GENWiki

Premier IT Outsourcing and Support Services within the UK

User Tools

Site Tools

Problem, Formatting or Query -  Send Feedback

Was this page helpful?-10+1


rfc:rfc7826

Internet Engineering Task Force (IETF) H. Schulzrinne Request for Comments: 7826 Columbia University Obsoletes: 2326 A. Rao Category: Standards Track Cisco ISSN: 2070-1721 R. Lanphier

                                                         M. Westerlund
                                                              Ericsson
                                                   M. Stiemerling, Ed.
                              University of Applied Sciences Darmstadt
                                                         December 2016
              Real-Time Streaming Protocol Version 2.0

Abstract

 This memorandum defines the Real-Time Streaming Protocol (RTSP)
 version 2.0, which obsoletes RTSP version 1.0 defined in RFC 2326.
 RTSP is an application-layer protocol for the setup and control of
 the delivery of data with real-time properties.  RTSP provides an
 extensible framework to enable controlled, on-demand delivery of
 real-time data, such as audio and video.  Sources of data can include
 both live data feeds and stored clips.  This protocol is intended to
 control multiple data delivery sessions; provide a means for choosing
 delivery channels such as UDP, multicast UDP, and TCP; and provide a
 means for choosing delivery mechanisms based upon RTP (RFC 3550).

Status of This Memo

 This is an Internet Standards Track document.
 This document is a product of the Internet Engineering Task Force
 (IETF).  It represents the consensus of the IETF community.  It has
 received public review and has been approved for publication by the
 Internet Engineering Steering Group (IESG).  Further information on
 Internet Standards is available in Section 2 of RFC 7841.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 http://www.rfc-editor.org/info/rfc7826.

Schulzrinne, et al. Standards Track [Page 1] RFC 7826 RTSP 2.0 December 2016

Copyright Notice

 Copyright (c) 2016 IETF Trust and the persons identified as the
 document authors.  All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (http://trustee.ietf.org/license-info) in effect on the date of
 publication of this document.  Please review these documents
 carefully, as they describe your rights and restrictions with respect
 to this document.  Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of
 the Trust Legal Provisions and are provided without warranty as
 described in the Simplified BSD License.
 This document may contain material from IETF Documents or IETF
 Contributions published or made publicly available before November
 10, 2008.  The person(s) controlling the copyright in some of this
 material may not have granted the IETF Trust the right to allow
 modifications of such material outside the IETF Standards Process.
 Without obtaining an adequate license from the person(s) controlling
 the copyright in such materials, this document may not be modified
 outside the IETF Standards Process, and derivative works of it may
 not be created outside the IETF Standards Process, except to format
 it for publication as an RFC or to translate it into languages other
 than English.

Table of Contents

 1. Introduction ...................................................10
 2. Protocol Overview ..............................................11
    2.1. Presentation Description ..................................12
    2.2. Session Establishment .....................................12
    2.3. Media Delivery Control ....................................14
    2.4. Session Parameter Manipulations ...........................15
    2.5. Media Delivery ............................................16
         2.5.1. Media Delivery Manipulations .......................16
    2.6. Session Maintenance and Termination .......................19
    2.7. Extending RTSP ............................................20
 3. Document Conventions ...........................................21
    3.1. Notational Conventions ....................................21
    3.2. Terminology ...............................................21
 4. Protocol Parameters ............................................25
    4.1. RTSP Version ..............................................25
    4.2. RTSP IRI and URI ..........................................25
    4.3. Session Identifiers .......................................28

Schulzrinne, et al. Standards Track [Page 2] RFC 7826 RTSP 2.0 December 2016

    4.4. Media-Time Formats ........................................28
         4.4.1. SMPTE-Relative Timestamps ..........................28
         4.4.2. Normal Play Time ...................................29
         4.4.3. Absolute Time ......................................30
    4.5. Feature Tags ..............................................31
    4.6. Message Body Tags .........................................32
    4.7. Media Properties ..........................................32
         4.7.1. Random Access and Seeking ..........................33
         4.7.2. Retention ..........................................34
         4.7.3. Content Modifications ..............................34
         4.7.4. Supported Scale Factors ............................34
         4.7.5. Mapping to the Attributes ..........................35
 5. RTSP Message ...................................................35
    5.1. Message Types .............................................36
    5.2. Message Headers ...........................................36
    5.3. Message Body ..............................................37
    5.4. Message Length ............................................37
 6. General-Header Fields ..........................................37
 7. Request ........................................................39
    7.1. Request Line ..............................................40
    7.2. Request-Header Fields .....................................42
 8. Response .......................................................43
    8.1. Status-Line ...............................................43
         8.1.1. Status Code and Reason Phrase ......................43
    8.2. Response Headers ..........................................47
 9. Message Body ...................................................47
    9.1. Message Body Header Fields ................................48
    9.2. Message Body ..............................................49
    9.3. Message Body Format Negotiation ...........................49
 10. Connections ...................................................50
    10.1. Reliability and Acknowledgements .........................50
    10.2. Using Connections ........................................51
    10.3. Closing Connections ......................................54
    10.4. Timing Out Connections and RTSP Messages .................56
    10.5. Showing Liveness .........................................57
    10.6. Use of IPv6 ..............................................58
    10.7. Overload Control .........................................58
 11. Capability Handling ...........................................60
    11.1. Feature Tag: play.basic ..................................62
 12. Pipelining Support ............................................62
 13. Method Definitions ............................................63
    13.1. OPTIONS ..................................................65
    13.2. DESCRIBE .................................................66
    13.3. SETUP ....................................................68
         13.3.1. Changing Transport Parameters .....................71
    13.4. PLAY .....................................................72
         13.4.1. General Usage .....................................72
         13.4.2. Aggregated Sessions ...............................77

Schulzrinne, et al. Standards Track [Page 3] RFC 7826 RTSP 2.0 December 2016

         13.4.3. Updating Current PLAY Requests ....................78
         13.4.4. Playing On-Demand Media ...........................81
         13.4.5. Playing Dynamic On-Demand Media ...................81
         13.4.6. Playing Live Media ................................81
         13.4.7. Playing Live with Recording .......................82
         13.4.8. Playing Live with Time-Shift ......................83
    13.5. PLAY_NOTIFY ..............................................83
         13.5.1. End-of-Stream .....................................84
         13.5.2. Media-Properties-Update ...........................86
         13.5.3. Scale-Change ......................................87
    13.6. PAUSE ....................................................89
    13.7. TEARDOWN .................................................92
         13.7.1. Client to Server ..................................92
         13.7.2. Server to Client ..................................93
    13.8. GET_PARAMETER ............................................94
    13.9. SET_PARAMETER ............................................96
    13.10. REDIRECT ................................................98
 14. Embedded (Interleaved) Binary Data ...........................101
 15. Proxies ......................................................103
    15.1. Proxies and Protocol Extensions .........................104
    15.2. Multiplexing and Demultiplexing of Messages .............105
 16. Caching ......................................................106
    16.1. Validation Model ........................................107
         16.1.1. Last-Modified Dates ..............................108
         16.1.2. Message Body Tag Cache Validators ................108
         16.1.3. Weak and Strong Validators .......................108
         16.1.4. Rules for When to Use Message Body Tags
                 and Last-Modified Dates ..........................110
         16.1.5. Non-validating Conditionals ......................112
    16.2. Invalidation after Updates or Deletions .................112
 17. Status Code Definitions ......................................113
    17.1. Informational 1xx .......................................113
         17.1.1. 100 Continue .....................................113
    17.2. Success 2xx .............................................113
         17.2.1. 200 OK ...........................................113
    17.3. Redirection 3xx .........................................113
         17.3.1. 300 ..............................................114
         17.3.2. 301 Moved Permanently ............................114
         17.3.3. 302 Found ........................................114
         17.3.4. 303 See Other ....................................115
         17.3.5. 304 Not Modified .................................115
         17.3.6. 305 Use Proxy ....................................115
    17.4. Client Error 4xx ........................................116
         17.4.1. 400 Bad Request ..................................116
         17.4.2. 401 Unauthorized .................................116
         17.4.3. 402 Payment Required .............................116
         17.4.4. 403 Forbidden ....................................116
         17.4.5. 404 Not Found ....................................116

Schulzrinne, et al. Standards Track [Page 4] RFC 7826 RTSP 2.0 December 2016

         17.4.6. 405 Method Not Allowed ...........................117
         17.4.7. 406 Not Acceptable ...............................117
         17.4.8. 407 Proxy Authentication Required ................117
         17.4.9. 408 Request Timeout ..............................117
         17.4.10. 410 Gone ........................................118
         17.4.11. 412 Precondition Failed .........................118
         17.4.12. 413 Request Message Body Too Large ..............118
         17.4.13. 414 Request-URI Too Long ........................118
         17.4.14. 415 Unsupported Media Type ......................119
         17.4.15. 451 Parameter Not Understood ....................119
         17.4.16. 452 Illegal Conference Identifier ...............119
         17.4.17. 453 Not Enough Bandwidth ........................119
         17.4.18. 454 Session Not Found ...........................119
         17.4.19. 455 Method Not Valid in This State ..............119
         17.4.20. 456 Header Field Not Valid for Resource .........119
         17.4.21. 457 Invalid Range ...............................120
         17.4.22. 458 Parameter Is Read-Only ......................120
         17.4.23. 459 Aggregate Operation Not Allowed .............120
         17.4.24. 460 Only Aggregate Operation Allowed ............120
         17.4.25. 461 Unsupported Transport .......................120
         17.4.26. 462 Destination Unreachable .....................120
         17.4.27. 463 Destination Prohibited ......................120
         17.4.28. 464 Data Transport Not Ready Yet ................121
         17.4.29. 465 Notification Reason Unknown .................121
         17.4.30. 466 Key Management Error ........................121
         17.4.31. 470 Connection Authorization Required ...........121
         17.4.32. 471 Connection Credentials Not Accepted .........121
         17.4.33. 472 Failure to Establish Secure Connection ......121
    17.5. Server Error 5xx ........................................122
         17.5.1. 500 Internal Server Error ........................122
         17.5.2. 501 Not Implemented ..............................122
         17.5.3. 502 Bad Gateway ..................................122
         17.5.4. 503 Service Unavailable ..........................122
         17.5.5. 504 Gateway Timeout ..............................123
         17.5.6. 505 RTSP Version Not Supported ...................123
         17.5.7. 551 Option Not Supported .........................123
         17.5.8. 553 Proxy Unavailable ............................123
 18. Header Field Definitions .....................................124
    18.1. Accept ..................................................134
    18.2. Accept-Credentials ......................................135
    18.3. Accept-Encoding .........................................135
    18.4. Accept-Language .........................................136
    18.5. Accept-Ranges ...........................................137
    18.6. Allow ...................................................138
    18.7. Authentication-Info .....................................138
    18.8. Authorization ...........................................138
    18.9. Bandwidth ...............................................139
    18.10. Blocksize ..............................................140

Schulzrinne, et al. Standards Track [Page 5] RFC 7826 RTSP 2.0 December 2016

    18.11. Cache-Control ..........................................140
    18.12. Connection .............................................143
    18.13. Connection-Credentials .................................143
    18.14. Content-Base ...........................................144
    18.15. Content-Encoding .......................................145
    18.16. Content-Language .......................................145
    18.17. Content-Length .........................................146
    18.18. Content-Location .......................................146
    18.19. Content-Type ...........................................148
    18.20. CSeq ...................................................148
    18.21. Date ...................................................150
    18.22. Expires ................................................151
    18.23. From ...................................................151
    18.24. If-Match ...............................................152
    18.25. If-Modified-Since ......................................152
    18.26. If-None-Match ..........................................153
    18.27. Last-Modified ..........................................154
    18.28. Location ...............................................154
    18.29. Media-Properties .......................................154
    18.30. Media-Range ............................................156
    18.31. MTag ...................................................157
    18.32. Notify-Reason ..........................................158
    18.33. Pipelined-Requests .....................................158
    18.34. Proxy-Authenticate .....................................159
    18.35. Proxy-Authentication-Info ..............................159
    18.36. Proxy-Authorization ....................................159
    18.37. Proxy-Require ..........................................160
    18.38. Proxy-Supported ........................................160
    18.39. Public .................................................161
    18.40. Range ..................................................162
    18.41. Referrer ...............................................164
    18.42. Request-Status .........................................164
    18.43. Require ................................................165
    18.44. Retry-After ............................................166
    18.45. RTP-Info ...............................................167
    18.46. Scale ..................................................169
    18.47. Seek-Style .............................................170
    18.48. Server .................................................171
    18.49. Session ................................................172
    18.50. Speed ..................................................173
    18.51. Supported ..............................................174
    18.52. Terminate-Reason .......................................175
    18.53. Timestamp ..............................................175
    18.54. Transport ..............................................176
    18.55. Unsupported ............................................183
    18.56. User-Agent .............................................184
    18.57. Via ....................................................184
    18.58. WWW-Authenticate .......................................185

Schulzrinne, et al. Standards Track [Page 6] RFC 7826 RTSP 2.0 December 2016

 19. Security Framework ...........................................185
    19.1. RTSP and HTTP Authentication ............................185
         19.1.1. Digest Authentication ............................186
    19.2. RTSP over TLS ...........................................187
    19.3. Security and Proxies ....................................188
         19.3.1. Accept-Credentials ...............................189
         19.3.2. User-Approved TLS Procedure ......................190
 20. Syntax .......................................................192
    20.1. Base Syntax .............................................193
    20.2. RTSP Protocol Definition ................................195
         20.2.1. Generic Protocol Elements ........................195
         20.2.2. Message Syntax ...................................198
         20.2.3. Header Syntax ....................................201
    20.3. SDP Extension Syntax ....................................209
 21. Security Considerations ......................................209
    21.1. Signaling Protocol Threats ..............................210
    21.2. Media Stream Delivery Threats ...........................213
         21.2.1. Remote DoS Attack ................................215
         21.2.2. RTP Security Analysis ............................216
 22. IANA Considerations ..........................................217
    22.1. Feature Tags ............................................218
         22.1.1. Description ......................................218
         22.1.2. Registering New Feature Tags with IANA ...........218
         22.1.3. Registered Entries ...............................219
    22.2. RTSP Methods ............................................219
         22.2.1. Description ......................................219
         22.2.2. Registering New Methods with IANA ................219
         22.2.3. Registered Entries ...............................220
    22.3. RTSP Status Codes .......................................220
         22.3.1. Description ......................................220
         22.3.2. Registering New Status Codes with IANA ...........220
         22.3.3. Registered Entries ...............................221
    22.4. RTSP Headers ............................................221
         22.4.1. Description ......................................221
         22.4.2. Registering New Headers with IANA ................221
         22.4.3. Registered Entries ...............................222
    22.5. Accept-Credentials ......................................223
         22.5.1. Accept-Credentials Policies ......................223
         22.5.2. Accept-Credentials Hash Algorithms ...............224
    22.6. Cache-Control Cache Directive Extensions ................224
    22.7. Media Properties ........................................225
         22.7.1. Description ......................................225
         22.7.2. Registration Rules ...............................226
         22.7.3. Registered Values ................................226
    22.8. Notify-Reason Values ....................................226
         22.8.1. Description ......................................226
         22.8.2. Registration Rules ...............................226
         22.8.3. Registered Values ................................227

Schulzrinne, et al. Standards Track [Page 7] RFC 7826 RTSP 2.0 December 2016

    22.9. Range Header Formats ....................................227
         22.9.1. Description ......................................227
         22.9.2. Registration Rules ...............................227
         22.9.3. Registered Values ................................228
    22.10. Terminate-Reason Header ................................228
         22.10.1. Redirect Reasons ................................228
         22.10.2. Terminate-Reason Header Parameters ..............229
    22.11. RTP-Info Header Parameters .............................229
         22.11.1. Description .....................................229
         22.11.2. Registration Rules ..............................229
         22.11.3. Registered Values ...............................230
    22.12. Seek-Style Policies ....................................230
         22.12.1. Description .....................................230
         22.12.2. Registration Rules ..............................230
         22.12.3. Registered Values ...............................230
    22.13. Transport Header Registries ............................231
         22.13.1. Transport Protocol Identifier ...................231
         22.13.2. Transport Modes .................................233
         22.13.3. Transport Parameters ............................233
    22.14. URI Schemes ............................................234
         22.14.1. The "rtsp" URI Scheme ...........................234
         22.14.2. The "rtsps" URI Scheme ..........................235
         22.14.3. The "rtspu" URI Scheme ..........................237
    22.15. SDP Attributes .........................................238
    22.16. Media Type Registration for text/parameters ............238
 23. References ...................................................240
    23.1. Normative References ....................................240
    23.2. Informative References ..................................245
 Appendix A. Examples .............................................248
    A.1. Media on Demand (Unicast) ................................248
    A.2. Media on Demand Using Pipelining .........................251
    A.3. Secured Media Session for On-Demand Content ..............254
    A.4. Media on Demand (Unicast) ................................257
    A.5. Single-Stream Container Files ............................260
    A.6. Live Media Presentation Using Multicast ..................263
    A.7. Capability Negotiation ...................................264
 Appendix B. RTSP Protocol State Machine ..........................265
    B.1. States ...................................................266
    B.2. State Variables ..........................................266
    B.3. Abbreviations ............................................266
    B.4. State Tables .............................................267
 Appendix C. Media-Transport Alternatives .........................272
    C.1. RTP ......................................................272
      C.1.1. AVP ..................................................272
      C.1.2. AVP/UDP ..............................................273
      C.1.3. AVPF/UDP .............................................274
      C.1.4. SAVP/UDP .............................................275
      C.1.5. SAVPF/UDP ............................................277

Schulzrinne, et al. Standards Track [Page 8] RFC 7826 RTSP 2.0 December 2016

      C.1.6. RTCP Usage with RTSP .................................278
    C.2. RTP over TCP .............................................279
      C.2.1. Interleaved RTP over TCP .............................280
      C.2.2. RTP over Independent TCP .............................280
    C.3. Handling Media-Clock Time Jumps in the RTP Media Layer ...284
    C.4. Handling RTP Timestamps after PAUSE ......................287
    C.5. RTSP/RTP Integration  ....................................290
    C.6. Scaling with RTP .........................................290
    C.7. Maintaining NPT Synchronization with RTP Timestamps ......290
    C.8. Continuous Audio .........................................290
    C.9. Multiple Sources in an RTP Session .......................290
    C.10. Usage of SSRCs and the RTCP BYE Message during an RTSP
          Session .................................................290
    C.11. Future Additions ........................................291
 Appendix D. Use of SDP for RTSP Session Descriptions .............292
    D.1. Definitions  .............................................292
      D.1.1. Control URI ..........................................292
      D.1.2. Media Streams ........................................294
      D.1.3. Payload Type(s) ......................................294
      D.1.4. Format-Specific Parameters ...........................294
      D.1.5. Directionality of Media Stream .......................295
      D.1.6. Range of Presentation ................................295
      D.1.7. Time of Availability .................................296
      D.1.8. Connection Information ...............................297
      D.1.9. Message Body Tag .....................................297
    D.2. Aggregate Control Not Available ..........................298
    D.3. Aggregate Control Available ..............................298
    D.4. Grouping of Media Lines in SDP ...........................299
    D.5. RTSP External SDP Delivery ...............................300
 Appendix E. RTSP Use Cases .......................................300
    E.1. On-Demand Playback of Stored Content .....................300
    E.2. Unicast Distribution of Live Content .....................302
    E.3. On-Demand Playback Using Multicast .......................303
    E.4. Inviting an RTSP Server into a Conference ................303
    E.5. Live Content Using Multicast .............................304
 Appendix F. Text Format for Parameters ...........................305
 Appendix G. Requirements for Unreliable Transport of RTSP ........305
 Appendix H. Backwards-Compatibility Considerations ...............306
    H.1. Play Request in Play State ...............................307
    H.2. Using Persistent Connections .............................307
 Appendix I. Changes ..............................................307
    I.1. Brief Overview ...........................................308
    I.2. Detailed List of Changes .................................309
 Acknowledgements .................................................316
 Contributors  ....................................................317
 Authors' Addresses ...............................................318

Schulzrinne, et al. Standards Track [Page 9] RFC 7826 RTSP 2.0 December 2016

1. Introduction

 This memo defines version 2.0 of the Real-Time Streaming Protocol
 (RTSP 2.0).  RTSP 2.0 is an application-layer protocol for the setup
 and control over the delivery of data with real-time properties,
 typically streaming media.  Streaming media is, for instance, video
 on demand or audio live streaming.  Put simply, RTSP acts as a
 "network remote control" for multimedia servers.
 The protocol operates between RTSP 2.0 clients and servers, but it
 also supports the use of proxies placed between clients and servers.
 Clients can request information about streaming media from servers by
 asking for a description of the media or use media description
 provided externally.  The media delivery protocol is used to
 establish the media streams described by the media description.
 Clients can then request to play out the media, pause it, or stop it
 completely.  The requested media can consist of multiple audio and
 video streams that are delivered as time-synchronized streams from
 servers to clients.
 RTSP 2.0 is a replacement of RTSP 1.0 [RFC2326] and this document
 obsoletes that specification.  This protocol is based on RTSP 1.0 but
 is not backwards compatible other than in the basic version
 negotiation mechanism.  The changes between the two documents are
 listed in Appendix I.  There are many reasons why RTSP 2.0 can't be
 backwards compatible with RTSP 1.0; some of the main ones are as
 follows:
 o  Most headers that needed to be extensible did not define the
    allowed syntax, preventing safe deployment of extensions;
 o  the changed behavior of the PLAY method when received in Play
    state;
 o  the changed behavior of the extensibility model and its mechanism;
    and
 o  the change of syntax for some headers.
 There are so many small updates that changing versions became
 necessary to enable clarification and consistent behavior.  Anyone
 implementing RTSP for a new use case in which they have not installed
 RTSP 1.0 should only implement RTSP 2.0 to avoid having to deal with
 RTSP 1.0 inconsistencies.
 This document is structured as follows.  It begins with an overview
 of the protocol operations and its functions in an informal way.
 Then, a set of definitions of terms used and document conventions is

Schulzrinne, et al. Standards Track [Page 10] RFC 7826 RTSP 2.0 December 2016

 introduced.  These are followed by the actual RTSP 2.0 core protocol
 specification.  The appendices describe and define some
 functionalities that are not part of the core RTSP specification, but
 which are still important to enable some usages.  Among them, the RTP
 usage is defined in Appendix C, the Session Description Protocol
 (SDP) usage with RTSP is defined in Appendix D, and the "text/
 parameters" file format Appendix F, are three normative specification
 appendices.  Other appendices include a number of informational parts
 discussing the changes, use cases, different considerations or
 motivations.

2. Protocol Overview

 This section provides an informative overview of the different
 mechanisms in the RTSP 2.0 protocol to give the reader a high-level
 understanding before getting into all the specific details.  In case
 of conflict with this description and the later sections, the later
 sections take precedence.  For more information about use cases
 considered for RTSP, see Appendix E.
 RTSP 2.0 is a bidirectional request and response protocol that first
 establishes a context including content resources (the media) and
 then controls the delivery of these content resources from the
 provider to the consumer.  RTSP has three fundamental parts: Session
 Establishment, Media Delivery Control, and an extensibility model
 described below.  The protocol is based on some assumptions about
 existing functionality to provide a complete solution for client-
 controlled real-time media delivery.
 RTSP uses text-based messages, requests and responses, that may
 contain a binary message body.  An RTSP request starts with a method
 line that identifies the method, the protocol, and version and the
 resource on which to act.  The resource is identified by a URI and
 the hostname part of the URI is used by RTSP client to resolve the
 IPv4 or IPv6 address of the RTSP server.  Following the method line
 are a number of RTSP headers.  These lines are ended by two
 consecutive carriage return line feed (CRLF) character pairs.  The
 message body, if present, follows the two CRLF character pairs, and
 the body's length is described by a message header.  RTSP responses
 are similar, but they start with a response line with the protocol
 and version followed by a status code and a reason phrase.  RTSP
 messages are sent over a reliable transport protocol between the
 client and server.  RTSP 2.0 requires clients and servers to
 implement TCP and TLS over TCP as mandatory transports for RTSP
 messages.

Schulzrinne, et al. Standards Track [Page 11] RFC 7826 RTSP 2.0 December 2016

2.1. Presentation Description

 RTSP exists to provide access to multimedia presentations and content
 but tries to be agnostic about the media type or the actual media
 delivery protocol that is used.  To enable a client to implement a
 complete system, an RTSP-external mechanism for describing the
 presentation and the delivery protocol(s) is used.  RTSP assumes that
 this description is either delivered completely out of band or as a
 data object in the response to a client's request using the DESCRIBE
 method (Section 13.2).
 Parameters that commonly have to be included in the presentation
 description are the following:
 o  The number of media streams;
 o  the resource identifier for each media stream/resource that is to
    be controlled by RTSP;
 o  the protocol that will be used to deliver each media stream;
 o  the transport protocol parameters that are not negotiated or vary
    with each client;
 o  the media-encoding information enabling a client to correctly
    decode the media upon reception; and
 o  an aggregate control resource identifier.
 RTSP uses its own URI schemes ("rtsp" and "rtsps") to reference media
 resources and aggregates under common control (see Section 4.2).
 This specification describes in Appendix D how one uses SDP [RFC4566]
 for describing the presentation.

2.2. Session Establishment

 The RTSP client can request the establishment of an RTSP session
 after having used the presentation description to determine which
 media streams are available, which media delivery protocol is used,
 and the resource identifiers of the media streams.  The RTSP session
 is a common context between the client and the server that consists
 of one or more media resources that are to be under common media
 delivery control.
 The client creates an RTSP session by sending a request using the
 SETUP method (Section 13.3) to the server.  In the Transport header
 (Section 18.54) of the SETUP request, the client also includes all

Schulzrinne, et al. Standards Track [Page 12] RFC 7826 RTSP 2.0 December 2016

 the transport parameters necessary to enable the media delivery
 protocol to function.  This includes parameters that are
 preestablished by the presentation description but necessary for any
 middlebox to correctly handle the media delivery protocols.  The
 Transport header in a request may contain multiple alternatives for
 media delivery in a prioritized list, which the server can select
 from.  These alternatives are typically based on information in the
 presentation description.
 When receiving a SETUP request, the server determines if the media
 resource is available and if one or more of the of the transport
 parameter specifications are acceptable.  If that is successful, an
 RTSP session context is created and the relevant parameters and state
 is stored.  An identifier is created for the RTSP session and
 included in the response in the Session header (Section 18.49).  The
 SETUP response includes a Transport header that specifies which of
 the alternatives has been selected and relevant parameters.
 A SETUP request that references an existing RTSP session but
 identifies a new media resource is a request to add that media
 resource under common control with the already-present media
 resources in an aggregated session.  A client can expect this to work
 for all media resources under RTSP control within a multimedia
 content container.  However, a server will likely refuse to aggregate
 resources from different content containers.  Even if an RTSP session
 contains only a single media stream, the RTSP session can be
 referenced by the aggregate control URI.
 To avoid an extra round trip in the session establishment of
 aggregated RTSP sessions, RTSP 2.0 supports pipelined requests; i.e.,
 the client can send multiple requests back-to-back without waiting
 first for the completion of any of them.  The client uses a client-
 selected identifier in the Pipelined-Requests header (Section 18.33)
 to instruct the server to bind multiple requests together as if they
 included the session identifier.
 The SETUP response also provides additional information about the
 established sessions in a couple of different headers.  The Media-
 Properties header (Section 18.29) includes a number of properties
 that apply for the aggregate that is valuable when doing media
 delivery control and configuring user interface.  The Accept-Ranges
 header (Section 18.5) informs the client about range formats that the
 server supports for these media resources.  The Media-Range header
 (Section 18.30) informs the client about the time range of the media
 currently available.

Schulzrinne, et al. Standards Track [Page 13] RFC 7826 RTSP 2.0 December 2016

2.3. Media Delivery Control

 After having established an RTSP session, the client can start
 controlling the media delivery.  The basic operations are "begin
 playback", using the PLAY method (Section 13.4) and "suspend (pause)
 playback" by using the PAUSE method (Section 13.6).  PLAY also allows
 for choosing the starting media position from which the server should
 deliver the media.  The positioning is done by using the Range header
 (Section 18.40) that supports several different time formats: Normal
 Play Time (NPT) (Section 4.4.2), Society of Motion Picture and
 Television Engineers (SMPTE) Timestamps (Section 4.4.1), and absolute
 time (Section 4.4.3).  The Range header also allows the client to
 specify a position where delivery should end, thus allowing a
 specific interval to be delivered.
 The support for positioning/searching within media content depends on
 the content's media properties.  Content exists in a number of
 different types, such as on-demand, live, and live with simultaneous
 recording.  Even within these categories, there are differences in
 how the content is generated and distributed, which affect how it can
 be accessed for playback.  The properties applicable for the RTSP
 session are provided by the server in the SETUP response using the
 Media-Properties header (Section 18.29).  These are expressed using
 one or several independent attributes.  A first attribute is Random-
 Access, which indicates whether positioning is possible, and with
 what granularity.  Another aspect is whether the content will change
 during the lifetime of the session.  While on-demand content will be
 provided in full from the beginning, a live stream being recorded
 results in the length of the accessible content growing as the
 session goes on.  There also exists content that is dynamically built
 by a protocol other than RTSP and, thus, also changes in steps during
 the session, but maybe not continuously.  Furthermore, when content
 is recorded, there are cases where the complete content is not
 maintained, but, for example, only the last hour.  All of these
 properties result in the need for mechanisms that will be discussed
 below.
 When the client accesses on-demand content that allows random access,
 the client can issue the PLAY request for any point in the content
 between the start and the end.  The server will deliver media from
 the closest random access point prior to the requested point and
 indicate that in its PLAY response.  If the client issues a PAUSE,
 the delivery will be halted and the point at which the server stopped
 will be reported back in the response.  The client can later resume
 by sending a PLAY request without a Range header.  When the server is
 about to complete the PLAY request by delivering the end of the
 content or the requested range, the server will send a PLAY_NOTIFY
 request (Section 13.5) indicating this.

Schulzrinne, et al. Standards Track [Page 14] RFC 7826 RTSP 2.0 December 2016

 When playing live content with no extra functions, such as recording,
 the client will receive the live media from the server after having
 sent a PLAY request.  Seeking in such content is not possible as the
 server does not store it, but only forwards it from the source of the
 session.  Thus, delivery continues until the client sends a PAUSE
 request, tears down the session, or the content ends.
 For live sessions that are being recorded, the client will need to
 keep track of how the recording progresses.  Upon session
 establishment, the client will learn the current duration of the
 recording from the Media-Range header.  Because the recording is
 ongoing, the content grows in direct relation to the time passed.
 Therefore, each server's response to a PLAY request will contain the
 current Media-Range header.  The server should also regularly send
 (approximately every 5 minutes) the current media range in a
 PLAY_NOTIFY request (Section 13.5.2).  If the live transmission ends,
 the server must send a PLAY_NOTIFY request with the updated Media-
 Properties indicating that the content stopped being a recorded live
 session and instead became on-demand content; the request also
 contains the final media range.  While the live delivery continues,
 the client can request to play the current live point by using the
 NPT timescale symbol "now", or it can request a specific point in the
 available content by an explicit range request for that point.  If
 the requested point is outside of the available interval, the server
 will adjust the position to the closest available point, i.e., either
 at the beginning or the end.
 A special case of recording is that where the recording is not
 retained longer than a specific time period; thus, as the live
 delivery continues, the client can access any media within a moving
 window that covers, for example, "now" to "now" minus 1 hour.  A
 client that pauses on a specific point within the content may not be
 able to retrieve the content anymore.  If the client waits too long
 before resuming the pause point, the content may no longer be
 available.  In this case, the pause point will be adjusted to the
 closest point in the available media.

2.4. Session Parameter Manipulations

 A session may have additional state or functionality that affects how
 the server or client treats the session or content, how it functions,
 or feedback on how well the session works.  Such extensions are not
 defined in this specification, but they may be covered in various
 extensions.  RTSP has two methods for retrieving and setting
 parameter values on either the client or the server: GET_PARAMETER
 (Section 13.8) and SET_PARAMETER (Section 13.9).  These methods carry
 the parameters in a message body of the appropriate format.  One can
 also use headers to query state with the GET_PARAMETER method.  As an

Schulzrinne, et al. Standards Track [Page 15] RFC 7826 RTSP 2.0 December 2016

 example, clients needing to know the current media range for a time-
 progressing session can use the GET_PARAMETER method and include the
 media range.  Furthermore, synchronization information can be
 requested by using a combination of RTP-Info (Section 18.45) and
 Range (Section 18.40).
 RTSP 2.0 does not have a strong mechanism for negotiating the headers
 or parameters and their formats.  However, responses will indicate
 request-headers or parameters that are not supported.  A priori
 determination of what features are available needs to be done through
 out-of-band mechanisms, like the session description, or through the
 usage of feature tags (Section 4.5).

2.5. Media Delivery

 This document specifies how media is delivered with RTP [RFC3550]
 over UDP [RFC768], TCP [RFC793], or the RTSP connection.  Additional
 protocols may be specified in the future as needed.
 The usage of RTP as a media delivery protocol requires some
 additional information to function well.  The PLAY response contains
 information to enable reliable and timely delivery of how a client
 should synchronize different sources in the different RTP sessions.
 It also provides a mapping between RTP timestamps and the content-
 time scale.  When the server wants to notify the client about the
 completion of the media delivery, it sends a PLAY_NOTIFY request to
 the client.  The PLAY_NOTIFY request includes information about the
 stream end, including the last RTP sequence number for each stream,
 thus enabling the client to empty the buffer smoothly.

2.5.1. Media Delivery Manipulations

 The basic playback functionality of RTSP enables delivery of a range
 of requested content to the client at the pace intended by the
 content's creator.  However, RTSP can also manipulate the delivery to
 the client in two ways.
 Scale:  The ratio of media-content time delivered per unit of
    playback time.
 Speed:  The ratio of playback time delivered per unit of wallclock
    time.
 Both affect the media delivery per time unit.  However, they
 manipulate two independent timescales and the effects are possible to
 combine.

Schulzrinne, et al. Standards Track [Page 16] RFC 7826 RTSP 2.0 December 2016

 Scale (Section 18.46) is used for fast-forward or slow-motion control
 as it changes the amount of content timescale that should be played
 back per time unit.  Scale > 1.0, means fast forward, e.g., scale =
 2.0 results in that 2 seconds of content being played back every
 second of playback.  Scale = 1.0 is the default value that is used if
 no scale is specified, i.e., playback at the content's original rate.
 Scale values between 0 and 1.0 provide for slow motion.  Scale can be
 negative to allow for reverse playback in either regular pace
 (scale = -1.0), fast backwards (scale < -1.0), or slow-motion
 backwards (-1.0 < scale < 0).  Scale = 0 would be equal to pause and
 is not allowed.
 In most cases, the realization of scale means server-side
 manipulation of the media to ensure that the client can actually play
 it back.  The nature of these media manipulations and when they are
 needed is highly media-type dependent.  Let's consider two common
 media types, audio and video.
 It is very difficult to modify the playback rate of audio.
 Typically, no more than a factor of two is possible while maintaining
 intelligibility by changing the pitch and rate of speech.  Music goes
 out of tune if one tries to manipulate the playback rate by
 resampling it.  This is a well-known problem, and audio is commonly
 muted or played back in short segments with skips to keep up with the
 current playback point.
 For video, it is possible to manipulate the frame rate, although the
 rendering capabilities are often limited to certain frame rates.
 Also, the allowed bitrates in decoding, the structure used in the
 encoding, and the dependency between frames and other capabilities of
 the rendering device limits the possible manipulations.  Therefore,
 the basic fast-forward capabilities often are implemented by
 selecting certain subsets of frames.
 Due to the media restrictions, the possible scale values are commonly
 restricted to the set of realizable scale ratios.  To enable the
 clients to select from the possible scale values, RTSP can signal the
 supported scale ratios for the content.  To support aggregated or
 dynamic content, where this may change during the ongoing session and
 dependent on the location within the content, a mechanism for
 updating the media properties and the scale factor currently in use,
 exists.
 Speed (Section 18.50) affects how much of the playback timeline is
 delivered in a given wallclock period.  The default is Speed = 1
 which means to deliver at the same rate the media is consumed.
 Speed > 1 means that the receiver will get content faster than it
 regularly would consume it.  Speed < 1 means that delivery is slower

Schulzrinne, et al. Standards Track [Page 17] RFC 7826 RTSP 2.0 December 2016

 than the regular media rate.  Speed values of 0 or lower have no
 meaning and are not allowed.  This mechanism enables two general
 functionalities.  One is client-side scale operations, i.e., the
 client receives all the frames and makes the adjustment to the
 playback locally.  The second is delivery control for the buffering
 of media.  By specifying a speed over 1.0, the client can build up
 the amount of playback time it has present in its buffers to a level
 that is sufficient for its needs.
 A naive implementation of Speed would only affect the transmission
 schedule of the media and has a clear impact on the needed bandwidth.
 This would result in the data rate being proportional to the speed
 factor.  Speed = 1.5, i.e., 50% faster than normal delivery, would
 result in a 50% increase in the data-transport rate.  Whether or not
 that can be supported depends solely on the underlying network path.
 Scale may also have some impact on the required bandwidth due to the
 manipulation of the content in the new playback schedule.  An example
 is fast forward where only the independently decodable intra-frames
 are included in the media stream.  This usage of solely intra-frames
 increases the data rate significantly compared to a normal sequence
 with the same number of frames, where most frames are encoded using
 prediction.
 This potential increase of the data rate needs to be handled by the
 media sender.  The client has requested that the media be delivered
 in a specific way, which should be honored.  However, the media
 sender cannot ignore if the network path between the sender and the
 receiver can't handle the resulting media stream.  In that case, the
 media stream needs to be adapted to fit the available resources of
 the path.  This can result in a reduced media quality.
 The need for bitrate adaptation becomes especially problematic in
 connection with the Speed semantics.  If the goal is to fill up the
 buffer, the client may not want to do that at the cost of reduced
 quality.  If the client wants to make local playout changes, then it
 may actually require that the requested speed be honored.  To resolve
 this issue, Speed uses a range so that both cases can be supported.
 The server is requested to use the highest possible speed value
 within the range, which is compatible with the available bandwidth.
 As long as the server can maintain a speed value within the range, it
 shall not change the media quality, but instead modify the actual
 delivery rate in response to available bandwidth and reflect this in
 the Speed value in the response.  However, if this is not possible,
 the server should instead modify the media quality to respect the
 lowest speed value and the available bandwidth.

Schulzrinne, et al. Standards Track [Page 18] RFC 7826 RTSP 2.0 December 2016

 This functionality enables the local scaling implementation to use a
 tight range, or even a range where the lower bound equals the upper
 bound, to identify that it requires the server to deliver the
 requested amount of media time per delivery time, independent of how
 much it needs to adapt the media quality to fit within the available
 path bandwidth.  For buffer filling, it is suitable to use a range
 with a reasonable span and with a lower bound at the nominal media
 rate 1.0, such as 1.0 - 2.5.  If the client wants to reduce the
 buffer, it can specify an upper bound that is below 1.0 to force the
 server to deliver slower than the nominal media rate.

2.6. Session Maintenance and Termination

 The session context that has been established is kept alive by having
 the client show liveness.  This is done in two main ways:
 o  Media-transport protocol keep-alive.  RTP Control Protocol (RTCP)
    may be used when using RTP.
 o  Any RTSP request referencing the session context.
 Section 10.5 discusses the methods for showing liveness in more
 depth.  If the client fails to show liveness for more than the
 established session timeout value (normally 60 seconds), the server
 may terminate the context.  Other values may be selected by the
 server through the inclusion of the timeout parameter in the session
 header.
 The session context is normally terminated by the client sending a
 TEARDOWN request (Section 13.7) to the server referencing the
 aggregated control URI.  An individual media resource can be removed
 from a session context by a TEARDOWN request referencing that
 particular media resource.  If all media resources are removed from a
 session context, the session context is terminated.
 A client may keep the session alive indefinitely if allowed by the
 server; however, a client is advised to release the session context
 when an extended period of time without media delivery activity has
 passed.  The client can re-establish the session context if required
 later.  What constitutes an extended period of time is dependent on
 the client, server, and their usage.  It is recommended that the
 client terminate the session before ten times the session timeout
 value has passed.  A server may terminate the session after one
 session timeout period without any client activity beyond keep-alive.
 When a server terminates the session context, it does so by sending a
 TEARDOWN request indicating the reason.

Schulzrinne, et al. Standards Track [Page 19] RFC 7826 RTSP 2.0 December 2016

 A server can also request that the client tear down the session and
 re-establish it at an alternative server, as may be needed for
 maintenance.  This is done by using the REDIRECT method
 (Section 13.10).  The Terminate-Reason header (Section 18.52) is used
 to indicate when and why.  The Location header indicates where it
 should connect if there is an alternative server available.  When the
 deadline expires, the server simply stops providing the service.  To
 achieve a clean closure, the client needs to initiate session
 termination prior to the deadline.  In case the server has no other
 server to redirect to, and it wants to close the session for
 maintenance, it shall use the TEARDOWN method with a Terminate-Reason
 header.

2.7. Extending RTSP

 RTSP is quite a versatile protocol that supports extensions in many
 different directions.  Even this core specification contains several
 blocks of functionality that are optional to implement.  The use case
 and need for the protocol deployment should determine what parts are
 implemented.  Allowing for extensions makes it possible for RTSP to
 address additional use cases.  However, extensions will affect the
 interoperability of the protocol; therefore, it is important that
 they can be added in a structured way.
 The client can learn the capability of a server by using the OPTIONS
 method (Section 13.1) and the Supported header (Section 18.51).  It
 can also try and possibly fail using new methods or require that
 particular features be supported using the Require (Section 18.43) or
 Proxy-Require (Section 18.37) header.
 The RTSP, in itself, can be extended in three ways, listed here in
 increasing order of the magnitude of changes supported:
 o  Existing methods can be extended with new parameters, for example,
    headers, as long as these parameters can be safely ignored by the
    recipient.  If the client needs negative acknowledgment when a
    method extension is not supported, a tag corresponding to the
    extension may be added in the field of the Require or Proxy-
    Require headers.
 o  New methods can be added.  If the recipient of the message does
    not understand the request, it must respond with error code 501
    (Not Implemented) so that the sender can avoid using this method
    again.  A client may also use the OPTIONS method to inquire about
    methods supported by the server.  The server must list the methods
    it supports using the Public response-header.

Schulzrinne, et al. Standards Track [Page 20] RFC 7826 RTSP 2.0 December 2016

 o  A new version of the protocol can be defined, allowing almost all
    aspects (except the position of the protocol version number) to
    change.  A new version of the protocol must be registered through
    a Standards Track document.
 The basic capability discovery mechanism can be used to both discover
 support for a certain feature and to ensure that a feature is
 available when performing a request.  For a detailed explanation of
 this, see Section 11.
 New media delivery protocols may be added and negotiated at session
 establishment, in addition to extensions to the core protocol.
 Certain types of protocol manipulations can be done through parameter
 formats using SET_PARAMETER and GET_PARAMETER.

3. Document Conventions

3.1. Notational Conventions

 All the mechanisms specified in this document are described in both
 prose and the Augmented Backus-Naur form (ABNF) described in detail
 in [RFC5234].
 Indented paragraphs are used to provide informative background and
 motivation.  This is intended to give readers who were not involved
 with the formulation of the specification an understanding of why
 things are the way they are in RTSP.
 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
 "OPTIONAL" in this document are to be interpreted as described in
 [RFC2119].
 The word, "unspecified" is used to indicate functionality or features
 that are not defined in this specification.  Such functionality
 cannot be used in a standardized manner without further definition in
 an extension specification to RTSP.

3.2. Terminology

 Aggregate control:  The concept of controlling multiple streams using
    a single timeline, generally one maintained by the server.  A
    client, for example, uses aggregate control when it issues a
    single play or pause message to simultaneously control both the
    audio and video in a movie.  A session that is under aggregate
    control is referred to as an "aggregated session".

Schulzrinne, et al. Standards Track [Page 21] RFC 7826 RTSP 2.0 December 2016

 Aggregate control URI:  The URI used in an RTSP request to refer to
    and control an aggregated session.  It normally, but not always,
    corresponds to the presentation URI specified in the session
    description.  See Section 13.3 for more information.
 Client:  The client is the requester of media service from the media
    server.
 Connection:  A transport-layer virtual circuit established between
    two programs for the purpose of communication.
 Container file:  A file that may contain multiple media streams that
    often constitute a presentation when played together.  The concept
    of a container file is not embedded in the protocol.  However,
    RTSP servers may offer aggregate control on the media streams
    within these files.
 Continuous media:  Data where there is a timing relationship between
    source and sink; that is, the sink needs to reproduce the timing
    relationship that existed at the source.  The most common examples
    of continuous media are audio and motion video.  Continuous media
    can be real time (interactive or conversational), where there is a
    "tight" timing relationship between source and sink or it can be
    streaming where the relationship is less strict.
 Feature tag:  A tag representing a certain set of functionality,
    i.e., a feature.
 IRI:  An Internationalized Resource Identifier is similar to a URI
    but allows characters from the whole Universal Character Set
    (Unicode/ISO 10646), rather than the US-ASCII only.  See [RFC3987]
    for more information.
 Live:  A live presentation or session originates media from an event
    taking place at the same time as the media delivery.  Live
    sessions often have an unbound or only loosely defined duration
    and seek operations may not be possible.
 Media initialization:  The datatype- or codec-specific
    initialization.  This includes such things as clock rates, color
    tables, etc.  Any transport-independent information that is
    required by a client for playback of a media stream occurs in the
    media initialization phase of stream setup.
 Media parameter:  A parameter specific to a media type that may be
    changed before or during stream delivery.

Schulzrinne, et al. Standards Track [Page 22] RFC 7826 RTSP 2.0 December 2016

 Media server:  The server providing media-delivery services for one
    or more media streams.  Different media streams within a
    presentation may originate from different media servers.  A media
    server may reside on the same host or on a different host from
    which the presentation is invoked.
 (Media) Stream:  A single media instance, e.g., an audio stream or a
    video stream as well as a single whiteboard or shared application
    group.  When using RTP, a stream consists of all RTP and RTCP
    packets created by a media source within an RTP session.
 Message:  The basic unit of RTSP communication, consisting of a
    structured sequence of octets matching the syntax defined in
    Section 20 and transmitted over a transport between RTSP agents.
    A message is either a request or a response.
 Message body:  The information transferred as the payload of a
    message (request or response).  A message body consists of meta-
    information in the form of message body headers and content in the
    form of an arbitrary number of data octets, as described in
    Section 9.
 Non-aggregated control:  Control of a single media stream.
 Presentation:  A set of one or more streams presented to the client
    as a complete media feed and described by a presentation
    description as defined below.  Presentations with more than one
    media stream are often handled in RTSP under aggregate control.
 Presentation description:  A presentation description contains
    information about one or more media streams within a presentation,
    such as the set of encodings, network addresses, and information
    about the content.  Other IETF protocols, such as SDP ([RFC4566]),
    use the term "session" for a presentation.  The presentation
    description may take several different formats, including but not
    limited to SDP format.
 Response:  An RTSP response to a request.  One type of RTSP message.
    If an HTTP response is meant, it is indicated explicitly.
 Request:  An RTSP request.  One type of RTSP message.  If an HTTP
    request is meant, it is indicated explicitly.
 Request-URI:  The URI used in a request to indicate the resource on
    which the request is to be performed.

Schulzrinne, et al. Standards Track [Page 23] RFC 7826 RTSP 2.0 December 2016

 RTSP agent:  Either an RTSP client, an RTSP server, or an RTSP proxy.
    In this specification, there are many capabilities that are common
    to these three entities such as the capability to send requests or
    receive responses.  This term will be used when describing
    functionality that is applicable to all three of these entities.
 RTSP session:  A stateful abstraction upon which the main control
    methods of RTSP operate.  An RTSP session is a common context; it
    is created and maintained on a client's request and can be
    destroyed by either the client or server.  It is established by an
    RTSP server upon the completion of a successful SETUP request
    (when a 200 OK response is sent) and is labeled with a session
    identifier at that time.  The session exists until timed out by
    the server or explicitly removed by a TEARDOWN request.  An RTSP
    session is a stateful entity; an RTSP server maintains an explicit
    session state machine (see Appendix B) where most state
    transitions are triggered by client requests.  The existence of a
    session implies the existence of state about the session's media
    streams and their respective transport mechanisms.  A given
    session can have one or more media streams associated with it.  An
    RTSP server uses the session to aggregate control over multiple
    media streams.
 Origin server:  The server on which a given resource resides.
 Seeking:  Requesting playback from a particular point in the content
    time line.
 Transport initialization:  The negotiation of transport information
    (e.g., port numbers, transport protocols) between the client and
    the server.
 URI:  A Universal Resource Identifier; see [RFC3986].  The URIs used
    in RTSP are generally URLs as they give a location for the
    resource.  As URLs are a subset of URIs, they will be referred to
    as URIs to cover also the cases when an RTSP URI would not be a
    URL.
 URL:  A Universal Resource Locator is a URI that identifies the
    resource through its primary access mechanism rather than
    identifying the resource by name or by some other attribute(s) of
    that resource.

Schulzrinne, et al. Standards Track [Page 24] RFC 7826 RTSP 2.0 December 2016

4. Protocol Parameters

4.1. RTSP Version

 This specification defines version 2.0 of RTSP.
 RTSP uses a "<major>.<minor>" numbering scheme to indicate versions
 of the protocol.  The protocol versioning policy is intended to allow
 the sender to indicate the format of a message and its capacity for
 understanding further RTSP communication rather than the features
 obtained via that communication.  No change is made to the version
 number for the addition of message components that do not affect
 communication behavior or that only add to extensible field values.
 The <minor> number is incremented when the changes made to the
 protocol add features that do not change the general message parsing
 algorithm but that may add to the message semantics and imply
 additional capabilities of the sender.  The <major> number is
 incremented when the format of a message within the protocol is
 changed.  The version of an RTSP message is indicated by an RTSP-
 Version field in the first line of the message.  Note that the major
 and minor numbers MUST be treated as separate integers and that each
 MAY be incremented higher than a single digit.  Thus, RTSP/2.4 is a
 lower version than RTSP/2.13, which, in turn, is lower than
 RTSP/12.3.  Leading zeros SHALL NOT be sent and MUST be ignored by
 recipients.

4.2. RTSP IRI and URI

 RTSP 2.0 defines and registers or updates three URI schemes "rtsp",
 "rtsps", and "rtspu".  The usage of the last, "rtspu", is unspecified
 in RTSP 2.0 and is defined here to register the URI scheme that was
 defined in RTSP 1.0.  The "rtspu" scheme indicates unspecified
 transport of the RTSP messages over unreliable transport means (UDP
 in RTSP 1.0).  An RTSP server MUST respond with an error code
 indicating the "rtspu" scheme is not implemented (501) to a request
 that carries a "rtspu" URI scheme.
 The details of the syntax of "rtsp" and "rtsps" URIs have been
 changed from RTSP 1.0.  These changes include the addition of:
 o  Support for an IPv6 literal in the host part and future IP
    literals through a mechanism defined in [RFC3986].
 o  A new relative format to use in the RTSP elements that is not
    required to start with "/".

Schulzrinne, et al. Standards Track [Page 25] RFC 7826 RTSP 2.0 December 2016

 Neither should have any significant impact on interoperability.  If
 IPv6 literals are needed in the RTSP URI, then that RTSP server must
 be IPv6 capable, and RTSP 1.0 is not a fully IPv6 capable protocol.
 If an RTSP 1.0 client attempts to process the URI, the URI will not
 match the allowed syntax, it will be considered invalid, and
 processing will be stopped.  This is clearly a failure to reach the
 resource; however, it is not a signification issue as RTSP 2.0
 support was needed anyway in both server and client.  Thus, failure
 will only occur in a later step when there is an RTSP version
 mismatch between client and server.  The second change will only
 occur inside RTSP message headers, as the Request-URI must be an
 absolute URI.  Thus, such usages will only occur after an agent has
 accepted and started processing RTSP 2.0 messages, and an agent using
 RTSP 1.0 only will not be required to parse such types of relative
 URIs.
 This specification also defines the format of RTSP IRIs [RFC3987]
 that can be used as RTSP resource identifiers and locators on web
 pages, user interfaces, on paper, etc.  However, the RTSP request
 message format only allows usage of the absolute URI format.  The
 RTSP IRI format MUST use the rules and transformation for IRIs to
 URIs, as defined in [RFC3987].  This allows a URI that matches the
 RTSP 2.0 specification, and so is suitable for use in a request, to
 be created from an RTSP IRI.
 The RTSP IRI and URI are both syntax restricted compared to the
 generic syntax defined in [RFC3986] and [RFC3987]:
 o  An absolute URI requires the authority part; i.e., a host identity
    MUST be provided.
 o  Parameters in the path element are prefixed with the reserved
    separator ";".
 The "scheme" and "host" parts of all URIs [RFC3986] and IRIs
 [RFC3987] are case insensitive.  All other parts of RTSP URIs and
 IRIs are case sensitive, and they MUST NOT be case mapped.
 The fragment identifier is used as defined in Sections 3.5 and 4.3 of
 [RFC3986], i.e., the fragment is to be stripped from the IRI by the
 requester and not included in the Request-URI.  The user agent needs
 to interpret the value of the fragment based on the media type the
 request relates to; i.e., the media type indicated in Content-Type
 header in the response to a DESCRIBE request.
 The syntax of any URI query string is unspecified and responder
 (usually the server) specific.  The query is, from the requester's
 perspective, an opaque string and needs to be handled as such.

Schulzrinne, et al. Standards Track [Page 26] RFC 7826 RTSP 2.0 December 2016

 Please note that relative URIs with queries are difficult to handle
 due to the relative URI handling rules of RFC 3986.  Any change of
 the path element using a relative URI results in the stripping of the
 query, which means the relative part needs to contain the query.
 The URI scheme "rtsp" requires that commands be issued via a reliable
 protocol (within the Internet, TCP), while the scheme "rtsps"
 identifies a reliable transport using secure transport (TLS
 [RFC5246]); see Section 19.
 For the scheme "rtsp", if no port number is provided in the authority
 part of the URI, the port number 554 MUST be used.  For the scheme
 "rtsps", if no port number is provided in the authority part of the
 URI port number, the TCP port 322 MUST be used.
 A presentation or a stream is identified by a textual media
 identifier, using the character set and escape conventions of URIs
 [RFC3986].  URIs may refer to a stream or an aggregate of streams;
 i.e., a presentation.  Accordingly, requests described in Section 13
 can apply to either the whole presentation or an individual stream
 within the presentation.  Note that some request methods can only be
 applied to streams, not presentations, and vice versa.
 For example, the RTSP URI:
    rtsp://media.example.com:554/twister/audiotrack
 may identify the audio stream within the presentation "twister",
 which can be controlled via RTSP requests issued over a TCP
 connection to port 554 of host media.example.com.
 Also, the RTSP URI:
    rtsp://media.example.com:554/twister
 identifies the presentation "twister", which may be composed of audio
 and video streams, but could also be something else, such as a random
 media redirector.
    This does not imply a standard way to reference streams in URIs.
    The presentation description defines the hierarchical
    relationships in the presentation and the URIs for the individual
    streams.  A presentation description may name a stream "a.mov" and
    the whole presentation "b.mov".
 The path components of the RTSP URI are opaque to the client and do
 not imply any particular file system structure for the server.

Schulzrinne, et al. Standards Track [Page 27] RFC 7826 RTSP 2.0 December 2016

    This decoupling also allows presentation descriptions to be used
    with non-RTSP media control protocols simply by replacing the
    scheme in the URI.

4.3. Session Identifiers

 Session identifiers are strings of a length between 8-128 characters.
 A session identifier MUST be generated using methods that make it
 cryptographically random (see [RFC4086]).  It is RECOMMENDED that a
 session identifier contain 128 bits of entropy, i.e., approximately
 22 characters from a high-quality generator (see Section 21).
 However, note that the session identifier does not provide any
 security against session hijacking unless it is kept confidential by
 the client, server, and trusted proxies.

4.4. Media-Time Formats

 RTSP currently supports three different media-time formats defined
 below.  Additional time formats may be specified in the future.
 These time formats can be used with the Range header (Section 18.40)
 to request playback and specify at which media position protocol
 requests actually will or have taken place.  They are also used in
 description of the media's properties using the Media-Range header
 (Section 18.30).  The unqualified format identifier is used on its
 own in Accept-Ranges header (Section 18.5) to declare supported time
 formats and also in the Range header (Section 18.40) to request the
 time format used in the response.

4.4.1. SMPTE-Relative Timestamps

 A timestamp may use a format derived from a Society of Motion Picture
 and Television Engineers (SMPTE) specification and expresses time
 offsets anchored at the start of the media clip.  Relative timestamps
 are expressed as SMPTE time codes [SMPTE-TC] for frame-level access
 accuracy.  The time code has the format:
    hours:minutes:seconds:frames.subframes
 with the origin at the start of the clip.  The default SMPTE format
 is "SMPTE 30 drop" format, with a frame rate of 29.97 frames per
 second.  Other SMPTE codes MAY be supported (such as "SMPTE 25")
 through the use of "smpte-type".  For SMPTE 30, the "frames" field in
 the time value can assume the values 0 through 29.  The difference
 between 30 and 29.97 frames per second is handled by dropping the
 first two frame indices (values 00 and 01) of every minute, except
 every tenth minute.  If the frame and the subframe values are zero,
 they may be omitted.  Subframes are measured in hundredths of a
 frame.

Schulzrinne, et al. Standards Track [Page 28] RFC 7826 RTSP 2.0 December 2016

 Examples:
   smpte=10:12:33:20-
   smpte=10:07:33-
   smpte=10:07:00-10:07:33:05.01
   smpte-25=10:07:00-10:07:33:05.01

4.4.2. Normal Play Time

 Normal Play Time (NPT) indicates the stream-absolute position
 relative to the beginning of the presentation.  The timestamp
 consists of two parts: The mandatory first part may be expressed in
 either seconds only or in hours, minutes, and seconds.  The optional
 second part consists of a decimal point and decimal figures and
 indicates fractions of a second.
 The beginning of a presentation corresponds to 0.0 seconds.  Negative
 values are not defined.
 The special constant "now" is defined as the current instant of a
 live event.  It MAY only be used for live events and MUST NOT be used
 for on-demand (i.e., non-live) content.
 NPT is defined as in Digital Storage Media Command and Control
 (DSMb;CC) [ISO.13818-6.1995]:
    Intuitively, NPT is the clock the viewer associates with a
    program.  It is often digitally displayed on a DVD player.  NPT
    advances normally when in normal play mode (scale = 1), advances
    at a faster rate when in fast-scan forward (high positive scale
    ratio), decrements when in scan reverse (negative scale ratio) and
    is fixed in pause mode.  NPT is (logically) equivalent to SMPTE
    time codes.
 Examples:
   npt=123.45-125
   npt=12:05:35.3-
   npt=now-

Schulzrinne, et al. Standards Track [Page 29] RFC 7826 RTSP 2.0 December 2016

 The syntax is based on ISO 8601 [ISO.8601.2000] and expresses the
 time elapsed since presentation start, with two different notations
 allowed:
 o  The npt-hhmmss notation uses an ISO 8601 extended complete
    representation of the time of the day format (Section 5.3.1.1 of
    [ISO.8601.2000] ) using colons (":") as separators between hours,
    minutes, and seconds (hh:mm:ss).  The hour counter is not limited
    to 0-24 hours; up to nineteen (19) hour digits are allowed.
  • In accordance with the requirements of the ISO 8601 time

format, the hours, minutes, and seconds MUST all be present,

       with two digits used for minutes and for seconds and with at
       least two digits for hours.  An NPT of 7 minutes and 0 seconds
       is represented as "00:07:00", and an NPT of 392 hours, 0
       minutes, and 6 seconds is represented as "392:00:06".
  • RTSP 1.0 allowed NPT in the npt-hhmmss notation without any

leading zeros to ensure that implementations don't fail; for

       backward compatibility, all RTSP 2.0 implementations are
       REQUIRED to support receiving NPT values, hours, minutes, or
       seconds, without leading zeros.
 o  The npt-sec notation expresses the time in seconds, using between
    one and nineteen (19) digits.
 Both notations allow decimal fractions of seconds as specified in
 Section 5.3.1.3 of [ISO.8601.2000], using at most nine digits, and
 allowing only "." (full stop) as the decimal separator.
 The npt-sec notation is optimized for automatic generation; the npt-
 hhmmss notation is optimized for consumption by human readers.  The
 "now" constant allows clients to request to receive the live feed
 rather than the stored or time-delayed version.  This is needed since
 neither absolute time nor zero time are appropriate for this case.

4.4.3. Absolute Time

 Absolute time is expressed using a timestamp based on ISO 8601
 [ISO.8601.2000].  The date is a complete representation of the
 calendar date in basic format (YYYYMMDD) without separators (per
 Section 5.2.1.1 of [ISO.8601.2000]).  The time of day is provided in
 the complete representation basic format (hhmmss) as specified in
 Section 5.3.1.1 of [ISO.8601.2000], allowing decimal fractions of
 seconds following Section 5.3.1.3 requiring "." (full stop) as
 decimal separator and limiting the number of digits to no more than
 nine.  The time expressed MUST use UTC (GMT), i.e., no time zone
 offsets are allowed.  The full date and time specification is the

Schulzrinne, et al. Standards Track [Page 30] RFC 7826 RTSP 2.0 December 2016

 eight-digit date followed by a "T" followed by the six-digit time
 value, optionally followed by a full stop followed by one to nine
 fractions of a second and ended by "Z", e.g., YYYYMMDDThhmmss.ssZ.
    The reasons for this time format rather than using "Date and Time
    on the Internet: Timestamps" [RFC3339] are historic.  We continue
    to use the format specified in RTSP 1.0.  The motivations raised
    in RFC 3339 apply to why a selection from ISO 8601 was made;
    however, a different and even more restrictive selection was
    applied in this case.
 Below are three examples of media time formats, first, a request for
 a clock format range request for a starting time of November 8, 1996
 at 14 h 37 min and 20 1/4 seconds UTC playing for 10 min and 5
 seconds, followed by a Media-Properties header's "Time-Limited" UTC
 property for the 24th of December 2014 at 15 hours and 00 minutes,
 and finally a Terminate-Reason header "time" property for the 18th of
 June 2013 at 16 hours, 12 minutes, and 56 seconds:
   clock=19961108T143720.25Z-19961108T144725.25Z
   Time-Limited=20141224T1500Z
   time=20130618T161256Z

4.5. Feature Tags

 Feature tags are unique identifiers used to designate features in
 RTSP.  These tags are used in Require (Section 18.43), Proxy-Require
 (Section 18.37), Proxy-Supported (Section 18.38), Supported
 (Section 18.51), and Unsupported (Section 18.55) header fields.
 A feature tag definition MUST indicate which combination of clients,
 servers, or proxies to which it applies.
 The creator of a new RTSP feature tag should either prefix the
 feature tag with a reverse domain name (e.g.,
 "com.example.mynewfeature" is an apt name for a feature whose
 inventor can be reached at "example.com") or register the new feature
 tag with the Internet Assigned Numbers Authority (IANA).  (See
 Section 22, "IANA Considerations".)
 The usage of feature tags is further described in Section 11, which
 deals with capability handling.

Schulzrinne, et al. Standards Track [Page 31] RFC 7826 RTSP 2.0 December 2016

4.6. Message Body Tags

 Message body tags are opaque strings that are used to compare two
 message bodies from the same resource, for example, in caches or to
 optimize setup after a redirect.  Message body tags can be carried in
 the MTag header (see Section 18.31) or in SDP (see Appendix D.1.9).
 MTag is similar to ETag in HTTP/1.1 (see Section 3.11 of [RFC2068]).
 A message body tag MUST be unique across all versions of all message
 bodies associated with a particular resource.  A given message body
 tag value MAY be used for message bodies obtained by requests on
 different URIs.  The use of the same message body tag value in
 conjunction with message bodies obtained by requests on different
 URIs does not imply the equivalence of those message bodies.
 Message body tags are used in RTSP to make some methods conditional.
 The methods are made conditional through the inclusion of headers;
 see Section 18.24 and Section 18.26 for information on the If-Match
 and If-None-Match headers, respectively.  Note that RTSP message body
 tags apply to the complete presentation, i.e., both the presentation
 description and the individual media streams.  Thus, message body
 tags can be used to verify at setup time after a redirect that the
 same session description applies to the media at the new location
 using the If-Match header.

4.7. Media Properties

 When an RTSP server handles media, it is important to consider the
 different properties a media instance for delivery and playback can
 have.  This specification considers the media properties listed below
 in its protocol operations.  They are derived from the differences
 between a number of supported usages.
 On-demand:  Media that has a fixed (given) duration that doesn't
    change during the lifetime of the RTSP session and is known at the
    time of the creation of the session.  It is expected that the
    content of the media will not change, even if the representation,
    such as encoding, or quality, may change.  Generally, one can
    seek, i.e., request any range, within the media.
 Dynamic On-demand:  This is a variation of the on-demand case where
    external methods are used to manipulate the actual content of the
    media setup for the RTSP session.  The main example is content
    defined by a playlist.

Schulzrinne, et al. Standards Track [Page 32] RFC 7826 RTSP 2.0 December 2016

 Live:  Live media represents a progressing content stream (such as
    broadcast TV) where the duration may or may not be known.  It is
    not seekable, only the content presently being delivered can be
    accessed.
 Live with Recording:  A live stream that is combined with a server-
    side capability to store and retain the content of the live
    session and allow for random access delivery within the part of
    the already-recorded content.  The actual behavior of the media
    stream is very much dependent on the retention policy for the
    media stream; either the server will be able to capture the
    complete media stream or it will have a limitation in how much
    will be retained.  The media range will dynamically change as the
    session progress.  For servers with a limited amount of storage
    available for recording, there will typically be a sliding window
    that moves forward while new data is made available and older data
    is discarded.
 To cover the above usages, the following media properties with
 appropriate values are specified.

4.7.1. Random Access and Seeking

 Random access is the ability to specify and get media delivered
 starting from any time (instant) within the content, an operation
 called "seeking".  The Media-Properties header will indicate the
 general capability for a media resource to perform random access.
 Random-Access:  The media is seekable to any out of a large number of
    points within the media.  Due to media-encoding limitations, a
    particular point may not be reachable, but seeking to a point
    close by is enabled.  A floating-point number of seconds may be
    provided to express the worst-case distance between random access
    points.
 Beginning-Only:  Seeking is only possible to the beginning of the
    content.
 No-Seeking:  Seeking is not possible at all.
 If random access is possible, as indicated by the Media-Properties
 header, the actual behavior policy when seeking can be controlled
 using the Seek-Style header (Section 18.47).

Schulzrinne, et al. Standards Track [Page 33] RFC 7826 RTSP 2.0 December 2016

4.7.2. Retention

 The following retention policies are used by media to limit possible
 protocol operations:
 Unlimited:  The media will not be removed as long as the RTSP session
    is in existence.
 Time-Limited:  The media will not be removed before the given
    wallclock time.  After that time, it may or may not be available
    anymore.
 Time-Duration:  The media (on fragment or unit basis) will be
    retained for the specified duration.

4.7.3. Content Modifications

 The media content and its timeline can be of different types, e.g.
 pre-produced content on demand, a live source that is being generated
 as time progresses, or something that is dynamically altered or
 recomposed during playback.  Therefore, a media property for content
 modifications is needed and the following initial values are defined:
 Immutable:  The content of the media will not change, even if the
    representation, such as encoding or quality changes.
 Dynamic:  The content can change due to external methods or triggers,
    such as playlists, but this will be announced by explicit updates.
 Time-Progressing:  As time progresses, new content will become
    available.  If the content is also retained, it will become longer
    as everything between the start point and the point currently
    being made available can be accessed.  If the media server uses a
    sliding-window policy for retention, the start point will also
    change as time progresses.

4.7.4. Supported Scale Factors

 A particular media content item often supports only a limited set or
 range of scales when delivering the media.  To enable the client to
 know what values or ranges of scale operations that the whole content
 or the current position supports, a media properties attribute for
 this is defined that contains a list with the values or ranges that
 are supported.  The attribute is named "Scales".  The "Scales"
 attribute may be updated at any point in the content due to content
 consisting of spliced pieces or content being dynamically updated by
 out-of-band mechanisms.

Schulzrinne, et al. Standards Track [Page 34] RFC 7826 RTSP 2.0 December 2016

4.7.5. Mapping to the Attributes

 This section shows examples of how one would map the above usages to
 the properties and their values.
 Example of On-Demand:
    Random Access: Random-Access=5.0, Content Modifications:
    Immutable, Retention: Unlimited or Time-Limited.
 Example of Dynamic On-Demand:
    Random Access: Random-Access=3.0, Content Modifications: Dynamic,
    Retention: Unlimited or Time-Limited.
 Example of Live:
    Random Access: No-Seeking, Content Modifications: Time-
    Progressing, Retention: Time-Duration=0.0
 Example of Live with Recording:
    Random Access: Random-Access=3.0, Content Modifications: Time-
    Progressing, Retention: Time-Duration=7200.0

5. RTSP Message

 RTSP is a text-based protocol that uses the ISO 10646 character set
 in UTF-8 encoding per RFC 3629 [RFC3629].  Lines MUST be terminated
 by a CRLF.
    Text-based protocols make it easier to add optional parameters in
    a self-describing manner.  Since the number of parameters and the
    frequency of commands is low, processing efficiency is not a
    concern.  Text-based protocols, if used carefully, also allow easy
    implementation of research prototypes in scripting languages such
    as Python, PHP, Perl and TCL.
 The ISO 10646 character set avoids character-set switching, but is
 invisible to the application as long as US-ASCII is being used.  This
 is also the encoding used for text fields in RTCP [RFC3550].
 A request contains a method, the object the method is operating upon,
 and parameters to further describe the method.  Methods are
 idempotent unless otherwise noted.  Methods are also designed to
 require little or no state maintenance at the media server.

Schulzrinne, et al. Standards Track [Page 35] RFC 7826 RTSP 2.0 December 2016

5.1. Message Types

 RTSP messages are either requests from client to server or from
 server to client, and responses in the reverse direction.  Request
 (Section 7) and response (Section 8) messages use a format based on
 the generic message format of RFC 5322 [RFC5322] for transferring
 bodies (the payload of the message).  Both types of messages consist
 of a start-line, zero or more header fields (also known as
 "headers"), an empty line (i.e., a line with nothing preceding the
 CRLF) indicating the end of the headers, and possibly the data of the
 message body.  The ABNF [RFC5234] below is for illustration only; the
 formal message specification is presented in Section 20.2.2.
 generic-message = start-line
                 *(rtsp-header CRLF)
                   CRLF
                 [ message-body-data ]
 start-line = Request-Line / Status-Line
 In the interest of robustness, agents MUST ignore any empty line(s)
 received where a Request-Line or Status-Line is expected.  In other
 words, if the agent is reading the protocol stream at the beginning
 of a message and receives any number of CRLFs first, it MUST ignore
 all of the CRLFs.

5.2. Message Headers

 RTSP header fields (see Section 18) include general-header, request-
 header, response-header, and message body header fields.
 The order in which header fields with differing field names are
 received is not significant.  However, it is "good practice" to send
 general-header fields first, followed by a request-header or
 response-header field, and ending with the message body header
 fields.
 Multiple header fields with the same field-name MAY be present in a
 message if and only if the entire field-value for that header field
 is defined as a comma-separated list.  It MUST be possible to combine
 the multiple header fields into one "field-name: field-value" pair,
 without changing the semantics of the message, by appending each
 subsequent field-value to the first, each separated by a comma.  The
 order in which header fields with the same field-name are received is
 therefore significant to the interpretation of the combined field
 value; thus, a proxy MUST NOT change the order of these field-values
 when a message is forwarded.

Schulzrinne, et al. Standards Track [Page 36] RFC 7826 RTSP 2.0 December 2016

 Unknown message headers MUST be ignored (skipping over the header to
 the next protocol element, and not causing an error) by an RTSP
 server or client.  An RTSP proxy MUST forward unknown message
 headers.  Message headers defined outside of this specification that
 are required to be interpreted by the RTSP agent will need to use
 feature tags (Section 4.5) and include them in the appropriate
 Require (Section 18.43) or Proxy-Require (Section 18.37) header.

5.3. Message Body

 The message body (if any) of an RTSP message is used to carry further
 information for a particular resource associated with the request or
 response.  An example of a message body is an SDP message.
 The presence of a message body in either a request or a response MUST
 be signaled by the inclusion of a Content-Length header (see
 Section 18.17) and Content-Type header (see Section 18.19).  A
 message body MUST NOT be included in a request or response if the
 specification of the particular method (see Method Definitions
 (Section 13)) does not allow sending a message body.  In case a
 message body is received in a message when not expected, the message
 body data SHOULD be discarded.  This is to allow future extensions to
 define optional use of a message body.

5.4. Message Length

 An RTSP message that does not contain any message body is terminated
 by the first empty line after the header fields (note: an empty line
 is a line with nothing preceding the CRLF.).  In RTSP messages that
 contain message bodies, the empty line is followed by the message
 body.  The length of that body is determined by the value of the
 Content-Length header (Section 18.17).  The value in the header
 represents the length of the message body in octets.  If this header
 field is not present, a value of zero is assumed, i.e., no message
 body present in the message.  Unlike an HTTP message, an RTSP message
 MUST contain a Content-Length header whenever it contains a message
 body.  Note that RTSP does not support the HTTP/1.1 "chunked"
 transfer coding (see Section 4.1 of [RFC7230]).
    Given the moderate length of presentation descriptions returned,
    the server should always be able to determine its length, even if
    it is generated dynamically, making the chunked transfer encoding
    unnecessary.

6. General-Header Fields

 General headers are headers that may be used in both requests and
 responses.  The general-headers are listed in Table 1:

Schulzrinne, et al. Standards Track [Page 37] RFC 7826 RTSP 2.0 December 2016

                +--------------------+----------------+
                | Header Name        | Defined in     |
                +--------------------+----------------+
                | Accept-Ranges      | Section 18.5   |
                |                    |                |
                | Cache-Control      | Section 18.11  |
                |                    |                |
                | Connection         | Section 18.12  |
                |                    |                |
                | CSeq               | Section 18.20  |
                |                    |                |
                | Date               | Section 18.21  |
                |                    |                |
                | Media-Properties   | Section 18.29  |
                |                    |                |
                | Media-Range        | Section 18.30  |
                |                    |                |
                | Pipelined-Requests | Section 18.33  |
                |                    |                |
                | Proxy-Supported    | Section 18.38  |
                |                    |                |
                | Range              | Section 18.40  |
                |                    |                |
                | RTP-Info           | Section 18.45  |
                |                    |                |
                | Scale              | Section 18.46  |
                |                    |                |
                | Seek-Style         | Section 18.47  |
                |                    |                |
                | Server             | Section 18.48  |
                |                    |                |
                | Session            | Section 18.49  |
                |                    |                |
                | Speed              | Section 18.50  |
                |                    |                |
                | Supported          | Section 18.51  |
                |                    |                |
                | Timestamp          | Section 18.53  |
                |                    |                |
                | Transport          | Section 18.54  |
                |                    |                |
                | User-Agent         | Section 18.56  |
                |                    |                |
                | Via                | Section 18.57  |
                +--------------------+----------------+
               Table 1: The General Headers Used in RTSP

Schulzrinne, et al. Standards Track [Page 38] RFC 7826 RTSP 2.0 December 2016

7. Request

 A request message uses the format outlined below regardless of the
 direction of a request, whether client to server or server to client:
 o  Request line, containing the method to be applied to the resource,
    the identifier of the resource, and the protocol version in use;
 o  Zero or more Header lines, which can be of the following types:
    general-headers (Section 6), request-headers (Section 7.2), or
    message body headers (Section 9.1);
 o  One empty line (CRLF) to indicate the end of the header section;
 o  Optionally, a message body, consisting of one or more lines.  The
    length of the message body in octets is indicated by the Content-
    Length message header.

Schulzrinne, et al. Standards Track [Page 39] RFC 7826 RTSP 2.0 December 2016

7.1. Request Line

 The request line provides the key information about the request: what
 method, on what resources, and using which RTSP version.  The methods
 that are defined by this specification are listed in Table 2.
                  +---------------+----------------+
                  | Method        | Defined in     |
                  +---------------+----------------+
                  | DESCRIBE      | Section 13.2   |
                  |               |                |
                  | GET_PARAMETER | Section 13.8   |
                  |               |                |
                  | OPTIONS       | Section 13.1   |
                  |               |                |
                  | PAUSE         | Section 13.6   |
                  |               |                |
                  | PLAY          | Section 13.4   |
                  |               |                |
                  | PLAY_NOTIFY   | Section 13.5   |
                  |               |                |
                  | REDIRECT      | Section 13.10  |
                  |               |                |
                  | SETUP         | Section 13.3   |
                  |               |                |
                  | SET_PARAMETER | Section 13.9   |
                  |               |                |
                  | TEARDOWN      | Section 13.7   |
                  +---------------+----------------+
                       Table 2: The RTSP Methods
 The syntax of the RTSP request line has the following:
    <Method> SP <Request-URI> SP <RTSP-Version> CRLF
 Note: This syntax cannot be freely changed in future versions of
 RTSP.  This line needs to remain parsable by older RTSP
 implementations since it indicates the RTSP version of the message.
 In contrast to HTTP/1.1 [RFC7230], RTSP requests identify the
 resource through an absolute RTSP URI (including scheme, host, and
 port) (see Section 4.2) rather than just the absolute path.
    HTTP/1.1 requires servers to understand the absolute URI, but
    clients are supposed to use the Host request-header.  This is
    purely needed for backward compatibility with HTTP/1.0 servers, a
    consideration that does not apply to RTSP.

Schulzrinne, et al. Standards Track [Page 40] RFC 7826 RTSP 2.0 December 2016

 An asterisk "*" can be used instead of an absolute URI in the
 Request-URI part to indicate that the request does not apply to a
 particular resource but to the server or proxy itself, and is only
 allowed when the request method does not necessarily apply to a
 resource.
 For example:
    OPTIONS * RTSP/2.0
 An OPTIONS in this form will determine the capabilities of the server
 or the proxy that first receives the request.  If the capability of
 the specific server needs to be determined, without regard to the
 capability of an intervening proxy, the server should be addressed
 explicitly with an absolute URI that contains the server's address.
 For example:
    OPTIONS rtsp://example.com RTSP/2.0

Schulzrinne, et al. Standards Track [Page 41] RFC 7826 RTSP 2.0 December 2016

7.2. Request-Header Fields

 The RTSP headers in Table 3 can be included in a request, as request-
 headers, to modify the specifics of the request.
               +---------------------+----------------+
               | Header              | Defined in     |
               +---------------------+----------------+
               | Accept              | Section 18.1   |
               |                     |                |
               | Accept-Credentials  | Section 18.2   |
               |                     |                |
               | Accept-Encoding     | Section 18.3   |
               |                     |                |
               | Accept-Language     | Section 18.4   |
               |                     |                |
               | Authorization       | Section 18.8   |
               |                     |                |
               | Bandwidth           | Section 18.9   |
               |                     |                |
               | Blocksize           | Section 18.10  |
               |                     |                |
               | From                | Section 18.23  |
               |                     |                |
               | If-Match            | Section 18.24  |
               |                     |                |
               | If-Modified-Since   | Section 18.25  |
               |                     |                |
               | If-None-Match       | Section 18.26  |
               |                     |                |
               | Notify-Reason       | Section 18.32  |
               |                     |                |
               | Proxy-Authorization | Section 18.36  |
               |                     |                |
               | Proxy-Require       | Section 18.37  |
               |                     |                |
               | Referrer            | Section 18.41  |
               |                     |                |
               | Request-Status      | Section 18.42  |
               |                     |                |
               | Require             | Section 18.43  |
               |                     |                |
               | Terminate-Reason    | Section 18.52  |
               +---------------------+----------------+
                   Table 3: The RTSP Request-Headers
 Detailed header definitions are provided in Section 18.

Schulzrinne, et al. Standards Track [Page 42] RFC 7826 RTSP 2.0 December 2016

 New request-headers may be defined.  If the receiver of the request
 is required to understand the request-header, the request MUST
 include a corresponding feature tag in a Require or Proxy-Require
 header to ensure the processing of the header.

8. Response

 After receiving and interpreting a request message, the recipient
 responds with an RTSP response message.  Normally, there is only one,
 final, response.  Responses using the response code class 1xx is the
 only class for which there MAY be sent one or more responses prior to
 the final response message.
 The valid response codes and the methods they can be used with are
 listed in Table 4.

8.1. Status-Line

 The first line of a response message is the Status-Line, consisting
 of the protocol version followed by a numeric status code and the
 textual phrase associated with the status code, with each element
 separated by SP characters.  No CR or LF is allowed except in the
 final CRLF sequence.
 <RTSP-Version> SP <Status-Code> SP <Reason Phrase> CRLF

8.1.1. Status Code and Reason Phrase

 The Status-Code element is a 3-digit integer result code of the
 attempt to understand and satisfy the request.  These codes are fully
 defined in Section 17.  The reason phrase is intended to give a short
 textual description of the Status-Code.  The Status-Code is intended
 for use by automata and the reason phrase is intended for the human
 user.  The client is not required to examine or display the reason
 phrase.
 The first digit of the Status-Code defines the class of response.
 The last two digits do not have any categorization role.  There are
 five values for the first digit:
 1xx:  Informational - Request received, continuing process
 2xx:  Success - The action was successfully received, understood, and
       accepted
 3rr:  Redirection - Further action needs to be taken in order to
       complete the request (3rr rather than 3xx is used as 304 is
       excluded; see Section 17.3)

Schulzrinne, et al. Standards Track [Page 43] RFC 7826 RTSP 2.0 December 2016

 4xx:  Client Error - The request contains bad syntax or cannot be
       fulfilled
 5xx:  Server Error - The server failed to fulfill an apparently valid
       request
 The individual values of the numeric status codes defined for RTSP
 2.0, and an example set of corresponding reason phrases, are
 presented in Table 4.  The reason phrases listed here are only
 recommended; they may be replaced by local equivalents without
 affecting the protocol.  Note that RTSP adopted most HTTP/1.1
 [RFC2068] status codes and then added RTSP-specific status codes
 starting at x50 to avoid conflicts with future HTTP status codes that
 are desirable to import into RTSP.  All these codes are RTSP specific
 and RTSP has its own registry separate from HTTP for status codes.
 RTSP status codes are extensible.  RTSP applications are not required
 to understand the meaning of all registered status codes, though such
 understanding is obviously desirable.  However, applications MUST
 understand the class of any status code, as indicated by the first
 digit, and treat any unrecognized response as being equivalent to the
 x00 status code of that class, with an exception for unknown 3xx
 codes, which MUST be treated as a 302 (Found).  The reason for that
 exception is that the status code 300 (Multiple Choices in HTTP) is
 not defined for RTSP.  A response with an unrecognized status code
 MUST NOT be cached.  For example, if an unrecognized status code of
 431 is received by the client, it can safely assume that there was
 something wrong with its request and treat the response as if it had
 received a 400 status code.  In such cases, user agents SHOULD
 present to the user the message body returned with the response,
 since that message body is likely to include human-readable
 information that will explain the unusual status.
 +------+---------------------------------+--------------------------+
 | Code | Reason                          | Method                   |
 +------+---------------------------------+--------------------------+
 | 100  | Continue                        | all                      |
 |      |                                 |                          |
 | 200  | OK                              | all                      |
 |      |                                 |                          |
 | 301  | Moved Permanently               | all                      |
 |      |                                 |                          |
 | 302  | Found                           | all                      |
 |      |                                 |                          |
 | 303  | See Other                       | n/a                      |
 |      |                                 |                          |
 | 304  | Not Modified                    | all                      |
 |      |                                 |                          |

Schulzrinne, et al. Standards Track [Page 44] RFC 7826 RTSP 2.0 December 2016

 | 305  | Use Proxy                       | all                      |
 |      |                                 |                          |
 | 400  | Bad Request                     | all                      |
 |      |                                 |                          |
 | 401  | Unauthorized                    | all                      |
 |      |                                 |                          |
 | 402  | Payment Required                | all                      |
 |      |                                 |                          |
 | 403  | Forbidden                       | all                      |
 |      |                                 |                          |
 | 404  | Not Found                       | all                      |
 |      |                                 |                          |
 | 405  | Method Not Allowed              | all                      |
 |      |                                 |                          |
 | 406  | Not Acceptable                  | all                      |
 |      |                                 |                          |
 | 407  | Proxy Authentication Required   | all                      |
 |      |                                 |                          |
 | 408  | Request Timeout                 | all                      |
 |      |                                 |                          |
 | 410  | Gone                            | all                      |
 |      |                                 |                          |
 | 412  | Precondition Failed             | DESCRIBE, SETUP          |
 |      |                                 |                          |
 | 413  | Request Message Body Too Large  | all                      |
 |      |                                 |                          |
 | 414  | Request-URI Too Long            | all                      |
 |      |                                 |                          |
 | 415  | Unsupported Media Type          | all                      |
 |      |                                 |                          |
 | 451  | Parameter Not Understood        | SET_PARAMETER,           |
 |      |                                 | GET_PARAMETER            |
 |      |                                 |                          |
 | 452  | reserved                        | n/a                      |
 |      |                                 |                          |
 | 453  | Not Enough Bandwidth            | SETUP                    |
 |      |                                 |                          |
 | 454  | Session Not Found               | all                      |
 |      |                                 |                          |
 | 455  | Method Not Valid in This State  | all                      |
 |      |                                 |                          |
 | 456  | Header Field Not Valid for      | all                      |
 |      | Resource                        |                          |
 |      |                                 |                          |
 | 457  | Invalid Range                   | PLAY, PAUSE              |
 |      |                                 |                          |
 | 458  | Parameter Is Read-Only          | SET_PARAMETER            |
 |      |                                 |                          |

Schulzrinne, et al. Standards Track [Page 45] RFC 7826 RTSP 2.0 December 2016

 | 459  | Aggregate Operation Not Allowed | all                      |
 |      |                                 |                          |
 | 460  | Only Aggregate Operation        | all                      |
 |      | Allowed                         |                          |
 |      |                                 |                          |
 | 461  | Unsupported Transport           | all                      |
 |      |                                 |                          |
 | 462  | Destination Unreachable         | all                      |
 |      |                                 |                          |
 | 463  | Destination Prohibited          | SETUP                    |
 |      |                                 |                          |
 | 464  | Data Transport Not Ready Yet    | PLAY                     |
 |      |                                 |                          |
 | 465  | Notification Reason Unknown     | PLAY_NOTIFY              |
 |      |                                 |                          |
 | 466  | Key Management Error            | all                      |
 |      |                                 |                          |
 | 470  | Connection Authorization        | all                      |
 |      | Required                        |                          |
 |      |                                 |                          |
 | 471  | Connection Credentials Not      | all                      |
 |      | Accepted                        |                          |
 |      |                                 |                          |
 | 472  | Failure to Establish Secure     | all                      |
 |      | Connection                      |                          |
 |      |                                 |                          |
 | 500  | Internal Server Error           | all                      |
 |      |                                 |                          |
 | 501  | Not Implemented                 | all                      |
 |      |                                 |                          |
 | 502  | Bad Gateway                     | all                      |
 |      |                                 |                          |
 | 503  | Service Unavailable             | all                      |
 |      |                                 |                          |
 | 504  | Gateway Timeout                 | all                      |
 |      |                                 |                          |
 | 505  | RTSP Version Not Supported      | all                      |
 |      |                                 |                          |
 | 551  | Option Not Supported            | all                      |
 |      |                                 |                          |
 | 553  | Proxy Unavailable               | all                      |
 +------+---------------------------------+--------------------------+
        Table 4: Status Codes and Their Usage with RTSP Methods

Schulzrinne, et al. Standards Track [Page 46] RFC 7826 RTSP 2.0 December 2016

8.2. Response Headers

 The response-headers allow the request recipient to pass additional
 information about the response that cannot be placed in the Status-
 Line.  This header gives information about the server and about
 further access to the resource identified by the Request-URI.  All
 headers currently classified as response-headers are listed in
 Table 5.
              +------------------------+----------------+
              | Header                 | Defined in     |
              +------------------------+----------------+
              | Authentication-Info    | Section 18.7   |
              |                        |                |
              | Connection-Credentials | Section 18.13  |
              |                        |                |
              | Location               | Section 18.28  |
              |                        |                |
              | MTag                   | Section 18.31  |
              |                        |                |
              | Proxy-Authenticate     | Section 18.34  |
              |                        |                |
              | Public                 | Section 18.39  |
              |                        |                |
              | Retry-After            | Section 18.44  |
              |                        |                |
              | Unsupported            | Section 18.55  |
              |                        |                |
              | WWW-Authenticate       | Section 18.58  |
              +------------------------+----------------+
                  Table 5: The RTSP Response Headers
 Response-header names can be extended reliably only in combination
 with a change in the protocol version.  However, the usage of feature
 tags in the request allows the responding party to learn the
 capability of the receiver of the response.  A new or experimental
 header can be given the semantics of response-header if all parties
 in the communication recognize them to be a response-header.
 Unrecognized headers in responses MUST be ignored.

9. Message Body

 Some request and response messages include a message body, if not
 otherwise restricted by the request method or response status code.
 The message body consists of the content data itself (see also
 Section 5.3).

Schulzrinne, et al. Standards Track [Page 47] RFC 7826 RTSP 2.0 December 2016

 The SET_PARAMETER and GET_PARAMETER requests and responses, and the
 DESCRIBE response as defined by this specification, can have a
 message body; the purpose of the message body is defined in each
 case.  All 4xx and 5xx responses MAY also have a message body to
 carry additional response information.  Generally, a message body MAY
 be attached to any RTSP 2.0 request or response, but the content of
 the message body MAY be ignored by the receiver.  Extensions to this
 specification can specify the purpose and content of message bodies,
 including requiring their inclusion.
 In this section, both sender and recipient refer to either the client
 or the server, depending on who sends and who receives the message
 body.

9.1. Message Body Header Fields

 Message body header fields define meta-information about the content
 data in the message body.  The message body header fields are listed
 in Table 6.
                 +------------------+----------------+
                 | Header           | Defined in     |
                 +------------------+----------------+
                 | Allow            | Section 18.6   |
                 |                  |                |
                 | Content-Base     | Section 18.14  |
                 |                  |                |
                 | Content-Encoding | Section 18.15  |
                 |                  |                |
                 | Content-Language | Section 18.16  |
                 |                  |                |
                 | Content-Length   | Section 18.17  |
                 |                  |                |
                 | Content-Location | Section 18.18  |
                 |                  |                |
                 | Content-Type     | Section 18.19  |
                 |                  |                |
                 | Expires          | Section 18.22  |
                 |                  |                |
                 | Last-Modified    | Section 18.27  |
                 +------------------+----------------+
                Table 6: The RTSP Message Body Headers

Schulzrinne, et al. Standards Track [Page 48] RFC 7826 RTSP 2.0 December 2016

 The extension-header mechanism allows additional message body header
 fields to be defined without changing the protocol, but these fields
 cannot be assumed to be recognizable by the recipient.  Unrecognized
 header fields MUST be ignored by the recipient and forwarded by
 proxies.

9.2. Message Body

 An RTSP message with a message body MUST include the Content-Type and
 Content-Length headers.  When a message body is included with a
 message, the data type of that content data is determined via the
 Content-Type and Content-Encoding header fields.
 Content-Type specifies the media type of the underlying data.  There
 is no default media format and the actual format used in the body is
 required to be explicitly stated in the Content-Type header.  By
 being explicit and always requiring the inclusion of the Content-Type
 header with accurate information, one avoids the many pitfalls in a
 heuristic-based interpretation of the body content.  The user
 experience of HTTP and email have suffered from relying on such
 heuristics.
 Content-Encoding may be used to indicate any additional content-
 codings applied to the data, usually for the purpose of data
 compression, that are a property of the requested resource.  The
 default encoding is 'identity', i.e. no transformation of the message
 body.
 The Content-Length of a message is the length of the content,
 measured in octets.

9.3. Message Body Format Negotiation

 The content format of the message body is provided using the Content-
 Type header (Section 18.19).  To enable the responder of a request to
 determine which media type it should use, the requester may include
 the Accept header (Section 18.1) in a request to identify supported
 media types or media type ranges suitable to the response.  In case
 the responder is not supporting any of the specified formats, then
 the request response will be a 406 (Not Acceptable) error code.
 The media types that may be used on requests with message bodies need
 to be determined through the use of feature tags, specification
 requirement, or trial and error.  Trial and error works because when
 the responder does not support the media type of the message body, it
 will respond with a 415 (Unsupported Media Type).

Schulzrinne, et al. Standards Track [Page 49] RFC 7826 RTSP 2.0 December 2016

 The formats supported and their negotiation is done individually on a
 per method and direction (request or response body) direction.
 Requirements on supporting particular media types for use as message
 bodies in requests and response SHALL also be specified on a per-
 method and per-direction basis.

10. Connections

 RTSP messages are transferred between RTSP agents and proxies using a
 transport connection.  This transport connection uses TCP or TCP/TLS.
 This transport connection is referred to as the "connection" or "RTSP
 connection" within this document.
 RTSP requests can be transmitted using the two different connection
 scenarios listed below:
 o  persistent - a transport connection is used for several request/
    response transactions;
 o  transient - a transport connection is used for each single
    request/response transaction.
 RFC 2326 attempted to specify an optional mechanism for transmitting
 RTSP messages in connectionless mode over a transport protocol such
 as UDP.  However, it was not specified in sufficient detail to allow
 for interoperable implementations.  In an attempt to reduce
 complexity and scope, and due to lack of interest, RTSP 2.0 does not
 attempt to define a mechanism for supporting RTSP over UDP or other
 connectionless transport protocols.  A side effect of this is that
 RTSP requests MUST NOT be sent to multicast groups since no
 connection can be established with a specific receiver in multicast
 environments.
 Certain RTSP headers, such as the CSeq header (Section 18.20), which
 may appear to be relevant only to connectionless transport scenarios,
 are still retained and MUST be implemented according to this
 specification.  In the case of CSeq, it is quite useful for matching
 responses to requests if the requests are pipelined (see Section 12).
 It is also useful in proxies for keeping track of the different
 requests when aggregating several client requests on a single TCP
 connection.

10.1. Reliability and Acknowledgements

 Since RTSP messages are transmitted using reliable transport
 protocols, they MUST NOT be retransmitted at the RTSP level.
 Instead, the implementation must rely on the underlying transport to

Schulzrinne, et al. Standards Track [Page 50] RFC 7826 RTSP 2.0 December 2016

 provide reliability.  The RTSP implementation may use any indication
 of reception acknowledgment of the message from the underlying
 transport protocols to optimize the RTSP behavior.
    If both the underlying reliable transport, such as TCP, and the
    RTSP application retransmit requests, each packet loss or message
    loss may result in two retransmissions.  The receiver typically
    cannot take advantage of the application-layer retransmission
    since the transport stack will not deliver the application-layer
    retransmission before the first attempt has reached the receiver.
    If the packet loss is caused by congestion, multiple
    retransmissions at different layers will exacerbate the
    congestion.
 Lack of acknowledgment of an RTSP request should be handled within
 the constraints of the connection timeout considerations described
 below (Section 10.4).

10.2. Using Connections

 A TCP transport can be used for both persistent connections (for
 several message exchanges) and transient connections (for a single
 message exchange).  Implementations of this specification MUST
 support RTSP over TCP.  The scheme of the RTSP URI (Section 4.2)
 allows the client to specify the port it will contact the server on,
 and defines the default port to use if one is not explicitly given.
 In addition to the registered default ports, i.e., 554 (rtsp) and 322
 (rtsps), there is an alternative port 8554 registered.  This port may
 provide some benefits over non-registered ports if an RTSP server is
 unable to use the default ports.  The benefits may include
 preconfigured security policies as well as classifiers in network
 monitoring tools.
 An RTSP client opening a TCP connection to access a particular
 resource as identified by a URI uses the IP address and port derived
 from the host and port parts of the URI.  The IP address is either
 the explicit address provided in the URI or any of the addresses
 provided when performing A and AAAA record DNS lookups of the
 hostname in the URI.
 A server MUST handle both persistent and transient connections.
    Transient connections facilitate mechanisms for fault tolerance.
    They also allow for application-layer mobility.  A server-and-
    client pair that supports transient connections can survive the

Schulzrinne, et al. Standards Track [Page 51] RFC 7826 RTSP 2.0 December 2016

    loss of a TCP connection; e.g., due to a NAT timeout.  When the
    client has discovered that the TCP connection has been lost, it
    can set up a new one when there is need to communicate again.
 A persistent connection is RECOMMENDED to be used for all
 transactions between the server and client, including messages for
 multiple RTSP sessions.  However, a persistent connection MAY be
 closed after a few message exchanges.  For example, a client may use
 a persistent connection for the initial SETUP and PLAY message
 exchanges in a session and then close the connection.  Later, when
 the client wishes to send a new request, such as a PAUSE for the
 session, a new connection would be opened.  This connection may be
 either transient or persistent.
 An RTSP agent MAY use one connection to handle multiple RTSP sessions
 on the same server.  The RTSP agent SHALL NOT use more than one
 connection per RTSP session at any given point.
    Having only one connection in use at any time avoids confusion
    regarding on which connection any server-to-client requests shall
    be sent.  Using a single connection for multiple RTSP sessions
    also saves complexity by enabling the server to maintain less
    state about its connection resources on the server.  Not using
    more than one connection at a time for a particular RTSP session
    avoids wasting connection resources and allows the server to track
    only the most recently used client-to-server connection for each
    RTSP session as being the currently valid server-to-client
    connection.
 RTSP allows a server to send requests to a client.  However, this can
 be supported only if a client establishes a persistent connection
 with the server.  In cases where a persistent connection does not
 exist between a server and its client, due to the lack of a signaling
 channel, the server may be forced to silently discard RTSP messages,
 and it may even drop an RTSP session without notifying the client.
 An example of such a case is when the server desires to send a
 REDIRECT request for an RTSP session to the client but is not able to
 do so because it cannot reach the client.  A server that attempts to
 send a request to a client that has no connection currently to the
 server SHALL discard the request.
    Without a persistent connection between the client and the server,
    the media server has no reliable way of reaching the client.
    Because of the likely failure of server-to-client established
    connections, the server will not even attempt establishing any
    connection.

Schulzrinne, et al. Standards Track [Page 52] RFC 7826 RTSP 2.0 December 2016

    Queuing of server-to-client requests has been considered.
    However, a security issue exists as to how it might be possible to
    authorize a client establishing a new connection as being a
    legitimate receiver of a request related to a particular RTSP
    session, without the client first issuing requests related to the
    pending request.  Thus, it would be likely to make any such
    requests even more delayed and less useful.
 The sending of client and server requests can be asynchronous events.
 To avoid deadlock situations, both client and server MUST be able to
 send and receive requests simultaneously.  As an RTSP response may be
 queued up for transmission, reception or processing behind the peer
 RTSP agent's own requests, all RTSP agents are required to have a
 certain capability of handling outstanding messages.  A potential
 issue is that outstanding requests may time out despite being
 processed by the peer; this can be due to the response being caught
 in the queue behind a number of requests that the RTSP agent is
 processing but that take some time to complete.  To avoid this
 problem, an RTSP agent should buffer incoming messages locally so
 that any response messages can be processed immediately upon
 reception.  If responses are separated from requests and directly
 forwarded for processing, not only can the result be used
 immediately, the state associated with that outstanding request can
 also be released.  However, buffering a number of requests on the
 receiving RTSP agent consumes resources and enables a resource
 exhaustion attack on the agent.  Therefore, this buffer should be
 limited so that an unreasonable number of requests or total message
 size is not allowed to consume the receiving agent's resources.  In
 most APIs, having the receiving agent stop reading from the TCP
 socket will result in TCP's window being clamped, thus forcing the
 buffering onto the sending agent when the load is larger than
 expected.  However, as both RTSP message sizes and frequency may be
 changed in the future by protocol extensions, an agent should be
 careful about taking harsher measurements against a potential attack.
 When under attack, an RTSP agent can close TCP connections and
 release state associated with that TCP connection.
 To provide some guidance on what is reasonable, the following
 guidelines are given.  It is RECOMMENDED that:
 o  an RTSP agent should not have more than 10 outstanding requests
    per RTSP session;
 o  an RTSP agent should not have more than 10 outstanding requests
    that are not related to an RTSP session or that are requesting to
    create an RTSP session.

Schulzrinne, et al. Standards Track [Page 53] RFC 7826 RTSP 2.0 December 2016

 In light of the above, it is RECOMMENDED that clients use persistent
 connections whenever possible.  A client that supports persistent
 connections MAY "pipeline" its requests (see Section 12).
 RTSP agents can send requests to multiple different destinations,
 either server or client contexts over the same connection to a proxy.
 Then, the proxy forks the message to the different destinations over
 proxy-to-agent connections.  In these cases when multiple requests
 are outstanding, the requesting agent MUST be ready to receive the
 responses out of order compared to the order they where sent on the
 connection.  The order between multiple messages for each destination
 will be maintained; however, the order between response from
 different destinations can be different.
    The reason for this is to avoid a head-of-line blocking situation.
    In a sequence of requests, an early outstanding request may take
    time to be processed at one destination.  Simultaneously, a
    response from any other destination that was later in the sequence
    of requests may have arrived at the proxy; thus, allowing out-of-
    order responses avoids forcing the proxy to buffer this response
    and instead deliver it as soon as possible.  Note, this will not
    affect the order in which the messages sent to each separate
    destination were processed at the request destination.
 This scenario can occur in two cases involving proxies.  The first is
 a client issuing requests for sessions on different servers using a
 common client-to-proxy connection.  The second is for server-to-
 client requests, like REDIRECT being sent by the server over a common
 transport connection the proxy created for its different connecting
 clients.

10.3. Closing Connections

 The client MAY close a connection at any point when no outstanding
 request/response transactions exist for any RTSP session being
 managed through the connection.  The server, however, SHOULD NOT
 close a connection until all RTSP sessions being managed through the
 connection have been timed out (Section 18.49).  A server SHOULD NOT
 close a connection immediately after responding to a session-level
 TEARDOWN request for the last RTSP session being controlled through
 the connection.  Instead, the server should wait for a reasonable
 amount of time for the client to receive and act upon the TEARDOWN

Schulzrinne, et al. Standards Track [Page 54] RFC 7826 RTSP 2.0 December 2016

 response and then initiate the connection closing.  The server SHOULD
 wait at least 10 seconds after sending the TEARDOWN response before
 closing the connection.
    This is to ensure that the client has time to issue a SETUP for a
    new session on the existing connection after having torn the last
    one down.  Ten seconds should give the client ample opportunity to
    get its message to the server.
 A server SHOULD NOT close the connection directly as a result of
 responding to a request with an error code.
    Certain error responses such as 460 (Only Aggregate Operation
    Allowed) (Section 17.4.24) are used for negotiating capabilities
    of a server with respect to content or other factors.  In such
    cases, it is inefficient for the server to close a connection on
    an error response.  Also, such behavior would prevent
    implementation of advanced or special types of requests or result
    in extra overhead for the client when testing for new features.
    On the other hand, keeping connections open after sending an error
    response poses a Denial-of-Service (DoS) security risk
    (Section 21).
 The server MAY close a connection if it receives an incomplete
 message and if the message is not completed within a reasonable
 amount of time.  It is RECOMMENDED that the server wait at least 10
 seconds for the completion of a message or for the next part of the
 message to arrive (which is an indication that the transport and the
 client are still alive).  Servers believing they are under attack or
 that are otherwise starved for resources during that event MAY
 consider using a shorter timeout.
 If a server closes a connection while the client is attempting to
 send a new request, the client will have to close its current
 connection, establish a new connection, and send its request over the
 new connection.
 An RTSP message SHOULD NOT be terminated by closing the connection.
 Such a message MAY be considered to be incomplete by the receiver and
 discarded.  An RTSP message is properly terminated as defined in
 Section 5.

Schulzrinne, et al. Standards Track [Page 55] RFC 7826 RTSP 2.0 December 2016

10.4. Timing Out Connections and RTSP Messages

 Receivers of a request (responders) SHOULD respond to requests in a
 timely manner even when a reliable transport such as TCP is used.
 Similarly, the sender of a request (requester) SHOULD wait for a
 sufficient time for a response before concluding that the responder
 will not be acting upon its request.
 A responder SHOULD respond to all requests within 5 seconds.  If the
 responder recognizes that the processing of a request will take
 longer than 5 seconds, it SHOULD send a 100 (Continue) response as
 soon as possible.  It SHOULD continue sending a 100 response every 5
 seconds thereafter until it is ready to send the final response to
 the requester.  After sending a 100 response, the responder MUST send
 a final response indicating the success or failure of the request.
 A requester SHOULD wait at least 10 seconds for a response before
 concluding that the responder will not be responding to its request.
 After receiving a 100 response, the requester SHOULD continue waiting
 for further responses.  If more than 10 seconds elapse without
 receiving any response, the requester MAY assume that the responder
 is unresponsive and abort the connection by closing the TCP
 connection.
 In some cases, multiple RTSP sessions share the same transport
 connection; abandoning a request and closing the connection may have
 significant impact on those other sessions.  First of all, other RTSP
 requests may have become queued up due to the request taking a long
 time to process.  Secondly, those sessions also lose the possibility
 to receive server-to-client requests.  To mitigate that situation,
 the RTSP client or server SHOULD establish a new connection and send
 any requests that are queued up or that haven't received a response
 on this new connection.  Thirdly, to ensure that the RTSP server
 knows which connection is valid for a particular RTSP session, the
 RTSP agent SHOULD send a keep-alive request, if no other request will
 be sent immediately for that RTSP session, for each RTSP session on
 the old connection.  The keep-alive request will normally be a
 SET_PARAMETER with a session header to inform the server that this
 agent cares about this RTSP session.
 A requester SHOULD wait longer than 10 seconds for a response if it
 is experiencing significant transport delays on its connection to the
 responder.  The requester is capable of determining the Round-Trip
 Time (RTT) of the request/response cycle using the Timestamp header
 (Section 18.53) in any RTSP request.

Schulzrinne, et al. Standards Track [Page 56] RFC 7826 RTSP 2.0 December 2016

    The 10-second wait was chosen for the following reasons.  It gives
    TCP time to perform a couple of retransmissions, even if operating
    on default values.  It is short enough that users may not abandon
    the process themselves.  However, it should be noted that 10
    seconds can be aggressive on certain types of networks.  The
    5-second value for 1xx messages is half the timeout giving a
    reasonable chance of successful delivery before timeout happens on
    the requester side.

10.5. Showing Liveness

 RTSP requires the client to periodically show its liveness to the
 server or the server may terminate any session state.  Several
 different protocol mechanism include in their usage a liveness proof
 from the client.  These mechanisms are RTSP requests with a Session
 header to the server; if RTP & RTCP is used for media data transport
 and the transport is established, the RTCP message proves liveness;
 or through any other used media-transport protocol capable of
 indicating liveness of the RTSP client.  It is RECOMMENDED that a
 client not wait to the last second of the timeout before trying to
 send a liveness message.  The RTSP message may take some time to
 arrive safely at the receiver, due to packet loss and TCP
 retransmissions.  To show liveness between RTSP requests being issued
 to accomplish other things, the following mechanisms can be used, in
 descending order of preference:
 RTCP: If RTP is used for media transport, RTCP SHOULD be used.  If
       RTCP is used to report transport statistics, it will
       necessarily also function as a keep-alive.  The server can
       determine the client by network address and port together with
       the fact that the client is reporting on the server's RTP
       sender sources (synchronization source (SSRCs)).  A downside of
       using RTCP is that it only gives statistical guarantees of
       reaching the server.  However, the probability of a false
       client timeout is so low that it can be ignored in most cases.
       For example, assume a session with a 60-second timeout and
       enough bitrate assigned to RTCP messages to send a message from
       client to server on average every 5 seconds.  That client has,
       for a network with 5% packet loss, a probability of failing to
       confirm liveness within the timeout interval for that session
       of 2.4*E-16.  Sessions with shorter timeouts, much higher
       packet loss, or small RTCP bandwidths SHOULD also implement one
       or more of the mechanisms below.

Schulzrinne, et al. Standards Track [Page 57] RFC 7826 RTSP 2.0 December 2016

 SET_PARAMETER:  When using SET_PARAMETER for keep-alives, a body
       SHOULD NOT be included.  This method is the RECOMMENDED RTSP
       method to use for a request intended only to perform keep-
       alives.  RTSP servers MUST support the SET_PARAMETER method, so
       that clients can always use this mechanism.
 GET_PARAMETER:  When using GET_PARAMETER for keep-alives, a body
       SHOULD NOT be included, dependent on implementation support in
       the server.  Use the OPTIONS method to determine if there is
       method support or simply try.
 OPTIONS:  This method is also usable, but it causes the server to
       perform more unnecessary processing and results in bigger
       responses than necessary for the task.  The reason is that the
       server needs to determine the capabilities associated with the
       media resource to correctly populate the Public and Allow
       headers.
 The timeout parameter of the Session header (Section 18.49) MAY be
 included in a SETUP response and MUST NOT be included in requests.
 The server uses it to indicate to the client how long the server is
 prepared to wait between RTSP commands or other signs of life before
 closing the session due to lack of activity (see Appendix B).  The
 timeout is measured in seconds, with a default of 60 seconds.  The
 length of the session timeout MUST NOT be changed in an established
 session.

10.6. Use of IPv6

 Explicit IPv6 [RFC2460] support was not present in RTSP 1.0.  RTSP
 2.0 has been updated for explicit IPv6 support.  Implementations of
 RTSP 2.0 MUST understand literal IPv6 addresses in URIs and RTSP
 headers.  Although the general URI format envisages potential future
 new versions of the literal IP address, usage of any such new version
 would require other modifications to the RTSP specification (e.g.,
 address fields in the Transport header (Section 18.54)).

10.7. Overload Control

 Overload in RTSP can occur when servers and proxies have insufficient
 resources to complete the processing of a request.  An improper
 handling of such an overload situation at proxies and servers can
 impact the operation of the RTSP deployment, and probably worsen the
 situation.  RTSP defines the 503 (Service Unavailable) response
 (Section 17.5.4) to let servers and proxies notify requesting proxies
 and RTSP clients about an overload situation.  In conjunction with

Schulzrinne, et al. Standards Track [Page 58] RFC 7826 RTSP 2.0 December 2016

 the Retry-After header (Section 18.44), the server or proxy can
 indicate the time after which the requesting entity can send another
 request to the proxy or server.
 There are two scopes of such 503 answers.  The first scope is for an
 established RTSP session, where the request resulting in the 503
 response as well as the response itself carries a Session header
 identifying the session that is suffering overload.  This response
 only applies to this particular session.  The other scope is the
 general RTSP server as identified by the host in the Request-URI.
 Such a 503 answer with any Retry-After header applies to all requests
 that are not session specific to that server, including a SETUP
 request intended to create a new RTSP session.
 Another scope for overload situations exists: the RTSP proxy.  To
 enable an RTSP proxy to signal that it is overloaded, or otherwise
 unavailable and unable to handle the request, a 553 response code has
 been defined with the meaning "Proxy Unavailable".  As with servers,
 there is a separation in response scopes between requests associated
 with existing RTSP sessions and requests to create new sessions or
 general proxy requests.
 Simply implementing and using the 503 (Service Unavailable) and 553
 (Proxy Unavailable) response codes is not sufficient for properly
 handling overload situations.  For instance, a simplistic approach
 would be to send the 503 response with a Retry-After header set to a
 fixed value.  However, this can cause a situation in which multiple
 RTSP clients again send requests to a proxy or server at roughly the
 same time, which may again cause an overload situation.  Another
 situation would be if the "old" overload situation is not yet
 resolved, i.e., the length indicated in the Retry-After header was
 too short for the overload situation to subside.
 An RTSP server or proxy in an overload situation must select the
 value of the Retry-After header carefully, bearing in mind its
 current load situation.  It is REQUIRED to increase the timeout
 period in proportion to the current load on the server, i.e., an
 increasing workload should result in an increased length of the
 indicated unavailability.  It is REQUIRED not to send the same value
 in the Retry-After header to all requesting proxies and clients, but
 to add a variation to the mean value of the Retry-After header.
 A more complex case may arise when a load-balancing RTSP proxy is in
 use.  This is the case when an RTSP proxy is used to select amongst a
 set of RTSP servers to handle the requests or when multiple server
 addresses are available for a given server name.  The proxy or client
 may receive a 503 (Service Unavailable) or 553 (Proxy Unavailable)
 response code from one of its RTSP servers or proxies, or a TCP

Schulzrinne, et al. Standards Track [Page 59] RFC 7826 RTSP 2.0 December 2016

 timeout (if the server is even unable to handle the request message).
 The proxy or client simply retries the other addresses or configured
 proxies, but it may also receive a 503 (Service Unavailable) or 553
 (Proxy Unavailable) response or TCP timeouts from those addresses.
 In such a situation, where none of the RTSP servers/proxies/addresses
 can handle the request, the RTSP agent has to wait before it can send
 any new requests to the RTSP server.  Any additional request to a
 specific address MUST be delayed according to the Retry-After headers
 received.  For addresses where no response was received or TCP
 timeout occurred, an initial wait timer SHOULD be set to 5 seconds.
 That timer MUST be doubled for each additional failure to connect or
 receive response until the value exceeds 30 minutes when the timer's
 mean value may be set to 30 minutes.  It is REQUIRED not to set the
 same value in the timer for each scheduling, but instead to add a
 variation to the mean value, resulting in picking a random value
 within the range of 0.5 to 1.5 times the mean value.

11. Capability Handling

 This section describes the available capability-handling mechanism
 that allows RTSP to be extended.  Extensions to this version of the
 protocol are basically done in two ways.  Firstly, new headers can be
 added.  Secondly, new methods can be added.  The capability-handling
 mechanism is designed to handle both cases.
 When a method is added, the involved parties can use the OPTIONS
 method to discover whether it is supported.  This is done by issuing
 an OPTIONS request to the other party.  Depending on the URI, it will
 either apply in regard to a certain media resource, the whole server
 in general, or simply the next hop.  The OPTIONS response MUST
 contain a Public header that declares all methods supported for the
 indicated resource.
 It is not necessary to use OPTIONS to discover support of a method,
 as the client could simply try the method.  If the receiver of the
 request does not support the method, it will respond with an error
 code indicating the method is either not implemented (501) or does
 not apply for the resource (405).  The choice between the two
 discovery methods depends on the requirements of the service.
 Feature tags are defined to handle functionality additions that are
 not new methods.  Each feature tag represents a certain block of
 functionality.  The amount of functionality that a feature tag
 represents can vary significantly.  For example, a feature tag can
 represent the functionality a single RTSP header provides.  Another
 feature tag can represent much more functionality, such as the
 "play.basic" feature tag (Section 11.1), which represents the minimal
 media delivery for playback implementation.

Schulzrinne, et al. Standards Track [Page 60] RFC 7826 RTSP 2.0 December 2016

 Feature tags are used to determine whether the client, server, or
 proxy supports the functionality that is necessary to achieve the
 desired service.  To determine support of a feature tag, several
 different headers can be used, each explained below:
 Supported:  This header is used to determine the complete set of
       functionality that both client and server have, in general, and
       is not dependent on a specific resource.  The intended usage is
       to determine before one needs to use a functionality that it is
       supported.  It can be used in any method, but OPTIONS is the
       most suitable as it simultaneously determines all methods that
       are implemented.  When sending a request, the requester
       declares all its capabilities by including all supported
       feature tags.  This results in the receiver learning the
       requester's feature support.  The receiver then includes its
       set of features in the response.
 Proxy-Supported:  This header is used in a similar fashion as the
       Supported header, but instead of giving the supported
       functionality of the client or server, it provides both the
       requester and the responder a view of the common functionality
       supported in general by all members of the proxy chain between
       the client and server; it does not depend on the resource.
       Proxies are required to add this header whenever the Supported
       header is present, but proxies may also add it independently of
       the requester.
 Require:  This header can be included in any request where the
       endpoint, i.e., the client or server, is required to understand
       the feature to correctly perform the request.  This can, for
       example, be a SETUP request, where the server is required to
       understand a certain parameter to be able to set up the media
       delivery correctly.  Ignoring this parameter would not have the
       desired effect and is not acceptable.  Therefore, the endpoint
       receiving a request containing a Require MUST negatively
       acknowledge any feature that it does not understand and not
       perform the request.  The response in cases where features are
       not supported is 551 (Option Not Supported).  Also, the
       features that are not supported are given in the Unsupported
       header in the response.
 Proxy-Require:  This header has the same purpose and behavior as
       Require except that it only applies to proxies and not the
       endpoint.  Features that need to be supported by both proxies
       and endpoints need to be included in both the Require and
       Proxy-Require header.

Schulzrinne, et al. Standards Track [Page 61] RFC 7826 RTSP 2.0 December 2016

 Unsupported:  This header is used in a 551 (Option Not Supported)
       error response, to indicate which features were not supported.
       Such a response is only the result of the usage of the Require
       or Proxy-Require headers where one or more features were not
       supported.  This information allows the requester to make the
       best of situations as it knows which features are not
       supported.

11.1. Feature Tag: play.basic

 An implementation supporting all normative parts of this
 specification for the setup and control of playback of media uses the
 feature tag "play.basic" to indicate this support.  The appendices
 (starting with letters) are not part of the functionality included in
 the feature tag unless the appendix is explicitly specified in a main
 section as being a required appendix.
    Note: This feature tag does not mandate any media delivery
    protocol, such as RTP.
    In RTSP 1.0, there was a minimal implementation section.  However,
    that was not consistent with the rest of the specification.  So,
    rather than making an attempt to explicitly enumerate the features
    for play.basic, this specification has to be taken as a whole and
    the necessary features normatively defined as being required are
    included.

12. Pipelining Support

 Pipelining is a general method to improve performance of request/
 response protocols by allowing the requesting agent to have more than
 one request outstanding and to send them over the same persistent
 connection.  For RTSP, where the relative order of requests will
 matter, it is important to maintain the order of the requests.
 Because of this, the responding agent MUST process the incoming
 requests in their sending order.  The sending order can be determined
 by the CSeq header and its sequence number.  For TCP, the delivery
 order will be the same, between two agents, as the sending order.
 The processing of the request MUST also have been finished before
 processing the next request from the same agent.  The responses MUST
 be sent in the order the requests were processed.
 RTSP 2.0 has extended support for pipelining beyond the capabilities
 in RTSP 1.0.  As a major improvement, all requests involved in
 setting up and initiating media delivery can now be pipelined,
 indicated by the Pipelined-Request header (see Section 18.33).  This
 header allows a client to request that two or more requests be
 processed in the same RTSP session context that the first request

Schulzrinne, et al. Standards Track [Page 62] RFC 7826 RTSP 2.0 December 2016

 creates.  In other words, a client can request that two or more media
 streams be set up and then played without needing to wait for a
 single response.  This speeds up the initial start-up time for an
 RTSP session by at least one RTT.
 If a pipelined request builds on the successful completion of one or
 more prior requests, the requester must verify that all requests were
 executed as expected.  A common example will be two SETUP requests
 and a PLAY request.  In case one of the SETUP requests fails
 unexpectedly, the PLAY request can still be successfully executed.
 However, the resulting presentation will not be as expected by the
 requesting client, as only a single media instead of two will be
 played.  In this case, the client can send a PAUSE request, correct
 the failing SETUP request, and then request it be played.

13. Method Definitions

 The method indicates what is to be performed on the resource
 identified by the Request-URI.  The method name is case sensitive.
 New methods may be defined in the future.  Method names MUST NOT
 start with a $ character (decimal 36) and MUST be a token as defined
 by the ABNF [RFC5234] in Section 20.  The methods are summarized in
 Table 7.

Schulzrinne, et al. Standards Track [Page 63] RFC 7826 RTSP 2.0 December 2016

  +---------------+-----------+--------+-------------+-------------+
  | method        | direction | object | Server req. | Client req. |
  +---------------+-----------+--------+-------------+-------------+
  | DESCRIBE      | C -> S    | P,S    | recommended | recommended |
  |               |           |        |             |             |
  | GET_PARAMETER | C -> S    | P,S    | optional    | optional    |
  |               |           |        |             |             |
  |               | S -> C    | P,S    | optional    | optional    |
  |               |           |        |             |             |
  | OPTIONS       | C -> S    | P,S    | required    | required    |
  |               |           |        |             |             |
  |               | S -> C    | P,S    | optional    | optional    |
  |               |           |        |             |             |
  | PAUSE         | C -> S    | P,S    | required    | required    |
  |               |           |        |             |             |
  | PLAY          | C -> S    | P,S    | required    | required    |
  |               |           |        |             |             |
  | PLAY_NOTIFY   | S -> C    | P,S    | required    | required    |
  |               |           |        |             |             |
  | REDIRECT      | S -> C    | P,S    | optional    | required    |
  |               |           |        |             |             |
  | SETUP         | C -> S    | S      | required    | required    |
  |               |           |        |             |             |
  | SET_PARAMETER | C -> S    | P,S    | required    | optional    |
  |               |           |        |             |             |
  |               | S -> C    | P,S    | optional    | optional    |
  |               |           |        |             |             |
  | TEARDOWN      | C -> S    | P,S    | required    | required    |
  |               |           |        |             |             |
  |               | S -> C    | P      | required    | required    |
  +---------------+-----------+--------+-------------+-------------+
                   Table 7: Overview of RTSP Methods
    Note on Table 7: This table covers RTSP methods, their direction,
    and on what objects (P: presentation, S: stream) they operate.
    Further, it indicates whether a server or a client implementation
    is required (mandatory), recommended, or optional.
    Further note on Table 7: the GET_PARAMETER is optional.  For
    example, a fully functional server can be built to deliver media
    without any parameters.  However, SET_PARAMETER is required, i.e.,
    mandatory to implement for the server; this is due to its usage
    for keep-alive.  PAUSE is required because it is the only way of
    leaving the Play state without terminating the whole session.

Schulzrinne, et al. Standards Track [Page 64] RFC 7826 RTSP 2.0 December 2016

 If an RTSP agent does not support a particular method, it MUST return
 a 501 (Not Implemented) response code and the requesting RTSP agent,
 in turn, SHOULD NOT try this method again for the given agent/
 resource combination.  An RTSP proxy whose main function is to log or
 audit and not modify transport or media handling in any way MAY
 forward RTSP messages with unknown methods.  Note that the proxy
 still needs to perform the minimal required processing, like adding
 the Via header.

13.1. OPTIONS

 The semantics of the RTSP OPTIONS method is similar to that of the
 HTTP OPTIONS method described in Section 4.3.7 of [RFC7231].
 However, in RTSP, OPTIONS is bidirectional in that a client can send
 the request to a server and vice versa.  A client MUST implement the
 capability to send an OPTIONS request and a server or a proxy MUST
 implement the capability to respond to an OPTIONS request.  In
 addition to this "MUST-implement" functionality, clients, servers and
 proxies MAY provide support both for sending OPTIONS requests and for
 generating responses to the requests.
 An OPTIONS request may be issued at any time.  Such a request does
 not modify the session state.  However, it may prolong the session
 lifespan (see below).  The URI in an OPTIONS request determines the
 scope of the request and the corresponding response.  If the Request-
 URI refers to a specific media resource on a given host, the scope is
 limited to the set of methods supported for that media resource by
 the indicated RTSP agent.  A Request-URI with only the host address
 limits the scope to the specified RTSP agent's general capabilities
 without regard to any specific media.  If the Request-URI is an
 asterisk ("*"), the scope is limited to the general capabilities of
 the next hop (i.e., the RTSP agent in direct communication with the
 request sender).
 Regardless of the scope of the request, the Public header MUST always
 be included in the OPTIONS response, listing the methods that are
 supported by the responding RTSP agent.  In addition, if the scope of
 the request is limited to a media resource, the Allow header MUST be
 included in the response to enumerate the set of methods that are
 allowed for that resource unless the set of methods completely
 matches the set in the Public header.  If the given resource is not
 available, the RTSP agent SHOULD return an appropriate response code,
 such as 3rr or 4xx.  The Supported header MAY be included in the
 request to query the set of features that are supported by the
 responding RTSP agent.

Schulzrinne, et al. Standards Track [Page 65] RFC 7826 RTSP 2.0 December 2016

 The OPTIONS method can be used to keep an RTSP session alive.
 However, this is not the preferred way of session keep-alive
 signaling; see Section 18.49.  An OPTIONS request intended for
 keeping alive an RTSP session MUST include the Session header with
 the associated session identifier.  Such a request SHOULD also use
 the media or the aggregated control URI as the Request-URI.
 Example:
   C->S:  OPTIONS rtsp://server.example.com RTSP/2.0
          CSeq: 1
          User-Agent: PhonyClient/1.2
          Proxy-Require: gzipped-messages
          Supported: play.basic
   S->C:  RTSP/2.0 200 OK
          CSeq: 1
          Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS
          Supported: play.basic, setup.rtp.rtcp.mux, play.scale
          Server: PhonyServer/1.1
 Note that the "gzipped-messages" feature tag in the Proxy-Require is
 a fictitious feature.

13.2. DESCRIBE

 The DESCRIBE method is used to retrieve the description of a
 presentation or media object from a server.  The Request-URI of the
 DESCRIBE request identifies the media resource of interest.  The
 client MAY include the Accept header in the request to list the
 description formats that it understands.  The server MUST respond
 with a description of the requested resource and return the
 description in the message body of the response, if the DESCRIBE
 method request can be successfully fulfilled.  The DESCRIBE reply-
 response pair constitutes the media initialization phase of RTSP.
 The DESCRIBE response SHOULD contain all media initialization
 information for the resource(s) that it describes.  Servers SHOULD
 NOT use the DESCRIBE response as a means of media indirection by
 having the description point at another server; instead, using the
 3rr responses is RECOMMENDED.
    By forcing a DESCRIBE response to contain all media initialization
    information for the set of streams that it describes, and
    discouraging the use of DESCRIBE for media indirection, any
    looping problems can be avoided that might have resulted from
    other approaches.

Schulzrinne, et al. Standards Track [Page 66] RFC 7826 RTSP 2.0 December 2016

 Example:
   C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/2.0
         CSeq: 312
         User-Agent: PhonyClient/1.2
         Accept: application/sdp, application/example
   S->C: RTSP/2.0 200 OK
         CSeq: 312
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer/1.1
         Content-Base: rtsp://server.example.com/fizzle/foo/
         Content-Type: application/sdp
         Content-Length: 358
         v=0
         o=MNobody 2890844526 2890842807 IN IP4 192.0.2.46
         s=SDP Seminar
         i=A Seminar on the session description protocol
         u=http://www.example.com/lectures/sdp.ps
         e=seminar@example.com (Seminar Management)
         c=IN IP4 0.0.0.0
         a=control:*
         t=2873397496 2873404696
         m=audio 3456 RTP/AVP 0
         a=control:audio
         m=video 2232 RTP/AVP 31
         a=control:video
 Media initialization is a requirement for any RTSP-based system, but
 the RTSP specification does not dictate that this is required to be
 done via the DESCRIBE method.  There are three ways that an RTSP
 client may receive initialization information:
 o  via an RTSP DESCRIBE request
 o  via some other protocol (HTTP, email attachment, etc.)
 o  via some form of user interface
 If a client obtains a valid description from an alternate source, the
 client MAY use this description for initialization purposes without
 issuing a DESCRIBE request for the same media.  The client should use
 any MTag to either validate the presentation description or make the
 session establishment conditional on being valid.

Schulzrinne, et al. Standards Track [Page 67] RFC 7826 RTSP 2.0 December 2016

 It is RECOMMENDED that minimal servers support the DESCRIBE method,
 and highly recommended that minimal clients support the ability to
 act as "helper applications" that accept a media initialization file
 from a user interface, or other means that are appropriate to the
 operating environment of the clients.

13.3. SETUP

 The description below uses the following states in a protocol state
 machine that is related to a specific session when that session has
 been created.  The state transitions are driven by protocol
 interactions.  For additional information about the state machine,
 see Appendix B.
 Init: Initial state.  No session exists.
 Ready:  Session is ready to start playing.
 Play: Session is playing, i.e., sending media-stream data in the
       direction S->C.
 The SETUP request for a URI specifies the transport mechanism to be
 used for the streamed media.  The SETUP method may be used in two
 different cases, namely, creating an RTSP session and changing the
 transport parameters of media streams that are already set up.  SETUP
 can be used in all three states, Init, Ready, and Play, to change the
 transport parameters.  Additionally, Init and Ready can also be used
 for the creation of the RTSP session.  The usage of the SETUP method
 in the Play state to add a media resource to the session is
 unspecified.
 The Transport header, see Section 18.54, specifies the media-
 transport parameters acceptable to the client for data transmission;
 the response will contain the transport parameters selected by the
 server.  This allows the client to enumerate, in descending order of
 preference, the transport mechanisms and parameters acceptable to it,
 so the server can select the most appropriate.  It is expected that
 the session description format used will enable the client to select
 a limited number of possible configurations that are offered as
 choices to the server.  All transport-related parameters SHALL be
 included in the Transport header; the use of other headers for this
 purpose is NOT RECOMMENDED due to middleboxes, such as firewalls or
 NATs.
 For the benefit of any intervening firewalls, a client MUST indicate
 the known transport parameters, even if it has no influence over
 these parameters, for example, where the server advertises a fixed-
 multicast address as destination.

Schulzrinne, et al. Standards Track [Page 68] RFC 7826 RTSP 2.0 December 2016

    Since SETUP includes all transport initialization information,
    firewalls and other intermediate network devices (which need this
    information) are spared the more arduous task of parsing the
    DESCRIBE response, which has been reserved for media
    initialization.
 The client MUST include the Accept-Ranges header in the request,
 indicating all supported unit formats in the Range header.  This
 allows the server to know which formats it may use in future session-
 related responses, such as a PLAY response without any range in the
 request.  If the client does not support a time format necessary for
 the presentation, the server MUST respond using 456 (Header Field Not
 Valid for Resource) and include the Accept-Ranges header with the
 range unit formats supported for the resource.
 In a SETUP response, the server MUST include the Accept-Ranges header
 (see Section 18.5) to indicate which time formats are acceptable to
 use for this media resource.
 The SETUP 200 OK response MUST include the Media-Properties header
 (see Section 18.29).  The combination of the parameters of the Media-
 Properties header indicates the nature of the content present in the
 session (see also Section 4.7).  For example, a live stream with time
 shifting is indicated by
 o  Random access set to Random-Access,
 o  Content Modifications set to Time-Progressing, and
 o  Retention set to Time-Duration (with specific recording window
    time value).
 The SETUP 200 OK response MUST include the Media-Range header (see
 Section 18.30) if the media is Time-Progressing.

Schulzrinne, et al. Standards Track [Page 69] RFC 7826 RTSP 2.0 December 2016

 A basic example for SETUP:
   C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
         CSeq: 302
         Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589",
                    RTP/AVP/TCP;unicast;interleaved=0-1
         Accept-Ranges: npt, clock
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 302
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer/1.1
         Session: QKyjN8nt2WqbWw4tIYof52;timeout=60
         Transport: RTP/AVP;unicast;dest_addr="192.0.2.53:4588"/
                    "192.0.2.53:4589"; src_addr="198.51.100.241:6256"/
                    "198.51.100.241:6257"; ssrc=2A3F93ED
         Accept-Ranges: npt
         Media-Properties: Random-Access=3.2, Time-Progressing,
                           Time-Duration=3600.0
         Media-Range: npt=0-2893.23
 In the above example, the client wants to create an RTSP session
 containing the media resource "rtsp://example.com/foo/bar/baz.rm".
 The transport parameters acceptable to the client are either RTP/AVP/
 UDP (UDP per default) to be received on client port 4588 and 4589 at
 the address the RTSP setup connection comes from or RTP/AVP
 interleaved on the RTSP control channel.  The server selects the
 RTP/AVP/UDP transport and adds the address and ports it will send and
 receive RTP and RTCP from, and the RTP SSRC that will be used by the
 server.
 The server MUST generate a session identifier in response to a
 successful SETUP request unless a SETUP request to a server includes
 a session identifier or a Pipelined-Requests header referencing an
 existing session context.  In that latter case, the server MUST
 bundle this SETUP request into the existing session (aggregated
 session) or return a 459 (Aggregate Operation Not Allowed) error code
 (see Section 17.4.23).  An aggregate control URI MUST be used to
 control an aggregated session.  This URI MUST be different from the
 stream control URIs of the individual media streams included in the
 aggregate (see Section 13.4.2 for aggregated sessions and for the
 particular URIs see Appendix D.1.1).  The aggregate control URI is to
 be specified by the session description if the server supports
 aggregated control and aggregated control is desired for the session.

Schulzrinne, et al. Standards Track [Page 70] RFC 7826 RTSP 2.0 December 2016

 However, even if aggregated control is offered, the client MAY choose
 not to set up the session in aggregated control.  If an aggregate
 control URI is not specified in the session description, it is
 normally an indication that non-aggregated control should be used.
 The SETUP of media streams in an aggregate that has not been given an
 aggregated control URI is unspecified.
    While the session ID sometimes carries enough information for
    aggregate control of a session, the aggregate control URI is still
    important for some methods such as SET_PARAMETER where the control
    URI enables the resource in question to be easily identified.  The
    aggregate control URI is also useful for proxies, enabling them to
    route the request to the appropriate server, and for logging,
    where it is useful to note the actual resource on which a request
    was operating.
 A session will exist until it is either removed by a TEARDOWN request
 or is timed out by the server.  The server MAY remove a session that
 has not demonstrated liveness signs from the client(s) within a
 certain timeout period.  The default timeout value is 60 seconds; the
 server MAY set this to a different value and indicate so in the
 timeout field of the Session header in the SETUP response.  For
 further discussion, see Section 18.49.  Signs of liveness for an RTSP
 session include any RTSP requests from a client that contain a
 Session header with the ID for that session, as well as RTCP sender
 or receiver reports if RTP is used to transport the underlying media
 stream.  RTCP sender reports may, for example, be received in session
 where the server is invited into a conference session and are thus
 valid as a liveness indicator.
 If a SETUP request on a session fails for any reason, the session
 state, as well as transport and other parameters for associated
 streams, MUST remain unchanged from their values as if the SETUP
 request had never been received by the server.

13.3.1. Changing Transport Parameters

 A client MAY issue a SETUP request for a stream that is already set
 up or playing in the session to change transport parameters, which a
 server MAY allow.  If it does not allow the changing of parameters,
 it MUST respond with error 455 (Method Not Valid in This State).  The
 reasons to support changing transport parameters include allowing
 application-layer mobility and flexibility to utilize the best
 available transport as it becomes available.  If a client receives a
 455 error when trying to change transport parameters while the server
 is in Play state, it MAY try to put the server in Ready state using
 PAUSE before issuing the SETUP request again.  If that also fails,

Schulzrinne, et al. Standards Track [Page 71] RFC 7826 RTSP 2.0 December 2016

 the changing of transport parameters will require that the client
 perform a TEARDOWN of the affected media and then set it up again.
 For an aggregated session, not tearing down all the media at the same
 time will avoid the creation of a new session.
 All transport parameters MAY be changed.  However, the primary usage
 expected is to either change the transport protocol completely, like
 switching from Interleaved TCP mode to UDP or vice versa, or to
 change the delivery address.
 In a SETUP response for a request to change the transport parameters
 while in Play state, the server MUST include the Range header to
 indicate at what point the new transport parameters will be used.
 Further, if RTP is used for delivery, the server MUST also include
 the RTP-Info header to indicate at what timestamp and RTP sequence
 number the change will take place.  If both RTP-Info and Range are
 included in the response, the "rtp_time" parameter and start point in
 the Range header MUST be for the corresponding time, i.e., be used in
 the same way as for PLAY to ensure the correct synchronization
 information is available.
 If the transport-parameters change that happened while in Play state
 results in a change of synchronization-related information, for
 example, changing RTP SSRC, the server MUST include the necessary
 synchronization information in the SETUP response.  However, the
 server SHOULD avoid changing the synchronization information if
 possible.

13.4. PLAY

 This section describes the usage of the PLAY method in general, for
 aggregated sessions, and in different usage scenarios.

13.4.1. General Usage

 The PLAY method tells the server to start sending data via the
 mechanism specified in SETUP and which part of the media should be
 played out.  PLAY requests are valid when the session is in Ready or
 Play state.  A PLAY request MUST include a Session header to indicate
 to which session the request applies.
 Upon receipt of the PLAY request, the server MUST position the normal
 play time to the beginning of the range specified in the received
 Range header, within the limits of the media resource and in
 accordance with the Seek-Style header (Section 18.47).  It MUST
 deliver stream data until the end of the range if given, until a new
 PLAY request is received, until a PAUSE request (Section 13.5) is
 received, or until the end of the media is reached.  If no Range

Schulzrinne, et al. Standards Track [Page 72] RFC 7826 RTSP 2.0 December 2016

 header is present in the PLAY request, the server SHALL play from
 current pause point until the end of media.  The pause point defaults
 at session start to the beginning of the media.  For media that is
 time-progressing and has no retention, the pause point will always be
 set equal to NPT "now", i.e., the current delivery point.  The pause
 point may also be set to a particular point in the media by the PAUSE
 method; see Section 13.6.  The pause point for media that is
 currently playing is equal to the current media position.  For time-
 progressing media with time-limited retention, if the pause point
 represents a position that is older than what is retained by the
 server, the pause point will be moved to the oldest retained
 position.
 What range values are valid depends on the type of content.  For
 content that isn't time-progressing, the range value is valid if the
 given range is part of any media within the aggregate.  In other
 words, the valid media range for the aggregate is the union of all of
 the media components in the aggregate.  If a given range value points
 outside of the media, the response MUST be the 457 (Invalid Range)
 error code and include the Media-Range header (Section 18.30) with
 the valid range for the media.  Except for time-progressing content
 where the client requests a start point prior to what is retained,
 the start point is adjusted to the oldest retained content.  For a
 start point that is beyond the media front edge, i.e., beyond the
 current value for "now", the server SHALL adjust the start value to
 the current front edge.  The Range header's stop point value may
 point beyond the current media edge.  In that case, the server SHALL
 deliver media from the requested (and possibly adjusted) start point
 until the first of either the provided stop point or the end of the
 media.  Please note that if one simply wants to play from a
 particular start point until the end of media, using a Range header
 with an implicit stop point is RECOMMENDED.
 If a client requests to start playing at the end of media, either
 explicitly with a Range header or implicitly with a pause point that
 is at the end of media, a 457 (Invalid Range) error MUST be sent and
 include the Media-Range header (Section 18.30).  It is specified
 below that the Range header also must be included in the response and
 that it will carry the pause point in the media, in the case of the
 session being in Ready State.  Note that this also applies if the
 pause point or requested start point is at the beginning of the media
 and a Scale header (Section 18.46) is included with a negative value
 (playing backwards).
 For media with random access properties, a client may indicate which
 policy for start point selection the server should use.  This is done
 by including the Seek-Style header (Section 18.47) in the PLAY

Schulzrinne, et al. Standards Track [Page 73] RFC 7826 RTSP 2.0 December 2016

 request.  The Seek-Style applied will affect the content of the Range
 header as it will be adjusted to indicate from what point the media
 actually is delivered.
 A client desiring to play the media from the beginning MUST send a
 PLAY request with a Range header pointing at the beginning, e.g.,
 "npt=0-".  If a PLAY request is received without a Range header and
 media delivery has stopped at the end, the server SHOULD respond with
 a 457 (Invalid Range) error response.  In that response, the current
 pause point MUST be included in a Range header.
 All range specifiers in this specification allow for ranges with an
 implicit start point (e.g., "npt=-30").  When used in a PLAY request,
 the server treats this as a request to start or resume delivery from
 the current pause point, ending at the end time specified in the
 Range header.  If the pause point is located later than the given end
 value, a 457 (Invalid Range) response MUST be returned.
 The example below will play seconds 10 through 25.  It also requests
 that the server deliver media from the first random access point
 prior to the indicated start point.
   C->S: PLAY rtsp://audio.example.com/audio RTSP/2.0
         CSeq: 835
         Session: ULExwZCXh2pd0xuFgkgZJW
         Range: npt=10-25
         Seek-Style: RAP
         User-Agent: PhonyClient/1.2
 Servers MUST include a Range header in any PLAY response, even if no
 Range header was present in the request.  The response MUST use the
 same format as the request's Range header contained.  If no Range
 header was in the request, the format used in any previous PLAY
 request within the session SHOULD be used.  If no format has been
 indicated in a previous request, the server MAY use any time format
 supported by the media and indicated in the Accept-Ranges header in
 the SETUP request.  It is RECOMMENDED that NPT is used if supported
 by the media.
 For any error response to a PLAY request, the server's response
 depends on the current session state.  If the session is in Ready
 state, the current pause point is returned using a Range header with
 the pause point as the explicit start point and an implicit stop
 point.  For time-progressing content, where the pause-point moves
 with real-time due to limited retention, the current pause point is
 returned.  For sessions in Play state, the current playout point and

Schulzrinne, et al. Standards Track [Page 74] RFC 7826 RTSP 2.0 December 2016

 the remaining parts of the range request are returned.  For any media
 with retention longer than 0 seconds, the currently valid Media-Range
 header SHALL also be included in the response.
 A PLAY response MAY include a header carrying synchronization
 information.  As the information necessary is dependent on the media-
 transport format, further rules specifying the header and its usage
 are needed.  For RTP the RTP-Info header is specified, see
 Section 18.45, and used in the following example.
 Here is a simple example for a single audio stream where the client
 requests the media starting from 3.52 seconds and to the end.  The
 server sends a 200 OK response with the actual play time, which is 10
 ms prior (3.51), and the RTP-Info header that contains the necessary
 parameters for the RTP stack.
 C->S: PLAY rtsp://example.com/audio RTSP/2.0
       CSeq: 836
       Session: ULExwZCXh2pd0xuFgkgZJW
       Range: npt=3.52-
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 836
       Date: Thu, 23 Jan 1997 15:35:06 GMT
       Server: PhonyServer/1.0
       Range: npt=3.51-324.39
       Seek-Style: First-Prior
           Session: ULExwZCXh2pd0xuFgkgZJW
       RTP-Info:url="rtsp://example.com/audio"
          ssrc=0D12F123:seq=14783;rtptime=2345962545
 S->C: RTP Packet TS=2345962545 => NPT=3.51
       Media duration=0.16 seconds
 The server replies with the actual start point that will be
 delivered.  This may differ from the requested range if alignment of
 the requested range to valid frame boundaries is required for the
 media source.  Note that some media streams in an aggregate may need
 to be delivered from even earlier points.  Also, some media formats
 have a very long duration per individual data unit; therefore, it
 might be necessary for the client to parse the data unit, and select
 where to start.  The server SHALL also indicate which policy it uses
 for selecting the actual start point by including a Seek-Style
 header.

Schulzrinne, et al. Standards Track [Page 75] RFC 7826 RTSP 2.0 December 2016

 In the following example, the client receives the first media packet
 that stretches all the way up and past the requested playtime.  Thus,
 it is the client's decision whether to render to the user the time
 between 3.52 and 7.05 or to skip it.  In most cases, it is probably
 most suitable not to render that time period.
 C->S: PLAY rtsp://example.com/audio RTSP/2.0
       CSeq: 836
       Session: ZGGyCJOs8xaLkdNK2dmxQO
       Range: npt=7.05-
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 836
       Date: Thu, 23 Jan 1997 15:35:06 GMT
       Server: PhonyServer/1.0
           Session: ZGGyCJOs8xaLkdNK2dmxQO
       Range: npt=3.52-
       Seek-Style: First-Prior
       RTP-Info:url="rtsp://example.com/audio"
          ssrc=0D12F123:seq=14783;rtptime=2345962545
 S->C: RTP Packet TS=2345962545 => NPT=3.52
       Duration=4.15 seconds
 After playing the desired range, the presentation does NOT change to
 the Ready state, media delivery simply stops.  If it is necessary to
 put the stream into the Ready state, a PAUSE request MUST be issued.
 A PLAY request while the stream is still in the Play state is legal
 and can be issued without an intervening PAUSE request.  Such a
 request MUST replace the current PLAY action with the new one
 requested, i.e., being handled in the same way as if as the request
 was received in Ready state.  In the case that the range in the Range
 header has an implicit start time ("-endtime"), the server MUST
 continue to play from where it currently was until the specified
 endpoint.  This is useful to change the end to at another point than
 in the previous request.
 The following example plays the whole presentation starting at SMPTE
 time code 0:10:20 until the end of the clip.  Note: the RTP-Info
 headers have been broken into several lines, where subsequent lines
 start with whitespace as allowed by the syntax.

Schulzrinne, et al. Standards Track [Page 76] RFC 7826 RTSP 2.0 December 2016

 C->S: PLAY rtsp://audio.example.com/twister.en RTSP/2.0
       CSeq: 833
       Session: N465Wvsv0cjUy6tLqINkcf
       Range: smpte=0:10:20-
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 833
       Date: Thu, 23 Jan 1997 15:35:06 GMT
       Session: N465Wvsv0cjUy6tLqINkcf
       Server: PhonyServer/1.0
       Range: smpte=0:10:22-0:15:45
       Seek-Style: Next
       RTP-Info:url="rtsp://example.com/twister.en"
          ssrc=0D12F123:seq=14783;rtptime=2345962545
 For playing back a recording of a live presentation, it may be
 desirable to use clock units:
 C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/2.0
       CSeq: 835
       Session: N465Wvsv0cjUy6tLqINkcf
       Range: clock=19961108T142300Z-19961108T143520Z
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 835
       Date: Thu, 23 Jan 1997 15:35:06 GMT
       Session: N465Wvsv0cjUy6tLqINkcf
       Server: PhonyServer/1.0
       Range: clock=19961108T142300Z-19961108T143520Z
       Seek-Style: Next
       RTP-Info:url="rtsp://example.com/meeting.en"
          ssrc=0D12F123:seq=53745;rtptime=484589019

13.4.2. Aggregated Sessions

 PLAY requests can operate on sessions controlling a single media
 stream and on aggregated sessions controlling multiple media streams.
 In an aggregated session, the PLAY request MUST contain an aggregated
 control URI.  A server MUST respond with a 460 error (Only Aggregate
 Operation Allowed) if the client PLAY Request-URI is for a single
 media.  The media in an aggregate MUST be played in sync.  If a
 client wants individual control of the media, it needs to use
 separate RTSP sessions for each media.

Schulzrinne, et al. Standards Track [Page 77] RFC 7826 RTSP 2.0 December 2016

 For aggregated sessions where the initial SETUP request (creating a
 session) is followed by one or more additional SETUP requests, a PLAY
 request MAY be pipelined (Section 12) after those additional SETUP
 requests without awaiting their responses.  This procedure can reduce
 the delay from the start of session establishment until media playout
 has started with one RTT.  However, a client needs to be aware that
 using this procedure will result in the playout of the server state
 established at the time of processing the PLAY, i.e., after the
 processing of all the requests prior to the PLAY request in the
 pipeline.  This state may not be the intended one due to failure of
 any of the prior requests.  A client can easily determine this based
 on the responses from those requests.  In case of failure, the client
 can halt the media playout using PAUSE and try to establish the
 intended state again before issuing another PLAY request.

13.4.3. Updating Current PLAY Requests

 Clients can issue PLAY requests while the stream is in Play state and
 thus updating their request.
 The important difference compared to a PLAY request in Ready state is
 the handling of the current play point and how the Range header in
 the request is constructed.  The session is actively playing media
 and the play point will be moving, making the exact time a request
 will take effect hard to predict.  Depending on how the PLAY header
 appears, two different cases exist: total replacement or
 continuation.  A total replacement is signaled by having the first
 range specification have an explicit start value, e.g., "npt=45-" or
 "npt=45-60", in which case the server stops playout at the current
 playout point and then starts delivering media according to the Range
 header.  This is equivalent to having the client first send a PAUSE
 and then a new PLAY request that isn't based on the pause point.  In
 the case of continuation, the first range specifier has an implicit
 start point and an explicit stop value (Z), e.g., "npt=-60", which
 indicate that it MUST convert the range specifier being played prior
 to this PLAY request (X to Y) into (X to Z) and continue as if this
 was the request originally played.  If the current delivery point is
 beyond the stop point, the server SHALL immediately pause delivery.
 As the request has been completed successfully, it shall be responded
 to with a 200 OK response.  A PLAY_NOTIFY with end-of-stream is also
 sent to indicate the actual stop point.  The pause point is set to
 the requested stop point.
 The following is an example of this behavior: The server has received
 requests to play ranges 10 to 15.  If the new PLAY request arrives at
 the server 4 seconds after the previous one, it will take effect

Schulzrinne, et al. Standards Track [Page 78] RFC 7826 RTSP 2.0 December 2016

 while the server still plays the first range (10-15).  The server
 changes the current play to continue to 25 seconds, i.e., the
 equivalent single request would be PLAY with "range: npt=10-25".
   C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 834
         Session: apzA8LnjQ5KWTdw0kUkiRh
         Range: npt=10-15
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 834
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Session: apzA8LnjQ5KWTdw0kUkiRh
         Server: PhonyServer/1.0
         Range: npt=10-15
         Seek-Style: Next
         RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=5712;rtptime=934207921,
                 url="rtsp://example.com/fizzle/videotrack"
                 ssrc=789DAF12:seq=57654;rtptime=2792482193
   C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 835
         Session: apzA8LnjQ5KWTdw0kUkiRh
         Range: npt=-25
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 835
         Date: Thu, 23 Jan 1997 15:35:09 GMT
         Session: apzA8LnjQ5KWTdw0kUkiRh
         Server: PhonyServer/1.0
         Range: npt=14-25
         Seek-Style: Next
         RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=5712;rtptime=934239921,
                 url="rtsp://example.com/fizzle/videotrack"
                 ssrc=789DAF12:seq=57654;rtptime=2792842193
 A common use of a PLAY request while in Play state is changing the
 scale of the media, i.e., entering or leaving fast forward or fast
 rewind.  The client can issue an updating PLAY request that is either
 a continuation or a complete replacement, as discussed above this
 section.  Below is an example of a client that is requesting a fast
 forward (scale = 2) without giving a stop point and then a change
 from fast forward to regular playout (scale = 1).  In the second PLAY

Schulzrinne, et al. Standards Track [Page 79] RFC 7826 RTSP 2.0 December 2016

 request, the time is set explicitly to be wherever the server
 currently plays out (npt=now-) and the server responds with the
 actual playback point where the new scale actually takes effect
 (npt=02:17:27.144-).
   C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 2034
         Session: apzA8LnjQ5KWTdw0kUkiRh
         Range: npt=now-
         Scale: 2.0
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 2034
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Session: apzA8LnjQ5KWTdw0kUkiRh
         Server: PhonyServer/1.0
         Range: npt=02:17:21.394-
         Seek-Style: Next
         Scale: 2.0
         RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=5712;rtptime=934207921,
                 url="rtsp://example.com/fizzle/videotrack"
                 ssrc=789DAF12:seq=57654;rtptime=2792482193
 [playing in fast forward and now returning to scale = 1]
   C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 2035
         Session: apzA8LnjQ5KWTdw0kUkiRh
         Range: npt=now-
         Scale: 1.0
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 2035
         Date: Thu, 23 Jan 1997 15:35:09 GMT
         Session: apzA8LnjQ5KWTdw0kUkiRh
         Server: PhonyServer/1.0
         Range: npt=02:17:27.144-
         Seek-Style: Next
         Scale: 1.0
         RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=5712;rtptime=934239921,
                 url="rtsp://example.com/fizzle/videotrack"
                 ssrc=789DAF12:seq=57654;rtptime=2792842193

Schulzrinne, et al. Standards Track [Page 80] RFC 7826 RTSP 2.0 December 2016

13.4.4. Playing On-Demand Media

 On-demand media is indicated by the content of the Media-Properties
 header in the SETUP response when (see also Section 18.29):
 o  the Random Access property is set to Random-Access;
 o  the Content Modifications property is set to Immutable;
 o  the Retention property is set to Unlimited or Time-Limited.
 Playing on-demand media follows the general usage as described in
 Section 13.4.1.

13.4.5. Playing Dynamic On-Demand Media

 Dynamic on-demand media is indicated by the content of the Media-
 Properties header in the SETUP response when (see also
 Section 18.29):
 o  the Random Access property is set to Random-Access;
 o  the Content Modifications property is set to Dynamic;
 o  the Retention property is set to Unlimited or Time-Limited.
 Playing on-demand media follows the general usage as described in
 Section 13.4.1 as long as the media has not been changed.
 There are two ways for the client to be informed about changes of
 media resources in Play state.  The first being that the client will
 receive a PLAY_NOTIFY request with the Notify-Reason header set to
 media-properties-update (see Section 13.5.2).  The client can use the
 value of the Media-Range header to decide further actions, if the
 Media-Range header is present in the PLAY_NOTIFY request.  The second
 way is that the client issues a GET_PARAMETER request without a body
 but including a Media-Range header.  The 200 OK response MUST include
 the current Media-Range header (see Section 18.30).

13.4.6. Playing Live Media

 Live media is indicated by the content of the Media-Properties header
 in the SETUP response when (see also Section 18.29):
 o  the Random Access property is set to No-Seeking;
 o  the Content Modifications property is set to Time-Progressing;

Schulzrinne, et al. Standards Track [Page 81] RFC 7826 RTSP 2.0 December 2016

 o  the Retention property's Time-Duration is set to 0.0.
 For live media, the SETUP 200 OK response MUST include the Media-
 Range header (see Section 18.30).
 A client MAY send PLAY requests without the Range header.  If the
 request includes the Range header, it MUST use a symbolic value
 representing "now".  For NPT, that range specification is "npt=now-".
 The server MUST include the Range header in the response, and it MUST
 indicate an explicit time value and not a symbolic value.  In other
 words, "npt=now-" cannot be used in the response.  Instead, the time
 since session start is recommended, expressed as an open interval,
 e.g., "npt=96.23-".  An absolute time value (clock) for the
 corresponding time MAY be given, i.e., "clock=20030213T143205Z-".
 The Absolute Time format can only be used if the client has shown
 support for it using the Accept-Ranges header.

13.4.7. Playing Live with Recording

 Certain media servers may offer recording services of live sessions
 to their clients.  This recording would normally be from the
 beginning of the media session.  Clients can randomly access the
 media between now and the beginning of the media session.  This live
 media with recording is indicated by the content of the Media-
 Properties header in the SETUP response when (see also
 Section 18.29):
 o  the Random Access property is set to Random-Access;
 o  the Content Modifications property is set to Time-Progressing;
 o  the Retention property is set to Time-Limited or Unlimited
 The SETUP 200 OK response MUST include the Media-Range header (see
 Section 18.30) for this type of media.  For live media with
 recording, the Range header indicates the current delivery point in
 the media and the Media-Range header indicates the currently
 available media window around the current time.  This window can
 cover recorded content in the past (seen from current time in the
 media) or recorded content in the future (seen from current time in
 the media).  The server adjusts the delivery point to the requested
 border of the window.  If the client requests a delivery point that
 is located outside the recording window, e.g., if the requested point
 is too far in the past, the server selects the oldest point in the
 recording.  The considerations in Section 13.5.3 apply if a client
 requests delivery with scale (Section 18.46) values other than 1.0
 (normal playback rate) while delivering live media with recording.

Schulzrinne, et al. Standards Track [Page 82] RFC 7826 RTSP 2.0 December 2016

13.4.8. Playing Live with Time-Shift

 Certain media servers may offer time-shift services to their clients.
 This time shift records a fixed interval in the past, i.e., a sliding
 window recording mechanism, but not past this interval.  Clients can
 randomly access the media between now and the interval.  This live
 media with recording is indicated by the content of the Media-
 Properties header in the SETUP response when (see also
 Section 18.29):
 o  the Random Access property is set to Random-Access;
 o  the Content Modifications property is set to Time-Progressing;
 o  the Retention property is set to Time-Duration and a value
    indicating the recording interval (>0).
 The SETUP 200 OK response MUST include the Media-Range header (see
 Section 18.30) for this type of media.  For live media with
 recording, the Range header indicates the current time in the media
 and the Media-Range header indicates a window around the current
 time.  This window can cover recorded content in the past (seen from
 current time in the media) or recorded content in the future (seen
 from current time in the media).  The server adjusts the play point
 to the requested border of the window, if the client requests a play
 point that is located outside the recording windows, e.g., if
 requested too far in the past, the server selects the oldest range in
 the recording.  The considerations in Section 13.5.3 apply if a
 client requests delivery using a scale (Section 18.46) value other
 than 1.0 (normal playback rate) while delivering live media with
 time-shift.

13.5. PLAY_NOTIFY

 The PLAY_NOTIFY method is issued by a server to inform a client about
 an asynchronous event for a session in Play state.  The Session
 header MUST be presented in a PLAY_NOTIFY request and indicates the
 scope of the request.  Sending of PLAY_NOTIFY requests requires a
 persistent connection between server and client; otherwise, there is
 no way for the server to send this request method to the client.
 PLAY_NOTIFY requests have an end-to-end (i.e., server-to-client)
 scope, as they carry the Session header, and apply only to the given
 session.  The client SHOULD immediately return a response to the
 server.

Schulzrinne, et al. Standards Track [Page 83] RFC 7826 RTSP 2.0 December 2016

 PLAY_NOTIFY requests MAY use both an aggregate control URI and
 individual media resource URIs, depending on the scope of the
 notification.  This scope may have important distinctions for
 aggregated sessions, and each reason for a PLAY_NOTIFY request needs
 to specify the interpretation as well as if aggregated control URIs
 or individual URIs may be used in requests.
 PLAY_NOTIFY requests can be used with a message body, depending on
 the value of the Notify-Reason header.  It is described in the
 particular section for each Notify-Reason if a message body is used.
 However, currently there is no Notify-Reason that allows the use of a
 message body.  In this case, there is a need to obey some limitations
 when adding new Notify-Reasons that intend to use a message body: the
 server can send any type of message body, but it is not ensured that
 the client can understand the received message body.  This is related
 to DESCRIBE (see Section 13.2 ); but, in this particular case, the
 client can state its acceptable message bodies by using the Accept
 header.  In the case of PLAY_NOTIFY, the server does not know which
 message bodies are understood by the client.
 The Notify-Reason header (see Section 18.32) specifies the reason why
 the server sends the PLAY_NOTIFY request.  This is extensible and new
 reasons can be added in the future (see Section 22.8).  In case the
 client does not understand the reason for the notification, it MUST
 respond with a 465 (Notification Reason Unknown) (Section 17.4.29)
 error code.  This document defines how servers can send PLAY_NOTIFY
 with Notify-Reason values of these types:
 o  end-of-stream (see Section 13.5.1);
 o  media-properties-update (see Section 13.5.2);
 o  scale-change (see Section 13.5.3).

13.5.1. End-of-Stream

 A PLAY_NOTIFY request with the Notify-Reason header set to end-of-
 stream indicates the completion or near completion of the PLAY
 request and the ending delivery of the media stream(s).  The request
 MUST NOT be issued unless the server is in the Play state.  The end
 of the media stream delivery notification may be used either to
 indicate a successful completion of the PLAY request currently being
 served or to indicate some error resulting in failure to complete the
 request.  The Request-Status header (Section 18.42) MUST be included
 to indicate which request the notification is for and its completion
 status.  The message response status codes (Section 8.1.1) are used
 to indicate how the PLAY request concluded.  The sender of a
 PLAY_NOTIFY MAY issue an updated PLAY_NOTIFY, in the case of a

Schulzrinne, et al. Standards Track [Page 84] RFC 7826 RTSP 2.0 December 2016

 PLAY_NOTIFY sent with wrong information.  For instance, a PLAY_NOTIFY
 was issued before reaching the end-of-stream, but some error occurred
 resulting in that the previously sent PLAY_NOTIFY contained a wrong
 time when the stream will end.  In this case, a new PLAY_NOTIFY MUST
 be sent including the correct status for the completion and all
 additional information.
 PLAY_NOTIFY requests with the Notify-Reason header set to end-of-
 stream MUST include a Range header and the Scale header if the scale
 value is not 1.  The Range header indicates the point in the stream
 or streams where delivery is ending with the timescale that was used
 by the server in the PLAY response for the request being fulfilled.
 The server MUST NOT use the "now" constant in the Range header; it
 MUST use the actual numeric end position in the proper timescale.
 When end-of-stream notifications are issued prior to having sent the
 last media packets, this is made evident because the end time in the
 Range header is beyond the current time in the media being received
 by the client, e.g., "npt=-15", if npt is currently at 14.2 seconds.
 The Scale header is to be included so that it is evident if the media
 timescale is moving backwards or has a non-default pace.  The end-of-
 stream notification does not prevent the client from sending a new
 PLAY request.
 If RTP is used as media transport, an RTP-Info header MUST be
 included, and the RTP-Info header MUST indicate the last sequence
 number in the sequence parameter.
 For an RTSP Session where media resources are under aggregated
 control, the media resources will normally end at approximately the
 same time, although some small differences may exist, on the scale of
 a few hundred milliseconds.  In those cases, an RTSP session under
 aggregated control SHOULD send only a single PLAY_NOTIFY request.  By
 using the aggregate control URI in the PLAY_NOTIFY request, the RTSP
 server indicates that this applies to all media resources within the
 session.  In cases in which RTP is used for media delivery,
 corresponding RTP-Info needs to be included for all media resources.
 In cases where one or more media resources have a significantly
 shorter duration than some other resources in the aggregated session,
 the server MAY send end-of-stream notifications using individual
 media resource URIs to indicate to agents that there will be no more
 media for this particular media resource related to the current
 active PLAY request.  In such cases, when the remaining media
 resources come to the end of the stream, they MUST send a PLAY_NOTIFY
 request using the aggregate control URI to indicate that no more
 resources remain.
 A PLAY_NOTIFY request with Notify-Reason header set to end-of-stream
 MUST NOT carry a message body.

Schulzrinne, et al. Standards Track [Page 85] RFC 7826 RTSP 2.0 December 2016

 This example request notifies the client about a future end-of-stream
 event:
   S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 854
         Notify-Reason: end-of-stream
         Request-Status: cseq=853 status=200 reason="OK"
         Range: npt=-145
         RTP-Info:url="rtsp://example.com/fizzle/foo/audio"
            ssrc=0D12F123:seq=14783;rtptime=2345962545,
            url="rtsp://example.com/fizzle/video"
            ssrc=789DAF12:seq=57654;rtptime=2792482193
         Session: CDtUJfDQXJWtJ7Iqua2xOi
         Date: Mon, 08 Mar 2010 13:37:16 GMT
   C->S: RTSP/2.0 200 OK
         CSeq: 854
         User-Agent: PhonyClient/1.2
         Session: CDtUJfDQXJWtJ7Iqua2xOi

13.5.2. Media-Properties-Update

 A PLAY_NOTIFY request with a Notify-Reason header set to media-
 properties-update indicates an update of the media properties for the
 given session (see Section 18.29) or the available media range that
 can be played as indicated by the Media-Range header (Section 18.30).
 PLAY_NOTIFY requests with Notify-Reason header set to media-
 properties-update MUST include a Media-Properties and Date header and
 SHOULD include a Media-Range header.  The Media-Properties header has
 session scope; thus, for aggregated sessions, the PLAY_NOTIFY request
 MUST use the aggregated control URI.
 This notification MUST be sent for media that are time-progressing
 every time an event happens that changes the basis for making
 estimates on how the available for play-back media range will
 progress with wall clock time.  In addition, it is RECOMMENDED that
 the server send these notifications approximately every 5 minutes for
 time-progressing content to ensure the long-term stability of the
 client estimation and allow for clock skew detection by the client.
 The time between notifications should be greater than 1 minute and
 less than 2 hours.  For the reasons just explained, requests MUST
 include a Media-Range header to provide current Media duration and a
 Range header to indicate the current playing point and any remaining
 parts of the requested range.

Schulzrinne, et al. Standards Track [Page 86] RFC 7826 RTSP 2.0 December 2016

    The recommendation for sending updates every 5 minutes is due to
    any clock skew issues.  In 5 minutes, the clock skew should not
    become too significant as this is not used for media playback and
    synchronization, it is only for determining which content is
    available to the user.
 A PLAY_NOTIFY request with Notify-Reason header set to media-
 properties-update MUST NOT carry a message body.
  S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
         Date: Tue, 14 Apr 2008 15:48:06 GMT
         CSeq: 854
         Notify-Reason: media-properties-update
         Session: CDtUJfDQXJWtJ7Iqua2xOi
         Media-Properties: Time-Progressing,
               Time-Limited=20080415T153919.36Z, Random-Access=5.0
         Media-Range: npt=00:00:00-01:37:21.394
         Range: npt=01:15:49.873-
   C->S: RTSP/2.0 200 OK
         CSeq: 854
         User-Agent: PhonyClient/1.2
         Session: CDtUJfDQXJWtJ7Iqua2xOi

13.5.3. Scale-Change

 The server may be forced to change the rate of media time per
 playback time when a client requests delivery using a scale
 (Section 18.46) value other than 1.0 (normal playback rate).  For
 time-progressing media with some retention, i.e., the server stores
 already-sent content, a client requesting to play with scale values
 larger than 1 may catch up with the front end of the media.  The
 server will then be unable to continue to provide content at scale
 larger than 1 as content is only made available by the server at
 scale = 1.  Another case is when scale < 1 and the media retention is
 Time-Duration limited.  In this case, the delivery point can reach
 the oldest media unit available, and further playback at this scale
 becomes impossible as there will be no media available.  To avoid
 having the client lose any media, the scale will need to be adjusted
 to the same rate at which the media is removed from the storage
 buffer, commonly scale = 1.0.
 Another case is when the content itself consists of spliced pieces or
 is dynamically updated.  In these cases, the server may be required
 to change from one supported scale value (different than scale = 1.0)
 to another.  In this case, the server will pick the closest value and

Schulzrinne, et al. Standards Track [Page 87] RFC 7826 RTSP 2.0 December 2016

 inform the client of what it has picked.  In these cases, the media
 properties will also be sent, updating the supported scale values.
 This enables a client to adjust the scale value used.
 To minimize impact on playback in any of the above cases, the server
 MUST modify the playback properties, set scale to a supportable
 value, and continue delivery of the media.  When doing this
 modification, it MUST send a PLAY_NOTIFY message with the Notify-
 Reason header set to "scale-change".  The request MUST contain a
 Range header with the media time when the change took effect, a Scale
 header with the new value in use, a Session header with the
 identifier for the session to which it applies, and a Date header
 with the server wallclock time of the change.  For time-progressing
 content, the Media-Range and the Media-Properties headers at this
 point in time also MUST be included.  The Media-Properties header
 MUST be included if the scale change was due to the content changing
 what scale values ("Scales") are supported.
 For media streams delivered using RTP, an RTP-Info header MUST also
 be included.  It MUST contain the rtptime parameter with a value
 corresponding to the point of change in that media and optionally the
 sequence number.
 PLAY_NOTIFY requests for aggregated sessions MUST use the aggregated
 control URI in the request.  The scale change for any aggregated
 session applies to all media streams that are part of the aggregate.
 A PLAY_NOTIFY request with Notify-Reason header set to "Scale-Change"
 MUST NOT carry a message body.

Schulzrinne, et al. Standards Track [Page 88] RFC 7826 RTSP 2.0 December 2016

   S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
         Date: Tue, 14 Apr 2008 15:48:06 GMT
         CSeq: 854
         Notify-Reason: scale-change
         Session: CDtUJfDQXJWtJ7Iqua2xOi
         Media-Properties: Time-Progressing,
               Time-Limited=20080415T153919.36Z, Random-Access=5.0
         Media-Range: npt=00:00:00-01:37:21.394
         Range: npt=01:37:21.394-
         Scale: 1
         RTP-Info: url="rtsp://example.com/fizzle/foo/audio"
             ssrc=0D12F123:rtptime=2345962545,
             url="rtsp://example.com/fizzle/foo/videotrack"
             ssrc=789DAF12:seq=57654;rtptime=2792482193
   C->S: RTSP/2.0 200 OK
         CSeq: 854
         User-Agent: PhonyClient/1.2
         Session: CDtUJfDQXJWtJ7Iqua2xOi

13.6. PAUSE

 The PAUSE request causes the stream delivery to immediately be
 interrupted (halted).  A PAUSE request MUST be made either with the
 aggregated control URI for aggregated sessions, resulting in all
 media being halted, or with the media URI for non-aggregated
 sessions.  Any attempt to mute a single media with a PAUSE request in
 an aggregated session MUST be responded to with a 460 (Only Aggregate
 Operation Allowed) error.  After resuming playback, synchronization
 of the tracks MUST be maintained.  Any server resources are kept,
 though servers MAY close the session and free resources after being
 paused for the duration specified with the timeout parameter of the
 Session header in the SETUP message.
 Example:
   C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 834
         Session: OoOUPyUwt0VeY9fFRHuZ6L
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 834
         Date: Thu, 23 Jan 1997 15:35:06 GMT
                 Session: OoOUPyUwt0VeY9fFRHuZ6L
         Range: npt=45.76-75.00

Schulzrinne, et al. Standards Track [Page 89] RFC 7826 RTSP 2.0 December 2016

 The PAUSE request causes stream delivery to be interrupted
 immediately on receipt of the message, and the pause point is set to
 the current point in the presentation.  That pause point in the media
 stream needs to be maintained.  A subsequent PLAY request without a
 Range header resumes from the pause point and plays until media end.
 The pause point after any PAUSE request MUST be returned to the
 client by adding a Range header with what remains unplayed of the
 PLAY request's range.  For media with random access properties, if
 one desires to resume playing a ranged request, one simply includes
 the Range header from the PAUSE response and includes the Seek-Style
 header with the Next policy in the PLAY request.  For media that is
 time-progressing and has retention duration=0, the follow-up PLAY
 request to start media delivery again MUST use "npt=now-" and not the
 answer given in the response to PAUSE.
   C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 834
         Session: OccldOFFq23KwjYpAnBbUr
         Range: npt=10-30
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 834
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer/1.0
         Range: npt=10-30
         Seek-Style: First-Prior
         RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=5712;rtptime=934207921,
                 url="rtsp://example.com/fizzle/videotrack"
                 ssrc=4FAD8726:seq=57654;rtptime=2792482193
         Session: OccldOFFq23KwjYpAnBbUr

Schulzrinne, et al. Standards Track [Page 90] RFC 7826 RTSP 2.0 December 2016

 After 11 seconds, i.e., at 21 seconds into the presentation:
   C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 835
         Session: OccldOFFq23KwjYpAnBbUr
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 835
         Date: 23 Jan 1997 15:35:17 GMT
         Server: PhonyServer/1.0
         Range: npt=21-30
         Session: OccldOFFq23KwjYpAnBbUr
 If a client issues a PAUSE request and the server acknowledges and
 enters the Ready state, the proper server response, if the player
 issues another PAUSE, is still 200 OK.  The 200 OK response MUST
 include the Range header with the current pause point.  See examples
 below:
   C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 834
         Session: OccldOFFq23KwjYpAnBbUr
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 834
         Session: OccldOFFq23KwjYpAnBbUr
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Range: npt=45.76-98.36
   C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 835
         Session: OccldOFFq23KwjYpAnBbUr
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 835
         Session: OccldOFFq23KwjYpAnBbUr
         Date: 23 Jan 1997 15:35:07 GMT
         Range: npt=45.76-98.36

Schulzrinne, et al. Standards Track [Page 91] RFC 7826 RTSP 2.0 December 2016

13.7. TEARDOWN

13.7.1. Client to Server

 The TEARDOWN client-to-server request stops the stream delivery for
 the given URI, freeing the resources associated with it.  A TEARDOWN
 request can be performed on either an aggregated or a media control
 URI.  However, some restrictions apply depending on the current
 state.  The TEARDOWN request MUST contain a Session header indicating
 to what session the request applies.  The TEARDOWN request MUST NOT
 include a Terminate-Reason header.
 A TEARDOWN using the aggregated control URI or the media URI in a
 session under non-aggregated control (single media session) MAY be
 done in any state (Ready and Play).  A successful request MUST result
 in that media delivery being immediately halted and the session state
 being destroyed.  This MUST be indicated through the lack of a
 Session header in the response.
 A TEARDOWN using a media URI in an aggregated session can only be
 done in Ready state.  Such a request only removes the indicated media
 stream and associated resources from the session.  This may result in
 a session returning to non-aggregated control, because it only
 contains a single media after the request's completion.  A session
 that will exist after the processing of the TEARDOWN request MUST, in
 the response to that TEARDOWN request, contain a Session header.
 Thus, the presence of the Session header indicates to the receiver of
 the response if the session is still extant or has been removed.

Schulzrinne, et al. Standards Track [Page 92] RFC 7826 RTSP 2.0 December 2016

 Example:
   C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 892
         Session: OccldOFFq23KwjYpAnBbUr
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 892
         Server: PhonyServer/1.0

13.7.2. Server to Client

 The server can send TEARDOWN requests in the server-to-client
 direction to indicate that the server has been forced to terminate
 the ongoing session.  This may happen for several reasons, such as
 server maintenance without available backup, or that the session has
 been inactive for extended periods of time.  The reason is provided
 in the Terminate-Reason header (Section 18.52).
 When an RTSP client has maintained an RTSP session that otherwise is
 inactive for an extended period of time, the server may reclaim the
 resources.  That is done by issuing a TEARDOWN request with the
 Terminate-Reason set to "Session-Timeout".  This MAY be done when the
 client has been inactive in the RTSP session for more than one
 Session Timeout period (Section 18.49).  However, the server is NOT
 RECOMMENDED to perform this operation until an extended period of
 inactivity of 10 times the Session-Timeout period has passed.  It is
 up to the operator of the RTSP server to actually configure how long
 this extended period of inactivity is.  An operator should take into
 account, when doing this configuration, what the served content is
 and what this means for the extended period of inactivity.
 In case the server needs to stop providing service to the established
 sessions and there is no server to point at in a REDIRECT request,
 then TEARDOWN SHALL be used to terminate the session.  This method
 can also be used when non-recoverable internal errors have happened
 and the server has no other option than to terminate the sessions.
 The TEARDOWN request MUST be made only on the session aggregate
 control URI (i.e., it is not allowed to terminate individual media
 streams, if it is a session aggregate), and it MUST include the
 following headers: Session and Terminate-Reason.  The request only
 applies to the session identified in the Session header.  The server
 may include a message to the client's user with the "user-msg"
 parameter.

Schulzrinne, et al. Standards Track [Page 93] RFC 7826 RTSP 2.0 December 2016

 The TEARDOWN request may alternatively be done on the wildcard URI
 "*" and without any session header.  The scope of such a request is
 limited to the next-hop (i.e., the RTSP agent in direct communication
 with the server) and applies, as well, to the RTSP connection between
 the next-hop RTSP agent and the server.  This request indicates that
 all sessions and pending requests being managed via the connection
 are terminated.  Any intervening proxies SHOULD do all of the
 following in the order listed:
 1.  respond to the TEARDOWN request
 2.  disconnect the control channel from the requesting server
 3.  pass the TEARDOWN request to each applicable client (typically
     those clients with an active session or an unanswered request)
    Note: The proxy is responsible for accepting TEARDOWN responses
    from its clients; these responses MUST NOT be passed on to either
    the original server or the target server in the redirect.

13.8. GET_PARAMETER

 The GET_PARAMETER request retrieves the value of any specified
 parameter or parameters for a presentation or stream specified in the
 URI.  If the Session header is present in a request, the value of a
 parameter MUST be retrieved in the specified session context.  There
 are two ways of specifying the parameters to be retrieved.
 The first approach includes headers that have been defined to be
 usable for this purpose.  Headers for this purpose should allow
 empty, or stripped value parts to avoid having to specify bogus data
 when indicating the desire to retrieve a value.  The successful
 completion of the request should also be evident from any filled out
 values in the response.  The headers in this specification that MAY
 be used for retrieving their current value using GET_PARAMETER are
 listed below; additional headers MAY be specified in the future:
 o  Accept-Ranges
 o  Media-Range
 o  Media-Properties
 o  Range
 o  RTP-Info

Schulzrinne, et al. Standards Track [Page 94] RFC 7826 RTSP 2.0 December 2016

 The other way is to specify a message body that lists the
 parameter(s) that are desired to be retrieved.  The Content-Type
 header (Section 18.19) is used to specify which format the message
 body has.  If the receiver of the request does not support the media
 type used for the message body, it SHALL respond using the error code
 415 (Unsupported Media Type).  The responder to a GET_PARAMETER
 request MUST use the media type of the request for the response.  For
 additional considerations regarding message body negotiation, see
 Section 9.3.
 RTSP agents implementing support for responding to GET_PARAMETER
 requests SHALL implement the "text/parameters" format (Appendix F).
 This to ensure that at least one known format for parameters is
 implemented and, thus, prevent parameter format negotiation failure.
 Parameters specified within the body of the message must all be
 understood by the request-receiving agent.  If one or more parameters
 are not understood a 451 (Parameter Not Understood) MUST be sent
 including a body listing the parameters that weren't understood.  If
 all parameters are understood, their values are filled in and
 returned in the response message body.
 The method can also be used without a message body or any header that
 requests parameters for keep-alive purposes.  The keep-alive timer
 has been updated for any request that is successful, i.e., a 200 OK
 response is received.  Any non-required header present in such a
 request may or may not have been processed.  Normally, the presence
 of filled-out values in the header will be indication that the header
 has been processed.  However, for cases when this is difficult to
 determine, it is recommended to use a feature tag and the Require
 header.  For this reason, it is usually easier if any parameters to
 be retrieved are sent in the body, rather than using any header.

Schulzrinne, et al. Standards Track [Page 95] RFC 7826 RTSP 2.0 December 2016

 Example:
   S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 431
         User-Agent: PhonyClient/1.2
         Session: OccldOFFq23KwjYpAnBbUr
         Content-Length: 26
         Content-Type: text/parameters
         packets_received
         jitter
   C->S: RTSP/2.0 200 OK
         CSeq: 431
         Session: OccldOFFq23KwjYpAnBbUr
         Server: PhonyServer/1.1
         Date: Mon, 08 Mar 2010 13:43:23 GMT
         Content-Length: 38
         Content-Type: text/parameters
         packets_received: 10
         jitter: 0.3838

13.9. SET_PARAMETER

 This method requests the setting of the value of a parameter or a set
 of parameters for a presentation or stream specified by the URI.  If
 the Session header is present in a request, the value of a parameter
 MUST be retrieved in the specified session context.  The method MAY
 also be used without a message body.  It is the RECOMMENDED method to
 be used in a request sent for the sole purpose of updating the keep-
 alive timer.  If this request is successful, i.e., a 200 OK response
 is received, then the keep-alive timer has been updated.  Any non-
 required header present in such a request may or may not have been
 processed.  To allow a client to determine if any such header has
 been processed, it is necessary to use a feature tag and the Require
 header.  Due to this reason it is RECOMMENDED that any parameters are
 sent in the body rather than using any header.
 When using a message body to list the parameter(s) desired to be set,
 the Content-Type header (Section 18.19) is used to specify which
 format the message body has.  If the receiver of the request is not
 supporting the media type used for the message body, it SHALL respond
 using the error code 415 (Unsupported Media Type).  For additional
 considerations regarding message body negotiation, see Section 9.3.
 The responder to a SET_PARAMETER request MUST use the media type of
 the request for the response.  For additional considerations
 regarding message body negotiation, see Section 9.3.

Schulzrinne, et al. Standards Track [Page 96] RFC 7826 RTSP 2.0 December 2016

 RTSP agents implementing support for responding to SET_PARAMETER
 requests SHALL implement the text/parameters format (Appendix F).
 This is to ensure that at least one known format for parameters is
 implemented and, thus, prevent parameter format negotiation failure.
 A request is RECOMMENDED to only contain a single parameter to allow
 the client to determine why a particular request failed.  If the
 request contains several parameters, the server MUST only act on the
 request if all of the parameters can be set successfully.  A server
 MUST allow a parameter to be set repeatedly to the same value, but it
 MAY disallow changing parameter values.  If the receiver of the
 request does not understand or cannot locate a parameter, error 451
 (Parameter Not Understood) MUST be used.  When a parameter is not
 allowed to change, the error code is 458 (Parameter Is Read-Only).
 The response body MUST contain only the parameters that have errors.
 Otherwise, a body MUST NOT be returned.  The response body MUST use
 the media type of the request for the response.
 Note: transport parameters for the media stream MUST only be set with
 the SETUP command.
    Restricting setting transport parameters to SETUP is for the
    benefit of firewalls connected to border RTSP proxies.
    The parameters are split in a fine-grained fashion so that there
    can be more meaningful error indications.  However, it may make
    sense to allow the setting of several parameters if an atomic
    setting is desirable.  Imagine device control where the client
    does not want the camera to pan unless it can also tilt to the
    right angle at the same time.

Schulzrinne, et al. Standards Track [Page 97] RFC 7826 RTSP 2.0 December 2016

 Example:
   C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 421
         User-Agent: PhonyClient/1.2
         Session: iixT43KLc
         Date: Mon, 08 Mar 2010 14:45:04 GMT
         Content-length: 20
         Content-type: text/parameters
         barparam: barstuff
   S->C: RTSP/2.0 451 Parameter Not Understood
         CSeq: 421
         Session: iixT43KLc
         Server: PhonyServer/1.0
         Date: Mon, 08 Mar 2010 14:44:56 GMT
         Content-length: 20
         Content-type: text/parameters
         barparam: barstuff

13.10. REDIRECT

 The REDIRECT method is issued by a server to inform a client that the
 service provided will be terminated and where a corresponding service
 can be provided instead.  This may happen for different reasons.  One
 is that the server is being administered such that it must stop
 providing service.  Thus, the client is required to connect to
 another server location to access the resource indicated by the
 Request-URI.
 The REDIRECT request SHALL contain a Terminate-Reason header
 (Section 18.52) to inform the client of the reason for the request.
 Additional parameters related to the reason may also be included.
 The intention here is to allow a server administrator to do a
 controlled shutdown of the RTSP server.  That requires sufficient
 time to inform all entities having associated state with the server
 and for them to perform a controlled migration from this server to a
 fall-back server.
 A REDIRECT request with a Session header has end-to-end (i.e.,
 server-to-client) scope and applies only to the given session.  Any
 intervening proxies SHOULD NOT disconnect the control channel while
 there are other remaining end-to-end sessions.  The REQUIRED Location
 header MUST contain a complete absolute URI pointing to the resource
 to which the client SHOULD reconnect.  Specifically, the Location

Schulzrinne, et al. Standards Track [Page 98] RFC 7826 RTSP 2.0 December 2016

 MUST NOT contain just the host and port.  A client may receive a
 REDIRECT request with a Session header, if and only if, an end-to-end
 session has been established.
 A client may receive a REDIRECT request without a Session header at
 any time when it has communication or a connection established with a
 server.  The scope of such a request is limited to the next-hop
 (i.e., the RTSP agent in direct communication with the server) and
 applies to all sessions controlled, as well as the connection between
 the next-hop RTSP agent and the server.  A REDIRECT request without a
 Session header indicates that all sessions and pending requests being
 managed via the connection MUST be redirected.  The Location header,
 if included in such a request, SHOULD contain an absolute URI with
 only the host address and the OPTIONAL port number of the server to
 which the RTSP agent SHOULD reconnect.  Any intervening proxies
 SHOULD do all of the following in the order listed:
 1.  respond to the REDIRECT request
 2.  disconnect the control channel from the requesting server
 3.  connect to the server at the given host address
 4.  pass the REDIRECT request to each applicable client (typically
     those clients with an active session or an unanswered request)
    Note: The proxy is responsible for accepting REDIRECT responses
    from its clients; these responses MUST NOT be passed on to either
    the original server or the redirected server.
 A server that needs to terminate a session or all its sessions and
 lacks an alternative server to redirect to, SHALL instead use
 TEARDOWN requests.
 When no Terminate-Reason "time" parameter is included in a REDIRECT
 request, the client SHALL perform the redirection immediately and
 return a response to the server.  The server shall consider the
 session to be terminated and can free any associated state after it
 receives the successful (2xx) response.  The server MAY close the
 signaling connection upon receiving the response, and the client
 SHOULD close the signaling connection after sending the 2xx response.
 The exception to this is when the client has several sessions on the
 server being managed by the given signaling connection.  In this
 case, the client SHOULD close the connection when it has received and
 responded to REDIRECT requests for all the sessions managed by the
 signaling connection.

Schulzrinne, et al. Standards Track [Page 99] RFC 7826 RTSP 2.0 December 2016

 The Terminate-Reason header "time" parameter MAY be used to indicate
 the wallclock time by which the redirection MUST have taken place.
 To allow a client to determine that redirect time without being time
 synchronized with the server, the server MUST include a Date header
 in the request.  The client should have terminated the session and
 closed the connection before the redirection time-line terminated.
 The server MAY simply cease to provide service when the deadline time
 has been reached, or it can issue a TEARDOWN requests to the
 remaining sessions.
 If the REDIRECT request times out following the rules in
 Section 10.4, the server MAY terminate the session or transport
 connection that would be redirected by the request.  This is a
 safeguard against misbehaving clients that refuse to respond to a
 REDIRECT request.  This action removes any incentive of not
 acknowledging the reception of a REDIRECT request.
 After a REDIRECT request has been processed, a client that wants to
 continue to receive media for the resource identified by the Request-
 URI will have to establish a new session with the designated host.
 If the URI given in the Location header is a valid resource URI, a
 client SHOULD issue a DESCRIBE request for the URI.
    Note: The media resource indicated by the Location header can be
    identical, slightly different, or totally different.  This is the
    reason why a new DESCRIBE request SHOULD be issued.
 If the Location header contains only a host address, the client may
 assume that the media on the new server is identical to the media on
 the old server, i.e., all media configuration information from the
 old session is still valid except for the host address.  However, the
 usage of conditional SETUP using MTag identifiers is RECOMMENDED as a
 means to verify the assumption.
 This example request redirects traffic for this session to the new
 server at the given absolute time:
   S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 732
         Location: rtsp://s2.example.com:8001/fizzle/foo
         Terminate-Reason: Server-Admin ;time=19960213T143205Z
         Session: uZ3ci0K+Ld-M
         Date: Thu, 13 Feb 1996 14:30:43 GMT
   C->S: RTSP/2.0 200 OK
         CSeq: 732
         User-Agent: PhonyClient/1.2
         Session: uZ3ci0K+Ld-M

Schulzrinne, et al. Standards Track [Page 100] RFC 7826 RTSP 2.0 December 2016

14. Embedded (Interleaved) Binary Data

 In order to fulfill certain requirements on the network side, e.g.,
 in conjunction with network address translators that block RTP
 traffic over UDP, it may be necessary to interleave RTSP messages and
 media-stream data.  This interleaving should generally be avoided
 unless necessary since it complicates client and server operation and
 imposes additional overhead.  Also, head-of-line blocking may cause
 problems.  Interleaved binary data SHOULD only be used if RTSP is
 carried over TCP.  Interleaved data is not allowed inside RTSP
 messages.
 Stream data, such as RTP packets, is encapsulated by an ASCII dollar
 sign (36 decimal) followed by a one-octet channel identifier and the
 length of the encapsulated binary data as a binary, two-octet
 unsigned integer in network octet order (Appendix B of [RFC791]).
 The stream data follows immediately afterwards, without a CRLF, but
 including the upper-layer protocol headers.  Each dollar sign block
 MUST contain exactly one upper-layer protocol data unit, e.g., one
 RTP packet.
    Note that this mechanism does not support PDUs larger than 65535
    octets, which matches the maximum payload size of regular, non-
    jumbo IPv4 and IPv6 packets.  If the media delivery protocol
    intended to be used has larger PDUs than that, a definition of a
    PDU fragmentation mechanism will be required to support embedded
    binary data.
     0                   1                   2                   3
     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    | "$" = 36      | Channel ID    | Length in octets              |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    : Binary data (Length according to Length field)                :
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           Figure 1: Embedded Interleaved Binary Data Format
 The channel identifier is defined in the Transport header with the
 interleaved parameter (Section 18.54).
 When the transport choice is RTP, RTCP messages are also interleaved
 by the server over the TCP connection.  The usage of RTCP messages is
 indicated by including an interval containing a second channel in the
 interleaved parameter of the Transport header (see Section 18.54).
 If RTCP is used, packets MUST be sent on the first available channel

Schulzrinne, et al. Standards Track [Page 101] RFC 7826 RTSP 2.0 December 2016

 that is higher than the RTP channel.  The channels are bidirectional,
 using the same Channel ID in both directions; therefore, RTCP traffic
 is sent on the second channel in both directions.
    RTCP is sometimes needed for synchronization when two or more
    streams are interleaved in such a fashion.  Also, this provides a
    convenient way to tunnel RTP/RTCP packets through the RTSP
    connection (TCP or TCP/TLS) when required by the network
    configuration and to transfer them onto UDP when possible.
   C->S: SETUP rtsp://example.com/bar.file RTSP/2.0
         CSeq: 2
         Transport: RTP/AVP/TCP;unicast;interleaved=0-1
         Accept-Ranges: npt, smpte, clock
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 2
         Date: Thu, 05 Jun 1997 18:57:18 GMT
         Transport: RTP/AVP/TCP;unicast;interleaved=5-6
         Session: OccldOFFq23KwjYpAnBbUr
         Accept-Ranges: npt
         Media-Properties: Random-Access=0.2, Immutable, Unlimited
   C->S: PLAY rtsp://example.com/bar.file RTSP/2.0
         CSeq: 3
         Session: OccldOFFq23KwjYpAnBbUr
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 3
         Session: OccldOFFq23KwjYpAnBbUr
         Date: Thu, 05 Jun 1997 18:57:19 GMT
         RTP-Info: url="rtsp://example.com/bar.file"
           ssrc=0D12F123:seq=232433;rtptime=972948234
         Range: npt=0-56.8
         Seek-Style: RAP
   S->C: $005{2 octet length}{"length" octets data, w/RTP header}
   S->C: $005{2 octet length}{"length" octets data, w/RTP header}
   S->C: $006{2 octet length}{"length" octets  RTCP packet}

Schulzrinne, et al. Standards Track [Page 102] RFC 7826 RTSP 2.0 December 2016

15. Proxies

 RTSP Proxies are RTSP agents that are located in between a client and
 a server.  A proxy can take on the roles of both client and server
 depending on what it tries to accomplish.  RTSP proxies use two
 transport-layer connections: one from the RTSP client to the RTSP
 proxy and a second from the RTSP proxy to the RTSP server.  Proxies
 are introduced for several different reasons; those listed below are
 often combined.
 Caching Proxy:  This type of proxy is used to reduce the workload on
       servers and connections.  By caching the description and media
       streams, i.e., the presentation, the proxy can serve a client
       with content, but without requesting it from the server once it
       has been cached and has not become stale.  See Section 16.
       This type of proxy is also expected to understand RTSP endpoint
       functionality, i.e., functionality identified in the Require
       header in addition to what Proxy-Require demands.
 Translator Proxy:  This type of proxy is used to ensure that an RTSP
       client gets access to servers and content on an external
       network or gets access by using content encodings not supported
       by the client.  The proxy performs the necessary translation of
       addresses, protocols, or encodings.  This type of proxy is
       expected also to understand RTSP endpoint functionality, i.e.,
       functionality identified in the Require header in addition to
       what Proxy-Require demands.
 Access Proxy:  This type of proxy is used to ensure that an RTSP
       client gets access to servers on an external network.  Thus,
       this proxy is placed on the border between two domains, e.g., a
       private address space and the public Internet.  The proxy
       performs the necessary translation, usually addresses.  This
       type of proxy is required to redirect the media to itself or a
       controlled gateway that performs the translation before the
       media can reach the client.
 Security Proxy:  This type of proxy is used to help facilitate
       security functions around RTSP.  For example, in the case of a
       firewalled network, the security proxy requests that the
       necessary pinholes in the firewall are opened when a client in
       the protected network wants to access media streams on the
       external side.  This proxy can perform its function without
       redirecting the media between the server and client.  However,
       in deployments with private address spaces, this proxy is
       likely to be combined with the access proxy.  The functionality
       of this proxy is usually closely tied into understanding all
       aspects of the media transport.

Schulzrinne, et al. Standards Track [Page 103] RFC 7826 RTSP 2.0 December 2016

 Auditing Proxy:  RTSP proxies can also provide network owners with a
       logging and auditing point for RTSP sessions, e.g., for
       corporations that track their employees usage of the network.
       This type of proxy can perform its function without inserting
       itself or any other node in the media transport.  This proxy
       type can also accept unknown methods as it doesn't interfere
       with the clients' requests.
 All types of proxies can also be used when using secured
 communication with TLS, as RTSP 2.0 allows the client to approve
 certificate chains used for connection establishment from a proxy;
 see Section 19.3.2.  However, that trust model may not be suitable
 for all types of deployment.  In those cases, the secured sessions do
 bypass the proxies.
 Access proxies SHOULD NOT be used in equipment like NATs and
 firewalls that aren't expected to be regularly maintained, like home
 or small office equipment.  In these cases, it is better to use the
 NAT traversal procedures defined for RTSP 2.0 [RFC7825].  The reason
 for these recommendations is that any extensions of RTSP resulting in
 new media-transport protocols or profiles, new parameters, etc., may
 fail in a proxy that isn't maintained.  This would impede RTSP's
 future development and usage.

15.1. Proxies and Protocol Extensions

 The existence of proxies must always be considered when developing
 new RTSP extensions.  Most types of proxies will need to implement
 any new method to operate correctly in the presence of that
 extension.  New headers can be introduced and will not be blocked by
 older proxies.  However, it is important to consider if this header
 and its function are required to be understood by the proxy or if it
 can be simply forwarded.  If the header needs to be understood, a
 feature tag representing the functionality MUST be included in the
 Proxy-Require header.  Below are guidelines for analysis whether the
 header needs to be understood.  The Transport header and its
 parameters are extensible, which requires handling rules for a proxy
 in order to ensure a correct interpretation.

Schulzrinne, et al. Standards Track [Page 104] RFC 7826 RTSP 2.0 December 2016

 Whether or not a proxy needs to understand a header is not easy to
 determine as they serve a broad variety of functions.  When
 evaluating if a header needs to be understood, one can divide the
 functionality into three main categories:
 Media modifying:  The caching and translator proxies modify the
    actual media and therefore need also to understand the request
    directed to the server that affects how the media is rendered.
    Thus, this type of proxy also needs to understand the server-side
    functionality.
 Transport modifying:  The access and the security proxy both need to
    understand how the media transport is performed, either for
    opening pinholes or translating the outer headers, e.g., IP and
    UDP or TCP.
 Non-modifying:  The audit proxy is special in that it does not modify
    the messages in other ways than to insert the Via header.  That
    makes it possible for this type to forward RTSP messages that
    contain different types of unknown methods, headers, or header
    parameters.
 An extension has to be classified as mandatory to be implemented for
 a proxy, if an extension has to be understood by a "Transport
 modifying" type of proxy.

15.2. Multiplexing and Demultiplexing of Messages

 RTSP proxies may have to multiplex several RTSP sessions from their
 clients towards RTSP servers.  This requires that RTSP requests from
 multiple clients be multiplexed onto a common connection for requests
 outgoing to an RTSP server, and, on the way back, the responses be
 demultiplexed from the server to per-client responses.  On the
 protocol level, this requires that request and response messages be
 handled in both directions, requiring that there be a mechanism to
 correlate which request/response pair exchanged between proxy and
 server is mapped to which client (or client request).
 This multiplexing of requests and demultiplexing of responses is done
 by using the CSeq header field.  The proxy has to rewrite the CSeq in
 requests to the server and responses from the server and remember
 which CSeq is mapped to which client.  The proxy also needs to ensure
 that the order of the message related to each client is maintained.
 Section 18.20 defines the handling of how requests and responses are
 rewritten.

Schulzrinne, et al. Standards Track [Page 105] RFC 7826 RTSP 2.0 December 2016

16. Caching

 In HTTP, request/response pairs are cached.  RTSP differs
 significantly in that respect.  Responses are not cacheable, with the
 exception of the presentation description returned by DESCRIBE.
 (Since the responses for anything but DESCRIBE and GET_PARAMETER do
 not return any data, caching is not really an issue for these
 requests.)  However, it is desirable for the continuous media data,
 typically delivered out-of-band with respect to RTSP, to be cached,
 as well as the session description.
 On receiving a SETUP or PLAY request, a proxy ascertains whether it
 has an up-to-date copy of the continuous media content and its
 description.  It can determine whether the copy is up to date by
 issuing a SETUP or DESCRIBE request, respectively, and comparing the
 Last-Modified header with that of the cached copy.  If the copy is
 not up to date, it modifies the SETUP transport parameters as
 appropriate and forwards the request to the origin server.
 Subsequent control commands such as PLAY or PAUSE then pass the proxy
 unmodified.  The proxy delivers the continuous media data to the
 client, while possibly making a local copy for later reuse.  The
 exact allowed behavior of the cache is given by the cache-response
 directives described in Section 18.11.  A cache MUST answer any
 DESCRIBE requests if it is currently serving the stream to the
 requester, as it is possible that low-level details of the stream
 description may have changed on the origin server.
 Note that an RTSP cache is of the "cut-through" variety.  Rather than
 retrieving the whole resource from the origin server, the cache
 simply copies the streaming data as it passes by on its way to the
 client.  Thus, it does not introduce additional latency.
 To the client, an RTSP proxy cache appears like a regular media
 server.  To the media origin server, an RTSP proxy cache appears like
 a client.  Just as an HTTP cache has to store the content type,
 content language, and so on for the objects it caches, a media cache
 has to store the presentation description.  Typically, a cache
 eliminates all transport references (e.g., multicast information)
 from the presentation description, since these are independent of the
 data delivery from the cache to the client.  Information on the
 encodings remains the same.  If the cache is able to translate the
 cached media data, it would create a new presentation description
 with all the encoding possibilities it can offer.

Schulzrinne, et al. Standards Track [Page 106] RFC 7826 RTSP 2.0 December 2016

16.1. Validation Model

 When a cache has a stale entry that it would like to use as a
 response to a client's request, it first has to check with the origin
 server (or possibly an intermediate cache with a fresh response) to
 see if its cached entry is still usable.  This is called "validating"
 the cache entry.  To avoid having to pay the overhead of
 retransmitting the full response if the cached entry is good, and at
 the same time avoiding having to pay the overhead of an extra round
 trip if the cached entry is invalid, RTSP supports the use of
 conditional methods.
 The key protocol features for supporting conditional methods are
 those concerned with "cache validators."  When an origin server
 generates a full response, it attaches some sort of validator to it,
 which is kept with the cache entry.  When a client (user agent or
 proxy cache) makes a conditional request for a resource for which it
 has a cache entry, it includes the associated validator in the
 request.
 The server then checks that validator against the current validator
 for the requested resource, and, if they match (see Section 16.1.3),
 it responds with a special status code (usually, 304 (Not Modified))
 and no message body.  Otherwise, it returns a full response
 (including message body).  Thus, avoiding transmitting the full
 response if the validator matches and avoiding an extra round trip if
 it does not match.
 In RTSP, a conditional request looks exactly the same as a normal
 request for the same resource, except that it carries a special
 header (which includes the validator) that implicitly turns the
 method (usually DESCRIBE or SETUP) into a conditional.
 The protocol includes both positive and negative senses of cache-
 validating conditions.  That is, it is possible to request that a
 method be performed either if and only if a validator matches or if
 and only if no validators match.
    Note: a response that lacks a validator may still be cached, and
    served from cache until it expires, unless this is explicitly
    prohibited by a cache directive (see Section 18.11).  However, a
    cache cannot perform a conditional retrieval if it does not have a
    validator for the resource, which means it will not be refreshable
    after it expires.

Schulzrinne, et al. Standards Track [Page 107] RFC 7826 RTSP 2.0 December 2016

 Media streams that are being adapted based on the transport capacity
 between the server and the cache make caching more difficult.  A
 server needs to consider how it views the caching of media streams
 that it adapts and potentially instruct any caches not to cache such
 streams.

16.1.1. Last-Modified Dates

 The Last-Modified header (Section 18.27) value is often used as a
 cache validator.  In simple terms, a cache entry is considered to be
 valid if the cache entry was created after the Last-Modified time.

16.1.2. Message Body Tag Cache Validators

 The MTag response-header field-value, a message body tag, provides
 for an "opaque" cache validator.  This might allow more reliable
 validation in situations where it is inconvenient to store
 modification dates, where the one-second resolution of RTSP-date
 values is not sufficient, or where the origin server wishes to avoid
 certain paradoxes that might arise from the use of modification
 dates.
 Message body tags are described in Section 4.6

16.1.3. Weak and Strong Validators

 Since both origin servers and caches will compare two validators to
 decide if they represent the same or different entities, one normally
 would expect that if the message body (i.e., the presentation
 description) or any associated message body headers changes in any
 way, then the associated validator would change as well.  If this is
 true, then this validator is a "strong validator".  The Message body
 (i.e., the presentation description) or any associated message body
 headers is named an entity for a better understanding.
 However, there might be cases when a server prefers to change the
 validator only on semantically significant changes and not when
 insignificant aspects of the entity change.  A validator that does
 not always change when the resource changes is a "weak validator".
 Message body tags are normally strong validators, but the protocol
 provides a mechanism to tag a message body tag as "weak".  One can
 think of a strong validator as one that changes whenever the bits of
 an entity changes, while a weak value changes whenever the meaning of
 an entity changes.  Alternatively, one can think of a strong
 validator as part of an identifier for a specific entity, while a
 weak validator is part of an identifier for a set of semantically
 equivalent entities.

Schulzrinne, et al. Standards Track [Page 108] RFC 7826 RTSP 2.0 December 2016

    Note: One example of a strong validator is an integer that is
    incremented in stable storage every time an entity is changed.
    An entity's modification time, if represented with one-second
    resolution, could be a weak validator, since it is possible that
    the resource might be modified twice during a single second.
    Support for weak validators is optional.  However, weak validators
    allow for more efficient caching of equivalent objects.
 A "use" of a validator is either when a client generates a request
 and includes the validator in a validating header field or when a
 server compares two validators.
 Strong validators are usable in any context.  Weak validators are
 only usable in contexts that do not depend on exact equality of an
 entity.  For example, either kind is usable for a conditional
 DESCRIBE of a full entity.  However, only a strong validator is
 usable for a subrange retrieval, since otherwise the client might end
 up with an internally inconsistent entity.
 Clients MAY issue DESCRIBE requests with either weak or strong
 validators.  Clients MUST NOT use weak validators in other forms of
 requests.
 The only function that RTSP defines on validators is comparison.
 There are two validator comparison functions, depending on whether or
 not the comparison context allows the use of weak validators:
 o  The strong comparison function: in order to be considered equal,
    both validators MUST be identical in every way, and both MUST NOT
    be weak.
 o  The weak comparison function: in order to be considered equal,
    both validators MUST be identical in every way, but either or both
    of them MAY be tagged as "weak" without affecting the result.
 A message body tag is strong unless it is explicitly tagged as weak.
 A Last-Modified time, when used as a validator in a request, is
 implicitly weak unless it is possible to deduce that it is strong,
 using the following rules:
 o  The validator is being compared by an origin server to the actual
    current validator for the entity and,

Schulzrinne, et al. Standards Track [Page 109] RFC 7826 RTSP 2.0 December 2016

 o  That origin server reliably knows that the associated entity did
    not change more than once during the second covered by the
    presented validator.
 OR
 o  The validator is about to be used by a client in an If-Modified-
    Since, because the client has a cache entry for the associated
    entity, and
 o  That cache entry includes a Date value, which gives the time when
    the origin server sent the original response, and
 o  The presented Last-Modified time is at least 60 seconds before the
    Date value.
 OR
 o  The validator is being compared by an intermediate cache to the
    validator stored in its cache entry for the entity, and
 o  That cache entry includes a Date value, which gives the time when
    the origin server sent the original response, and
 o  The presented Last-Modified time is at least 60 seconds before the
    Date value.
 This method relies on the fact that if two different responses were
 sent by the origin server during the same second, but both had the
 same Last-Modified time, then at least one of those responses would
 have a Date value equal to its Last-Modified time.  The arbitrary
 60-second limit guards against the possibility that the Date and
 Last-Modified values are generated from different clocks or at
 somewhat different times during the preparation of the response.  An
 implementation MAY use a value larger than 60 seconds, if it is
 believed that 60 seconds is too short.
 If a client wishes to perform a subrange retrieval on a value for
 which it has only a Last-Modified time and no opaque validator, it
 MAY do this only if the Last-Modified time is strong in the sense
 described here.

16.1.4. Rules for When to Use Message Body Tags and Last-Modified Dates

 This document adopts a set of rules and recommendations for origin
 servers, clients, and caches regarding when various validator types
 ought to be used, and for what purposes.

Schulzrinne, et al. Standards Track [Page 110] RFC 7826 RTSP 2.0 December 2016

 RTSP origin servers:
 o  SHOULD send a message body tag validator unless it is not feasible
    to generate one.
 o  MAY send a weak message body tag instead of a strong message body
    tag, if performance considerations support the use of weak message
    body tags, or if it is unfeasible to send a strong message body
    tag.
 o  SHOULD send a Last-Modified value if it is feasible to send one,
    unless the risk of a breakdown in semantic transparency that could
    result from using this date in an If-Modified-Since header would
    lead to serious problems.
 In other words, the preferred behavior for an RTSP origin server is
 to send both a strong message body tag and a Last-Modified value.
 In order to be legal, a strong message body tag MUST change whenever
 the associated entity value changes in any way.  A weak message body
 tag SHOULD change whenever the associated entity changes in a
 semantically significant way.
    Note: in order to provide semantically transparent caching, an
    origin server MUST avoid reusing a specific strong message body
    tag value for two different entities or reusing a specific weak
    message body tag value for two semantically different entities.
    Cache entries might persist for arbitrarily long periods,
    regardless of expiration times, so it might be inappropriate to
    expect that a cache will never again attempt to validate an entry
    using a validator that it obtained at some point in the past.
 RTSP clients:
 o  If a message body tag has been provided by the origin server, MUST
    use that message body tag in any cache-conditional request (using
    If-Match or If-None-Match).
 o  If only a Last-Modified value has been provided by the origin
    server, SHOULD use that value in non-subrange cache-conditional
    requests (using If-Modified-Since).
 o  If both a message body tag and a Last-Modified value have been
    provided by the origin server, SHOULD use both validators in
    cache-conditional requests.
 An RTSP origin server, upon receiving a conditional request that
 includes both a Last-Modified date (e.g., in an If-Modified-Since
 header) and one or more message body tags (e.g., in an If-Match,

Schulzrinne, et al. Standards Track [Page 111] RFC 7826 RTSP 2.0 December 2016

 If-None-Match, or If-Range header field) as cache validators, MUST
 NOT return a response status of 304 (Not Modified) unless doing so is
 consistent with all of the conditional header fields in the request.
    Note: The general principle behind these rules is that RTSP
    servers and clients should transmit as much non-redundant
    information as is available in their responses and requests.  RTSP
    systems receiving this information will make the most conservative
    assumptions about the validators they receive.

16.1.5. Non-validating Conditionals

 The principle behind message body tags is that only the service
 author knows the semantics of a resource well enough to select an
 appropriate cache validation mechanism, and the specification of any
 validator comparison function more complex than octet equality would
 open up a can of worms.  Thus, comparisons of any other headers are
 never used for purposes of validating a cache entry.

16.2. Invalidation after Updates or Deletions

 The effect of certain methods performed on a resource at the origin
 server might cause one or more existing cache entries to become non-
 transparently invalid.  That is, although they might continue to be
 "fresh," they do not accurately reflect what the origin server would
 return for a new request on that resource.
 There is no way for RTSP to guarantee that all such cache entries are
 marked invalid.  For example, the request that caused the change at
 the origin server might not have gone through the proxy where a cache
 entry is stored.  However, several rules help reduce the likelihood
 of erroneous behavior.
 In this section, the phrase "invalidate an entity" means that the
 cache will either remove all instances of that entity from its
 storage or mark these as "invalid" and in need of a mandatory
 revalidation before they can be returned in response to a subsequent
 request.
 Some RTSP methods MUST cause a cache to invalidate an entity.  This
 is either the entity referred to by the Request-URI or by the
 Location or Content-Location headers (if present).  These methods
 are:
 o  DESCRIBE
 o  SETUP

Schulzrinne, et al. Standards Track [Page 112] RFC 7826 RTSP 2.0 December 2016

 In order to prevent DoS attacks, an invalidation based on the URI in
 a Location or Content-Location header MUST only be performed if the
 host part is the same as in the Request-URI.
 A cache that passes through requests for methods it does not
 understand SHOULD invalidate any entities referred to by the Request-
 URI.

17. Status Code Definitions

 Where applicable, HTTP status codes (see Section 6 of [RFC7231]) are
 reused.  See Table 4 in Section 8.1 for a listing of which status
 codes may be returned by which requests.  All error messages, 4xx and
 5xx, MAY return a body containing further information about the
 error.

17.1. Informational 1xx

17.1.1. 100 Continue

 The requesting agent SHOULD continue with its request.  This interim
 response is used to inform the requesting agent that the initial part
 of the request has been received and has not yet been rejected by the
 responding agent.  The requesting agent SHOULD continue by sending
 the remainder of the request or, if the request has already been
 completed, continue to wait for a final response (see Section 10.4).
 The responding agent MUST send a final response after the request has
 been completed.

17.2. Success 2xx

 This class of status code indicates that the agent's request was
 successfully received, understood, and accepted.

17.2.1. 200 OK

 The request has succeeded.  The information returned with the
 response is dependent on the method used in the request.

17.3. Redirection 3xx

 The notation "3xx" indicates response codes from 300 to 399 inclusive
 that are meant for redirection.  We use the notation "3rr" to
 indicate all 3xx codes used for redirection, i.e., excluding 304.
 The 304 response code appears here, rather than a 2xx response code,
 which would have been appropriate; 304 has also been used in RTSP 1.0
 [RFC2326].

Schulzrinne, et al. Standards Track [Page 113] RFC 7826 RTSP 2.0 December 2016

 Within RTSP, redirection may be used for load-balancing or
 redirecting stream requests to a server topologically closer to the
 agent.  Mechanisms to determine topological proximity are beyond the
 scope of this specification.
 A 3rr code MAY be used to respond to any request.  The Location
 header MUST be included in any 3rr response.  It is RECOMMENDED that
 they are used if necessary before a session is established, i.e., in
 response to DESCRIBE or SETUP.  However, in cases where a server is
 not able to send a REDIRECT request to the agent, the server MAY need
 to resort to using 3rr responses to inform an agent with an
 established session about the need for redirecting the session.  If a
 3rr response is received for a request in relation to an established
 session, the agent SHOULD send a TEARDOWN request for the session and
 MAY reestablish the session using the resource indicated by the
 Location.
 If the Location header is used in a response, it MUST contain an
 absolute URI pointing out the media resource the agent is redirected
 to; the URI MUST NOT only contain the hostname.
 In the event that an unknown 3rr status code is received, the agent
 SHOULD behave as if a 302 response code had been received
 (Section 17.3.3).

17.3.1. 300

 The 300 response code is not used in RTSP 2.0.

17.3.2. 301 Moved Permanently

 The requested resource is moved permanently and resides now at the
 URI given by the Location header.  The user agent SHOULD redirect
 automatically to the given URI.  This response MUST NOT contain a
 message body.  The Location header MUST be included in the response.

17.3.3. 302 Found

 The requested resource resides temporarily at the URI given by the
 Location header.  This response is intended to be used for many types
 of temporary redirects, e.g., load balancing.  It is RECOMMENDED that
 the server set the reason phrase to something more meaningful than
 "Found" in these cases.  The Location header MUST be included in the
 response.  The user agent SHOULD redirect automatically to the given
 URI.  This response MUST NOT contain a message body.

Schulzrinne, et al. Standards Track [Page 114] RFC 7826 RTSP 2.0 December 2016

 This example shows a client being redirected to a different server:
   C->S: SETUP rtsp://example.com/fizzle/foo RTSP/2.0
         CSeq: 2
         Transport: RTP/AVP/TCP;unicast;interleaved=0-1
         Accept-Ranges: npt, smpte, clock
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 302 Try Other Server
         CSeq: 2
         Location: rtsp://s2.example.com:8001/fizzle/foo

17.3.4. 303 See Other

 This status code MUST NOT be used in RTSP 2.0.  However, it was
 allowed in RTSP 1.0 [RFC2326].

17.3.5. 304 Not Modified

 If the agent has performed a conditional DESCRIBE or SETUP (see
 Sections 18.25 and 18.26) and the requested resource has not been
 modified, the server SHOULD send a 304 response.  This response MUST
 NOT contain a message body.
 The response MUST include the following header fields:
 o  Date
 o  MTag or Content-Location, if the headers would have been sent in a
    200 response to the same request.
 o  Expires and Cache-Control if the field-value might differ from
    that sent in any previous response for the same variant.
 This response is independent for the DESCRIBE and SETUP requests.
 That is, a 304 response to DESCRIBE does NOT imply that the resource
 content is unchanged (only the session description) and a 304
 response to SETUP does NOT imply that the resource description is
 unchanged.  The MTag and If-Match header (Section 18.24) may be used
 to link the DESCRIBE and SETUP in this manner.

17.3.6. 305 Use Proxy

 The requested resource MUST be accessed through the proxy given by
 the Location header that MUST be included.  The Location header
 field-value gives the URI of the proxy.  The recipient is expected to
 repeat this single request via the proxy. 305 responses MUST only be
 generated by origin servers.

Schulzrinne, et al. Standards Track [Page 115] RFC 7826 RTSP 2.0 December 2016

17.4. Client Error 4xx

17.4.1. 400 Bad Request

 The request could not be understood by the agent due to malformed
 syntax.  The agent SHOULD NOT repeat the request without
 modifications.  If the request does not have a CSeq header, the agent
 MUST NOT include a CSeq in the response.

17.4.2. 401 Unauthorized

 The request requires user authentication using the HTTP
 authentication mechanism [RFC7235].  The usage of the error code is
 defined in [RFC7235] and any applicable HTTP authentication scheme,
 such as Digest [RFC7616].  The response is to include a WWW-
 Authenticate header (Section 18.58) field containing a challenge
 applicable to the requested resource.  The agent can repeat the
 request with a suitable Authorization header field.  If the request
 already included authorization credentials, then the 401 response
 indicates that authorization has been refused for those credentials.
 If the 401 response contains the same challenge as the prior
 response, and the user agent has already attempted authentication at
 least once, then the user SHOULD be presented the message body that
 was given in the response, since that message body might include
 relevant diagnostic information.

17.4.3. 402 Payment Required

 This code is reserved for future use.

17.4.4. 403 Forbidden

 The agent understood the request, but is refusing to fulfill it.
 Authorization will not help, and the request SHOULD NOT be repeated.
 If the agent wishes to make public why the request has not been
 fulfilled, it SHOULD describe the reason for the refusal in the
 message body.  If the agent does not wish to make this information
 available to the agent, the status code 404 (Not Found) can be used
 instead.

17.4.5. 404 Not Found

 The agent has not found anything matching the Request-URI.  No
 indication is given of whether the condition is temporary or
 permanent.  The 410 (Gone) status code SHOULD be used if the agent
 knows, through some internally configurable mechanism, that an old
 resource is permanently unavailable and has no forwarding address.

Schulzrinne, et al. Standards Track [Page 116] RFC 7826 RTSP 2.0 December 2016

 This status code is commonly used when the agent does not wish to
 reveal exactly why the request has been refused, or when no other
 response is applicable.

17.4.6. 405 Method Not Allowed

 The method specified in the request is not allowed for the resource
 identified by the Request-URI.  The response MUST include an Allow
 header containing a list of valid methods for the requested resource.
 This status code is also to be used if a request attempts to use a
 method not indicated during SETUP.

17.4.7. 406 Not Acceptable

 The resource identified by the request is only capable of generating
 response message bodies that have content characteristics not
 acceptable according to the Accept headers sent in the request.
 The response SHOULD include a message body containing a list of
 available message body characteristics and location(s) from which the
 user or user agent can choose the one most appropriate.  The message
 body format is specified by the media type given in the Content-Type
 header field.  Depending upon the format and the capabilities of the
 user agent, selection of the most appropriate choice MAY be performed
 automatically.  However, this specification does not define any
 standard for such automatic selection.
 If the response could be unacceptable, a user agent SHOULD
 temporarily stop receipt of more data and query the user for a
 decision on further actions.

17.4.8. 407 Proxy Authentication Required

 This code is similar to 401 (Unauthorized) (Section 17.4.2), but it
 indicates that the client must first authenticate itself with the
 proxy.  The usage of this error code is defined in [RFC7235] and any
 applicable HTTP authentication scheme, such as Digest [RFC7616].  The
 proxy MUST return a Proxy-Authenticate header field (Section 18.34)
 containing a challenge applicable to the proxy for the requested
 resource.

17.4.9. 408 Request Timeout

 The agent did not produce a request within the time that the agent
 was prepared to wait.  The agent MAY repeat the request without
 modifications at any later time.

Schulzrinne, et al. Standards Track [Page 117] RFC 7826 RTSP 2.0 December 2016

17.4.10. 410 Gone

 The requested resource is no longer available at the server and the
 forwarding address is not known.  This condition is expected to be
 considered permanent.  If the server does not know, or has no
 facility to determine, whether or not the condition is permanent, the
 status code 404 (Not Found) SHOULD be used instead.  This response is
 cacheable unless indicated otherwise.
 The 410 response is primarily intended to assist the task of
 repository maintenance by notifying the recipient that the resource
 is intentionally unavailable and that the server owners desire that
 remote links to that resource be removed.  Such an event is common
 for limited-time, promotional services and for resources belonging to
 individuals no longer working at the server's site.  It is not
 necessary to mark all permanently unavailable resources as "gone" or
 to keep the mark for any length of time -- that is left to the
 discretion of the owner of the server.

17.4.11. 412 Precondition Failed

 The precondition given in one or more of the 'if-' request-header
 fields evaluated to false when it was tested on the agent.  See these
 sections for the 'if-' headers: If-Match Section 18.24, If-Modified-
 Since Section 18.25, and If-None-Match Section 18.26.  This response
 code allows the agent to place preconditions on the current resource
 meta-information (header field data) and, thus, prevent the requested
 method from being applied to a resource other than the one intended.

17.4.12. 413 Request Message Body Too Large

 The agent is refusing to process a request because the request
 message body is larger than the agent is willing or able to process.
 The agent MAY close the connection to prevent the requesting agent
 from continuing the request.
 If the condition is temporary, the agent SHOULD include a Retry-After
 header field to indicate that it is temporary and after what time the
 requesting agent MAY try again.

17.4.13. 414 Request-URI Too Long

 The responding agent is refusing to service the request because the
 Request-URI is longer than the agent is willing to interpret.  This
 rare condition is only likely to occur when an agent has used a
 request with long query information, when the agent has descended
 into a URI "black hole" of redirection (e.g., a redirected URI prefix
 that points to a suffix of itself), or when the agent is under attack

Schulzrinne, et al. Standards Track [Page 118] RFC 7826 RTSP 2.0 December 2016

 by an agent attempting to exploit security holes present in some
 agents using fixed-length buffers for reading or manipulating the
 Request-URI.

17.4.14. 415 Unsupported Media Type

 The server is refusing to service the request because the message
 body of the request is in a format not supported by the requested
 resource for the requested method.

17.4.15. 451 Parameter Not Understood

 The recipient of the request does not support one or more parameters
 contained in the request.  When returning this error message the
 agent SHOULD return a message body containing the offending
 parameter(s).

17.4.16. 452 Illegal Conference Identifier

 This status code MUST NOT be used in RTSP 2.0.  However, it was
 allowed in RTSP 1.0 [RFC2326].

17.4.17. 453 Not Enough Bandwidth

 The request was refused because there was insufficient bandwidth.
 This may, for example, be the result of a resource reservation
 failure.

17.4.18. 454 Session Not Found

 The RTSP session identifier in the Session header is missing, is
 invalid, or has timed out.

17.4.19. 455 Method Not Valid in This State

 The agent cannot process this request in its current state.  The
 response MUST contain an Allow header to make error recovery
 possible.

17.4.20. 456 Header Field Not Valid for Resource

 The targeted agent could not act on a required request-header.  For
 example, if PLAY request contains the Range header field but the
 stream does not allow seeking.  This error message may also be used
 for specifying when the time format in Range is impossible for the
 resource.  In that case, the Accept-Ranges header MUST be returned to
 inform the agent of which formats are allowed.

Schulzrinne, et al. Standards Track [Page 119] RFC 7826 RTSP 2.0 December 2016

17.4.21. 457 Invalid Range

 The Range value given is out of bounds, e.g., beyond the end of the
 presentation.

17.4.22. 458 Parameter Is Read-Only

 The parameter to be set by SET_PARAMETER can be read but not
 modified.  When returning this error message, the sender SHOULD
 return a message body containing the offending parameter(s).

17.4.23. 459 Aggregate Operation Not Allowed

 The requested method may not be applied on the URI in question since
 it is an aggregate (presentation) URI.  The method may be applied on
 a media URI.

17.4.24. 460 Only Aggregate Operation Allowed

 The requested method may not be applied on the URI in question since
 it is not an aggregate control (presentation) URI.  The method may be
 applied on the aggregate control URI.

17.4.25. 461 Unsupported Transport

 The Transport field did not contain a supported transport
 specification.

17.4.26. 462 Destination Unreachable

 The data transmission channel could not be established because the
 agent address could not be reached.  This error will most likely be
 the result of an agent attempt to place an invalid dest_addr
 parameter in the Transport field.

17.4.27. 463 Destination Prohibited

 The data transmission channel was not established because the server
 prohibited access to the agent address.  This error is most likely
 the result of an agent attempt to redirect media traffic to another
 destination with a dest_addr parameter in the Transport header.

Schulzrinne, et al. Standards Track [Page 120] RFC 7826 RTSP 2.0 December 2016

17.4.28. 464 Data Transport Not Ready Yet

 The data transmission channel to the media destination is not yet
 ready for carrying data.  However, the responding agent still expects
 that the data transmission channel will be established at some point
 in time.  Note, however, that this may result in a permanent failure
 like 462 (Destination Unreachable).
 An example of when this error may occur is in the case in which a
 client sends a PLAY request to a server prior to ensuring that the
 TCP connections negotiated for carrying media data were successfully
 established (in violation of this specification).  The server would
 use this error code to indicate that the requested action could not
 be performed due to the failure of completing the connection
 establishment.

17.4.29. 465 Notification Reason Unknown

 This indicates that the client has received a PLAY_NOTIFY
 (Section 13.5) with a Notify-Reason header (Section 18.32) unknown to
 the client.

17.4.30. 466 Key Management Error

 This indicates that there has been an error in a Key Management
 function used in conjunction with a request.  For example, usage of
 Multimedia Internet KEYing (MIKEY) [RFC3830] according to
 Appendix C.1.4.1 may result in this error.

17.4.31. 470 Connection Authorization Required

 The secured connection attempt needs user or client authorization
 before proceeding.  The next hop's certificate is included in this
 response in the Accept-Credentials header.

17.4.32. 471 Connection Credentials Not Accepted

 When performing a secure connection over multiple connections, an
 intermediary has refused to connect to the next hop and carry out the
 request due to unacceptable credentials for the used policy.

17.4.33. 472 Failure to Establish Secure Connection

 A proxy fails to establish a secure connection to the next-hop RTSP
 agent.  This is primarily caused by a fatal failure at the TLS
 handshake, for example, due to the agent not accepting any cipher
 suites.

Schulzrinne, et al. Standards Track [Page 121] RFC 7826 RTSP 2.0 December 2016

17.5. Server Error 5xx

 Response status codes beginning with the digit "5" indicate cases in
 which the server is aware that it has erred or is incapable of
 performing the request.  The server SHOULD include a message body
 containing an explanation of the error situation and whether it is a
 temporary or permanent condition.  User agents SHOULD display any
 included message body to the user.  These response codes are
 applicable to any request method.

17.5.1. 500 Internal Server Error

 The agent encountered an unexpected condition that prevented it from
 fulfilling the request.

17.5.2. 501 Not Implemented

 The agent does not support the functionality required to fulfill the
 request.  This is the appropriate response when the agent does not
 recognize the request method and is not capable of supporting it for
 any resource.

17.5.3. 502 Bad Gateway

 The agent, while acting as a gateway or proxy, received an invalid
 response from the upstream agent it accessed in attempting to fulfill
 the request.

17.5.4. 503 Service Unavailable

 The server is currently unable to handle the request due to a
 temporary overloading or maintenance of the server.  The implication
 is that this is a temporary condition that will be alleviated after
 some delay.  If known, the length of the delay MAY be indicated in a
 Retry-After header.  If no Retry-After is given, the agent SHOULD
 handle the response as it would for a 500 response.  The agent MUST
 honor the length, if given, in the Retry-After header.
       Note: The existence of the 503 status code does not imply that
       a server must use it when becoming overloaded.  Some servers
       may wish to simply refuse the transport connection.
 The response scope is dependent on the request.  If the request was
 in relation to an existing RTSP session, the scope of the overload
 response is to this individual RTSP session.  If the request was not
 session specific or intended to form an RTSP session, it applies to
 the RTSP server identified by the hostname in the Request-URI.

Schulzrinne, et al. Standards Track [Page 122] RFC 7826 RTSP 2.0 December 2016

17.5.5. 504 Gateway Timeout

 The agent, while acting as a proxy, did not receive a timely response
 from the upstream agent specified by the URI or some other auxiliary
 server (e.g., DNS) that it needed to access in attempting to complete
 the request.

17.5.6. 505 RTSP Version Not Supported

 The agent does not support, or refuses to support, the RTSP version
 that was used in the request message.  The agent is indicating that
 it is unable or unwilling to complete the request using the same
 major version as the agent other than with this error message.  The
 response SHOULD contain a message body describing why that version is
 not supported and what other protocols are supported by that agent.

17.5.7. 551 Option Not Supported

 A feature tag given in the Require or the Proxy-Require fields was
 not supported.  The Unsupported header MUST be returned stating the
 feature for which there is no support.

17.5.8. 553 Proxy Unavailable

 The proxy is currently unable to handle the request due to a
 temporary overloading or maintenance of the proxy.  The implication
 is that this is a temporary condition that will be alleviated after
 some delay.  If known, the length of the delay MAY be indicated in a
 Retry-After header.  If no Retry-After is given, the agent SHOULD
 handle the response as it would for a 500 response.  The agent MUST
 honor the length, if given in the Retry-After header.
       Note: The existence of the 553 status code does not imply that
       a proxy must use it when becoming overloaded.  Some proxies may
       wish to simply refuse the connection.
 The response scope is dependent on the Request.  If the request was
 in relation to an existing RTSP session, the scope of the overload
 response is to this individual RTSP session.  If the request was non-
 session specific or intended to form an RTSP session, it applies to
 all such requests to this proxy.

Schulzrinne, et al. Standards Track [Page 123] RFC 7826 RTSP 2.0 December 2016

18. Header Field Definitions

     +---------------+----------------+--------+---------+------+
     | method        | direction      | object | acronym | Body |
     +---------------+----------------+--------+---------+------+
     | DESCRIBE      | C -> S         | P,S    | DES     | r    |
     |               |                |        |         |      |
     | GET_PARAMETER | C -> S, S -> C | P,S    | GPR     | R,r  |
     |               |                |        |         |      |
     | OPTIONS       | C -> S, S -> C | P,S    | OPT     |      |
     |               |                |        |         |      |
     | PAUSE         | C -> S         | P,S    | PSE     |      |
     |               |                |        |         |      |
     | PLAY          | C -> S         | P,S    | PLY     |      |
     |               |                |        |         |      |
     | PLAY_NOTIFY   | S -> C         | P,S    | PNY     | R    |
     |               |                |        |         |      |
     | REDIRECT      | S -> C         | P,S    | RDR     |      |
     |               |                |        |         |      |
     | SETUP         | C -> S         | S      | STP     |      |
     |               |                |        |         |      |
     | SET_PARAMETER | C -> S, S -> C | P,S    | SPR     | R,r  |
     |               |                |        |         |      |
     | TEARDOWN      | C -> S         | P,S    | TRD     |      |
     |               |                |        |         |      |
     |               | S -> C         | P      | TRD     |      |
     +---------------+----------------+--------+---------+------+
 This table is an overview of RTSP methods, their direction, and what
 objects (P: presentation, S: stream) they operate on.  "Body" denotes
   if a method is allowed to carry body and in which direction; R =
  request, r=response.  Note: All error messages for statuses 4xx and
                   5xx are allowed to carry a body.
                   Table 8: Overview of RTSP Methods
 The general syntax for header fields is covered in Section 5.2.  This
 section lists the full set of header fields along with notes on
 meaning and usage.  The syntax definitions for header fields are
 present in Section 20.2.3.  Examples of each header field are given.
 Information about header fields in relation to methods and proxy
 processing is summarized in Figures 2, 3, 4, and 5.

Schulzrinne, et al. Standards Track [Page 124] RFC 7826 RTSP 2.0 December 2016

 The "where" column describes the request and response types in which
 the header field can be used.  Values in this column are:
 R:                header field may only appear in requests;
 r:                header field may only appear in responses;
 2xx, 4xx, etc.:   numerical value or range indicates response codes
                   with which the header field can be used;
 c:                header field is copied from the request to the
                   response.
 G:                header field is a general-header and may be present
                   in both requests and responses.
 Note: General headers do not always use the "G" value in the "where"
 column.  This is due to differences when the header may be applied in
 requests compared to responses.  When such differences exist, they
 are expressed using two different rows: one with "where" being "R"
 and one with it being "r".
 The "proxy" column describes the operations a proxy may perform on a
 header field.  An empty proxy column indicates that the proxy MUST
 NOT make any changes to that header, all allowed operations are
 explicitly stated:
 a:    A proxy can add or concatenate the header field if not present.
 m:    A proxy can modify an existing header field value.
 d:    A proxy can delete a header field-value.
 r:    A proxy needs to be able to read the header field; thus, this
       header field cannot be encrypted.
 The rest of the columns relate to the presence of a header field in a
 method.  The method names when abbreviated, are according to Table 8:
 c:    Conditional; requirements on the header field depend on the
       context of the message.
 m:    The header field is mandatory.
 m*:   The header field SHOULD be sent, but agents need to be prepared
       to receive messages without that header field.
 o:    The header field is optional.

Schulzrinne, et al. Standards Track [Page 125] RFC 7826 RTSP 2.0 December 2016

  • : The header field MUST be present if the message body is not

empty. See Sections 18.17, 18.19 and 5.3 for details.

  1. : The header field is not applicable.
 "Optional" means that an agent MAY include the header field in a
 request or response.  The agent behavior when receiving such headers
 varies; for some, it may ignore the header field.  In other cases, it
 is a request to process the header.  This is regulated by the method
 and header descriptions.  Examples of headers that require processing
 are the Require and Proxy-Require header fields discussed in Sections
 18.43 and 18.37.  A "mandatory" header field MUST be present in a
 request, and it MUST be understood by the agent receiving the
 request.  A mandatory response-header field MUST be present in the
 response, and the header field MUST be understood by the processing
 the response.  "Not applicable" means that the header field MUST NOT
 be present in a request.  If one is placed in a request by mistake,
 it MUST be ignored by the agent receiving the request.  Similarly, a
 header field labeled "not applicable" for a response means that the
 agent MUST NOT place the header field in the response, and the agent
 MUST ignore the header field in the response.
 An RTSP agent MUST ignore extension headers that are not understood.
 The From and Location header fields contain a URI.  If the URI
 contains a comma (') or semicolon (;), the URI MUST be enclosed in
 double quotes (").  Any URI parameters are contained within these
 quotes.  If the URI is not enclosed in double quotes, any semicolon-
 delimited parameters are header-parameters, not URI parameters.

Schulzrinne, et al. Standards Track [Page 126] RFC 7826 RTSP 2.0 December 2016

 +-------------------+------+------+----+----+-----+-----+-----+-----+
 | Header            |Where |Proxy |DES | OPT| STP | PLY | PSE | TRD |
 +-------------------+------+------+----+----+-----+-----+-----+-----+
 | Accept            | R    |      | o  | -  | -   | -   | -   | -   |
 | Accept-           | R    | rm   | o  | o  | o   | o   | o   | o   |
 | Credentials       |      |      |    |    |     |     |     |     |
 | Accept-Encoding   | R    | r    | o  | -  | -   | -   | -   | -   |
 | Accept-Language   | R    | r    | o  | -  | -   | -   | -   | -   |
 | Accept-Ranges     | G    | r    | -  | -  | m   | -   | -   | -   |
 | Accept-Ranges     | 456  | r    | -  | -  | -   | m   | -   | -   |
 | Allow             | r    | am   | c  | c  | c   | -   | -   | -   |
 | Allow             | 405  | am   | m  | m  | m   | m   | m   | m   |
 | Authentication-   | r    |      | o  | o  | o   | o   | o   | o/- |
 | Info              |      |      |    |    |     |     |     |     |
 | Authorization     | R    |      | o  | o  | o   | o   | o   | o/- |
 | Bandwidth         | R    |      | o  | o  | o   | o   | -   | -   |
 | Blocksize         | R    |      | o  | -  | o   | o   | -   | -   |
 | Cache-Control     | G    | r    | o  | -  | o   | -   | -   | -   |
 | Connection        | G    | ad   | o  | o  | o   | o   | o   | o   |
 | Connection-       | 470, | ar   | o  | o  | o   | o   | o   | o   |
 | Credentials       | 407  |      |    |    |     |     |     |     |
 | Content-Base      | r    |      | o  | -  | -   | -   | -   | -   |
 | Content-Base      | 4xx, |      | o  | o  | o   | o   | o   | o   |
 |                   | 5xx  |      |    |    |     |     |     |     |
 | Content-Encoding  | R    | r    | -  | -  | -   | -   | -   | -   |
 | Content-Encoding  | r    | r    | o  | -  | -   | -   | -   | -   |
 | Content-Encoding  | 4xx, | r    | o  | o  | o   | o   | o   | o   |
 |                   | 5xx  |      |    |    |     |     |     |     |
 | Content-Language  | R    | r    | -  | -  | -   | -   | -   | -   |
 | Content-Language  | r    | r    | o  | -  | -   | -   | -   | -   |
 | Content-Language  | 4xx, | r    | o  | o  | o   | o   | o   | o   |
 |                   | 5xx  |      |    |    |     |     |     |     |
 | Content-Length    | r    | r    | *  | -  | -   | -   | -   | -   |
 | Content-Length    | 4xx, | r    | *  | *  | *   | *   | *   | *   |
 |                   | 5xx  |      |    |    |     |     |     |     |
 | Content-Location  | r    | r    | o  | -  | -   | -   | -   | -   |
 | Content-Location  | 4xx, | r    | o  | o  | o   | o   | o   | o   |
 |                   | 5xx  |      |    |    |     |     |     |     |
 | Content-Type      | r    | r    | *  | -  | -   | -   | -   | -   |
 | Content-Type      | 4xx, | ar   | *  | *  | *   | *   | *   | *   |
 |                   | 5xx  |      |    |    |     |     |     |     |
 | CSeq              | Gc   | rm   | m  | m  | m   | m   | m   | m   |
 | Date              | G    | am   | o/*| o/*| o/* | o/* | o/* | o/* |
 | Expires           | r    | r    | o  | -  | o   | -   | -   | -   |
 | From              | R    | r    | o  | o  | o   | o   | o   | o   |
 | If-Match          | R    | r    | -  | -  | o   | -   | -   | -   |
 | If-Modified-Since | R    | r    | o  | -  | o   | -   | -   | -   |
 | If-None-Match     | R    | r    | o  | -  | o   | -   | -   | -   |

Schulzrinne, et al. Standards Track [Page 127] RFC 7826 RTSP 2.0 December 2016

 | Last-Modified     | r    | r    | o  | -  | o   | -   | -   | -   |
 | Location          | 3rr  |      | m  | m  | m   | m   | m   | m   |
 +-------------------+------+------+----+----+-----+-----+-----+-----+
 | Header            |Where |Proxy |DES | OPT| STP | PLY | PSE | TRD |
 +-------------------+------+------+----+----+-----+-----+-----+-----+
   Figure 2: Overview of RTSP Header Fields (A-L) Related to Methods
          DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN

Schulzrinne, et al. Standards Track [Page 128] RFC 7826 RTSP 2.0 December 2016

 +------------------+---------+-----+----+----+----+-----+-----+-----+
 | Header           | Where   |Proxy|DES |OPT |STP | PLY | PSE | TRD |
 +------------------+---------+-----+----+----+----+-----+-----+-----+
 | Media-Properties | r       |     | -  | -  | m  | o   | o   | -   |
 | Media-Range      | r       |     | -  | -  | c  | c   | c   | -   |
 | MTag             | r       | r   | o  | -  | o  | -   | -   | -   |
 | Pipelined-       | G       | amd | -  | o  | o  | o   | o   | o   |
 | Requests         |         | r   |    |    |    |     |     |     |
 | Proxy-           | 407     | amr | m  | m  | m  | m   | m   | m   |
 | Authenticate     |         |     |    |    |    |     |     |     |
 | Proxy-           | r       | amd | o  | o  | o  | o   | o   | o/- |
 | Authentication-  |         | r   |    |    |    |     |     |     |
 | Info             |         |     |    |    |    |     |     |     |
 | Proxy-           | R       | rd  | o  | o  | o  | o   | o   | o   |
 | Authorization    |         |     |    |    |    |     |     |     |
 | Proxy-Require    | R       | ar  | o  | o  | o  | o   | o   | o   |
 | Proxy-Require    | r       | r   | c  | c  | c  | c   | c   | c   |
 | Proxy-Supported  | R       | amr | c  | c  | c  | c   | c   | c   |
 | Proxy-Supported  | r       |     | c  | c  | c  | c   | c   | c   |
 | Public           | r       | amr | -  | m  | -  | -   | -   | -   |
 | Public           | 501     | amr | m  | m  | m  | m   | m   | m   |
 | Range            | R       |     | -  | -  | -  | o   | -   | -   |
 | Range            | r       |     | -  | -  | c  | m   | m   | -   |
 | Referrer         | R       |     | o  | o  | o  | o   | o   | o   |
 | Request-Status   | R       |     | -  | -  | -  | -   | -   | -   |
 | Require          | R       |     | o  | o  | o  | o   | o   | o   |
 | Retry-After      | 3rr,503 |     | o  | o  | o  | o   | o   | -   |
 |                  | ,553    |     |    |    |    |     |     |     |
 | Retry-After      | 413     |     | o  | -  | -  | -   | -   | -   |
 | RTP-Info         | r       |     | -  | -  | c  | c   | -   | -   |
 | Scale            | R       | r   | -  | -  | -  | o   | -   | -   |
 | Scale            | r       | amr | -  | -  | c  | c   | c   | -   |
 | Seek-Style       | R       |     | -  | -  | -  | o   | -   | -   |
 | Seek-Style       | r       |     | -  | -  | -  | m   | -   | -   |
 | Server           | R       | r   | -  | o  | -  | -   | -   | o   |
 | Server           | r       | r   | o  | o  | o  | o   | o   | o   |
 | Session          | R       | r   | -  | o  | o  | m   | m   | m   |
 | Session          | r       | r   | -  | c  | m  | m   | m   | o   |
 | Speed            | R       | admr| -  | -  | -  | o   | -   | -   |
 | Speed            | r       | admr| -  | -  | -  | c   | -   | -   |
 | Supported        | R       | r   | o  | o  | o  | o   | o   | o   |
 | Supported        | r       | r   | c  | c  | c  | c   | c   | c   |
 | Terminate-Reason | R       | r   | -  | -  | -  | -   | -   | -/o |
 | Timestamp        | R       | admr| o  | o  | o  | o   | o   | o   |
 | Timestamp        | c       | admr| m  | m  | m  | m   | m   | m   |
 | Transport        | G       | mr  | -  | -  | m  | -   | -   | -   |
 | Unsupported      | r       |     | c  | c  | c  | c   | c   | c   |
 | User-Agent       | R       |     | m* | m* | m* | m*  | m*  | m*  |

Schulzrinne, et al. Standards Track [Page 129] RFC 7826 RTSP 2.0 December 2016

 | Via              | R       | amr | c  | c  | c  | c   | c   | c   |
 | Via              | r       | amr | c  | c  | c  | c   | c   | c   |
 | WWW-Authenticate | 401     |     | m  | m  | m  | m   | m   | m   |
 +------------------+---------+-----+----+----+----+-----+-----+-----+
 | Header           | Where   |Proxy|DES |OPT |STP | PLY | PSE | TRD |
 +------------------+---------+-----+----+----+----+-----+-----+-----+
   Figure 3: Overview of RTSP Header Fields (M-W) Related to Methods
          DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN

Schulzrinne, et al. Standards Track [Page 130] RFC 7826 RTSP 2.0 December 2016

 +---------------------------+-------+-------+-----+-----+-----+-----+
 | Header                    | Where | Proxy | GPR | SPR | RDR | PNY |
 +---------------------------+-------+-------+-----+-----+-----+-----+
 | Accept-Credentials        | R     | rm    | o   | o   | o   | -   |
 | Accept-Encoding           | R     | r     | o   | o   | o   | -   |
 | Accept-Language           | R     | r     | o   | o   | o   | -   |
 | Accept-Ranges             | G     | rm    | o   | -   | -   | -   |
 | Allow                     | 405   | amr   | m   | m   | m   | m   |
 | Authentication-Info       | r     |       | o/- | o/- | -   | -   |
 | Authorization             | R     |       | o   | o   | o   | -   |
 | Bandwidth                 | R     |       | -   | o   | -   | -   |
 | Blocksize                 | R     |       | -   | o   | -   | -   |
 | Cache-Control             | G     | r     | o   | o   | -   | -   |
 | Connection                | G     |       | o   | o   | o   | o   |
 | Connection-Credentials    | 470,  | ar    | o   | o   | o   | -   |
 |                           | 407   |       |     |     |     |     |
 | Content-Base              | R     |       | o   | o   | -   | o   |
 | Content-Base              | r     |       | o   | o   | -   | -   |
 | Content-Base              | 4xx,  |       | o   | o   | o   | o   |
 |                           | 5xx   |       |     |     |     |     |
 | Content-Encoding          | R     | r     | o   | o   | -   | o   |
 | Content-Encoding          | r     | r     | o   | o   | -   | -   |
 | Content-Encoding          | 4xx,  | r     | o   | o   | o   | o   |
 |                           | 5xx   |       |     |     |     |     |
 | Content-Language          | R     | r     | o   | o   | -   | o   |
 | Content-Language          | r     | r     | o   | o   | -   | -   |
 | Content-Language          | 4xx,  | r     | o   | o   | o   | o   |
 |                           | 5xx   |       |     |     |     |     |
 | Content-Length            | R     | r     | *   | *   | -   | *   |
 | Content-Length            | r     | r     | *   | *   | -   | -   |
 | Content-Length            | 4xx,  | r     | *   | *   | *   | *   |
 |                           | 5xx   |       |     |     |     |     |
 | Content-Location          | R     |       | o   | o   | -   | o   |
 | Content-Location          | r     |       | o   | o   | -   | -   |
 | Content-Location          | 4xx,  |       | o   | o   | o   | o   |
 |                           | 5xx   |       |     |     |     |     |
 | Content-Type              | R     |       | *   | *   | -   | *   |
 | Content-Type              | r     |       | *   | *   | -   | -   |
 | Content-Type              | 4xx,  |       | *   | *   | *   | *   |
 |                           | 5xx   |       |     |     |     |     |
 | CSeq                      | R,c   | mr    | m   | m   | m   | m   |
 | Date                      | R     | a     | o/* | o/* | m   | o/* |
 | Date                      | r     | am    | o/* | o/* | o/* | o/* |
 | Expires                   | r     | r     | -   | -   | -   | -   |
 | From                      | R     | r     | o   | o   | o   | -   |
 | If-Match                  | R     | r     | -   | -   | -   | -   |
 | If-Modified-Since         | R     | am    | o   | -   | -   | -   |
 | If-None-Match             | R     | am    | o   | -   | -   | -   |

Schulzrinne, et al. Standards Track [Page 131] RFC 7826 RTSP 2.0 December 2016

 | Last-Modified             | R     | r     | -   | -   | -   | -   |
 | Last-Modified             | r     | r     | o   | -   | -   | -   |
 | Location                  | 3rr   |       | m   | m   | m   | -   |
 | Location                  | R     |       | -   | -   | m   | -   |
 +---------------------------+-------+-------+-----+-----+-----+-----+
 | Header                    | Where | Proxy | GPR | SPR | RDR | PNY |
 +---------------------------+-------+-------+-----+-----+-----+-----+
   Figure 4: Overview of RTSP Header Fields (A-L) Related to Methods
        GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY

Schulzrinne, et al. Standards Track [Page 132] RFC 7826 RTSP 2.0 December 2016

+—————————+———+——-+—–+—–+—–+—–+ | Header | Where | Proxy | GPR | SPR | RDR | PNY | +—————————+———+——-+—–+—–+—–+—–+ | Media-Properties | R | amr | o | - | - | c | | Media-Properties | r | mr | c | - | - | - | | Media-Range | R | | o | - | - | c | | Media-Range | r | | c | - | - | - | | MTag | r | r | o | - | - | - | | Notify-Reason | R | | - | - | - | m | | Pipelined-Requests | R | amdr | o | o | - | - | | Proxy-Authenticate | 407 | amdr | m | m | m | - | | Proxy-Authentication-Info | r | amdr | o/- | o/- | - | - | | Proxy-Authorization | R | amdr | o | o | o | - | | Proxy-Require | R | ar | o | o | o | - | | Proxy-Supported | R | amr | c | c | c | - | | Proxy-Supported | r | | c | c | c | - | | Public | 501 | admr | m | m | m | - | | Range | R | | o | - | - | m | | Range | r | | c | - | - | - | | Referrer | R | | o | o | o | - | | Request-Status | R | mr | - | - | - | c | | Require | R | r | o | o | o | o | | Retry-After | 3rr,503,| | o | o | - | - | | | 553 | | | | | | | Retry-After | 413 | | o | o | - | - | | RTP-Info | R | r | o | - | - | C | | RTP-Info | r | r | c | - | - | - | | Scale | G | | c | - | c | c | | Seek-Style | G | | - | - | - | - | | Server | R | r | o | o | o | o | | Server | r | r | o | o | - | - | | Session | R | r | o | o | o | m | | Session | r | r | c | c | o | m | | Speed | G | | - | - | - | - | | Supported | R | r | o | o | o | - | | Supported | r | r | c | c | c | - | | Terminate-Reason | R | r | - | - | m | - | | Timestamp | R | adrm | o | o | o | o | | Timestamp | c | adrm | m | m | m | m | | Transport | G | mr | - | - | - | - | | Unsupported | r | arm | c | c | c | c | | User-Agent | R | r | m* | m* | - | - | | User-Agent | r | r | m* | m* | m* | m* | | Via | R | amr | c | c | c | c |

Schulzrinne, et al. Standards Track [Page 133] RFC 7826 RTSP 2.0 December 2016

| Via | r | amr | c | c | c | c | | WWW-Authenticate | 401 | | m | m | m | - | +—————————+———+——-+—–+—–+—–+—–+ | Header | Where | Proxy | GPR | SPR | RDR | PNY | +—————————+———+——-+—–+—–+—–+—–+

   Figure 5: Overview of RTSP Header Fields (M-W) Related to Methods
        GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY

18.1. Accept

 The Accept request-header field can be used to specify certain
 presentation description and parameter media types [RFC6838] that are
 acceptable for the response to the DESCRIBE request.
 See Section 20.2.3 for the syntax.
 The asterisk "*" character is used to group media types into ranges,
 with "*/*" indicating all media types and "type/*" indicating all
 subtypes of that type.  The range MAY include media type parameters
 that are generally applicable to that range.
 Each media type or range MAY be followed by one or more accept-
 params, beginning with the "q" parameter to indicate a relative
 quality factor.  The first "q" parameter (if any) separates the media
 type or range's parameters from the accept-params.  Quality factors
 allow the user or user agent to indicate the relative degree of
 preference for that media type, using the qvalue scale from 0 to 1
 (Section 5.3.1 of [RFC7231]).  The default value is q=1.
 Example of use:
   Accept: application/example ;q=0.7, application/sdp
 Indicates that the requesting agent prefers the media type
 application/sdp through the default 1.0 rating but also accepts the
 application/example media type with a 0.7 quality rating.
 If no Accept header field is present, then it is assumed that the
 client accepts all media types.  If an Accept header field is
 present, and if the server cannot send a response that is acceptable
 according to the combined Accept field-value, then the server SHOULD
 send a 406 (Not Acceptable) response.

Schulzrinne, et al. Standards Track [Page 134] RFC 7826 RTSP 2.0 December 2016

18.2. Accept-Credentials

 The Accept-Credentials header is a request-header used to indicate to
 any trusted intermediary how to handle further secured connections to
 proxies or servers.  It MUST NOT be included in server-to-client
 requests.  See Section 19 for the usage of this header
 In a request, the header MUST contain the method (User, Proxy, or
 Any) for approving credentials selected by the requester.  The method
 MUST NOT be changed by any proxy, unless it is "Proxy" when a proxy
 MAY change it to "user" to take the role of user approving each
 further hop.  If the method is "User", the header contains zero or
 more of the credentials that the client accepts.  The header may
 contain zero credentials in the first RTSP request to an RTSP server
 via a proxy when using the "User" method.  This is because the client
 has not yet received any credentials to accept.  Each credential MUST
 consist of one URI identifying the proxy or server, the hash
 algorithm identifier, and the hash over that agent's ASN.1 DER-
 encoded certificate [RFC5280] in Base64, according to Section 4 of
 [RFC4648] and where the padding bits are set to zero.  All RTSP
 clients and proxies MUST implement the SHA-256 [FIPS180-4] algorithm
 for computation of the hash of the DER-encoded certificate.  The
 SHA-256 algorithm is identified by the token "sha-256".
 The intention of allowing for other hash algorithms is to enable the
 future retirement of algorithms that are not implemented somewhere
 other than here.  Thus, the definition of future algorithms for this
 purpose is intended to be extremely limited.  A feature tag can be
 used to ensure that support for the replacement algorithm exists.
 Example:
 Accept-Credentials:User
   "rtsps://proxy2.example.com/";sha-256;exaIl9VMbQMOFGClx5rXnPJKVNI=,
   "rtsps://server.example.com/";sha-256;lurbjj5khhB0NhIuOXtt4bBRH1M=

18.3. Accept-Encoding

 The Accept-Encoding request-header field is similar to Accept, but it
 restricts the content-codings (see Section 18.15), i.e.,
 transformation codings of the message body, such as gzip compression,
 that are acceptable in the response.

Schulzrinne, et al. Standards Track [Page 135] RFC 7826 RTSP 2.0 December 2016

 A server tests whether a content-coding is acceptable, according to
 an Accept-Encoding field, using these rules:
 1.  If the content-coding is one of the content-codings listed in the
     Accept-Encoding field, then it is acceptable, unless it is
     accompanied by a qvalue of 0.  (As defined in Section 5.3.1 of
     [RFC7231], a qvalue of 0 means "not acceptable.")
 2.  The special "*" symbol in an Accept-Encoding field matches any
     available content-coding not explicitly listed in the header
     field.
 3.  If multiple content-codings are acceptable, then the acceptable
     content-coding with the highest non-zero qvalue is preferred.
 4.  The "identity" content-coding is always acceptable, i.e., no
     transformation at all, unless specifically refused because the
     Accept-Encoding field includes "identity;q=0" or because the
     field includes "*;q=0" and does not explicitly include the
     "identity" content-coding.  If the Accept-Encoding field-value is
     empty, then only the "identity" encoding is acceptable.
 If an Accept-Encoding field is present in a request, and if the
 server cannot send a response that is acceptable according to the
 Accept-Encoding header, then the server SHOULD send an error response
 with the 406 (Not Acceptable) status code.
 If no Accept-Encoding field is present in a request, the server MAY
 assume that the client will accept any content-coding.  In this case,
 if "identity" is one of the available content-codings, then the
 server SHOULD use the "identity" content-coding, unless it has
 additional information that a different content-coding is meaningful
 to the client.

18.4. Accept-Language

 The Accept-Language request-header field is similar to Accept, but
 restricts the set of natural languages that are preferred as a
 response to the request.  Note that the language specified applies to
 the presentation description (response message body) and any reason
 phrases, but not the media content.
 A language tag identifies a natural language spoken, written, or
 otherwise conveyed by human beings for communication of information
 to other human beings.  Computer languages are explicitly excluded.
 The syntax and registry of RTSP 2.0 language tags are the same as
 those defined by [RFC5646].

Schulzrinne, et al. Standards Track [Page 136] RFC 7826 RTSP 2.0 December 2016

 Each language-range MAY be given an associated quality value that
 represents an estimate of the user's preference for the languages
 specified by that range.  The quality value defaults to "q=1".  For
 example:
    Accept-Language: da, en-gb;q=0.8, en;q=0.7
 would mean: "I prefer Danish, but will accept British English and
 other types of English."  A language-range matches a language tag if
 it exactly equals the full tag or if it exactly equals a prefix of
 the tag, i.e., the primary-tag in the ABNF, such that the character
 following primary-tag is "-".  The special range "*", if present in
 the Accept-Language field, matches every tag not matched by any other
 range present in the Accept-Language field.
    Note: This use of a prefix matching rule does not imply that
    language tags are assigned to languages in such a way that it is
    always true that if a user understands a language with a certain
    tag, then this user will also understand all languages with tags
    for which this tag is a prefix.  The prefix rule simply allows the
    use of prefix tags if this is the case.
 In the process of selecting a language, each language tag is assigned
 a qualification factor, i.e., if a language being supported by the
 client is actually supported by the server and what "preference"
 level the language achieves.  The quality value (q-value) of the
 longest language-range in the field that matches the language tag is
 assigned as the qualification factor for a particular language tag.
 If no language-range in the field matches the tag, the language
 qualification factor assigned is 0.  If no Accept-Language header is
 present in the request, the server SHOULD assume that all languages
 are equally acceptable.  If an Accept-Language header is present,
 then all languages that are assigned a qualification factor greater
 than 0 are acceptable.

18.5. Accept-Ranges

 The Accept-Ranges general-header field allows indication of the
 format supported in the Range header.  The client MUST include the
 header in SETUP requests to indicate which formats are acceptable
 when received in PLAY and PAUSE responses and REDIRECT requests.  The
 server MUST include the header in SETUP responses and 456 (Header
 Field Not Valid for Resource) error responses to indicate the formats
 supported for the resource indicated by the Request-URI.  The header
 MAY be included in GET_PARAMETER request and response pairs.  The
 GET_PARAMETER request MUST contain a Session header to identify the

Schulzrinne, et al. Standards Track [Page 137] RFC 7826 RTSP 2.0 December 2016

 session context the request is related to.  The requester and
 responder will indicate their capabilities regarding Range formats
 respectively.
    Accept-Ranges: npt, smpte, clock
 The syntax is defined in Section 20.2.3.

18.6. Allow

 The Allow message body header field lists the methods supported by
 the resource identified by the Request-URI.  The purpose of this
 field is to inform the recipient of the complete set of valid methods
 associated with the resource.  An Allow header field MUST be present
 in a 405 (Method Not Allowed) response.  The Allow header MUST also
 be present in all OPTIONS responses where the content of the header
 will not include exactly the same methods as listed in the Public
 header.
 The Allow message body header MUST also be included in SETUP and
 DESCRIBE responses, if the methods allowed for the resource are
 different from the complete set of methods defined in this memo.
 Example of use:
    Allow: SETUP, PLAY, SET_PARAMETER, DESCRIBE

18.7. Authentication-Info

 The Authentication-Info response-header is used by the server to
 communicate some information regarding the successful HTTP
 authentication [RFC7235] in the response message.  The definition of
 the header is in [RFC7615], and any applicable HTTP authentication
 schemes appear in other RFCs, such as Digest [RFC7616].  This header
 MUST only be used in response messages related to client to server
 requests.

18.8. Authorization

 An RTSP client that wishes to authenticate itself with a server using
 the authentication mechanism from HTTP [RFC7235], usually (but not
 necessarily) after receiving a 401 response, does so by including an
 Authorization request-header field with the request.  The
 Authorization field-value consists of credentials containing the
 authentication information of the user agent for the realm of the
 resource being requested.  The definition of the header is in

Schulzrinne, et al. Standards Track [Page 138] RFC 7826 RTSP 2.0 December 2016

 [RFC7235], and any applicable HTTP authentication schemes appear in
 other RFCs, such as Digest [RFC7616] and Basic [RFC7617].  This
 header MUST only be used in client-to-server requests.
 If a request is authenticated and a realm specified, the same
 credentials SHOULD be valid for all other requests within this realm
 (assuming that the authentication scheme itself does not require
 otherwise, such as credentials that vary according to a challenge
 value or using synchronized clocks).  Each client-to-server request
 MUST be individually authorized by including the Authorization header
 with the information.
 When a shared cache (see Section 16) receives a request containing an
 Authorization field, it MUST NOT return the corresponding response as
 a reply to any other request, unless one of the following specific
 exceptions holds:
 1.  If the response includes the "max-age" cache directive, the cache
     MAY use that response in replying to a subsequent request.  But
     (if the specified maximum age has passed) a proxy cache MUST
     first revalidate it with the origin server, using the request-
     headers from the new request to allow the origin server to
     authenticate the new request.  (This is the defined behavior for
     max-age.)  If the response includes "max-age=0", the proxy MUST
     always revalidate it before reusing it.
 2.  If the response includes the "must-revalidate" cache-control
     directive, the cache MAY use that response in replying to a
     subsequent request.  But if the response is stale, all caches
     MUST first revalidate it with the origin server, using the
     request-headers from the new request to allow the origin server
     to authenticate the new request.
 3.  If the response includes the "public" cache directive, it MAY be
     returned in reply to any subsequent request.

18.9. Bandwidth

 The Bandwidth request-header field describes the estimated bandwidth
 available to the client, expressed as a positive integer and measured
 in kilobits per second.  The bandwidth available to the client may
 change during an RTSP session, e.g., due to mobility, congestion,
 etc.
 Clients may not be able to accurately determine the available
 bandwidth, for example, because the first hop is not a bottleneck.
 Such a case is when the local area network (LAN) is not the
 bottleneck, instead the LAN's Internet access link is, if the server

Schulzrinne, et al. Standards Track [Page 139] RFC 7826 RTSP 2.0 December 2016

 is not in the same LAN.  Thus, link speeds of WLAN or Ethernet
 networks are normally not a basis for estimating the available
 bandwidth.  Cellular devices or other devices directly connected to a
 modem or connection-enabling device may more accurately estimate the
 bottleneck bandwidth and what is a reasonable share of it for RTSP-
 controlled media.  The client will also need to take into account
 other traffic sharing the bottleneck.  For example, by only assigning
 a certain fraction to RTSP and its media streams.  It is RECOMMENDED
 that only clients that have accurate and explicit information about
 bandwidth bottlenecks use this header.
 This header is not a substitute for proper congestion control.  It is
 only a method providing an initial estimate and coarsely determines
 if the selected content can be delivered at all.
 Example:
   Bandwidth: 62360

18.10. Blocksize

 The Blocksize request-header field is sent from the client to the
 media server asking the server for a particular media packet size.
 This packet size does not include lower-layer headers such as IP,
 UDP, or RTP.  The server is free to use a blocksize that is lower
 than the one requested.  The server MAY truncate this packet size to
 the closest multiple of the minimum, media-specific block size or
 override it with the media-specific size, if necessary.  The block
 size MUST be a positive decimal number measured in octets.  The
 server only returns an error (4xx) if the value is syntactically
 invalid.

18.11. Cache-Control

 The Cache-Control general-header field is used to specify directives
 that MUST be obeyed by all caching mechanisms along the request/
 response chain.
 Cache directives MUST be passed through by a proxy or gateway
 application, regardless of their significance to that application,
 since the directives may be applicable to all recipients along the
 request/response chain.  It is not possible to specify a cache-
 directive for a specific cache.
 Cache-Control should only be specified in a DESCRIBE, GET_PARAMETER,
 SET_PARAMETER, and SETUP request and its response.  Note: Cache-
 Control does not govern only the caching of responses for the RTSP
 messages, instead it also applies to the media stream identified by

Schulzrinne, et al. Standards Track [Page 140] RFC 7826 RTSP 2.0 December 2016

 the SETUP request.  The RTSP requests are generally not cacheable;
 for further information, see Section 16.  Below are the descriptions
 of the cache directives that can be included in the Cache-Control
 header.
 no-cache:  Indicates that the media stream or RTSP response MUST NOT
       be cached anywhere.  This allows an origin server to prevent
       caching even by caches that have been configured to return
       stale responses to client requests.  Note: there is no security
       function preventing the caching of content.
 public:  Indicates that the media stream or RTSP response is
       cacheable by any cache.
 private:  Indicates that the media stream or RTSP response is
       intended for a single user and MUST NOT be cached by a shared
       cache.  A private (non-shared) cache may cache the media
       streams.
 no-transform:  An intermediate cache (proxy) may find it useful to
       convert the media type of a certain stream.  A proxy might, for
       example, convert between video formats to save cache space or
       to reduce the amount of traffic on a slow link.  Serious
       operational problems may occur, however, when these
       transformations have been applied to streams intended for
       certain kinds of applications.  For example, applications for
       medical imaging, scientific data analysis and those using end-
       to-end authentication all depend on receiving a stream that is
       bit-for-bit identical to the original media stream or RTSP
       response.  Therefore, if a response includes the no-transform
       directive, an intermediate cache or proxy MUST NOT change the
       encoding of the stream or response.  Unlike HTTP, RTSP does not
       provide for partial transformation at this point, e.g.,
       allowing translation into a different language.
 only-if-cached:  In some cases, such as times of extremely poor
       network connectivity, a client may want a cache to return only
       those media streams or RTSP responses that it currently has
       stored and not to receive these from the origin server.  To do
       this, the client may include the only-if-cached directive in a
       request.  If the cache receives this directive, it SHOULD
       either respond using a cached media stream or response that is
       consistent with the other constraints of the request or respond
       with a 504 (Gateway Timeout) status.  However, if a group of
       caches is being operated as a unified system with good internal
       connectivity, such a request MAY be forwarded within that group
       of caches.

Schulzrinne, et al. Standards Track [Page 141] RFC 7826 RTSP 2.0 December 2016

 max-stale:  Indicates that the client is willing to accept a media
       stream or RTSP response that has exceeded its expiration time.
       If max-stale is assigned a value, then the client is willing to
       accept a response that has exceeded its expiration time by no
       more than the specified number of seconds.  If no value is
       assigned to max-stale, then the client is willing to accept a
       stale response of any age.
 min-fresh:  Indicates that the client is willing to accept a media
       stream or RTSP response whose freshness lifetime is no less
       than its current age plus the specified time in seconds.  That
       is, the client wants a response that will still be fresh for at
       least the specified number of seconds.
 must-revalidate:  When the must-revalidate directive is present in a
       SETUP response received by a cache, that cache MUST NOT use the
       cache entry after it becomes stale to respond to a subsequent
       request without first revalidating it with the origin server.
       That is, the cache is required to do an end-to-end revalidation
       every time, if, based solely on the origin server's Expires,
       the cached response is stale.
 proxy-revalidate:  The proxy-revalidate directive has the same
       meaning as the must-revalidate directive, except that it does
       not apply to non-shared user agent caches.  It can be used on a
       response to an authenticated request to permit the user's cache
       to store and later return the response without needing to
       revalidate it (since it has already been authenticated once by
       that user), while still requiring proxies that service many
       users to revalidate each time (in order to make sure that each
       user has been authenticated).  Note that such authenticated
       responses also need the "public" cache directive in order to
       allow them to be cached at all.
 max-age:  When an intermediate cache is forced, by means of a max-
       age=0 directive, to revalidate its own cache entry, and the
       client has supplied its own validator in the request, the
       supplied validator might differ from the validator currently
       stored with the cache entry.  In this case, the cache MAY use
       either validator in making its own request without affecting
       semantic transparency.
       However, the choice of validator might affect performance.  The
       best approach is for the intermediate cache to use its own
       validator when making its request.  If the server replies with
       304 (Not Modified), then the cache can return its now validated
       copy to the client with a 200 (OK) response.  If the server
       replies with a new message body and cache validator, however,

Schulzrinne, et al. Standards Track [Page 142] RFC 7826 RTSP 2.0 December 2016

       the intermediate cache can compare the returned validator with
       the one provided in the client's request, using the strong
       comparison function.  If the client's validator is equal to the
       origin server's, then the intermediate cache simply returns 304
       (Not Modified).  Otherwise, it returns the new message body
       with a 200 (OK) response.

18.12. Connection

 The Connection general-header field allows the sender to specify
 options that are desired for that particular connection.  It MUST NOT
 be communicated by proxies over further connections.
 RTSP 2.0 proxies MUST parse the Connection header field before a
 message is forwarded and, for each connection-token in this field,
 remove any header field(s) from the message with the same name as the
 connection-token.  Connection options are signaled by the presence of
 a connection-token in the Connection header field, not by any
 corresponding additional header field(s), since the additional header
 field may not be sent if there are no parameters associated with that
 connection option.
 Message headers listed in the Connection header MUST NOT include end-
 to-end headers, such as Cache-Control.
 RTSP 2.0 defines the "close" connection option for the sender to
 signal that the connection will be closed after completion of the
 response.  For example, "Connection: close in either the request or
 the response-header fields" indicates that the connection SHOULD NOT
 be considered "persistent" (Section 10.2) after the current request/
 response is complete.
 The use of the connection option "close" in RTSP messages SHOULD be
 limited to error messages when the server is unable to recover and
 therefore sees it necessary to close the connection.  The reason
 being that the client has the choice of continuing using a connection
 indefinitely, as long as it sends valid messages.

18.13. Connection-Credentials

 The Connection-Credentials response-header is used to carry the chain
 of credentials for any next hop that needs to be approved by the
 requester.  It MUST only be used in server-to-client responses.
 The Connection-Credentials header in an RTSP response MUST, if
 included, contain the credential information (in the form of a list
 of certificates providing the chain of certification) of the next hop
 to which an intermediary needs to securely connect.  The header MUST

Schulzrinne, et al. Standards Track [Page 143] RFC 7826 RTSP 2.0 December 2016

 include the URI of the next hop (proxy or server) and a
 Base64-encoded (according to Section 4 of [RFC4648] and where the
 padding bits are set to zero) binary structure containing a sequence
 of DER-encoded X.509v3 certificates [RFC5280].
 The binary structure starts with the number of certificates
 (NR_CERTS) included as a 16-bit unsigned integer.  This is followed
 by an NR_CERTS number of 16-bit unsigned integers providing the size,
 in octets, of each DER-encoded certificate.  This is followed by an
 NR_CERTS number of DER-encoded X.509v3 certificates in a sequence
 (chain).  This format is exemplified in Figure 6.  The certificate of
 the proxy or server must come first in the structure.  Each following
 certificate must directly certify the one preceding it.  Because
 certificate validation requires that root keys be distributed
 independently, the self-signed certificate that specifies the root
 certificate authority may optionally be omitted from the chain, under
 the assumption that the remote end must already possess it in order
 to validate it in any case.
 Example:
 Connection-Credentials:"rtsps://proxy2.example.com/";MIIDNTCC...
 Where MIIDNTCC... is a Base64 encoding of the following structure:
      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |  Number of certificates       | Size of certificate #1        |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     | Size of certificate #2        | Size of certificate #3        |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     : DER Encoding of Certificate #1                                :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     : DER Encoding of Certificate #2                                :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     : DER Encoding of Certificate #3                                :
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 Figure 6: Format Example of Connection-Credentials Header Certificate

18.14. Content-Base

 The Content-Base message body header field may be used to specify the
 base URI for resolving relative URIs within the message body.
 Content-Base: rtsp://media.example.com/movie/twister/

Schulzrinne, et al. Standards Track [Page 144] RFC 7826 RTSP 2.0 December 2016

 If no Content-Base field is present, the base URI of a message body
 is defined by either its Content-Location (if that Content-Location
 URI is an absolute URI) or the URI used to initiate the request, in
 that order of precedence.  Note, however, that the base URI of the
 contents within the message body may be redefined within that message
 body.

18.15. Content-Encoding

 The Content-Encoding message body header field is used as a modifier
 of the media-type.  When present, its value indicates what additional
 content-codings have been applied to the message body, and thus what
 decoding mechanisms must be applied in order to obtain the media-type
 referenced by the Content-Type header field.  Content-Encoding is
 primarily used to allow a document to be compressed without losing
 the identity of its underlying media type.
 The content-coding is a characteristic of the message body identified
 by the Request-URI.  Typically, the message body is stored with this
 encoding and is only decoded before rendering or analogous usage.
 However, an RTSP proxy MAY modify the content-coding if the new
 coding is known to be acceptable to the recipient, unless the "no-
 transform" cache directive is present in the message.
 If the content-coding of a message body is not "identity", then the
 message MUST include a Content-Encoding message body header that
 lists the non-identity content-coding(s) used.
 If the content-coding of a message body in a request message is not
 acceptable to the origin server, the server SHOULD respond with a
 status code of 415 (Unsupported Media Type).
 If multiple encodings have been applied to a message body, the
 content-codings MUST be listed in the order in which they were
 applied, first to last from left to right.  Additional information
 about the encoding parameters MAY be provided by other header fields
 not defined by this specification.

18.16. Content-Language

 The Content-Language message body header field describes the natural
 language(s) of the intended audience for the enclosed message body.
 Note that this might not be equivalent to all the languages used
 within the message body.

Schulzrinne, et al. Standards Track [Page 145] RFC 7826 RTSP 2.0 December 2016

 Language tags are mentioned in Section 18.4.  The primary purpose of
 Content-Language is to allow a user to identify and differentiate
 entities according to the user's own preferred language.  Thus, if
 the body content is intended only for a Danish-literate audience, the
 appropriate field is
    Content-Language: da
 If no Content-Language is specified, the default is that the content
 is intended for all language audiences.  This might mean that the
 sender does not consider it to be specific to any natural language or
 that the sender does not know for which language it is intended.
 Multiple languages MAY be listed for content that is intended for
 multiple audiences.  For example, a rendition of the "Treaty of
 Waitangi", presented simultaneously in the original Maori and English
 versions, would call for
    Content-Language: mi, en
 However, just because multiple languages are present within a message
 body does not mean that it is intended for multiple linguistic
 audiences.  An example would be a beginner's language primer, such as
 "A First Lesson in Latin", which is clearly intended to be used by an
 English-literate audience.  In this case, the Content-Language would
 properly only include "en".
 Content-Language MAY be applied to any media type -- it is not
 limited to textual documents.

18.17. Content-Length

 The Content-Length message body header field contains the length of
 the message body of the RTSP message (i.e., after the double CRLF
 following the last header) in octets of bits.  Unlike HTTP, it MUST
 be included in all messages that carry a message body beyond the
 header portion of the RTSP message.  If it is missing, a default
 value of zero is assumed.  Any Content-Length greater than or equal
 to zero is a valid value.

18.18. Content-Location

 The Content-Location message body header field MAY be used to supply
 the resource location for the message body enclosed in the message
 when that body is accessible from a location separate from the
 requested resource's URI.  A server SHOULD provide a Content-Location
 for the variant corresponding to the response message body;
 especially in the case where a resource has multiple variants

Schulzrinne, et al. Standards Track [Page 146] RFC 7826 RTSP 2.0 December 2016

 associated with it, and those entities actually have separate
 locations by which they might be individually accessed, the server
 SHOULD provide a Content-Location for the particular variant that is
 returned.
 As an example, if an RTSP client performs a DESCRIBE request on a
 given resource, e.g., "rtsp://a.example.com/movie/
 Plan9FromOuterSpace", then the server may use additional information,
 such as the User-Agent header, to determine the capabilities of the
 agent.  The server will then return a media description tailored to
 that class of RTSP agents.  To indicate which specific description
 the agent receives, the resource identifier
 ("rtsp://a.example.com/movie/Plan9FromOuterSpace/FullHD.sdp") is
 provided in Content-Location, while the description is still a valid
 response for the generic resource identifier, thus enabling both
 debugging and cache operation as discussed below.
 The Content-Location value is not a replacement for the original
 requested URI; it is only a statement of the location of the resource
 corresponding to this particular variant at the time of the request.
 Future requests MAY specify the Content-Location URI as the Request-
 URI if the desire is to identify the source of that particular
 variant.  This is useful if the RTSP agent desires to verify if the
 resource variant is current through a conditional request.
 A cache cannot assume that a message body with a Content-Location
 different from the URI used to retrieve it can be used to respond to
 later requests on that Content-Location URI.  However, the Content-
 Location can be used to differentiate between multiple variants
 retrieved from a single requested resource.
 If the Content-Location is a relative URI, the relative URI is
 interpreted relative to the Request-URI.
 Note that Content-Location can be used in some cases to derive the
 base-URI for relative URI(s) present in session description formats.
 This needs to be taken into account when Content-Location is used.
 The easiest way to avoid needing to consider that issue is to include
 the Content-Base whenever the Content-Location is included.
 Note also, when using Media Tags in conjunction with Content-
 Location, it is important that the different versions have different
 MTags, even if provided under different Content-Location URIs.  This
 is because the different content variants still have been provided in
 response to the same request URI.

Schulzrinne, et al. Standards Track [Page 147] RFC 7826 RTSP 2.0 December 2016

 Note also, as in most cases, the URIs used in the DESCRIBE and the
 SETUP requests are different: the URI provided in a DESCRIBE Content-
 Location response can't directly be used in a SETUP request.
 Instead, the steps of deriving the media resource URIs are necessary.
 This commonly involves combing the media description's relative URIs,
 e.g., from the SDP's a=control attribute, with the base-URI to create
 the absolute URIs needed in the SETUP request.

18.19. Content-Type

 The Content-Type message body header indicates the media type of the
 message body sent to the recipient.  Note that the content types
 suitable for RTSP are likely to be restricted in practice to
 presentation descriptions and parameter-value types.

18.20. CSeq

 The CSeq general-header field specifies the sequence number (integer)
 for an RTSP request/response pair.  This field MUST be present in all
 requests and responses.  RTSP agents maintain a sequence number
 series for each responder to which they have an open message
 transport channel.  For each new RTSP request an agent originates on
 a particular RTSP message transport, the CSeq value MUST be
 incremented by one.  The initial sequence number can be any number;
 however, it is RECOMMENDED to start at 0.  Each sequence number
 series is unique between each requester and responder, i.e., the
 client has one series for its requests to a server and the server has
 another when sending requests to the client.  Each requester and
 responder is identified by its socket address (IP address and port
 number), i.e., per direction of a TCP connection.  Any retransmitted
 request MUST contain the same sequence number as the original, i.e.,
 the sequence number is not incremented for retransmissions of the
 same request.  The RTSP agent receiving requests MUST process the
 requests arriving on a particular transport in the order of the
 sequence numbers.  Responses are sent in the order that they are
 generated.  The RTSP response MUST have the same sequence number as
 was present in the corresponding request.  An RTSP agent receiving a
 response MAY receive the responses out of order compared to the order
 of the requests it sent.  Thus, the agent MUST use the sequence
 number in the response to pair it with the corresponding request.
    The main purpose of the sequence number is to map responses to
    requests.
    The requirement to use a sequence-number increment of one for each
    new request is to support any future specification of RTSP message
    transport over a protocol that does not provide in-order delivery
    or is unreliable.

Schulzrinne, et al. Standards Track [Page 148] RFC 7826 RTSP 2.0 December 2016

    The above rules relating to the initial sequence number may appear
    unnecessarily loose.  The reason for this is to cater to some
    common behavior of existing implementations: when using multiple
    reliable connections in sequence, it may still be easiest to use a
    single sequence-number series for a client connecting with a
    particular server.  Thus, the initial sequence number may be
    arbitrary depending on the number of previous requests.  For any
    unreliable transport, a stricter definition or other solution will
    be required to enable detection of any loss of the first request.
    When using multiple sequential transport connections, there is no
    protocol mechanism to ensure in-order processing as the sequence
    number is scoped on the individual transport connection and its
    five tuple.  Thus, there are potential issues with opening a new
    transport connection to the same host for which there already
    exists a transport connection with outstanding requests and
    previously dispatched requests related to the same RTSP session.
 RTSP Proxies also need to follow the above rules.  This implies that
 proxies that aggregate requests from multiple clients onto a single
 transport towards a server or a next-hop proxy need to renumber these
 requests to form a unified sequence on that transport, fulfilling the
 above rules.  A proxy capable of fulfilling some agent's request
 without emitting its own request (e.g., a caching proxy that fulfills
 a request from its cache) also causes a need to renumber as the
 number of received requests with a particular target may not be the
 same as the number of emitted requests towards that target agent.  A
 proxy that needs to renumber needs to perform the corresponding
 renumbering back to the original sequence number for any received
 response before forwarding it back to the originator of the request.
    A client connected to a proxy, and using that transport to send
    requests to multiple servers, creates a situation where it is
    quite likely to receive the responses out of order.  This is
    because the proxy will establish separate transports from the
    proxy to the servers on which to forward the client's requests.
    When the responses arrive from the different servers, they will be
    forwarded to the client in the order they arrive at the proxy and
    can be processed, not the order of the client's original sequence
    numbers.  This is intentional to avoid some session's requests
    being blocked by another server's slow processing of requests.

Schulzrinne, et al. Standards Track [Page 149] RFC 7826 RTSP 2.0 December 2016

18.21. Date

 The Date general-header field represents the date and time at which
 the message was originated.  The inclusion of the Date header in an
 RTSP message follows these rules:
 o  An RTSP message, sent by either the client or the server,
    containing a body MUST include a Date header, if the sending host
    has a clock;
 o  Clients and servers are RECOMMENDED to include a Date header in
    all other RTSP messages, if the sending host has a clock;
 o  If the server does not have a clock that can provide a reasonable
    approximation of the current time, its responses MUST NOT include
    a Date header field.  In this case, this rule MUST be followed:
    some origin-server implementations might not have a clock
    available.  An origin server without a clock MUST NOT assign
    Expires or Last-Modified values to a response, unless these values
    were associated with the resource by a system or user with a
    reliable clock.  It MAY assign an Expires value that is known, at
    or before server-configuration time, to be in the past (this
    allows "pre-expiration" of responses without storing separate
    Expires values for each resource).
 A received message that does not have a Date header field MUST be
 assigned one by the recipient if the message will be cached by that
 recipient.  An RTSP implementation without a clock MUST NOT cache
 responses without revalidating them on every use.  An RTSP cache,
 especially a shared cache, SHOULD use a mechanism, such as the
 Network Time Protocol (NTP) [RFC5905], to synchronize its clock with
 a reliable external standard.
 The RTSP-date, a full date as specified by Section 3.3 of [RFC5322],
 sent in a Date header SHOULD NOT represent a date and time subsequent
 to the generation of the message.  It SHOULD represent the best
 available approximation of the date and time of message generation,
 unless the implementation has no means of generating a reasonably
 accurate date and time.  In theory, the date ought to represent the
 moment just before the message body is generated.  In practice, the
 date can be generated at any time during the message origination
 without affecting its semantic value.
    Note: The RTSP 2.0 date format is defined to be the full-date
    format in RFC 5322.  This format is more flexible than the date
    format in RFC 1123 used by RTSP 1.0.  Thus, implementations should
    use single spaces as separators, as recommended by RFC 5322, and
    support receiving the obsolete format.

Schulzrinne, et al. Standards Track [Page 150] RFC 7826 RTSP 2.0 December 2016

    Further, note that the syntax allows for a comment to be added at
    the end of the date.

18.22. Expires

 The Expires message body header field gives a date and time after
 which the description or media-stream should be considered stale.
 The interpretation depends on the method:
 DESCRIBE response:  The Expires header indicates a date and time
       after which the presentation description (body) SHOULD be
       considered stale.
 SETUP response:  The Expires header indicates a date and time after
       which the media stream SHOULD be considered stale.
 A stale cache entry should not be returned by a cache (either a proxy
 cache or a user agent cache) unless it is first validated with the
 origin server (or with an intermediate cache that has a fresh copy of
 the message body).  See Section 16 for further discussion of the
 expiration model.
 The presence of an Expires field does not imply that the original
 resource will change or cease to exist at, before, or after that
 time.
 The format is an absolute date and time as defined by RTSP-date.  An
 example of its use is
   Expires: Wed, 23 Jan 2013 15:36:52 +0000
 RTSP 2.0 clients and caches MUST treat other invalid date formats,
 especially those including the value "0", as having occurred in the
 past (i.e., already expired).
 To mark a response as "already expired," an origin server should use
 an Expires date that is equal to the Date header value.  To mark a
 response as "never expires", an origin server SHOULD use an Expires
 date approximately one year from the time the response is sent.  RTSP
 2.0 servers SHOULD NOT send Expires dates that are more than one year
 in the future.

18.23. From

 The From request-header field, if given, SHOULD contain an Internet
 email address for the human user who controls the requesting user
 agent.  The address SHOULD be machine usable, as defined by "mailbox"
 in [RFC1123].

Schulzrinne, et al. Standards Track [Page 151] RFC 7826 RTSP 2.0 December 2016

 This header field MAY be used for logging purposes and as a means for
 identifying the source of invalid or unwanted requests.  It SHOULD
 NOT be used as an insecure form of access protection.  The
 interpretation of this field is that the request is being performed
 on behalf of the person given, who accepts responsibility for the
 method performed.  In particular, robot agents SHOULD include this
 header so that the person responsible for running the robot can be
 contacted if problems occur on the receiving end.
 The Internet email address in this field MAY be separate from the
 Internet host that issued the request.  For example, when a request
 is passed through a proxy, the original issuer's address SHOULD be
 used.
 The client SHOULD NOT send the From header field without the user's
 approval, as it might conflict with the user's privacy interests or
 their site's security policy.  It is strongly recommended that the
 user be able to disable, enable, and modify the value of this field
 at any time prior to a request.

18.24. If-Match

 The If-Match request-header field is especially useful for ensuring
 the integrity of the presentation description, independent of how the
 presentation description was received.  The presentation description
 can be fetched via means external to RTSP (such as HTTP) or via the
 DESCRIBE message.  In the case of retrieving the presentation
 description via RTSP, the server implementation is guaranteeing the
 integrity of the description between the time of the DESCRIBE message
 and the SETUP message.  By including the MTag given in or with the
 session description in an If-Match header part of the SETUP request,
 the client ensures that resources set up are matching the
 description.  A SETUP request with the If-Match header for which the
 MTag validation check fails MUST generate a response using 412
 (Precondition Failed).
 This validation check is also very useful if a session has been
 redirected from one server to another.

18.25. If-Modified-Since

 The If-Modified-Since request-header field is used with the DESCRIBE
 and SETUP methods to make them conditional.  If the requested variant
 has not been modified since the time specified in this field, a
 description will not be returned from the server (DESCRIBE) or a
 stream will not be set up (SETUP).  Instead, a 304 (Not Modified)
 response MUST be returned without any message body.

Schulzrinne, et al. Standards Track [Page 152] RFC 7826 RTSP 2.0 December 2016

 An example of the field is:
   If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT

18.26. If-None-Match

 This request-header can be used with one or several message body tags
 to make DESCRIBE requests conditional.  A client that has one or more
 message bodies previously obtained from the resource can verify that
 none of those entities is current by including a list of their
 associated message body tags in the If-None-Match header field.  The
 purpose of this feature is to allow efficient updates of cached
 information with a minimum amount of transaction overhead.  As a
 special case, the value "*" matches any current entity of the
 resource.
 If any of the message body tags match the message body tag of the
 message body that would have been returned in the response to a
 similar DESCRIBE request (without the If-None-Match header) on that
 resource, or if "*" is given and any current entity exists for that
 resource, then the server MUST NOT perform the requested method,
 unless required to do so because the resource's modification date
 fails to match that supplied in an If-Modified-Since header field in
 the request.  Instead, if the request method was DESCRIBE, the server
 SHOULD respond with a 304 (Not Modified) response, including the
 cache-related header fields (particularly MTag) of one of the message
 bodies that matched.  For all other request methods, the server MUST
 respond with a status of 412 (Precondition Failed).
 See Section 16.1.3 for rules on how to determine if two message body
 tags match.
 If none of the message body tags match, then the server MAY perform
 the requested method as if the If-None-Match header field did not
 exist, but MUST also ignore any If-Modified-Since header field(s) in
 the request.  That is, if no message body tags match, then the server
 MUST NOT return a 304 (Not Modified) response.
 If the request would, without the If-None-Match header field, result
 in anything other than a 2xx or 304 status, then the If-None-Match
 header MUST be ignored.  (See Section 16.1.4 for a discussion of
 server behavior when both If-Modified-Since and If-None-Match appear
 in the same request.)
 The result of a request having both an If-None-Match header field and
 an If-Match header field is unspecified and MUST be considered an
 illegal request.

Schulzrinne, et al. Standards Track [Page 153] RFC 7826 RTSP 2.0 December 2016

18.27. Last-Modified

 The Last-Modified message body header field indicates the date and
 time at which the origin server believes the presentation description
 or media stream was last modified.  For the DESCRIBE method, the
 header field indicates the last modification date and time of the
 description, for the SETUP of the media stream.
 An origin server MUST NOT send a Last-Modified date that is later
 than the server's time of message origination.  In such cases, where
 the resource's last modification would indicate some time in the
 future, the server MUST replace that date with the message
 origination date.
 An origin server SHOULD obtain the Last-Modified value of the message
 body as close as possible to the time that it generates the Date
 value of its response.  This allows a recipient to make an accurate
 assessment of the message body's modification time, especially if the
 message body changes near the time that the response is generated.
 RTSP servers SHOULD send Last-Modified whenever feasible.

18.28. Location

 The Location response-header field is used to redirect the recipient
 to a location other than the Request-URI for completion of the
 request or identification of a new resource.  For 3rr responses, the
 location SHOULD indicate the server's preferred URI for automatic
 redirection to the resource.  The field-value consists of a single
 absolute URI.
 Note: The Content-Location header field (Section 18.18) differs from
 Location in that the Content-Location identifies the original
 location of the message body enclosed in the request.  Therefore, it
 is possible for a response to contain header fields for both Location
 and Content-Location.  Also, see Section 16.2 for cache requirements
 of some methods.

18.29. Media-Properties

 This general-header is used in SETUP responses or PLAY_NOTIFY
 requests to indicate the media's properties that currently are
 applicable to the RTSP session.  PLAY_NOTIFY MAY be used to modify
 these properties at any point.  However, the client SHOULD have
 received the update prior to any action related to the new media
 properties taking effect.  For aggregated sessions, the Media-
 Properties header will be returned in each SETUP response.  The
 header received in the latest response is the one that applies on the

Schulzrinne, et al. Standards Track [Page 154] RFC 7826 RTSP 2.0 December 2016

 whole session from this point until any future update.  The header
 MAY be included without value in GET_PARAMETER requests to the server
 with a Session header included to query the current Media-Properties
 for the session.  The responder MUST include the current session's
 media properties.
 The media properties expressed by this header are the ones applicable
 to all media in the RTSP session.  For aggregated sessions, the
 header expressed the combined media-properties.  As a result,
 aggregation of media MAY result in a change of the media properties
 and, thus, the content of the Media-Properties header contained in
 subsequent SETUP responses.
 The header contains a list of property values that are applicable to
 the currently setup media or aggregate of media as indicated by the
 RTSP URI in the request.  No ordering is enforced within the header.
 Property values should be placed into a single group that handles a
 particular orthogonal property.  Values or groups that express
 multiple properties SHOULD NOT be used.  The list of properties that
 can be expressed MAY be extended at any time.  Unknown property
 values MUST be ignored.
 This specification defines the following four groups and their
 property values:
 Random Access:
    Random-Access:  Indicates that random access is possible.  May
       optionally include a floating-point value in seconds indicating
       the longest duration between any two random access points in
       the media.
    Beginning-Only:  Seeking is limited to the beginning only.
    No-Seeking:  No seeking is possible.
 Content Modifications:
    Immutable:  The content will not be changed during the lifetime of
       the RTSP session.
    Dynamic:  The content may be changed based on external methods or
       triggers.
    Time-Progressing:  The media accessible progresses as wallclock
       time progresses.

Schulzrinne, et al. Standards Track [Page 155] RFC 7826 RTSP 2.0 December 2016

 Retention:
    Unlimited:  Content will be retained for the duration of the
       lifetime of the RTSP session.
    Time-Limited:  Content will be retained at least until the
       specified wallclock time.  The time must be provided in the
       absolute time format specified in Section 4.4.3.
    Time-Duration:  Each individual media unit is retained for at
       least the specified Time-Duration.  This definition allows for
       retaining data with a time-based sliding window.  The time
       duration is expressed as floating-point number in seconds.  The
       value 0.0 is a valid as this indicates that no data is retained
       in a time-progressing session.
 Supported Scale:
    Scales:  A quoted comma-separated list of one or more decimal
       values or ranges of scale values supported by the content in
       arbitrary order.  A range has a start and stop value separated
       by a colon.  A range indicates that the content supports a
       fine-grained selection of scale values.  Fine-graining allows
       for steps at least as small as one tenth of a scale value.
       Content is considered to support fine-grained selection when
       the server in response to a given scale value can produce
       content with an actual scale that is less than one tenth of
       scale unit, i.e., 0.1, from the requested value.  Negative
       values are supported.  The value 0 has no meaning and MUST NOT
       be used.
 Examples of this header for on-demand content and a live stream
 without recording are:
 On-demand:
 Media-Properties: Random-Access=2.5, Unlimited, Immutable,
      Scales="-20, -10, -4, 0.5:1.5, 4, 8, 10, 15, 20"
 Live stream without recording/timeshifting:
 Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0.0

18.30. Media-Range

 The Media-Range general-header is used to give the range of the media
 at the time of sending the RTSP message.  This header MUST be
 included in the SETUP response, PLAY and PAUSE responses for media
 that are time-progressing, PLAY and PAUSE responses after any change
 for media that are Dynamic, and in PLAY_NOTIFY requests that are sent

Schulzrinne, et al. Standards Track [Page 156] RFC 7826 RTSP 2.0 December 2016

 due to Media-Property-Update.  A Media-Range header without any range
 specifications MAY be included in GET_PARAMETER requests to the
 server to request the current range.  In this case, the server MUST
 include the current range at the time of sending the response.
 The header MUST include range specifications for all time formats
 supported for the media, as indicated in Accept-Ranges header
 (Section 18.5) when setting up the media.  The server MAY include
 more than one range specification of any given time format to
 indicate media that has non-continuous range.  The range
 specifications SHALL be ordered with the range with the lowest value
 or earliest start time first, followed by ranges with increasingly
 higher values or later start time.
 For media that has the time-progressing property, the Media-Range
 header values will only be valid for the particular point in time
 when it was issued.  As the wallclock progresses, so will the media
 range.  However, it shall be assumed that media time progresses in
 direct relationship to wallclock time (with the exception of clock
 skew) so that a reasonably accurate estimation of the media range can
 be calculated.

18.31. MTag

 The MTag response-header MAY be included in DESCRIBE, GET_PARAMETER,
 or SETUP responses.  The message body tags (Section 4.6) returned in
 a DESCRIBE response and the one in SETUP refer to the presentation,
 i.e., both the returned session description and the media stream.
 This allows for verification that one has the right session
 description to a media resource at the time of the SETUP request.
 However, it has the disadvantage that a change in any of the parts
 results in invalidation of all the parts.
 If the MTag is provided both inside the message body, e.g., within
 the "a=mtag" attribute in SDP, and in the response message, then both
 tags MUST be identical.  It is RECOMMENDED that the MTag be primarily
 given in the RTSP response message, to ensure that caches can use the
 MTag without requiring content inspection.  However, for session
 descriptions that are distributed outside of RTSP, for example, using
 HTTP, etc., it will be necessary to include the message body tag in
 the session description as specified in Appendix D.1.9.
 SETUP and DESCRIBE requests can be made conditional upon the MTag
 using the headers If-Match (Section 18.24) and If-None-Match
 (Section 18.26).

Schulzrinne, et al. Standards Track [Page 157] RFC 7826 RTSP 2.0 December 2016

18.32. Notify-Reason

 The Notify-Reason response-header is solely used in the PLAY_NOTIFY
 method.  It indicates the reason why the server has sent the
 asynchronous PLAY_NOTIFY request (see Section 13.5).

18.33. Pipelined-Requests

 The Pipelined-Requests general-header is used to indicate that a
 request is to be executed in the context created by a previous
 request(s).  The primary usage of this header is to allow pipelining
 of SETUP requests so that any additional SETUP request after the
 first one does not need to wait for the session ID to be sent back to
 the requesting agent.  The header contains a unique identifier that
 is scoped by the persistent connection used to send the requests.
 Upon receiving a request with the Pipelined-Requests, the responding
 agent MUST look up if there exists a binding between this Pipelined-
 Requests identifier for the current persistent connection and an RTSP
 session ID.  If the binding exists, then the received request is
 processed the same way as if it contained the Session header with the
 found session ID.  If there does not exist a mapping and no Session
 header is included in the request, the responding agent MUST create a
 binding upon the successful completion of a session creating request,
 i.e., SETUP.  A binding MUST NOT be created, if the request failed to
 create an RTSP session.  In case the request contains both a Session
 header and the Pipelined-Requests header, the Pipelined-Requests
 header MUST be ignored.
 Note: Based on the above definition, at least the first request
 containing a new unique Pipelined-Requests header will be required to
 be a SETUP request (unless the protocol is extended with new methods
 of creating a session).  After that first one, additional SETUP
 requests or requests of any type using the RTSP session context may
 include the Pipelined-Requests header.
 When responding to any request that contained the Pipelined-Requests
 header, the server MUST also include the Session header when a
 binding to a session context exists.  An RTSP agent that knows the
 session identifier SHOULD NOT use the Pipelined-Requests header in
 any request and only use the Session header.  This as the Session
 identifier is persistent across transport contexts, like TCP
 connections, which the Pipelined-Requests identifier is not.
 The RTSP agent sending the request with a Pipelined-Requests header
 has the responsibility for using a unique and previously unused
 identifier within the transport context.  Currently, only a TCP
 connection is defined as such a transport context.  A server MUST

Schulzrinne, et al. Standards Track [Page 158] RFC 7826 RTSP 2.0 December 2016

 delete the Pipelined-Requests identifier and its binding to a session
 upon the termination of that session.  Despite the previous mandate,
 RTSP agents are RECOMMENDED not to reuse identifiers to allow for
 better error handling and logging.
 RTSP Proxies may need to translate Pipelined-Requests identifier
 values from incoming requests to outgoing to allow for aggregation of
 requests onto a persistent connection.

18.34. Proxy-Authenticate

 The Proxy-Authenticate response-header field MUST be included as part
 of a 407 (Proxy Authentication Required) response.  The field-value
 consists of a challenge that indicates the authentication scheme and
 parameters applicable to the proxy for this Request-URI.  The
 definition of the header is in [RFC7235], and any applicable HTTP
 authentication schemes appear in other RFCs, such as Digest [RFC7616]
 and Basic [RFC7617].
 The HTTP access authentication process is described in [RFC7235].
 This header MUST only be used in response messages related to client-
 to-server requests.

18.35. Proxy-Authentication-Info

 The Proxy-Authentication-Info response-header is used by the proxy to
 communicate some information regarding the successful authentication
 to the proxy in the message response in some authentication schemes,
 such as the Digest scheme [RFC7616].  The definition of the header is
 in [RFC7615], and any applicable HTTP authentication schemes appear
 in other RFCs.  This header MUST only be used in response messages
 related to client-to-server requests.  This header has hop-by-hop
 scope.

18.36. Proxy-Authorization

 The Proxy-Authorization request-header field allows the client to
 identify itself (or its user) to a proxy that requires
 authentication.  The Proxy-Authorization field-value consists of
 credentials containing the authentication information of the user
 agent for the proxy or realm of the resource being requested.  The
 definition of the header is in [RFC7235], and any applicable HTTP
 authentication schemes appear in other RFCs, such as Digest [RFC7616]
 and Basic [RFC7617].

Schulzrinne, et al. Standards Track [Page 159] RFC 7826 RTSP 2.0 December 2016

 The HTTP access authentication process is described in [RFC7235].
 Unlike Authorization, the Proxy-Authorization header field applies
 only to the next-hop proxy.  This header MUST only be used in client-
 to-server requests.

18.37. Proxy-Require

 The Proxy-Require request-header field is used to indicate proxy-
 sensitive features that MUST be supported by the proxy.  Any Proxy-
 Require header features that are not supported by the proxy MUST be
 negatively acknowledged by the proxy to the client using the
 Unsupported header.  The proxy MUST use the 551 (Option Not
 Supported) status code in the response.  Any feature tag included in
 the Proxy-Require does not apply to the endpoint (server or client).
 To ensure that a feature is supported by both proxies and servers,
 the tag needs to be included in also a Require header.
 See Section 18.43 for more details on the mechanics of this message
 and a usage example.  See discussion in the proxies section
 (Section 15.1) about when to consider that a feature requires proxy
 support.
 Example of use:
    Proxy-Require: play.basic

18.38. Proxy-Supported

 The Proxy-Supported general-header field enumerates all the
 extensions supported by the proxy using feature tags.  The header
 carries the intersection of extensions supported by the forwarding
 proxies.  The Proxy-Supported header MAY be included in any request
 by a proxy.  It MUST be added by any proxy if the Supported header is
 present in a request.  When present in a request, the receiver MUST
 copy the received Proxy-Supported header in the response.
 The Proxy-Supported header field contains a list of feature tags
 applicable to proxies, as described in Section 4.5.  The list is the
 intersection of all feature tags understood by the proxies.  To
 achieve an intersection, the proxy adding the Proxy-Supported header
 includes all proxy feature tags it understands.  Any proxy receiving
 a request with the header MUST check the list and remove any feature
 tag(s) it does not support.  A Proxy-Supported header present in the
 response MUST NOT be modified by the proxies.  These feature tags are
 the ones the proxy chains support in general and are not specific to
 the request resource.

Schulzrinne, et al. Standards Track [Page 160] RFC 7826 RTSP 2.0 December 2016

 Example:
   C->P1: OPTIONS rtsp://example.com/ RTSP/2.0
          Supported: foo, bar, blech
          User-Agent: PhonyClient/1.2
  P1->P2: OPTIONS rtsp://example.com/ RTSP/2.0
          Supported: foo, bar, blech
          Proxy-Supported: proxy-foo, proxy-bar, proxy-blech
          Via: 2.0 pro.example.com
  P2->S:  OPTIONS rtsp://example.com/ RTSP/2.0
          Supported: foo, bar, blech
          Proxy-Supported: proxy-foo, proxy-blech
          Via: 2.0 pro.example.com, 2.0 prox2.example.com
   S->C:  RTSP/2.0 200 OK
          Supported: foo, bar, baz
          Proxy-Supported: proxy-foo, proxy-blech
          Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
          Via: 2.0 pro.example.com, 2.0 prox2.example.com

18.39. Public

 The Public response-header field lists the set of methods supported
 by the response sender.  This header applies to the general
 capabilities of the sender, and its only purpose is to indicate the
 sender's capabilities to the recipient.  The methods listed may or
 may not be applicable to the Request-URI; the Allow header field
 (Section 18.6) MAY be used to indicate methods allowed for a
 particular URI.
 Example of use:
    Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
 In the event that there are proxies between the sender and the
 recipient of a response, each intervening proxy MUST modify the
 Public header field to remove any methods that are not supported via
 that proxy.  The resulting Public header field will contain an
 intersection of the sender's methods and the methods allowed through
 by the intervening proxies.
    In general, proxies should allow all methods to transparently pass
    through from the sending RTSP agent to the receiving RTSP agent,
    but there may be cases where this is not desirable for a given
    proxy.  Modification of the Public response-header field by the

Schulzrinne, et al. Standards Track [Page 161] RFC 7826 RTSP 2.0 December 2016

    intervening proxies ensures that the request sender gets an
    accurate response indicating the methods that can be used on the
    target agent via the proxy chain.

18.40. Range

 The Range general-header specifies a time range in PLAY
 (Section 13.4), PAUSE (Section 13.6), SETUP (Section 13.3), and
 PLAY_NOTIFY (Section 13.5) requests and responses.  It MAY be
 included in GET_PARAMETER requests from the client to the server with
 only a Range format and no value to request the current media
 position, whether the session is in Play or Ready state in the
 included format.  The server SHALL, if supporting the range format,
 respond with the current playing point or pause point as the start of
 the range.  If an explicit stop point was used in the previous PLAY
 request, then that value shall be included as stop point.  Note that
 if the server is currently under any type of media playback
 manipulation affecting the interpretation of the Range header, like
 scale value other than 1, that fact is also required to be included
 in any GET_PARAMETER response by including the Scale header to
 provide complete information.
 The range can be specified in a number of units.  This specification
 defines smpte (Section 4.4.1), npt (Section 4.4.2), and clock
 (Section 4.4.3) range units.  While octet ranges (Byte Ranges) (see
 Section 2.1 of [RFC7233]) and other extended units MAY be used, their
 behavior is unspecified since they are not normally meaningful in
 RTSP.  Servers supporting the Range header MUST understand the NPT
 range format and SHOULD understand the SMPTE range format.  If the
 Range header is sent in a time format that is not understood, the
 recipient SHOULD return 456 (Header Field Not Valid for Resource) and
 include an Accept-Ranges header indicating the supported time formats
 for the given resource.
 Example:
   Range: clock=19960213T143205Z-
 The Range header contains a range of one single range format.  A
 range is a half-open interval with a start and an end point,
 including the start point but excluding the end point.  A range may
 either be fully specified with explicit values for start point and
 end point or have either the start or end point be implicit.  An
 implicit start point indicates the session's pause point, and if no
 pause point is set, the start of the content.  An implicit end point
 indicates the end of the content.  The usage of both implicit start

Schulzrinne, et al. Standards Track [Page 162] RFC 7826 RTSP 2.0 December 2016

 and end points is not allowed in the same Range header; however, the
 omission of the Range header has that meaning, i.e., from pause point
 (or start) until end of content.
    As noted, Range headers define half-open intervals.  A range of
    A-B starts exactly at time A, but ends just before B.  Only the
    start time of a media unit such as a video or audio frame is
    relevant.  For example, assume that video frames are generated
    every 40 ms.  A range of 10.0-10.1 would include a video frame
    starting at 10.0 or later time and would include a video frame
    starting at 10.08, even though it lasted beyond the interval.  A
    range of 10.0-10.08, on the other hand, would exclude the frame at
    10.08.
    Please note the difference between NPT timescales' "now" and an
    implicit start value.  Implicit values reference the current
    pause-point, while "now" is the current time.  In a time-
    progressing session with recording (retention for some or full
    time), the pause point may be 2 min into the session while now
    could be 1 hour into the session.
 By default, range intervals increase, where the second point is
 larger than the first point.
 Example:
     Range: npt=10-15
 However, range intervals can also decrease if the Scale header (see
 Section 18.46) indicates a negative scale value.  For example, this
 would be the case when a playback in reverse is desired.
 Example:
     Scale: -1
     Range: npt=15-10
 Decreasing ranges are still half-open intervals as described above.
 Thus, for range A-B, A is closed and B is open.  In the above
 example, 15 is closed and 10 is open.  An exception to this rule is
 the case when B=0 is in a decreasing range.  In this case, the range
 is closed on both ends, as otherwise there would be no way to reach 0
 on a reverse playback for formats that have such a notion, like NPT
 and SMPTE.

Schulzrinne, et al. Standards Track [Page 163] RFC 7826 RTSP 2.0 December 2016

 Example:
     Scale: -1
     Range: npt=15-0
 In this range, both 15 and 0 are closed.
 A decreasing range interval without a corresponding negative value in
 the Scale header is not valid.

18.41. Referrer

 The Referrer request-header field allows the client to specify, for
 the server's benefit, the address (URI) of the resource from which
 the Request-URI was obtained.  The URI refers to that of the
 presentation description, typically retrieved via HTTP.  The Referrer
 request-header allows a server to generate lists of back-links to
 resources for interest, logging, optimized caching, etc.  It also
 allows obsolete or mistyped links to be traced for maintenance.  The
 Referrer field MUST NOT be sent if the Request-URI was obtained from
 a source that does not have its own URI, such as input from the user
 keyboard.
 If the field-value is a relative URI, it SHOULD be interpreted
 relative to the Request-URI.  The URI MUST NOT include a fragment
 identifier.
 Because the source of a link might be private information or might
 reveal an otherwise private information source, it is strongly
 recommended that the user be able to select whether or not the
 Referrer field is sent.  For example, a streaming client could have a
 toggle switch for openly/anonymously, which would respectively
 enable/disable the sending of Referrer and From information.
 Clients SHOULD NOT include a Referrer header field in an (non-secure)
 RTSP request if the referring page was transferred with a secure
 protocol.

18.42. Request-Status

 This request-header is used to indicate the end result for requests
 that take time to complete, such as PLAY (Section 13.4).  It is sent
 in PLAY_NOTIFY (Section 13.5) with the end-of-stream reason to report
 how the PLAY request concluded, either in success or in failure.  The
 header carries a reference to the request it reports on using the
 CSeq number and the Session ID used in the request reported on.  This
 is not ensured to be unambiguous due to the fact that the CSeq number
 is scoped by the transport connection.  Agents originating requests

Schulzrinne, et al. Standards Track [Page 164] RFC 7826 RTSP 2.0 December 2016

 can reduce the issue by using a monotonically increasing counter
 across all sequential transports used.  The header provides both a
 numerical status code (according to Section 8.1.1) and a human-
 readable reason phrase.
 Example:
 Request-Status: cseq=63 status=500 reason="Media data unavailable"
 Proxies that renumber the CSeq header need to perform corresponding
 remapping of the cseq parameter in this header when forwarding the
 request to the next-hop agent.

18.43. Require

 The Require request-header field is used by agents to ensure that the
 other endpoint supports features that are required in respect to this
 request.  It can also be used to query if the other endpoint supports
 certain features; however, the use of the Supported general-header
 (Section 18.51) is much more effective in this purpose.  In case any
 of the feature tags listed by the Require header are not supported by
 the server or client receiving the request, it MUST respond to the
 request using the error code 551 (Option Not Supported) and include
 the Unsupported header listing those feature tags that are NOT
 supported.  This header does not apply to proxies; for the same
 functionality with respect to proxies, see the Proxy-Require header
 (Section 18.37) with the exception of media-modifying proxies.
 Media-modifying proxies, due to their nature of handling media in a
 way that is very similar to a server, do need to understand also the
 server's features to correctly serve the client.
    This is to make sure that the client-server interaction will
    proceed without delay when all features are understood by both
    sides and only slow down if features are not understood (as in the
    example below).  For a well-matched client-server pair, the
    interaction proceeds quickly, saving a round trip often required
    by negotiation mechanisms.  In addition, it also removes state
    ambiguity when the client requires features that the server does
    not understand.

Schulzrinne, et al. Standards Track [Page 165] RFC 7826 RTSP 2.0 December 2016

 Example (Not complete):
 C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/2.0
         CSeq: 302
         Require: funky-feature
         Funky-Parameter: funkystuff
 S->C:   RTSP/2.0 551 Option not supported
         CSeq: 302
         Unsupported: funky-feature
 In this example, "funky-feature" is the feature tag that indicates to
 the client that the fictional Funky-Parameter field is required.  The
 relationship between "funky-feature" and Funky-Parameter is not
 communicated via the RTSP exchange, since that relationship is an
 immutable property of "funky-feature" and thus should not be
 transmitted with every exchange.
 Proxies and other intermediary devices MUST ignore this header.  If a
 particular extension requires that intermediate devices support it,
 the extension should be tagged in the Proxy-Require field instead
 (see Section 18.37).  See discussion in the proxies section
 (Section 15.1) about when to consider that a feature requires proxy
 support.

18.44. Retry-After

 The Retry-After response-header field can be used with a 503 (Service
 Unavailable) or 553 (Proxy Unavailable) response to indicate how long
 the service is expected to be unavailable to the requesting client.
 This field MAY also be used with any 3rr (Redirection) response to
 indicate the minimum time the user agent is asked to wait before
 issuing the redirected request.  A response using 413 (Request
 Message Body Too Large) when the restriction is temporary MAY also
 include the Retry-After header.  The value of this field can be
 either an RTSP-date or an integer number of seconds (in decimal)
 after the time of the response.
 Example:
 Retry-After: Fri, 31 Dec 1999 23:59:59 GMT
 Retry-After: 120
 In the latter example, the delay is 2 minutes.

Schulzrinne, et al. Standards Track [Page 166] RFC 7826 RTSP 2.0 December 2016

18.45. RTP-Info

 The RTP-Info general-header field is used to set RTP-specific
 parameters in the PLAY and GET_PARAMETER responses or PLAY_NOTIFY and
 GET_PARAMETER requests.  For streams using RTP as transport protocol,
 the RTP-Info header SHOULD be part of a 200 response to PLAY.
    The exclusion of the RTP-Info in a PLAY response for RTP-
    transported media will result in a client needing to synchronize
    the media streams using RTCP.  This may have negative impact as
    the RTCP can be lost and does not need to be particularly timely
    in its arrival.  Also, functionality that informs the client from
    which packet a seek has occurred is affected.
 The RTP-Info MAY be included in SETUP responses to provide
 synchronization information when changing transport parameters, see
 Section 13.3.  The RTP-Info header and the Range header MAY be
 included in a GET_PARAMETER request from client to server without any
 values to request the current playback point and corresponding RTP
 synchronization information.  When the RTP-Info header is included in
 a Request, the Range header MUST also be included.  The server
 response SHALL include both the Range header and the RTP-Info header.
 If the session is in Play state, then the value of the Range header
 SHALL be filled in with the current playback point and with the
 corresponding RTP-Info values.  If the server is in another state, no
 values are included in the RTP-Info header.  The header is included
 in PLAY_NOTIFY requests with the Notify-Reason of the end of stream
 to provide RTP information about the end of the stream.
 The header can carry the following parameters:
 url:  Indicates the stream URI for which the following RTP parameters
       correspond; this URI MUST be the same as used in the SETUP
       request for this media stream.  Any relative URI MUST use the
       Request-URI as base URI.  This parameter MUST be present.
 ssrc: The SSRC to which the RTP timestamp and sequence number
       provided applies.  This parameter MUST be present.
 seq:  Indicates the sequence number of the first packet of the stream
       that is direct result of the request.  This allows clients to
       gracefully deal with packets when seeking.  The client uses
       this value to differentiate packets that originated before the
       seek from packets that originated after the seek.  Note that a
       client may not receive the packet with the expressed sequence
       number and instead may receive packets with a higher sequence
       number due to packet loss or reordering.  This parameter is
       RECOMMENDED to be present.

Schulzrinne, et al. Standards Track [Page 167] RFC 7826 RTSP 2.0 December 2016

 rtptime:  MUST indicate the RTP timestamp value corresponding to the
       start time value in the Range response-header or, if not
       explicitly given, the implied start point.  The client uses
       this value to calculate the mapping of RTP time to NPT or other
       media timescale.  This parameter SHOULD be present to ensure
       inter-media synchronization is achieved.  There exists no
       requirement that any received RTP packet will have the same RTP
       timestamp value as the one in the parameter used to establish
       synchronization.
    A mapping from RTP timestamps to NTP format timestamps (wallclock)
    is available via RTCP.  However, this information is not
    sufficient to generate a mapping from RTP timestamps to media
    clock time (NPT, etc.).  Furthermore, in order to ensure that this
    information is available at the necessary time (immediately at
    startup or after a seek), and that it is delivered reliably, this
    mapping is placed in the RTSP control channel.
    In order to compensate for drift for long, uninterrupted
    presentations, RTSP clients should additionally map NPT to NTP,
    using initial RTCP sender reports to do the mapping, and later
    reports to check drift against the mapping.
 Example:
 Range:npt=3.25-15
 RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102;
          rtptime=12345678,url="rtsp://example.com/foo/video"
          ssrc=9A9DE123:seq=30211;rtptime=29567112
 Lets assume that Audio uses a 16 kHz RTP timestamp clock and Video
 a 90 kHz RTP timestamp clock.  Then, the media synchronization is
 depicted in the following way.
 NPT    3.0---3.1---3.2-X-3.3---3.4---3.5---3.6
 Audio               PA A
 Video                  V    PV
 X: NPT time value = 3.25, from Range header.
 A: RTP timestamp value for Audio from RTP-Info header (12345678).
 V: RTP timestamp value for Video from RTP-Info header (29567112).
 PA: RTP audio packet carrying an RTP timestamp of 12344878, which
     corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2
 PV: RTP video packet carrying an RTP timestamp of 29573412, which
     corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32

Schulzrinne, et al. Standards Track [Page 168] RFC 7826 RTSP 2.0 December 2016

18.46. Scale

 The Scale general-header indicates the requested or used view rate
 for the media resource being played back.  A scale value of 1
 indicates normal play at the normal forward viewing rate.  If not 1,
 the value corresponds to the rate with respect to normal viewing
 rate.  For example, a value of 2 indicates twice the normal viewing
 rate ("fast forward") and a value of 0.5 indicates half the normal
 viewing rate.  In other words, a value of 2 has content time increase
 at twice the playback time.  For every second of elapsed (wallclock)
 time, 2 seconds of content time will be delivered.  A negative value
 indicates reverse direction.  For certain media transports, this may
 require certain considerations to work consistently; see Appendix C.1
 for description on how RTP handles this.
 The transmitted-data rate SHOULD NOT be changed by selection of a
 different scale value.  The resulting bitrate should be reasonably
 close to the nominal bitrate of the content for scale = 1.  The
 server has to actively manipulate the data when needed to meet the
 bitrate constraints.  Implementation of scale changes depends on the
 server and media type.  For video, a server may, for example, deliver
 only key frames or selected frames.  For audio, it may time-scale the
 audio while preserving pitch or, less desirably, deliver fragments of
 audio, or completely mute the audio.
 The server and content may restrict the range of scale values that it
 supports.  The supported values are indicated by the Media-Properties
 header (Section 18.29).  The client SHOULD only indicate request
 values to be supported.  However, as the values may change as the
 content progresses, a requested value may no longer be valid when the
 request arrives.  Thus, a non-supported value in a request does not
 generate an error, it only forces the server to choose the closest
 value.  The response MUST always contain the actual scale value
 chosen by the server.
 If the server does not implement the possibility to scale, it will
 not return a Scale header.  A server supporting scale operations for
 PLAY MUST indicate this with the use of the "play.scale" feature tag.
 When indicating a negative scale for a reverse playback, the Range
 header MUST indicate a decreasing range as described in
 Section 18.40.
 Example of playing in reverse at 3.5 times normal rate:
   Scale: -3.5
   Range: npt=15-10

Schulzrinne, et al. Standards Track [Page 169] RFC 7826 RTSP 2.0 December 2016

18.47. Seek-Style

 When a client sends a PLAY request with a Range header to perform a
 random access to the media, the client does not know if the server
 will pick the first media samples or the first random access point
 prior to the request range.  Depending on the use case, the client
 may have a strong preference.  To express this preference and provide
 the client with information on how the server actually acted on that
 preference, the Seek-Style general-header is defined.
 Seek-Style is a general-header that MAY be included in any PLAY
 request to indicate the client's preference for any media stream that
 has the random access properties.  The server MUST always include the
 header in any PLAY response for media with random access properties
 to indicate what policy was applied.  A server that receives an
 unknown Seek-Style policy MUST ignore it and select the server
 default policy.  A client receiving an unknown policy MUST ignore it
 and use the Range header and any media synchronization information as
 basis to determine what the server did.
 This specification defines the following seek policies that may be
 requested (see also Section 4.7.1):
 RAP:  Random Access Point (RAP) is the behavior of requesting the
    server to locate the closest previous random access point that
    exists in the media aggregate and deliver from that.  By
    requesting a RAP, media quality will be the best possible as all
    media will be delivered from a point where full media state can be
    established in the media decoder.
 CoRAP:  Conditional Random Access Point (CoRAP) is a variant of the
    above RAP behavior.  This policy is primarily intended for cases
    where there is larger distance between the random access points in
    the media.  CoRAP uses the RAP policy if the condition that there
    is a Random Access Point closer to the requested start point than
    to the current pause point is fulfilled.  Otherwise, no seeking is
    performed and playback will continue from the current pause point.
    This policy assumes that the media state existing prior to the
    pause is usable if delivery is continued.  If the client or server
    knows that this is not the fact, the RAP policy should be used.
    In other words, in most cases when the client requests a start
    point prior to the current pause point, a valid decoding
    dependency chain from the media delivered prior to the pause and
    to the requested media unit will not exist.  If the server
    searched to a random access point, the server MUST return the
    CoRAP policy in the Seek-Style header and adjust the Range header
    to reflect the position of the selected RAP.  In case the random
    access point is farther away and the server chooses to continue

Schulzrinne, et al. Standards Track [Page 170] RFC 7826 RTSP 2.0 December 2016

    from the current pause point, it MUST include the "Next" policy in
    the Seek-Style header and adjust the Range header start point to
    the current pause point.
 First-Prior:  The first-prior policy will start delivery with the
    media unit that has a playout time first prior to the requested
    time.  For discrete media, that would only include media units
    that would still be rendered at the request time.  For continuous
    media, that is media that will be rendered during the requested
    start time of the range.
 Next:  The next media units after the provided start time of the
    range: for continuous framed media, that would mean the first next
    frame after the provided time and for discrete media, the first
    unit that is to be rendered after the provided time.  The main
    usage for this case is when the client knows it has all media up
    to a certain point and would like to continue delivery so that a
    complete uninterrupted media playback can be achieved.  An example
    of such a scenario would be switching from a broadcast/multicast
    delivery to a unicast-based delivery.  This policy MUST only be
    used on the client's explicit request.
 Please note that these expressed preferences exist for optimizing the
 startup time or the media quality.  The "Next" policy breaks the
 normal definition of the Range header to enable a client to request
 media with minimal overlap, although some may still occur for
 aggregated sessions.  RAP and First-Prior both fulfill the
 requirement of providing media from the requested range and forward.
 However, unless RAP is used, the media quality for many media codecs
 using predictive methods can be severely degraded unless additional
 data is available as, for example, already buffered, or through other
 side channels.

18.48. Server

 The Server general-header field contains information about the
 software used by the origin server to create or handle the request.
 This field can contain multiple product tokens and comments
 identifying the server and any significant subproducts.  The product
 tokens are listed in order of their significance for identifying the
 application.
 Example:
 Server: PhonyServer/1.0

Schulzrinne, et al. Standards Track [Page 171] RFC 7826 RTSP 2.0 December 2016

 If the response is being forwarded through a proxy, the proxy
 application MUST NOT modify the Server response-header.  Instead, it
 SHOULD include a Via field (Section 18.57).  If the response is
 generated by the proxy, the proxy application MUST return the Server
 response-header as previously returned by the server.

18.49. Session

 The Session general-header field identifies an RTSP session.  An RTSP
 session is created by the server as a result of a successful SETUP
 request, and in the response, the session identifier is given to the
 client.  The RTSP session exists until destroyed by a TEARDOWN or a
 REDIRECT or is timed out by the server.
 The session identifier is chosen by the server (see Section 4.3) and
 MUST be returned in the SETUP response.  Once a client receives a
 session identifier, it MUST be included in any request related to
 that session.  This means that the Session header MUST be included in
 a request, using the following methods: PLAY, PAUSE, PLAY_NOTIFY and
 TEARDOWN.  It MAY be included in SETUP, OPTIONS, SET_PARAMETER,
 GET_PARAMETER, and REDIRECT.  It MUST NOT be included in DESCRIBE.
 The Session header MUST NOT be included in the following methods, if
 these requests are pipelined and if the session identifier is not yet
 known: PLAY, PAUSE, TEARDOWN, SETUP, OPTIONS SET_PARAMETER, and
 GET_PARAMETER.
 In an RTSP response, the session header MUST be included in methods,
 SETUP, PLAY, PAUSE, and PLAY_NOTIFY, and it MAY be included in
 methods TEARDOWN and REDIRECT.  If included in the request of the
 following methods it MUST also be included in the response: OPTIONS,
 GET_PARAMETER, and SET_PARAMETER.  It MUST NOT be included in
 DESCRIBE responses.
 Note that a session identifier identifies an RTSP session across
 transport sessions or connections.  RTSP requests for a given session
 can use different URIs (Presentation and media URIs).  Note, that
 there are restrictions depending on the session as to which URIs are
 acceptable for a given method.  However, multiple "user" sessions for
 the same URI from the same client will require use of different
 session identifiers.
    The session identifier is needed to distinguish several delivery
    requests for the same URI coming from the same client.
 The response 454 (Session Not Found) MUST be returned if the session
 identifier is invalid.

Schulzrinne, et al. Standards Track [Page 172] RFC 7826 RTSP 2.0 December 2016

 The header MAY include a parameter for session timeout period.  If
 not explicitly provided, this value is set to 60 seconds.  As this
 affects how often session keep-alives are needed, values smaller than
 30 seconds are not recommended.  However, larger-than-default values
 can be useful in applications of RTSP that have inactive but
 established sessions for longer time periods.
    The 60-second value was chosen as the session timeout value as it
    results in keep-alive messages that are not too frequent and low
    sensitivity to variations in request/response timing.  If one
    reduces the timeout value to below 30 seconds, the corresponding
    request/response timeout becomes a significant part of the session
    timeout.  The 60-second value also allows for reasonably rapid
    recovery of committed server resources in case of client failure.

18.50. Speed

 The Speed general-header field requests the server to deliver
 specific amounts of nominal media time per unit of delivery time,
 contingent on the server's ability and desire to serve the media
 stream at the given speed.  The client requests the delivery speed to
 be within a given range with a lower and upper bound.  The server
 SHALL deliver at the highest possible speed within the range, but not
 faster than the upper bound, for which the underlying network path
 can support the resulting transport data rates.  As long as any speed
 value within the given range can be provided, the server SHALL NOT
 modify the media quality.  Only if the server is unable to deliver
 media at the speed value provided by the lower bound shall it reduce
 the media quality.
 Implementation of the Speed functionality by the server is OPTIONAL.
 The server can indicate its support through a feature tag,
 play.speed.  The lack of a Speed header in the response is an
 indication of lack of support of this functionality.
 The speed parameter values are expressed as a positive decimal value,
 e.g., a value of 2.0 indicates that data is to be delivered twice as
 fast as normal.  A speed value of zero is invalid.  The range is
 specified in the form "lower bound - upper bound".  The lower-bound
 value may be smaller or equal to the upper bound.  All speeds may not
 be possible to support.  Therefore, the server MAY modify the
 requested values to the closest supported.  The actual supported
 speed MUST be included in the response.  However, note that the use
 cases may vary and that Speed value ranges such as 0.7-0.8, 0.3-2.0,
 1.0-2.5, and 2.5-2.5 all have their usages.

Schulzrinne, et al. Standards Track [Page 173] RFC 7826 RTSP 2.0 December 2016

 Example:
   Speed: 1.0-2.5
 Use of this header changes the bandwidth used for data delivery.  It
 is meant for use in specific circumstances where delivery of the
 presentation at a higher or lower rate is desired.  The main use
 cases are buffer operations or local scale operations.  Implementers
 should keep in mind that bandwidth for the session may be negotiated
 beforehand (by means other than RTSP) and, therefore, renegotiation
 may be necessary.  To perform Speed operations, the server needs to
 ensure that the network path can support the resulting bitrate.
 Thus, the media transport needs to support feedback so that the
 server can react and adapt to the available bitrate.

18.51. Supported

 The Supported general-header enumerates all the extensions supported
 by the client or server using feature tags.  The header carries the
 extensions supported by the message-sending client or server.  The
 Supported header MAY be included in any request.  When present in a
 request, the receiver MUST respond with its corresponding Supported
 header.  Note that the Supported header is also included in 4xx and
 5xx responses.
 The Supported header contains a list of feature tags, described in
 Section 4.5, that are understood by the client or server.  These
 feature tags are the ones the server or client supports in general
 and are not specific to the request resource.
 Example:
   C->S:  OPTIONS rtsp://example.com/ RTSP/2.0
          Supported: foo, bar, blech
          User-Agent: PhonyClient/1.2
   S->C:  RTSP/2.0 200 OK
          Supported: bar, blech, baz

Schulzrinne, et al. Standards Track [Page 174] RFC 7826 RTSP 2.0 December 2016

18.52. Terminate-Reason

 The Terminate-Reason request-header allows the server, when sending a
 REDIRECT or TEARDOWN request, to provide a reason for the session
 termination and any additional information.  This specification
 identifies three reasons for Redirections and may be extended in the
 future:
 Server-Admin:  The server needs to be shut down for some
    administrative reason.
 Session-Timeout:  A client's session has been kept alive for extended
    periods of time and the server has determined that it needs to
    reclaim the resources associated with this session.
 Internal-Error  An internal error that is impossible to recover from
    has occurred, forcing the server to terminate the session.
 The Server may provide additional parameters containing information
 around the redirect.  This specification defines the following ones.
 time:  Provides a wallclock time when the server will stop providing
    any service.
 user-msg:  A UTF-8 text string with a message from the server to the
    user.  This message SHOULD be displayed to the user.

18.53. Timestamp

 The Timestamp general-header describes when the agent sent the
 request.  The value of the timestamp is of significance only to the
 agent and may use any timescale.  The responding agent MUST echo the
 exact same value and MAY, if it has accurate information about this,
 add a floating-point number indicating the number of seconds that has
 elapsed since it has received the request.  The timestamp can be used
 by the agent to compute the round-trip time to the responding agent
 so that it can adjust the timeout value for retransmissions when
 running over an unreliable protocol.  It also resolves retransmission
 ambiguities for unreliable transport of RTSP.
 Note that the present specification provides only for reliable
 transport of RTSP messages.  The Timestamp general-header is
 specified in case the protocol is extended in the future to use
 unreliable transport.

Schulzrinne, et al. Standards Track [Page 175] RFC 7826 RTSP 2.0 December 2016

18.54. Transport

 The Transport general-header indicates which transport protocol is to
 be used and configures its parameters such as destination address,
 compression, multicast time-to-live and destination port for a single
 stream.  It sets those values not already determined by a
 presentation description.
 A Transport request-header MAY contain a list of transport options
 acceptable to the client, in the form of multiple transport
 specification entries.  Transport specifications are comma separated
 and listed in decreasing order of preference.  Each transport
 specification consists of a transport protocol identifier, followed
 by any number of parameters separated by semicolons.  A Transport
 request-header MAY contain multiple transport specifications using
 the same transport protocol identifier.  The server MUST return a
 Transport response-header in the response to indicate the values
 actually chosen, if any.  If no transport specification is supported,
 no transport header is returned and the response MUST use the status
 code 461 (Unsupported Transport) (Section 17.4.25).  In case more
 than one transport specification was present in the request, the
 server MUST return the single transport specification (transport-
 spec) that was actually chosen, if any.  The number of transport-spec
 entries is expected to be limited as the client will receive guidance
 on what configurations are possible from the presentation
 description.
 The Transport header MAY also be used in subsequent SETUP requests to
 change transport parameters.  A server MAY refuse to change
 parameters of an existing stream.
 The transport protocol identifier defines, for each transport
 specification, which transport protocol to use and any related rules.
 Each transport protocol identifier defines the parameters that are
 required to occur; additional optional parameters MAY occur.  This
 flexibility is provided as parameters may be different and provide
 different options to the RTSP agent.  A transport specification may
 only contain one of any given parameter within it.  A parameter
 consists of a name and optionally a value string.  Parameters MAY be
 given in any order.  Additionally, a transport specification may only
 contain either the unicast or the multicast transport type parameter.
 The transport protocol identifier, and all parameters, need to be
 understood in a transport specification; if not, the transport
 specification MUST be ignored.  An RTSP proxy of any type that uses
 or modifies the transport specification, e.g., access proxy or
 security proxy, MUST remove specifications with unknown parameters

Schulzrinne, et al. Standards Track [Page 176] RFC 7826 RTSP 2.0 December 2016

 before forwarding the RTSP message.  If that results in no remaining
 transport specification, the proxy SHALL send a 461 (Unsupported
 Transport) (Section 17.4.25) response without any Transport header.
    The Transport header is restricted to describing a single media
    stream.  (RTSP can also control multiple streams as a single
    entity.)  Making it part of RTSP rather than relying on a
    multitude of session description formats greatly simplifies
    designs of firewalls.
 The general syntax for the transport protocol identifier is a list of
 slash-separated tokens:
 Value1/Value2/Value3...
 Which, for RTP transports, takes the form:
 RTP/profile/lower-transport.
 The default value for the "lower-transport" parameters is specific to
 the profile.  For RTP/AVP, the default is UDP.
 There are two different methods for how to specify where the media
 should be delivered for unicast transport:
 dest_addr:  The presence of this parameter and its values indicates
       the destination address or addresses (host address and port
       pairs for IP flows) necessary for the media transport.
 No dest_addr:  The lack of the dest_addr parameter indicates that the
       server MUST send media to the same address from which the RTSP
       messages originates.
 The choice of method for indicating where the media is to be
 delivered depends on the use case.  In some cases, the only allowed
 method will be to use no explicit address indication and have the
 server deliver media to the source of the RTSP messages.
 For multicast, there are several methods for specifying addresses,
 but they are different in how they work compared with unicast:
 dest_addr with client picked address:  The address and relevant
       parameters, like TTL (scope), for the actual multicast group to
       deliver the media to.  There are security implications
       (Section 21) with this method that need to be addressed because
       an RTSP server can be used as a DoS attacker on an existing
       multicast group.

Schulzrinne, et al. Standards Track [Page 177] RFC 7826 RTSP 2.0 December 2016

 dest_addr using Session Description Information:  The information
       included in the transport header can all be coming from the
       session description, e.g., the SDP "c=" and "m=" lines.  This
       mitigates some of the security issues of the previous methods
       as it is the session provider that picks the multicast group
       and scope.  The client MUST include the information if it is
       available in the session description.
 No dest_addr:  The behavior when no explicit multicast group is
       present in a request is not defined.
 An RTSP proxy will need to take care.  If the media is not desired to
 be routed through the proxy, the proxy will need to introduce the
 destination indication.
 Below are the configuration parameters associated with transport:
 General parameters:
 unicast / multicast:  This parameter is a mutually exclusive
       indication of whether unicast or multicast delivery will be
       attempted.  One of the two values MUST be specified.  Clients
       that are capable of handling both unicast and multicast
       transmission need to indicate such capability by including two
       full transport-specs with separate parameters for each.
 layers:  The number of multicast layers to be used for this media
       stream.  The layers are sent to consecutive addresses starting
       at the dest_addr address.  If the parameter is not included, it
       defaults to a single layer.
 dest_addr:  A general destination address parameter that can contain
       one or more address specifications.  Each combination of
       protocol/profile/lower transport needs to have the format and
       interpretation of its address specification defined.  For
       RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a
       tuple containing a host address and port.  Note, only a single
       destination parameter per transport spec is intended.  The
       usage of multiple destinations to distribute a single media to
       multiple entities is unspecified.
       The client originating the RTSP request MAY specify the
       destination address of the stream recipient with the host
       address as part of the tuple.  When the destination address is
       specified, the recipient may be a different party than the
       originator of the request.  To avoid becoming the unwitting
       perpetrator of a remote-controlled DoS attack, a server MUST
       perform security checks (see Section 21.2.1) and SHOULD log

Schulzrinne, et al. Standards Track [Page 178] RFC 7826 RTSP 2.0 December 2016

       such attempts before allowing the client to direct a media
       stream to a recipient address not chosen by the server.
       Implementations cannot rely on TCP as a reliable means of
       client identification.  If the server does not allow the host
       address part of the tuple to be set, it MUST return 463
       (Destination Prohibited).
       The host address part of the tuple MAY be empty, for example
       ":58044", in cases when it is desired to specify only the
       destination port.  Responses to requests including the
       Transport header with a dest_addr parameter SHOULD include the
       full destination address that is actually used by the server.
       The server MUST NOT remove address information that is already
       present in the request when responding, unless the protocol
       requires it.
 src_addr:  A general source address parameter that can contain one or
       more address specifications.  Each combination of
       protocol/profile/lower transport needs to have the format and
       interpretation of its address specification defined.  For
       RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a
       tuple containing a host address and port.
       This parameter MUST be specified by the server if it transmits
       media packets from an address other than the one RTSP messages
       are sent to.  This will allow the client to verify the source
       address and give it a destination address for its RTCP feedback
       packets, if RTP is used.  The address or addresses indicated in
       the src_addr parameter SHOULD be used both for the sending and
       receiving of the media stream's data packets.  The main reasons
       are threefold: First, indicating the port and source address(s)
       lets the receiver know where from the packets is expected to
       originate.  Second, traversal of NATs is greatly simplified
       when traffic is flowing symmetrically over a NAT binding.
       Third, certain NAT traversal mechanisms need to know to which
       address and port to send so-called "binding packets" from the
       receiver to the sender, thus creating an address binding in the
       NAT that the sender-to-receiver packet flow can use.
          This information may also be available through SDP.
          However, since this is more a feature of transport than
          media initialization, the authoritative source for this
          information should be in the SETUP response.

Schulzrinne, et al. Standards Track [Page 179] RFC 7826 RTSP 2.0 December 2016

 mode: The mode parameter indicates the methods to be supported for
       this session.  The currently defined valid value is "PLAY".  If
       not provided, the default is "PLAY".  The "RECORD" value was
       defined in RFC 2326; in this specification, it is unspecified
       but reserved.  RECORD and other values may be specified in the
       future.
 interleaved:  The interleaved parameter implies mixing the media
       stream with the control stream in whatever protocol is being
       used by the control stream, using the mechanism defined in
       Section 14.  The argument provides the channel number to be
       used in the $ block (see Section 14) and MUST be present.  This
       parameter MAY be specified as an interval, e.g.,
       interleaved=4-5 in cases where the transport choice for the
       media stream requires it, e.g., for RTP with RTCP.  The channel
       number given in the request is only a guidance from the client
       to the server on what channel number(s) to use.  The server MAY
       set any valid channel number in the response.  The declared
       channels are bidirectional, so both end parties MAY send data
       on the given channel.  One example of such usage is the second
       channel used for RTCP, where both server and client send RTCP
       packets on the same channel.
          This allows RTP/RTCP to be handled similarly to the way that
          it is done with UDP, i.e., one channel for RTP and the other
          for RTCP.
 MIKEY:  This parameter is used in conjunction with transport
       specifications that can utilize MIKEY [RFC3830] for security
       context establishment.  So far, only the SRTP-based RTP
       profiles SAVP and SAVPF can utilize MIKEY, and this is defined
       in Appendix C.1.4.1.  This parameter can be included both in
       request and response messages.  The binary MIKEY message SHALL
       be Base64-encoded [RFC4648] before being included in the value
       part of the parameter, where the encoding adheres to the
       definition in Section 4 of RFC 4648 and where the padding bits
       are set to zero.
 Multicast-specific:
 ttl:  multicast time-to-live for IPv4.  When included in requests,
       the value indicates the TTL value that the client requests the
       server to use.  In a response, the value actually being used by
       the server is returned.  A server will need to consider what
       values that are reasonable and also the authority of the user
       to set this value.  Corresponding functions are not needed for
       IPv6 as the scoping is part of the IPv6 multicast address
       [RFC4291].

Schulzrinne, et al. Standards Track [Page 180] RFC 7826 RTSP 2.0 December 2016

 RTP-specific:
 These parameters MAY only be used if the media-transport protocol is
 RTP.
 ssrc: The ssrc parameter, if included in a SETUP response, indicates
       the RTP SSRC [RFC3550] value(s) that will be used by the media
       server for RTP packets within the stream.  The values are
       expressed as a slash-separated sequence of SSRC values, each
       SSRC expressed as an eight-digit hexadecimal value.
       The ssrc parameter MUST NOT be specified in requests.  The
       functionality of specifying the ssrc parameter in a SETUP
       request is deprecated as it is incompatible with the
       specification of RTP [RFC3550].  If the parameter is included
       in the Transport header of a SETUP request, the server SHOULD
       ignore it, and choose appropriate SSRCs for the stream.  The
       server SHOULD set the ssrc parameter in the Transport header of
       the response.
 RTCP-mux:  Used to negotiate the usage of RTP and RTCP multiplexing
       [RFC5761] on a single underlying transport stream/flow.  The
       presence of this parameter in a SETUP request indicates the
       client's support and requires the server to use RTP and RTCP
       multiplexing.  The client SHALL only include one transport
       stream in the Transport header specification.  To provide the
       server with a choice between using RTP/RTCP multiplexing or
       not, two different transport header specifications must be
       included.
 The parameter setup and connection defined below MAY only be used if
 the media-transport protocol of the lower-level transport is
 connection oriented (such as TCP).  However, these parameters MUST
 NOT be used when interleaving data over the RTSP connection.
 setup:  Clients use the setup parameter on the Transport line in a
       SETUP request to indicate the roles it wishes to play in a TCP
       connection.  This parameter is adapted from [RFC4145].  The use
       of this parameter in RTP/AVP/TCP non-interleaved transport is
       discussed in Appendix C.2.2; the discussion below is limited to
       syntactic issues.  Clients may specify the following values for
       the setup parameter:
       active:  The client will initiate an outgoing connection.
       passive:  The client will accept an incoming connection.

Schulzrinne, et al. Standards Track [Page 181] RFC 7826 RTSP 2.0 December 2016

       actpass:  The client is willing to accept an incoming
          connection or to initiate an outgoing connection.
       If a client does not specify a setup value, the "active" value
       is assumed.
       In response to a client SETUP request where the setup parameter
       is set to "active", a server's 2xx reply MUST assign the setup
       parameter to "passive" on the Transport header line.
       In response to a client SETUP request where the setup parameter
       is set to "passive", a server's 2xx reply MUST assign the setup
       parameter to "active" on the Transport header line.
       In response to a client SETUP request where the setup parameter
       is set to "actpass", a server's 2xx reply MUST assign the setup
       parameter to "active" or "passive" on the Transport header
       line.
       Note that the "holdconn" value for setup is not defined for
       RTSP use, and MUST NOT appear on a Transport line.
 connection:  Clients use the connection parameter in a transport
       specification part of the Transport header in a SETUP request
       to indicate the client's preference for either reusing an
       existing connection between client and server (in which case
       the client sets the "connection" parameter to "existing") or
       requesting the creation of a new connection between client and
       server (in which cast the client sets the "connection"
       parameter to "new").  Typically, clients use the "new" value
       for the first SETUP request for a URL, and "existing" for
       subsequent SETUP requests for a URL.
       If a client SETUP request assigns the "new" value to
       "connection", the server response MUST also assign the "new"
       value to "connection" on the Transport line.
       If a client SETUP request assigns the "existing" value to
       "connection", the server response MUST assign a value of
       "existing" or "new" to "connection" on the Transport line, at
       its discretion.
       The default value of "connection" is "existing", for all SETUP
       requests (initial and subsequent).
 The combination of transport protocol, profile and lower transport
 needs to be defined.  A number of combinations are defined in the
 Appendix C.

Schulzrinne, et al. Standards Track [Page 182] RFC 7826 RTSP 2.0 December 2016

 Below is a usage example, showing a client advertising the capability
 to handle multicast or unicast, preferring multicast.  Since this is
 a unicast-only stream, the server responds with the proper transport
 parameters for unicast.
   C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
         CSeq: 302
         Transport: RTP/AVP;multicast;mode="PLAY",
             RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
             "192.0.2.5:3457";mode="PLAY"
         Accept-Ranges: npt, smpte, clock
         User-Agent: PhonyClient/1.2
   S->C: RTSP/2.0 200 OK
         CSeq: 302
         Date: Fri, 20 Dec 2013 10:20:32 +0000
         Session: rQi1hBrGlFdiYld241FxUO
         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
            "192.0.2.5:3457";src_addr="192.0.2.224:6256"/
            "192.0.2.224:6257";mode="PLAY"
         Accept-Ranges: npt
         Media-Properties: Random-Access=0.6, Dynamic,
                           Time-Limited=20081128T165900

18.55. Unsupported

 The Unsupported response-header lists the features not supported by
 the responding RTSP agent.  In the case where the feature was
 specified via the Proxy-Require field (Section 18.37), if there is a
 proxy on the path between the client and the server, the proxy MUST
 send a response message with a status code of 551 (Option Not
 Supported).  The request MUST NOT be forwarded.
 See Section 18.43 for a usage example.

Schulzrinne, et al. Standards Track [Page 183] RFC 7826 RTSP 2.0 December 2016

18.56. User-Agent

 The User-Agent general-header field contains information about the
 user agent originating the request or producing a response.  This is
 for statistical purposes, the tracing of protocol violations, and
 automated recognition of user agents for the sake of tailoring
 responses to avoid particular user agent limitations.  User agents
 SHOULD include this field with requests.  The field can contain
 multiple product tokens and comments identifying the agent and any
 subproducts which form a significant part of the user agent.  By
 convention, the product tokens are listed in order of their
 significance for identifying the application.
 Example:
 User-Agent: PhonyClient/1.2

18.57. Via

 The Via general-header field MUST be used by proxies to indicate the
 intermediate protocols and recipients between the user agent and the
 server on requests and between the origin server and the client on
 responses.  The field is intended to be used for tracking message
 forwards, avoiding request loops, and identifying the protocol
 capabilities of all senders along the request/response chain.
 Each of multiple values in the Via field represents each proxy that
 has forwarded the message.  Each recipient MUST append its
 information such that the end result is ordered according to the
 sequence of forwarding applications.  So messages originating with
 the client or server do not include the Via header.  The first proxy
 or other intermediate adds the header and its information into the
 field.  Any additional intermediate adds additional field-values.
 Resulting in the server seeing the chains of intermediates in a
 client-to-server request and the client seeing the full chain in the
 response message.
 Proxies (e.g., Access Proxy or Translator Proxy) SHOULD NOT, by
 default, forward the names and ports of hosts within the private/
 protected region.  This information SHOULD only be propagated if
 explicitly enabled.  If not enabled, the via-received of any host
 behind the firewall/NAT SHOULD be replaced by an appropriate
 pseudonym for that host.

Schulzrinne, et al. Standards Track [Page 184] RFC 7826 RTSP 2.0 December 2016

 For organizations that have strong privacy requirements for hiding
 internal structures, a proxy MAY combine an ordered subsequence of
 Via header field entries with identical sent-protocol values into a
 single such entry.  Applications MUST NOT combine entries that have
 different received-protocol values.

18.58. WWW-Authenticate

 The WWW-Authenticate header is specified in [RFC7235]; its usage
 depends on the used authentication schemes, such as Digest [RFC7616]
 and Basic [RFC7617].  The WWW-Authenticate response-header field MUST
 be included in 401 (Unauthorized) response messages.  The field-value
 consists of at least one challenge that indicates the authentication
 scheme(s) and parameters applicable to the Request-URI.  This header
 MUST only be used in response messages related to client to server
 requests.
 The HTTP access authentication process is described in [RFC7235] with
 some clarification in Section 19.1.  User agents are advised to take
 special care in parsing the WWW-Authenticate field-value as it might
 contain more than one challenge, or if more than one WWW-Authenticate
 header field is provided, the contents of a challenge itself can
 contain a comma-separated list of authentication parameters.

19. Security Framework

 The RTSP security framework consists of two high-level components:
 the pure authentication mechanisms based on HTTP authentication and
 the message transport protection based on TLS, which is independent
 of RTSP.  Because of the similarity in syntax and usage between RTSP
 servers and HTTP servers, the security for HTTP is reused to a large
 extent.

19.1. RTSP and HTTP Authentication

 RTSP and HTTP share common authentication schemes; thus, they follow
 the same framework as specified in [RFC7235].  RTSP uses the
 corresponding RTSP error codes (401 and 407) and headers (WWW-
 Authenticate, Authorization, Proxy-Authenticate, Proxy-Authorization)
 by importing the definitions from [RFC7235].  Servers SHOULD
 implement both the Basic [RFC7617] and the Digest [RFC7616]
 authentication schemes.  Clients MUST implement both the Basic and
 the Digest authentication schemes so that a server that requires the
 client to authenticate can trust that the capability is present.  If
 implementing the Digest authentication scheme, then the additional
 considerations specified below in Section 19.1.1 MUST be followed.

Schulzrinne, et al. Standards Track [Page 185] RFC 7826 RTSP 2.0 December 2016

 It should be stressed that using the HTTP authentication alone does
 not provide full RTSP message security.  Therefore, TLS SHOULD be
 used; see Section 19.2.  Any RTSP message containing an Authorization
 header using the Basic authentication scheme MUST be using a TLS
 connection with confidentiality protection enabled, i.e., no NULL
 encryption.
 In cases where there is a chain of proxies between the client and the
 server, each proxy may individually request the client or previous
 proxy to authenticate itself.  This is done using the Proxy-
 Authenticate (Section 18.34), the Proxy-Authorization
 (Section 18.36), and the Proxy-Authentication-Info (Section 18.35)
 headers.  These headers are hop-by-hop headers and are only scoped to
 the current connection and hop.  Thus, if a proxy chain exists, a
 proxy connecting to another proxy will have to act as a client to
 authorize itself towards the next proxy.  The WWW-Authenticate
 (Section 18.58), Authorization (Section 18.8), and Authentication-
 Info (Section 18.7) headers are end-to-end and MUST NOT be modified
 by proxies.
 This authentication mechanism works only for client-to-server
 requests as currently defined.  This leaves server-to-client request
 outside of the context of TLS-based communication more vulnerable to
 message-injection attacks on the client.  Based on the server-to-
 client methods that exist, the potential risks are various: hijacking
 (REDIRECT), denial of service (TEARDOWN and PLAY_NOTIFY), or attacks
 with uncertain results (SET_PARAMETER).

19.1.1. Digest Authentication

 This section describes the modifications and clarifications required
 to apply the HTTP Digest authentication scheme to RTSP.  The RTSP
 scheme usage is almost completely identical to that for HTTP
 [RFC7616].  These modifications are based on the procedures defined
 for SIP 2.0 [RFC3261] (in Section 22.4) but updated to use RFC 7235,
 RFC 7616 and RFC 7615 instead of RFC 2617.
 Digest authentication uses two additional headers, Authentication-
 Info and Proxy-Authentication-Info, that are defined as in [RFC7615].
 The rules for Digest authentication follow those defined in
 [RFC7616], with "HTTP/1.1" replaced by "RTSP/2.0" in addition to the
 following differences:
 1.  Use the ABNF specified in the referenced documents, with the
     difference that the URI parameter uses the request URI format for
     RTSP, i.e. the ABNF element: Request-URI (see Section 20.2.1).
     The domain parameter uses the RTSP-URI-Ref element for absolute
     and relative URIs.

Schulzrinne, et al. Standards Track [Page 186] RFC 7826 RTSP 2.0 December 2016

 2.  If MTags are used, then the example procedure for choosing a
     nonce based on ETag can work, based on replacing the ETag with
     the MTag.
 3.  As a clarification to the calculation of the A2 value for message
     integrity assurance in the Digest authentication scheme,
     implementers should assume, when the entity-body is empty (that
     is, when the RTSP messages have no message body) that the hash of
     the message body resolves to the hash of an empty string, or:
     H(entity-body), example MD5("") =
     "d41d8cd98f00b204e9800998ecf8427e".

19.2. RTSP over TLS

 RTSP agents MUST implement RTSP over TLS as defined in this section
 and the next Section 19.3.  RTSP MUST follow the same guidelines with
 regard to TLS [RFC5246] usage as specified for HTTP; see [RFC2818].
 RTSP over TLS is separated from unsecured RTSP both on the URI level
 and the port level.  Instead of using the "rtsp" scheme identifier in
 the URI, the "rtsps" scheme identifier MUST be used to signal RTSP
 over TLS.  If no port is given in a URI with the "rtsps" scheme, port
 322 MUST be used for TLS over TCP/IP.
 When a client tries to set up an insecure channel to the server
 (using the "rtsp" URI), and the policy for the resource requires a
 secure channel, the server MUST redirect the client to the secure
 service by sending a 301 redirect response code together with the
 correct Location URI (using the "rtsps" scheme).  A user or client
 MAY upgrade a non secured URI to a secured by changing the scheme
 from "rtsp" to "rtsps".  A server implementing support for "rtsps"
 MUST allow this.
 It should be noted that TLS allows for mutual authentication (when
 using both server and client certificates).  Still, one of the more
 common ways TLS is used is to provide only server-side authentication
 (often to avoid client certificates).  TLS is then used in addition
 to HTTP authentication, providing transport security and server
 authentication, while HTTP Authentication is used to authenticate the
 client.
 RTSP includes the possibility to keep a TCP session up between the
 client and server, throughout the RTSP session lifetime.  It may be
 convenient to keep the TCP session, not only to save the extra setup
 time for TCP, but also the extra setup time for TLS (even if TLS uses
 the resume function, there will be almost two extra round trips).
 Still, when TLS is used, such behavior introduces extra active state
 in the server, not only for TCP and RTSP, but also for TLS.  This may
 increase the vulnerability to DoS attacks.

Schulzrinne, et al. Standards Track [Page 187] RFC 7826 RTSP 2.0 December 2016

 There exists a potential security vulnerability when reusing TCP and
 TLS state for different resources (URIs).  If two different hostnames
 point at the same IP address, it can be desirable to reuse the TCP/
 TLS connection to that server.  In that case, the RTSP agent having
 the TCP/TLS connection MUST verify that the server certificate
 associated with the connection has a SubjectAltName matching the
 hostname present in the URI for the resource an RTSP request is to be
 issued.
 In addition to these recommendations, Section 19.3 gives further
 recommendations of TLS usage with proxies.

19.3. Security and Proxies

 The nature of a proxy is often to act as a "man in the middle", while
 security is often about preventing the existence of one.  This
 section provides clients with the possibility to use proxies even
 when applying secure transports (TLS) between the RTSP agents.  The
 TLS proxy mechanism allows for server and proxy identification using
 certificates.  However, the client cannot be identified based on
 certificates.  The client needs to select between using the procedure
 specified below or using a TLS connection directly (bypassing any
 proxies) to the server.  The choice may be dependent on policies.
 In general, there are two categories of proxies: the transparent
 proxies (of which the client is not aware) and the non-transparent
 proxies (of which the client is aware).  This memo specifies only
 non-transparent RTSP proxies, i.e., proxies visible to the RTSP
 client and RTSP server.  An infrastructure based on proxies requires
 that the trust model be such that both client and server can trust
 the proxies to handle the RTSP messages correctly.  To be able to
 trust a proxy, the client and server also need to be aware of the
 proxy.  Hence, transparent proxies cannot generally be seen as
 trusted and will not work well with security (unless they work only
 at the transport layer).  In the rest of this section, any reference
 to "proxy" will be to a non-transparent proxy, which inspects or
 manipulates the RTSP messages.
 HTTP Authentication is built on the assumption of proxies and can
 provide user-proxy authentication and proxy-proxy/server
 authentication in addition to the client-server authentication.
 When TLS is applied and a proxy is used, the client will connect to
 the proxy's address when connecting to any RTSP server.  This implies
 that for TLS, the client will authenticate the proxy server and not
 the end server.  Note that when the client checks the server

Schulzrinne, et al. Standards Track [Page 188] RFC 7826 RTSP 2.0 December 2016

 certificate in TLS, it MUST check the proxy's identity (URI or
 possibly other known identity) against the proxy's identity as
 presented in the proxy's Certificate message.
 The problem is that for a proxy accepted by the client, the proxy
 needs to be provided information on which grounds it should accept
 the next-hop certificate.  Both the proxy and the user may have rules
 for this, and the user should have the possibility to select the
 desired behavior.  To handle this case, the Accept-Credentials header
 (see Section 18.2) is used, where the client can request the proxy or
 proxies to relay back the chain of certificates used to authenticate
 any intermediate proxies as well as the server.  The assumption that
 the proxies are viewed as trusted gives the user a possibility to
 enforce policies on each trusted proxy of whether it should accept
 the next agent in the chain.  However, it should be noted that not
 all deployments will return the chain of certificates used to
 authenticate any intermediate proxies as well as the server.  An
 operator of such a deployment may want to hide its topology from the
 client.  It should be noted well that the client does not have any
 insight into the proxy's operation.  Even if the proxy is trusted, it
 can still return an incomplete chain of certificates.
 A proxy MUST use TLS for the next hop if the RTSP request includes an
 "rtsps" URI.  TLS MAY be applied on intermediate links (e.g., between
 client and proxy or between proxy and proxy) even if the resource and
 the end server are not required to use it.  The chain of proxies used
 by a client to reach a server and its TLS sessions MUST have
 commensurate security.  Therefore, a proxy MUST, when initiating the
 next-hop TLS connection, use the incoming TLS connections cipher-
 suite list, only modified by removing any cipher suites that the
 proxy does not support.  In case a proxy fails to establish a TLS
 connection due to cipher-suite mismatch between proxy and next-hop
 proxy or server, this is indicated using error code 472 (Failure to
 Establish Secure Connection).

19.3.1. Accept-Credentials

 The Accept-Credentials header can be used by the client to distribute
 simple authorization policies to intermediate proxies.  The client
 includes the Accept-Credentials header to dictate how the proxy
 treats the server / next proxy certificate.  There are currently
 three methods defined:
 Any:  With "any", the proxy (or proxies) MUST accept whatever
       certificate is presented.  Of course, this is not a recommended
       option to use, but it may be useful in certain circumstances
       (such as testing).

Schulzrinne, et al. Standards Track [Page 189] RFC 7826 RTSP 2.0 December 2016

 Proxy:  For the "proxy" method, the proxy (or proxies) MUST use its
       own policies to validate the certificate and decide whether or
       not to accept it.  This is convenient in cases where the user
       has a strong trust relation with the proxy.  Reasons why a
       strong trust relation may exist are personal/company proxy or
       the proxy has an out-of-band policy configuration mechanism.
 User: For the "user" method, the proxy (or proxies) MUST send
       credential information about the next hop to the client for
       authorization.  The client can then decide whether or not the
       proxy should accept the certificate.  See Section 19.3.2 for
       further details.
 If the Accept-Credentials header is not included in the RTSP request
 from the client, then the "Proxy" method MUST be used as default.  If
 a method other than the "Proxy" is to be used, then the Accept-
 Credentials header MUST be included in all of the RTSP requests from
 the client.  This is because it cannot be assumed that the proxy
 always keeps the TLS state or the user's previous preference between
 different RTSP messages (in particular, if the time interval between
 the messages is long).
 With the "Any" and "Proxy" methods, the proxy will apply the policy
 as defined for each method.  If the policy does not accept the
 credentials of the next hop, the proxy MUST respond with a message
 using status code 471 (Connection Credentials Not Accepted).
 An RTSP request in the direction server to client MUST NOT include
 the Accept-Credentials header.  As for the non-secured communication,
 the possibility for these requests depends on the presence of a
 client established connection.  However, if the server-to-client
 request is in relation to a session established over a TLS secured
 channel, it MUST be sent in a TLS secured connection.  That secured
 connection MUST also be the one used by the last client-to-server
 request.  If no such transport connection exists at the time when the
 server desires to send the request, the server MUST discard the
 message.
 Further policies MAY be defined and registered, but this should be
 done with caution.

19.3.2. User-Approved TLS Procedure

 For the "User" method, each proxy MUST perform the following
 procedure for each RTSP request:
 o  Set up the TLS session to the next hop if not already present
    (i.e., run the TLS handshake, but do not send the RTSP request).

Schulzrinne, et al. Standards Track [Page 190] RFC 7826 RTSP 2.0 December 2016

 o  Extract the peer certificate chain for the TLS session.
 o  Check if a matching identity and hash of the peer certificate are
    present in the Accept-Credentials header.  If present, send the
    message to the next hop and conclude these procedures.  If not, go
    to the next step.
 o  The proxy responds to the RTSP request with a 470 or 407 response
    code.  The 407 response code MAY be used when the proxy requires
    both user and connection authorization from user or client.  In
    this message the proxy MUST include a Connection-Credentials
    header, see Section 18.13, with the next hop's identity and
    certificate.
 The client MUST upon receiving a 470 (Connection Authorization
 Required) or 407 (Proxy Authentication Required) response with
 Connection-Credentials header take the decision on whether or not to
 accept the certificate (if it cannot do so, the user SHOULD be
 consulted).  Using IP addresses in the next-hop URI and certificates
 rather than domain names makes it very difficult for a user to
 determine whether or not it should approve the next hop.  Proxies are
 RECOMMENDED to use domain names to identify themselves in URIs and in
 the certificates.  If the certificate is accepted, the client has to
 again send the RTSP request.  In that request, the client has to
 include the Accept-Credentials header including the hash over the
 DER-encoded certificate for all trusted proxies in the chain.

Schulzrinne, et al. Standards Track [Page 191] RFC 7826 RTSP 2.0 December 2016

 Example:
 C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
       CSeq: 2
       Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
                  "192.0.2.5:4589"
       Accept-Ranges: npt, smpte, clock
       Accept-Credentials: User
 P->C: RTSP/2.0 470 Connection Authorization Required
       CSeq: 2
       Connection-Credentials: "rtsps://test.example.org";
       MIIDNTCCAp...
 C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
       CSeq: 3
       Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
                  "192.0.2.5:4589"
       Accept-Credentials: User "rtsps://test.example.org";sha-256;
       dPYD7txpoGTbAqZZQJ+vaeOkyH4=
       Accept-Ranges: npt, smpte, clock
 P->S: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
       CSeq: 3
       Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
                  "192.0.2.5:4589"
       Via: RTSP/2.0 proxy.example.org
       Accept-Credentials: User "rtsps://test.example.org";sha-256;
       dPYD7txpoGTbAqZZQJ+vaeOkyH4=
       Accept-Ranges: npt, smpte, clock
 One implication of this process is that the connection for secured
 RTSP messages may take significantly more round-trip times for the
 first message.  A complete extra message exchange between the proxy
 connecting to the next hop and the client results because of the
 process for approval for each hop.  However, if each message contains
 the chain of proxies that the requester accepts, the remaining
 message exchange should not be delayed.  The procedure of including
 the credentials in each request rather than building state in each
 proxy avoids the need for revocation procedures.

20. Syntax

 The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF)
 as defined in RFC 5234 [RFC5234].  It uses the basic definitions
 present in RFC 5234.

Schulzrinne, et al. Standards Track [Page 192] RFC 7826 RTSP 2.0 December 2016

 Please note that ABNF strings, e.g., "Accept", are case insensitive
 as specified in Section 2.3 of RFC 5234.
 The RTSP syntax makes use of the ISO 10646 character set in UTF-8
 encoding [RFC3629].

20.1. Base Syntax

 RTSP header values can be folded onto multiple lines if the
 continuation line begins with a space or horizontal tab.  All linear
 whitespace, including folding, has the same semantics as SP.  A
 recipient MAY replace any linear whitespace with a single SP before
 interpreting the field-value or forwarding the message downstream.
 The SWS construct is used when linear whitespace is optional,
 generally between tokens and separators.
 To separate the header name from the rest of value, a colon is used,
 which, by the above rule, allows whitespace before, but no line
 break, and whitespace after, including a line break.  The HCOLON
 defines this construct.
 OCTET           =  %x00-FF ; any 8-bit sequence of data
 CHAR            =  %x01-7F ; any US-ASCII character (octets 1 - 127)
 UPALPHA         =  %x41-5A ; any US-ASCII uppercase letter "A".."Z"
 LOALPHA         =  %x61-7A ; any US-ASCII lowercase letter "a".."z"
 ALPHA           =  UPALPHA / LOALPHA
 DIGIT           =  %x30-39 ; any US-ASCII digit "0".."9"
 CTL             =  %x00-1F / %x7F  ; any US-ASCII control character
                    ; (octets 0 - 31) and DEL (127)
 CR              =  %x0D ; US-ASCII CR, carriage return (13)
 LF              =  %x0A  ; US-ASCII LF, linefeed (10)
 SP              =  %x20  ; US-ASCII SP, space (32)
 HT              =  %x09  ; US-ASCII HT, horizontal-tab (9)
 BACKSLASH       =  %x5C  ; US-ASCII backslash (92)
 CRLF            =  CR LF
 LWS             =  [CRLF] 1*( SP / HT ) ; Line-breaking whitespace
 SWS             =  [LWS] ; Separating whitespace
 HCOLON          =  *( SP / HT ) ":" SWS
 TEXT            =  %x20-7E / %x80-FF  ; any OCTET except CTLs
 tspecials       =  "(" / ")" / "<" / ">" / "@"
                 /  "," / ";" / ":" / BACKSLASH / DQUOTE
                 /  "/" / "[" / "]" / "?" / "="
                 /  "{" / "}" / SP / HT
 token           =  1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
                 /  %x41-5A / %x5E-7A / %x7C / %x7E)
                    ; 1*<any CHAR except CTLs or tspecials>
 quoted-string   =  ( DQUOTE *qdtext DQUOTE )

Schulzrinne, et al. Standards Track [Page 193] RFC 7826 RTSP 2.0 December 2016

 qdtext          = %x20-21 / %x23-5B / %x5D-7E / quoted-pair
                 / UTF8-NONASCII
                 ; No DQUOTE and no "\"
 quoted-pair     = "\\" / ( "\" DQUOTE )
 ctext           =  %x20-27 / %x2A-7E
                 /  %x80-FF  ; any OCTET except CTLs, "(" and ")"
 generic-param   =  token [ EQUAL gen-value ]
 gen-value       =  token / host / quoted-string
 safe            =  "$" / "-" / "_" / "." / "+"
 extra           =  "!" / "*" / "'" / "(" / ")" / ","
 rtsp-extra      =  "!" / "*" / "'" / "(" / ")"
 HEX             =  DIGIT / "A" / "B" / "C" / "D" / "E" / "F"
                 /  "a" / "b" / "c" / "d" / "e" / "f"
 LHEX            =  DIGIT /  "a" / "b" / "c" / "d" / "e" / "f"
                    ; lowercase "a-f" Hex
 reserved        =  ";" / "/" / "?" / ":" / "@" / "&" / "="
 unreserved      =  ALPHA / DIGIT / safe / extra
 rtsp-unreserved  =  ALPHA / DIGIT / safe / rtsp-extra
 base64          =  *base64-unit [base64-pad]
 base64-unit     =  4base64-char
 base64-pad      =  (2base64-char "==") / (3base64-char "=")
 base64-char     =  ALPHA / DIGIT / "+" / "/"
 SLASH    =  SWS "/" SWS ; slash
 EQUAL    =  SWS "=" SWS ; equal
 LPAREN   =  SWS "(" SWS ; left parenthesis
 RPAREN   =  SWS ")" SWS ; right parenthesis
 COMMA    =  SWS "," SWS ; comma
 SEMI     =  SWS ";" SWS ; semicolon
 COLON    =  SWS ":" SWS ; colon
 MINUS    =  SWS "-" SWS ; minus/dash
 LDQUOT   =  SWS DQUOTE ; open double quotation mark
 RDQUOT   =  DQUOTE SWS ; close double quotation mark
 RAQUOT   =  ">" SWS ; right angle quote
 LAQUOT   =  SWS "<" ; left angle quote
 TEXT-UTF8char    =  %x21-7E / UTF8-NONASCII
 UTF8-NONASCII    = UTF8-2 / UTF8-3 / UTF8-4
 UTF8-1           = <As defined in RFC 3629>
 UTF8-2           = <As defined in RFC 3629>
 UTF8-3           = <As defined in RFC 3629>
 UTF8-4           = <As defined in RFC 3629>
 UTF8-tail        = <As defined in RFC 3629>

Schulzrinne, et al. Standards Track [Page 194] RFC 7826 RTSP 2.0 December 2016

 POS-FLOAT        = 1*12DIGIT ["." 1*9DIGIT]
 FLOAT            = ["-"] POS-FLOAT

20.2. RTSP Protocol Definition

20.2.1. Generic Protocol Elements

 RTSP-IRI       =  schemes ":" IRI-rest
 IRI-rest       =  ihier-part [ "?" iquery ]
 ihier-part     =  "//" iauthority ipath-abempty
 RTSP-IRI-ref   =  RTSP-IRI / irelative-ref
 irelative-ref  =  irelative-part [ "?" iquery ]
 irelative-part =  "//" iauthority ipath-abempty
                   / ipath-absolute
                   / ipath-noscheme
                   / ipath-empty
 iauthority     =  < As defined in RFC 3987>
 ipath          =  ipath-abempty   ; begins with "/" or is empty
                   / ipath-absolute  ; begins with "/" but not "//"
                   / ipath-noscheme  ; begins with a non-colon segment
                   / ipath-rootless  ; begins with a segment
                   / ipath-empty     ; zero characters
 ipath-abempty   =  *( "/" isegment )
 ipath-absolute  =  "/" [ isegment-nz *( "/" isegment ) ]
 ipath-noscheme  =  isegment-nz-nc *( "/" isegment )
 ipath-rootless  =  isegment-nz *( "/" isegment )
 ipath-empty     =  0<ipchar>
 isegment        =  *ipchar [";" *ipchar]
 isegment-nz     =  1*ipchar [";" *ipchar]
                    / ";" *ipchar
 isegment-nz-nc  =  (1*ipchar-nc [";" *ipchar-nc])
                    / ";" *ipchar-nc
                    ; non-zero-length segment without any colon ":"
                    ; No parameter (; delimited) inside path.
 ipchar         =  iunreserved / pct-encoded / sub-delims / ":" / "@"
 ipchar-nc      =  iunreserved / pct-encoded / sub-delims / "@"
                   ; sub-delims is different from RFC 3987
                   ; not including ";"
 iquery         =  < As defined in RFC 3987>
 iunreserved    =  < As defined in RFC 3987>
 pct-encoded    =  < As defined in RFC 3987>

Schulzrinne, et al. Standards Track [Page 195] RFC 7826 RTSP 2.0 December 2016

 RTSP-URI       =  schemes ":" URI-rest
 RTSP-REQ-URI   =  schemes ":" URI-req-rest
 RTSP-URI-Ref   =  RTSP-URI / RTSP-Relative
 RTSP-REQ-Ref   =  RTSP-REQ-URI / RTSP-REQ-Rel
 schemes        =  "rtsp" / "rtsps" / scheme
 scheme         =  < As defined in RFC 3986>
 URI-rest       =  hier-part [ "?" query ]
 URI-req-rest   =  hier-part [ "?" query ]
                   ; Note fragment part not allowed in requests
 hier-part      =  "//" authority path-abempty
 RTSP-Relative  =  relative-part [ "?" query ]
 RTSP-REQ-Rel   =  relative-part [ "?" query ]
 relative-part  =  "//" authority path-abempty
                   / path-absolute
                   / path-noscheme
                   / path-empty
 authority      =  < As defined in RFC 3986>
 query          =  < As defined in RFC 3986>
 path           =  path-abempty    ; begins with "/" or is empty
                   / path-absolute ; begins with "/" but not "//"
                   / path-noscheme ; begins with a non-colon segment
                   / path-rootless ; begins with a segment
                   / path-empty    ; zero characters
 path-abempty   =  *( "/" segment )
 path-absolute  =  "/" [ segment-nz *( "/" segment ) ]
 path-noscheme  =  segment-nz-nc *( "/" segment )
 path-rootless  =  segment-nz *( "/" segment )
 path-empty     =  0<pchar>
 segment        =  *pchar [";" *pchar]
 segment-nz     =  ( 1*pchar [";" *pchar]) / (";" *pchar)
 segment-nz-nc  =  ( 1*pchar-nc [";" *pchar-nc]) / (";" *pchar-nc)
                   ; non-zero-length segment without any colon ":"
                   ; No parameter (; delimited) inside path.
 pchar          =  unreserved / pct-encoded / sub-delims / ":" / "@"
 pchar-nc       =  unreserved / pct-encoded / sub-delims / "@"
 sub-delims     =  "!" / "$" / "&" / "'" / "(" / ")"
                   / "*" / "+" / "," / "="
                   ; sub-delims is different from RFC 3986/3987
                   ; not including ";"

Schulzrinne, et al. Standards Track [Page 196] RFC 7826 RTSP 2.0 December 2016

 smpte-range        =  smpte-type [EQUAL smpte-range-spec]
                       ; See section 4.4
 smpte-range-spec   =  ( smpte-time "-" [ smpte-time ] )
                    /  ( "-" smpte-time )
 smpte-type         =  "smpte" / "smpte-30-drop"
                    /  "smpte-25" / smpte-type-extension
                    ; other timecodes may be added
 smpte-type-extension  =  "smpte" token
 smpte-time         =  1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
                       [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]
 npt-range        =  "npt" [EQUAL npt-range-spec]
 npt-range-spec   =  ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
 npt-time         =  "now" / npt-sec / npt-hhmmss / npt-hhmmss-comp
 npt-sec          =  1*19DIGIT [ "." 1*9DIGIT ]
 npt-hhmmss       =  npt-hh ":" npt-mm ":" npt-ss [ "." 1*9DIGIT ]
 npt-hh           =  2*19DIGIT   ; any positive number
 npt-mm           =  2*2DIGIT  ; 0-59
 npt-ss           =  2*2DIGIT  ; 0-59
 npt-hhmmss-comp  =  npt-hh-comp ":" npt-mm-comp ":" npt-ss-comp
                     [ "." 1*9DIGIT ] ; Compatibility format
 npt-hh-comp      =  1*19DIGIT   ; any positive number
 npt-mm-comp      =  1*2DIGIT  ; 0-59
 npt-ss-comp      =  1*2DIGIT  ; 0-59
 utc-range        =  "clock" [EQUAL utc-range-spec]
 utc-range-spec   =  ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
 utc-time         =  utc-date "T" utc-clock "Z"
 utc-date         =  8DIGIT
 utc-clock        =  6DIGIT [ "." 1*9DIGIT ]
 feature-tag       =  token
 session-id        =  1*256( ALPHA / DIGIT / safe )
 extension-header  =  header-name HCOLON header-value
 header-name       =  token
 header-value      =  *(TEXT-UTF8char / LWS)

Schulzrinne, et al. Standards Track [Page 197] RFC 7826 RTSP 2.0 December 2016

20.2.2. Message Syntax

 RTSP-message  = Request / Response  ; RTSP/2.0 messages
 Request       = Request-Line
                 *((general-header
                 /  request-header
                 /  message-body-header) CRLF)
                 CRLF
                 [ message-body-data ]
 Response     = Status-Line
                *((general-header
                /  response-header
                /  message-body-header) CRLF)
                CRLF
                [ message-body-data ]
 Request-Line = Method SP Request-URI SP RTSP-Version CRLF
 Status-Line  = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
 Method  =  "DESCRIBE"
         /  "GET_PARAMETER"
         /  "OPTIONS"
         /  "PAUSE"
         /  "PLAY"
         /  "PLAY_NOTIFY"
         /  "REDIRECT"
         /  "SETUP"
         /  "SET_PARAMETER"
         /  "TEARDOWN"
         /  extension-method
 extension-method  =  token
 Request-URI  =  "*" / RTSP-REQ-URI
 RTSP-Version =  "RTSP/" 1*DIGIT "." 1*DIGIT
 message-body-data = 1*OCTET
 Status-Code  =  "100"  ; Continue
              /  "200"  ; OK
              /  "301"  ; Moved Permanently
              /  "302"  ; Found
              /  "303"  ; See Other
              /  "304"  ; Not Modified
              /  "305"  ; Use Proxy

Schulzrinne, et al. Standards Track [Page 198] RFC 7826 RTSP 2.0 December 2016

              /  "400"  ; Bad Request
              /  "401"  ; Unauthorized
              /  "402"  ; Payment Required
              /  "403"  ; Forbidden
              /  "404"  ; Not Found
              /  "405"  ; Method Not Allowed
              /  "406"  ; Not Acceptable
              /  "407"  ; Proxy Authentication Required
              /  "408"  ; Request Timeout
              /  "410"  ; Gone
              /  "412"  ; Precondition Failed
              /  "413"  ; Request Message Body Too Large
              /  "414"  ; Request-URI Too Long
              /  "415"  ; Unsupported Media Type
              /  "451"  ; Parameter Not Understood
              /  "452"  ; reserved
              /  "453"  ; Not Enough Bandwidth
              /  "454"  ; Session Not Found
              /  "455"  ; Method Not Valid In This State
              /  "456"  ; Header Field Not Valid for Resource
              /  "457"  ; Invalid Range
              /  "458"  ; Parameter Is Read-Only
              /  "459"  ; Aggregate Operation Not Allowed
              /  "460"  ; Only Aggregate Operation Allowed
              /  "461"  ; Unsupported Transport
              /  "462"  ; Destination Unreachable
              /  "463"  ; Destination Prohibited
              /  "464"  ; Data Transport Not Ready Yet
              /  "465"  ; Notification Reason Unknown
              /  "466"  ; Key Management Error
              /  "470"  ; Connection Authorization Required
              /  "471"  ; Connection Credentials Not Accepted
              /  "472"  ; Failure to Establish Secure Connection
              /  "500"  ; Internal Server Error
              /  "501"  ; Not Implemented
              /  "502"  ; Bad Gateway
              /  "503"  ; Service Unavailable
              /  "504"  ; Gateway Timeout
              /  "505"  ; RTSP Version Not Supported
              /  "551"  ; Option Not Supported
              /  "553"  ; Proxy Unavailable
              /  extension-code
 extension-code  =  3DIGIT
 Reason-Phrase   =  1*(TEXT-UTF8char / HT / SP)

Schulzrinne, et al. Standards Track [Page 199] RFC 7826 RTSP 2.0 December 2016

 rtsp-header     = general-header
                 / request-header
                 / response-header
                 / message-body-header
 general-header  =  Accept-Ranges
                 /  Cache-Control
                 /  Connection
                 /  CSeq
                 /  Date
                 /  Media-Properties
                 /  Media-Range
                 /  Pipelined-Requests
                 /  Proxy-Supported
                 /  Range
                 /  RTP-Info
                 /  Scale
                 /  Seek-Style
                 /  Server
                 /  Session
                 /  Speed
                 /  Supported
                 /  Timestamp
                 /  Transport
                 /  User-Agent
                 /  Via
                 /  extension-header
 request-header  =  Accept
                 /  Accept-Credentials
                 /  Accept-Encoding
                 /  Accept-Language
                 /  Authorization
                 /  Bandwidth
                 /  Blocksize
                 /  From
                 /  If-Match
                 /  If-Modified-Since
                 /  If-None-Match
                 /  Notify-Reason
                 /  Proxy-Authorization
                 /  Proxy-Require
                 /  Referrer
                 /  Request-Status
                 /  Require
                 /  Terminate-Reason
                 /  extension-header

Schulzrinne, et al. Standards Track [Page 200] RFC 7826 RTSP 2.0 December 2016

 response-header  =  Authentication-Info
                  /  Connection-Credentials
                  /  Location
                  /  MTag
                  /  Proxy-Authenticate
                  /  Proxy-Authentication-Info
                  /  Public
                  /  Retry-After
                  /  Unsupported
                  /  WWW-Authenticate
                  /  extension-header
 message-body-header    =  Allow
                  /  Content-Base
                  /  Content-Encoding
                  /  Content-Language
                  /  Content-Length
                  /  Content-Location
                  /  Content-Type
                  /  Expires
                  /  Last-Modified
                  /  extension-header

20.2.3. Header Syntax

 Accept            =  "Accept" HCOLON
                      [ accept-range *(COMMA accept-range) ]
 accept-range      =  media-type-range [SEMI accept-params]
 media-type-range  =  ( "*/*"
                      / ( m-type SLASH "*" )
                      / ( m-type SLASH m-subtype )
                     ) *( SEMI m-parameter )
 accept-params     =  "q" EQUAL qvalue *(SEMI generic-param )
 qvalue            =  ( "0" [ "." *3DIGIT ] )
                   /  ( "1" [ "." *3("0") ] )
 Accept-Credentials =  "Accept-Credentials" HCOLON cred-decision
 cred-decision     =  ("User" [LWS cred-info])
                   /  "Proxy"
                   /  "Any"
                   /  (token [LWS 1*header-value])
                                   ; For future extensions
 cred-info         =  cred-info-data *(COMMA cred-info-data)
 cred-info-data    =  DQUOTE RTSP-REQ-URI DQUOTE SEMI hash-alg
                      SEMI base64
 hash-alg          =  "sha-256" / extension-alg
 extension-alg     =  token
 Accept-Encoding   =  "Accept-Encoding" HCOLON

Schulzrinne, et al. Standards Track [Page 201] RFC 7826 RTSP 2.0 December 2016

                      [ encoding *(COMMA encoding) ]
 encoding          =  codings [SEMI accept-params]
 codings           =  content-coding / "*"
 content-coding    =  "identity" / token
 Accept-Language   =  "Accept-Language" HCOLON
                      language *(COMMA language)
 language          =  language-range [SEMI accept-params]
 language-range    =  language-tag / "*"
 language-tag      =  primary-tag *( "-" subtag )
 primary-tag       =  1*8ALPHA
 subtag            =  1*8ALPHA
 Accept-Ranges     =  "Accept-Ranges" HCOLON acceptable-ranges
 acceptable-ranges =  (range-unit *(COMMA range-unit))
 range-unit        =  "npt" / "smpte" / "smpte-30-drop" / "smpte-25"
                      / "clock" / extension-format
 extension-format  =  token
 Allow             =  "Allow" HCOLON Method *(COMMA Method)
 Authentication-Info = "Authentication-Info" HCOLON auth-param-list
 auth-param-list   =  <As the Authentication-Info element in RFC 7615>
 Authorization     =  "Authorization" HCOLON credentials
 credentials       =  <As defined by RFC 7235>
 Bandwidth         =  "Bandwidth" HCOLON 1*19DIGIT
 Blocksize         =  "Blocksize" HCOLON 1*9DIGIT
 Cache-Control     =  "Cache-Control" HCOLON cache-directive
                      *(COMMA cache-directive)
 cache-directive   =  cache-rqst-directive
                   /  cache-rspns-directive
 cache-rqst-directive =  "no-cache"
                      /  "max-stale" [EQUAL delta-seconds]
                      /  "min-fresh" EQUAL delta-seconds
                      /  "only-if-cached"
                      /  cache-extension
 cache-rspns-directive =  "public"
                          /  "private"
                          /  "no-cache"
                          /  "no-transform"
                          /  "must-revalidate"
                          /  "proxy-revalidate"
                          /  "max-age" EQUAL delta-seconds
                          /  cache-extension
 cache-extension   =  token [EQUAL (token / quoted-string)]
 delta-seconds     =  1*19DIGIT

Schulzrinne, et al. Standards Track [Page 202] RFC 7826 RTSP 2.0 December 2016

 Connection         =  "Connection" HCOLON connection-token
                       *(COMMA connection-token)
 connection-token   =  "close" / token
 Connection-Credentials = "Connection-Credentials" HCOLON cred-chain
 cred-chain         =  DQUOTE RTSP-REQ-URI DQUOTE SEMI base64
 Content-Base       =  "Content-Base" HCOLON RTSP-URI
 Content-Encoding   =  "Content-Encoding" HCOLON
                       content-coding *(COMMA content-coding)
 Content-Language   =  "Content-Language" HCOLON
                       language-tag *(COMMA language-tag)
 Content-Length     =  "Content-Length" HCOLON 1*19DIGIT
 Content-Location   =  "Content-Location" HCOLON RTSP-REQ-Ref
 Content-Type       =  "Content-Type" HCOLON media-type
 media-type         =  m-type SLASH m-subtype *(SEMI m-parameter)
 m-type             =  discrete-type / composite-type
 discrete-type      =  "text" / "image" / "audio" / "video"
                    /  "application" / extension-token
 composite-type   =  "message" / "multipart" / extension-token
 extension-token  =  ietf-token / x-token
 ietf-token       =  token
 x-token          =  "x-" token
 m-subtype        =  extension-token / iana-token
 iana-token       =  token
 m-parameter      =  m-attribute EQUAL m-value
 m-attribute      =  token
 m-value          =  token / quoted-string
 CSeq           =  "CSeq" HCOLON cseq-nr
 cseq-nr        =  1*9DIGIT
 Date           =  "Date" HCOLON RTSP-date
 RTSP-date      =  date-time ;
 date-time      =  <As defined in RFC 5322>
 Expires        =  "Expires" HCOLON RTSP-date
 From           =  "From" HCOLON from-spec
 from-spec      =  ( name-addr / addr-spec ) *( SEMI from-param )
 name-addr      =  [ display-name ] LAQUOT addr-spec RAQUOT
 addr-spec      =  RTSP-REQ-URI / absolute-URI
 absolute-URI   =  < As defined in RFC 3986>
 display-name   =  *(token LWS) / quoted-string
 from-param     =  tag-param / generic-param
 tag-param      =  "tag" EQUAL token
 If-Match       =  "If-Match" HCOLON ("*" / message-tag-list)
 message-tag-list =  message-tag *(COMMA message-tag)
 message-tag      =  [ weak ] opaque-tag
 weak             =  "W/"
 opaque-tag       =  quoted-string

Schulzrinne, et al. Standards Track [Page 203] RFC 7826 RTSP 2.0 December 2016

 If-Modified-Since  =  "If-Modified-Since" HCOLON RTSP-date
 If-None-Match    =  "If-None-Match" HCOLON ("*" / message-tag-list)
 Last-Modified    =  "Last-Modified" HCOLON RTSP-date
 Location         =  "Location" HCOLON RTSP-REQ-URI
 Media-Properties = "Media-Properties" HCOLON [media-prop-list]
 media-prop-list  = media-prop-value *(COMMA media-prop-value)
 media-prop-value = ("Random-Access" [EQUAL POS-FLOAT])
                  / "Beginning-Only"
                  / "No-Seeking"
                  / "Immutable"
                  / "Dynamic"
                  / "Time-Progressing"
                  / "Unlimited"
                  / ("Time-Limited" EQUAL utc-time)
                  / ("Time-Duration" EQUAL POS-FLOAT)
                  / ("Scales" EQUAL scale-value-list)
                  / media-prop-ext
 media-prop-ext   = token [EQUAL (1*rtsp-unreserved / quoted-string)]
 scale-value-list = DQUOTE scale-entry *(COMMA scale-entry) DQUOTE
 scale-entry      = scale-value / (scale-value COLON scale-value)
 scale-value      = FLOAT
 Media-Range      = "Media-Range" HCOLON [ranges-list]
 ranges-list      =  ranges-spec *(COMMA ranges-spec)
 MTag             =  "MTag" HCOLON message-tag
 Notify-Reason    = "Notify-Reason" HCOLON Notify-Reas-val
 Notify-Reas-val  = "end-of-stream"
                  / "media-properties-update"
                  / "scale-change"
                  / Notify-Reason-extension
 Notify-Reason-extension  = token
 Pipelined-Requests = "Pipelined-Requests" HCOLON startup-id
 startup-id  = 1*8DIGIT
 Proxy-Authenticate =  "Proxy-Authenticate" HCOLON challenge-list
 challenge-list     = <As defined by the WWW-Authenticate in RFC 7235>
 Proxy-Authentication-Info = "Proxy-Authentication-Info" HCOLON
                       auth-param-list
 Proxy-Authorization = "Proxy-Authorization" HCOLON credentials
 Proxy-Require      =  "Proxy-Require" HCOLON feature-tag-list
 feature-tag-list   =  feature-tag *(COMMA feature-tag)
 Proxy-Supported    =  "Proxy-Supported" HCOLON [feature-tag-list]
 Public             =  "Public" HCOLON Method *(COMMA Method)
 Range              =  "Range" HCOLON ranges-spec
 ranges-spec        =  npt-range / utc-range / smpte-range
                    /  range-ext

Schulzrinne, et al. Standards Track [Page 204] RFC 7826 RTSP 2.0 December 2016

 range-ext          =  extension-format [EQUAL range-value]
 range-value        =  1*(rtsp-unreserved / quoted-string / ":" )
 Referrer           =  "Referrer" HCOLON (absolute-URI / RTSP-URI-Ref)
 Request-Status     =  "Request-Status" HCOLON req-status-info
 req-status-info    =  cseq-info LWS status-info LWS reason-info
 cseq-info          =  "cseq" EQUAL cseq-nr
 status-info        =  "status" EQUAL Status-Code
 reason-info        =  "reason" EQUAL DQUOTE Reason-Phrase DQUOTE
 Require            =  "Require" HCOLON feature-tag-list

Schulzrinne, et al. Standards Track [Page 205] RFC 7826 RTSP 2.0 December 2016

 RTP-Info         =  "RTP-Info" HCOLON [rtsp-info-spec
                     *(COMMA rtsp-info-spec)]
 rtsp-info-spec   =  stream-url 1*ssrc-parameter
 stream-url       =  "url" EQUAL DQUOTE RTSP-REQ-Ref DQUOTE
 ssrc-parameter   =  LWS "ssrc" EQUAL ssrc HCOLON
                     ri-parameter *(SEMI ri-parameter)
 ri-parameter     =  ("seq" EQUAL 1*5DIGIT)
                  /  ("rtptime" EQUAL 1*10DIGIT)
                  /  generic-param
 Retry-After      =  "Retry-After" HCOLON (RTSP-date / delta-seconds)
 Scale            =  "Scale" HCOLON scale-value
 Seek-Style       =  "Seek-Style" HCOLON Seek-S-values
 Seek-S-values    =  "RAP"
                  /  "CoRAP"
                  /  "First-Prior"
                  /  "Next"
                  /  Seek-S-value-ext
 Seek-S-value-ext =  token
 Server           =  "Server" HCOLON ( product / comment )
                     *(LWS (product / comment))
 product          =  token [SLASH product-version]
 product-version  =  token
 comment          =  LPAREN *( ctext / quoted-pair) RPAREN
 Session          =  "Session" HCOLON session-id
                     [ SEMI "timeout" EQUAL delta-seconds ]
 Speed            =  "Speed" HCOLON lower-bound MINUS upper-bound
 lower-bound      =  POS-FLOAT
 upper-bound      =  POS-FLOAT
 Supported        =  "Supported" HCOLON [feature-tag-list]

Schulzrinne, et al. Standards Track [Page 206] RFC 7826 RTSP 2.0 December 2016

 Terminate-Reason      =  "Terminate-Reason" HCOLON TR-Info
 TR-Info              =  TR-Reason *(SEMI TR-Parameter)
 TR-Reason            =  "Session-Timeout"
                      /  "Server-Admin"
                      /  "Internal-Error"
                      /  token
 TR-Parameter         =  TR-time / TR-user-msg / generic-param
 TR-time              =  "time" EQUAL utc-time
 TR-user-msg          =  "user-msg" EQUAL quoted-string
 Timestamp        =  "Timestamp" HCOLON timestamp-value [LWS delay]
 timestamp-value  =  *19DIGIT [ "." *9DIGIT ]
 delay            =  *9DIGIT [ "." *9DIGIT ]
 Transport        =  "Transport" HCOLON transport-spec
                     *(COMMA transport-spec)
 transport-spec   =  transport-id *trns-parameter
 transport-id     =  trans-id-rtp / other-trans
 trans-id-rtp     =  "RTP/" profile ["/" lower-transport]
                     ; no LWS is allowed inside transport-id
 other-trans      =  token *("/" token)

Schulzrinne, et al. Standards Track [Page 207] RFC 7826 RTSP 2.0 December 2016

 profile           = "AVP" / "SAVP" / "AVPF" / "SAVPF" / token
 lower-transport   = "TCP" / "UDP" / token
 trns-parameter    = (SEMI ( "unicast" / "multicast" ))
                   / (SEMI "interleaved" EQUAL channel ["-" channel])
                   / (SEMI "ttl" EQUAL ttl)
                   / (SEMI "layers" EQUAL 1*DIGIT)
                   / (SEMI "ssrc" EQUAL ssrc *(SLASH ssrc))
                   / (SEMI "mode" EQUAL mode-spec)
                   / (SEMI "dest_addr" EQUAL addr-list)
                   / (SEMI "src_addr" EQUAL addr-list)
                   / (SEMI "setup" EQUAL contrans-setup)
                   / (SEMI "connection" EQUAL contrans-con)
                   / (SEMI "RTCP-mux")
                   / (SEMI "MIKEY" EQUAL MIKEY-Value)
                   / (SEMI trn-param-ext)
 contrans-setup    = "active" / "passive" / "actpass"
 contrans-con      = "new" / "existing"
 trn-param-ext     = par-name [EQUAL trn-par-value]
 par-name          = token
 trn-par-value     = *(rtsp-unreserved / quoted-string)
 ttl               = 1*3DIGIT ; 0 to 255
 ssrc              = 8HEX
 channel           = 1*3DIGIT ; 0 to 255
 MIKEY-Value       = base64
 mode-spec         = ( DQUOTE mode *(COMMA mode) DQUOTE )
 mode              = "PLAY" / token
 addr-list         = quoted-addr *(SLASH quoted-addr)
 quoted-addr       = DQUOTE (host-port / extension-addr) DQUOTE
 host-port         = ( host [":" port] )
                   / ( ":" port )
 extension-addr    = 1*qdtext
 host              = < As defined in RFC 3986>
 port              = < As defined in RFC 3986>

Schulzrinne, et al. Standards Track [Page 208] RFC 7826 RTSP 2.0 December 2016

 Unsupported     = "Unsupported" HCOLON feature-tag-list
 User-Agent      = "User-Agent" HCOLON ( product / comment )
                   *(LWS (product / comment))
 Via             = "Via" HCOLON via-parm *(COMMA via-parm)
 via-parm        = sent-protocol LWS sent-by *( SEMI via-params )
 via-params      = via-ttl / via-maddr
                 / via-received / via-extension
 via-ttl         = "ttl" EQUAL ttl
 via-maddr       = "maddr" EQUAL host
 via-received    = "received" EQUAL (IPv4address / IPv6address)
 IPv4address     = < As defined in RFC 3986>
 IPv6address     = < As defined in RFC 3986>
 via-extension   = generic-param
 sent-protocol   = protocol-name SLASH protocol-version
                   SLASH transport-prot
 protocol-name   = "RTSP" / token
 protocol-version = token
 transport-prot  = "UDP" / "TCP" / "TLS" / other-transport
 other-transport = token
 sent-by         = host [ COLON port ]
 WWW-Authenticate = "WWW-Authenticate" HCOLON challenge-list

20.3. SDP Extension Syntax

 This section defines in ABNF the SDP extensions defined for RTSP.
 See Appendix D for the definition of the extensions in text.
 control-attribute   =  "a=control:" *SP RTSP-REQ-Ref CRLF
 a-range-def         =  "a=range:" ranges-spec CRLF
 a-mtag-def          =  "a=mtag:" message-tag CRLF

21. Security Considerations

 The security considerations and threats around RTSP and its usage can
 be divided into considerations around the signaling protocol itself
 and the issues related to the media-stream delivery.  However, when
 it comes to mitigation of security threats, a threat depending on the
 media-stream delivery may in fact be mitigated by a mechanism in the
 signaling protocol.

Schulzrinne, et al. Standards Track [Page 209] RFC 7826 RTSP 2.0 December 2016

 There are several chapters and an appendix in this document that
 define security solutions for the protocol.  These sections will be
 referenced when discussing the threats below.  However, the reader
 should take special notice of the Security Framework (Section 19) and
 the specification of how to use SRTP and its key-management
 (Appendix C.1.4) to achieve certain aspects of the media security.

21.1. Signaling Protocol Threats

 This section focuses on issues related to the signaling protocol.
 Because of the similarity in syntax and usage between RTSP servers
 and HTTP servers, the security considerations outlined in [RFC7230],
 [RFC7231], [RFC7232], [RFC7233], [RFC7234], and [RFC7235] apply as
 well.
 Specifically, please note the following:
 Abuse of Server Log Information:  A server is in the position to save
       personal data about a user's requests that might identify their
       media consumption patterns or subjects of interest.  This
       information is clearly confidential in nature, and its handling
       can be constrained by law in certain countries.  Log
       information needs to be securely stored and appropriate
       guidelines followed for its analysis.  See Section 9.8 of
       [RFC7230] for additional guidelines.
 Transfer of Sensitive Information:  There is no reason to believe
       that information transferred in RTSP message, such as the URI
       and the content of headers, especially the Server, Via,
       Referrer, and From headers, may be any less sensitive than when
       used in HTTP.  Therefore, all of the precautions regarding the
       protection of data privacy and user privacy apply to
       implementers of RTSP clients, servers, and proxies.  See
       Sections 9.3-9.6 of [RFC7231] for further details.
       The RTSP methods defined in this document are primarily used to
       establish and control the delivery of the media data
       represented by the URI; thus, the RTSP message bodies are
       generally less sensitive than the ones in HTTP.  Where HTTP
       bodies could contain, for example, your medical records, in
       RTSP, the sensitive video of your medical operation would be in
       the media stream over the media-transport protocol, not in the
       RTSP message.  Still, one has to take note of what potential
       sensitive information is included in RTSP.  The protection of
       the media data is separate, can be applied directly between
       client and server, and is dependent on the media-transport
       protocol in use.  See Section 21.2 for further discussion.
       This possibility for separation of security between media-

Schulzrinne, et al. Standards Track [Page 210] RFC 7826 RTSP 2.0 December 2016

       resource content and the signaling protocol mitigates the risk
       of exposing the media content when using hop-by-hop security
       for RTSP signaling using proxies (Section 19.3).
 Attacks Based On File and Path Names:  Though RTSP URIs are opaque
       handles that do not necessarily have file-system semantics, it
       is anticipated that many implementations will translate
       portions of the Request-URIs directly to file-system calls.  In
       such cases, file systems SHOULD follow the precautions outlined
       in Section 9.1 of [RFC7231], such as checking for ".." in path
       components.
 Personal Information:  RTSP clients are often privy to the same
       information that HTTP clients are (username, location, etc.)
       and thus should be equally sensitive.  See Section 9.8 of
       [RFC7230], Sections 9.3-9.7 of [RFC7231], and Section 8 of
       [RFC7234] for further recommendations.
 Privacy Issues Connected to Accept Headers:  Since similar usages of
       the "Accept" headers exist in RTSP as in HTTP, the same caveats
       outlined in Section 9.4 of [RFC7231] with regard to their use
       should be followed.
 Establishing Authority:  RTSP shares with HTTP the question of how a
       client communicates with the authoritative source for media
       streams (Section 9.1 of [RFC7230]).  The used DNS servers, the
       security of the communication, and any possibility of a man in
       the middle, and the trust in any RTSP proxies all affect the
       possibility that a client has received a non-authoritative
       response to a request.  Ensuring that a client receives an
       authoritative response is challenging, although using the
       secure communication for RTSP signaling (rtsps) simplifies it
       significantly as the server can provide a hostname identity
       assertion in the TLS handshake.
 Location Headers and Spoofing:  If a single server supports multiple
       organizations that do not trust each another, then it MUST
       check the values of the Content-Location header fields in
       responses that are generated under control of said
       organizations to make sure that they do not attempt to
       invalidate resources over which they have no authority (see
       Section 15.4 of [RFC2616]).
 In addition to the recommendations in the current HTTP specifications
 ([RFC7230], [RFC7231], [RFC7232], [RFC7233], [RFC7234], and [RFC7235]
 as of this writing) and also those of the previous relevant RFCs
 [RFC2068] [RFC2616], future HTTP specifications may provide
 additional guidance on security issues.

Schulzrinne, et al. Standards Track [Page 211] RFC 7826 RTSP 2.0 December 2016

 The following are added considerations for RTSP implementations.
 Session Hijacking:  Since there is no or little relation between a
       transport-layer connection and an RTSP session, it is possible
       for a malicious client to issue requests with random session
       identifiers that could affect other clients of an unsuspecting
       server.  To mitigate this, the server SHALL use a large, random
       and non-sequential session identifier to minimize the
       possibility of this kind of attack.  However, unless the RTSP
       signaling is always confidentiality protected, e.g., using TLS,
       an on-path attacker will be able to hijack a session.  Another
       choice for preventing session hijacking is to use client
       authentication and only allow the authenticated client creating
       the session to access that session.
 Authentication:  Servers SHOULD implement both basic and Digest
       [RFC2617] authentication.  In environments requiring tighter
       security for the control messages, the transport-layer
       mechanism TLS [RFC5246] SHOULD be used.
 Suspicious Behavior:  Upon detecting instances of behavior that is
       deemed a security risk, RTSP servers SHOULD return error code
       403 (Forbidden).  RTSP servers SHOULD also be aware of attempts
       to probe the server for weaknesses and entry points and MAY
       arbitrarily disconnect and ignore further requests from clients
       that are deemed to be in violation of local security policy.
 TLS through Proxies:  If one uses the possibility to connect TLS in
       multiple legs (Section 19.3), one really needs to be aware of
       the trust model.  This procedure requires trust in all proxies
       part of the path to the server.  The proxies one connects
       through are identified, assuming the proxies so far connected
       through are well behaved and fulfilling the trust.  The
       accepted proxies are men in the middle and have access to all
       that goes on over the TLS connection.  Thus, it is important to
       consider if that trust model is acceptable in the actual
       application.  Further discussion of the actual trust model is
       in Section 19.3.  It is important to note what difference in
       security properties, if any, may exist with the used media-
       transport protocol and its security mechanism.  Using SRTP and
       the MIKEY-based key-establishment defined in Appendix C.1.4.1
       enables media key-establishment to be done end-to-end without
       revealing the keys to the proxies.

Schulzrinne, et al. Standards Track [Page 212] RFC 7826 RTSP 2.0 December 2016

 Resource Exhaustion:  As RTSP is a stateful protocol and establishes
       resource usage on the server, there is a clear possibility to
       attack the server by trying to overbook these resources to
       perform a DoS attack.  This attack can be both against ongoing
       sessions and to prevent others from establishing sessions.
       RTSP agents will need to have mechanisms to prevent single
       peers from consuming extensive amounts of resources.  The
       methods for guarding against this are varied and depend on the
       agent's role and capabilities and policies.  Each
       implementation has to carefully consider its methods and
       policies to mitigate this threat.  There are recommendations
       regarding the handling of connections in Section 10.7.
 The above threats and considerations have resulted in a set of
 security functions and mechanisms built into or used by the protocol.
 The signaling protocol relies on two security features defined in the
 Security Framework (Section 19): namely client authentication using
 HTTP authentication and TLS-based transport protection of the
 signaling messages.  Both of these mechanisms are required to be
 implemented by any RTSP agent.
 A number of different security mitigations have been designed into
 the protocol and will be instantiated if the specification is
 implemented as written, for example, by ensuring sufficient amounts
 of entropy in the randomly generated session identifiers when not
 using client authentication to minimize the risk of session
 hijacking.  When client authentication is used, protection against
 hijacking will be greatly improved by scoping the accessible sessions
 to the one this client identity has created.  Some of the above
 threats are such that the implementation of the RTSP functionality
 itself needs to consider which policy and strategy it uses to
 mitigate them.

21.2. Media Stream Delivery Threats

 The fact that RTSP establishes and controls a media-stream delivery
 results in a set of security issues related to the media streams.
 This section will attempt to analyze general threats; however, the
 choice of media-stream transport protocol, such as RTP, will result
 in some differences in threats and what mechanisms exist to mitigate
 them.  Thus, it becomes important that each specification of a new
 media-stream transport and delivery protocol usable by RTSP requires
 its own security analysis.  This section includes one for RTP.

Schulzrinne, et al. Standards Track [Page 213] RFC 7826 RTSP 2.0 December 2016

 The set of general threats from or by the media-stream delivery
 itself are:
 Concentrated Denial-of-Service Attack:  The protocol offers the
    opportunity for a remote-controlled DoS attack, where the media
    stream is the hammer in that DoS attack.  See Section 21.2.1.
 Media Confidentiality:  The media delivery may contain content of any
    type, and it is not possible, in general, to determine how
    sensitive this content is from a confidentiality point.  Thus, it
    is a strong requirement that any media delivery protocol supply a
    method for providing confidentiality of the actual media content.
    In addition to the media-level confidentiality, it becomes
    critical that no resource identifiers used in the signaling be
    exposed to an attacker as they may have human-understandable names
    or may be available to the attacker, allowing it to determine the
    content the user received.  Thus, the signaling protocol must also
    provide confidentiality protection of any information related to
    the media resource.
 Media Integrity and Authentication:  There are several reasons why an
    attacker will be interested in substituting the media stream sent
    out from the RTSP server with one of the attacker's creation or
    selection, such as discrediting the target and misinformation
    about the target.  Therefore, it is important that the media
    protocol provide mechanisms to verify the source authentication
    and integrity and to prevent replay attacks on the media stream.
 Scope of Multicast:  If RTSP is used to control the transmission of
    media onto a multicast network, the scope of the delivery must be
    considered.  RTSP supports the TTL Transport header parameter to
    indicate this scope for IPv4.  IPv6 has a different mechanism for
    the scope boundary.  However, such scope control has risks, as it
    may be set too large and distribute media beyond the intended
    scope.
 Below (Section 21.2.2) a protocol-specific analysis of security
 considerations for RTP-based media transport is included.  In that
 section, the requirements on implementing security functions for RTSP
 agents supporting media delivery over RTP are made clear.

Schulzrinne, et al. Standards Track [Page 214] RFC 7826 RTSP 2.0 December 2016

21.2.1. Remote DoS Attack

 The attacker may initiate traffic flows to one or more IP addresses
 by specifying them as the destination in SETUP requests.  While the
 attacker's IP address may be known in this case, this is not always
 useful in the prevention of more attacks or ascertaining the
 attacker's identity.  Thus, an RTSP server MUST only allow client-
 specified destinations for RTSP-initiated traffic flows if the server
 has ensured that the specified destination address accepts receiving
 media through different security mechanisms.  Security mechanisms
 that are acceptable in order of increasing generality are:
 o  Verification of the client's identity against a database of known
    users using RTSP authentication mechanisms (preferably Digest
    authentication or stronger)
 o  A list of addresses that have consented to be media destinations,
    especially considering user identity
 o  Verification based on media path
 The server SHOULD NOT allow the destination field to be set unless a
 mechanism exists in the system to authorize the request originator to
 direct streams to the recipient.  It is preferred that this
 authorization be performed by the media recipient (destination)
 itself and the credentials be passed along to the server.  However,
 in certain cases, such as when the recipient address is a multicast
 group or when the recipient is unable to communicate with the server
 in an out-of-band manner, this may not be possible.  In these cases,
 the server may choose another method such as a server-resident
 authorization list to ensure that the request originator has the
 proper credentials to request stream delivery to the recipient.
 One solution that performs the necessary verification of acceptance
 of media suitable for unicast-based delivery is the NAT traversal
 method based on Interactive Connectivity Establishment (ICE)
 [RFC5245] described in [RFC7825].  This mechanism uses random
 passwords and a username so that the probability of unintended
 indication as a valid media destination is very low.  In addition,
 the server includes in its Session Traversal Utilities for NAT (STUN)
 [RFC5389] requests a cookie (consisting of random material) that the
 destination echoes back; thus, the solution also safeguards against
 having an off-path attacker being able to spoof the STUN checks.
 This leaves this solution vulnerable only to on-path attackers that
 can see the STUN requests go to the target of attack and thus forge a
 response.

Schulzrinne, et al. Standards Track [Page 215] RFC 7826 RTSP 2.0 December 2016

 For delivery to multicast addresses, there is a need for another
 solution that is not specified in this memo.

21.2.2. RTP Security Analysis

 RTP is a commonly used media-transport protocol and has been the most
 common choice for RTSP 1.0 implementations.  The core RTP protocol
 has been in use for a long time, and it has well-known security
 properties and the RTP security consideration (Section 9 of
 [RFC3550]) needs to be reviewed.  In perspective of the usage of RTP
 in the context of RTSP, the following properties should be noted:
 Stream Additions:  RTP has support for multiple simultaneous media
    streams in each RTP session.  As some use cases require support
    for non-synchronized adding and removal of media streams and their
    identifiers, an attacker can easily insert additional media
    streams into a session context that, according to protocol design,
    is intended to be played out.  Another threat vector is one of DoS
    by exhausting the resources of the RTP session receiver, for
    example, by using a large number of SSRC identifiers
    simultaneously.  The strong mitigation of this is to ensure that
    one cryptographically authenticates any incoming packet flow to
    the RTP session.  Weak mitigations like blocking additional media
    streams in session contexts easily lead to a DoS vulnerability in
    addition to preventing certain RTP extensions or use cases that
    rely on multiple media streams, such as RTP retransmission
    [RFC4588] to function.
 Forged Feedback:  The built-in RTCP also offers a large attack
    surface for a couple of different types of attacks.  One venue is
    to send RTCP feedback to the media sender indicating large amounts
    of packet loss and thus trigger a media bitrate adaptation
    response from the sender resulting in lowered media quality and
    potentially a shutdown of the media stream.  Another attack is to
    perform a resource-exhaustion attack on the receiver by using many
    SSRC identifiers to create large state tables and increase the
    RTCP-related processing demands.
 RTP/RTCP Extensions:  RTP and RTCP extensions generally provide
    additional and sometimes extremely powerful tools for DoS attacks
    or service disruption.  For example, the Code Control Message
    [RFC5104] RTCP extensions enables both the lock down of the
    bitrate to low values and disruption of video quality by
    requesting intra-frames.
 Taking into account the above general discussion in Section 21.2 and
 the RTP-specific discussion in this section, it is clear that it is
 necessary that a strong security mechanism be supported to protect

Schulzrinne, et al. Standards Track [Page 216] RFC 7826 RTSP 2.0 December 2016

 RTP.  Therefore, this specification has the following requirements on
 RTP security functions for all RTSP agents that handle media streams
 and where media-stream transport is completed using RTP.
 RTSP agents supporting RTP MUST implement Secure RTP (SRTP) [RFC3711]
 and, thus, SAVP.  In addition, SAVPF [RFC5124] MUST also be supported
 if AVPF is implemented.  This specification requires no additional
 cryptographic transforms or configuration values beyond those
 specified as mandatory to implement in RFC 3711, i.e., AES-CM and
 HMAC-SHA1.  The default key-management mechanism that MUST be
 implemented is the one defined in MIKEY Key Establishment
 (Appendix C.1.4.1).  The MIKEY implementation MUST implement the
 necessary functions for MIKEY-RSA-R mode [RFC4738] and the SRTP
 parameter negotiation necessary to negotiate the supported SRTP
 transforms and parameters.

22. IANA Considerations

 This section describes a number of registries for RTSP 2.0 that have
 been established and are maintained by IANA.  These registries are
 separate from any registries existing for RTSP 1.0.  For each
 registry, there is a description of the required content, the
 registration procedures, and the entries that this document
 registers.  For more information on extending RTSP, see Section 2.7.
 In addition, this document registers three SDP attributes.
 Registries or entries in registries that have been made for RTSP 1.0
 are not moved to RTSP 2.0: the registries and entries of RTSP 1.0 and
 RTSP 2.0 are independent.  If any registry or entry in a registry is
 also required in RTSP 2.0, it MUST follow the procedure defined below
 to allocate the registry or entry in a registry.
 The sections describing how to register an item use some of the
 registration policies described in [RFC5226] -- namely, "First Come
 First Served", "Expert Review", "Specification Required", and
 "Standards Action".
 In case a registry requires a contact person, the authors (with
 Magnus Westerlund <magnus.westerlund@ericsson.com> as primary) are
 the contact persons for any entries created by this document.
 IANA will request the following information for any registration
 request:
 o  A name of the item to register according to the rules specified by
    the intended registry

Schulzrinne, et al. Standards Track [Page 217] RFC 7826 RTSP 2.0 December 2016

 o  Indication of who has change control over the feature (for
    example, the IETF, ISO, ITU-T, other international standardization
    bodies, a consortium, a particular company or group of companies,
    or an individual)
 o  A reference to a further description, if available, for example
    (in decreasing order of preference), an RFC, a published standard,
    a published paper, a patent filing, a technical report, documented
    source code or a computer manual
 o  For proprietary features, contact information (postal and email
    address)

22.1. Feature Tags

22.1.1. Description

 When a client and server try to determine what part and functionality
 of the RTSP specification and any future extensions that its
 counterpart implements, there is need for a namespace.  This registry
 contains named entries representing certain functionality.
 The usage of feature tags is explained in Section 11 and
 Section 13.1.

22.1.2. Registering New Feature Tags with IANA

 The registering of feature tags is done on a First Come, First Served
 [RFC5226] basis.
 The registry entry for a feature tag has the following information:
 o  The name of the feature tag
  • If the registrant indicates that the feature is proprietary,

IANA should request a vendor "prefix" portion of the name. The

       name will then be the vendor prefix followed by a "." followed
       by the rest of the provided feature name.
  • If the feature is not proprietary, then IANA need not collect a

prefix for the name.

 o  A one-paragraph description of what the feature tag represents
 o  The applicability (server, client, proxy, or some combination)
 o  A reference to a specification, if applicable

Schulzrinne, et al. Standards Track [Page 218] RFC 7826 RTSP 2.0 December 2016

 Feature tag names (including the vendor prefix) may contain any non-
 space and non-control characters.  There is no length limit on
 feature tags.
 Examples for a vendor tag describing a proprietary feature are:
       vendorA.specfeat01
       vendorA.specfeat02

22.1.3. Registered Entries

 The following feature tags are defined in this specification and
 hereby registered.  The change control belongs to the IETF.
 play.basic:  The implementation for delivery and playback operations
       according to the core RTSP specification, as defined in this
       memo.  Applies for clients, servers, and proxies.  See
       Section 11.1.
 play.scale:  Support of scale operations for media playback.  Applies
       only for servers.  See Section 18.46.
 play.speed:  Support of the speed functionality for media delivery.
       Applies only for servers.  See Section 18.50.
 setup.rtp.rtcp.mux:  Support of the RTP and RTCP multiplexing as
       discussed in Appendix C.1.6.4.  Applies for both client and
       servers and any media caching proxy.
 The IANA registry is a table with the name, description, and
 reference for each feature tag.

22.2. RTSP Methods

22.2.1. Description

 Methods are described in Section 13.  Extending the protocol with new
 methods allows for totally new functionality.

22.2.2. Registering New Methods with IANA

 A new method is registered through a Standards Action [RFC5226]
 because new methods may radically change the protocol's behavior and
 purpose.

Schulzrinne, et al. Standards Track [Page 219] RFC 7826 RTSP 2.0 December 2016

 A specification for a new RTSP method consists of the following
 items:
 o  A method name that follows the ABNF rules for methods.
 o  A clear specification of what a request using the method does and
    what responses are expected.  In which directions the method is
    used: C->S, S->C, or both.  How the use of headers, if any,
    modifies the behavior and effect of the method.
 o  A list or table specifying which of the IANA-registered headers
    that are allowed to be used with the method in the request or/and
    response.  The list or table SHOULD follow the format of tables in
    Section 18.
 o  Describe how the method relates to network proxies.

22.2.3. Registered Entries

 This specification, RFC 7826, registers 10 methods: DESCRIBE,
 GET_PARAMETER, OPTIONS, PAUSE, PLAY, PLAY_NOTIFY, REDIRECT, SETUP,
 SET_PARAMETER, and TEARDOWN.  The initial table of the registry is
 provided below.
 Method         Directionality           Reference
 -----------------------------------------------------
 DESCRIBE       C->S                     RFC 7826
 GET_PARAMETER  C->S, S->C               RFC 7826
 OPTIONS        C->S, S->C               RFC 7826
 PAUSE          C->S                     RFC 7826
 PLAY           C->S                     RFC 7826
 PLAY_NOTIFY    S->C                     RFC 7826
 REDIRECT       S->C                     RFC 7826
 SETUP          C->S                     RFC 7826
 SET_PARAMETER  C->S, S->C               RFC 7826
 TEARDOWN       C->S, S->C               RFC 7826

22.3. RTSP Status Codes

22.3.1. Description

 A status code is the three-digit number used to convey information in
 RTSP response messages; see Section 8.  The number space is limited,
 and care should be taken not to fill the space.

Schulzrinne, et al. Standards Track [Page 220] RFC 7826 RTSP 2.0 December 2016

22.3.2. Registering New Status Codes with IANA

 A new status code registration follows the policy of IETF Review
 [RFC5226].  New RTSP functionality requiring Status Codes should
 first be registered in the range of x50-x99.  Only when the range is
 full should registrations be made in the x00-x49 range, unless it is
 to adopt an HTTP extension to RTSP.  This is done to enable any HTTP
 extension to be adopted to RTSP without needing to renumber any
 related status codes.  A specification for a new status code must
 include the following:
 o  The registered number.
 o  A description of what the status code means and the expected
    behavior of the sender and receiver of the code.

22.3.3. Registered Entries

 RFC 7826 (this document) registers the numbered status code defined
 in the ABNF entry "Status-Code", except "extension-code" (that
 defines the syntax allowed for future extensions) in Section 20.2.2.

22.4. RTSP Headers

22.4.1. Description

 By specifying new headers, one or more methods can be enhanced in
 many different ways.  An unknown header will be ignored by the
 receiving agent.  If the new header is vital for certain
 functionality, a feature tag for the functionality can be created and
 demanded to be used by the counterpart with the inclusion of a
 Require header carrying the feature tag.

22.4.2. Registering New Headers with IANA

 Registrations can be made following the Expert Review policy
 [RFC5226].  A specification is recommended to be provided, preferably
 an RFC or other specification from a Standards Developing
 Organization.  The minimal information in a registration request is
 the header name and the contact information.
 The expert reviewer verifies that the registration request contains
 the following information:
 o  The name of the header.
 o  An ABNF specification of the header syntax.

Schulzrinne, et al. Standards Track [Page 221] RFC 7826 RTSP 2.0 December 2016

 o  A list or table specifying when the header may be used,
    encompassing all methods, their request or response, and the
    direction (C->S or S->C).
 o  How the header is to be handled by proxies.
 o  A description of the purpose of the header.

22.4.3. Registered Entries

 All headers specified in Section 18 in RFC 7826 have been registered.
 The registry includes the header name and reference.
 Furthermore, the following legacy RTSP headers defined in other
 specifications are registered with header name, and reference
 according to below list.  Note: these references may not fulfill all
 of the above rules for registrations due to their legacy status.
 o  x-wap-profile defined in [TS-26234].  The x-wap-profile request-
    header contains one or more absolute URLs to the requesting
    agent's device-capability profile.
 o  x-wap-profile-diff defined in [TS-26234].  The x-wap-profile-diff
    request-header contains a subset of a device-capability profile.
 o  x-wap-profile-warning defined in [TS-26234].  The x-wap-profile-
    warning is a response-header that contains error codes explaining
    to what extent the server has been able to match the terminal
    request in regard to device-capability profiles, as described
    using x-wap-profile and x-wap-profile-diff headers.
 o  x-predecbufsize defined in [TS-26234].  This response-header
    provides an RTSP agent with the TS 26.234 Annex G hypothetical
    pre-decoder buffer size.
 o  x-initpredecbufperiod defined in [TS-26234].  This response-header
    provides an RTSP agent with the TS 26.234 Annex G hypothetical
    pre-decoder buffering period.
 o  x-initpostdecbufperiod defined in [TS-26234].  This response-
    header provides an RTSP agent with the TS 26.234 Annex G post-
    decoder buffering period.
 o  3gpp-videopostdecbufsize defined in [TS-26234].  This response-
    header provides an RTSP agent with the TS 26.234 defined post-
    decoder buffer size usable for H.264 (AVC) video streams.

Schulzrinne, et al. Standards Track [Page 222] RFC 7826 RTSP 2.0 December 2016

 o  3GPP-Link-Char defined in [TS-26234].  This request-header
    provides the RTSP server with the RTSP client's link
    characteristics as determined from the radio interface.  The
    information that can be provided are guaranteed bitrate, maximum
    bitrate and maximum transfer delay.
 o  3GPP-Adaptation defined in [TS-26234].  This general-header is
    part of the bitrate adaptation solution specified for the Packet-
    switched Streaming Service (PSS).  It provides the RTSP client's
    buffer sizes and target buffer levels to the server, and responses
    are used to acknowledge the support and values.
 o  3GPP-QoE-Metrics defined in [TS-26234].  This general-header is
    used by PSS RTSP agents to negotiate the quality of experience
    metrics that a client should gather and report to the server.
 o  3GPP-QoE-Feedback defined in [TS-26234].  This request-header is
    used by RTSP clients supporting PSS to report the actual values of
    the metrics gathered in its quality of experience metering.
 The use of "x-" is NOT RECOMMENDED, but the above headers in the list
 were defined prior to the clarification.

22.5. Accept-Credentials

 The security framework's TLS connection mechanism has two
 registerable entities.

22.5.1. Accept-Credentials Policies

 This registry is for policies for an RTSP proxy's handling and
 verification of TLS certificates when establishing an outbound TLS
 connection on behalf of a client.  In Section 19.3.1, three policies
 for how to handle certificates are specified.  Further policies may
 be defined; registration is made through Standards Action [RFC5226].
 A registration request is required to contain the following
 information:
 o  Name of the policy.
 o  Text that describes how the policy works for handling the
    certificates.
 o  A contact person.

Schulzrinne, et al. Standards Track [Page 223] RFC 7826 RTSP 2.0 December 2016

 This specification registers the following values:
 Any:  A policy requiring the proxy to accept any received
       certificate.
 Proxy:  A policy where the proxy applies its own policies to
       determine which certificates are accepted.
 User: A policy where the certificate is required to be forwarded down
       the proxy chain to the client, thus allowing the user to
       decided to accept or refuse a certificate.

22.5.2. Accept-Credentials Hash Algorithms

 The Accept-Credentials header (see Section 18.2) allows for the usage
 of other algorithms for hashing the DER records of accepted entities.
 The registration of any future algorithm is expected to be extremely
 rare and could also cause interoperability problems.  Therefore, the
 bar for registering new algorithms is intentionally placed high.
 Any registration of a new hash algorithm requires Standards Action
 [RFC5226].  The registration needs to fulfill the following
 requirement:
 o  The algorithms identifier meeting the "token" ABNF requirement.
 o  Provide a definition of the algorithm.
 The registered value is:
 Hash Alg. ID   Reference
 ------------------------
 sha-256        RFC 7826

22.6. Cache-Control Cache Directive Extensions

 There exist a number of cache directives that can be sent in the
 Cache-Control header.  A registry for these cache directives has been
 established by IANA.  New registrations in this registry require
 Standards Action or IESG Approval [RFC5226].  A registration request
 needs to contain the following information.
 o  The name of the cache directive.
 o  A definition of the parameter value, if any is allowed.
 o  The specification if it is a request or response directive.

Schulzrinne, et al. Standards Track [Page 224] RFC 7826 RTSP 2.0 December 2016

 o  Text that explains how the cache directive is used for RTSP-
    controlled media streams.
 o  A contact person.
 This specification registers the following values:
    no-cache:
    public:
    private:
    no-transform:
    only-if-cached:
    max-stale:
    min-fresh:
    must-revalidate:
    proxy-revalidate:
    max-age:
 The registry contains the name of the directive and the reference.

22.7. Media Properties

22.7.1. Description

 The media streams being controlled by RTSP can have many different
 properties.  The media properties required to cover the use cases
 that were in mind when writing the specification are defined.
 However, it can be expected that further innovation will result in
 new use cases or media streams with properties not covered by the
 ones specified here.  Thus, new media properties can be specified.
 As new media properties may need a substantial amount of new
 definitions to correctly specify behavior for this property, the bar
 is intended to be high.

Schulzrinne, et al. Standards Track [Page 225] RFC 7826 RTSP 2.0 December 2016

22.7.2. Registration Rules

 Registering a new media property is done following the Specification
 Required policy [RFC5226].  The expert reviewer verifies that a
 registration request fulfills the following requirements.
 o  An ABNF definition of the media property value name that meets
    "media-prop-ext" definition is included.
 o  A definition of which media property group it belongs to or define
    a new group is included.
 o  A description of all changes to the behavior of RTSP as result of
    these changes is included.
 o  A contact person for the registration is listed.

22.7.3. Registered Values

 This specification registers the ten values listed in Section 18.29.
 The registry contains the property group, the name of the media
 property, and the reference.

22.8. Notify-Reason Values

22.8.1. Description

 Notify-Reason values are used to indicate the reason the notification
 was sent.  Each reason has its associated rules on what headers and
 information may or must be included in the notification.  New
 notification behaviors need to be specified to enable interoperable
 usage; thus, a specification of each new value is required.

22.8.2. Registration Rules

 Registrations for new Notify-Reason values follow the Specification
 Required policy [RFC5226].  The expert reviewer verifies that the
 request fulfills the following requirements:
 o  An ABNF definition of the Notify-Reason value name that meets
    "Notify-Reason-extension" definition is included.
 o  A description of which headers shall be included in the request
    and response, when it should be sent, and any effect it has on the
    server client state is made clear.
 o  A contact person for the registration is listed.

Schulzrinne, et al. Standards Track [Page 226] RFC 7826 RTSP 2.0 December 2016

22.8.3. Registered Values

 This specification registers three values defined in the Notify-Reas-
 val ABNF, Section 20.2.3:
 end-of-stream:  This Notify-Reason value indicates the end of a media
    stream.
 media-properties-update:  This Notify-Reason value allows the server
    to indicate that the properties of the media have changed during
    the playout.
 scale-change:  This Notify-Reason value allows the server to notify
    the client about a change in the scale of the media.
 The registry contains the name, description, and reference.

22.9. Range Header Formats

22.9.1. Description

 The Range header (Section 18.40) allows for different range formats.
 These range formats also need an identifier to be used in the Accept-
 Ranges header (Section 18.5).  New range formats may be registered,
 but moderation should be applied as it makes interoperability more
 difficult.

22.9.2. Registration Rules

 A registration follows the Specification Required policy [RFC5226].
 The expert reviewer verifies that the request fulfills the following
 requirements:
 o  An ABNF definition of the range format that fulfills the "range-
    ext" definition is included.
 o  The range format identifier used in Accept-Ranges header according
    to the "extension-format" definition is defined.
 o  Rules for how one handles the range when using a negative Scale
    are included.
 o  A contact person for the registration is listed.

Schulzrinne, et al. Standards Track [Page 227] RFC 7826 RTSP 2.0 December 2016

22.9.3. Registered Values

 The registry contains the Range header format identifier, the name of
 the range format, and the reference.  This specification registers
 the following values.
 npt:  Normal Play Time
 clock:  UTC Absolute Time format
 smpte:  SMPTE Timestamps
 smpte-30-drop:  SMPTE Timestamps 29.97 Frames/sec (30 Hz with Drop)
 smpte-25:  SMPTE Timestamps 25 Frames/sec

22.10. Terminate-Reason Header

 The Terminate-Reason header (Section 18.52) has two registries for
 extensions.

22.10.1. Redirect Reasons

 This registry contains reasons for session termination that can be
 included in a Terminate-Reason header (Section 18.52).  Registrations
 follow the Expert Review policy [RFC5226].  The expert reviewer
 verifies that the registration request contains the following
 information:
 o  That the value follows the Terminate-Reason ABNF, i.e., be a
    token.
 o  That the specification provide a definition of what procedures are
    to be followed when a client receives this redirect reason.
 o  A contact person
 This specification registers three values:
 o  Session-Timeout
 o  Server-Admin
 o  Internal-Error
 The registry contains the name of the Redirect Reason and the
 reference.

Schulzrinne, et al. Standards Track [Page 228] RFC 7826 RTSP 2.0 December 2016

22.10.2. Terminate-Reason Header Parameters

 This registry contains parameters that may be included in the
 Terminate-Reason header (Section 18.52) in addition to a reason.
 Registrations are made under the Specification Required policy
 [RFC5226].  The expert reviewer verifies that the registration
 request contains the following:
 o  A parameter name.
 o  A parameter following the syntax allowed by the RTSP 2.0
    specification.
 o  A reference to the specification.
 o  A contact person.
 This specification registers:
 o  time
 o  user-msg
 The registry contains the name of the Terminate Reason and the
 reference.

22.11. RTP-Info Header Parameters

22.11.1. Description

 The RTP-Info header (Section 18.45) carries one or more parameter
 value pairs with information about a particular point in the RTP
 stream.  RTP extensions or new usages may need new types of
 information.  As RTP information that could be needed is likely to be
 generic enough, and to maximize the interoperability, new
 registration is made under the Specification Required policy.

22.11.2. Registration Rules

 Registrations for new RTP-Info values follow the policy of
 Specification Required [RFC5226].  The expert reviewer verifies that
 the registration request contains the following information.
 o  An ABNF definition that meets the "generic-param" definition.
 o  A reference to the specification.
 o  A contact person for the registration.

Schulzrinne, et al. Standards Track [Page 229] RFC 7826 RTSP 2.0 December 2016

22.11.3. Registered Values

 This specification registers the following parameter value pairs:
 o  url
 o  ssrc
 o  seq
 o  rtptime
 The registry contains the name of the parameter and the reference.

22.12. Seek-Style Policies

22.12.1. Description

 Seek-Style policy defines how the RTSP agent seeks in media content
 when given a position within the media content.  New seek policies
 may be registered; however, a large number of these will complicate
 implementation substantially.  The impact of unknown policies is that
 the server will not honor the unknown and will use the server default
 policy instead.

22.12.2. Registration Rules

 Registrations of new Seek-Style policies follow the Specification
 Required policy [RFC5226].  The expert reviewer verifies that the
 registration request fulfills the following requirements:
 o  Has an ABNF definition of the Seek-Style policy name that meets
    "Seek-S-value-ext" definition.
 o  Includes a short description.
 o  Lists a contact person for the registration.
 o  Includes a description of which headers shall be included in the
    request and response, when it should be sent, and any affect it
    has on the server-client state.

22.12.3. Registered Values

 This specification registers four values (Name - Short Description):
 o  RAP - Using the closest Random Access Point prior to or at the
    requested start position.

Schulzrinne, et al. Standards Track [Page 230] RFC 7826 RTSP 2.0 December 2016

 o  CoRAP - Conditional Random Access Point is like RAP, but only if
    the RAP is closer prior to the requested start position than
    current pause point.
 o  First-Prior - The first-prior policy will start delivery with the
    media unit that has a playout time first prior to the requested
    start position.
 o  Next - The next media units after the provided start position.
 The registry contains the name of the Seek-Style policy, the
 description, and the reference.

22.13. Transport Header Registries

 The transport header (Section 18.54) contains a number of parameters
 that have possibilities for future extensions.  Therefore, registries
 for these are defined below.

22.13.1. Transport Protocol Identifier

 A Transport Protocol specification consists of a transport protocol
 identifier, representing some combination of transport protocols, and
 any number of transport header parameters required or optional to use
 with the identified protocol specification.  This registry contains
 the identifiers used by registered transport protocol identifiers.
 A registration for the parameter transport protocol identifier
 follows the Specification Required policy [RFC5226].  The expert
 reviewer verifies that the registration request fulfills the
 following requirements:
 o  A contact person or organization with address and email.
 o  A value definition that follows the ABNF syntax definition of
    "transport-id" Section 20.2.3.
 o  A descriptive text that explains how the registered values are
    used in RTSP, which underlying transport protocols are used, and
    any required Transport header parameters.
 The registry contains the protocol ID string and the reference.

Schulzrinne, et al. Standards Track [Page 231] RFC 7826 RTSP 2.0 December 2016

 This specification registers the following values:
 RTP/AVP:  Use of the RTP [RFC3550] protocol for media transport in
       combination with the "RTP Profile for Audio and Video
       Conferences with Minimal Control" [RFC3551] over UDP.  The
       usage is explained in RFC 7826, Appendix C.1.
 RTP/AVP/UDP:  the same as RTP/AVP.
 RTP/AVPF:  Use of the RTP [RFC3550] protocol for media transport in
       combination with the "Extended RTP Profile for RTCP-based
       Feedback (RTP/AVPF)" [RFC4585] over UDP.  The usage is
       explained in RFC 7826, Appendix C.1.
 RTP/AVPF/UDP:  the same as RTP/AVPF.
 RTP/SAVP:  Use of the RTP [RFC3550] protocol for media transport in
       combination with the "The Secure Real-time Transport Protocol
       (SRTP)" [RFC3711] over UDP.  The usage is explained in RFC
       7826, Appendix C.1.
 RTP/SAVP/UDP:  the same as RTP/SAVP.
 RTP/SAVPF:  Use of the RTP [RFC3550] protocol for media transport in
       combination with the "Extended Secure RTP Profile for Real-time
       Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)"
       [RFC5124] over UDP.  The usage is explained in RFC 7826,
       Appendix C.1.
 RTP/SAVPF/UDP:  the same as RTP/SAVPF.
 RTP/AVP/TCP:  Use of the RTP [RFC3550] protocol for media transport
       in combination with the "RTP profile for audio and video
       conferences with minimal control" [RFC3551] over TCP.  The
       usage is explained in RFC 7826, Appendix C.2.2.
 RTP/AVPF/TCP:  Use of the RTP [RFC3550] protocol for media transport
       in combination with the "Extended RTP Profile for Real-time
       Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)"
       [RFC4585] over TCP.  The usage is explained in RFC 7826,
       Appendix C.2.2.
 RTP/SAVP/TCP:  Use of the RTP [RFC3550] protocol for media transport
       in combination with the "The Secure Real-time Transport
       Protocol (SRTP)" [RFC3711] over TCP.  The usage is explained in
       RFC 7826, Appendix C.2.2.

Schulzrinne, et al. Standards Track [Page 232] RFC 7826 RTSP 2.0 December 2016

 RTP/SAVPF/TCP:  Use of the RTP [RFC3550] protocol for media transport
       in combination with the "Extended Secure RTP Profile for Real-
       time Transport Control Protocol (RTCP)-Based Feedback (RTP/
       SAVPF)" [RFC5124] over TCP.  The usage is explained in RFC
       7826, Appendix C.2.2.

22.13.2. Transport Modes

 The Transport Mode is a Transport header (Section 18.54) parameter.
 It is used to identify the general mode of media transport.  The PLAY
 value registered defines a PLAYBACK mode, where media flows from
 server to client.
 A registration for the transport parameter mode follows the Standards
 Action policy [RFC5226].  The registration request needs to meet the
 following requirements:
 o  A value definition that follows the ABNF "token" definition
    Section 20.2.3.
 o  Text that explains how the registered value is used in RTSP.
 This specification registers one value:
 PLAY: See RFC 7826.
 The registry contains the transport mode value and the reference.

22.13.3. Transport Parameters

 Transport Parameters are different parameters used in a Transport
 header's transport specification (Section 18.54) to provide
 additional information required beyond the transport protocol
 identifier to establish a functioning transport.
 A registration for parameters that may be included in the Transport
 header follows the Specification Required policy [RFC5226].  The
 expert reviewer verifies that the registration request fulfills the
 following requirements:
 o  A Transport Parameter Name following the "token" ABNF definition.
 o  A value definition, if the parameter takes a value, that follows
    the ABNF definition of "trn-par-value" Section 20.2.3.
 o  Text that explains how the registered value is used in RTSP.

Schulzrinne, et al. Standards Track [Page 233] RFC 7826 RTSP 2.0 December 2016

 This specification registers all the transport parameters defined in
 Section 18.54.  This is a copy of that list:
 o  unicast
 o  multicast
 o  interleaved
 o  ttl
 o  layers
 o  ssrc
 o  mode
 o  dest_addr
 o  src_addr
 o  setup
 o  connection
 o  RTCP-mux
 o  MIKEY
 The registry contains the transport parameter name and the reference.

22.14. URI Schemes

 This specification updates two URI schemes: one previously
 registered, "rtsp", and one missing in the registry, "rtspu"
 (previously only defined in RTSP 1.0 [RFC2326]).  One new URI scheme,
 "rtsps", is also registered.  These URI schemes are registered in an
 existing registry ("Uniform Resource Identifier (URI) Schemes") not
 created by this memo.  Registrations follow [RFC7595].

22.14.1. The "rtsp" URI Scheme

 URI scheme name:  rtsp
 Status:  Permanent
 URI scheme syntax:  See Section 20.2.1 of RFC 7826.

Schulzrinne, et al. Standards Track [Page 234] RFC 7826 RTSP 2.0 December 2016

 URI scheme semantics:  The rtsp scheme is used to indicate resources
       accessible through the usage of the Real-Time Streaming
       Protocol (RTSP).  RTSP allows different operations on the
       resource identified by the URI, but the primary purpose is the
       streaming delivery of the resource to a client.  However, the
       operations that are currently defined are DESCRIBE,
       GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
       SETUP, SET_PARAMETER, and TEARDOWN.
 Encoding considerations:  IRIs in this scheme are defined and need to
       be encoded as RTSP URIs when used within RTSP.  That encoding
       is done according to RFC 3987.
 Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC
       2326), RTSP 2.0 (RFC 7826).
 Interoperability considerations:  The extensions in the URI syntax
       performed between RTSP 1.0 and 2.0 can create interoperability
       issues.  The changes are:
          Support for IPv6 literals in the host part and future IP
          literals through a mechanism as defined in RFC 3986.
          A new relative format to use in RTSP elements that is not
          required to start with "/".
       The above changes should have no impact on interoperability as
       discussed in detail in Section 4.2 of RFC 7826.
 Security considerations:  All the security threats identified in
       Section 7 of RFC 3986 also apply to this scheme.  They need to
       be reviewed and considered in any implementation utilizing this
       scheme.
 Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com
 Author/Change controller:  IETF
 References:  RFC 2326, RFC 3986, RFC 3987, and RFC 7826

22.14.2. The "rtsps" URI Scheme

 URI scheme name:  rtsps
 Status:  Permanent
 URI scheme syntax:  See Section 20.2.1 of RFC 7826.

Schulzrinne, et al. Standards Track [Page 235] RFC 7826 RTSP 2.0 December 2016

 URI scheme semantics:  The rtsps scheme is used to indicate resources
       accessible through the usage of the Real-Time Streaming
       Protocol (RTSP) over TLS.  RTSP allows different operations on
       the resource identified by the URI, but the primary purpose is
       the streaming delivery of the resource to a client.  However,
       the operations that are currently defined are DESCRIBE,
       GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
       SETUP, SET_PARAMETER, and TEARDOWN.
 Encoding considerations:  IRIs in this scheme are defined and need to
       be encoded as RTSP URIs when used within RTSP.  That encoding
       is done according to RFC 3987.
 Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC
       2326), RTSP 2.0 (RFC 7826).
 Interoperability considerations:  The "rtsps" scheme was never
       officially defined for RTSP 1.0; however, it has seen
       widespread use in actual deployments of RTSP 1.0.  Therefore,
       this section discusses the believed changes between the
       unspecified RTSP 1.0 "rtsps" scheme and RTSP 2.0 definition.
       The extensions in the URI syntax performed between RTSP 1.0 and
       2.0 can create interoperability issues.  The changes are:
          Support for IPv6 literals in the host part and future IP
          literals through a mechanism as defined by RFC 3986.
          A new relative format to use in RTSP elements that is not
          required to start with "/".
       The above changes should have no impact on interoperability as
       discussed in detail in Section 4.2 of RFC 7826.
 Security considerations:  All the security threats identified in
       Section 7 of RFC 3986 also apply to this scheme.  They need to
       be reviewed and considered in any implementation utilizing this
       scheme.
 Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com
 Author/Change controller:  IETF
 References:  RFC 2326, RFC 3986, RFC 3987, and RFC 7826

Schulzrinne, et al. Standards Track [Page 236] RFC 7826 RTSP 2.0 December 2016

22.14.3. The "rtspu" URI Scheme

 URI scheme name:  rtspu
 Status:  Permanent
 URI scheme syntax:  See Section 3.2 of RFC 2326.
 URI scheme semantics:  The rtspu scheme is used to indicate resources
       accessible through the usage of the Real-Time Streaming
       Protocol (RTSP) over unreliable datagram transport.  RTSP
       allows different operations on the resource identified by the
       URI, but the primary purpose is the streaming delivery of the
       resource to a client.  However, the operations that are
       currently defined are DESCRIBE, GET_PARAMETER, OPTIONS,
       REDIRECT,PLAY, PLAY_NOTIFY, PAUSE, SETUP, SET_PARAMETER, and
       TEARDOWN.
 Encoding considerations:  This scheme is not intended to be used with
       characters outside the US-ASCII repertoire.
 Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC
       2326).
 Interoperability considerations:  The definition of the transport
       mechanism of RTSP over UDP has interoperability issues.  That
       makes the usage of this scheme problematic.
 Security considerations:  All the security threats identified in
       Section 7 of RFC 3986 also apply to this scheme.  They need to
       be reviewed and considered in any implementation utilizing this
       scheme.
 Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com
 Author/Change controller:  IETF
 References:  RFC 2326

Schulzrinne, et al. Standards Track [Page 237] RFC 7826 RTSP 2.0 December 2016

22.15. SDP Attributes

 This specification defines three SDP [RFC4566] attributes that have
 been registered by IANA.
 SDP Attribute ("att-field"):
      Attribute name:     range
      Long form:          Media Range Attribute
      Type of name:       att-field
      Type of attribute:  both session and media level
      Subject to charset: No
      Purpose:            RFC 7826
      Reference:          RFC 2326, RFC 7826
      Values:             See ABNF definition.
      Attribute name:     control
      Long form:          RTSP control URI
      Type of name:       att-field
      Type of attribute:  both session and media level
      Subject to charset: No
      Purpose:            RFC 7826
      Reference:          RFC 2326, RFC 7826
      Values:             Absolute or Relative URIs.
      Attribute name:     mtag
      Long form:          Message Tag
      Type of name:       att-field
      Type of attribute:  both session and media level
      Subject to charset: No
      Purpose:            RFC 7826
      Reference:          RFC 7826
      Values:             See ABNF definition

22.16. Media Type Registration for text/parameters

 Type name:  text
 Subtype name:  parameters
 Required parameters:
 Optional parameters:  charset: The charset parameter is applicable to
    the encoding of the parameter values.  The default charset is
    UTF-8, if the 'charset' parameter is not present.
 Encoding considerations:  8bit

Schulzrinne, et al. Standards Track [Page 238] RFC 7826 RTSP 2.0 December 2016

 Security considerations:  This format may carry any type of
    parameters.  Some can have security requirements, like privacy,
    confidentiality, or integrity requirements.  The format has no
    built-in security protection.  For the usage, the transport can be
    protected between server and client using TLS.  However, care must
    be taken to consider if the proxies are also trusted with the
    parameters in case hop-by-hop security is used.  If stored as a
    file in a file system, the necessary precautions need to be taken
    in relation to the parameter requirements including object
    security such as S/MIME [RFC5751].
 Interoperability considerations:  This media type was mentioned as a
    fictional example in [RFC2326], but was not formally specified.
    This has resulted in usage of this media type that may not match
    its formal definition.
 Published specification:  RFC 7826, Appendix F.
 Applications that use this media type:  Applications that use RTSP
    and have additional parameters they like to read and set using the
    RTSP GET_PARAMETER and SET_PARAMETER methods.
 Additional information:
 Magic number(s):  N/A
 File extension(s):  N/A
 Macintosh file type code(s):  N/A
 Person & email address to contact for further information:
    Magnus Westerlund (magnus.westerlund@ericsson.com)
 Intended usage:   Common
 Restrictions on usage:   None
 Author:  Magnus Westerlund (magnus.westerlund@ericsson.com)
 Change controller:  IETF
 Addition Notes:

Schulzrinne, et al. Standards Track [Page 239] RFC 7826 RTSP 2.0 December 2016

23. References

23.1. Normative References

 [FIPS180-4]
            National Institute of Standards and Technology (NIST),
            "Federal Information Processing Standards Publication:
            Secure Hash Standard (SHS)", DOI 10.6028/NIST.FIPS.180-4,
            August 2015, <http://nvlpubs.nist.gov/nistpubs/FIPS/
            NIST.FIPS.180-4.pdf>.
 [RFC768]   Postel, J., "User Datagram Protocol", STD 6, RFC 768,
            DOI 10.17487/RFC0768, August 1980,
            <http://www.rfc-editor.org/info/rfc768>.
 [RFC793]   Postel, J., "Transmission Control Protocol", STD 7,
            RFC 793, DOI 10.17487/RFC0793, September 1981,
            <http://www.rfc-editor.org/info/rfc793>.
 [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
            Requirement Levels", BCP 14, RFC 2119,
            DOI 10.17487/RFC2119, March 1997,
            <http://www.rfc-editor.org/info/rfc2119>.
 [RFC2460]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
            (IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460,
            December 1998, <http://www.rfc-editor.org/info/rfc2460>.
 [RFC2616]  Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
            Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
            Transfer Protocol -- HTTP/1.1", RFC 2616,
            DOI 10.17487/RFC2616, June 1999,
            <http://www.rfc-editor.org/info/rfc2616>.
 [RFC2617]  Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
            Leach, P., Luotonen, A., and L. Stewart, "HTTP
            Authentication: Basic and Digest Access Authentication",
            RFC 2617, DOI 10.17487/RFC2617, June 1999,
            <http://www.rfc-editor.org/info/rfc2617>.
 [RFC2818]  Rescorla, E., "HTTP Over TLS", RFC 2818,
            DOI 10.17487/RFC2818, May 2000,
            <http://www.rfc-editor.org/info/rfc2818>.
 [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
            July 2003, <http://www.rfc-editor.org/info/rfc3550>.

Schulzrinne, et al. Standards Track [Page 240] RFC 7826 RTSP 2.0 December 2016

 [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
            Video Conferences with Minimal Control", STD 65, RFC 3551,
            DOI 10.17487/RFC3551, July 2003,
            <http://www.rfc-editor.org/info/rfc3551>.
 [RFC3629]  Yergeau, F., "UTF-8, a transformation format of ISO
            10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
            2003, <http://www.rfc-editor.org/info/rfc3629>.
 [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
            Norrman, "The Secure Real-time Transport Protocol (SRTP)",
            RFC 3711, DOI 10.17487/RFC3711, March 2004,
            <http://www.rfc-editor.org/info/rfc3711>.
 [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
            Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
            DOI 10.17487/RFC3830, August 2004,
            <http://www.rfc-editor.org/info/rfc3830>.
 [RFC3986]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
            Resource Identifier (URI): Generic Syntax", STD 66,
            RFC 3986, DOI 10.17487/RFC3986, January 2005,
            <http://www.rfc-editor.org/info/rfc3986>.
 [RFC3987]  Duerst, M. and M. Suignard, "Internationalized Resource
            Identifiers (IRIs)", RFC 3987, DOI 10.17487/RFC3987,
            January 2005, <http://www.rfc-editor.org/info/rfc3987>.
 [RFC4086]  Eastlake 3rd, D., Schiller, J., and S. Crocker,
            "Randomness Requirements for Security", BCP 106, RFC 4086,
            DOI 10.17487/RFC4086, June 2005,
            <http://www.rfc-editor.org/info/rfc4086>.
 [RFC4291]  Hinden, R. and S. Deering, "IP Version 6 Addressing
            Architecture", RFC 4291, DOI 10.17487/RFC4291, February
            2006, <http://www.rfc-editor.org/info/rfc4291>.
 [RFC7595]  Thaler, D., Ed., Hansen, T., and T. Hardie, "Guidelines
            and Registration Procedures for URI Schemes", BCP 35, RFC
            7595, DOI 10.17487/RFC7595, June 2015, <http://www.rfc-
            editor.org/info/rfc7595>.
 [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
            Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
            July 2006, <http://www.rfc-editor.org/info/rfc4566>.

Schulzrinne, et al. Standards Track [Page 241] RFC 7826 RTSP 2.0 December 2016

 [RFC4571]  Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
            and RTP Control Protocol (RTCP) Packets over Connection-
            Oriented Transport", RFC 4571, DOI 10.17487/RFC4571, July
            2006, <http://www.rfc-editor.org/info/rfc4571>.
 [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
            "Extended RTP Profile for Real-time Transport Control
            Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
            DOI 10.17487/RFC4585, July 2006,
            <http://www.rfc-editor.org/info/rfc4585>.
 [RFC4648]  Josefsson, S., "The Base16, Base32, and Base64 Data
            Encodings", RFC 4648, DOI 10.17487/RFC4648, October 2006,
            <http://www.rfc-editor.org/info/rfc4648>.
 [RFC4738]  Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY-
            RSA-R: An Additional Mode of Key Distribution in
            Multimedia Internet KEYing (MIKEY)", RFC 4738,
            DOI 10.17487/RFC4738, November 2006,
            <http://www.rfc-editor.org/info/rfc4738>.
 [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
            Real-time Transport Control Protocol (RTCP)-Based Feedback
            (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
            2008, <http://www.rfc-editor.org/info/rfc5124>.
 [RFC5226]  Narten, T. and H. Alvestrand, "Guidelines for Writing an
            IANA Considerations Section in RFCs", BCP 26, RFC 5226,
            DOI 10.17487/RFC5226, May 2008,
            <http://www.rfc-editor.org/info/rfc5226>.
 [RFC5234]  Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
            Specifications: ABNF", STD 68, RFC 5234,
            DOI 10.17487/RFC5234, January 2008,
            <http://www.rfc-editor.org/info/rfc5234>.
 [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
            (TLS) Protocol Version 1.2", RFC 5246,
            DOI 10.17487/RFC5246, August 2008,
            <http://www.rfc-editor.org/info/rfc5246>.
 [RFC5280]  Cooper, D., Santesson, S., Farrell, S., Boeyen, S.,
            Housley, R., and W. Polk, "Internet X.509 Public Key
            Infrastructure Certificate and Certificate Revocation List
            (CRL) Profile", RFC 5280, DOI 10.17487/RFC5280, May 2008,
            <http://www.rfc-editor.org/info/rfc5280>.

Schulzrinne, et al. Standards Track [Page 242] RFC 7826 RTSP 2.0 December 2016

 [RFC5322]  Resnick, P., Ed., "Internet Message Format", RFC 5322,
            DOI 10.17487/RFC5322, October 2008,
            <http://www.rfc-editor.org/info/rfc5322>.
 [RFC5646]  Phillips, A., Ed. and M. Davis, Ed., "Tags for Identifying
            Languages", BCP 47, RFC 5646, DOI 10.17487/RFC5646,
            September 2009, <http://www.rfc-editor.org/info/rfc5646>.
 [RFC5751]  Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
            Mail Extensions (S/MIME) Version 3.2 Message
            Specification", RFC 5751, DOI 10.17487/RFC5751, January
            2010, <http://www.rfc-editor.org/info/rfc5751>.
 [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
            Control Packets on a Single Port", RFC 5761,
            DOI 10.17487/RFC5761, April 2010,
            <http://www.rfc-editor.org/info/rfc5761>.
 [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
            Protocol (SDP) Grouping Framework", RFC 5888,
            DOI 10.17487/RFC5888, June 2010,
            <http://www.rfc-editor.org/info/rfc5888>.
 [RFC6838]  Freed, N., Klensin, J., and T. Hansen, "Media Type
            Specifications and Registration Procedures", BCP 13,
            RFC 6838, DOI 10.17487/RFC6838, January 2013,
            <http://www.rfc-editor.org/info/rfc6838>.
 [RFC7230]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
            Protocol (HTTP/1.1): Message Syntax and Routing",
            RFC 7230, DOI 10.17487/RFC7230, June 2014,
            <http://www.rfc-editor.org/info/rfc7230>.
 [RFC7231]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
            Protocol (HTTP/1.1): Semantics and Content", RFC 7231,
            DOI 10.17487/RFC7231, June 2014,
            <http://www.rfc-editor.org/info/rfc7231>.
 [RFC7232]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
            Protocol (HTTP/1.1): Conditional Requests", RFC 7232,
            DOI 10.17487/RFC7232, June 2014,
            <http://www.rfc-editor.org/info/rfc7232>.
 [RFC7233]  Fielding, R., Ed., Lafon, Y., Ed., and J. Reschke, Ed.,
            "Hypertext Transfer Protocol (HTTP/1.1): Range Requests",
            RFC 7233, DOI 10.17487/RFC7233, June 2014,
            <http://www.rfc-editor.org/info/rfc7233>.

Schulzrinne, et al. Standards Track [Page 243] RFC 7826 RTSP 2.0 December 2016

 [RFC7234]  Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke,
            Ed., "Hypertext Transfer Protocol (HTTP/1.1): Caching",
            RFC 7234, DOI 10.17487/RFC7234, June 2014,
            <http://www.rfc-editor.org/info/rfc7234>.
 [RFC7235]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
            Protocol (HTTP/1.1): Authentication", RFC 7235,
            DOI 10.17487/RFC7235, June 2014,
            <http://www.rfc-editor.org/info/rfc7235>.
 [RFC7615]  Reschke, J., "HTTP Authentication-Info and Proxy-
            Authentication-Info Response Header Fields", RFC 7615,
            DOI 10.17487/RFC7615, September 2015,
            <http://www.rfc-editor.org/info/rfc7615>.
 [RFC7616]  Shekh-Yusef, R., Ed., Ahrens, D., and S. Bremer, "HTTP
            Digest Access Authentication", RFC 7616,
            DOI 10.17487/RFC7616, September 2015,
            <http://www.rfc-editor.org/info/rfc7616>.
 [RFC7617]  Reschke, J., "The 'Basic' HTTP Authentication Scheme",
            RFC 7617, DOI 10.17487/RFC7617, September 2015,
            <http://www.rfc-editor.org/info/rfc7617>.
 [RFC7825]  Goldberg, J., Westerlund, M., and T. Zeng, "A Network
            Address Translator (NAT) Traversal Mechanism for Media
            Controlled by Real-Time Streaming Protocol (RTSP)",
            RFC 7825, DOI 10.17487/RFC7825, December 2016,
            <http://www.rfc-editor.org/info/rfc7825>.
 [RTP-CIRCUIT-BREAKERS]
            Perkins, C. and V. Singh, "Multimedia Congestion Control:
            Circuit Breakers for Unicast RTP Sessions", Work in
            Progress, draft-ietf-avtcore-rtp-circuit-breakers-13,
            February 2016.
 [SMPTE-TC] Society of Motion Picture and Television Engineers, "ST
            12-1:2008 For Television -- Time and Control Code",
            DOI 10.5594/SMPTE.ST12-1.2008, February 2008,
            <http://ieeexplore.ieee.org/servlet/
            opac?punumber=7289818>.
 [TS-26234] 3rd Generation Partnership Project (3GPP), "Transparent
            end-to-end Packet-switched Streaming Service (PSS);
            Protocols and codecs", Technical Specification 26.234,
            Release 13, September 2015,
            <http://www.3gpp.org/DynaReport/26234.htm>.

Schulzrinne, et al. Standards Track [Page 244] RFC 7826 RTSP 2.0 December 2016

23.2. Informative References

 [ISO.13818-6.1995]
            International Organization for Standardization,
            "Information technology -- Generic coding of moving
            pictures and associated audio information - part 6:
            Extension for DSM-CC", ISO Draft Standard 13818-6:1998,
            October 1998,
            <http://www.iso.org/iso/home/store/catalogue_tc/
            catalogue_detail.htm?csnumber=25039>.
 [ISO.8601.2000]
            International Organization for Standardization, "Data
            elements and interchange formats - Information interchange
            - Representation of dates and times", ISO/IEC Standard
            8601, December 2000.
 [RFC791]   Postel, J., "Internet Protocol", STD 5, RFC 791,
            DOI 10.17487/RFC0791, September 1981,
            <http://www.rfc-editor.org/info/rfc791>.
 [RFC1123]  Braden, R., Ed., "Requirements for Internet Hosts -
            Application and Support", STD 3, RFC 1123,
            DOI 10.17487/RFC1123, October 1989,
            <http://www.rfc-editor.org/info/rfc1123>.
 [RFC2068]  Fielding, R., Gettys, J., Mogul, J., Frystyk, H., and T.
            Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1",
            RFC 2068, DOI 10.17487/RFC2068, January 1997,
            <http://www.rfc-editor.org/info/rfc2068>.
 [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
            Streaming Protocol (RTSP)", RFC 2326,
            DOI 10.17487/RFC2326, April 1998,
            <http://www.rfc-editor.org/info/rfc2326>.
 [RFC2663]  Srisuresh, P. and M. Holdrege, "IP Network Address
            Translator (NAT) Terminology and Considerations",
            RFC 2663, DOI 10.17487/RFC2663, August 1999,
            <http://www.rfc-editor.org/info/rfc2663>.
 [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
            Announcement Protocol", RFC 2974, DOI 10.17487/RFC2974,
            October 2000, <http://www.rfc-editor.org/info/rfc2974>.

Schulzrinne, et al. Standards Track [Page 245] RFC 7826 RTSP 2.0 December 2016

 [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
            A., Peterson, J., Sparks, R., Handley, M., and E.
            Schooler, "SIP: Session Initiation Protocol", RFC 3261,
            DOI 10.17487/RFC3261, June 2002,
            <http://www.rfc-editor.org/info/rfc3261>.
 [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
            with Session Description Protocol (SDP)", RFC 3264,
            DOI 10.17487/RFC3264, June 2002,
            <http://www.rfc-editor.org/info/rfc3264>.
 [RFC3339]  Klyne, G. and C. Newman, "Date and Time on the Internet:
            Timestamps", RFC 3339, DOI 10.17487/RFC3339, July 2002,
            <http://www.rfc-editor.org/info/rfc3339>.
 [RFC4145]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in
            the Session Description Protocol (SDP)", RFC 4145,
            DOI 10.17487/RFC4145, September 2005,
            <http://www.rfc-editor.org/info/rfc4145>.
 [RFC4567]  Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
            Carrara, "Key Management Extensions for Session
            Description Protocol (SDP) and Real Time Streaming
            Protocol (RTSP)", RFC 4567, DOI 10.17487/RFC4567, July
            2006, <http://www.rfc-editor.org/info/rfc4567>.
 [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
            Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
            DOI 10.17487/RFC4588, July 2006,
            <http://www.rfc-editor.org/info/rfc4588>.
 [RFC4855]  Casner, S., "Media Type Registration of RTP Payload
            Formats", RFC 4855, DOI 10.17487/RFC4855, February 2007,
            <http://www.rfc-editor.org/info/rfc4855>.
 [RFC4856]  Casner, S., "Media Type Registration of Payload Formats in
            the RTP Profile for Audio and Video Conferences",
            RFC 4856, DOI 10.17487/RFC4856, February 2007,
            <http://www.rfc-editor.org/info/rfc4856>.
 [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
            "Codec Control Messages in the RTP Audio-Visual Profile
            with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
            February 2008, <http://www.rfc-editor.org/info/rfc5104>.

Schulzrinne, et al. Standards Track [Page 246] RFC 7826 RTSP 2.0 December 2016

 [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
            (ICE): A Protocol for Network Address Translator (NAT)
            Traversal for Offer/Answer Protocols", RFC 5245,
            DOI 10.17487/RFC5245, April 2010,
            <http://www.rfc-editor.org/info/rfc5245>.
 [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
            "Session Traversal Utilities for NAT (STUN)", RFC 5389,
            DOI 10.17487/RFC5389, October 2008,
            <http://www.rfc-editor.org/info/rfc5389>.
 [RFC5583]  Schierl, T. and S. Wenger, "Signaling Media Decoding
            Dependency in the Session Description Protocol (SDP)",
            RFC 5583, DOI 10.17487/RFC5583, July 2009,
            <http://www.rfc-editor.org/info/rfc5583>.
 [RFC5905]  Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch,
            "Network Time Protocol Version 4: Protocol and Algorithms
            Specification", RFC 5905, DOI 10.17487/RFC5905, June 2010,
            <http://www.rfc-editor.org/info/rfc5905>.
 [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
            "Computing TCP's Retransmission Timer", RFC 6298,
            DOI 10.17487/RFC6298, June 2011,
            <http://www.rfc-editor.org/info/rfc6298>.
 [Stevens98]
            Stevens, W., Fenner, B., and A. Rudoff, "Unix Networking
            Programming, Volume 1: The Sockets Networking API (3rd
            Edition)", 1998.

Schulzrinne, et al. Standards Track [Page 247] RFC 7826 RTSP 2.0 December 2016

Appendix A. Examples

 This section contains several different examples trying to illustrate
 possible ways of using RTSP.  The examples can also help with the
 understanding of how functions of RTSP work.  However, remember that
 these are examples and the normative and syntax descriptions in the
 other sections take precedence.  Please also note that many of the
 examples have been broken into several lines, where following lines
 start with whitespace as allowed by the syntax.

A.1. Media on Demand (Unicast)

 This is an example of media-on-demand streaming of media stored in a
 container file.  For the purposes of this example, a container file
 is a storage entity in which multiple continuous media types
 pertaining to the same end-user presentation are present.  In effect,
 the container file represents an RTSP presentation, with each of its
 components being RTSP-controlled media streams.  Container files are
 a widely used means to store such presentations.  While the
 components are transported as independent streams, it is desirable to
 maintain a common context for those streams at the server end.
    This enables the server to keep a single storage handle open
    easily.  It also allows treating all the streams equally in case
    of any prioritization of streams by the server.
 It is also possible that the presentation author may wish to prevent
 selective retrieval of the streams by the client in order to preserve
 the artistic effect of the combined media presentation.  Similarly,
 in such a tightly bound presentation, it is desirable to be able to
 control all the streams via a single control message using an
 aggregate URI.
 The following is an example of using a single RTSP session to control
 multiple streams.  It also illustrates the use of aggregate URIs.  In
 a container file, it is also desirable not to write any URI parts
 that are not kept when the container is distributed, like the host
 and most of the path element.  Therefore, this example also uses the
 "*" and relative URI in the delivered SDP.
 Also, this presentation description (SDP) is not cacheable, as the
 Expires header is set to an equal value with date indicating
 immediate expiration of its validity.
 Client C requests a presentation from media server M.  The movie is
 stored in a container file.  The client has obtained an RTSP URI to
 the container file.

Schulzrinne, et al. Standards Track [Page 248] RFC 7826 RTSP 2.0 December 2016

 C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
       CSeq: 1
       User-Agent: PhonyClient/1.2
 M->C: RTSP/2.0 200 OK
       CSeq: 1
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:20:32 +0000
       Content-Type: application/sdp
       Content-Length: 271
       Content-Base: rtsp://example.com/twister.3gp/
       Expires: Fri, 20 Dec 2013 12:20:32 +0000
       v=0
       o=- 2890844256 2890842807 IN IP4 198.51.100.5
       s=RTSP Session
       i=An Example of RTSP Session Usage
       e=adm@example.com
       c=IN IP4 0.0.0.0
       a=control: *
       a=range:npt=00:00:00-00:10:34.10
       t=0 0
       m=audio 0 RTP/AVP 0
       a=control: trackID=1
       m=video 0 RTP/AVP 26
       a=control: trackID=4
 C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
       CSeq: 2
       User-Agent: PhonyClient/1.2
       Require: play.basic
       Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
       Accept-Ranges: npt, smpte, clock
 M->C: RTSP/2.0 200 OK
       CSeq: 2
       Server: PhonyServer/1.0
       Transport: RTP/AVP;unicast; ssrc=93CB001E;
                  dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
                  src_addr="198.51.100.5:9000"/"198.51.100.5:9001"
       Session: OccldOFFq23KwjYpAnBbUr
       Expires: Fri, 20 Dec 2013 12:20:33 +0000
       Date: Fri, 20 Dec 2013 10:20:33 +0000
       Accept-Ranges: npt
       Media-Properties: Random-Access=0.02, Immutable, Unlimited

Schulzrinne, et al. Standards Track [Page 249] RFC 7826 RTSP 2.0 December 2016

 C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
       CSeq: 3
       User-Agent: PhonyClient/1.2
       Require: play.basic
       Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
       Session: OccldOFFq23KwjYpAnBbUr
       Accept-Ranges: npt, smpte, clock
 M->C: RTSP/2.0 200 OK
       CSeq: 3
       Server: PhonyServer/1.0
       Transport: RTP/AVP;unicast; ssrc=A813FC13;
                  dest_addr="192.0.2.53:8002"/"192.0.2.53:8003";
                  src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
       Session: OccldOFFq23KwjYpAnBbUr
       Expires: Fri, 20 Dec 2013 12:20:33 +0000
       Date: Fri, 20 Dec 2013 10:20:33 +0000
       Accept-Range: NPT
       Media-Properties: Random-Access=0.8, Immutable, Unlimited
 C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
       CSeq: 4
       User-Agent: PhonyClient/1.2
       Range: npt=30-
       Seek-Style: RAP
       Session: OccldOFFq23KwjYpAnBbUr
 M->C: RTSP/2.0 200 OK
       CSeq: 4
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:20:34 +0000
       Session: OccldOFFq23KwjYpAnBbUr
       Range: npt=30-634.10
       Seek-Style: RAP
       RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
          ssrc=0D12F123:seq=12345;rtptime=3450012,
         url="rtsp://example.com/twister.3gp/trackID=1"
          ssrc=4F312DD8:seq=54321;rtptime=2876889
 C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/2.0
       CSeq: 5
       User-Agent: PhonyClient/1.2
       Session: OccldOFFq23KwjYpAnBbUr
 # Pause happens 0.87 seconds after starting to play

Schulzrinne, et al. Standards Track [Page 250] RFC 7826 RTSP 2.0 December 2016

 M->C: RTSP/2.0 200 OK
       CSeq: 5
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:20:35 +0000
       Session: OccldOFFq23KwjYpAnBbUr
       Range: npt=30.87-634.10
 C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
       CSeq: 6
       User-Agent: PhonyClient/1.2
       Range: npt=30.87-634.10
       Seek-Style: Next
       Session: OccldOFFq23KwjYpAnBbUr
 M->C: RTSP/2.0 200 OK
       CSeq: 6
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:22:13 +0000
       Session: OccldOFFq23KwjYpAnBbUr
       Range: npt=30.87-634.10
       Seek-Style: Next
       RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
          ssrc=0D12F123:seq=12555;rtptime=6330012,
         url="rtsp://example.com/twister.3gp/trackID=1"
          ssrc=4F312DD8:seq=55021;rtptime=3132889
 C->M: TEARDOWN rtsp://example.com/twister.3gp/ RTSP/2.0
       CSeq: 7
       User-Agent: PhonyClient/1.2
       Session: OccldOFFq23KwjYpAnBbUr
 M->C: RTSP/2.0 200 OK
       CSeq: 7
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:31:53 +0000

A.2. Media on Demand Using Pipelining

 This example is basically the example above (Appendix A.1), but now
 utilizing pipelining to speed up the setup.  It requires only two
 round-trip times until the media starts flowing.  First of all, the
 session description is retrieved to determine what media resources
 need to be set up.  In the second step, one sends the necessary SETUP
 requests and the PLAY request to initiate media delivery.

Schulzrinne, et al. Standards Track [Page 251] RFC 7826 RTSP 2.0 December 2016

 Client C requests a presentation from media server M.  The movie is
 stored in a container file.  The client has obtained an RTSP URI to
 the container file.
 C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
       CSeq: 1
       User-Agent: PhonyClient/1.2
 M->C: RTSP/2.0 200 OK
       CSeq: 1
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:20:32 +0000
       Content-Type: application/sdp
       Content-Length: 271
       Content-Base: rtsp://example.com/twister.3gp/
       Expires: Fri, 20 Dec 2013 12:20:32 +0000
       v=0
       o=- 2890844256 2890842807 IN IP4 192.0.2.5
       s=RTSP Session
       i=An Example of RTSP Session Usage
       e=adm@example.com
       c=IN IP4 0.0.0.0
       a=control: *
       a=range:npt=00:00:00-00:10:34.10
       t=0 0
       m=audio 0 RTP/AVP 0
       a=control: trackID=1
       m=video 0 RTP/AVP 26
       a=control: trackID=4
 C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
       CSeq: 2
       User-Agent: PhonyClient/1.2
       Require: play.basic
       Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
       Accept-Ranges: npt, smpte, clock
       Pipelined-Requests: 7654

Schulzrinne, et al. Standards Track [Page 252] RFC 7826 RTSP 2.0 December 2016

 C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
       CSeq: 3
       User-Agent: PhonyClient/1.2
       Require: play.basic
       Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
       Accept-Ranges: npt, smpte, clock
       Pipelined-Requests: 7654
 C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
       CSeq: 4
       User-Agent: PhonyClient/1.2
       Range: npt=0-
       Seek-Style: RAP
       Pipelined-Requests: 7654
 M->C: RTSP/2.0 200 OK
       CSeq: 2
       Server: PhonyServer/1.0
       Transport: RTP/AVP;unicast;
                  dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
                  src_addr="198.51.100.5:9000"/"198.51.100.5:9001";
                  ssrc=93CB001E
       Session: OccldOFFq23KwjYpAnBbUr
       Expires: Fri, 20 Dec 2013 12:20:32 +0000
       Date: Fri, 20 Dec 2013 10:20:32 +0000
       Accept-Ranges: npt
       Pipelined-Requests: 7654
       Media-Properties: Random-Access=0.2, Immutable, Unlimited
 M->C: RTSP/2.0 200 OK
       CSeq: 3
       Server: PhonyServer/1.0
       Transport: RTP/AVP;unicast;
                  dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
                  src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
                  ssrc=A813FC13
       Session: OccldOFFq23KwjYpAnBbUr
       Expires: Sat, 21 Dec 2013 10:20:32 +0000
       Date: Fri, 20 Dec 2013 10:20:32 +0000
       Accept-Range: NPT
       Pipelined-Requests: 7654
       Media-Properties: Random-Access=0.8, Immutable, Unlimited

Schulzrinne, et al. Standards Track [Page 253] RFC 7826 RTSP 2.0 December 2016

 M->C: RTSP/2.0 200 OK
       CSeq: 4
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:20:32 +0000
       Session: OccldOFFq23KwjYpAnBbUr
       Range: npt=0-623.10
       Seek-Style: RAP
       RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
          ssrc=0D12F123:seq=12345;rtptime=3450012,
         url="rtsp://example.com/twister.3gp/trackID=1"
          ssrc=4F312DD8:seq=54321;rtptime=2876889
       Pipelined-Requests: 7654

A.3. Secured Media Session for On-Demand Content

 This example is basically the above example (Appendix A.2), but now
 including establishment of SRTP crypto contexts to get a secured
 media delivery.  First of all, the client attempts to fetch this
 insecurely, but the server redirects to a URI indicating a
 requirement on using a secure connection for the RTSP messages.  The
 client establishes a TCP/TLS connection, and the session description
 is retrieved to determine what media resources need to be set up.  In
 the this session description, secure media (SRTP) is indicated.  In
 the next step, the client sends the necessary SETUP requests
 including MIKEY messages.  This is pipelined with a PLAY request to
 initiate media delivery.
 Client C requests a presentation from media server M.  The movie is
 stored in a container file.  The client has obtained an RTSP URI to
 the container file.
 Note: The MIKEY messages below are not valid MIKEY messages and are
 Base64-encoded random data to represent where the MIKEY messages
 would go.
 C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
       CSeq: 1
       User-Agent: PhonyClient/1.2
 M->C: RTSP/2.0 301 Moved Permanently
       CSeq: 1
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:25:32 +0000
       Location: rtsps://example.com/twister.3gp
 C->M: Establish TCP/TLS connection and verify server's
       certificate that represents example.com.
       Used for all below RTSP messages.

Schulzrinne, et al. Standards Track [Page 254] RFC 7826 RTSP 2.0 December 2016

 C->M: DESCRIBE rtsps://example.com/twister.3gp RTSP/2.0
       CSeq: 2
       User-Agent: PhonyClient/1.2
 M->C: RTSP/2.0 200 OK
       CSeq: 2
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:25:33 +0000
       Content-Type: application/sdp
       Content-Length: 271
       Content-Base: rtsps://example.com/twister.3gp/
       Expires: Fri, 20 Dec 2013 12:25:33 +0000
       v=0
       o=- 2890844256 2890842807 IN IP4 192.0.2.5
       s=RTSP Session
       i=An Example of RTSP Session Usage
       e=adm@example.com
       c=IN IP4 0.0.0.0
       a=control: *
       a=range:npt=00:00:00-00:10:34.10
       t=0 0
       m=audio 0 RTP/SAVP 0
       a=control: trackID=1
       m=video 0 RTP/SAVP 26
       a=control: trackID=4
 C->M: SETUP rtsps://example.com/twister.3gp/trackID=1 RTSP/2.0
       CSeq: 3
       User-Agent: PhonyClient/1.2
       Require: play.basic
       Transport: RTP/SAVP;unicast;dest_addr=":8000"/":8001";
          MIKEY=VGhpcyBpcyB0aGUgZmlyc3Qgc3RyZWFtcyBNSUtFWSBtZXNzYWdl
       Accept-Ranges: npt, smpte, clock
       Pipelined-Requests: 7654
 C->M: SETUP rtsps://example.com/twister.3gp/trackID=4 RTSP/2.0
       CSeq: 4
       User-Agent: PhonyClient/1.2
       Require: play.basic
       Transport: RTP/SAVP;unicast;dest_addr=":8002"/":8003";
          MIKEY=TUlLRVkgZm9yIHN0cmVhbSB0d2lzdGVyLjNncC90cmFja0lEPTQ=
       Accept-Ranges: npt, smpte, clock
       Pipelined-Requests: 7654

Schulzrinne, et al. Standards Track [Page 255] RFC 7826 RTSP 2.0 December 2016

 C->M: PLAY rtsps://example.com/twister.3gp/ RTSP/2.0
       CSeq: 5
       User-Agent: PhonyClient/1.2
       Range: npt=0-
       Seek-Style: RAP
       Pipelined-Requests: 7654
 M->C: RTSP/2.0 200 OK
       CSeq: 3
       Server: PhonyServer/1.0
       Transport: RTP/SAVP;unicast;
          dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
          src_addr="198.51.100.5:9000"/"198.51.100.5:9001";
          ssrc=93CB001E;
          MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD0x
       Session: OccldOFFq23KwjYpAnBbUr
       Expires: Fri, 20 Dec 2013 12:25:34 +0000
       Date: Fri, 20 Dec 2013 10:25:34 +0000
       Accept-Ranges: npt
       Pipelined-Requests: 7654
       Media-Properties: Random-Access=0.2, Immutable, Unlimited
 M->C: RTSP/2.0 200 OK
       CSeq: 4
       Server: PhonyServer/1.0
       Transport: RTP/SAVP;unicast;
          dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
          src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
          ssrc=A813FC13;
          MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD00
       Session: OccldOFFq23KwjYpAnBbUr
       Expires: Fri, 20 Dec 2013 12:25:34 +0000
       Date: Fri, 20 Dec 2013 10:25:34 +0000
       Accept-Range: NPT
       Pipelined-Requests: 7654
       Media-Properties: Random-Access=0.8, Immutable, Unlimited

Schulzrinne, et al. Standards Track [Page 256] RFC 7826 RTSP 2.0 December 2016

 M->C: RTSP/2.0 200 OK
       CSeq: 5
       Server: PhonyServer/1.0
       Date: Fri, 20 Dec 2013 10:25:34 +0000
       Session: OccldOFFq23KwjYpAnBbUr
       Range: npt=0-623.10
       Seek-Style: RAP
       RTP-Info: url="rtsps://example.com/twister.3gp/trackID=4"
          ssrc=0D12F123:seq=12345;rtptime=3450012,
         url="rtsps://example.com/twister.3gp/trackID=1"
          ssrc=4F312DD8:seq=54321;rtptime=2876889;
       Pipelined-Requests: 7654

A.4. Media on Demand (Unicast)

 An alternative example of media on demand with a few more tweaks is
 the following.  Client C requests a movie distributed from two
 different media servers A (audio.example.com) and V
 (video.example.com).  The media description is stored on a web server
 W.  The media description contains descriptions of the presentation
 and all its streams, including the codecs that are available and the
 protocol stack.
 In this example, the client is only interested in the last part of
 the movie.
 C->W: GET /twister.sdp HTTP/1.1
       Host: www.example.com
       Accept: application/sdp
 W->C: HTTP/1.1 200 OK
       Date: Wed, 23 Jan 2013 15:35:06 GMT
       Content-Type: application/sdp
       Content-Length: 278
       Expires: Thu, 24 Jan 2013 15:35:06 GMT
       v=0
       o=- 2890844526 2890842807 IN IP4 198.51.100.5
       s=RTSP Session
       e=adm@example.com
       c=IN IP4 0.0.0.0
       a=range:npt=00:00:00-01:49:34
       t=0 0
       m=audio 0 RTP/AVP 0
       a=control:rtsp://audio.example.com/twister/audio.en
       m=video 0 RTP/AVP 31
       a=control:rtsp://video.example.com/twister/video

Schulzrinne, et al. Standards Track [Page 257] RFC 7826 RTSP 2.0 December 2016

 C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/2.0
       CSeq: 1
       User-Agent: PhonyClient/1.2
       Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",
                  RTP/AVP/TCP;unicast;interleaved=0-1
       Accept-Ranges: npt, smpte, clock
 A->C: RTSP/2.0 200 OK
       CSeq: 1
       Session: OccldOFFq23KwjYpAnBbUr
       Transport: RTP/AVP/UDP;unicast;
                  dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
                  src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
       Date: Wed, 23 Jan 2013 15:35:12 +0000
       Server: PhonyServer/1.0
       Expires: Thu, 24 Jan 2013 15:35:12 +0000
       Cache-Control: public
       Accept-Ranges: npt, smpte
       Media-Properties: Random-Access=0.02, Immutable, Unlimited
 C->V: SETUP rtsp://video.example.com/twister/video RTSP/2.0
       CSeq: 1
       User-Agent: PhonyClient/1.2
       Transport: RTP/AVP/UDP;unicast;
                  dest_addr="192.0.2.53:3058"/"192.0.2.53:3059",
                  RTP/AVP/TCP;unicast;interleaved=0-1
       Accept-Ranges: npt, smpte, clock

Schulzrinne, et al. Standards Track [Page 258] RFC 7826 RTSP 2.0 December 2016

 V->C: RTSP/2.0 200 OK
       CSeq: 1
       Session: P5it3pMo6xHkjUcDrNkBjf
       Transport: RTP/AVP/UDP;unicast;
          dest_addr="192.0.2.53:3058"/"192.0.2.53:3059";
          src_addr="198.51.100.5:5002"/"198.51.100.5:5003"
       Date: Wed, 23 Jan 2013 15:35:12 +0000
       Server: PhonyServer/1.0
       Cache-Control: public
       Expires: Thu, 24 Jan 2013 15:35:12 +0000
       Accept-Ranges: npt, smpte
       Media-Properties: Random-Access=1.2, Immutable, Unlimited
 C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0
       CSeq: 2
       User-Agent: PhonyClient/1.2
       Session: P5it3pMo6xHkjUcDrNkBjf
       Range: smpte=0:10:00-
 V->C: RTSP/2.0 200 OK
       CSeq: 2
       Session: P5it3pMo6xHkjUcDrNkBjf
       Range: smpte=0:10:00-1:49:23
       Seek-Style: First-Prior
       RTP-Info: url="rtsp://video.example.com/twister/video"
                 ssrc=A17E189D:seq=12312232;rtptime=78712811
       Server: PhonyServer/2.0
       Date: Wed, 23 Jan 2013 15:35:13 +0000
 C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/2.0
       CSeq: 2
       User-Agent: PhonyClient/1.2
       Session: OccldOFFq23KwjYpAnBbUr
       Range: smpte=0:10:00-
 A->C: RTSP/2.0 200 OK
       CSeq: 2
       Session: OccldOFFq23KwjYpAnBbUr
       Range: smpte=0:10:00-1:49:23
       Seek-Style: First-Prior
       RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
                 ssrc=3D124F01:seq=876655;rtptime=1032181
       Server: PhonyServer/1.0
       Date: Wed, 23 Jan 2013 15:35:13 +0000

Schulzrinne, et al. Standards Track [Page 259] RFC 7826 RTSP 2.0 December 2016

 C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/2.0
       CSeq: 3
       User-Agent: PhonyClient/1.2
       Session: OccldOFFq23KwjYpAnBbUr
 A->C: RTSP/2.0 200 OK
       CSeq: 3
       Server: PhonyServer/1.0
       Date: Wed, 23 Jan 2013 15:36:52 +0000
 C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/2.0
       CSeq: 3
       User-Agent: PhonyClient/1.2
       Session: P5it3pMo6xHkjUcDrNkBjf
 V->C: RTSP/2.0 200 OK
       CSeq: 3
       Server: PhonyServer/2.0
       Date: Wed, 23 Jan 2013 15:36:52 +0000
 Even though the audio and video track are on two different servers
 that may start at slightly different times and may drift with respect
 to each other over time, the client can perform initial
 synchronization of the two media using RTP-Info and Range received in
 the PLAY responses.  If the two servers are time synchronized, the
 RTCP packets can also be used to maintain synchronization.

A.5. Single-Stream Container Files

 Some RTSP servers may treat all files as though they are "container
 files", yet other servers may not support such a concept.  Because of
 this, clients needs to use the rules set forth in the session
 description for Request-URIs rather than assuming that a consistent
 URI may always be used throughout.  Below is an example of how a
 multi-stream server might expect a single-stream file to be served:

Schulzrinne, et al. Standards Track [Page 260] RFC 7826 RTSP 2.0 December 2016

 C->S: DESCRIBE rtsp://foo.example.com/test.wav RTSP/2.0
       Accept: application/x-rtsp-mh, application/sdp
       CSeq: 1
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 1
       Content-base: rtsp://foo.example.com/test.wav/
       Content-type: application/sdp
       Content-length: 163
       Server: PhonyServer/1.0
       Date: Wed, 23 Jan 2013 15:36:52 +0000
       Expires: Thu, 24 Jan 2013 15:36:52 +0000
       v=0
       o=- 872653257 872653257 IN IP4 192.0.2.5
       s=mu-law wave file
       i=audio test
       c=IN IP4 0.0.0.0
       t=0 0
       a=control: *
       m=audio 0 RTP/AVP 0
       a=control:streamid=0
 C->S: SETUP rtsp://foo.example.com/test.wav/streamid=0 RTSP/2.0
       Transport: RTP/AVP/UDP;unicast;
          dest_addr=":6970"/":6971";mode="PLAY"
       CSeq: 2
       User-Agent: PhonyClient/1.2
       Accept-Ranges: npt, smpte, clock
 S->C: RTSP/2.0 200 OK
       Transport: RTP/AVP/UDP;unicast;
           dest_addr="192.0.2.53:6970"/"192.0.2.53:6971";
           src_addr="198.51.100.5:6970"/"198.51.100.5:6971";
           mode="PLAY";ssrc=EAB98712
       CSeq: 2
       Session: NYkqQYKk0bb12BY3goyoyO
       Expires: Thu, 24 Jan 2013 15:36:52 +0000
       Server: PhonyServer/1.0
       Date: Wed, 23 Jan 2013 15:36:52 +0000
       Accept-Ranges: npt
       Media-Properties: Random-Access=0.5, Immutable, Unlimited

Schulzrinne, et al. Standards Track [Page 261] RFC 7826 RTSP 2.0 December 2016

 C->S: PLAY rtsp://foo.example.com/test.wav/ RTSP/2.0
       CSeq: 3
       User-Agent: PhonyClient/1.2
       Session: NYkqQYKk0bb12BY3goyoyO
 S->C: RTSP/2.0 200 OK
       CSeq: 3
       Server: PhonyServer/1.0
       Date: Wed, 23 Jan 2013 15:36:52 +0000
       Session: NYkqQYKk0bb12BY3goyoyO
       Range: npt=0-600
       Seek-Style: RAP
       RTP-Info: url="rtsp://foo.example.com/test.wav/streamid=0"
          ssrc=0D12F123:seq=981888;rtptime=3781123
 Note the different URI in the SETUP command and then the switch back
 to the aggregate URI in the PLAY command.  This makes complete sense
 when there are multiple streams with aggregate control, but it is
 less than intuitive in the special case where the number of streams
 is one.  However, the server has declared the aggregated control URI
 in the SDP; therefore, this is legal.
 In this case, it is also required that servers accept implementations
 that use the non-aggregated interpretation and use the individual
 media URI, like this:
 C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0
       CSeq: 3
       User-Agent: PhonyClient/1.2
       Session: NYkqQYKk0bb12BY3goyoyO

Schulzrinne, et al. Standards Track [Page 262] RFC 7826 RTSP 2.0 December 2016

A.6. Live Media Presentation Using Multicast

 The media server M chooses the multicast address and port.  Here, it
 is assumed that the web server only contains a pointer to the full
 description, while the media server M maintains the full description.
 C->W: GET /sessions.html HTTP/1.1
       Host: www.example.com
 W->C: HTTP/1.1 200 OK
       Content-Type: text/html
       <html>
         ...
         <a href "rtsp://live.example.com/concert/audio">
            Streamed Live Music performance </a>
         ...
       </html>
 C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/2.0
       CSeq: 1
       Supported: play.basic, play.scale
       User-Agent: PhonyClient/1.2
 M->C: RTSP/2.0 200 OK
       CSeq: 1
       Content-Type: application/sdp
       Content-Length: 183
       Server: PhonyServer/1.0
       Date: Wed, 23 Jan 2013 15:36:52 +0000
       Supported: play.basic
       v=0
       o=- 2890844526 2890842807 IN IP4 192.0.2.5
       s=RTSP Session
       t=0 0
       m=audio 3456 RTP/AVP 0
       c=IN IP4 233.252.0.54/16
       a=control: rtsp://live.example.com/concert/audio
       a=range:npt=0-

Schulzrinne, et al. Standards Track [Page 263] RFC 7826 RTSP 2.0 December 2016

 C->M: SETUP rtsp://live.example.com/concert/audio RTSP/2.0
       CSeq: 2
       Transport: RTP/AVP;multicast;
            dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
       Accept-Ranges: npt, smpte, clock
       User-Agent: PhonyClient/1.2
 M->C: RTSP/2.0 200 OK
       CSeq: 2
       Server: PhonyServer/1.0
       Date: Wed, 23 Jan 2013 15:36:52 +0000
       Transport: RTP/AVP;multicast;
            dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
            ;ssrc=4D12AB92/0DF876A3
       Session: qHj4jidpmF6zy9v9tNbtxr
       Accept-Ranges: npt, clock
       Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0
 C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0
       CSeq: 3
       Session: qHj4jidpmF6zy9v9tNbtxr
       User-Agent: PhonyClient/1.2
 M->C: RTSP/2.0 200 OK
       CSeq: 3
       Server: PhonyServer/1.0
       Date: Wed, 23 Jan 2013 15:36:52 +0000
       Session: qHj4jidpmF6zy9v9tNbtxr
       Seek-Style: Next
       Range:npt=1256-
       RTP-Info: url="rtsp://live.example.com/concert/audio"
                 ssrc=0D12F123:seq=1473; rtptime=80000

A.7. Capability Negotiation

 This example illustrates how the client and server determine their
 capability to support a special feature, in this case, "play.scale".
 The server, through the client request and the included Supported
 header, learns that the client supports RTSP 2.0 and also supports
 the playback time scaling feature of RTSP.  The server's response
 contains the following feature-related information to the client; it
 supports the basic media delivery functions (play.basic), the
 extended functionality of time scaling of content (play.scale), and
 one "example.com" proprietary feature (com.example.flight).  The
 client also learns the methods supported (Public header) by the
 server for the indicated resource.

Schulzrinne, et al. Standards Track [Page 264] RFC 7826 RTSP 2.0 December 2016

 C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/2.0
       CSeq: 1
       Supported: play.basic, play.scale
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 1
       Public:OPTIONS,SETUP,PLAY,PAUSE,TEARDOWN,DESCRIBE,GET_PARAMETER
       Allow: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN, DESCRIBE
       Server: PhonyServer/2.0
       Supported: play.basic, play.scale, com.example.flight
 When the client sends its SETUP request, it tells the server that it
 requires support of the play.scale feature for this session by
 including the Require header.
 C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/2.0
       CSeq: 3
       User-Agent: PhonyClient/1.2
       Transport: RTP/AVP/UDP;unicast;
                  dest_addr="192.0.2.53:3056"/"192.0.2.53:3057",
                  RTP/AVP/TCP;unicast;interleaved=0-1
       Require: play.scale
       Accept-Ranges: npt, smpte, clock
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 3
       Session: OccldOFFq23KwjYpAnBbUr
       Transport: RTP/AVP/UDP;unicast;
          dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
          src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
       Server: PhonyServer/2.0
       Accept-Ranges: npt, smpte
       Media-Properties: Random-Access=0.8, Immutable, Unlimited

Appendix B. RTSP Protocol State Machine

 The RTSP session state machine describes the behavior of the protocol
 from RTSP session initialization through RTSP session termination.
 It is probably easiest to think of this as the server's state and
 then view the client as needing to track what it believes the
 server's state will be based on sent or received RTSP messages.
 Thus, in most cases, the state tables below can be read as: if the
 client does X, and assuming it fulfills any prerequisite(s), the
 (server) state will move to the new state and the indicated response
 will returned.  However, there are also server-to-client
 notifications or requests, where the action describes what

Schulzrinne, et al. Standards Track [Page 265] RFC 7826 RTSP 2.0 December 2016

 notification or request will occur, its requisites, what new state
 will result after the server has received the response, as well as
 describing the client's response to the action.
 The State machine is defined on a per-session basis, which is
 uniquely identified by the RTSP session identifier.  The session may
 contain one or more media streams depending on state.  If a single
 media stream is part of the session, it is in non-aggregated control.
 If two or more are part of the session, it is in aggregated control.
 The below state machine is an informative description of the
 protocol's behavior.  In case of ambiguity with the earlier parts of
 this specification, the description in the earlier parts take
 precedence.

B.1. States

 The state machine contains three states, described below.  For each
 state, there exists a table that shows which requests and events are
 allowed and whether they will result in a state change.
 Init: Initial state, no session exists.
 Ready:  Session is ready to start playing.
 Play: Session is playing, i.e., sending media-stream data in the
       direction S->C.

B.2. State Variables

 This representation of the state machine needs more than its state to
 work.  A small number of variables are also needed, and they are
 explained below.
 NRM:  The number of media streams that are part of this session.
 RP:   Resume point, the point in the presentation time line at which
       a request to continue playing will resume from.  A time format
       for the variable is not mandated.

B.3. Abbreviations

 To make the state tables more compact, a number of abbreviations are
 used, which are explained below.
 IFI:  IF Implemented.
 md:   Media

Schulzrinne, et al. Standards Track [Page 266] RFC 7826 RTSP 2.0 December 2016

 PP:   Pause Point, the point in the presentation timeline at which
       the presentation was paused.
 Prs:  Presentation, the complete multimedia presentation.
 RedP: Redirect Point, the point in the presentation timeline at which
       a REDIRECT was specified to occur.
 SES:  Session.

B.4. State Tables

 This section contains a table for each state.  The table contains all
 the requests and events on which this state is allowed to act.  The
 events that are method names are, unless noted, requests with the
 given method in the direction client to server (C->S).  In some
 cases, there exists one or more requisites.  The response column
 tells what type of response actions should be performed.  Possible
 actions that are requested for an event include: response codes,
 e.g., 200, headers that need to be included in the response, setting
 of state variables, or settings of other session-related parameters.
 The new state column tells which state the state machine changes to.
 The response to a valid request meeting the requisites is normally a
 2xx (SUCCESS) unless otherwise noted in the response column.  The
 exceptions need to be given a response according to the response
 column.  If the request does not meet the requisite, is erroneous, or
 some other type of error occurs, the appropriate response code is to
 be sent.  If the response code is a 4xx, the session state is
 unchanged.  A response code of 3rr will result in that the session
 being ended and its state changed to Init.  A response code of 304
 results in no state change.  However, there are restrictions to when
 a 3rr response may be used.  A 5xx response does not result in any
 change of the session state, except if the error is not possible to
 recover from.  An unrecoverable error results in the ending of the
 session.  In the general case, if it can't be determined whether or
 not it was an unrecoverable error, the client will be required to
 test.  In the case that the next request after a 5xx is responded to
 with a 454 (Session Not Found), the client knows that the session has
 ended.  For any request message that cannot be responded to within
 the time defined in Section 10.4, a 100 response must be sent.
 The server will time out the session after the period of time
 specified in the SETUP response, if no activity from the client is
 detected.  Therefore, there exists a timeout event for all states
 except Init.

Schulzrinne, et al. Standards Track [Page 267] RFC 7826 RTSP 2.0 December 2016

 In the case that NRM = 1, the presentation URI is equal to the media
 URI or a specified presentation URI.  For NRM > 1, the presentation
 URI needs to be other than any of the media that are part of the
 session.  This applies to all states.
 +---------------+-----------------+---------------------------------+
 | Event         | Prerequisite    | Response                        |
 +---------------+-----------------+---------------------------------+
 | DESCRIBE      | Needs REDIRECT  | 3rr, Redirect                   |
 |               |                 |                                 |
 | DESCRIBE      |                 | 200, Session description        |
 |               |                 |                                 |
 | OPTIONS       | Session ID      | 200, Reset session timeout      |
 |               |                 | timer                           |
 |               |                 |                                 |
 | OPTIONS       |                 | 200                             |
 |               |                 |                                 |
 | SET_PARAMETER | Valid parameter | 200, change value of parameter  |
 |               |                 |                                 |
 | GET_PARAMETER | Valid parameter | 200, return value of parameter  |
 +---------------+-----------------+---------------------------------+
              Table 9: Non-State-Machine Changing Events
 The methods in Table 9 do not have any effect on the state machine or
 the state variables.  However, some methods do change other session-
 related parameters, for example, SET_PARAMETER, which will set the
 parameter(s) specified in its body.  Also, all of these methods that
 allow the Session header will also update the keep-alive timer for
 the session.
 +------------------+----------------+-----------+-------------------+
 | Action           | Requisite      | New State | Response          |
 +------------------+----------------+-----------+-------------------+
 | SETUP            |                | Ready     | NRM=1, RP=0.0     |
 |                  |                |           |                   |
 | SETUP            | Needs Redirect | Init      | 3rr Redirect      |
 |                  |                |           |                   |
 | S -> C: REDIRECT | No Session hdr | Init      | Terminate all SES |
 +------------------+----------------+-----------+-------------------+
                         Table 10: State: Init
 The initial state of the state machine (Table 10) can only be left by
 processing a correct SETUP request.  As seen in the table, the two
 state variables are also set by a correct request.  This table also
 shows that a correct SETUP can in some cases be redirected to another
 URI or server by a 3rr response.

Schulzrinne, et al. Standards Track [Page 268] RFC 7826 RTSP 2.0 December 2016

 +-------------+------------------------+---------+------------------+
 | Action      | Requisite              | New     | Response         |
 |             |                        | State   |                  |
 +-------------+------------------------+---------+------------------+
 | SETUP       | New URI                | Ready   | NRM +=1          |
 |             |                        |         |                  |
 | SETUP       | URI Setup prior        | Ready   | Change transport |
 |             |                        |         | param            |
 |             |                        |         |                  |
 | TEARDOWN    | Prs URI,               | Init    | No session hdr,  |
 |             |                        |         | NRM = 0          |
 |             |                        |         |                  |
 | TEARDOWN    | md URI,NRM=1           | Init    | No Session hdr,  |
 |             |                        |         | NRM = 0          |
 |             |                        |         |                  |
 | TEARDOWN    | md URI,NRM>1           | Ready   | Session hdr, NRM |
 |             |                        |         | -= 1             |
 |             |                        |         |                  |
 | PLAY        | Prs URI, No range      | Play    | Play from RP     |
 |             |                        |         |                  |
 | PLAY        | Prs URI, Range         | Play    | According to     |
 |             |                        |         | range            |
 |             |                        |         |                  |
 | PLAY        | md URI, NRM=1, Range   | Play    | According to     |
 |             |                        |         | range            |
 |             |                        |         |                  |
 | PLAY        | md URI, NRM=1          | Play    | Play from RP     |
 |             |                        |         |                  |
 | PAUSE       | Prs URI                | Ready   | Return PP        |
 |             |                        |         |                  |
 | SC:REDIRECT | Terminate-Reason       | Ready   | Set RedP         |
 |             |                        |         |                  |
 | SC:REDIRECT | No Terminate-Reason    | Init    | Session is       |
 |             | time parameter         |         | removed          |
 |             |                        |         |                  |
 | Timeout     |                        | Init    |                  |
 |             |                        |         |                  |
 | RedP        |                        | Init    | TEARDOWN of      |
 | reached     |                        |         | session          |
 +-------------+------------------------+---------+------------------+
                        Table 11: State: Ready
 In the Ready state (Table 11), some of the actions depend on the
 number of media streams (NRM) in the session, i.e., aggregated or
 non-aggregated control.  A SETUP request in the Ready state can
 either add one more media stream to the session or, if the media
 stream (same URI) already is part of the session, change the

Schulzrinne, et al. Standards Track [Page 269] RFC 7826 RTSP 2.0 December 2016

 transport parameters.  TEARDOWN depends on both the Request-URI and
 the number of media streams within the session.  If the Request-URI
 is the presentation URI, the whole session is torn down.  If a media
 URI is used in the TEARDOWN request and more than one media exists in
 the session, the session will remain and a session header is returned
 in the response.  If only a single media stream remains in the
 session when performing a TEARDOWN with a media URI, the session is
 removed.  The number of media streams remaining after tearing down a
 media stream determines the new state.

Schulzrinne, et al. Standards Track [Page 270] RFC 7826 RTSP 2.0 December 2016

 +----------------+-----------------------+--------+-----------------+
 | Action         | Requisite             | New    | Response        |
 |                |                       | State  |                 |
 +----------------+-----------------------+--------+-----------------+
 | PAUSE          | Prs URI               | Ready  | Set RP to       |
 |                |                       |        | present point   |
 |                |                       |        |                 |
 | End of media   | All media             | Play   | Set RP = End of |
 |                |                       |        | media           |
 |                |                       |        |                 |
 | End of range   |                       | Play   | Set RP = End of |
 |                |                       |        | range           |
 |                |                       |        |                 |
 | PLAY           | Prs URI, No range     | Play   | Play from       |
 |                |                       |        | present point   |
 |                |                       |        |                 |
 | PLAY           | Prs URI, Range        | Play   | According to    |
 |                |                       |        | range           |
 |                |                       |        |                 |
 | SC:PLAY_NOTIFY |                       | Play   | 200             |
 |                |                       |        |                 |
 | SETUP          | New URI               | Play   | 455             |
 |                |                       |        |                 |
 | SETUP          | md URI                | Play   | 455             |
 |                |                       |        |                 |
 | SETUP          | md URI, IFI           | Play   | Change          |
 |                |                       |        | transport param.|
 |                |                       |        |                 |
 | TEARDOWN       | Prs URI               | Init   | No session hdr  |
 |                |                       |        |                 |
 | TEARDOWN       | md URI,NRM=1          | Init   | No Session hdr, |
 |                |                       |        | NRM=0           |
 |                |                       |        |                 |
 | TEARDOWN       | md URI                | Play   | 455             |
 |                |                       |        |                 |
 | SC:REDIRECT    | Terminate Reason with | Play   | Set RedP        |
 |                | Time parameter        |        |                 |
 |                |                       |        |                 |
 | SC:REDIRECT    |                       | Init   | Session is      |
 |                |                       |        | removed         |
 |                |                       |        |                 |
 | RedP reached   |                       | Init   | TEARDOWN of     |
 |                |                       |        | session         |
 |                |                       |        |                 |
 | Timeout        |                       | Init   | Stop Media      |
 |                |                       |        | playout         |
 +----------------+-----------------------+--------+-----------------+
                         Table 12: State: Play

Schulzrinne, et al. Standards Track [Page 271] RFC 7826 RTSP 2.0 December 2016

 The Play state table (Table 12) contains a number of requests that
 need a presentation URI (labeled as Prs URI) to work on (i.e., the
 presentation URI has to be used as the Request-URI).  This is due to
 the exclusion of non-aggregated stream control in sessions with more
 than one media stream.
 To avoid inconsistencies between the client and server, automatic
 state transitions are avoided.  This can be seen at, for example, an
 "End of media" event when all media has finished playing but the
 session still remains in Play state.  An explicit PAUSE request needs
 to be sent to change the state to Ready.  It may appear that there
 exist automatic transitions in "RedP reached" and "PP reached".
 However, they are requested and acknowledged before they take place.
 The time at which the transition will happen is known by looking at
 the Terminate-Reason header's time parameter and Range header,
 respectively.  If the client sends a request close in time to these
 transitions, it needs to be prepared for receiving error messages, as
 the state may or may not have changed.

Appendix C. Media-Transport Alternatives

 This section defines how certain combinations of protocols, profiles,
 and lower transports are used.  This includes the usage of the
 Transport header's source and destination address parameters:
 "src_addr" and "dest_addr".

C.1. RTP

 This section defines the interaction of RTSP with respect to the RTP
 protocol [RFC3550].  It also defines any necessary media-transport
 signaling with regard to RTP.
 The available RTP profiles and lower-layer transports are described
 below along with rules on signaling the available combinations.

C.1.1. AVP

 The usage of the "RTP Profile for Audio and Video Conferences with
 Minimal Control" [RFC3551] when using RTP for media transport over
 different lower-layer transport protocols is defined below in regard
 to RTSP.
 One such case is defined within this document: the use of embedded
 (interleaved) binary data as defined in Section 14.  The usage of
 this method is indicated by including the "interleaved" parameter.

Schulzrinne, et al. Standards Track [Page 272] RFC 7826 RTSP 2.0 December 2016

 When using embedded binary data, "src_addr" and "dest_addr" MUST NOT
 be used.  This addressing and multiplexing is used as defined with
 use of channel numbers and the interleaved parameter.

C.1.2. AVP/UDP

 This part describes the sending of RTP [RFC3550] over lower-
 transport-layer UDP [RFC768] according to the profile "RTP Profile
 for Audio and Video Conferences with Minimal Control" defined in
 [RFC3551].  Implementations of RTP/AVP/UDP MUST implement RTCP
 (Appendix C.1.6).  This profile requires one or two unidirectional or
 bidirectional UDP flows per media stream.  The first UDP flow is for
 RTP and the second is for RTCP.  Multiplexing of RTP and RTCP
 (Appendix C.1.6.4) MAY be used, in which case, a single UDP flow is
 used for both parts.  Embedding of RTP data with the RTSP messages,
 in accordance with Section 14, SHOULD NOT be performed when RTSP
 messages are transported over unreliable transport protocols, like
 UDP [RFC768].
 The RTP/UDP and RTCP/UDP flows can be established using the Transport
 header's "src_addr" and "dest_addr" parameters.
 In RTSP PLAY mode, the transmission of RTP packets from client to
 server is unspecified.  The behavior in regard to such RTP packets
 MAY be defined in future.
 The "src_addr" and "dest_addr" parameters are used in the following
 way for media delivery and playback mode, i.e., Mode=PLAY:
 o  The "src_addr" and "dest_addr" parameters MUST contain either 1 or
    2 address specifications.  Note that two address specifications
    MAY be provided even if RTP and RTCP multiplexing is negotiated.
 o  Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST
    contain either:
  • both an address and a port number, or
  • a port number without an address.
 o  The first address specification given in either of the parameters
    applies to the RTP stream.  The second specification, if present,
    applies to the RTCP stream, unless in the case RTP and RTCP
    multiplexing is negotiated where both RTP and RTCP will use the
    first specification.

Schulzrinne, et al. Standards Track [Page 273] RFC 7826 RTSP 2.0 December 2016

 o  The RTP/UDP packets from the server to the client MUST be sent to
    the address and port given by the first address specification of
    the "dest_addr" parameter.
 o  The RTCP/UDP packets from the server to the client MUST be sent to
    the address and port given by the second address specification of
    the "dest_addr" parameter, unless RTP and RTCP multiplexing has
    been negotiated, in which case RTCP MUST be sent to the first
    address specification.  If no second pair is specified and RTP and
    RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.
 o  The RTCP/UDP packets from the client to the server MUST be sent to
    the address and port given by the second address specification of
    the "src_addr" parameter, unless RTP and RTCP multiplexing has
    been negotiated, in which case RTCP MUST be sent to the first
    address specification.  If no second pair is specified and RTP and
    RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.
 o  The RTP/UDP packets from the client to the server MUST be sent to
    the address and port given by the first address specification of
    the "src_addr" parameter.
 o  RTP and RTCP packets SHOULD be sent from the corresponding
    receiver port, i.e., RTCP packets from the server should be sent
    from the "src_addr" parameters second address port pair, unless
    RTP and RTCP multiplexing has been negotiated in which case the
    first address port pair is used.

C.1.3. AVPF/UDP

 The RTP profile "Extended RTP Profile for RTCP-based Feedback (RTP/
 AVPF)" [RFC4585] MAY be used as RTP profiles in sessions using RTP.
 All that is defined for AVP MUST also apply for AVPF.
 The usage of AVPF is indicated by the media initialization protocol
 used.  In the case of SDP, it is indicated by media lines ("m=")
 containing the profile RTP/AVPF.  That SDP MAY also contain further
 AVPF-related SDP attributes configuring the AVPF session regarding
 reporting interval and feedback messages to be used [RFC4585].  This
 configuration MUST be followed.

Schulzrinne, et al. Standards Track [Page 274] RFC 7826 RTSP 2.0 December 2016

C.1.4. SAVP/UDP

 The RTP profile "The Secure Real-time Transport Protocol (SRTP)"
 [RFC3711] is an RTP profile (SAVP) that MAY be used in RTSP sessions
 using RTP.  All that is defined for AVP MUST also apply for SAVP.
 The usage of SRTP requires that a security context be established.
 The default key-management unless otherwise signaled SHALL be MIKEY
 in RSA-R mode as defined in Appendix C.1.4.1 and not according to the
 procedure defined in "Key Management Extensions for Session
 Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)"
 [RFC4567].  The reason is that RFC 4567 sends the initial MIKEY
 message in SDP, thus, both requiring the usage of the DESCRIBE method
 and forcing the server to keep state for clients performing DESCRIBE
 in anticipation that they might require key management.
 MIKEY is selected as the default method for establishing SRTP
 cryptographic context within an RTSP session as it can be embedded in
 the RTSP messages while still ensuring confidentiality of content of
 the keying material, even when using hop-by-hop TLS security for the
 RTSP messages.  This method also supports pipelining of the RTSP
 messages.

C.1.4.1. MIKEY Key Establishment

 This method for using MIKEY [RFC3830] to establish the SRTP
 cryptographic context is initiated in the client's SETUP request, and
 the server's response to the SETUP carries the MIKEY response.  This
 ensures that the crypto context establishment happens simultaneously
 with the establishment of the media stream being protected.  By using
 MIKEY's RSA-R mode [RFC4738] the client can be the initiator and
 still allow the server to set the parameters in accordance with the
 actual media stream.
 The SRTP cryptographic context establishment is done according to the
 following process:
 1.   The client determines that SAVP or SAVPF shall be used from the
      media-description format, e.g., SDP.  If no other key-management
      method is explicitly signaled, then MIKEY SHALL be used as
      defined herein.  The use of SRTP with RTSP is only defined with
      MIKEY with keys established as defined in this section.  Future
      documents may define how an RTSP implementation treats SDP that
      indicates some other key mechanism to be used.  The need for
      such specification includes [RFC4567], which is not defined for
      use in RTSP 2.0 within this document.

Schulzrinne, et al. Standards Track [Page 275] RFC 7826 RTSP 2.0 December 2016

 2.   The client SHALL establish a TLS connection for RTSP messages,
      directly or hop-by-hop with the server.  If hop-by-hop TLS
      security is used, the User method SHALL be indicated in the
      Accept-Credentials header.  Note that using hop-by-hop does
      allow the proxy to insert itself as a man in the middle.  This
      can also occur in the MIKEY exchange by the proxy providing one
      of its certificates rather than the server's in the Connection-
      Credentials header.  Therefore, the client SHALL validate the
      server certificate.
 3.   The client retrieves the server's certificate from a direct TLS
      connection or hop-by-hop from a Connection-Credentials header.
      The client then checks that the server certificate is valid and
      belongs to the server.
 4.   The client forms the MIKEY Initiator message using RSA-R mode in
      unicast mode as specified in [RFC4738].  The client SHOULD use
      the same certificate for TLS and MIKEY to enable the server to
      bind the two together.  The client's certificate SHALL be
      included in the MIKEY message.  The client SHALL indicate its
      SRTP capabilities in the message.
 5.   The MIKEY message from the previous step is base64-encoded
      [RFC4648] and becomes the value of the MIKEY parameter that is
      included in the transport specification(s) that specifies an
      SRTP-based profile (SAVP, SAVPF) in the SETUP request.
 6.   Any proxy encountering the MIKEY parameter SHALL forward it
      without modification.  A proxy that is required to understand
      the Transport specifications will need to understand SAVP/SAVPF
      with MIKEY to enable the default keying for SRTP-protected media
      streams.  If such a proxy does not support SAVP/SAVPF with
      MIKEY, it will discard the whole transport specification.  Most
      types of proxies can easily support SAVP and SAVPF with MIKEY.
      If a client encounters a proxy not supporting SAVP/SAVPF with
      MIKEY, the client should attempt bypassing that proxy.
 7.   The server, upon receiving the SETUP request, will need to
      decide upon the transport specification to use, if multiple are
      included by the client.  In the determination of which transport
      specifications are supported and preferred, the server SHOULD
      decode the MIKEY message to take the embedded SRTP parameters
      into account.  If all transport spec require SRTP but no MIKEY
      parameter or other supported keying method is included, the
      server SHALL respond with 403 (Forbidden).

Schulzrinne, et al. Standards Track [Page 276] RFC 7826 RTSP 2.0 December 2016

 8.   Upon generating a response, the following outcomes can occur:
  • A transport spec not using SRTP and MIKEY is selected. Thus,

the response will not contain any MIKEY parameters.

  • A transport spec using SRTP and MIKEY is selected but an

error is encountered in the MIKEY processing. In this case,

         an RTSP error response code of 466 (Key Management Error)
         SHALL be used.  A MIKEY message describing the error MAY be
         included.
  • A transport spec using SRTP and MIKEY is selected and a MIKEY

response message can be created. The server SHOULD use the

         same certificate for TLS and in MIKEY to enable the client to
         bind the two together.  If a different certificate is used,
         it SHALL be included in the MIKEY message.  It is RECOMMENDED
         that the envelope key-cache type be set to 'Cache' and that a
         single envelope key is reused for all MIKEY messages to the
         client.  That message is included in the MIKEY parameter part
         of the single selected transport specification in the SETUP
         response.  The server will set the SRTP parameters as
         preferred for this media stream within the supported range by
         the client.
 9.   The server transmits the SETUP response back to the client.
 10.  The client receives the SETUP response and, if the response code
      indicates a successful request, it decodes the MIKEY message and
      establishes the SRTP cryptographic context from the parameters
      in the MIKEY response.
 In the above method, the client's certificate may be self signed in
 cases where the client's identity is not necessary to authenticate
 and the security goal is only to ensure that the RTSP signaling
 client is the same as the one receiving the SRTP security context.

C.1.5. SAVPF/UDP

 The RTP profile "Extended Secure RTP Profile for Real-time Transport
 Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124] is an
 RTP profile (SAVPF) that MAY be used in RTSP sessions using RTP.  All
 that is defined for AVPF MUST also apply for SAVPF.
 The usage of SRTP requires that a cryptographic context be
 established.  The default mechanism for establishing that security
 association is to use MIKEY[RFC3830] with RTSP as defined in
 Appendix C.1.4.1.

Schulzrinne, et al. Standards Track [Page 277] RFC 7826 RTSP 2.0 December 2016

C.1.6. RTCP Usage with RTSP

 RTCP has several usages when RTP is implemented for media transport
 as explained below.  Thus, RTCP MUST be supported if an RTSP agent
 handles RTP.

C.1.6.1. Media Synchronization

 RTCP provides media synchronization and clock-drift compensation.
 The initial media synchronization is available from RTP-Info header.
 However, to be able to handle any clock drift between the media
 streams, RTCP is needed.

C.1.6.2. RTSP Session Keep-Alive

 RTCP traffic from the RTSP client to the RTSP server MUST function as
 keep-alive.  This requires an RTSP server supporting RTP to use the
 received RTCP packets as indications that the client desires the
 related RTSP session to be kept alive.

C.1.6.3. Bitrate Adaption

 RTCP Receiver reports and any additional feedback from the client
 MUST be used to adapt the bitrate used over the transport for all
 cases when RTP is sent over UDP.  An RTP sender without reserved
 resources MUST NOT use more than its fair share of the available
 resources.  This can be determined by comparing on short-to-medium
 terms (some seconds) the used bitrate and adapting it so that the RTP
 sender sends at a bitrate comparable to what a TCP sender would
 achieve on average over the same path.
 To ensure that the implementation's adaptation mechanism has a well-
 defined outer envelope, all implementations using a non-congestion-
 controlled unicast transport protocol, like UDP, MUST implement
 "Multimedia Congestion Control: Circuit Breakers for Unicast RTP
 Sessions" [RTP-CIRCUIT-BREAKERS].

C.1.6.4. RTP and RTCP Multiplexing

 RTSP can be used to negotiate the usage of RTP and RTCP multiplexing
 as described in [RFC5761].  This allows servers and client to reduce
 the amount of resources required for the session by only requiring
 one underlying transport stream per media stream instead of two when
 using RTP and RTCP.  This lessens the server-port consumption and
 also the necessary state and keep-alive work when operating across
 NATs [RFC2663].

Schulzrinne, et al. Standards Track [Page 278] RFC 7826 RTSP 2.0 December 2016

 Content must be prepared with some consideration for RTP and RTCP
 multiplexing, mainly ensuring that the RTP payload types used do not
 collide with the ones used for RTCP packet types.  This option likely
 needs explicit support from the content unless the RTP payload types
 can be remapped by the server and that is correctly reflected in the
 session description.  Beyond that, support of this feature should
 come at little cost and much gain.
 It is recommended that, if the content and server support RTP and
 RTCP multiplexing, this is indicated in the session description, for
 example, using the SDP attribute "a=rtcp-mux".  If the SDP message
 contains the "a=rtcp-mux" attribute for a media stream, the server
 MUST support RTP and RTCP multiplexing.  If indicated or otherwise
 desired by the client, it can include the Transport parameter "RTCP-
 mux" in any transport specification where it desires to use "RTCP-
 mux".  The server will indicate if it supports "RTCP-mux".  Servers
 and Clients SHOULD support RTP and RTCP multiplexing.
 For capability exchange, an RTSP feature tag for RTP and RTCP
 multiplexing is defined: "setup.rtp.rtcp.mux".
 To minimize the risk of negotiation failure while using RTP and RTCP
 multiplexing, some recommendations are here provided.  If the session
 description includes explicit indication of support ("a=rtcp-mux" in
 SDP), then an RTSP agent can safely create a SETUP request with a
 transport specification with only a single "dest_addr" parameter
 address specification.  If no such explicit indication is provided,
 then even if the feature tag "setup.rtp.rtcp.mux" is provided in a
 Supported header by the RTSP server or the feature tag included in
 the Required header in the SETUP request, the media resource may not
 support RTP and RTCP multiplexing.  Thus, to maximize the probability
 of successful negotiation, the RTSP agent is recommended to include
 two "dest_addr" parameter address specifications in the first or
 first set (if pipelining is used) of SETUP request(s) for any media
 resource aggregate.  That way, the RTSP server can accept RTP and
 RTCP multiplexing and only use the first address specification or, if
 not, use both specifications.  The RTSP agent, after having received
 the response for a successful negotiation of the usage of RTP and
 RTCP multiplexing, can then release the resources associated with the
 second address specification.

C.2. RTP over TCP

 Transport of RTP over TCP can be done in two ways: over independent
 TCP connections using [RFC4571] or interleaved in the RTSP
 connection.  In both cases, the protocol MUST be "rtp" and the lower-
 layer MUST be TCP.  The profile may be any of the above specified
 ones: AVP, AVPF, SAVP, or SAVPF.

Schulzrinne, et al. Standards Track [Page 279] RFC 7826 RTSP 2.0 December 2016

C.2.1. Interleaved RTP over TCP

 The use of embedded (interleaved) binary data transported on the RTSP
 connection is possible as specified in Section 14.  When using this
 declared combination of interleaved binary data, the RTSP messages
 MUST be transported over TCP.  TLS may or may not be used.  If TLS is
 used, both RTSP messages and the binary data will be protected by
 TLS.
 One should, however, consider that this will result in all media
 streams going through any proxy.  Using independent TCP connections
 can avoid that issue.

C.2.2. RTP over Independent TCP

 In this section, the sending of RTP [RFC3550] over lower-layer
 transport TCP [RFC793] according to "Framing Real-time Transport
 Protocol (RTP) and RTP Control Protocol (RTCP) Packets over
 Connection-Oriented Transport" [RFC4571] is described.  This section
 adapts the guidelines for using RTP over TCP within SIP/SDP [RFC4145]
 to work with RTSP.
 A client codes the support of RTP over independent TCP by specifying
 an RTP/AVP/TCP transport option without an interleaved parameter in
 the Transport line of a SETUP request.  This transport option MUST
 include the "unicast" parameter.
 If the client wishes to use RTP with RTCP, two address specifications
 need to be included in the "dest_addr" parameter.  If the client
 wishes to use RTP without RTCP, one address specification is included
 in the "dest_addr" parameter.  If the client wishes to multiplex RTP
 and RTCP on a single transport flow (see Appendix C.1.6.4), one or
 two address specifications are included in the "dest_addr" parameter
 in addition to the "RTCP-mux" transport parameter.  Two address
 specifications are allowed to facilitate successful negotiation when
 the server or content can't support RTP and RTCP multiplexing.
 Ordering rules of dest_addr ports follow the rules for RTP/AVP/UDP.
 If the client wishes to play the active role in initiating the TCP
 connection, it MAY set the setup parameter (see Section 18.54) on the
 Transport line to be "active", or it MAY omit the setup parameter, as
 active is the default.  If the client signals the active role, the
 ports in the address specifications in the "dest_addr" parameter MUST
 be set to 9 (the discard port).
 If the client wishes to play the passive role in TCP connection
 initiation, it MUST set the setup parameter on the Transport line to
 be "passive".  If the client is able to assume the active or the

Schulzrinne, et al. Standards Track [Page 280] RFC 7826 RTSP 2.0 December 2016

 passive role, it MUST set the setup parameter on the Transport line
 to be "actpass".  In either case, the "dest_addr" parameter's address
 specification port value for RTP MUST be set to the TCP port number
 on which the client is expecting to receive the TCP connection for
 RTP, and the "dest_addr" address specification port value for RTCP
 MUST be set to the TCP port number on which the client is expecting
 to receive the TCP connection for RTCP.  In the case that the client
 wishes to multiplex RTP and RTCP on a single transport flow, the
 "RTCP-mux" parameter is included and one or two "dest_addr" parameter
 address specifications are included, as mentioned earlier in this
 section.
 Upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, if a
 server decides to accept this requested option, the 2xx reply MUST
 contain a Transport option that specifies RTP/AVP/TCP (without using
 the interleaved parameter and using the unicast parameter).  The
 "dest_addr" parameter value MUST be echoed from the parameter value
 in the client request unless the destination address (only port) was
 not provided; in which case, the server MAY include the source
 address of the RTSP TCP connection with the port number unchanged.
 In addition, the server reply MUST set the setup parameter on the
 Transport line, to indicate the role the server will play in the
 connection setup.  Permissible values are "active" (if a client set
 setup to "passive" or "actpass") and "passive" (if a client set setup
 to "active" or "actpass").
 If a server sets setup to "passive", the "src_addr" in the reply MUST
 indicate the ports on which the server is willing to receive a TCP
 connection for RTP and (if the client requested a TCP connection for
 RTCP by specifying two "dest_addr" address specifications) a TCP/
 RTCP connection.  If a server sets setup to "active", the ports
 specified in "src_addr" address specifications MUST be set to 9.  The
 server MAY use the "ssrc" parameter, following the guidance in
 Section 18.54.  The server sets only one address specification in the
 case that the client has indicated only a single address
 specification or in case RTP and RTCP multiplexing was requested and
 accepted by the server.  Port ordering for "src_addr" follows the
 rules for RTP/AVP/UDP.
 Servers MUST support taking the passive role and MAY support taking
 the active role.  Servers with a public IP address take the passive
 role, thus enabling clients behind NATs and firewalls a better chance
 of successful connect to the server by actively connecting outwards.
 Therefore, the clients are RECOMMENDED to take the active role.

Schulzrinne, et al. Standards Track [Page 281] RFC 7826 RTSP 2.0 December 2016

 After sending (receiving) a 2xx reply for a SETUP method for a non-
 interleaved RTP/AVP/TCP media stream, the active party SHOULD
 initiate the TCP connection as soon as possible.  The client MUST NOT
 send a PLAY request prior to the establishment of all the TCP
 connections negotiated using SETUP for the session.  In case the
 server receives a PLAY request in a session that has not yet
 established all the TCP connections, it MUST respond using the 464
 (Data Transport Not Ready Yet) (Section 17.4.28) error code.
 Once the PLAY request for a media resource transported over non-
 interleaved RTP/AVP/TCP occurs, media begins to flow from server to
 client over the RTP TCP connection, and RTCP packets flow
 bidirectionally over the RTCP TCP connection.  Unless RTP and RTCP
 multiplexing has been negotiated; in which case, RTP and RTCP will
 flow over a common TCP connection.  As in the RTP/UDP case, client-
 to-server traffic on an RTP-only TCP session is unspecified by this
 memo.  The packets that travel on these connections MUST be framed
 using the protocol defined in [RFC4571], not by the framing defined
 for interleaving RTP over the RTSP connection defined in Section 14.
 A successful PAUSE request for media being transported over RTP/AVP/
 TCP pauses the flow of packets over the connections, without closing
 the connections.  A successful TEARDOWN request signals that the TCP
 connections for RTP and RTCP are to be closed by the RTSP client as
 soon as possible.
 Subsequent SETUP requests using a URI already set up in an RTSP
 session using an RTP/AVP/TCP transport specification may be ambiguous
 in the following way: does the client wish to open up a new TCP
 connection for RTP or RTCP for the URI, or does the client wish to
 continue using the existing TCP connections?  The client SHOULD use
 the "connection" parameter (defined in Section 18.54) on the
 Transport line to make its intention clear (by setting "connection"
 to "new" if new connections are needed, and by setting "connection"
 to "existing" if the existing connections are to be used).  After a
 2xx reply for a SETUP request for a new connection, parties should
 close the preexisting connections, after waiting a suitable period
 for any stray RTP or RTCP packets to arrive.
 The usage of SRTP, i.e., either SAVP or SAVPF profiles, requires that
 a security association be established.  The default mechanism for
 establishing that security association is to use MIKEY[RFC3830] with
 RTSP as defined Appendix C.1.4.1.

Schulzrinne, et al. Standards Track [Page 282] RFC 7826 RTSP 2.0 December 2016

 Below, a rewritten version of the example "Media on Demand"
 (Appendix A.1) shows the use of RTP/AVP/TCP non-interleaved:
    C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
          CSeq: 1
          User-Agent: PhonyClient/1.2
    M->C: RTSP/2.0 200 OK
          CSeq: 1
          Server: PhonyServer/1.0
          Date: Wed, 23 Jan 2013 15:36:52 +0000
          Content-Type: application/sdp
          Content-Length: 227
          Content-Base: rtsp://example.com/twister.3gp/
          Expires: Thu, 24 Jan 2013 15:36:52 +0000
          v=0
          o=- 2890844256 2890842807 IN IP4 198.51.100.34
          s=RTSP Session
          i=An Example of RTSP Session Usage
          e=adm@example.com
          c=IN IP4 0.0.0.0
          a=control: *
          a=range:npt=00:00:00-00:10:34.10
          t=0 0
          m=audio 0 RTP/AVP 0
          a=control: trackID=1
    C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
          CSeq: 2
          User-Agent: PhonyClient/1.2
          Require: play.basic
          Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9";
                     setup=active;connection=new
          Accept-Ranges: npt, smpte, clock
    M->C: RTSP/2.0 200 OK
          CSeq: 2
          Server: PhonyServer/1.0
          Transport: RTP/AVP/TCP;unicast;
                     dest_addr=":9"/":9";
                     src_addr="198.51.100.5:53478"/"198.51.100:54091";
                     setup=passive;connection=new;ssrc=93CB001E
          Session: OccldOFFq23KwjYpAnBbUr
          Expires: Thu, 24 Jan 2013 15:36:52 +0000
          Date: Wed, 23 Jan 2013 15:36:52 +0000
          Accept-Ranges: npt
          Media-Properties: Random-Access=0.8, Immutable, Unlimited

Schulzrinne, et al. Standards Track [Page 283] RFC 7826 RTSP 2.0 December 2016

    C->M: TCP Connection Establishment x2
    C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
          CSeq: 4
          User-Agent: PhonyClient/1.2
          Range: npt=30-
          Session: OccldOFFq23KwjYpAnBbUr
    M->C: RTSP/2.0 200 OK
          CSeq: 4
          Server: PhonyServer/1.0
          Date: Wed, 23 Jan 2013 15:36:54 +0000
          Session: OccldOFFq23KwjYpAnBbUr
          Range: npt=30-623.10
          Seek-Style: First-Prior
          RTP-Info:  url="rtsp://example.com/twister.3gp/trackID=1"
             ssrc=4F312DD8:seq=54321;rtptime=2876889

C.3. Handling Media-Clock Time Jumps in the RTP Media Layer

 RTSP allows media clients to control selected, non-contiguous
 sections of media presentations, rendering those streams with an RTP
 media layer [RFC3550].  Two cases occur, the first is when a new PLAY
 request replaces an old ongoing request and the new request results
 in a jump in the media.  This should produce continuous media stream
 at the RTP layer.  A client may also immediately follow a completed
 PLAY request with a new PLAY request.  This will result in some gap
 in the media layer.  The below text will look into both cases.
 A PLAY request that replaces an ongoing PLAY request allows the media
 layer rendering the RTP stream to do so continuously without being
 affected by jumps in media-clock time.  The RTP timestamps for the
 new media range are set so that they become continuous with the
 previous media range in the previous request.  The RTP sequence
 number for the first packet in the new range will be the next
 following the last packet in the previous range, i.e., monotonically
 increasing.  The goal is to allow the media-rendering layer to work
 without interruption or reconfiguration across the jumps in media
 clock.  This should be possible in all cases of replaced PLAY
 requests for media that has random access properties.  In this case,
 care is needed to align frames or similar media-dependent structures.
 In cases where jumps in media-clock time are a result of RTSP
 signaling operations arriving after a completed PLAY operation, the
 request timing will result in that media becoming non-continuous.
 The server becomes unable to send the media so that it arrives timely
 and still carries timestamps to make the media stream continuous.  In
 these situations, the server will produce RTP streams where there are

Schulzrinne, et al. Standards Track [Page 284] RFC 7826 RTSP 2.0 December 2016

 gaps in the RTP timeline for the media.  If the media has frame
 structure, aligning the timestamp for the next frame with the
 previous structure reduces the burden to render this media.  The gap
 should represent the time the server hasn't been serving media, e.g.,
 the time between the end of the media stream or a PAUSE request and
 the new PLAY request.  In these cases, the RTP sequence number would
 normally be monotonically increasing across the gap.
 For RTSP sessions with media that lacks random access properties,
 such as live streams, any media-clock jump is commonly the result of
 a correspondingly long pause of delivery.  The RTP timestamp will
 have increased in direct proportion to the duration of the paused
 delivery.  Note also that in this case the RTP sequence number should
 be the next packet number.  If not, the RTCP packet loss reporting
 will indicate as loss all packets not received between the point of
 pausing and later resuming.  This may trigger congestion avoidance
 mechanisms.  An allowed exception from the above recommendation on
 monotonically increasing RTP sequence number is live media streams,
 likely being relayed.  In this case, when the client resumes
 delivery, it will get the media that is currently being delivered to
 the server itself.  For this type of basic delivery of live streams
 to multiple users over unicast, individual rewriting of RTP sequence
 numbers becomes quite a burden.  For solutions that already cache
 media or perform time shifting, the rewriting should impose only a
 minor burden.
 The goal when handling jumps in media-clock time is that the provided
 stream is continuous without gaps in RTP timestamp or sequence
 number.  However, when delivery has been halted for some reason, the
 RTP timestamp, when resuming, MUST represent the duration that the
 delivery was halted.  An RTP sequence number MUST generally be the
 next number, i.e., monotonically increasing modulo 65536.  For media
 resources with the properties Time-Progressing and Time-Duration=0.0,
 the server MAY create RTP media streams with RTP sequence number
 jumps in them due to the client first halting delivery and later
 resuming it (PAUSE and then later PLAY).  However, servers utilizing
 this exception must take into consideration the resulting RTCP
 receiver reports that likely contain loss reports for all the packets
 that were a part of the discontinuity.  A client cannot rely on the
 fact that a server will align when resuming play, even if it is
 RECOMMENDED.  The RTP-Info header will provide information on how the
 server acts in each case.
    One cannot assume that the RTSP client can communicate with the
    RTP media agent, as the two may be independent processes.  If the
    RTP timestamp shows the same gap as the NPT, the media agent will
    assume that there is a pause in the presentation.  If the jump in
    NPT is large enough, the RTP timestamp may roll over and the media

Schulzrinne, et al. Standards Track [Page 285] RFC 7826 RTSP 2.0 December 2016

    agent may believe later packets to be duplicates of packets just
    played out.  Having the RTP timestamp jump will also affect the
    RTCP measurements based on this.
 As an example, assume an RTP timestamp frequency of 8000 Hz, a
 packetization interval of 100 ms, and an initial sequence number and
 timestamp of zero.
    C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
      CSeq: 4
      Session: ymIqLXufHkMHGdtENdblWK
      Range: npt=10-15
      User-Agent: PhonyClient/1.2
    S->C: RTSP/2.0 200 OK
      CSeq: 4
      Session: ymIqLXufHkMHGdtENdblWK
      Range: npt=10-15
      RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                ssrc=0D12F123:seq=0;rtptime=0
 The ensuing RTP data stream is depicted below:
    S -> C: RTP packet - seq = 0,  rtptime = 0,     NPT time = 10s
    S -> C: RTP packet - seq = 1,  rtptime = 800,   NPT time = 10.1s
     . . .
    S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s
 Upon the completion of the requested delivery, the server sends a
 PLAY_NOTIFY.
      S->C: PLAY_NOTIFY rtsp://example.com/fizzle RTSP/2.0
            CSeq: 5
            Notify-Reason: end-of-stream
            Request-Status: cseq=4 status=200 reason="OK"
            Range: npt=-15
            RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
               ssrc=0D12F123:seq=49;rtptime=39200
            Session: ymIqLXufHkMHGdtENdblWK
      C->S: RTSP/2.0 200 OK
            CSeq: 5
            User-Agent: PhonyClient/1.2
 Upon the completion of the play range, the client follows up with a
 request to PLAY from a new NPT.

Schulzrinne, et al. Standards Track [Page 286] RFC 7826 RTSP 2.0 December 2016

 C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
       CSeq: 6
       Session: ymIqLXufHkMHGdtENdblWK
       Range: npt=18-20
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 6
       Session: ymIqLXufHkMHGdtENdblWK
       Range: npt=18-20
       RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=50;rtptime=40100
 The ensuing RTP data stream is depicted below:
    S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
    S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
     . . .
    S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s
 In this example, first, NPT 10 through 15 are played, then the client
 requests the server to skip ahead and play NPT 18 through 20.  The
 first segment is presented as RTP packets with sequence numbers 0
 through 49 and timestamps 0 through 39,200.  The second segment
 consists of RTP packets with sequence numbers 50 through 69, with
 timestamps 40,100 through 55,200.  While there is a gap in the NPT,
 there is no gap in the sequence-number space of the RTP data stream.
 The RTP timestamp gap is present in the above example due to the time
 it takes to perform the second play request, in this case, 12.5 ms
 (100/8000).

C.4. Handling RTP Timestamps after PAUSE

 During a PAUSE/PLAY interaction in an RTSP session, the duration of
 time for which the RTP transmission was halted MUST be reflected in
 the RTP timestamp of each RTP stream.  The duration can be calculated
 for each RTP stream as the time elapsed from when the last RTP packet
 was sent before the PAUSE request was received and when the first RTP
 packet was sent after the subsequent PLAY request was received.  The
 duration includes all latency incurred and processing time required
 to complete the request.
    RFC 3550 [RFC3550] states that: "the RTP timestamp for each unit
    [packet] would be related to the wallclock time at which the unit
    becomes current on the virtual presentation timeline".

Schulzrinne, et al. Standards Track [Page 287] RFC 7826 RTSP 2.0 December 2016

    In order to satisfy the requirements of [RFC3550], the RTP
    timestamp space needs to increase continuously with real time.
    While this is not optimal for stored media, it is required for RTP
    and RTCP to function as intended.  Using a continuous RTP
    timestamp space allows the same timestamp model for both stored
    and live media and allows better opportunity to integrate both
    types of media under a single control.
 As an example, assume a clock frequency of 8000 Hz, a packetization
 interval of 100 ms, and an initial sequence number and timestamp of
 zero.
 C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
       CSeq: 4
       Session: ymIqLXufHkMHGdtENdblWK
       Range: npt=10-15
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 4
       Session: ymIqLXufHkMHGdtENdblWK
       Range: npt=10-15
       RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=0;rtptime=0
 The ensuing RTP data stream is depicted below:
    S -> C: RTP packet - seq = 0, rtptime = 0,    NPT time = 10s
    S -> C: RTP packet - seq = 1, rtptime = 800,  NPT time = 10.1s
    S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
    S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s

Schulzrinne, et al. Standards Track [Page 288] RFC 7826 RTSP 2.0 December 2016

 The client then sends a PAUSE request:
 C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0
       CSeq: 5
       Session: ymIqLXufHkMHGdtENdblWK
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 5
       Session: ymIqLXufHkMHGdtENdblWK
       Range: npt=10.4-15
 20 seconds elapse and then the client sends a PLAY request.  In
 addition, the server requires 15 ms to process the request:
 C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
       CSeq: 6
       Session: ymIqLXufHkMHGdtENdblWK
       User-Agent: PhonyClient/1.2
 S->C: RTSP/2.0 200 OK
       CSeq: 6
       Session: ymIqLXufHkMHGdtENdblWK
       Range: npt=10.4-15
       RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=4;rtptime=164400
 The ensuing RTP data stream is depicted below:
    S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
    S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
    S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s
 First, NPT 10 through 10.3 is played, then a PAUSE is received by the
 server.  After 20 seconds, a PLAY is received by the server that
 takes 15 ms to process.  The duration of time for which the session
 was paused is reflected in the RTP timestamp of the RTP packets sent
 after this PLAY request.
 A client can use the RTSP Range header and RTP-Info header to map NPT
 time of a presentation with the RTP timestamp.
 Note: in RFC 2326 [RFC2326], this matter was not clearly defined and
 was misunderstood commonly.  However, for RTSP 2.0, it is expected
 that this will be handled correctly and no exception handling will be
 required.

Schulzrinne, et al. Standards Track [Page 289] RFC 7826 RTSP 2.0 December 2016

 Note further: it may be required to reset some of the state to ensure
 the correct media decoding and the usual jitter-buffer handling when
 issuing a PLAY request.

C.5. RTSP/RTP Integration

 For certain data types, tight integration between the RTSP layer and
 the RTP layer will be necessary.  This by no means precludes the
 above restrictions.  Combined RTSP/RTP media clients should use the
 RTP-Info field to determine whether incoming RTP packets were sent
 before or after a seek or before or after a PAUSE.

C.6. Scaling with RTP

 For scaling (see Section 18.46), RTP timestamps should correspond to
 the rendering timing.  For example, when playing video recorded at 30
 frames per second at a scale of two and speed (Section 18.50) of one,
 the server would drop every second frame to maintain and deliver
 video packets with the normal timestamp spacing of 3,000 per frame,
 but NPT would increase by 1/15 second for each video frame.
    Note: the above scaling puts requirements on the media codec or a
    media stream to support it.  For example, motion JPEG or other
    non-predictive video coding can easier handle the above example.

C.7. Maintaining NPT Synchronization with RTP Timestamps

 The client can maintain a correct display of NPT by noting the RTP
 timestamp value of the first packet arriving after repositioning.
 The sequence parameter of the RTP-Info (Section 18.45) header
 provides the first sequence number of the next segment.

C.8. Continuous Audio

 For continuous audio, the server SHOULD set the RTP marker bit at the
 beginning of serving a new PLAY request or at jumps in timeline.
 This allows the client to perform playout delay adaptation.

C.9. Multiple Sources in an RTP Session

 Note that more than one SSRC MAY be sent in the media stream.  If it
 happens, all sources are expected to be rendered simultaneously.

C.10. Usage of SSRCs and the RTCP BYE Message during an RTSP Session

 The RTCP BYE message indicates the end of use of a given SSRC.  If
 all sources leave an RTP session, it can, in most cases, be assumed
 to have ended.  Therefore, a client or server MUST NOT send an RTCP

Schulzrinne, et al. Standards Track [Page 290] RFC 7826 RTSP 2.0 December 2016

 BYE message until it has finished using a SSRC.  A server SHOULD keep
 using an SSRC until the RTP session is terminated.  Prolonging the
 use of a SSRC allows the established synchronization context
 associated with that SSRC to be used to synchronize subsequent PLAY
 requests even if the PLAY response is late.
 An SSRC collision with the SSRC that transmits media does also have
 consequences, as it will normally force the media sender to change
 its SSRC in accordance with the RTP specification [RFC3550].
 However, an RTSP server may wait and see if the client changes and
 thus resolve the conflict to minimize the impact.  As media sender,
 SSRC change will result in a loss of synchronization context and
 require any receiver to wait for RTCP sender reports for all media
 requiring synchronization before being able to play out synchronized.
 Due to these reasons, a client joining a session should take care not
 to select the same SSRC(s) as the server indicates in the ssrc
 Transport header parameter.  Any SSRC signaled in the Transport
 header MUST be avoided.  A client detecting a collision prior to
 sending any RTP or RTCP messages SHALL also select a new SSRC.

C.11. Future Additions

 It is the intention that any future protocol or profile regarding
 media delivery and lower transport should be easy to add to RTSP.
 This section provides the necessary steps that need to be met.
 The following things need to be considered when adding a new protocol
 or profile for use with RTSP:
 o  The protocol or profile needs to define a name tag representing
    it.  This tag is required to be an ABNF "token" to be possible to
    use in the Transport header specification.
 o  The useful combinations of protocol, profiles, and lower-layer
    transport for this extension need to be defined.  For each
    combination, declare the necessary parameters to use in the
    Transport header.
 o  For new media protocols, the interaction with RTSP needs to be
    addressed.  One important factor will be the media
    synchronization.  It may be necessary to have new headers similar
    to RTP info to carry this information.
 o  Discussion needs to occur regarding congestion control for media,
    especially if transport without built-in congestion control is
    used.

Schulzrinne, et al. Standards Track [Page 291] RFC 7826 RTSP 2.0 December 2016

 See the IANA Considerations section (Section 22) for information on
 how to register new attributes.

Appendix D. Use of SDP for RTSP Session Descriptions

 The Session Description Protocol (SDP, [RFC4566]) may be used to
 describe streams or presentations in RTSP.  This description is
 typically returned in reply to a DESCRIBE request on a URI from a
 server to a client or received via HTTP from a server to a client.
 This appendix describes how an SDP file determines the operation of
 an RTSP session.  Thus, it is worth pointing out that the
 interpretation of the SDP is done in the context of the SDP receiver,
 which is the one being configured.  This is the same as in SAP
 [RFC2974]; this differs from SDP Offer/Answer [RFC3264] where each
 SDP is interpreted in the context of the agent providing it.
 SDP as is provides no mechanism by which a client can distinguish,
 without human guidance, between several media streams to be rendered
 simultaneously and a set of alternatives (e.g., two audio streams
 spoken in different languages).  The SDP extension found in "The
 Session Description Protocol (SDP) Grouping Framework" [RFC5888]
 provides such functionality to some degree.  Appendix D.4 describes
 the usage of SDP media line grouping for RTSP.

D.1. Definitions

 The terms "session-level", "media-level", and other key/attribute
 names and values used in this appendix are to be used as defined in
 SDP [RFC4566]:

D.1.1. Control URI

 The "a=control" attribute is used to convey the control URI.  This
 attribute is used both for the session and media descriptions.  If
 used for individual media, it indicates the URI to be used for
 controlling that particular media stream.  If found at the session
 level, the attribute indicates the URI for aggregate control
 (presentation URI).  The session-level URI MUST be different from any
 media-level URI.  The presence of a session-level control attribute
 MUST be interpreted as support for aggregated control.  The control
 attribute MUST be present on the media level unless the presentation
 only contains a single media stream; in which case, the attribute MAY
 be present on the session level only and then also apply to that
 single media stream.
 ABNF for the attribute is defined in Section 20.3.

Schulzrinne, et al. Standards Track [Page 292] RFC 7826 RTSP 2.0 December 2016

 Example:
   a=control:rtsp://example.com/foo
 This attribute MAY contain either relative or absolute URIs,
 following the rules and conventions set out in RFC 3986 [RFC3986].
 Implementations MUST look for a base URI in the following order:
 1.  the RTSP Content-Base field;
 2.  the RTSP Content-Location field;
 3.  the RTSP Request-URI.
 If this attribute contains only an asterisk (*), then the URI MUST be
 treated as if it were an empty embedded URI; thus, it will inherit
 the entire base URI.
    Note: RFC 2326 was very unclear on the processing of relative URIs
    and several RTSP 1.0 implementations at the point of publishing
    this document did not perform RFC 3986 processing to determine the
    resulting URI; instead, simple concatenation is common.  To avoid
    this issue completely, it is recommended to use absolute URIs in
    the SDP.
 The URI handling for SDPs from container files needs special
 consideration.  For example, let's assume that a container file has
 the URI: "rtsp://example.com/container.mp4".  Let's further assume
 this URI is the base URI and that there is an absolute media-level
 URI: "rtsp://example.com/container.mp4/trackID=2".  A relative media-
 level URI that resolves in accordance with RFC 3986 [RFC3986] to the
 above given media URI is "container.mp4/trackID=2".  It is usually
 not desirable to need to include or modify the SDP stored within the
 container file with the server local name of the container file.  To
 avoid this, one can modify the base URI used to include a trailing
 slash, e.g., "rtsp://example.com/container.mp4/".  In this case, the
 relative URI for the media will only need to be "trackID=2".
 However, this will also mean that using "*" in the SDP will result in
 the control URI including the trailing slash, i.e.,
 "rtsp://example.com/container.mp4/".
    Note: the usage of TrackID in the above is not a standardized
    form, but one example out of several similar strings such as
    TrackID, Track_ID, StreamID that is used by different server
    vendors to indicate a particular piece of media inside a container
    file.

Schulzrinne, et al. Standards Track [Page 293] RFC 7826 RTSP 2.0 December 2016

D.1.2. Media Streams

 The "m=" field is used to enumerate the streams.  It is expected that
 all the specified streams will be rendered with appropriate
 synchronization.  If the session is over multicast, the port number
 indicated SHOULD be used for reception.  The client MAY try to
 override the destination port, through the Transport header.  The
 servers MAY allow this: the response will indicate whether or not
 this is allowed.  If the session is unicast, the port numbers are the
 ones RECOMMENDED by the server to the client, about which receiver
 ports to use; the client MUST still include its receiver ports in its
 SETUP request.  The client MAY ignore this recommendation.  If the
 server has no preference, it SHOULD set the port number value to
 zero.
 The "m=" lines contain information about which transport protocol,
 profile, and possibly lower-layer are to be used for the media
 stream.  The combination of transport, profile, and lower layer, like
 RTP/AVP/UDP, needs to be defined for how to be used with RTSP.  The
 currently defined combinations are discussed in Appendix C; further
 combinations MAY be specified.
 Example:
   m=audio 0 RTP/AVP 31

D.1.3. Payload Type(s)

 The payload type or types are specified in the "m=" line.  In case
 the payload type is a static payload type from RFC 3551 [RFC3551], no
 other information may be required.  In case it is a dynamic payload
 type, the media attribute "rtpmap" is used to specify what the media
 is.  The "encoding name" within the "rtpmap" attribute may be one of
 those specified in [RFC4856], a media type registered with IANA
 according to [RFC4855], or an experimental encoding as specified in
 SDP [RFC4566]).  Codec-specific parameters are not specified in this
 field, but rather in the "fmtp" attribute described below.
 The selection of the RTP payload type numbers used may be required to
 consider RTP and RTCP Multiplexing [RFC5761], if that is to be
 supported by the server.

D.1.4. Format-Specific Parameters

 Format-specific parameters are conveyed using the "fmtp" media
 attribute.  The syntax of the "fmtp" attribute is specific to the
 encoding(s) to which the attribute refers.  Note that some of the

Schulzrinne, et al. Standards Track [Page 294] RFC 7826 RTSP 2.0 December 2016

 format-specific parameters may be specified outside of the "fmtp"
 parameters, for example, like the "ptime" attribute for most audio
 encodings.

D.1.5. Directionality of Media Stream

 The SDP attributes "a=sendrecv", "a=recvonly", and "a=sendonly"
 provide instructions about the direction the media streams flow
 within a session.  When using RTSP, the SDP can be delivered to a
 client using either RTSP DESCRIBE or a number of RTSP external
 methods, like HTTP, FTP, and email.  Based on this, the SDP applies
 to how the RTSP client will see the complete session.  Thus, media
 streams delivered from the RTSP server to the client would be given
 the "a=recvonly" attribute.
 "a=recvonly" in an SDP provided to the RTSP client indicates that
 media delivery will only occur in the direction from the RTSP server
 to the client.  SDP provided to the RTSP client that lacks any of the
 directionality attributes ("a=recvonly", "a=sendonly", "a=sendrecv")
 would be interpreted as having "a=sendrecv".  At the time of writing,
 there exists no RTSP mode suitable for media traffic in the direction
 from the RTSP client to the server.  Thus, all RTSP SDP SHOULD have
 an "a=recvonly" attribute when using the PLAY mode defined in this
 document.  If future modes are defined for media in the client-to-
 server direction, then usage of "a=sendonly" or "a=sendrecv" may
 become suitable to indicate intended media directions.

D.1.6. Range of Presentation

 The "a=range" attribute defines the total time range of the stored
 session or an individual media.  Live sessions that are not seekable
 can be indicated as specified below; whereas the length of live
 sessions can be deduced from the "t=" and "r=" SDP parameters.
 The attribute is both a session- and a media-level attribute.  For
 presentations that contain media streams of the same duration, the
 range attribute SHOULD only be used at the session level.  In case of
 different lengths, the range attribute MUST be given at media level
 for all media and SHOULD NOT be given at the session level.  If the
 attribute is present at both media level and session level, the
 media-level values MUST be used.
 Note: usually one will specify the same length for all media, even if
 there isn't media available for the full duration on all media.
 However, that requires that the server accept PLAY requests within
 that range.

Schulzrinne, et al. Standards Track [Page 295] RFC 7826 RTSP 2.0 December 2016

 Servers MUST take care to provide RTSP Range (see Section 18.40)
 values that are consistent with what is presented in the SDP for the
 content.  There is no reason for non dynamic content, like media
 clips provided on demand to have inconsistent values.  Inconsistent
 values between the SDP and the actual values for the content handled
 by the server is likely to generate some failure, like 457 "Invalid
 Range", in case the client uses PLAY requests with a Range header.
 In case the content is dynamic in length and it is infeasible to
 provide a correct value in the SDP, the server is recommended to
 describe this as content that is not seekable (see below).  The
 server MAY override that property in the response to a PLAY request
 using the correct values in the Range header.
 The unit is specified first, followed by the value range.  The units
 and their values are as defined in Section 4.4.1, Section 4.4.2, and
 Section 4.4.3 and MAY be extended with further formats.  Any open-
 ended range (start-), i.e., without stop range, is of unspecified
 duration and MUST be considered as content that is not seekable
 unless this property is overridden.  Multiple instances carrying
 different clock formats MAY be included at either session or media
 level.
 ABNF for the attribute is defined in Section 20.3.
 Examples:
   a=range:npt=0-34.4368
   a=range:clock=19971113T211503Z-19971113T220300Z
   Non-seekable stream of unknown duration:
   a=range:npt=0-

D.1.7. Time of Availability

 The "t=" field defines when the SDP is valid.  For on-demand content,
 the server SHOULD indicate a stop time value for which it guarantees
 the description to be valid and a start time that is equal to or
 before the time at which the DESCRIBE request was received.  It MAY
 also indicate start and stop times of 0, meaning that the session is
 always available.
 For sessions that are of live type, i.e., specific start time,
 unknown stop time, likely not seekable, the "t=" and "r=" field
 SHOULD be used to indicate the start time of the event.  The stop
 time SHOULD be given so that the live event will have ended at that
 time, while still not being unnecessary far into the future.

Schulzrinne, et al. Standards Track [Page 296] RFC 7826 RTSP 2.0 December 2016

D.1.8. Connection Information

 In SDP used with RTSP, the "c=" field contains the destination
 address for the media stream.  If a multicast address is specified,
 the client SHOULD use this address in any SETUP request as
 destination address, including any additional parameters, such as
 TTL.  For on-demand unicast streams and some multicast streams, the
 destination address MAY be specified by the client via the SETUP
 request, thus overriding any specified address.  To identify streams
 without a fixed destination address, where the client is required to
 specify a destination address, the "c=" field SHOULD be set to a null
 value.  For addresses of type "IP4", this value MUST be "0.0.0.0";
 and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0" (can also be
 written as "::"), i.e., the unspecified address according to RFC 4291
 [RFC4291].

D.1.9. Message Body Tag

 The optional "a=mtag" attribute identifies a version of the session
 description.  It is opaque to the client.  SETUP requests may include
 this identifier in the If-Match field (see Section 18.24) to allow
 session establishment only if this attribute value still corresponds
 to that of the current description.  The attribute value is opaque
 and may contain any character allowed within SDP attribute values.
 ABNF for the attribute is defined in Section 20.3.
 Example:
   a=mtag:"158bb3e7c7fd62ce67f12b533f06b83a"
    One could argue that the "o=" field provides identical
    functionality.  However, it does so in a manner that would put
    constraints on servers that need to support multiple session
    description types other than SDP for the same piece of media
    content.

Schulzrinne, et al. Standards Track [Page 297] RFC 7826 RTSP 2.0 December 2016

D.2. Aggregate Control Not Available

 If a presentation does not support aggregate control, no session-
 level "a=control" attribute is specified.  For an SDP with multiple
 media sections specified, each section will have its own control URI
 specified via the "a=control" attribute.
 Example:
 v=0
 o=- 2890844256 2890842807 IN IP4 192.0.2.56
 s=I came from a web page
 e=adm@example.com
 c=IN IP4 0.0.0.0
 t=0 0
 m=video 8002 RTP/AVP 31
 a=control:rtsp://audio.example.com/movie.aud
 m=audio 8004 RTP/AVP 3
 a=control:rtsp://video.example.com/movie.vid
 Note that the position of the control URI in the description implies
 that the client establishes separate RTSP control sessions to the
 servers audio.example.com and video.example.com.
 It is recommended that an SDP file contain the complete media-
 initialization information even if it is delivered to the media
 client through non-RTSP means.  This is necessary as there is no
 mechanism to indicate that the client should request more detailed
 media stream information via DESCRIBE.

D.3. Aggregate Control Available

 In this scenario, the server has multiple streams that can be
 controlled as a whole.  In this case, there are both a media-level
 "a=control" attribute, which is used to specify the stream URIs, and
 a session-level "a=control" attribute, which is used as the Request-
 URI for aggregate control.  If the media-level URI is relative, it is
 resolved to absolute URIs according to Appendix D.1.1 above.

Schulzrinne, et al. Standards Track [Page 298] RFC 7826 RTSP 2.0 December 2016

 Example:
 C->M: DESCRIBE rtsp://example.com/movie RTSP/2.0
       CSeq: 1
       User-Agent: PhonyClient/1.2
 M->C: RTSP/2.0 200 OK
       CSeq: 1
       Date: Wed, 23 Jan 2013 15:36:52 +0000
       Expires: Wed, 23 Jan 2013 16:36:52 +0000
       Content-Type: application/sdp
       Content-Base: rtsp://example.com/movie/
       Content-Length: 227
       v=0
       o=- 2890844256 2890842807 IN IP4 192.0.2.211
       s=I contain
       i=<more info>
       e=adm@example.com
       c=IN IP4 0.0.0.0
       a=control:*
       t=0 0
       m=video 8002 RTP/AVP 31
       a=control:trackID=1
       m=audio 8004 RTP/AVP 3
       a=control:trackID=2
 In this example, the client is recommended to establish a single RTSP
 session to the server, and it uses the URIs rtsp://example.com/movie/
 trackID=1 and rtsp://example.com/movie/trackID=2 to set up the video
 and audio streams, respectively.  The URI rtsp://example.com/movie/,
 which is resolved from the "*", controls the whole presentation
 (movie).
 A client is not required to issue SETUP requests for all streams
 within an aggregate object.  Servers should allow the client to ask
 for only a subset of the streams.

D.4. Grouping of Media Lines in SDP

 For some types of media, it is desirable to express a relationship
 between various media components, for instance, for lip
 synchronization or Scalable Video Codec (SVC) [RFC5583].  This
 relationship is expressed on the SDP level by grouping of media
 lines, as described in [RFC5888], and can be exposed to RTSP.

Schulzrinne, et al. Standards Track [Page 299] RFC 7826 RTSP 2.0 December 2016

 For RTSP, it is mainly important to know how to handle grouped media
 received by means of SDP, i.e., if the media are under aggregate
 control (see Appendix D.3) or if aggregate control is not available
 (see Appendix D.2).
 It is RECOMMENDED that grouped media are handled by aggregate
 control, to give the client the ability to control either the whole
 presentation or single media.

D.5. RTSP External SDP Delivery

 There are some considerations that need to be made when the session
 description is delivered to the client outside of RTSP, for example
 via HTTP or email.
 First of all, the SDP needs to contain absolute URIs, since relative
 will, in most cases, not work as the delivery will not correctly
 forward the base URI.
 The writing of the SDP session availability information, i.e., "t="
 and "r=", needs to be carefully considered.  When the SDP is fetched
 by the DESCRIBE method, the probability that it is valid is very
 high.  However, the same is much less certain for SDPs distributed
 using other methods.  Therefore, the publisher of the SDP should take
 care to follow the recommendations about availability in the SDP
 specification [RFC4566] in Section 4.2.

Appendix E. RTSP Use Cases

 This appendix describes the most important and considered use cases
 for RTSP.  They are listed in descending order of importance in
 regard to ensuring that all necessary functionality is present.  This
 specification only fully supports usage of the two first.  Also, in
 these first two cases, there are special cases or exceptions that are
 not supported without extensions, e.g., the redirection of media
 delivery to an address other than the controlling agent's (client's).

E.1. On-Demand Playback of Stored Content

 An RTSP-capable server stores content suitable for being streamed to
 a client.  A client desiring playback of any of the stored content
 uses RTSP to set up the media transport required to deliver the
 desired content.  RTSP is then used to initiate, halt, and manipulate
 the actual transmission (playout) of the content.  RTSP is also
 required to provide the necessary description and synchronization
 information for the content.

Schulzrinne, et al. Standards Track [Page 300] RFC 7826 RTSP 2.0 December 2016

 The above high-level description can be broken down into a number of
 functions of which RTSP needs to be capable.
 Presentation Description:  Provide initialization information about
       the presentation (content); for example, which media codecs are
       needed for the content.  Other information that is important
       includes the number of media streams the presentation contains,
       the transport protocols used for the media streams, and
       identifiers for these media streams.  This information is
       required before setup of the content is possible and to
       determine if the client is even capable of using the content.
       This information need not be sent using RTSP; other external
       protocols can be used to transmit the transport presentation
       descriptions.  Two good examples are the use of HTTP [RFC7230]
       or email to fetch or receive presentation descriptions like SDP
       [RFC4566]
 Setup:  Set up some or all of the media streams in a presentation.
       The setup itself consists of selecting the protocol for media
       transport and the necessary parameters for the protocol, like
       addresses and ports.
 Control of Transmission:  After the necessary media streams have been
       established, the client can request the server to start
       transmitting the content.  The client must be allowed to start
       or stop the transmission of the content at arbitrary times.
       The client must also be able to start the transmission at any
       point in the timeline of the presentation.
 Synchronization:  For media-transport protocols like RTP [RFC3550],
       it might be beneficial to carry synchronization information
       within RTSP.  This may be due to either the lack of inter-media
       synchronization within the protocol itself or the potential
       delay before the synchronization is established (which is the
       case for RTP when using RTCP).
 Termination:  Terminate the established contexts.
 For this use case, there are a number of assumptions about how it
 works.  These are:
 On-Demand content:  The content is stored at the server and can be
       accessed at any time during a time period when it is intended
       to be available.

Schulzrinne, et al. Standards Track [Page 301] RFC 7826 RTSP 2.0 December 2016

 Independent sessions:  A server is capable of serving a number of
       clients simultaneously, including from the same piece of
       content at different points in that presentations timeline.
 Unicast Transport:  Content for each individual client is transmitted
       to them using unicast traffic.
 It is also possible to redirect the media traffic to a different
 destination than that of the agent controlling the traffic.  However,
 allowing this without appropriate mechanisms for checking that the
 destination approves of this allows for Distributed DoS (DDoS).

E.2. Unicast Distribution of Live Content

 This use case is similar to the above on-demand content case (see
 Appendix E.1), the difference is the nature of the content itself.
 Live content is continuously distributed as it becomes available from
 a source; i.e., the main difference from on-demand is that one starts
 distributing content before the end of it has become available to the
 server.
 In many cases, the consumer of live content is only interested in
 consuming what actually happens "now"; i.e., very similar to
 broadcast TV.  However, in this case, it is assumed that there exists
 no broadcast or multicast channel to the users, and instead the
 server functions as a distribution node, sending the same content to
 multiple receivers, using unicast traffic between server and client.
 This unicast traffic and the transport parameters are individually
 negotiated for each receiving client.
 Another aspect of live content is that it often has a very limited
 time of availability, as it is only available for the duration of the
 event the content covers.  An example of such live content could be a
 music concert that lasts two hours and starts at a predetermined
 time.  Thus, there is a need to announce when and for how long the
 live content is available.
 In some cases, the server providing live content may be saving some
 or all of the content to allow clients to pause the stream and resume
 it from the paused point, or to "rewind" and play continuously from a
 point earlier than the live point.  Hence, this use case does not
 necessarily exclude playing from other than the live point of the
 stream, playing with scales other than 1.0, etc.

Schulzrinne, et al. Standards Track [Page 302] RFC 7826 RTSP 2.0 December 2016

E.3. On-Demand Playback Using Multicast

 It is possible to use RTSP to request that media be delivered to a
 multicast group.  The entity setting up the session (the controller)
 will then control when and what media is delivered to the group.
 This use case has some potential for DoS attacks by flooding a
 multicast group.  Therefore, a mechanism is needed to indicate that
 the group actually accepts the traffic from the RTSP server.
 An open issue in this use case is how one ensures that all receivers
 listening to the multicast or broadcast receives the session
 presentation configuring the receivers.  This specification has to
 rely on an external solution to solve this issue.

E.4. Inviting an RTSP Server into a Conference

 If one has an established conference or group session, it is possible
 to have an RTSP server distribute media to the whole group.
 Transmission to the group is simplest when controlled by a single
 participant or leader of the conference.  Shared control might be
 possible, but would require further investigation and possibly
 extensions.
 This use case assumes that there exists either a multicast or a
 conference focus that redistributes media to all participants.
 This use case is intended to be able to handle the following
 scenario: a conference leader or participant (hereafter called the
 "controller") has some pre-stored content on an RTSP server that he
 wants to share with the group.  The controller sets up an RTSP
 session at the streaming server for this content and retrieves the
 session description for the content.  The destination for the media
 content is set to the shared multicast group or conference focus.
 When desired by the controller, he/she can start and stop the
 transmission of the media to the conference group.
 There are several issues with this use case that are not solved by
 this core specification for RTSP:
 DoS:  To avoid an RTSP server from being an unknowing participant in
       a DoS attack, the server needs to be able to verify the
       destination's acceptance of the media.  Such a mechanism to
       verify the approval of received media does not yet exist;
       instead, only policies can be used, which can be made to work
       in controlled environments.

Schulzrinne, et al. Standards Track [Page 303] RFC 7826 RTSP 2.0 December 2016

 Distributing the presentation description to all participants in the
 group:
          To enable a media receiver to correctly decode the content,
          the media configuration information needs to be distributed
          reliably to all participants.  This will most likely require
          support from an external protocol.
    Passing control of the session:  If it is desired to pass control
          of the RTSP session between the participants, some support
          will be required by an external protocol to exchange state
          information and possibly floor control of who is controlling
          the RTSP session.

E.5. Live Content Using Multicast

 This use case in its simplest form does not require any use of RTSP
 at all; this is what multicast conferences being announced with SAP
 [RFC2974] and SDP are intended to handle.  However, in use cases
 where more advanced features like access control to the multicast
 session are desired, RTSP could be used for session establishment.
 A client desiring to join a live multicasted media session with
 cryptographic (encryption) access control could use RTSP in the
 following way.  The source of the session announces the session and
 gives all interested an RTSP URI.  The client connects to the server
 and requests the presentation description, allowing configuration for
 reception of the media.  In this step, it is possible for the client
 to use secured transport and any desired level of authentication; for
 example, for billing or access control.  An RTSP link also allows for
 load balancing between multiple servers.
 If these were the only goals, they could be achieved by simply using
 HTTP.  However, for cases where the sender likes to keep track of
 each individual receiver of a session, and possibly use the session
 as a side channel for distributing key-updates or other information
 on a per-receiver basis, and the full set of receivers is not known
 prior to the session start, the state establishment that RTSP
 provides can be beneficial.  In this case, a client would establish
 an RTSP session for this multicast group with the RTSP server.  The
 RTSP server will not transmit any media, but instead will point to
 the multicast group.  The client and server will be able to keep the
 session alive for as long as the receiver participates in the session
 thus enabling, for example, the server to push updates to the client.
 This use case will most likely not be able to be implemented without
 some extensions to the server-to-client push mechanism.  Here the
 PLAY_NOTIFY method (see Section 13.5) with a suitable extension could
 provide clear benefits.

Schulzrinne, et al. Standards Track [Page 304] RFC 7826 RTSP 2.0 December 2016

Appendix F. Text Format for Parameters

 A resource of type "text/parameters" consists of either 1) a list of
 parameters (for a query) or 2) a list of parameters and associated
 values (for a response or setting of the parameter).  Each entry of
 the list is a single line of text.  Parameters are separated from
 values by a colon.  The parameter name MUST only use US-ASCII visible
 characters while the values are UTF-8 text strings.  The media type
 registration form is in Section 22.16.
 There is a potential interoperability issue for this format.  It was
 named in RFC 2326 but never defined, even if used in examples that
 hint at the syntax.  This format matches the purpose and its syntax
 supports the examples provided.  However, it goes further by allowing
 UTF-8 in the value part; thus, usage of UTF-8 strings may not be
 supported.  However, as individual parameters are not defined, the
 implementing application needs to have out-of-band agreement or using
 feature tag anyway to determine if the endpoint supports the
 parameters.
 The ABNF [RFC5234] grammar for "text/parameters" content is:
 file             = *((parameter / parameter-value) CRLF)
 parameter        = 1*visible-except-colon
 parameter-value  = parameter *WSP ":" value
 visible-except-colon = %x21-39 / %x3B-7E    ; VCHAR - ":"
 value            = *(TEXT-UTF8char / WSP)
 TEXT-UTF8char    = <as defined in Section 20.1>
 WSP              = <See RFC 5234> ; Space or HTAB
 VCHAR            = <See RFC 5234>
 CRLF             = <See RFC 5234>

Appendix G. Requirements for Unreliable Transport of RTSP

 This appendix provides guidance for those who want to implement RTSP
 messages over unreliable transports as has been defined in RTSP 1.0
 [RFC2326].  RFC 2326 defined the "rtspu" URI scheme and provided some
 basic information for the transport of RTSP messages over UDP.  The
 information is being provided here as there has been at least one
 commercial implementation and compatibility with that should be
 maintained.

Schulzrinne, et al. Standards Track [Page 305] RFC 7826 RTSP 2.0 December 2016

 The following points should be considered for an interoperable
 implementation:
 o  Requests shall be acknowledged by the receiver.  If there is no
    acknowledgement, the sender may resend the same message after a
    timeout of one round-trip time (RTT).  Any retransmissions due to
    lack of acknowledgement must carry the same sequence number as the
    original request.
 o  The RTT can be estimated as in TCP (RFC 6298) [RFC6298], with an
    initial round-trip value of 500 ms.  An implementation may cache
    the last RTT measurement as the initial value for future
    connections.
 o  The Timestamp header (Section 18.53) is used to avoid the
    retransmission ambiguity problem [Stevens98].
 o  The registered default port for RTSP over UDP for the server is
    554.
 o  RTSP messages can be carried over any lower-layer transport
    protocol that is 8-bit clean.
 o  RTSP messages are vulnerable to bit errors and should not be
    subjected to them.
 o  Source authentication, or at least validation that RTSP messages
    comes from the same entity becomes extremely important, as session
    hijacking may be substantially easier for RTSP message transport
    using an unreliable protocol like UDP than for TCP.
 There are two RTSP headers that are primarily intended for being used
 by the unreliable handling of RTSP messages and which will be
 maintained:
 o  CSeq: See Section 18.20.  It should be noted that the CSeq header
    is also required to match requests and responses independent
    whether a reliable or unreliable transport is used.
 o  Timestamp: See Section 18.53

Appendix H. Backwards-Compatibility Considerations

 This section contains notes on issues about backwards compatibility
 with clients or servers being implemented according to RFC 2326
 [RFC2326].  Note that there exists no requirement to implement RTSP
 1.0; in fact, this document recommends against it as it is difficult
 to do in an interoperable way.

Schulzrinne, et al. Standards Track [Page 306] RFC 7826 RTSP 2.0 December 2016

 A server implementing RTSP 2.0 MUST include an RTSP-Version of
 "RTSP/2.0" in all responses to requests containing RTSP-Version value
 of "RTSP/2.0".  If a server receives an RTSP 1.0 request, it MAY
 respond with an RTSP 1.0 response if it chooses to support RFC 2326.
 If the server chooses not to support RFC 2326, it MUST respond with a
 505 (RTSP Version Not Supported) status code.  A server MUST NOT
 respond to an RTSP 1.0 request with an RTSP 2.0 response.
 Clients implementing RTSP 2.0 MAY use an OPTIONS request with an
 RTSP-Version of "RTSP/2.0" to determine whether a server supports
 RTSP 2.0.  If the server responds with either an RTSP-Version of
 "RTSP/1.0" or a status code of 505 (RTSP Version Not Supported), the
 client will have to use RTSP 1.0 requests if it chooses to support
 RFC 2326.

H.1. Play Request in Play State

 The behavior in the server when a Play is received in Play state has
 changed (Section 13.4).  In RFC 2326, the new PLAY request would be
 queued until the current Play completed.  Any new PLAY request now
 takes effect immediately replacing the previous request.

H.2. Using Persistent Connections

 Some server implementations of RFC 2326 maintain a one-to-one
 relationship between a connection and an RTSP session.  Such
 implementations require clients to use a persistent connection to
 communicate with the server and when a client closes its connection,
 the server may remove the RTSP session.  This is worth noting if an
 RTSP 2.0 client also supporting 1.0 connects to a 1.0 server.

Appendix I. Changes

 This appendix briefly lists the differences between RTSP 1.0
 [RFC2326] and RTSP 2.0 for an informational purpose.  For
 implementers of RTSP 2.0, it is recommended to read carefully through
 this memo and not to rely on the list of changes below to adapt from
 RTSP 1.0 to RTSP 2.0, as RTSP 2.0 is not intended to be backwards
 compatible with RTSP 1.0 [RFC2326] other than the version negotiation
 mechanism.

Schulzrinne, et al. Standards Track [Page 307] RFC 7826 RTSP 2.0 December 2016

I.1. Brief Overview

 The following protocol elements were removed in RTSP 2.0 compared to
 RTSP 1.0:
 o  the RECORD and ANNOUNCE methods and all related functionality
    (including 201 (Created) and 250 (Low On Storage Space) status
    codes);
 o  the use of UDP for RTSP message transport (due to missing interest
    and to broken specification);
 o  the use of PLAY method for keep-alive in Play state.
 The following protocol elements were added or changed in RTSP 2.0
 compared to RTSP 1.0:
 o  RTSP session TEARDOWN from the server to the client;
 o  IPv6 support;
 o  extended IANA registries (e.g., transport headers parameters,
    transport-protocol, profile, lower-transport, and mode);
 o  request pipelining for quick session start-up;
 o  fully reworked state machine;
 o  RTSP messages now use URIs rather than URLs;
 o  incorporated much of related HTTP text ([RFC2616]) in this memo,
    compared to just referencing the sections in HTTP, to avoid
    ambiguities;
 o  the REDIRECT method was expanded and diversified for different
    situations;
 o  Includes a new section about how to set up different media-
    transport alternatives and their profiles in addition to lower-
    layer protocols.  This caused the appendix on RTP interaction to
    be moved to the new section instead of being in the part that
    describes RTP.  The section also includes guidelines what to
    consider when writing usage guidelines for new protocols and
    profiles;

Schulzrinne, et al. Standards Track [Page 308] RFC 7826 RTSP 2.0 December 2016

 o  Added an asynchronous notification method PLAY_NOTIFY.  This
    method is used by the RTSP server to asynchronously notify clients
    about session changes while in Play state.  To a limited extent,
    this is comparable with some implementations of ANNOUNCE in RTSP
    1.0 not intended for Recording.

I.2. Detailed List of Changes

 The below changes have been made to RTSP 1.0 (RFC 2326) when defining
 RTSP 2.0.  Note that this list does not reflect minor changes in
 wording or correction of typographical errors.
 o  The section on minimal implementation was deleted.  Instead, the
    main part of the specification defines the core of RTSP 2.0.
 o  The Transport header has been changed in the following ways:
  • The ABNF has been changed to define that extensions are

possible and that unknown parameters result in servers ignoring

       the transport specification.
  • To prevent backwards compatibility issues, any extension or new

parameter requires the usage of a feature tag combined with the

       Require header.
  • Syntax ambiguities with the Mode parameter have been resolved.
  • Syntax error with ";" for multicast and unicast has been

resolved.

  • Two new addressing parameters have been defined: src_addr and

dest_addr. These replace the parameters "port", "client_port",

       "server_port", "destination", and "source".
  • Support for IPv6 explicit addresses in all address fields has

been included.

  • To handle URI definitions that contain ";" or ",", a quoted-URI

format has been introduced and is required.

  • IANA registries for the transport header parameters, transport-

protocol, profile, lower-transport, and mode have been defined.

  • The Transport header's interleaved parameter's text was made

more strict and uses formal requirements levels. It was also

       clarified that the interleaved channels are symmetric and that
       it is the server that sets the channel numbers.

Schulzrinne, et al. Standards Track [Page 309] RFC 7826 RTSP 2.0 December 2016

  • It has been clarified that the client can't request of the

server to use a certain RTP SSRC, using a request with the

       transport parameter SSRC.
  • Syntax definition for SSRC has been clarified to require 8HEX.

It has also been extended to allow multiple values for clients

       supporting this version.
  • Clarified the text on the Transport header's "dest_addr"

parameters regarding what security precautions the server is

       required to perform.
 o  The Range formats have been changed in the following way:
  • The NPT format has been given an initial NPT identifier that

must now be used.

  • All formats now support initial open-ended formats of type

"npt=-10" and also format only "Range: smpte" ranges for usage

       with GET_PARAMETER requests.
  • The npt-hhmmss notation now follows ISO 8601 more strictly.
 o  RTSP message handling has been changed in the following ways:
  • RTSP messages now use URIs rather than URLs.
  • It has been clarified that a 4xx message due to a missing CSeq

header shall be returned without a CSeq header.

  • The 300 (Multiple Choices) response code has been removed.
  • Rules for how to handle the timing out RTSP messages have been

added.

  • Extended Pipelining rules allowing for quick session startup.
  • Sequence numbering and proxy handling of sequence numbers have

been defined, including cases when responses arrive out of

       order.
 o  The HTTP references have been updated to first RFCs 2616 and 2617
    and then to RFC 7230-7235.  Most of the text has been copied and
    then altered to fit RTSP into this specification.  The Public and
    the Content-Base headers have also been imported from RFC 2068 so
    that they are defined in the RTSP specification.  Known effects on
    RTSP due to HTTP clarifications:

Schulzrinne, et al. Standards Track [Page 310] RFC 7826 RTSP 2.0 December 2016

  • Content-Encoding header can include encoding of type

"identity".

 o  The state machine section has been completely rewritten.  It now
    includes more details and is also more clear about the model used.
 o  An IANA section has been included that contains a number of
    registries and their rules.  This will allow us to use IANA to
    keep track of RTSP extensions.
 o  The transport of RTSP messages has seen the following changes:
  • The use of UDP for RTSP message transport has been deprecated

due to missing interest and to broken specification.

  • The rules for how TCP connections are to be handled have been

clarified. Now it is made clear that servers should not close

       the TCP connection unless they have been unused for significant
       time.
  • Strong recommendations why servers and clients should use

persistent connections have also been added.

  • There is now a requirement on the servers to handle non-

persistent connections as this provides fault tolerance.

  • Added wording on the usage of Connection:Close for RTSP.
  • Specified usage of TLS for RTSP messages, including a scheme to

approve a proxy's TLS connection to the next hop.

 o  The following header-related changes have been made:
  • Accept-Ranges response-header has been added. This header

clarifies which range formats can be used for a resource.

  • Fixed the missing definitions for the Cache-Control header.

Also added to the syntax definition the missing delta-seconds

       for max-stale and min-fresh parameters.
  • Put requirement on CSeq header that the value is increased by

one for each new RTSP request. A recommendation to start at 0

       has also been added.
  • Added a requirement that the Date header must be used for all

messages with a message body and the Server should always

       include it.

Schulzrinne, et al. Standards Track [Page 311] RFC 7826 RTSP 2.0 December 2016

  • Removed the possibility of using Range header with Scale header

to indicate when it is to be activated, since it can't work as

       defined.  Also, added a rule that lack of Scale header in a
       response indicates lack of support for the header.  feature
       tags for scaled playback have been defined.
  • The Speed header must now be responded to in order to indicate

support and the actual speed going to be used. A feature tag

       is defined.  Notes on congestion control were also added.
  • The Supported header was borrowed from SIP [RFC3261] to help

with the feature negotiation in RTSP.

  • Clarified that the Timestamp header can be used to resolve

retransmission ambiguities.

  • The Session header text has been expanded with an explanation

on keep-alive and which methods to use. SET_PARAMETER is now

       recommended to use if only keep-alive within RTSP is desired.
  • It has been clarified how the Range header formats are used to

indicate pause points in the PAUSE response.

  • Clarified that RTP-Info URIs that are relative use the Request-

URI as base URI. Also clarified that the used URI must be the

       one that was used in the SETUP request.  The URIs are now also
       required to be quoted.  The header also expresses the SSRC for
       the provided RTP timestamp and sequence number values.
  • Added text that requires the Range to always be present in PLAY

responses. Clarified what should be sent in case of live

       streams.
  • The headers table has been updated using a structure borrowed

from SIP. Those tables convey much more information and should

       provide a good overview of the available headers.
  • It has been clarified that any message with a message body is

required to have a Content-Length header. This was the case in

       RFC 2326, but could be misinterpreted.
  • ETag has changed its name to MTag.
  • To resolve functionality around MTag, the MTag and If-None-

Match header have been added from HTTP with necessary

       clarification in regard to RTSP operation.

Schulzrinne, et al. Standards Track [Page 312] RFC 7826 RTSP 2.0 December 2016

  • Imported the Public header from HTTP (RFC 2068 [RFC2068]) since

it has been removed from HTTP due to lack of use. Public is

       used quite frequently in RTSP.
  • Clarified rules for populating the Public header so that it is

an intersection of the capabilities of all the RTSP agents in a

       chain.
  • Added the Media-Range header for listing the current

availability of the media range.

  • Added the Notify-Reason header for giving the reason when

sending PLAY_NOTIFY requests.

  • A new header Seek-Style has been defined to direct and inform

how any seek operation should/have been performed.

 o  The Protocol Syntax has been changed in the following way:
  • All ABNF definitions are updated according to the rules defined

in RFC 5234 [RFC5234] and have been gathered in a separate

       section (Section 20).
  • The ABNF for the User-Agent and Server headers have been

corrected.

  • Some definitions in the introduction regarding the RTSP session

have been changed.

  • The protocol has been made fully IPv6 capable.
  • The CHAR rule has been changed to exclude NULL.
 o  The Status codes have been changed in the following ways:
  • The use of status code 303 (See Other) has been deprecated as

it does not make sense to use in RTSP.

  • The never-defined status code 411 "Length Required" has been

completely removed.

  • When sending response 451 (Parameter Not Understood) and 458

(Parameter Is Read-Only), the response body should contain the

       offending parameters.

Schulzrinne, et al. Standards Track [Page 313] RFC 7826 RTSP 2.0 December 2016

  • Clarification on when a 3rr redirect status code can be

received has been added. This includes receiving 3rr as a

       result of a request within an established session.  This
       provides clarification to a previous unspecified behavior.
  • Removed the 201 (Created) and 250 (Low On Storage Space) status

codes as they are only relevant to recording, which is

       deprecated.
  • Several new status codes have been defined: 464 (Data Transport

Not Ready Yet), 465 (Notification Reason Unknown), 470

       (Connection Authorization Required), 471 (Connection
       Credentials Not Accepted), and 472 (Failure to Establish Secure
       Connection).
 o  The following functionality has been deprecated from the protocol:
  • The use of Queued Play.
  • The use of PLAY method for keep-alive in Play state.
  • The RECORD and ANNOUNCE methods and all related functionality.

Some of the syntax has been removed.

  • The possibility to use timed execution of methods with the time

parameter in the Range header.

  • The description on how rtspu works is not part of the core

specification and will require external description. Only that

       it exists is mentioned here and some requirements for the
       transport are provided.
 o  The following changes have been made in relation to methods:
  • The OPTIONS method has been clarified with regard to the use of

the Public and Allow headers.

  • Added text clarifying the usage of SET_PARAMETER for keep-alive

and usage without a body.

  • PLAY method is now allowed to be pipelined with the pipelining

of one or more SETUP requests following the initial that

       generates the session for aggregated control.
  • REDIRECT has been expanded and diversified for different

situations.

Schulzrinne, et al. Standards Track [Page 314] RFC 7826 RTSP 2.0 December 2016

  • Added a new method PLAY_NOTIFY. This method is used by the

RTSP server to asynchronously notify clients about session

       changes.
 o  Wrote a new section about how to set up different media-transport
    alternatives and their profiles as well as lower-layer protocols.
    This caused the appendix on RTP interaction to be moved to the new
    section instead of being in the part that describes RTP.  The new
    section also includes guidelines what to consider when writing
    usage guidelines for new protocols and profiles.
 o  Setup and usage of independent TCP connections for transport of
    RTP has been specified.
 o  Added a new section describing the available mechanisms to
    determine if functionality is supported, called "Capability
    Handling".  Renamed option-tags to feature tags.
 o  Added a Contributors section with people who have contributed
    actual text to the specification.
 o  Added a section "Use Cases" that describes the major use cases for
    RTSP.
 o  Clarified the usage of a=range and how to indicate live content
    that are not seekable with this header.
 o  Text specifying the special behavior of PLAY for live content.
 o  Security features of RTSP have been clarified:
  • HTTP-based authorization has been clarified requiring both

Basic and Digest support

  • TLS support has been mandated
  • If one implements RTP, then SRTP and defined MIKEY-based key-

exchange must be supported

  • Various minor mitigations discussed or resulted in protocol

changes.

Schulzrinne, et al. Standards Track [Page 315] RFC 7826 RTSP 2.0 December 2016

Acknowledgements

 This memorandum defines RTSP version 2.0, which is a revision of the
 Proposed Standard RTSP version 1.0 defined in [RFC2326].  The authors
 of RFC 2326 are Henning Schulzrinne, Anup Rao, and Robert Lanphier.
 Both RTSP version 1.0 and RTSP version 2.0 borrow format and
 descriptions from HTTP/1.1.
 Robert Sparks and especially Elwyn Davies provided very valuable and
 detailed reviews in the IETF Last Call that greatly improved the
 document and resolved many issues, especially regarding consistency.
 This document has benefited greatly from the comments of all those
 participating in the MMUSIC WG.  In addition to those already
 mentioned, the following individuals have contributed to this
 specification:
 Rahul Agarwal, Claudio Allocchio, Jeff Ayars, Milko Boic, Torsten
 Braun, Brent Browning, Bruce Butterfield, Steve Casner, Maureen
 Chesire, Jinhang Choi, Francisco Cortes, Elwyn Davies, Spencer
 Dawkins, Kelly Djahandari, Martin Dunsmuir, Adrian Farrel, Stephen
 Farrell, Ross Finlayson, Eric Fleischman, Jay Geagan, Andy Grignon,
 Christian Groves, V.  Guruprasad, Peter Haight, Mark Handley, Brad
 Hefta-Gaub, Volker Hilt, John K.  Ho, Patrick Hoffman, Go Hori,
 Philipp Hoschka, Anne Jones, Ingemar Johansson, Jae-Hwan Kim, Anders
 Klemets, Ruth Lang, Barry Leiba, Stephanie Leif, Jonathan Lennox,
 Eduardo F.  Llach, Chris Lonvick, Xavier Marjou, Thomas Marshall, Rob
 McCool, Martti Mela, David Oran, Joerg Ott, Joe Pallas, Maria
 Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan, Colin Perkins,
 Pekka Pessi, Igor Plotnikov, Pete Resnick, Peter Saint-Andre, Holger
 Schmidt, Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff
 Smith, Alexander Sokolsky, Dale Stammen, John Francis Stracke, Geetha
 Srikantan, Scott Taylor, David Walker, Stephan Wenger, Dale R.
 Worley, and Byungjo Yoon, and especially Flemming Andreasen.

Schulzrinne, et al. Standards Track [Page 316] RFC 7826 RTSP 2.0 December 2016

Contributors

 The following people have made written contributions that were
 included in the specification:
 o  Tom Marshall contributed text on the usage of 3rr status codes.
 o  Thomas Zheng contributed text on the usage of the Range in PLAY
    responses and proposed an earlier version of the PLAY_NOTIFY
    method.
 o  Sean Sheedy contributed text on the timeout behavior of RTSP
    messages and connections, the 463 (Destination Prohibited) status
    code, and proposed an earlier version of the PLAY_NOTIFY method.
 o  Greg Sherwood proposed an earlier version of the PLAY_NOTIFY
    method.
 o  Fredrik Lindholm contributed text about the RTSP security
    framework.
 o  John Lazzaro contributed the text for RTP over Independent TCP.
 o  Aravind Narasimhan contributed by rewriting "Media-Transport
    Alternatives" (Appendix C) and making editorial improvements on a
    number of places in the specification.
 o  Torbjorn Einarsson has done some editorial improvements of the
    text.

Schulzrinne, et al. Standards Track [Page 317] RFC 7826 RTSP 2.0 December 2016

Authors' Addresses

 Henning Schulzrinne
 Columbia University
 1214 Amsterdam Avenue
 New York, NY  10027
 United States of America
 Email: schulzrinne@cs.columbia.edu
 Anup Rao
 Cisco
 United States of America
 Email: anrao@cisco.com
 Rob Lanphier
 San Francisco, CA
 United States of America
 Email: robla@robla.net
 Magnus Westerlund
 Ericsson
 Faeroegatan 2
 Stockholm  SE-164 80
 Sweden
 Email: magnus.westerlund@ericsson.com
 Martin Stiemerling (editor)
 University of Applied Sciences Darmstadt
 Haardtring 100
 64295 Darmstadt
 Germany
 Email: mls.ietf@gmail.com
 URI:   http://www.stiemerling.org

Schulzrinne, et al. Standards Track [Page 318]

/data/webs/external/dokuwiki/data/pages/rfc/rfc7826.txt · Last modified: 2016/12/27 19:28 (external edit)