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rfc:rfc2326

Network Working Group H. Schulzrinne Request for Comments: 2326 Columbia U. Category: Standards Track A. Rao

                                                              Netscape
                                                           R. Lanphier
                                                          RealNetworks
                                                            April 1998
                Real Time Streaming Protocol (RTSP)

Status of this Memo

 This document specifies an Internet standards track protocol for the
 Internet community, and requests discussion and suggestions for
 improvements.  Please refer to the current edition of the "Internet
 Official Protocol Standards" (STD 1) for the standardization state
 and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

 Copyright (C) The Internet Society (1998).  All Rights Reserved.

Abstract

 The Real Time Streaming Protocol, or RTSP, is an application-level
 protocol for control over 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 1889).

Table of Contents

  • 1 Introduction …………………………………………. 5

+ 1.1 Purpose ……………………………………….. 5

      + 1.2 Requirements ..........................................  6
      + 1.3 Terminology ...........................................  6
      + 1.4 Protocol Properties ...................................  9
      + 1.5 Extending RTSP ........................................ 11
      + 1.6 Overall Operation ..................................... 11
      + 1.7 RTSP States ........................................... 12
      + 1.8 Relationship with Other Protocols ..................... 13
 * 2 Notational Conventions ....................................... 14
 * 3 Protocol Parameters .......................................... 14
      + 3.1 RTSP Version .......................................... 14

Schulzrinne, et. al. Standards Track [Page 1] RFC 2326 Real Time Streaming Protocol April 1998

      + 3.2 RTSP URL .............................................. 14
      + 3.3 Conference Identifiers ................................ 16
      + 3.4 Session Identifiers ................................... 16
      + 3.5 SMPTE Relative Timestamps ............................. 16
      + 3.6 Normal Play Time ...................................... 17
      + 3.7 Absolute Time ......................................... 18
      + 3.8 Option Tags ........................................... 18
           o 3.8.1 Registering New Option Tags with IANA .......... 18
 * 4 RTSP Message ................................................. 19
      + 4.1 Message Types ......................................... 19
      + 4.2 Message Headers ....................................... 19
      + 4.3 Message Body .......................................... 19
      + 4.4 Message Length ........................................ 20
 * 5 General Header Fields ........................................ 20
 * 6 Request ...................................................... 20
      + 6.1 Request Line .......................................... 21
      + 6.2 Request Header Fields ................................. 21
 * 7 Response ..................................................... 22
      + 7.1 Status-Line ........................................... 22
           o 7.1.1 Status Code and Reason Phrase .................. 22
           o 7.1.2 Response Header Fields ......................... 26
 * 8 Entity ....................................................... 27
      + 8.1 Entity Header Fields .................................. 27
      + 8.2 Entity Body ........................................... 28
 * 9 Connections .................................................. 28
      + 9.1 Pipelining ............................................ 28
      + 9.2 Reliability and Acknowledgements ...................... 28
 * 10 Method Definitions .......................................... 29
      + 10.1 OPTIONS .............................................. 30
      + 10.2 DESCRIBE ............................................. 31
      + 10.3 ANNOUNCE ............................................. 32
      + 10.4 SETUP ................................................ 33
      + 10.5 PLAY ................................................. 34
      + 10.6 PAUSE ................................................ 36
      + 10.7 TEARDOWN ............................................. 37
      + 10.8 GET_PARAMETER ........................................ 37
      + 10.9 SET_PARAMETER ........................................ 38
      + 10.10 REDIRECT ............................................ 39
      + 10.11 RECORD .............................................. 39
      + 10.12 Embedded (Interleaved) Binary Data .................. 40
 * 11 Status Code Definitions ..................................... 41
      + 11.1 Success 2xx .......................................... 41
           o 11.1.1 250 Low on Storage Space ...................... 41
      + 11.2 Redirection 3xx ...................................... 41
      + 11.3 Client Error 4xx ..................................... 42
           o 11.3.1 405 Method Not Allowed ........................ 42
           o 11.3.2 451 Parameter Not Understood .................. 42
           o 11.3.3 452 Conference Not Found ...................... 42

Schulzrinne, et. al. Standards Track [Page 2] RFC 2326 Real Time Streaming Protocol April 1998

           o 11.3.4 453 Not Enough Bandwidth ...................... 42
           o 11.3.5 454 Session Not Found ......................... 42
           o 11.3.6 455 Method Not Valid in This State ............ 42
           o 11.3.7 456 Header Field Not Valid for Resource ....... 42
           o 11.3.8 457 Invalid Range ............................. 43
           o 11.3.9 458 Parameter Is Read-Only .................... 43
           o 11.3.10 459 Aggregate Operation Not Allowed .......... 43
           o 11.3.11 460 Only Aggregate Operation Allowed ......... 43
           o 11.3.12 461 Unsupported Transport .................... 43
           o 11.3.13 462 Destination Unreachable .................. 43
           o 11.3.14 551 Option not supported ..................... 43
 * 12 Header Field Definitions .................................... 44
      + 12.1 Accept ............................................... 46
      + 12.2 Accept-Encoding ...................................... 46
      + 12.3 Accept-Language ...................................... 46
      + 12.4 Allow ................................................ 46
      + 12.5 Authorization ........................................ 46
      + 12.6 Bandwidth ............................................ 46
      + 12.7 Blocksize ............................................ 47
      + 12.8 Cache-Control ........................................ 47
      + 12.9 Conference ........................................... 49
      + 12.10 Connection .......................................... 49
      + 12.11 Content-Base ........................................ 49
      + 12.12 Content-Encoding .................................... 49
      + 12.13 Content-Language .................................... 50
      + 12.14 Content-Length ...................................... 50
      + 12.15 Content-Location .................................... 50
      + 12.16 Content-Type ........................................ 50
      + 12.17 CSeq ................................................ 50
      + 12.18 Date ................................................ 50
      + 12.19 Expires ............................................. 50
      + 12.20 From ................................................ 51
      + 12.21 Host ................................................ 51
      + 12.22 If-Match ............................................ 51
      + 12.23 If-Modified-Since ................................... 52
      + 12.24 Last-Modified........................................ 52
      + 12.25 Location ............................................ 52
      + 12.26 Proxy-Authenticate .................................. 52
      + 12.27 Proxy-Require ....................................... 52
      + 12.28 Public .............................................. 53
      + 12.29 Range ............................................... 53
      + 12.30 Referer ............................................. 54
      + 12.31 Retry-After ......................................... 54
      + 12.32 Require ............................................. 54
      + 12.33 RTP-Info ............................................ 55
      + 12.34 Scale ............................................... 56
      + 12.35 Speed ............................................... 57
      + 12.36 Server .............................................. 57

Schulzrinne, et. al. Standards Track [Page 3] RFC 2326 Real Time Streaming Protocol April 1998

      + 12.37 Session ............................................. 57
      + 12.38 Timestamp ........................................... 58
      + 12.39 Transport ........................................... 58
      + 12.40 Unsupported ......................................... 62
      + 12.41 User-Agent .......................................... 62
      + 12.42 Vary ................................................ 62
      + 12.43 Via ................................................. 62
      + 12.44 WWW-Authenticate .................................... 62
 * 13 Caching ..................................................... 62
 * 14 Examples .................................................... 63
      + 14.1 Media on Demand (Unicast) ............................ 63
      + 14.2 Streaming of a Container file ........................ 65
      + 14.3 Single Stream Container Files ........................ 67
      + 14.4 Live Media Presentation Using Multicast .............. 69
      + 14.5 Playing media into an existing session ............... 70
      + 14.6 Recording ............................................ 71
 * 15 Syntax ...................................................... 72
      + 15.1 Base Syntax .......................................... 72
 * 16 Security Considerations ..................................... 73
 * A RTSP Protocol State Machines ................................. 76
      + A.1 Client State Machine .................................. 76
      + A.2 Server State Machine .................................. 77
 * B Interaction with RTP ......................................... 79
 * C Use of SDP for RTSP Session Descriptions ..................... 80
      + C.1 Definitions ........................................... 80
           o C.1.1 Control URL .................................... 80
           o C.1.2 Media streams .................................. 81
           o C.1.3 Payload type(s) ................................ 81
           o C.1.4 Format-specific parameters ..................... 81
           o C.1.5 Range of presentation .......................... 82
           o C.1.6 Time of availability ........................... 82
           o C.1.7 Connection Information ......................... 82
           o C.1.8 Entity Tag ..................................... 82
      + C.2 Aggregate Control Not Available ....................... 83
      + C.3 Aggregate Control Available ........................... 83
 * D Minimal RTSP implementation .................................. 85
      + D.1 Client ................................................ 85
           o D.1.1 Basic Playback ................................. 86
           o D.1.2 Authentication-enabled ......................... 86
      + D.2 Server ................................................ 86
           o D.2.1 Basic Playback ................................. 87
           o D.2.2 Authentication-enabled ......................... 87
 * E Authors' Addresses ........................................... 88
 * F Acknowledgements ............................................. 89
 * References ..................................................... 90
 * Full Copyright Statement ....................................... 92

Schulzrinne, et. al. Standards Track [Page 4] RFC 2326 Real Time Streaming Protocol April 1998

1 Introduction

1.1 Purpose

 The Real-Time Streaming Protocol (RTSP) establishes and controls
 either a single or several time-synchronized streams of continuous
 media such as audio and video. It does not typically deliver the
 continuous streams itself, although interleaving of the continuous
 media stream with the control stream is possible (see Section 10.12).
 In other words, RTSP acts as a "network remote control" for
 multimedia servers.
 The set of streams to be controlled is defined by a presentation
 description. This memorandum does not define a format for a
 presentation description.
 There is no notion of an RTSP connection; instead, a server maintains
 a session labeled by an identifier. An RTSP session is in no way tied
 to a transport-level connection such as a TCP connection. During an
 RTSP session, an RTSP client may open and close many reliable
 transport connections to the server to issue RTSP requests.
 Alternatively, it may use a connectionless transport protocol such as
 UDP.
 The streams controlled by RTSP may use RTP [1], but the operation of
 RTSP does not depend on the transport mechanism used to carry
 continuous media.  The protocol is intentionally similar in syntax
 and operation to HTTP/1.1 [2] so that extension mechanisms to HTTP
 can in most cases also be added to RTSP. However, RTSP differs in a
 number of important aspects from HTTP:
  • RTSP introduces a number of new methods and has a different

protocol identifier.

  • An RTSP server needs to maintain state by default in almost all

cases, as opposed to the stateless nature of HTTP.

  • Both an RTSP server and client can issue requests.
  • Data is carried out-of-band by a different protocol. (There is an

exception to this.)

  • RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,

consistent with current HTML internationalization efforts [3].

  • The Request-URI always contains the absolute URI. Because of

backward compatibility with a historical blunder, HTTP/1.1 [2]

     carries only the absolute path in the request and puts the host
     name in a separate header field.
   This makes "virtual hosting" easier, where a single host with one
   IP address hosts several document trees.

Schulzrinne, et. al. Standards Track [Page 5] RFC 2326 Real Time Streaming Protocol April 1998

 The protocol supports the following operations:
 Retrieval of media from media server:
        The client can request a presentation description via HTTP or
        some other method. If the presentation is being multicast, the
        presentation description contains the multicast addresses and
        ports to be used for the continuous media. If the presentation
        is to be sent only to the client via unicast, the client
        provides the destination for security reasons.
 Invitation of a media server to a conference:
        A media server can be "invited" to join an existing
        conference, either to play back media into the presentation or
        to record all or a subset of the media in a presentation. This
        mode is useful for distributed teaching applications. Several
        parties in the conference may take turns "pushing the remote
        control buttons."
 Addition of media to an existing presentation:
        Particularly for live presentations, it is useful if the
        server can tell the client about additional media becoming
        available.
 RTSP requests may be handled by proxies, tunnels and caches as in
 HTTP/1.1 [2].

1.2 Requirements

 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
 document are to be interpreted as described in RFC 2119 [4].

1.3 Terminology

 Some of the terminology has been adopted from HTTP/1.1 [2]. Terms not
 listed here are defined as in HTTP/1.1.
 Aggregate control:
        The control of the multiple streams using a single timeline by
        the server. For audio/video feeds, this means that the client
        may issue a single play or pause message to control both the
        audio and video feeds.
 Conference:
        a multiparty, multimedia presentation, where "multi" implies
        greater than or equal to one.

Schulzrinne, et. al. Standards Track [Page 6] RFC 2326 Real Time Streaming Protocol April 1998

 Client:
        The client requests continuous media data from the media
        server.
 Connection:
        A transport layer virtual circuit established between two
        programs for the purpose of communication.
 Container file:
        A file which may contain multiple media streams which often
        comprise a presentation when played together. RTSP servers may
        offer aggregate control on these files, though the concept of
        a container file is not embedded in the protocol.
 Continuous media:
        Data where there is a timing relationship between source and
        sink; that is, the sink must 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), where there is a "tight"
        timing relationship between source and sink, or streaming
        (playback), where the relationship is less strict.
 Entity:
        The information transferred as the payload of a request or
        response. An entity consists of metainformation in the form of
        entity-header fields and content in the form of an entity-
        body, as described in Section 8.
 Media initialization:
        Datatype/codec specific initialization. This includes such
        things as clockrates, color tables, etc. Any transport-
        independent information which is required by a client for
        playback of a media stream occurs in the media initialization
        phase of stream setup.
 Media parameter:
        Parameter specific to a media type that may be changed before
        or during stream playback.
 Media server:
        The server providing playback or recording 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 or a different host as the
        web server the presentation is invoked from.

Schulzrinne, et. al. Standards Track [Page 7] RFC 2326 Real Time Streaming Protocol April 1998

 Media server indirection:
        Redirection of a media client to a different media server.
 (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 source within an RTP session. This is
        equivalent to the definition of a DSM-CC stream([5]).
 Message:
        The basic unit of RTSP communication, consisting of a
        structured sequence of octets matching the syntax defined in
        Section 15 and transmitted via a connection or a
        connectionless protocol.
 Participant:
        Member of a conference. A participant may be a machine, e.g.,
        a media record or playback server.
 Presentation:
        A set of one or more streams presented to the client as a
        complete media feed, using a presentation description as
        defined below. In most cases in the RTSP context, this implies
        aggregate control of those streams, but does not have to.
 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 (RFC 2327 [6]) use
        the term "session" for a live presentation. The presentation
        description may take several different formats, including but
        not limited to the session description format SDP.
 Response:
        An RTSP response. If an HTTP response is meant, that is
        indicated explicitly.
 Request:
        An RTSP request. If an HTTP request is meant, that is
        indicated explicitly.
 RTSP session:
        A complete RTSP "transaction", e.g., the viewing of a movie.
        A session typically consists of a client setting up a
        transport mechanism for the continuous media stream (SETUP),
        starting the stream with PLAY or RECORD, and closing the

Schulzrinne, et. al. Standards Track [Page 8] RFC 2326 Real Time Streaming Protocol April 1998

        stream with TEARDOWN.
 Transport initialization:
        The negotiation of transport information (e.g., port numbers,
        transport protocols) between the client and the server.

1.4 Protocol Properties

 RTSP has the following properties:
 Extendable:
        New methods and parameters can be easily added to RTSP.
 Easy to parse:
        RTSP can be parsed by standard HTTP or MIME parsers.
 Secure:
        RTSP re-uses web security mechanisms. All HTTP authentication
        mechanisms such as basic (RFC 2068 [2, Section 11.1]) and
        digest authentication (RFC 2069 [8]) are directly applicable.
        One may also reuse transport or network layer security
        mechanisms.
 Transport-independent:
        RTSP may use either an unreliable datagram protocol (UDP) (RFC
        768 [9]), a reliable datagram protocol (RDP, RFC 1151, not
        widely used [10]) or a reliable stream protocol such as TCP
        (RFC 793 [11]) as it implements application-level reliability.
 Multi-server capable:
        Each media stream within a presentation can reside on a
        different server. The client automatically establishes several
        concurrent control sessions with the different media servers.
        Media synchronization is performed at the transport level.
 Control of recording devices:
        The protocol can control both recording and playback devices,
        as well as devices that can alternate between the two modes
        ("VCR").
 Separation of stream control and conference initiation:
        Stream control is divorced from inviting a media server to a
        conference. The only requirement is that the conference
        initiation protocol either provides or can be used to create a
        unique conference identifier. In particular, SIP [12] or H.323
        [13] may be used to invite a server to a conference.

Schulzrinne, et. al. Standards Track [Page 9] RFC 2326 Real Time Streaming Protocol April 1998

 Suitable for professional applications:
        RTSP supports frame-level accuracy through SMPTE time stamps
        to allow remote digital editing.
 Presentation description neutral:
        The protocol does not impose a particular presentation
        description or metafile format and can convey the type of
        format to be used. However, the presentation description must
        contain at least one RTSP URI.
 Proxy and firewall friendly:
        The protocol should be readily handled by both application and
        transport-layer (SOCKS [14]) firewalls. A firewall may need to
        understand the SETUP method to open a "hole" for the UDP media
        stream.
 HTTP-friendly:
        Where sensible, RTSP reuses HTTP concepts, so that the
        existing infrastructure can be reused. This infrastructure
        includes PICS (Platform for Internet Content Selection
        [15,16]) for associating labels with content. However, RTSP
        does not just add methods to HTTP since the controlling
        continuous media requires server state in most cases.
 Appropriate server control:
        If a client can start a stream, it must be able to stop a
        stream. Servers should not start streaming to clients in such
        a way that clients cannot stop the stream.
 Transport negotiation:
        The client can negotiate the transport method prior to
        actually needing to process a continuous media stream.
 Capability negotiation:
        If basic features are disabled, there must be some clean
        mechanism for the client to determine which methods are not
        going to be implemented. This allows clients to present the
        appropriate user interface. For example, if seeking is not
        allowed, the user interface must be able to disallow moving a
        sliding position indicator.
   An earlier requirement in RTSP was multi-client capability.
   However, it was determined that a better approach was to make sure
   that the protocol is easily extensible to the multi-client
   scenario. Stream identifiers can be used by several control
   streams, so that "passing the remote" would be possible. The
   protocol would not address how several clients negotiate access;
   this is left to either a "social protocol" or some other floor

Schulzrinne, et. al. Standards Track [Page 10] RFC 2326 Real Time Streaming Protocol April 1998

   control mechanism.

1.5 Extending RTSP

 Since not all media servers have the same functionality, media
 servers by necessity will support different sets of requests. For
 example:
  • A server may only be capable of playback thus has no need to

support the RECORD request.

  • A server may not be capable of seeking (absolute positioning) if

it is to support live events only.

  • Some servers may not support setting stream parameters and thus

not support GET_PARAMETER and SET_PARAMETER.

 A server SHOULD implement all header fields described in Section 12.
 It is up to the creators of presentation descriptions not to ask the
 impossible of a server. This situation is similar in HTTP/1.1 [2],
 where the methods described in [H19.6] are not likely to be supported
 across all servers.
 RTSP can be extended in three ways, listed here in order of the
 magnitude of changes supported:
  • Existing methods can be extended with new parameters, as long as

these parameters can be safely ignored by the recipient. (This is

     equivalent to adding new parameters to an HTML tag.) If the
     client needs negative acknowledgement when a method extension is
     not supported, a tag corresponding to the extension may be added
     in the Require: field (see Section 12.32).
   * New methods can be added. If the recipient of the message does
     not understand the request, it responds with error code 501 (Not
     implemented) and the sender should not attempt to use this method
     again. A client may also use the OPTIONS method to inquire about
     methods supported by the server. The server SHOULD list the
     methods it supports using the Public response header.
   * A new version of the protocol can be defined, allowing almost all
     aspects (except the position of the protocol version number) to
     change.

1.6 Overall Operation

 Each presentation and media stream may be identified by an RTSP URL.
 The overall presentation and the properties of the media the
 presentation is made up of are defined by a presentation description
 file, the format of which is outside the scope of this specification.
 The presentation description file may be obtained by the client using

Schulzrinne, et. al. Standards Track [Page 11] RFC 2326 Real Time Streaming Protocol April 1998

 HTTP or other means such as email and may not necessarily be stored
 on the media server.
 For the purposes of this specification, a presentation description is
 assumed to describe one or more presentations, each of which
 maintains a common time axis. For simplicity of exposition and
 without loss of generality, it is assumed that the presentation
 description contains exactly one such presentation. A presentation
 may contain several media streams.
 The presentation description file contains a description of the media
 streams making up the presentation, including their encodings,
 language, and other parameters that enable the client to choose the
 most appropriate combination of media. In this presentation
 description, each media stream that is individually controllable by
 RTSP is identified by an RTSP URL, which points to the media server
 handling that particular media stream and names the stream stored on
 that server. Several media streams can be located on different
 servers; for example, audio and video streams can be split across
 servers for load sharing. The description also enumerates which
 transport methods the server is capable of.
 Besides the media parameters, the network destination address and
 port need to be determined. Several modes of operation can be
 distinguished:
 Unicast:
        The media is transmitted to the source of the RTSP request,
        with the port number chosen by the client. Alternatively, the
        media is transmitted on the same reliable stream as RTSP.
 Multicast, server chooses address:
        The media server picks the multicast address and port. This is
        the typical case for a live or near-media-on-demand
        transmission.
 Multicast, client chooses address:
        If the server is to participate in an existing multicast
        conference, the multicast address, port and encryption key are
        given by the conference description, established by means
        outside the scope of this specification.

1.7 RTSP States

 RTSP controls a stream which may be sent via a separate protocol,
 independent of the control channel. For example, RTSP control may
 occur on a TCP connection while the data flows via UDP. Thus, data
 delivery continues even if no RTSP requests are received by the media

Schulzrinne, et. al. Standards Track [Page 12] RFC 2326 Real Time Streaming Protocol April 1998

 server. Also, during its lifetime, a single media stream may be
 controlled by RTSP requests issued sequentially on different TCP
 connections. Therefore, the server needs to maintain "session state"
 to be able to correlate RTSP requests with a stream. The state
 transitions are described in Section A.
 Many methods in RTSP do not contribute to state. However, the
 following play a central role in defining the allocation and usage of
 stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and
 TEARDOWN.
 SETUP:
        Causes the server to allocate resources for a stream and start
        an RTSP session.
 PLAY and RECORD:
        Starts data transmission on a stream allocated via SETUP.
 PAUSE:
        Temporarily halts a stream without freeing server resources.
 TEARDOWN:
        Frees resources associated with the stream. The RTSP session
        ceases to exist on the server.
        RTSP methods that contribute to state use the Session header
        field (Section 12.37) to identify the RTSP session whose state
        is being manipulated. The server generates session identifiers
        in response to SETUP requests (Section 10.4).

1.8 Relationship with Other Protocols

 RTSP has some overlap in functionality with HTTP. It also may
 interact with HTTP in that the initial contact with streaming content
 is often to be made through a web page. The current protocol
 specification aims to allow different hand-off points between a web
 server and the media server implementing RTSP. For example, the
 presentation description can be retrieved using HTTP or RTSP, which
 reduces roundtrips in web-browser-based scenarios, yet also allows
 for standalone RTSP servers and clients which do not rely on HTTP at
 all.
 However, RTSP differs fundamentally from HTTP in that data delivery
 takes place out-of-band in a different protocol. HTTP is an
 asymmetric protocol where the client issues requests and the server
 responds. In RTSP, both the media client and media server can issue
 requests. RTSP requests are also not stateless; they may set
 parameters and continue to control a media stream long after the

Schulzrinne, et. al. Standards Track [Page 13] RFC 2326 Real Time Streaming Protocol April 1998

 request has been acknowledged.
   Re-using HTTP functionality has advantages in at least two areas,
   namely security and proxies. The requirements are very similar, so
   having the ability to adopt HTTP work on caches, proxies and
   authentication is valuable.
 While most real-time media will use RTP as a transport protocol, RTSP
 is not tied to RTP.
 RTSP assumes the existence of a presentation description format that
 can express both static and temporal properties of a presentation
 containing several media streams.

2 Notational Conventions

 Since many of the definitions and syntax are identical to HTTP/1.1,
 this specification only points to the section where they are defined
 rather than copying it. For brevity, [HX.Y] is to be taken to refer
 to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [2]).
 All the mechanisms specified in this document are described in both
 prose and an augmented Backus-Naur form (BNF) similar to that used in
 [H2.1]. It is described in detail in RFC 2234 [17], with the
 difference that this RTSP specification maintains the "1#" notation
 for comma-separated lists.
 In this memo, we use indented and smaller-type paragraphs to provide
 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 that they are in RTSP.

3 Protocol Parameters

3.1 RTSP Version

 [H3.1] applies, with HTTP replaced by RTSP.

3.2 RTSP URL

 The "rtsp" and "rtspu" schemes are used to refer to network resources
 via the RTSP protocol. This section defines the scheme-specific
 syntax and semantics for RTSP URLs.
 rtsp_URL  =   ( "rtsp:" | "rtspu:" )
               "//" host [ ":" port ] [ abs_path ]
 host      =   <A legal Internet host domain name of IP address
               (in dotted decimal form), as defined by Section 2.1

Schulzrinne, et. al. Standards Track [Page 14] RFC 2326 Real Time Streaming Protocol April 1998

               of RFC 1123 \cite{rfc1123}>
 port      =   *DIGIT
 abs_path is defined in [H3.2.1].
   Note that fragment and query identifiers do not have a well-defined
   meaning at this time, with the interpretation left to the RTSP
   server.
 The scheme rtsp requires that commands are issued via a reliable
 protocol (within the Internet, TCP), while the scheme rtspu identifies
 an unreliable protocol (within the Internet, UDP).
 If the port is empty or not given, port 554 is assumed. The semantics
 are that the identified resource can be controlled by RTSP at the
 server listening for TCP (scheme "rtsp") connections or UDP (scheme
 "rtspu") packets on that port of host, and the Request-URI for the
 resource is rtsp_URL.
 The use of IP addresses in URLs SHOULD be avoided whenever possible
 (see RFC 1924 [19]).
 A presentation or a stream is identified by a textual media
 identifier, using the character set and escape conventions [H3.2] of
 URLs (RFC 1738 [20]). URLs may refer to a stream or an aggregate of
 streams, i.e., a presentation. Accordingly, requests described in
 Section 10 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 URL:
   rtsp://media.example.com:554/twister/audiotrack
 identifies 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 URL:
   rtsp://media.example.com:554/twister
 identifies the presentation "twister", which may be composed of
 audio and video streams.
 This does not imply a standard way to reference streams in URLs.
 The presentation description defines the hierarchical relationships
 in the presentation and the URLs for the individual streams. A
 presentation description may name a stream "a.mov" and the whole
 presentation "b.mov".

Schulzrinne, et. al. Standards Track [Page 15] RFC 2326 Real Time Streaming Protocol April 1998

 The path components of the RTSP URL are opaque to the client and do
 not imply any particular file system structure for the server.
   This decoupling also allows presentation descriptions to be used
   with non-RTSP media control protocols simply by replacing the
   scheme in the URL.

3.3 Conference Identifiers

 Conference identifiers are opaque to RTSP and are encoded using
 standard URI encoding methods (i.e., LWS is escaped with %). They can
 contain any octet value. The conference identifier MUST be globally
 unique. For H.323, the conferenceID value is to be used.

conference-id = 1*xchar

   Conference identifiers are used to allow RTSP sessions to obtain
   parameters from multimedia conferences the media server is
   participating in. These conferences are created by protocols
   outside the scope of this specification, e.g., H.323 [13] or SIP
   [12]. Instead of the RTSP client explicitly providing transport
   information, for example, it asks the media server to use the
   values in the conference description instead.

3.4 Session Identifiers

 Session identifiers are opaque strings of arbitrary length. Linear
 white space must be URL-escaped. A session identifier MUST be chosen
 randomly and MUST be at least eight octets long to make guessing it
 more difficult. (See Section 16.)
   session-id   =   1*( ALPHA | DIGIT | safe )

3.5 SMPTE Relative Timestamps

 A SMPTE relative timestamp expresses time relative to the start of
 the clip. Relative timestamps are expressed as SMPTE time codes 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
 frame rate is 29.97 frames per second. Other SMPTE codes MAY be
 supported (such as "SMPTE 25") through the use of alternative use of
 "smpte time". For 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
 value is zero, it may be omitted. Subframes are measured in
 one-hundredth of a frame.

Schulzrinne, et. al. Standards Track [Page 16] RFC 2326 Real Time Streaming Protocol April 1998

 smpte-range  =   smpte-type "=" smpte-time "-" [ smpte-time ]
 smpte-type   =   "smpte" | "smpte-30-drop" | "smpte-25"
                                 ; other timecodes may be added
 smpte-time   =   1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT ]
                     [ "." 1*2DIGIT ]
 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

3.6 Normal Play Time

 Normal play time (NPT) indicates the stream absolute position
 relative to the beginning of the presentation. The timestamp consists
 of a decimal fraction. The part left of the decimal may be expressed
 in either seconds or hours, minutes, and seconds. The part right of
 the decimal point measures 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 be used only for live events.
 NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the
 viewer associates with a program. It is often digitally displayed on
 a VCR. 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 (high negative scale
 ratio) and is fixed in pause mode. NPT is (logically) equivalent to
 SMPTE time codes." [5]
 npt-range    =   ( npt-time "-" [ npt-time ] ) | ( "-" npt-time )
 npt-time     =   "now" | npt-sec | npt-hhmmss
 npt-sec      =   1*DIGIT [ "." *DIGIT ]
 npt-hhmmss   =   npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
 npt-hh       =   1*DIGIT     ; any positive number
 npt-mm       =   1*2DIGIT    ; 0-59
 npt-ss       =   1*2DIGIT    ; 0-59
 Examples:
   npt=123.45-125
   npt=12:05:35.3-
   npt=now-
   The syntax conforms to ISO 8601. The npt-sec notation is optimized
   for automatic generation, the ntp-hhmmss notation for consumption
   by human readers. The "now" constant allows clients to request to

Schulzrinne, et. al. Standards Track [Page 17] RFC 2326 Real Time Streaming Protocol April 1998

   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.

3.7 Absolute Time

   Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT).
   Fractions of a second may be indicated.
   utc-range    =   "clock" "=" utc-time "-" [ utc-time ]
   utc-time     =   utc-date "T" utc-time "Z"
   utc-date     =   8DIGIT                    ; < YYYYMMDD >
   utc-time     =   6DIGIT [ "." fraction ]   ; < HHMMSS.fraction >
   Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
   UTC:
   19961108T143720.25Z

3.8 Option Tags

 Option tags are unique identifiers used to designate new options in
 RTSP. These tags are used in Require (Section 12.32) and Proxy-
 Require (Section 12.27) header fields.
 Syntax:
   option-tag   =   1*xchar
 The creator of a new RTSP option should either prefix the option with
 a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name
 for a feature whose inventor can be reached at "foo.com"), or
 register the new option with the Internet Assigned Numbers Authority
 (IANA).

3.8.1 Registering New Option Tags with IANA

 When registering a new RTSP option, the following information should
 be provided:
  • Name and description of option. The name may be of any length,

but SHOULD be no more than twenty characters long. The name MUST

     not contain any spaces, control characters or periods.
   * Indication of who has change control over the option (for
     example, IETF, ISO, ITU-T, other international standardization
     bodies, a consortium or a particular company or group of
     companies);

Schulzrinne, et. al. Standards Track [Page 18] RFC 2326 Real Time Streaming Protocol April 1998

  • A reference to a further description, if available, for example

(in order of preference) an RFC, a published paper, a patent

     filing, a technical report, documented source code or a computer
     manual;
   * For proprietary options, contact information (postal and email
     address);

4 RTSP Message

 RTSP is a text-based protocol and uses the ISO 10646 character set in
 UTF-8 encoding (RFC 2279 [21]). Lines are terminated by CRLF, but
 receivers should be prepared to also interpret CR and LF by
 themselves as line terminators.
   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 done carefully, also allow easy
   implementation of research prototypes in scripting languages such
   as Tcl, Visual Basic and Perl.
   The 10646 character set avoids tricky character set switching, but
   is invisible to the application as long as US-ASCII is being used.
   This is also the encoding used for RTCP. ISO 8859-1 translates
   directly into Unicode with a high-order octet of zero. ISO 8859-1
   characters with the most-significant bit set are represented as
   1100001x 10xxxxxx. (See RFC 2279 [21])
 RTSP messages can be carried over any lower-layer transport protocol
 that is 8-bit clean.
 Requests contain methods, 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.

4.1 Message Types

 See [H4.1]

4.2 Message Headers

 See [H4.2]

4.3 Message Body

 See [H4.3]

Schulzrinne, et. al. Standards Track [Page 19] RFC 2326 Real Time Streaming Protocol April 1998

4.4 Message Length

 When a message body is included with a message, the length of that
 body is determined by one of the following (in order of precedence):
 1.     Any response message which MUST NOT include a message body
        (such as the 1xx, 204, and 304 responses) is always terminated
        by the first empty line after the header fields, regardless of
        the entity-header fields present in the message. (Note: An
        empty line consists of only CRLF.)
 2.     If a Content-Length header field (section 12.14) is present,
        its value in bytes represents the length of the message-body.
        If this header field is not present, a value of zero is
        assumed.
 3.     By the server closing the connection. (Closing the connection
        cannot be used to indicate the end of a request body, since
        that would leave no possibility for the server to send back a
        response.)
 Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
 transfer coding(see [H3.6]) and requires the presence of the
 Content-Length header field.
   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. Even though Content-Length must be present if there is
   any entity body, the rules ensure reasonable behavior even if the
   length is not given explicitly.

5 General Header Fields

 See [H4.5], except that Pragma, Transfer-Encoding and Upgrade headers
 are not defined:
    general-header     =     Cache-Control     ; Section 12.8
                       |     Connection        ; Section 12.10
                       |     Date              ; Section 12.18
                       |     Via               ; Section 12.43

6 Request

 A request message from a client to a server or vice versa includes,
 within the first line of that message, the method to be applied to
 the resource, the identifier of the resource, and the protocol
 version in use.

Schulzrinne, et. al. Standards Track [Page 20] RFC 2326 Real Time Streaming Protocol April 1998

     Request      =       Request-Line          ; Section 6.1
                  *(      general-header        ; Section 5
                  |       request-header        ; Section 6.2
                  |       entity-header )       ; Section 8.1
                          CRLF
                          [ message-body ]      ; Section 4.3

6.1 Request Line

Request-Line = Method SP Request-URI SP RTSP-Version CRLF
 Method         =         "DESCRIBE"              ; Section 10.2
                |         "ANNOUNCE"              ; Section 10.3
                |         "GET_PARAMETER"         ; Section 10.8
                |         "OPTIONS"               ; Section 10.1
                |         "PAUSE"                 ; Section 10.6
                |         "PLAY"                  ; Section 10.5
                |         "RECORD"                ; Section 10.11
                |         "REDIRECT"              ; Section 10.10
                |         "SETUP"                 ; Section 10.4
                |         "SET_PARAMETER"         ; Section 10.9
                |         "TEARDOWN"              ; Section 10.7
                |         extension-method
extension-method = token
Request-URI = "*" | absolute_URI
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT

6.2 Request Header Fields

request-header  =          Accept                   ; Section 12.1
                |          Accept-Encoding          ; Section 12.2
                |          Accept-Language          ; Section 12.3
                |          Authorization            ; Section 12.5
                |          From                     ; Section 12.20
                |          If-Modified-Since        ; Section 12.23
                |          Range                    ; Section 12.29
                |          Referer                  ; Section 12.30
                |          User-Agent               ; Section 12.41
 Note that in contrast to HTTP/1.1 [2], RTSP requests always contain
 the absolute URL (that is, including the scheme, host and port)
 rather than just the absolute path.

Schulzrinne, et. al. Standards Track [Page 21] RFC 2326 Real Time Streaming Protocol April 1998

   HTTP/1.1 requires servers to understand the absolute URL, 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.
 The asterisk "*" in the Request-URI means that the request does not
 apply to a particular resource, but to the server itself, and is only
 allowed when the method used does not necessarily apply to a
 resource.  One example would be:
   OPTIONS * RTSP/1.0

7 Response

 [H6] applies except that HTTP-Version is replaced by RTSP-Version.
 Also, RTSP defines additional status codes and does not define some
 HTTP codes. The valid response codes and the methods they can be used
 with are defined in Table 1.
 After receiving and interpreting a request message, the recipient
 responds with an RTSP response message.
   Response    =     Status-Line         ; Section 7.1
               *(    general-header      ; Section 5
               |     response-header     ; Section 7.1.2
               |     entity-header )     ; Section 8.1
                     CRLF
                     [ message-body ]    ; Section 4.3

7.1 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.
 Status-Line =   RTSP-Version SP Status-Code SP Reason-Phrase CRLF

7.1.1 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 11. 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.

Schulzrinne, et. al. Standards Track [Page 22] RFC 2326 Real Time Streaming Protocol April 1998

 The first digit of the Status-Code defines the class of response. The
 last two digits do not have any categorization role. There are 5
 values for the first digit:
  • 1xx: Informational - Request received, continuing process
  • 2xx: Success - The action was successfully received, understood,

and accepted

  • 3xx: Redirection - Further action must be taken in order to

complete the request

  • 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/1.0, and an example set of corresponding Reason-Phrase's, are
 presented below. The reason phrases listed here are only recommended
 - they may be replaced by local equivalents without affecting the
 protocol. Note that RTSP adopts most HTTP/1.1 [2] status codes and
 adds RTSP-specific status codes starting at x50 to avoid conflicts
 with newly defined HTTP status codes.

Schulzrinne, et. al. Standards Track [Page 23] RFC 2326 Real Time Streaming Protocol April 1998

 Status-Code  =     "100"      ; Continue
              |     "200"      ; OK
              |     "201"      ; Created
              |     "250"      ; Low on Storage Space
              |     "300"      ; Multiple Choices
              |     "301"      ; Moved Permanently
              |     "302"      ; Moved Temporarily
              |     "303"      ; See Other
              |     "304"      ; Not Modified
              |     "305"      ; Use Proxy
              |     "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 Time-out
              |     "410"      ; Gone
              |     "411"      ; Length Required
              |     "412"      ; Precondition Failed
              |     "413"      ; Request Entity Too Large
              |     "414"      ; Request-URI Too Large
              |     "415"      ; Unsupported Media Type
              |     "451"      ; Parameter Not Understood
              |     "452"      ; Conference Not Found
              |     "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
              |     "500"      ; Internal Server Error
              |     "501"      ; Not Implemented
              |     "502"      ; Bad Gateway
              |     "503"      ; Service Unavailable
              |     "504"      ; Gateway Time-out
              |     "505"      ; RTSP Version not supported
              |     "551"      ; Option not supported
              |     extension-code

Schulzrinne, et. al. Standards Track [Page 24] RFC 2326 Real Time Streaming Protocol April 1998

 extension-code  =     3DIGIT
 Reason-Phrase  =     *<TEXT, excluding CR, LF>
 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 the exception that an
 unrecognized response 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 entity returned
 with the response, since that entity is likely to include human-
 readable information which will explain the unusual status.
 Code           reason
 100            Continue                         all
 200            OK                               all
 201            Created                          RECORD
 250            Low on Storage Space             RECORD
 300            Multiple Choices                 all
 301            Moved Permanently                all
 302            Moved Temporarily                all
 303            See Other                        all
 305            Use Proxy                        all

Schulzrinne, et. al. Standards Track [Page 25] RFC 2326 Real Time Streaming Protocol April 1998

 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
 411            Length Required                  all
 412            Precondition Failed              DESCRIBE, SETUP
 413            Request Entity Too Large         all
 414            Request-URI Too Long             all
 415            Unsupported Media Type           all
 451            Invalid parameter                SETUP
 452            Illegal Conference Identifier    SETUP
 453            Not Enough Bandwidth             SETUP
 454            Session Not Found                all
 455            Method Not Valid In This State   all
 456            Header Field Not Valid           all
 457            Invalid Range                    PLAY
 458            Parameter Is Read-Only           SET_PARAMETER
 459            Aggregate Operation Not Allowed  all
 460            Only Aggregate Operation Allowed all
 461            Unsupported Transport            all
 462            Destination Unreachable          all
 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 support               all
    Table 1: Status codes and their usage with RTSP methods

7.1.2 Response Header Fields

 The response-header fields allow the request recipient to pass
 additional information about the response which cannot be placed in
 the Status-Line. These header fields give information about the
 server and about further access to the resource identified by the
 Request-URI.

Schulzrinne, et. al. Standards Track [Page 26] RFC 2326 Real Time Streaming Protocol April 1998

 response-header  =     Location             ; Section 12.25
                  |     Proxy-Authenticate   ; Section 12.26
                  |     Public               ; Section 12.28
                  |     Retry-After          ; Section 12.31
                  |     Server               ; Section 12.36
                  |     Vary                 ; Section 12.42
                  |     WWW-Authenticate     ; Section 12.44
 Response-header field names can be extended reliably only in
 combination with a change in the protocol version. However, new or
 experimental header fields MAY be given the semantics of response-
 header fields if all parties in the communication recognize them to
 be response-header fields. Unrecognized header fields are treated as
 entity-header fields.

8 Entity

 Request and Response messages MAY transfer an entity if not otherwise
 restricted by the request method or response status code. An entity
 consists of entity-header fields and an entity-body, although some
 responses will only include the entity-headers.
 In this section, both sender and recipient refer to either the client
 or the server, depending on who sends and who receives the entity.

8.1 Entity Header Fields

 Entity-header fields define optional metainformation about the
 entity-body or, if no body is present, about the resource identified
 by the request.
   entity-header       =    Allow               ; Section 12.4
                       |    Content-Base        ; Section 12.11
                       |    Content-Encoding    ; Section 12.12
                       |    Content-Language    ; Section 12.13
                       |    Content-Length      ; Section 12.14
                       |    Content-Location    ; Section 12.15
                       |    Content-Type        ; Section 12.16
                       |    Expires             ; Section 12.19
                       |    Last-Modified       ; Section 12.24
                       |    extension-header
   extension-header    =    message-header
 The extension-header mechanism allows additional entity-header fields
 to be defined without changing the protocol, but these fields cannot
 be assumed to be recognizable by the recipient. Unrecognized header
 fields SHOULD be ignored by the recipient and forwarded by proxies.

Schulzrinne, et. al. Standards Track [Page 27] RFC 2326 Real Time Streaming Protocol April 1998

8.2 Entity Body

 See [H7.2]

9 Connections

 RTSP requests can be transmitted in several different ways:
  • persistent transport connections used for several

request-response transactions;

  • one connection per request/response transaction;
  • connectionless mode.
 The type of transport connection is defined by the RTSP URI (Section
 3.2). For the scheme "rtsp", a persistent connection is assumed,
 while the scheme "rtspu" calls for RTSP requests to be sent without
 setting up a connection.
 Unlike HTTP, RTSP allows the media server to send requests to the
 media client. However, this is only supported for persistent
 connections, as the media server otherwise has no reliable way of
 reaching the client. Also, this is the only way that requests from
 media server to client are likely to traverse firewalls.

9.1 Pipelining

 A client that supports persistent connections or connectionless mode
 MAY "pipeline" its requests (i.e., send multiple requests without
 waiting for each response). A server MUST send its responses to those
 requests in the same order that the requests were received.

9.2 Reliability and Acknowledgements

 Requests are acknowledged by the receiver unless they are sent to a
 multicast group. If there is no acknowledgement, the sender may
 resend the same message after a timeout of one round-trip time (RTT).
 The round-trip time is estimated as in TCP (RFC 1123) [18], with an
 initial round-trip value of 500 ms. An implementation MAY cache the
 last RTT measurement as the initial value for future connections.
 If a reliable transport protocol is used to carry RTSP, requests MUST
 NOT be retransmitted; the RTSP application MUST instead rely on the
 underlying transport to provide reliability.
   If both the underlying reliable transport such as TCP and the RTSP
   application retransmit requests, it is possible that each packet
   loss results in two retransmissions. The receiver cannot typically
   take advantage of the application-layer retransmission since the

Schulzrinne, et. al. Standards Track [Page 28] RFC 2326 Real Time Streaming Protocol April 1998

   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.
   If RTSP is used over a small-RTT LAN, standard procedures for
   optimizing initial TCP round trip estimates, such as those used in
   T/TCP (RFC 1644) [22], can be beneficial.
 The Timestamp header (Section 12.38) is used to avoid the
 retransmission ambiguity problem [23, p. 301] and obviates the need
 for Karn's algorithm.
 Each request carries a sequence number in the CSeq header (Section
 12.17), which is incremented by one for each distinct request
 transmitted. If a request is repeated because of lack of
 acknowledgement, the request MUST carry the original sequence number
 (i.e., the sequence number is not incremented).
 Systems implementing RTSP MUST support carrying RTSP over TCP and MAY
 support UDP. The default port for the RTSP server is 554 for both UDP
 and TCP.
 A number of RTSP packets destined for the same control end point may
 be packed into a single lower-layer PDU or encapsulated into a TCP
 stream. RTSP data MAY be interleaved with RTP and RTCP packets.
 Unlike HTTP, an RTSP message MUST contain a Content-Length header
 whenever that message contains a payload. Otherwise, an RTSP packet
 is terminated with an empty line immediately following the last
 message header.

10 Method Definitions

 The method token indicates the method to be performed on the resource
 identified by the Request-URI. The method is case-sensitive.  New
 methods may be defined in the future. Method names may not start with
 a $ character (decimal 24) and must be a token. Methods are
 summarized in Table 2.

Schulzrinne, et. al. Standards Track [Page 29] RFC 2326 Real Time Streaming Protocol April 1998

    method            direction        object     requirement
    DESCRIBE          C->S             P,S        recommended
    ANNOUNCE          C->S, S->C       P,S        optional
    GET_PARAMETER     C->S, S->C       P,S        optional
    OPTIONS           C->S, S->C       P,S        required
                                                  (S->C: optional)
    PAUSE             C->S             P,S        recommended
    PLAY              C->S             P,S        required
    RECORD            C->S             P,S        optional
    REDIRECT          S->C             P,S        optional
    SETUP             C->S             S          required
    SET_PARAMETER     C->S, S->C       P,S        optional
    TEARDOWN          C->S             P,S        required
    Table 2: Overview of RTSP methods, their direction, and what
    objects (P: presentation, S: stream) they operate on
 Notes on Table 2: PAUSE is recommended, but not required in that a
 fully functional server can be built that does not support this
 method, for example, for live feeds. If a server does not support a
 particular method, it MUST return "501 Not Implemented" and a client
 SHOULD not try this method again for this server.

10.1 OPTIONS

 The behavior is equivalent to that described in [H9.2]. An OPTIONS
 request may be issued at any time, e.g., if the client is about to
 try a nonstandard request. It does not influence server state.
 Example:
   C->S:  OPTIONS * RTSP/1.0
          CSeq: 1
          Require: implicit-play
          Proxy-Require: gzipped-messages
   S->C:  RTSP/1.0 200 OK
          CSeq: 1
          Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
 Note that these are necessarily fictional features (one would hope
 that we would not purposefully overlook a truly useful feature just
 so that we could have a strong example in this section).

Schulzrinne, et. al. Standards Track [Page 30] RFC 2326 Real Time Streaming Protocol April 1998

10.2 DESCRIBE

 The DESCRIBE method retrieves the description of a presentation or
 media object identified by the request URL from a server. It may use
 the Accept header to specify the description formats that the client
 understands. The server responds with a description of the requested
 resource. The DESCRIBE reply-response pair constitutes the media
 initialization phase of RTSP.
 Example:
   C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
         CSeq: 312
         Accept: application/sdp, application/rtsl, application/mheg
   S->C: RTSP/1.0 200 OK
         CSeq: 312
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 376
         v=0
         o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
         s=SDP Seminar
         i=A Seminar on the session description protocol
         u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
         e=mjh@isi.edu (Mark Handley)
         c=IN IP4 224.2.17.12/127
         t=2873397496 2873404696
         a=recvonly
         m=audio 3456 RTP/AVP 0
         m=video 2232 RTP/AVP 31
         m=whiteboard 32416 UDP WB
         a=orient:portrait
 The DESCRIBE response MUST contain all media initialization
 information for the resource(s) that it describes. If a media client
 obtains a presentation description from a source other than DESCRIBE
 and that description contains a complete set of media initialization
 parameters, the client SHOULD use those parameters and not then
 request a description for the same media via RTSP.
 Additionally, servers SHOULD NOT use the DESCRIBE response as a means
 of media indirection.
   Clear ground rules need to be established so that clients have an
   unambiguous means of knowing when to request media initialization
   information via DESCRIBE, and when not to. By forcing a DESCRIBE

Schulzrinne, et. al. Standards Track [Page 31] RFC 2326 Real Time Streaming Protocol April 1998

   response to contain all media initialization for the set of streams
   that it describes, and discouraging use of DESCRIBE for media
   indirection, we avoid looping problems that might result from other
   approaches.
   Media initialization is a requirement for any RTSP-based system,
   but the RTSP specification does not dictate that this must be done
   via the DESCRIBE method. There are three ways that an RTSP client
   may receive initialization information:
  • via RTSP's DESCRIBE method;
  • via some other protocol (HTTP, email attachment, etc.);
  • via the command line or standard input (thus working as a browser

helper application launched with an SDP file or other media

     initialization format).
   In the interest of practical interoperability, it is highly
   recommended that minimal servers support the DESCRIBE method, and
   highly recommended that minimal clients support the ability to act
   as a "helper application" that accepts a media initialization file
   from standard input, command line, and/or other means that are
   appropriate to the operating environment of the client.

10.3 ANNOUNCE

 The ANNOUNCE method serves two purposes:
 When sent from client to server, ANNOUNCE posts the description of a
 presentation or media object identified by the request URL to a
 server. When sent from server to client, ANNOUNCE updates the session
 description in real-time.
 If a new media stream is added to a presentation (e.g., during a live
 presentation), the whole presentation description should be sent
 again, rather than just the additional components, so that components
 can be deleted.
 Example:
   C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0
         CSeq: 312
         Date: 23 Jan 1997 15:35:06 GMT
         Session: 47112344
         Content-Type: application/sdp
         Content-Length: 332
         v=0
         o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4

Schulzrinne, et. al. Standards Track [Page 32] RFC 2326 Real Time Streaming Protocol April 1998

         s=SDP Seminar
         i=A Seminar on the session description protocol
         u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
         e=mjh@isi.edu (Mark Handley)
         c=IN IP4 224.2.17.12/127
         t=2873397496 2873404696
         a=recvonly
         m=audio 3456 RTP/AVP 0
         m=video 2232 RTP/AVP 31
   S->C: RTSP/1.0 200 OK
         CSeq: 312

10.4 SETUP

 The SETUP request for a URI specifies the transport mechanism to be
 used for the streamed media. A client can issue a SETUP request for a
 stream that is already playing to change transport parameters, which
 a server MAY allow. If it does not allow this, it MUST respond with
 error "455 Method Not Valid In This State". For the benefit of any
 intervening firewalls, a client must indicate the transport
 parameters even if it has no influence over these parameters, for
 example, where the server advertises a fixed multicast address.
   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 Transport header specifies the transport parameters acceptable to
 the client for data transmission; the response will contain the
 transport parameters selected by the server.
  C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
        CSeq: 302
        Transport: RTP/AVP;unicast;client_port=4588-4589
  S->C: RTSP/1.0 200 OK
        CSeq: 302
        Date: 23 Jan 1997 15:35:06 GMT
        Session: 47112344
        Transport: RTP/AVP;unicast;
          client_port=4588-4589;server_port=6256-6257
 The server generates session identifiers in response to SETUP
 requests. If a SETUP request to a server includes a session
 identifier, the server MUST bundle this setup request into the

Schulzrinne, et. al. Standards Track [Page 33] RFC 2326 Real Time Streaming Protocol April 1998

 existing session or return error "459 Aggregate Operation Not
 Allowed" (see Section 11.3.10).

10.5 PLAY

 The PLAY method tells the server to start sending data via the
 mechanism specified in SETUP. A client MUST NOT issue a PLAY request
 until any outstanding SETUP requests have been acknowledged as
 successful.
 The PLAY request positions the normal play time to the beginning of
 the range specified and delivers stream data until the end of the
 range is reached. PLAY requests may be pipelined (queued); a server
 MUST queue PLAY requests to be executed in order. That is, a PLAY
 request arriving while a previous PLAY request is still active is
 delayed until the first has been completed.
   This allows precise editing.
 For example, regardless of how closely spaced the two PLAY requests
 in the example below arrive, the server will first play seconds 10
 through 15, then, immediately following, seconds 20 to 25, and
 finally seconds 30 through the end.
   C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
         CSeq: 835
         Session: 12345678
         Range: npt=10-15
   C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
         CSeq: 836
         Session: 12345678
         Range: npt=20-25
   C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
         CSeq: 837
         Session: 12345678
         Range: npt=30-
 See the description of the PAUSE request for further examples.
 A PLAY request without a Range header is legal. It starts playing a
 stream from the beginning unless the stream has been paused. If a
 stream has been paused via PAUSE, stream delivery resumes at the
 pause point. If a stream is playing, such a PLAY request causes no
 further action and can be used by the client to test server liveness.

Schulzrinne, et. al. Standards Track [Page 34] RFC 2326 Real Time Streaming Protocol April 1998

 The Range header may also contain a time parameter. This parameter
 specifies a time in UTC at which the playback should start. If the
 message is received after the specified time, playback is started
 immediately. The time parameter may be used to aid in synchronization
 of streams obtained from different sources.
 For a on-demand stream, the server replies with the actual range that
 will be played back. This may differ from the requested range if
 alignment of the requested range to valid frame boundaries is
 required for the media source. If no range is specified in the
 request, the current position is returned in the reply. The unit of
 the range in the reply is the same as that in the request.
 After playing the desired range, the presentation is automatically
 paused, as if a PAUSE request had been issued.
 The following example plays the whole presentation starting at SMPTE
 time code 0:10:20 until the end of the clip. The playback is to start
 at 15:36 on 23 Jan 1997.
   C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
         CSeq: 833
         Session: 12345678
         Range: smpte=0:10:20-;time=19970123T153600Z
   S->C: RTSP/1.0 200 OK
         CSeq: 833
         Date: 23 Jan 1997 15:35:06 GMT
         Range: smpte=0:10:22-;time=19970123T153600Z
 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/1.0
         CSeq: 835
         Session: 12345678
         Range: clock=19961108T142300Z-19961108T143520Z
   S->C: RTSP/1.0 200 OK
         CSeq: 835
         Date: 23 Jan 1997 15:35:06 GMT
 A media server only supporting playback MUST support the npt format
 and MAY support the clock and smpte formats.

Schulzrinne, et. al. Standards Track [Page 35] RFC 2326 Real Time Streaming Protocol April 1998

10.6 PAUSE

 The PAUSE request causes the stream delivery to be interrupted
 (halted) temporarily. If the request URL names a stream, only
 playback and recording of that stream is halted. For example, for
 audio, this is equivalent to muting. If the request URL names a
 presentation or group of streams, delivery of all currently active
 streams within the presentation or group is halted. After resuming
 playback or recording, 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/1.0
         CSeq: 834
         Session: 12345678
   S->C: RTSP/1.0 200 OK
         CSeq: 834
         Date: 23 Jan 1997 15:35:06 GMT
 The PAUSE request may contain a Range header specifying when the
 stream or presentation is to be halted. We refer to this point as the
 "pause point". The header must contain exactly one value rather than
 a time range. The normal play time for the stream is set to the pause
 point. The pause request becomes effective the first time the server
 is encountering the time point specified in any of the currently
 pending PLAY requests. If the Range header specifies a time outside
 any currently pending PLAY requests, the error "457 Invalid Range" is
 returned. If a media unit (such as an audio or video frame) starts
 presentation at exactly the pause point, it is not played or
 recorded.  If the Range header is missing, stream delivery is
 interrupted immediately on receipt of the message and the pause point
 is set to the current normal play time.
 A PAUSE request discards all queued PLAY requests. However, the pause
 point in the media stream MUST be maintained. A subsequent PLAY
 request without Range header resumes from the pause point.
 For example, if the server has play requests for ranges 10 to 15 and
 20 to 29 pending and then receives a pause request for NPT 21, it
 would start playing the second range and stop at NPT 21. If the pause
 request is for NPT 12 and the server is playing at NPT 13 serving the
 first play request, the server stops immediately. If the pause
 request is for NPT 16, the server stops after completing the first

Schulzrinne, et. al. Standards Track [Page 36] RFC 2326 Real Time Streaming Protocol April 1998

 play request and discards the second play request.
 As another example, if a server has received requests to play ranges
 10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
 request for NPT=14 would take effect while the server plays the first
 range, with the second PLAY request effectively being ignored,
 assuming the PAUSE request arrives before the server has started
 playing the second, overlapping range. Regardless of when the PAUSE
 request arrives, it sets the NPT to 14.
 If the server has already sent data beyond the time specified in the
 Range header, a PLAY would still resume at that point in time, as it
 is assumed that the client has discarded data after that point. This
 ensures continuous pause/play cycling without gaps.

10.7 TEARDOWN

 The TEARDOWN request stops the stream delivery for the given URI,
 freeing the resources associated with it. If the URI is the
 presentation URI for this presentation, any RTSP session identifier
 associated with the session is no longer valid. Unless all transport
 parameters are defined by the session description, a SETUP request
 has to be issued before the session can be played again.
 Example:
   C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
         CSeq: 892
         Session: 12345678
   S->C: RTSP/1.0 200 OK
         CSeq: 892

10.8 GET_PARAMETER

 The GET_PARAMETER request retrieves the value of a parameter of a
 presentation or stream specified in the URI. The content of the reply
 and response is left to the implementation. GET_PARAMETER with no
 entity body may be used to test client or server liveness ("ping").
 Example:
   S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
         CSeq: 431
         Content-Type: text/parameters
         Session: 12345678
         Content-Length: 15
         packets_received
         jitter

Schulzrinne, et. al. Standards Track [Page 37] RFC 2326 Real Time Streaming Protocol April 1998

   C->S: RTSP/1.0 200 OK
         CSeq: 431
         Content-Length: 46
         Content-Type: text/parameters
         packets_received: 10
         jitter: 0.3838
   The "text/parameters" section is only an example type for
   parameter. This method is intentionally loosely defined with the
   intention that the reply content and response content will be
   defined after further experimentation.

10.9 SET_PARAMETER

   This method requests to set the value of a parameter for a
   presentation or stream specified by the URI.
   A request SHOULD 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.
   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.
   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.
 Example:
   C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
         CSeq: 421
         Content-length: 20
         Content-type: text/parameters
         barparam: barstuff
   S->C: RTSP/1.0 451 Invalid Parameter

Schulzrinne, et. al. Standards Track [Page 38] RFC 2326 Real Time Streaming Protocol April 1998

         CSeq: 421
         Content-length: 10
         Content-type: text/parameters
         barparam
   The "text/parameters" section is only an example type for
   parameter. This method is intentionally loosely defined with the
   intention that the reply content and response content will be
   defined after further experimentation.

10.10 REDIRECT

 A redirect request informs the client that it must connect to another
 server location. It contains the mandatory header Location, which
 indicates that the client should issue requests for that URL. It may
 contain the parameter Range, which indicates when the redirection
 takes effect. If the client wants to continue to send or receive
 media for this URI, the client MUST issue a TEARDOWN request for the
 current session and a SETUP for the new session at the designated
 host.
 This example request redirects traffic for this URI to the new server
 at the given play time:
   S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
         CSeq: 732
         Location: rtsp://bigserver.com:8001
         Range: clock=19960213T143205Z-

10.11 RECORD

 This method initiates recording a range of media data according to
 the presentation description. The timestamp reflects start and end
 time (UTC). If no time range is given, use the start or end time
 provided in the presentation description. If the session has already
 started, commence recording immediately.
 The server decides whether to store the recorded data under the
 request-URI or another URI. If the server does not use the request-
 URI, the response SHOULD be 201 (Created) and contain an entity which
 describes the status of the request and refers to the new resource,
 and a Location header.
 A media server supporting recording of live presentations MUST
 support the clock range format; the smpte format does not make sense.

Schulzrinne, et. al. Standards Track [Page 39] RFC 2326 Real Time Streaming Protocol April 1998

 In this example, the media server was previously invited to the
 conference indicated.
   C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0
         CSeq: 954
         Session: 12345678
         Conference: 128.16.64.19/32492374

10.12 Embedded (Interleaved) Binary Data

 Certain firewall designs and other circumstances may force a server
 to interleave RTSP methods and stream data. This interleaving should
 generally be avoided unless necessary since it complicates client and
 server operation and imposes additional overhead. Interleaved binary
 data SHOULD only be used if RTSP is carried over TCP.
 Stream data such as RTP packets is encapsulated by an ASCII dollar
 sign (24 hexadecimal), followed by a one-byte channel identifier,
 followed by the length of the encapsulated binary data as a binary,
 two-byte integer in network byte order. The stream data follows
 immediately afterwards, without a CRLF, but including the upper-layer
 protocol headers. Each $ block contains exactly one upper-layer
 protocol data unit, e.g., one RTP packet.
 The channel identifier is defined in the Transport header with the
 interleaved parameter(Section 12.39).
 When the transport choice is RTP, RTCP messages are also interleaved
 by the server over the TCP connection. As a default, RTCP packets are
 sent on the first available channel higher than the RTP channel. The
 client MAY explicitly request RTCP packets on another channel. This
 is done by specifying two channels in the interleaved parameter of
 the Transport header(Section 12.39).
   RTCP is 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 TCP control connection when
   required by the network configuration and transfer them onto UDP
   when possible.
   C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP/TCP;interleaved=0-1
   S->C: RTSP/1.0 200 OK
         CSeq: 2
         Date: 05 Jun 1997 18:57:18 GMT
         Transport: RTP/AVP/TCP;interleaved=0-1

Schulzrinne, et. al. Standards Track [Page 40] RFC 2326 Real Time Streaming Protocol April 1998

         Session: 12345678
   C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0
         CSeq: 3
         Session: 12345678
   S->C: RTSP/1.0 200 OK
         CSeq: 3
         Session: 12345678
         Date: 05 Jun 1997 18:59:15 GMT
         RTP-Info: url=rtsp://foo.com/bar.file;
           seq=232433;rtptime=972948234
   S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
   S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
   S->C: $\001{2 byte length}{"length" bytes  RTCP packet}

11 Status Code Definitions

 Where applicable, HTTP status [H10] codes are reused. Status codes
 that have the same meaning are not repeated here. See Table 1 for a
 listing of which status codes may be returned by which requests.

11.1 Success 2xx

11.1.1 250 Low on Storage Space

 The server returns this warning after receiving a RECORD request that
 it may not be able to fulfill completely due to insufficient storage
 space. If possible, the server should use the Range header to
 indicate what time period it may still be able to record. Since other
 processes on the server may be consuming storage space
 simultaneously, a client should take this only as an estimate.

11.2 Redirection 3xx

 See [H10.3].
 Within RTSP, redirection may be used for load balancing or
 redirecting stream requests to a server topologically closer to the
 client.  Mechanisms to determine topological proximity are beyond the
 scope of this specification.

Schulzrinne, et. al. Standards Track [Page 41] RFC 2326 Real Time Streaming Protocol April 1998

11.3 Client Error 4xx

11.3.1 405 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, e.g., if a RECORD request is
 issued even though the mode parameter in the Transport header only
 specified PLAY.

11.3.2 451 Parameter Not Understood

 The recipient of the request does not support one or more parameters
 contained in the request.

11.3.3 452 Conference Not Found

 The conference indicated by a Conference header field is unknown to
 the media server.

11.3.4 453 Not Enough Bandwidth

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

11.3.5 454 Session Not Found

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

11.3.6 455 Method Not Valid in This State

 The client or server cannot process this request in its current
 state.  The response SHOULD contain an Allow header to make error
 recovery easier.

11.3.7 456 Header Field Not Valid for Resource

 The server could not act on a required request header. For example,
 if PLAY contains the Range header field but the stream does not allow
 seeking.

Schulzrinne, et. al. Standards Track [Page 42] RFC 2326 Real Time Streaming Protocol April 1998

11.3.8 457 Invalid Range

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

11.3.9 458 Parameter Is Read-Only

 The parameter to be set by SET_PARAMETER can be read but not
 modified.

11.3.10 459 Aggregate Operation Not Allowed

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

11.3.11 460 Only Aggregate Operation Allowed

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

11.3.12 461 Unsupported Transport

 The Transport field did not contain a supported transport
 specification.

11.3.13 462 Destination Unreachable

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

11.3.14 551 Option not supported

 An option given in the Require or the Proxy-Require fields was not
 supported. The Unsupported header should be returned stating the
 option for which there is no support.

Schulzrinne, et. al. Standards Track [Page 43] RFC 2326 Real Time Streaming Protocol April 1998

12 Header Field Definitions

 HTTP/1.1 [2] or other, non-standard header fields not listed here
 currently have no well-defined meaning and SHOULD be ignored by the
 recipient.
 Table 3 summarizes the header fields used by RTSP. Type "g"
 designates general request headers to be found in both requests and
 responses, type "R" designates request headers, type "r" designates
 response headers, and type "e" designates entity header fields.
 Fields marked with "req." in the column labeled "support" MUST be
 implemented by the recipient for a particular method, while fields
 marked "opt." are optional. Note that not all fields marked "req."
 will be sent in every request of this type. The "req."  means only
 that client (for response headers) and server (for request headers)
 MUST implement the fields. The last column lists the method for which
 this header field is meaningful; the designation "entity" refers to
 all methods that return a message body. Within this specification,
 DESCRIBE and GET_PARAMETER fall into this class.

Schulzrinne, et. al. Standards Track [Page 44] RFC 2326 Real Time Streaming Protocol April 1998

 Header               type   support   methods
 Accept               R      opt.      entity
 Accept-Encoding      R      opt.      entity
 Accept-Language      R      opt.      all
 Allow                r      opt.      all
 Authorization        R      opt.      all
 Bandwidth            R      opt.      all
 Blocksize            R      opt.      all but OPTIONS, TEARDOWN
 Cache-Control        g      opt.      SETUP
 Conference           R      opt.      SETUP
 Connection           g      req.      all
 Content-Base         e      opt.      entity
 Content-Encoding     e      req.      SET_PARAMETER
 Content-Encoding     e      req.      DESCRIBE, ANNOUNCE
 Content-Language     e      req.      DESCRIBE, ANNOUNCE
 Content-Length       e      req.      SET_PARAMETER, ANNOUNCE
 Content-Length       e      req.      entity
 Content-Location     e      opt.      entity
 Content-Type         e      req.      SET_PARAMETER, ANNOUNCE
 Content-Type         r      req.      entity
 CSeq                 g      req.      all
 Date                 g      opt.      all
 Expires              e      opt.      DESCRIBE, ANNOUNCE
 From                 R      opt.      all
 If-Modified-Since    R      opt.      DESCRIBE, SETUP
 Last-Modified        e      opt.      entity
 Proxy-Authenticate
 Proxy-Require        R      req.      all
 Public               r      opt.      all
 Range                R      opt.      PLAY, PAUSE, RECORD
 Range                r      opt.      PLAY, PAUSE, RECORD
 Referer              R      opt.      all
 Require              R      req.      all
 Retry-After          r      opt.      all
 RTP-Info             r      req.      PLAY
 Scale                Rr     opt.      PLAY, RECORD
 Session              Rr     req.      all but SETUP, OPTIONS
 Server               r      opt.      all
 Speed                Rr     opt.      PLAY
 Transport            Rr     req.      SETUP
 Unsupported          r      req.      all
 User-Agent           R      opt.      all
 Via                  g      opt.      all
 WWW-Authenticate     r      opt.      all

Schulzrinne, et. al. Standards Track [Page 45] RFC 2326 Real Time Streaming Protocol April 1998

 Overview of RTSP header fields

12.1 Accept

 The Accept request-header field can be used to specify certain
 presentation description content types which are acceptable for the
 response.
   The "level" parameter for presentation descriptions is properly
   defined as part of the MIME type registration, not here.
 See [H14.1] for syntax.
 Example of use:
   Accept: application/rtsl, application/sdp;level=2

12.2 Accept-Encoding

   See [H14.3]

12.3 Accept-Language

 See [H14.4]. Note that the language specified applies to the
 presentation description and any reason phrases, not the media
 content.

12.4 Allow

 The Allow response header field lists the methods supported by the
 resource identified by the request-URI. The purpose of this field is
 to strictly inform the recipient of valid methods associated with the
 resource. An Allow header field must be present in a 405 (Method not
 allowed) response.
 Example of use:
   Allow: SETUP, PLAY, RECORD, SET_PARAMETER

12.5 Authorization

   See [H14.8]

12.6 Bandwidth

 The Bandwidth request header field describes the estimated bandwidth
 available to the client, expressed as a positive integer and measured
 in bits per second. The bandwidth available to the client may change
 during an RTSP session, e.g., due to modem retraining.

Schulzrinne, et. al. Standards Track [Page 46] RFC 2326 Real Time Streaming Protocol April 1998

 Bandwidth = "Bandwidth" ":" 1*DIGIT
 Example:
   Bandwidth: 4000

12.7 Blocksize

 This 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 which 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 (416) if the value is syntactically invalid.

12.8 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 SETUP request and its
 response. Note: Cache-Control does not govern the caching of
 responses as for HTTP, but rather of the stream identified by the
 SETUP request.  Responses to RTSP requests are not cacheable, except
 for responses to DESCRIBE.
 Cache-Control            =   "Cache-Control" ":" 1#cache-directive
 cache-directive          =   cache-request-directive
                          |   cache-response-directive
 cache-request-directive  =   "no-cache"
                          |   "max-stale"
                          |   "min-fresh"
                          |   "only-if-cached"
                          |   cache-extension
 cache-response-directive =   "public"
                          |   "private"
                          |   "no-cache"
                          |   "no-transform"
                          |   "must-revalidate"

Schulzrinne, et. al. Standards Track [Page 47] RFC 2326 Real Time Streaming Protocol April 1998

                          |   "proxy-revalidate"
                          |   "max-age" "=" delta-seconds
                          |   cache-extension
 cache-extension          =   token [ "=" ( token | quoted-string ) ]
 no-cache:
        Indicates that the media stream 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.
 public:
        Indicates that the media stream is cacheable by any cache.
 private:
        Indicates that the media stream is intended for a single user
        and MUST NOT be cached by a shared cache. A private (non-
        shared) cache may cache the media stream.
 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 entity-body.
        Therefore, if a response includes the no-transform directive,
        an intermediate cache or proxy MUST NOT change the encoding of
        the stream. 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 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 it
        receives this directive, a cache SHOULD either respond using a
        cached media stream 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 48] RFC 2326 Real Time Streaming Protocol April 1998

 max-stale:
        Indicates that the client is willing to accept a media stream
        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
        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
        entry after it becomes stale to respond to a subsequent
        request without first revalidating it with the origin server.
        That is, the cache must do an end-to-end revalidation every
        time, if, based solely on the origin server's Expires, the
        cached response is stale.)

12.9 Conference

 This request header field establishes a logical connection between a
 pre-established conference and an RTSP stream. The conference-id must
 not be changed for the same RTSP session.
 Conference = "Conference" ":" conference-id Example:
   Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
 A response code of 452 (452 Conference Not Found) is returned if the
 conference-id is not valid.

12.10 Connection

 See [H14.10]

12.11 Content-Base

 See [H14.11]

12.12 Content-Encoding

 See [H14.12]

Schulzrinne, et. al. Standards Track [Page 49] RFC 2326 Real Time Streaming Protocol April 1998

12.13 Content-Language

 See [H14.13]

12.14 Content-Length

 This field contains the length of the content of the method (i.e.
 after the double CRLF following the last header). Unlike HTTP, it
 MUST be included in all messages that carry content beyond the header
 portion of the message. If it is missing, a default value of zero is
 assumed. It is interpreted according to [H14.14].

12.15 Content-Location

 See [H14.15]

12.16 Content-Type

 See [H14.18]. Note that the content types suitable for RTSP are
 likely to be restricted in practice to presentation descriptions and
 parameter-value types.

12.17 CSeq

 The CSeq field specifies the sequence number for an RTSP request-
 response pair. This field MUST be present in all requests and
 responses. For every RTSP request containing the given sequence
 number, there will be a corresponding response having the same
 number.  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).

12.18 Date

 See [H14.19].

12.19 Expires

 The Expires entity-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
        description should be considered stale.

Schulzrinne, et. al. Standards Track [Page 50] RFC 2326 Real Time Streaming Protocol April 1998

 A stale cache entry may not normally be returned by a cache (either a
 proxy cache or an user agent cache) unless it is first validated with
 the origin server (or with an intermediate cache that has a fresh
 copy of the entity). See section 13 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 HTTP-date in
 [H3.3]; it MUST be in RFC1123-date format:
 Expires = "Expires" ":" HTTP-date
 An example of its use is
   Expires: Thu, 01 Dec 1994 16:00:00 GMT
 RTSP/1.0 clients and caches MUST treat other invalid date formats,
 especially 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/1.0 servers should not send Expires dates more than one year in
 the future.
 The presence of an Expires header field with a date value of some
 time in the future on a media stream that otherwise would by default
 be non-cacheable indicates that the media stream is cacheable, unless
 indicated otherwise by a Cache-Control header field (Section 12.8).

12.20 From

 See [H14.22].

12.21 Host

 This HTTP request header field is not needed for RTSP. It should be
 silently ignored if sent.

12.22 If-Match

 See [H14.25].

Schulzrinne, et. al. Standards Track [Page 51] RFC 2326 Real Time Streaming Protocol April 1998

 This field is especially useful for ensuring the integrity of the
 presentation description, in both the case where it is fetched via
 means external to RTSP (such as HTTP), or in the case where the
 server implementation is guaranteeing the integrity of the
 description between the time of the DESCRIBE message and the SETUP
 message.
 The identifier is an opaque identifier, and thus is not specific to
 any particular session description language.

12.23 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 will be returned without any message-body.
 If-Modified-Since = "If-Modified-Since" ":" HTTP-date
 An example of the field is:
   If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT

12.24 Last-Modified

 The Last-Modified entity-header field indicates the date and time at
 which the origin server believes the presentation description or
 media stream was last modified. See [H14.29]. For the methods
 DESCRIBE or ANNOUNCE, the header field indicates the last
 modification date and time of the description, for SETUP that of the
 media stream.

12.25 Location

 See [H14.30].

12.26 Proxy-Authenticate

 See [H14.33].

12.27 Proxy-Require

 The Proxy-Require header 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 if not supported. Servers

Schulzrinne, et. al. Standards Track [Page 52] RFC 2326 Real Time Streaming Protocol April 1998

 should treat this field identically to the Require field.
 See Section 12.32 for more details on the mechanics of this message
 and a usage example.

12.28 Public

 See [H14.35].

12.29 Range

 This request and response header field specifies a range of time.
 The range can be specified in a number of units. This specification
 defines the smpte (Section 3.5), npt (Section 3.6), and clock
 (Section 3.7) range units. Within RTSP, byte ranges [H14.36.1] are
 not meaningful and MUST NOT be used. The header may also contain a
 time parameter in UTC, specifying the time at which the operation is
 to be made effective. Servers supporting the Range header MUST
 understand the NPT range format and SHOULD understand the SMPTE range
 format. The Range response header indicates what range of time is
 actually being played or recorded. If the Range header is given in a
 time format that is not understood, the recipient should return "501
 Not Implemented".
 Ranges are half-open intervals, including the lower point, but
 excluding the upper point. In other words, a range of a-b starts
 exactly at time a, but stops just before b. Only the start time of a
 media unit such as a video or audio frame is relevant. As an 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.
 Range            = "Range" ":" 1\#ranges-specifier
                        [ ";" "time" "=" utc-time ]
 ranges-specifier = npt-range | utc-range | smpte-range
 Example:
   Range: clock=19960213T143205Z-;time=19970123T143720Z
   The notation is similar to that used for the HTTP/1.1 [2] byte-
   range header. It allows clients to select an excerpt from the media
   object, and to play from a given point to the end as well as from
   the current location to a given point. The start of playback can be
   scheduled for any time in the future, although a server may refuse
   to keep server resources for extended idle periods.

Schulzrinne, et. al. Standards Track [Page 53] RFC 2326 Real Time Streaming Protocol April 1998

12.30 Referer

 See [H14.37]. The URL refers to that of the presentation description,
 typically retrieved via HTTP.

12.31 Retry-After

 See [H14.38].

12.32 Require

 The Require header is used by clients to query the server about
 options that it may or may not support. The server MUST respond to
 this header by using the Unsupported header to negatively acknowledge
 those options which are NOT supported.
   This is to make sure that the client-server interaction will
   proceed without delay when all options are understood by both
   sides, and only slow down if options are not understood (as in the
   case above). 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.
 Require =   "Require" ":"  1#option-tag
 Example:
   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Require: funky-feature
           Funky-Parameter: funkystuff
   S->C:   RTSP/1.0 551 Option not supported
           CSeq: 302
           Unsupported: funky-feature
   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 303
   S->C:   RTSP/1.0 200 OK
           CSeq: 303
 In this example, "funky-feature" is the feature tag which 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.

Schulzrinne, et. al. Standards Track [Page 54] RFC 2326 Real Time Streaming Protocol April 1998

 Proxies and other intermediary devices SHOULD ignore features that
 are not understood in this field. If a particular extension requires
 that intermediate devices support it, the extension should be tagged
 in the Proxy-Require field instead (see Section 12.27).

12.33 RTP-Info

 This field is used to set RTP-specific parameters in the PLAY
 response.
 url:
        Indicates the stream URL which for which the following RTP
        parameters correspond.
 seq:
        Indicates the sequence number of the first packet of the
        stream. 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.
 rtptime:
        Indicates the RTP timestamp corresponding to the time value in
        the Range response header. (Note: For aggregate control, a
        particular stream may not actually generate a packet for the
        Range time value returned or implied. Thus, there is no
        guarantee that the packet with the sequence number indicated
        by seq actually has the timestamp indicated by rtptime.) The
        client uses this value to calculate the mapping of RTP time to
        NPT.
   A mapping from RTP timestamps to NTP timestamps (wall clock) is
   available via RTCP. However, this information is not sufficient to
   generate a mapping from RTP timestamps to NPT. 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.

Schulzrinne, et. al. Standards Track [Page 55] RFC 2326 Real Time Streaming Protocol April 1998

 Syntax:
 RTP-Info        = "RTP-Info" ":" 1#stream-url 1*parameter
 stream-url      = "url" "=" url
 parameter       = ";" "seq" "=" 1*DIGIT
                 | ";" "rtptime" "=" 1*DIGIT
 Example:
   RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
             url=rtsp://foo.com/bar.avi/streamid=1;seq=30211

12.34 Scale

 A scale value of 1 indicates normal play or record at the normal
 forward viewing rate. If not 1, the value corresponds to the rate
 with respect to normal viewing rate. For example, a ratio of 2
 indicates twice the normal viewing rate ("fast forward") and a ratio
 of 0.5 indicates half the normal viewing rate. In other words, a
 ratio of 2 has normal play time increase at twice the wallclock rate.
 For every second of elapsed (wallclock) time, 2 seconds of content
 will be delivered. A negative value indicates reverse direction.
 Unless requested otherwise by the Speed parameter, the data rate
 SHOULD not be changed. Implementation of scale changes depends on the
 server and media type. For video, a server may, for example, deliver
 only key frames or selected key frames. For audio, it may time-scale
 the audio while preserving pitch or, less desirably, deliver
 fragments of audio.
 The server should try to approximate the viewing rate, but may
 restrict the range of scale values that it supports. The response
 MUST contain the actual scale value chosen by the server.
 If the request contains a Range parameter, the new scale value will
 take effect at that time.
 Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
 Example of playing in reverse at 3.5 times normal rate:
   Scale: -3.5

Schulzrinne, et. al. Standards Track [Page 56] RFC 2326 Real Time Streaming Protocol April 1998

12.35 Speed

 This request header fields parameter requests the server to deliver
 data to the client at a particular speed, contingent on the server's
 ability and desire to serve the media stream at the given speed.
 Implementation by the server is OPTIONAL. The default is the bit rate
 of the stream.
 The parameter value is expressed as a decimal ratio, e.g., a value of
 2.0 indicates that data is to be delivered twice as fast as normal. A
 speed of zero is invalid. If the request contains a Range parameter,
 the new speed value will take effect at that time.
 Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]
 Example:
   Speed: 2.5
 Use of this field changes the bandwidth used for data delivery. It is
 meant for use in specific circumstances where preview of the
 presentation at a higher or lower rate is necessary. Implementors
 should keep in mind that bandwidth for the session may be negotiated
 beforehand (by means other than RTSP), and therefore re-negotiation
 may be necessary. When data is delivered over UDP, it is highly
 recommended that means such as RTCP be used to track packet loss
 rates.

12.36 Server

 See [H14.39]

12.37 Session

 This request and response header field identifies an RTSP session
 started by the media server in a SETUP response and concluded by
 TEARDOWN on the presentation URL. The session identifier is chosen by
 the media server (see Section 3.4). Once a client receives a Session
 identifier, it MUST return it for any request related to that
 session.  A server does not have to set up a session identifier if it
 has other means of identifying a session, such as dynamically
 generated URLs.

Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]

 The timeout parameter is only allowed in a response header. The
 server uses it to indicate to the client how long the server is
 prepared to wait between RTSP commands before closing the session due
 to lack of activity (see Section A). The timeout is measured in

Schulzrinne, et. al. Standards Track [Page 57] RFC 2326 Real Time Streaming Protocol April 1998

 seconds, with a default of 60 seconds (1 minute).
 Note that a session identifier identifies a RTSP session across
 transport sessions or connections. Control messages for more than one
 RTSP URL may be sent within a single RTSP session. Hence, it is
 possible that clients use the same session for controlling many
 streams constituting a presentation, as long as all the streams come
 from the same server. (See example in Section 14). However, multiple
 "user" sessions for the same URL from the same client MUST use
 different session identifiers.
   The session identifier is needed to distinguish several delivery
   requests for the same URL coming from the same client.
 The response 454 (Session Not Found) is returned if the session
 identifier is invalid.

12.38 Timestamp

 The timestamp general header describes when the client sent the
 request to the server. The value of the timestamp is of significance
 only to the client and may use any timescale. The server 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 is
 used by the client to compute the round-trip time to the server so
 that it can adjust the timeout value for retransmissions.
 Timestamp  = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
 delay      =  *(DIGIT) [ "." *(DIGIT) ]

12.39 Transport

 This request 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.
 Transports are comma separated, listed in order of preference.
 Parameters may be added to each transport, separated by a semicolon.
 The Transport header MAY also be used to change certain transport
 parameters. A server MAY refuse to change parameters of an existing
 stream.
 The server MAY return a Transport response header in the response to
 indicate the values actually chosen.

Schulzrinne, et. al. Standards Track [Page 58] RFC 2326 Real Time Streaming Protocol April 1998

 A Transport request header field may contain a list of transport
 options acceptable to the client. In that case, the server MUST
 return a single option which was actually chosen.
 The syntax for the transport specifier is
     transport/profile/lower-transport.
 The default value for the "lower-transport" parameters is specific to
 the profile. For RTP/AVP, the default is UDP.
 Below are the configuration parameters associated with transport:
 General parameters:
 unicast | multicast:
        mutually exclusive indication of whether unicast or multicast
        delivery will be attempted. Default value is multicast.
        Clients that are capable of handling both unicast and
        multicast transmission MUST indicate such capability by
        including two full transport-specs with separate parameters
        for each.
 destination:
        The address to which a stream will be sent. The client may
        specify the multicast address with the destination parameter.
        To avoid becoming the unwitting perpetrator of a remote-
        controlled denial-of-service attack, a server SHOULD
        authenticate the client and SHOULD log such attempts before
        allowing the client to direct a media stream to an address not
        chosen by the server. This is particularly important if RTSP
        commands are issued via UDP, but implementations cannot rely
        on TCP as reliable means of client identification by itself. A
        server SHOULD not allow a client to direct media streams to an
        address that differs from the address commands are coming
        from.
 source:
        If the source address for the stream is different than can be
        derived from the RTSP endpoint address (the server in playback
        or the client in recording), the source MAY be specified.
   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 59] RFC 2326 Real Time Streaming Protocol April 1998

 layers:
        The number of multicast layers to be used for this media
        stream. The layers are sent to consecutive addresses starting
        at the destination address.
 mode:
        The mode parameter indicates the methods to be supported for
        this session. Valid values are PLAY and RECORD. If not
        provided, the default is PLAY.
 append:
        If the mode parameter includes RECORD, the append parameter
        indicates that the media data should append to the existing
        resource rather than overwrite it. If appending is requested
        and the server does not support this, it MUST refuse the
        request rather than overwrite the resource identified by the
        URI. The append parameter is ignored if the mode parameter
        does not contain RECORD.
 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 10.12.
        The argument provides the channel number to be used in the $
        statement. This parameter may be specified as a range, e.g.,
        interleaved=4-5 in cases where the transport choice for the
        media stream requires it.
   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.
 Multicast specific:
 ttl:
        multicast time-to-live
 RTP Specific:
 port:
        This parameter provides the RTP/RTCP port pair for a multicast
        session. It is specified as a range, e.g., port=3456-3457.
 client_port:
        This parameter provides the unicast RTP/RTCP port pair on
        which the client has chosen to receive media data and control
        information.  It is specified as a range, e.g.,
        client_port=3456-3457.

Schulzrinne, et. al. Standards Track [Page 60] RFC 2326 Real Time Streaming Protocol April 1998

 server_port:
        This parameter provides the unicast RTP/RTCP port pair on
        which the server has chosen to receive media data and control
        information.  It is specified as a range, e.g.,
        server_port=3456-3457.
 ssrc:
        The ssrc parameter indicates the RTP SSRC [24, Sec. 3] value
        that should be (request) or will be (response) used by the
        media server. This parameter is only valid for unicast
        transmission. It identifies the synchronization source to be
        associated with the media stream.
 Transport           =    "Transport" ":"
                          1\#transport-spec
 transport-spec      =    transport-protocol/profile[/lower-transport]
                          *parameter
 transport-protocol  =    "RTP"
 profile             =    "AVP"
 lower-transport     =    "TCP" | "UDP"
 parameter           =    ( "unicast" | "multicast" )
                     |    ";" "destination" [ "=" address ]
                     |    ";" "interleaved" "=" channel [ "-" channel ]
                     |    ";" "append"
                     |    ";" "ttl" "=" ttl
                     |    ";" "layers" "=" 1*DIGIT
                     |    ";" "port" "=" port [ "-" port ]
                     |    ";" "client_port" "=" port [ "-" port ]
                     |    ";" "server_port" "=" port [ "-" port ]
                     |    ";" "ssrc" "=" ssrc
                     |    ";" "mode" = <"> 1\#mode <">
 ttl                 =    1*3(DIGIT)
 port                =    1*5(DIGIT)
 ssrc                =    8*8(HEX)
 channel             =    1*3(DIGIT)
 address             =    host
 mode                =    <"> *Method <"> | Method
 Example:
   Transport: RTP/AVP;multicast;ttl=127;mode="PLAY",
              RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"
   The Transport header is restricted to describing a single RTP
   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.

Schulzrinne, et. al. Standards Track [Page 61] RFC 2326 Real Time Streaming Protocol April 1998

12.40 Unsupported

 The Unsupported response header lists the features not supported by
 the server. In the case where the feature was specified via the
 Proxy-Require field (Section 12.32), if there is a proxy on the path
 between the client and the server, the proxy MUST insert a message
 reply with an error message "551 Option Not Supported".
 See Section 12.32 for a usage example.

12.41 User-Agent

 See [H14.42]

12.42 Vary

 See [H14.43]

12.43 Via

 See [H14.44].

12.44 WWW-Authentica

 See [H14.46].

13 Caching

 In HTTP, response-request pairs are cached. RTSP differs
 significantly in that respect. Responses are not cacheable, with the
 exception of the presentation description returned by DESCRIBE or
 included with ANNOUNCE. (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 behavior
 allowed to the cache is given by the cache-response directives

Schulzrinne, et. al. Standards Track [Page 62] RFC 2326 Real Time Streaming Protocol April 1998

 described in Section 12.8. A cache MUST answer any DESCRIBE requests
 if it is currently serving the stream to the requestor, as it is
 possible that low-level details of the stream description may have
 changed on the origin-server.
 Note that an RTSP cache, unlike the HTTP 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 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
 (that is, 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.

14 Examples

 The following examples refer to stream description formats that are
 not standards, such as RTSL. The following examples are not to be
 used as a reference for those formats.

14.1 Media on Demand (Unicast)

 Client C requests a movie from 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, dynamic RTP payload types, the protocol stack, and content
 information such as language or copyright restrictions. It may also
 give an indication about the timeline of the movie.
 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/1.0 200 OK
         Content-Type: application/sdp

Schulzrinne, et. al. Standards Track [Page 63] RFC 2326 Real Time Streaming Protocol April 1998

         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202
         s=RTSP Session
         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
   C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 1
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057
   A->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
                    server_port=5000-5001
   C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 1
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059
   V->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 23456789
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
                    server_port=5002-5003
   C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-
   V->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-0:20:00
         RTP-Info: url=rtsp://video.example.com/twister/video;
           seq=12312232;rtptime=78712811
   C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 2
         Session: 12345678
         Range: smpte=0:10:00-
   A->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 12345678

Schulzrinne, et. al. Standards Track [Page 64] RFC 2326 Real Time Streaming Protocol April 1998

         Range: smpte=0:10:00-0:20:00
         RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
           seq=876655;rtptime=1032181
   C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 3
         Session: 12345678
   A->C: RTSP/1.0 200 OK
         CSeq: 3
   C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 3
         Session: 23456789
   V->C: RTSP/1.0 200 OK
         CSeq: 3
 Even though the audio and video track are on two different servers,
 and may start at slightly different times and may drift with respect
 to each other, the client can synchronize the two using standard RTP
 methods, in particular the time scale contained in the RTCP sender
 reports.

14.2 Streaming of a Container file

 For 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 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 URL.
 The following is an example of using a single RTSP session to control
 multiple streams. It also illustrates the use of aggregate URLs.

Schulzrinne, et. al. Standards Track [Page 65] RFC 2326 Real Time Streaming Protocol April 1998

 Client C requests a presentation from media server M . The movie is
 stored in a container file. The client has obtained an RTSP URL to
 the container file.
   C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
         CSeq: 1
   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Content-Type: application/sdp
         Content-Length: 164
         v=0
         o=- 2890844256 2890842807 IN IP4 172.16.2.93
         s=RTSP Session
         i=An Example of RTSP Session Usage
         a=control:rtsp://foo/twister
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control:rtsp://foo/twister/audio
         m=video 0 RTP/AVP 26
         a=control:rtsp://foo/twister/video
   C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP;unicast;client_port=8000-8001
   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Transport: RTP/AVP;unicast;client_port=8000-8001;
                    server_port=9000-9001
         Session: 12345678
   C->M: SETUP rtsp://foo/twister/video RTSP/1.0
         CSeq: 3
         Transport: RTP/AVP;unicast;client_port=8002-8003
         Session: 12345678
   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Transport: RTP/AVP;unicast;client_port=8002-8003;
                    server_port=9004-9005
         Session: 12345678
   C->M: PLAY rtsp://foo/twister RTSP/1.0
         CSeq: 4
         Range: npt=0-
         Session: 12345678

Schulzrinne, et. al. Standards Track [Page 66] RFC 2326 Real Time Streaming Protocol April 1998

   M->C: RTSP/1.0 200 OK
         CSeq: 4
         Session: 12345678
         RTP-Info: url=rtsp://foo/twister/video;
           seq=9810092;rtptime=3450012
   C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
         CSeq: 5
         Session: 12345678
   M->C: RTSP/1.0 460 Only aggregate operation allowed
         CSeq: 5
   C->M: PAUSE rtsp://foo/twister RTSP/1.0
         CSeq: 6
         Session: 12345678
   M->C: RTSP/1.0 200 OK
         CSeq: 6
         Session: 12345678
   C->M: SETUP rtsp://foo/twister RTSP/1.0
         CSeq: 7
         Transport: RTP/AVP;unicast;client_port=10000
   M->C: RTSP/1.0 459 Aggregate operation not allowed
         CSeq: 7
 In the first instance of failure, the client tries to pause one
 stream (in this case video) of the presentation. This is disallowed
 for that presentation by the server. In the second instance, the
 aggregate URL may not be used for SETUP and one control message is
 required per stream to set up transport parameters.
   This keeps the syntax of the Transport header simple and allows
   easy parsing of transport information by firewalls.

14.3 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 SHOULD use the rules set forth in the session
 description for request URLs, rather than assuming that a consistent
 URL may always be used throughout. Here's an example of how a multi-
 stream server might expect a single-stream file to be served:
        Accept: application/x-rtsp-mh, application/sdp

Schulzrinne, et. al. Standards Track [Page 67] RFC 2326 Real Time Streaming Protocol April 1998

        CSeq: 1
  S->C  RTSP/1.0 200 OK
        CSeq: 1
        Content-base: rtsp://foo.com/test.wav/
        Content-type: application/sdp
        Content-length: 48
        v=0
        o=- 872653257 872653257 IN IP4 172.16.2.187
        s=mu-law wave file
        i=audio test
        t=0 0
        m=audio 0 RTP/AVP 0
        a=control:streamid=0
  C->S  SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
        Transport: RTP/AVP/UDP;unicast;
                   client_port=6970-6971;mode=play
        CSeq: 2
  S->C  RTSP/1.0 200 OK
        Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
                   server_port=6970-6971;mode=play
        CSeq: 2
        Session: 2034820394
  C->S  PLAY rtsp://foo.com/test.wav RTSP/1.0
        CSeq: 3
        Session: 2034820394
  S->C  RTSP/1.0 200 OK
        CSeq: 3
        Session: 2034820394
        RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
          seq=981888;rtptime=3781123
 Note the different URL in the SETUP command, and then the switch back
 to the aggregate URL in the PLAY command. This makes complete sense
 when there are multiple streams with aggregate control, but is less
 than intuitive in the special case where the number of streams is
 one.
 In this special case, it is recommended that servers be forgiving of
 implementations that send:
  C->S  PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
        CSeq: 3

Schulzrinne, et. al. Standards Track [Page 68] RFC 2326 Real Time Streaming Protocol April 1998

 In the worst case, servers should send back:
  S->C  RTSP/1.0 460 Only aggregate operation allowed
        CSeq: 3
 One would also hope that server implementations are also forgiving of
 the following:
  C->S  SETUP rtsp://foo.com/test.wav RTSP/1.0
        Transport: rtp/avp/udp;client_port=6970-6971;mode=play
        CSeq: 2
 Since there is only a single stream in this file, it's not ambiguous
 what this means.

14.4 Live Media Presentation Using Multicast

 The media server M chooses the multicast address and port. Here, we
 assume that the web server only contains a pointer to the full
 description, while the media server M maintains the full description.
   C->W: GET /concert.sdp HTTP/1.1
         Host: www.example.com
   W->C: HTTP/1.1 200 OK
         Content-Type: application/x-rtsl
         <session>
           <track src="rtsp://live.example.com/concert/audio">
         </session>
   C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 1
   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Content-Type: application/sdp
         Content-Length: 44
         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202
         s=RTSP Session
         m=audio 3456 RTP/AVP 0
         a=control:rtsp://live.example.com/concert/audio
         c=IN IP4 224.2.0.1/16
   C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 2

Schulzrinne, et. al. Standards Track [Page 69] RFC 2326 Real Time Streaming Protocol April 1998

         Transport: RTP/AVP;multicast
   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Transport: RTP/AVP;multicast;destination=224.2.0.1;
                    port=3456-3457;ttl=16
         Session: 0456804596
   C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 3
         Session: 0456804596
   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Session: 0456804596

14.5 Playing media into an existing session

 A conference participant C wants to have the media server M play back
 a demo tape into an existing conference. C indicates to the media
 server that the network addresses and encryption keys are already
 given by the conference, so they should not be chosen by the server.
 The example omits the simple ACK responses.
   C->M: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0
         CSeq: 1
         Accept: application/sdp
   M->C: RTSP/1.0 200 1 OK
         Content-type: application/sdp
         Content-Length: 44
         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202
         s=RTSP Session
         i=See above
         t=0 0
         m=audio 0 RTP/AVP 0
   C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP;multicast;destination=225.219.201.15;
                    port=7000-7001;ttl=127
         Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Transport: RTP/AVP;multicast;destination=225.219.201.15;

Schulzrinne, et. al. Standards Track [Page 70] RFC 2326 Real Time Streaming Protocol April 1998

                    port=7000-7001;ttl=127
         Session: 91389234234
         Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
   C->M: PLAY rtsp://server.example.com/demo/548/sound RTSP/1.0
         CSeq: 3
         Session: 91389234234
   M->C: RTSP/1.0 200 OK
         CSeq: 3

14.6 Recording

 The conference participant client C asks the media server M to record
 the audio and video portions of a meeting. The client uses the
 ANNOUNCE method to provide meta-information about the recorded
 session to the server.
   C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
         CSeq: 90
         Content-Type: application/sdp
         Content-Length: 121
         v=0
         o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
         s=IETF Meeting, Munich - 1
         i=The thirty-ninth IETF meeting will be held in Munich, Germany
         u=http://www.ietf.org/meetings/Munich.html
         e=IETF Channel 1 <ietf39-mbone@uni-koeln.de>
         p=IETF Channel 1 +49-172-2312 451
         c=IN IP4 224.0.1.11/127
         t=3080271600 3080703600
         a=tool:sdr v2.4a6
         a=type:test
         m=audio 21010 RTP/AVP 5
         c=IN IP4 224.0.1.11/127
         a=ptime:40
         m=video 61010 RTP/AVP 31
         c=IN IP4 224.0.1.12/127
   M->C: RTSP/1.0 200 OK
         CSeq: 90
   C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
         CSeq: 91
         Transport: RTP/AVP;multicast;destination=224.0.1.11;
                    port=21010-21011;mode=record;ttl=127

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   M->C: RTSP/1.0 200 OK
         CSeq: 91
         Session: 50887676
         Transport: RTP/AVP;multicast;destination=224.0.1.11;
                    port=21010-21011;mode=record;ttl=127
   C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
         CSeq: 92
         Session: 50887676
         Transport: RTP/AVP;multicast;destination=224.0.1.12;
                    port=61010-61011;mode=record;ttl=127
   M->C: RTSP/1.0 200 OK
         CSeq: 92
         Transport: RTP/AVP;multicast;destination=224.0.1.12;
                    port=61010-61011;mode=record;ttl=127
   C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
         CSeq: 93
         Session: 50887676
         Range: clock=19961110T1925-19961110T2015
   M->C: RTSP/1.0 200 OK
         CSeq: 93

15 Syntax

 The RTSP syntax is described in an augmented Backus-Naur form (BNF)
 as used in RFC 2068 [2].

15.1 Base Syntax

 OCTET              =      <any 8-bit sequence of data>
 CHAR               =      <any US-ASCII character (octets 0 - 127)>
 UPALPHA            =      <any US-ASCII uppercase letter "A".."Z">
 LOALPHA            =      <any US-ASCII lowercase letter "a".."z">
 ALPHA              =      UPALPHA | LOALPHA
 DIGIT              =      <any US-ASCII digit "0".."9">
 CTL                =      <any US-ASCII control character
                            (octets 0 - 31) and DEL (127)>
 CR                 =      <US-ASCII CR, carriage return (13)>
 LF                 =      <US-ASCII LF, linefeed (10)>
 SP                 =      <US-ASCII SP, space (32)>
 HT                 =      <US-ASCII HT, horizontal-tab (9)>
 <">                =      <US-ASCII double-quote mark (34)>
 CRLF               =      CR LF

Schulzrinne, et. al. Standards Track [Page 72] RFC 2326 Real Time Streaming Protocol April 1998

 LWS                =      [CRLF] 1*( SP | HT )
 TEXT               =      <any OCTET except CTLs>
 tspecials          =      "(" | ")" | "<" | ">" | "@"
                    |       "," | ";" | ":" | "\" | <">
                    |       "/" | "[" | "]" | "?" | "="
                    |       "{" | "}" | SP | HT
 token              =      1*<any CHAR except CTLs or tspecials>
 quoted-string      =      ( <"> *(qdtext) <"> )
 qdtext             =      <any TEXT except <">>
 quoted-pair        =      "\" CHAR
 message-header     =      field-name ":" [ field-value ] CRLF
 field-name         =      token
 field-value        =      *( field-content | LWS )
 field-content      =      <the OCTETs making up the field-value and
                            consisting of either *TEXT or
                            combinations of token, tspecials, and
                            quoted-string>
 safe               =  "\$" | "-" | "_" | "." | "+"
 extra              =  "!" | "*" | "$'$" | "(" | ")" | ","
 hex                =  DIGIT | "A" | "B" | "C" | "D" | "E" | "F" |
                      "a" | "b" | "c" | "d" | "e" | "f"
 escape             =  "\%" hex hex
 reserved           =  ";" | "/" | "?" | ":" | "@" | "&" | "="
 unreserved         =  alpha | digit | safe | extra
 xchar              =  unreserved | reserved | escape

16 Security Considerations

 Because of the similarity in syntax and usage between RTSP servers
 and HTTP servers, the security considerations outlined in [H15]
 apply.  Specifically, please note the following:
 Authentication Mechanisms:
        RTSP and HTTP share common authentication schemes, and thus
        should follow the same prescriptions with regards to
        authentication. See [H15.1] for client authentication issues,
        and [H15.2] for issues regarding support for multiple
        authentication mechanisms.
 Abuse of Server Log Information:
        RTSP and HTTP servers will presumably have similar logging
        mechanisms, and thus should be equally guarded in protecting
        the contents of those logs, thus protecting the privacy of the

Schulzrinne, et. al. Standards Track [Page 73] RFC 2326 Real Time Streaming Protocol April 1998

        users of the servers. See [H15.3] for HTTP server
        recommendations regarding server logs.
 Transfer of Sensitive Information:
        There is no reason to believe that information transferred via
        RTSP may be any less sensitive than that normally transmitted
        via HTTP. Therefore, all of the precautions regarding the
        protection of data privacy and user privacy apply to
        implementors of RTSP clients, servers, and proxies. See
        [H15.4] for further details.
 Attacks Based On File and Path Names:
        Though RTSP URLs are opaque handles that do not necessarily
        have file system semantics, it is anticipated that many
        implementations will translate portions of the request URLs
        directly to file system calls. In such cases, file systems
        SHOULD follow the precautions outlined in [H15.5], such as
        checking for ".." in path components.
 Personal Information:
        RTSP clients are often privy to the same information that HTTP
        clients are (user name, location, etc.) and thus should be
        equally. See [H15.6] for further recommendations.
 Privacy Issues Connected to Accept Headers:
        Since may of the same "Accept" headers exist in RTSP as in
        HTTP, the same caveats outlined in [H15.7] with regards to
        their use should be followed.
 DNS Spoofing:
        Presumably, given the longer connection times typically
        associated to RTSP sessions relative to HTTP sessions, RTSP
        client DNS optimizations should be less prevalent.
        Nonetheless, the recommendations provided in [H15.8] are still
        relevant to any implementation which attempts to rely on a
        DNS-to-IP mapping to hold beyond a single use of the mapping.
 Location Headers and Spoofing:
        If a single server supports multiple organizations that do not
        trust one another, then it must check the values of Location
        and Content-Location headers 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. ([H15.9])
 In addition to the recommendations in the current HTTP specification
 (RFC 2068 [2], as of this writing), future HTTP specifications may
 provide additional guidance on security issues.

Schulzrinne, et. al. Standards Track [Page 74] RFC 2326 Real Time Streaming Protocol April 1998

 The following are added considerations for RTSP implementations.
 Concentrated denial-of-service attack:
        The protocol offers the opportunity for a remote-controlled
        denial-of-service 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
        prevention of more attacks or ascertaining the attackers
        identity. Thus, an RTSP server SHOULD only allow client-
        specified destinations for RTSP-initiated traffic flows if the
        server has verified the client's identity, either against a
        database of known users using RTSP authentication mechanisms
        (preferably digest authentication or stronger), or other
        secure means.
 Session hijacking:
        Since there is no relation between a transport layer
        connection and an RTSP session, it is possible for a malicious
        client to issue requests with random session identifiers which
        would affect unsuspecting clients. The server SHOULD use a
        large, random and non-sequential session identifier to
        minimize the possibility of this kind of attack.
 Authentication:
        Servers SHOULD implement both basic and digest [8]
        authentication. In environments requiring tighter security for
        the control messages, the RTSP control stream may be
        encrypted.
 Stream issues:
        RTSP only provides for stream control. Stream delivery issues
        are not covered in this section, nor in the rest of this memo.
        RTSP implementations will most likely rely on other protocols
        such as RTP, IP multicast, RSVP and IGMP, and should address
        security considerations brought up in those and other
        applicable specifications.
 Persistently suspicious behavior:
        RTSP servers SHOULD return error code 403 (Forbidden) upon
        receiving a single instance of behavior which is deemed a
        security risk. 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 clients
        which are deemed to be in violation of local security policy.

Schulzrinne, et. al. Standards Track [Page 75] RFC 2326 Real Time Streaming Protocol April 1998

Appendix A: RTSP Protocol State Machines

 The RTSP client and server state machines describe the behavior of
 the protocol from RTSP session initialization through RTSP session
 termination.
 State is defined on a per object basis. An object is uniquely
 identified by the stream URL and the RTSP session identifier. Any
 request/reply using aggregate URLs denoting RTSP presentations
 composed of multiple streams will have an effect on the individual
 states of all the streams. For example, if the presentation /movie
 contains two streams, /movie/audio and /movie/video, then the
 following command:
   PLAY rtsp://foo.com/movie RTSP/1.0
   CSeq: 559
   Session: 12345678
 will have an effect on the states of movie/audio and movie/video.
   This example does not imply a standard way to represent streams in
   URLs or a relation to the filesystem. See Section 3.2.
 The requests OPTIONS, ANNOUNCE, DESCRIBE, GET_PARAMETER,
 SET_PARAMETER do not have any effect on client or server state and
 are therefore not listed in the state tables.

A.1 Client State Machine

 The client can assume the following states:
 Init:
        SETUP has been sent, waiting for reply.
 Ready:
        SETUP reply received or PAUSE reply received while in Playing
        state.
 Playing:
        PLAY reply received
 Recording:
        RECORD reply received
 In general, the client changes state on receipt of replies to
 requests. Note that some requests are effective at a future time or
 position (such as a PAUSE), and state also changes accordingly. If no
 explicit SETUP is required for the object (for example, it is

Schulzrinne, et. al. Standards Track [Page 76] RFC 2326 Real Time Streaming Protocol April 1998

 available via a multicast group), state begins at Ready. In this
 case, there are only two states, Ready and Playing. The client also
 changes state from Playing/Recording to Ready when the end of the
 requested range is reached.
 The "next state" column indicates the state assumed after receiving a
 success response (2xx). If a request yields a status code of 3xx, the
 state becomes Init, and a status code of 4xx yields no change in
 state. Messages not listed for each state MUST NOT be issued by the
 client in that state, with the exception of messages not affecting
 state, as listed above. Receiving a REDIRECT from the server is
 equivalent to receiving a 3xx redirect status from the server.
 state       message sent     next state after response
 Init        SETUP            Ready
             TEARDOWN         Init
 Ready       PLAY             Playing
             RECORD           Recording
             TEARDOWN         Init
             SETUP            Ready
 Playing     PAUSE            Ready
             TEARDOWN         Init
             PLAY             Playing
             SETUP            Playing (changed transport)
 Recording   PAUSE            Ready
             TEARDOWN         Init
             RECORD           Recording
             SETUP            Recording (changed transport)

A.2 Server State Machine

 The server can assume the following states:
 Init:
        The initial state, no valid SETUP has been received yet.
 Ready:
        Last SETUP received was successful, reply sent or after
        playing, last PAUSE received was successful, reply sent.
 Playing:
        Last PLAY received was successful, reply sent. Data is being
        sent.
 Recording:
        The server is recording media data.

Schulzrinne, et. al. Standards Track [Page 77] RFC 2326 Real Time Streaming Protocol April 1998

 In general, the server changes state on receiving requests. If the
 server is in state Playing or Recording and in unicast mode, it MAY
 revert to Init and tear down the RTSP session if it has not received
 "wellness" information, such as RTCP reports or RTSP commands, from
 the client for a defined interval, with a default of one minute. The
 server can declare another timeout value in the Session response
 header (Section 12.37). If the server is in state Ready, it MAY
 revert to Init if it does not receive an RTSP request for an interval
 of more than one minute. Note that some requests (such as PAUSE) may
 be effective at a future time or position, and server state changes
 at the appropriate time. The server reverts from state Playing or
 Recording to state Ready at the end of the range requested by the
 client.
 The REDIRECT message, when sent, is effective immediately unless it
 has a Range header specifying when the redirect is effective. In such
 a case, server state will also change at the appropriate time.
 If no explicit SETUP is required for the object, the state starts at
 Ready and there are only two states, Ready and Playing.
 The "next state" column indicates the state assumed after sending a
 success response (2xx). If a request results in a status code of 3xx,
 the state becomes Init. A status code of 4xx results in no change.
   state           message received  next state
   Init            SETUP             Ready
                   TEARDOWN          Init
   Ready           PLAY              Playing
                   SETUP             Ready
                   TEARDOWN          Init
                   RECORD            Recording
   Playing         PLAY              Playing
                   PAUSE             Ready
                   TEARDOWN          Init
                   SETUP             Playing
   Recording       RECORD            Recording
                   PAUSE             Ready
                   TEARDOWN          Init
                   SETUP             Recording

Schulzrinne, et. al. Standards Track [Page 78] RFC 2326 Real Time Streaming Protocol April 1998

Appendix B: Interaction with RTP

 RTSP allows media clients to control selected, non-contiguous
 sections of media presentations, rendering those streams with an RTP
 media layer[24]. The media layer rendering the RTP stream should not
 be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP
 timestamps MUST be continuous and monotonic across jumps of NPT.
 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. First we play NPT 10 through 15, then skip ahead and play NPT
 18 through 20. The first segment is presented as RTP packets with
 sequence numbers 0 through 49 and timestamp 0 through 39,200. The
 second segment consists of RTP packets with sequence number 50
 through 69, with timestamps 40,000 through 55,200.
   We 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
   agent may believe later packets to be duplicates of packets just
   played out.
   For certain datatypes, tight integration between the RTSP layer and
   the RTP layer will be necessary. This by no means precludes the
   above restriction. Combined RTSP/RTP media clients should use the
   RTP-Info field to determine whether incoming RTP packets were sent
   before or after a seek.
 For continuous audio, the server SHOULD set the RTP marker bit at the
 beginning of serving a new PLAY request. This allows the client to
 perform playout delay adaptation.
 For scaling (see Section 12.34), RTP timestamps should correspond to
 the playback timing. For example, when playing video recorded at 30
 frames/second at a scale of two and speed (Section 12.35) 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.
 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 12.33) header provides
 the first sequence number of the next segment.

Schulzrinne, et. al. Standards Track [Page 79] RFC 2326 Real Time Streaming Protocol April 1998

Appendix C: Use of SDP for RTSP Session Descriptions

 The Session Description Protocol (SDP, RFC 2327 [6]) may be used to
 describe streams or presentations in RTSP. Such usage is limited to
 specifying means of access and encoding(s) for:
 aggregate control:
        A presentation composed of streams from one or more servers
        that are not available for aggregate control. Such a
        description is typically retrieved by HTTP or other non-RTSP
        means. However, they may be received with ANNOUNCE methods.
 non-aggregate control:
        A presentation composed of multiple streams from a single
        server that are available for aggregate control. Such a
        description is typically returned in reply to a DESCRIBE
        request on a URL, or received in an ANNOUNCE method.
 This appendix describes how an SDP file, retrieved, for example,
 through HTTP, determines the operation of an RTSP session. It also
 describes how a client should interpret SDP content returned in reply
 to a DESCRIBE request. SDP 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).

C.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 (RFC 2327 [6]):

C.1.1 Control URL

 The "a=control:" attribute is used to convey the control URL. This
 attribute is used both for the session and media descriptions. If
 used for individual media, it indicates the URL to be used for
 controlling that particular media stream. If found at the session
 level, the attribute indicates the URL for aggregate control.
 Example:
   a=control:rtsp://example.com/foo
 This attribute may contain either relative and absolute URLs,
 following the rules and conventions set out in RFC 1808 [25].
 Implementations should look for a base URL in the following order:

Schulzrinne, et. al. Standards Track [Page 80] RFC 2326 Real Time Streaming Protocol April 1998

 1.     The RTSP Content-Base field
 2.     The RTSP Content-Location field
 3.     The RTSP request URL
 If this attribute contains only an asterisk (*), then the URL is
 treated as if it were an empty embedded URL, and thus inherits the
 entire base URL.

C.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 unicast, the port number serves as
 a recommendation from the server to the client; the client still has
 to include it in its SETUP request and may ignore this
 recommendation.  If the server has no preference, it SHOULD set the
 port number value to zero.
 Example:
   m=audio 0 RTP/AVP 31

C.1.3 Payload type(s)

 The payload type(s) are specified in the "m=" field. In case the
 payload type is a static payload type from RFC 1890 [1], no other
 information is 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 RFC 1890 (Sections 5 and 6), or an experimental encoding
 with a "X-" prefix as specified in SDP (RFC 2327 [6]).  Codec-
 specific parameters are not specified in this field, but rather in
 the "fmtp" attribute described below. Implementors seeking to
 register new encodings should follow the procedure in RFC 1890 [1].
 If the media type is not suited to the RTP AV profile, then it is
 recommended that a new profile be created and the appropriate profile
 name be used in lieu of "RTP/AVP" in the "m=" field.

C.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) that the attribute refers to. Note that the packetization
 interval is conveyed using the "ptime" attribute.

Schulzrinne, et. al. Standards Track [Page 81] RFC 2326 Real Time Streaming Protocol April 1998

C.1.5 Range of presentation

 The "a=range" attribute defines the total time range of the stored
 session. (The length of live sessions can be deduced from the "t" and
 "r" parameters.) Unless the presentation contains media streams of
 different durations, the range attribute is a session-level
 attribute. The unit is specified first, followed by the value range.
 The units and their values are as defined in Section 3.5, 3.6 and
 3.7.
 Examples:
   a=range:npt=0-34.4368
   a=range:clock=19971113T2115-19971113T2203

C.1.6 Time of availability

 The "t=" field MUST contain suitable values for the start and stop
 times for both aggregate and non-aggregate stream control. With
 aggregate control, 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. With non-aggregate control, the
 values should reflect the actual period for which the session is
 available in keeping with SDP semantics, and not depend on other
 means (such as the life of the web page containing the description)
 for this purpose.

C.1.7 Connection Information

 In SDP, the "c=" field contains the destination address for the media
 stream. However, for on-demand unicast streams and some multicast
 streams, the destination address is specified by the client via the
 SETUP request. Unless the media content has a fixed destination
 address, the "c=" field is to be set to a suitable null value. For
 addresses of type "IP4", this value is "0.0.0.0".
C.1.8 Entity Tag
 The optional "a=etag" 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 12.22) to only
 allow session establishment 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.
 Example:
   a=etag:158bb3e7c7fd62ce67f12b533f06b83a

Schulzrinne, et. al. Standards Track [Page 82] RFC 2326 Real Time Streaming Protocol April 1998

   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.

C.2 Aggregate Control Not Available

 If a presentation does not support aggregate control and multiple
 media sections are specified, each section MUST have the control URL
 specified via the "a=control:" attribute.
 Example:
   v=0
   o=- 2890844256 2890842807 IN IP4 204.34.34.32
   s=I came from a web page
   t=0 0
   c=IN IP4 0.0.0.0
   m=video 8002 RTP/AVP 31
   a=control:rtsp://audio.com/movie.aud
   m=audio 8004 RTP/AVP 3
   a=control:rtsp://video.com/movie.vid
 Note that the position of the control URL in the description implies
 that the client establishes separate RTSP control sessions to the
 servers audio.com and video.com.
 It is recommended that an SDP file contains 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.

C.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 media-level
 "a=control:" attributes, which are used to specify the stream URLs,
 and a session-level "a=control:" attribute which is used as the
 request URL for aggregate control. If the media-level URL is
 relative, it is resolved to absolute URLs according to Section C.1.1
 above.
 If the presentation comprises only a single stream, the media-level
 "a=control:" attribute may be omitted altogether. However, if the
 presentation contains more than one stream, each media stream section
 MUST contain its own "a=control" attribute.

Schulzrinne, et. al. Standards Track [Page 83] RFC 2326 Real Time Streaming Protocol April 1998

 Example:
   v=0
   o=- 2890844256 2890842807 IN IP4 204.34.34.32
   s=I contain
   i=<more info>
   t=0 0
   c=IN IP4 0.0.0.0
   a=control:rtsp://example.com/movie/
   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 required to establish a single RTSP
 session to the server, and uses the URLs
 rtsp://example.com/movie/trackID=1 and
 rtsp://example.com/movie/trackID=2 to set up the video and audio
 streams, respectively. The URL rtsp://example.com/movie/ controls the
 whole movie.

Schulzrinne, et. al. Standards Track [Page 84] RFC 2326 Real Time Streaming Protocol April 1998

Appendix D: Minimal RTSP implementation

D.1 Client

 A client implementation MUST be able to do the following :
  • Generate the following requests: SETUP, TEARDOWN, and one of PLAY

(i.e., a minimal playback client) or RECORD (i.e., a minimal

     recording client). If RECORD is implemented, ANNOUNCE must be
     implemented as well.
   * Include the following headers in requests: CSeq, Connection,
     Session, Transport. If ANNOUNCE is implemented, the capability to
     include headers Content-Language, Content-Encoding, Content-
     Length, and Content-Type should be as well.
   * Parse and understand the following headers in responses: CSeq,
     Connection, Session, Transport, Content-Language, Content-
     Encoding, Content-Length, Content-Type. If RECORD is implemented,
     the Location header must be understood as well.  RTP-compliant
     implementations should also implement RTP-Info.
   * Understand the class of each error code received and notify the
     end-user, if one is present, of error codes in classes 4xx and
     5xx. The notification requirement may be relaxed if the end-user
     explicitly does not want it for one or all status codes.
   * Expect and respond to asynchronous requests from the server, such
     as ANNOUNCE. This does not necessarily mean that it should
     implement the ANNOUNCE method, merely that it MUST respond
     positively or negatively to any request received from the server.
 Though not required, the following are highly recommended at the time
 of publication for practical interoperability with initial
 implementations and/or to be a "good citizen".
  • Implement RTP/AVP/UDP as a valid transport.
  • Inclusion of the User-Agent header.
  • Understand SDP session descriptions as defined in Appendix C
  • Accept media initialization formats (such as SDP) from standard

input, command line, or other means appropriate to the operating

     environment to act as a "helper application" for other
     applications (such as web browsers).
   There may be RTSP applications different from those initially
   envisioned by the contributors to the RTSP specification for which
   the requirements above do not make sense. Therefore, the
   recommendations above serve only as guidelines instead of strict
   requirements.

Schulzrinne, et. al. Standards Track [Page 85] RFC 2326 Real Time Streaming Protocol April 1998

D.1.1 Basic Playback

 To support on-demand playback of media streams, the client MUST
 additionally be able to do the following:
   * generate the PAUSE request;
   * implement the REDIRECT method, and the Location header.

D.1.2 Authentication-enabled

 In order to access media presentations from RTSP servers that require
 authentication, the client MUST additionally be able to do the
 following:
   * recognize the 401 status code;
   * parse and include the WWW-Authenticate header;
   * implement Basic Authentication and Digest Authentication.

D.2 Server

 A minimal server implementation MUST be able to do the following:
  • Implement the following methods: SETUP, TEARDOWN, OPTIONS and

either PLAY (for a minimal playback server) or RECORD (for a

     minimal recording server).  If RECORD is implemented, ANNOUNCE
     should be implemented as well.
   * Include the following headers in responses: Connection,
     Content-Length, Content-Type, Content-Language, Content-Encoding,
     Transport, Public. The capability to include the Location header
     should be implemented if the RECORD method is. RTP-compliant
     implementations should also implement the RTP-Info field.
   * Parse and respond appropriately to the following headers in
     requests: Connection, Session, Transport, Require.
 Though not required, the following are highly recommended at the time
 of publication for practical interoperability with initial
 implementations and/or to be a "good citizen".
  • Implement RTP/AVP/UDP as a valid transport.
  • Inclusion of the Server header.
  • Implement the DESCRIBE method.
  • Generate SDP session descriptions as defined in Appendix C
   There may be RTSP applications different from those initially
   envisioned by the contributors to the RTSP specification for which
   the requirements above do not make sense. Therefore, the
   recommendations above serve only as guidelines instead of strict
   requirements.

Schulzrinne, et. al. Standards Track [Page 86] RFC 2326 Real Time Streaming Protocol April 1998

D.2.1 Basic Playback

 To support on-demand playback of media streams, the server MUST
 additionally be able to do the following:
  • Recognize the Range header, and return an error if seeking is not

supported.

  • Implement the PAUSE method.
 In addition, in order to support commonly-accepted user interface
 features, the following are highly recommended for on-demand media
 servers:
  • Include and parse the Range header, with NPT units.

Implementation of SMPTE units is recommended.

  • Include the length of the media presentation in the media

initialization information.

  • Include mappings from data-specific timestamps to NPT. When RTP

is used, the rtptime portion of the RTP-Info field may be used to

     map RTP timestamps to NPT.
   Client implementations may use the presence of length information
   to determine if the clip is seekable, and visibly disable seeking
   features for clips for which the length information is unavailable.
   A common use of the presentation length is to implement a "slider
   bar" which serves as both a progress indicator and a timeline
   positioning tool.
   Mappings from RTP timestamps to NPT are necessary to ensure correct
   positioning of the slider bar.

D.2.2 Authentication-enabled

 In order to correctly handle client authentication, the server MUST
 additionally be able to do the following:
  • Generate the 401 status code when authentication is required for

the resource.

  • Parse and include the WWW-Authenticate header
  • Implement Basic Authentication and Digest Authentication

Schulzrinne, et. al. Standards Track [Page 87] RFC 2326 Real Time Streaming Protocol April 1998

Appendix E: Authors' Addresses

 Henning Schulzrinne
 Dept. of Computer Science
 Columbia University
 1214 Amsterdam Avenue
 New York, NY 10027
 USA
 EMail: schulzrinne@cs.columbia.edu
 Anup Rao
 Netscape Communications Corp.
 501 E. Middlefield Road
 Mountain View, CA 94043
 USA
 EMail: anup@netscape.com
 Robert Lanphier
 RealNetworks
 1111 Third Avenue Suite 2900
 Seattle, WA 98101
 USA
 EMail: robla@real.com

Schulzrinne, et. al. Standards Track [Page 88] RFC 2326 Real Time Streaming Protocol April 1998

Appendix F: Acknowledgements

 This memo is based on the functionality of the original RTSP document
 submitted in October 96. It also borrows format and descriptions from
 HTTP/1.1.
 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, Torsten Braun, Brent Browning, Bruce Butterfield,
 Steve Casner, Francisco Cortes, Kelly Djahandari, Martin Dunsmuir,
 Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad, Peter
 Haight, Mark Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka,
 Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan
 Lennox, Eduardo F. Llach, Rob McCool, David Oran, Maria Papadopouli,
 Sujal Patel, Ema Patki, Alagu Periyannan, Igor Plotnikov, Pinaki
 Shah, David Singer, Jeff Smith, Alexander Sokolsky, Dale Stammen, and
 John Francis Stracke.

Schulzrinne, et. al. Standards Track [Page 89] RFC 2326 Real Time Streaming Protocol April 1998

References

 1      Schulzrinne, H., "RTP profile for audio and video conferences
        with minimal control", RFC 1890, January 1996.
 2      Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
        Berners-Lee, "Hypertext transfer protocol - HTTP/1.1", RFC
        2068, January 1997.
 3      Yergeau, F., Nicol, G., Adams, G., and M. Duerst,
        "Internationalization of the hypertext markup language", RFC
        2070, January 1997.
 4      Bradner, S., "Key words for use in RFCs to indicate
        requirement levels", BCP 14, RFC 2119, March 1997.
 5      ISO/IEC, "Information technology - generic coding of moving
        pictures and associated audio information - part 6: extension
        for digital storage media and control," Draft International
        Standard ISO 13818-6, International Organization for
        Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland,
        Nov. 1995.
 6      Handley, M., and V. Jacobson, "SDP: Session Description
        Protocol", RFC 2327, April 1998.
 7      Franks, J., Hallam-Baker, P., and J. Hostetler, "An extension to
        HTTP: digest access authentication", RFC 2069, January 1997.
 8      Postel, J., "User Datagram Protocol", STD 6, RFC 768, August
        1980.
 9      Hinden, B. and C. Partridge, "Version 2 of the reliable data
        protocol (RDP)", RFC 1151, April 1990.
 10     Postel, J., "Transmission control protocol", STD 7, RFC 793,
        September 1981.
 11     H. Schulzrinne, "A comprehensive multimedia control
        architecture for the Internet," in Proc. International
        Workshop on Network and Operating System Support for Digital
        Audio and Video (NOSSDAV), (St. Louis, Missouri), May 1997.
 12     International Telecommunication Union, "Visual telephone
        systems and equipment for local area networks which provide a
        non-guaranteed quality of service," Recommendation H.323,
        Telecommunication Standardization Sector of ITU, Geneva,
        Switzerland, May 1996.

Schulzrinne, et. al. Standards Track [Page 90] RFC 2326 Real Time Streaming Protocol April 1998

 13     McMahon, P., "GSS-API authentication method for SOCKS version
        5", RFC 1961, June 1996.
 14     J. Miller, P. Resnick, and D. Singer, "Rating services and
        rating systems (and their machine readable descriptions),"
        Recommendation REC-PICS-services-961031, W3C (World Wide Web
        Consortium), Boston, Massachusetts, Oct. 1996.
 15     J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS
        label distribution label syntax and communication protocols,"
        Recommendation REC-PICS-labels-961031, W3C (World Wide Web
        Consortium), Boston, Massachusetts, Oct. 1996.
 16     Crocker, D. and P. Overell, "Augmented BNF for syntax
        specifications: ABNF", RFC 2234, November 1997.
 17     Braden, B., "Requirements for internet hosts - application and
        support", STD 3, RFC 1123, October 1989.
 18     Elz, R., "A compact representation of IPv6 addresses", RFC
        1924, April 1996.
 19     Berners-Lee, T., Masinter, L. and M. McCahill, "Uniform
        resource locators (URL)", RFC 1738, December 1994.
 20     Yergeau, F., "UTF-8, a transformation format of ISO 10646",
        RFC 2279, January 1998.
 22     Braden, B., "T/TCP - TCP extensions for transactions
        functional specification", RFC 1644, July 1994.
 22     W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
        Reading, Massachusetts: Addison-Wesley, 1994.
 23     Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
        "RTP: a transport protocol for real-time applications", RFC
        1889, January 1996.
 24     Fielding, R., "Relative uniform resource locators", RFC 1808,
        June 1995.

Schulzrinne, et. al. Standards Track [Page 91] RFC 2326 Real Time Streaming Protocol April 1998

Full Copyright Statement

 Copyright (C) The Internet Society (1998). All Rights Reserved.
 This document and translations of it may be copied and furnished to
 others, and derivative works that comment on or otherwise explain it
 or assist in its implementation may be prepared, copied, published
 and distributed, in whole or in part, without restriction of any
 kind, provided that the above copyright notice and this paragraph are
 included on all such copies and derivative works. However, this
 document itself may not be modified in any way, such as by removing
 the copyright notice or references to the Internet Society or other
 Internet organizations, except as needed for the purpose of
 developing Internet standards in which case the procedures for
 copyrights defined in the Internet Standards process must be
 followed, or as required to translate it into languages other than
 English.
 The limited permissions granted above are perpetual and will not be
 revoked by the Internet Society or its successors or assigns.
 This document and the information contained herein is provided on an
 "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
 TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
 BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
 HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
 MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Schulzrinne, et. al. Standards Track [Page 92]

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