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rfc:rfc3261

Network Working Group J. Rosenberg Request for Comments: 3261 dynamicsoft Obsoletes: 2543 H. Schulzrinne Category: Standards Track Columbia U.

                                                          G. Camarillo
                                                              Ericsson
                                                           A. Johnston
                                                              WorldCom
                                                           J. Peterson
                                                               Neustar
                                                             R. Sparks
                                                           dynamicsoft
                                                            M. Handley
                                                                  ICIR
                                                           E. Schooler
                                                                  AT&T
                                                             June 2002
                  SIP: Session Initiation Protocol

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 (2002).  All Rights Reserved.

Abstract

 This document describes Session Initiation Protocol (SIP), an
 application-layer control (signaling) protocol for creating,
 modifying, and terminating sessions with one or more participants.
 These sessions include Internet telephone calls, multimedia
 distribution, and multimedia conferences.
 SIP invitations used to create sessions carry session descriptions
 that allow participants to agree on a set of compatible media types.
 SIP makes use of elements called proxy servers to help route requests
 to the user's current location, authenticate and authorize users for
 services, implement provider call-routing policies, and provide
 features to users.  SIP also provides a registration function that
 allows users to upload their current locations for use by proxy
 servers.  SIP runs on top of several different transport protocols.

Rosenberg, et. al. Standards Track [Page 1] RFC 3261 SIP: Session Initiation Protocol June 2002

Table of Contents

 1          Introduction ........................................    8
 2          Overview of SIP Functionality .......................    9
 3          Terminology .........................................   10
 4          Overview of Operation ...............................   10
 5          Structure of the Protocol ...........................   18
 6          Definitions .........................................   20
 7          SIP Messages ........................................   26
 7.1        Requests ............................................   27
 7.2        Responses ...........................................   28
 7.3        Header Fields .......................................   29
 7.3.1      Header Field Format .................................   30
 7.3.2      Header Field Classification .........................   32
 7.3.3      Compact Form ........................................   32
 7.4        Bodies ..............................................   33
 7.4.1      Message Body Type ...................................   33
 7.4.2      Message Body Length .................................   33
 7.5        Framing SIP Messages ................................   34
 8          General User Agent Behavior .........................   34
 8.1        UAC Behavior ........................................   35
 8.1.1      Generating the Request ..............................   35
 8.1.1.1    Request-URI .........................................   35
 8.1.1.2    To ..................................................   36
 8.1.1.3    From ................................................   37
 8.1.1.4    Call-ID .............................................   37
 8.1.1.5    CSeq ................................................   38
 8.1.1.6    Max-Forwards ........................................   38
 8.1.1.7    Via .................................................   39
 8.1.1.8    Contact .............................................   40
 8.1.1.9    Supported and Require ...............................   40
 8.1.1.10   Additional Message Components .......................   41
 8.1.2      Sending the Request .................................   41
 8.1.3      Processing Responses ................................   42
 8.1.3.1    Transaction Layer Errors ............................   42
 8.1.3.2    Unrecognized Responses ..............................   42
 8.1.3.3    Vias ................................................   43
 8.1.3.4    Processing 3xx Responses ............................   43
 8.1.3.5    Processing 4xx Responses ............................   45
 8.2        UAS Behavior ........................................   46
 8.2.1      Method Inspection ...................................   46
 8.2.2      Header Inspection ...................................   46
 8.2.2.1    To and Request-URI ..................................   46
 8.2.2.2    Merged Requests .....................................   47
 8.2.2.3    Require .............................................   47
 8.2.3      Content Processing ..................................   48
 8.2.4      Applying Extensions .................................   49
 8.2.5      Processing the Request ..............................   49

Rosenberg, et. al. Standards Track [Page 2] RFC 3261 SIP: Session Initiation Protocol June 2002

 8.2.6      Generating the Response .............................   49
 8.2.6.1    Sending a Provisional Response ......................   49
 8.2.6.2    Headers and Tags ....................................   50
 8.2.7      Stateless UAS Behavior ..............................   50
 8.3        Redirect Servers ....................................   51
 9          Canceling a Request .................................   53
 9.1        Client Behavior .....................................   53
 9.2        Server Behavior .....................................   55
 10         Registrations .......................................   56
 10.1       Overview ............................................   56
 10.2       Constructing the REGISTER Request ...................   57
 10.2.1     Adding Bindings .....................................   59
 10.2.1.1   Setting the Expiration Interval of Contact Addresses    60
 10.2.1.2   Preferences among Contact Addresses .................   61
 10.2.2     Removing Bindings ...................................   61
 10.2.3     Fetching Bindings ...................................   61
 10.2.4     Refreshing Bindings .................................   61
 10.2.5     Setting the Internal Clock ..........................   62
 10.2.6     Discovering a Registrar .............................   62
 10.2.7     Transmitting a Request ..............................   62
 10.2.8     Error Responses .....................................   63
 10.3       Processing REGISTER Requests ........................   63
 11         Querying for Capabilities ...........................   66
 11.1       Construction of OPTIONS Request .....................   67
 11.2       Processing of OPTIONS Request .......................   68
 12         Dialogs .............................................   69
 12.1       Creation of a Dialog ................................   70
 12.1.1     UAS behavior ........................................   70
 12.1.2     UAC Behavior ........................................   71
 12.2       Requests within a Dialog ............................   72
 12.2.1     UAC Behavior ........................................   73
 12.2.1.1   Generating the Request ..............................   73
 12.2.1.2   Processing the Responses ............................   75
 12.2.2     UAS Behavior ........................................   76
 12.3       Termination of a Dialog .............................   77
 13         Initiating a Session ................................   77
 13.1       Overview ............................................   77
 13.2       UAC Processing ......................................   78
 13.2.1     Creating the Initial INVITE .........................   78
 13.2.2     Processing INVITE Responses .........................   81
 13.2.2.1   1xx Responses .......................................   81
 13.2.2.2   3xx Responses .......................................   81
 13.2.2.3   4xx, 5xx and 6xx Responses ..........................   81
 13.2.2.4   2xx Responses .......................................   82
 13.3       UAS Processing ......................................   83
 13.3.1     Processing of the INVITE ............................   83
 13.3.1.1   Progress ............................................   84
 13.3.1.2   The INVITE is Redirected ............................   84

Rosenberg, et. al. Standards Track [Page 3] RFC 3261 SIP: Session Initiation Protocol June 2002

 13.3.1.3   The INVITE is Rejected ..............................   85
 13.3.1.4   The INVITE is Accepted ..............................   85
 14         Modifying an Existing Session .......................   86
 14.1       UAC Behavior ........................................   86
 14.2       UAS Behavior ........................................   88
 15         Terminating a Session ...............................   89
 15.1       Terminating a Session with a BYE Request ............   90
 15.1.1     UAC Behavior ........................................   90
 15.1.2     UAS Behavior ........................................   91
 16         Proxy Behavior ......................................   91
 16.1       Overview ............................................   91
 16.2       Stateful Proxy ......................................   92
 16.3       Request Validation ..................................   94
 16.4       Route Information Preprocessing .....................   96
 16.5       Determining Request Targets .........................   97
 16.6       Request Forwarding ..................................   99
 16.7       Response Processing .................................  107
 16.8       Processing Timer C ..................................  114
 16.9       Handling Transport Errors ...........................  115
 16.10      CANCEL Processing ...................................  115
 16.11      Stateless Proxy .....................................  116
 16.12      Summary of Proxy Route Processing ...................  118
 16.12.1    Examples ............................................  118
 16.12.1.1  Basic SIP Trapezoid .................................  118
 16.12.1.2  Traversing a Strict-Routing Proxy ...................  120
 16.12.1.3  Rewriting Record-Route Header Field Values ..........  121
 17         Transactions ........................................  122
 17.1       Client Transaction ..................................  124
 17.1.1     INVITE Client Transaction ...........................  125
 17.1.1.1   Overview of INVITE Transaction ......................  125
 17.1.1.2   Formal Description ..................................  125
 17.1.1.3   Construction of the ACK Request .....................  129
 17.1.2     Non-INVITE Client Transaction .......................  130
 17.1.2.1   Overview of the non-INVITE Transaction ..............  130
 17.1.2.2   Formal Description ..................................  131
 17.1.3     Matching Responses to Client Transactions ...........  132
 17.1.4     Handling Transport Errors ...........................  133
 17.2       Server Transaction ..................................  134
 17.2.1     INVITE Server Transaction ...........................  134
 17.2.2     Non-INVITE Server Transaction .......................  137
 17.2.3     Matching Requests to Server Transactions ............  138
 17.2.4     Handling Transport Errors ...........................  141
 18         Transport ...........................................  141
 18.1       Clients .............................................  142
 18.1.1     Sending Requests ....................................  142
 18.1.2     Receiving Responses .................................  144
 18.2       Servers .............................................  145
 18.2.1     Receiving Requests ..................................  145

Rosenberg, et. al. Standards Track [Page 4] RFC 3261 SIP: Session Initiation Protocol June 2002

 18.2.2     Sending Responses ...................................  146
 18.3       Framing .............................................  147
 18.4       Error Handling ......................................  147
 19         Common Message Components ...........................  147
 19.1       SIP and SIPS Uniform Resource Indicators ............  148
 19.1.1     SIP and SIPS URI Components .........................  148
 19.1.2     Character Escaping Requirements .....................  152
 19.1.3     Example SIP and SIPS URIs ...........................  153
 19.1.4     URI Comparison ......................................  153
 19.1.5     Forming Requests from a URI .........................  156
 19.1.6     Relating SIP URIs and tel URLs ......................  157
 19.2       Option Tags .........................................  158
 19.3       Tags ................................................  159
 20         Header Fields .......................................  159
 20.1       Accept ..............................................  161
 20.2       Accept-Encoding .....................................  163
 20.3       Accept-Language .....................................  164
 20.4       Alert-Info ..........................................  164
 20.5       Allow ...............................................  165
 20.6       Authentication-Info .................................  165
 20.7       Authorization .......................................  165
 20.8       Call-ID .............................................  166
 20.9       Call-Info ...........................................  166
 20.10      Contact .............................................  167
 20.11      Content-Disposition .................................  168
 20.12      Content-Encoding ....................................  169
 20.13      Content-Language ....................................  169
 20.14      Content-Length ......................................  169
 20.15      Content-Type ........................................  170
 20.16      CSeq ................................................  170
 20.17      Date ................................................  170
 20.18      Error-Info ..........................................  171
 20.19      Expires .............................................  171
 20.20      From ................................................  172
 20.21      In-Reply-To .........................................  172
 20.22      Max-Forwards ........................................  173
 20.23      Min-Expires .........................................  173
 20.24      MIME-Version ........................................  173
 20.25      Organization ........................................  174
 20.26      Priority ............................................  174
 20.27      Proxy-Authenticate ..................................  174
 20.28      Proxy-Authorization .................................  175
 20.29      Proxy-Require .......................................  175
 20.30      Record-Route ........................................  175
 20.31      Reply-To ............................................  176
 20.32      Require .............................................  176
 20.33      Retry-After .........................................  176
 20.34      Route ...............................................  177

Rosenberg, et. al. Standards Track [Page 5] RFC 3261 SIP: Session Initiation Protocol June 2002

 20.35      Server ..............................................  177
 20.36      Subject .............................................  177
 20.37      Supported ...........................................  178
 20.38      Timestamp ...........................................  178
 20.39      To ..................................................  178
 20.40      Unsupported .........................................  179
 20.41      User-Agent ..........................................  179
 20.42      Via .................................................  179
 20.43      Warning .............................................  180
 20.44      WWW-Authenticate ....................................  182
 21         Response Codes ......................................  182
 21.1       Provisional 1xx .....................................  182
 21.1.1     100 Trying ..........................................  183
 21.1.2     180 Ringing .........................................  183
 21.1.3     181 Call Is Being Forwarded .........................  183
 21.1.4     182 Queued ..........................................  183
 21.1.5     183 Session Progress ................................  183
 21.2       Successful 2xx ......................................  183
 21.2.1     200 OK ..............................................  183
 21.3       Redirection 3xx .....................................  184
 21.3.1     300 Multiple Choices ................................  184
 21.3.2     301 Moved Permanently ...............................  184
 21.3.3     302 Moved Temporarily ...............................  184
 21.3.4     305 Use Proxy .......................................  185
 21.3.5     380 Alternative Service .............................  185
 21.4       Request Failure 4xx .................................  185
 21.4.1     400 Bad Request .....................................  185
 21.4.2     401 Unauthorized ....................................  185
 21.4.3     402 Payment Required ................................  186
 21.4.4     403 Forbidden .......................................  186
 21.4.5     404 Not Found .......................................  186
 21.4.6     405 Method Not Allowed ..............................  186
 21.4.7     406 Not Acceptable ..................................  186
 21.4.8     407 Proxy Authentication Required ...................  186
 21.4.9     408 Request Timeout .................................  186
 21.4.10    410 Gone ............................................  187
 21.4.11    413 Request Entity Too Large ........................  187
 21.4.12    414 Request-URI Too Long ............................  187
 21.4.13    415 Unsupported Media Type ..........................  187
 21.4.14    416 Unsupported URI Scheme ..........................  187
 21.4.15    420 Bad Extension ...................................  187
 21.4.16    421 Extension Required ..............................  188
 21.4.17    423 Interval Too Brief ..............................  188
 21.4.18    480 Temporarily Unavailable .........................  188
 21.4.19    481 Call/Transaction Does Not Exist .................  188
 21.4.20    482 Loop Detected ...................................  188
 21.4.21    483 Too Many Hops ...................................  189
 21.4.22    484 Address Incomplete ..............................  189

Rosenberg, et. al. Standards Track [Page 6] RFC 3261 SIP: Session Initiation Protocol June 2002

 21.4.23    485 Ambiguous .......................................  189
 21.4.24    486 Busy Here .......................................  189
 21.4.25    487 Request Terminated ..............................  190
 21.4.26    488 Not Acceptable Here .............................  190
 21.4.27    491 Request Pending .................................  190
 21.4.28    493 Undecipherable ..................................  190
 21.5       Server Failure 5xx ..................................  190
 21.5.1     500 Server Internal Error ...........................  190
 21.5.2     501 Not Implemented .................................  191
 21.5.3     502 Bad Gateway .....................................  191
 21.5.4     503 Service Unavailable .............................  191
 21.5.5     504 Server Time-out .................................  191
 21.5.6     505 Version Not Supported ...........................  192
 21.5.7     513 Message Too Large ...............................  192
 21.6       Global Failures 6xx .................................  192
 21.6.1     600 Busy Everywhere .................................  192
 21.6.2     603 Decline .........................................  192
 21.6.3     604 Does Not Exist Anywhere .........................  192
 21.6.4     606 Not Acceptable ..................................  192
 22         Usage of HTTP Authentication ........................  193
 22.1       Framework ...........................................  193
 22.2       User-to-User Authentication .........................  195
 22.3       Proxy-to-User Authentication ........................  197
 22.4       The Digest Authentication Scheme ....................  199
 23         S/MIME ..............................................  201
 23.1       S/MIME Certificates .................................  201
 23.2       S/MIME Key Exchange .................................  202
 23.3       Securing MIME bodies ................................  205
 23.4       SIP Header Privacy and Integrity using S/MIME:
            Tunneling SIP .......................................  207
 23.4.1     Integrity and Confidentiality Properties of SIP
            Headers .............................................  207
 23.4.1.1   Integrity ...........................................  207
 23.4.1.2   Confidentiality .....................................  208
 23.4.2     Tunneling Integrity and Authentication ..............  209
 23.4.3     Tunneling Encryption ................................  211
 24         Examples ............................................  213
 24.1       Registration ........................................  213
 24.2       Session Setup .......................................  214
 25         Augmented BNF for the SIP Protocol ..................  219
 25.1       Basic Rules .........................................  219
 26         Security Considerations: Threat Model and Security
            Usage Recommendations ...............................  232
 26.1       Attacks and Threat Models ...........................  233
 26.1.1     Registration Hijacking ..............................  233
 26.1.2     Impersonating a Server ..............................  234
 26.1.3     Tampering with Message Bodies .......................  235
 26.1.4     Tearing Down Sessions ...............................  235

Rosenberg, et. al. Standards Track [Page 7] RFC 3261 SIP: Session Initiation Protocol June 2002

 26.1.5     Denial of Service and Amplification .................  236
 26.2       Security Mechanisms .................................  237
 26.2.1     Transport and Network Layer Security ................  238
 26.2.2     SIPS URI Scheme .....................................  239
 26.2.3     HTTP Authentication .................................  240
 26.2.4     S/MIME ..............................................  240
 26.3       Implementing Security Mechanisms ....................  241
 26.3.1     Requirements for Implementers of SIP ................  241
 26.3.2     Security Solutions ..................................  242
 26.3.2.1   Registration ........................................  242
 26.3.2.2   Interdomain Requests ................................  243
 26.3.2.3   Peer-to-Peer Requests ...............................  245
 26.3.2.4   DoS Protection ......................................  246
 26.4       Limitations .........................................  247
 26.4.1     HTTP Digest .........................................  247
 26.4.2     S/MIME ..............................................  248
 26.4.3     TLS .................................................  249
 26.4.4     SIPS URIs ...........................................  249
 26.5       Privacy .............................................  251
 27         IANA Considerations .................................  252
 27.1       Option Tags .........................................  252
 27.2       Warn-Codes ..........................................  252
 27.3       Header Field Names ..................................  253
 27.4       Method and Response Codes ...........................  253
 27.5       The "message/sip" MIME type.  .......................  254
 27.6       New Content-Disposition Parameter Registrations .....  255
 28         Changes From RFC 2543 ...............................  255
 28.1       Major Functional Changes ............................  255
 28.2       Minor Functional Changes ............................  260
 29         Normative References ................................  261
 30         Informative References ..............................  262
 A          Table of Timer Values ...............................  265
 Acknowledgments ................................................  266
 Authors' Addresses .............................................  267
 Full Copyright Statement .......................................  269

1 Introduction

 There are many applications of the Internet that require the creation
 and management of a session, where a session is considered an
 exchange of data between an association of participants.  The
 implementation of these applications is complicated by the practices
 of participants: users may move between endpoints, they may be
 addressable by multiple names, and they may communicate in several
 different media - sometimes simultaneously.  Numerous protocols have
 been authored that carry various forms of real-time multimedia
 session data such as voice, video, or text messages.  The Session
 Initiation Protocol (SIP) works in concert with these protocols by

Rosenberg, et. al. Standards Track [Page 8] RFC 3261 SIP: Session Initiation Protocol June 2002

 enabling Internet endpoints (called user agents) to discover one
 another and to agree on a characterization of a session they would
 like to share.  For locating prospective session participants, and
 for other functions, SIP enables the creation of an infrastructure of
 network hosts (called proxy servers) to which user agents can send
 registrations, invitations to sessions, and other requests.  SIP is
 an agile, general-purpose tool for creating, modifying, and
 terminating sessions that works independently of underlying transport
 protocols and without dependency on the type of session that is being
 established.

2 Overview of SIP Functionality

 SIP is an application-layer control protocol that can establish,
 modify, and terminate multimedia sessions (conferences) such as
 Internet telephony calls.  SIP can also invite participants to
 already existing sessions, such as multicast conferences.  Media can
 be added to (and removed from) an existing session.  SIP
 transparently supports name mapping and redirection services, which
 supports personal mobility [27] - users can maintain a single
 externally visible identifier regardless of their network location.
 SIP supports five facets of establishing and terminating multimedia
 communications:
    User location: determination of the end system to be used for
         communication;
    User availability: determination of the willingness of the called
         party to engage in communications;
    User capabilities: determination of the media and media parameters
         to be used;
    Session setup: "ringing", establishment of session parameters at
         both called and calling party;
    Session management: including transfer and termination of
         sessions, modifying session parameters, and invoking
         services.
 SIP is not a vertically integrated communications system.  SIP is
 rather a component that can be used with other IETF protocols to
 build a complete multimedia architecture.  Typically, these
 architectures will include protocols such as the Real-time Transport
 Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and
 providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC
 2326 [29]) for controlling delivery of streaming media, the Media

Rosenberg, et. al. Standards Track [Page 9] RFC 3261 SIP: Session Initiation Protocol June 2002

 Gateway Control Protocol (MEGACO) (RFC 3015 [30]) for controlling
 gateways to the Public Switched Telephone Network (PSTN), and the
 Session Description Protocol (SDP) (RFC 2327 [1]) for describing
 multimedia sessions.  Therefore, SIP should be used in conjunction
 with other protocols in order to provide complete services to the
 users.  However, the basic functionality and operation of SIP does
 not depend on any of these protocols.
 SIP does not provide services.  Rather, SIP provides primitives that
 can be used to implement different services.  For example, SIP can
 locate a user and deliver an opaque object to his current location.
 If this primitive is used to deliver a session description written in
 SDP, for instance, the endpoints can agree on the parameters of a
 session.  If the same primitive is used to deliver a photo of the
 caller as well as the session description, a "caller ID" service can
 be easily implemented.  As this example shows, a single primitive is
 typically used to provide several different services.
 SIP does not offer conference control services such as floor control
 or voting and does not prescribe how a conference is to be managed.
 SIP can be used to initiate a session that uses some other conference
 control protocol.  Since SIP messages and the sessions they establish
 can pass through entirely different networks, SIP cannot, and does
 not, provide any kind of network resource reservation capabilities.
 The nature of the services provided make security particularly
 important.  To that end, SIP provides a suite of security services,
 which include denial-of-service prevention, authentication (both user
 to user and proxy to user), integrity protection, and encryption and
 privacy services.
 SIP works with both IPv4 and IPv6.

3 Terminology

 In this document, the key words "MUST", "MUST NOT", "REQUIRED",
 "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT
 RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as
 described in BCP 14, RFC 2119 [2] and indicate requirement levels for
 compliant SIP implementations.

4 Overview of Operation

 This section introduces the basic operations of SIP using simple
 examples.  This section is tutorial in nature and does not contain
 any normative statements.

Rosenberg, et. al. Standards Track [Page 10] RFC 3261 SIP: Session Initiation Protocol June 2002

 The first example shows the basic functions of SIP: location of an
 end point, signal of a desire to communicate, negotiation of session
 parameters to establish the session, and teardown of the session once
 established.
 Figure 1 shows a typical example of a SIP message exchange between
 two users, Alice and Bob.  (Each message is labeled with the letter
 "F" and a number for reference by the text.)  In this example, Alice
 uses a SIP application on her PC (referred to as a softphone) to call
 Bob on his SIP phone over the Internet.  Also shown are two SIP proxy
 servers that act on behalf of Alice and Bob to facilitate the session
 establishment.  This typical arrangement is often referred to as the
 "SIP trapezoid" as shown by the geometric shape of the dotted lines
 in Figure 1.
 Alice "calls" Bob using his SIP identity, a type of Uniform Resource
 Identifier (URI) called a SIP URI. SIP URIs are defined in Section
 19.1.  It has a similar form to an email address, typically
 containing a username and a host name.  In this case, it is
 sip:bob@biloxi.com, where biloxi.com is the domain of Bob's SIP
 service provider.  Alice has a SIP URI of sip:alice@atlanta.com.
 Alice might have typed in Bob's URI or perhaps clicked on a hyperlink
 or an entry in an address book.  SIP also provides a secure URI,
 called a SIPS URI.  An example would be sips:bob@biloxi.com.  A call
 made to a SIPS URI guarantees that secure, encrypted transport
 (namely TLS) is used to carry all SIP messages from the caller to the
 domain of the callee.  From there, the request is sent securely to
 the callee, but with security mechanisms that depend on the policy of
 the domain of the callee.
 SIP is based on an HTTP-like request/response transaction model.
 Each transaction consists of a request that invokes a particular
 method, or function, on the server and at least one response.  In
 this example, the transaction begins with Alice's softphone sending
 an INVITE request addressed to Bob's SIP URI.  INVITE is an example
 of a SIP method that specifies the action that the requestor (Alice)
 wants the server (Bob) to take.  The INVITE request contains a number
 of header fields.  Header fields are named attributes that provide
 additional information about a message.  The ones present in an
 INVITE include a unique identifier for the call, the destination
 address, Alice's address, and information about the type of session
 that Alice wishes to establish with Bob.  The INVITE (message F1 in
 Figure 1) might look like this:

Rosenberg, et. al. Standards Track [Page 11] RFC 3261 SIP: Session Initiation Protocol June 2002

                   atlanta.com  . . . biloxi.com
               .      proxy              proxy     .
             .                                       .
     Alice's  . . . . . . . . . . . . . . . . . . . .  Bob's
    softphone                                        SIP Phone
       |                |                |                |
       |    INVITE F1   |                |                |
       |--------------->|    INVITE F2   |                |
       |  100 Trying F3 |--------------->|    INVITE F4   |
       |<---------------|  100 Trying F5 |--------------->|
       |                |<-------------- | 180 Ringing F6 |
       |                | 180 Ringing F7 |<---------------|
       | 180 Ringing F8 |<---------------|     200 OK F9  |
       |<---------------|    200 OK F10  |<---------------|
       |    200 OK F11  |<---------------|                |
       |<---------------|                |                |
       |                       ACK F12                    |
       |------------------------------------------------->|
       |                   Media Session                  |
       |<================================================>|
       |                       BYE F13                    |
       |<-------------------------------------------------|
       |                     200 OK F14                   |
       |------------------------------------------------->|
       |                                                  |
       Figure 1: SIP session setup example with SIP trapezoid
    INVITE sip:bob@biloxi.com SIP/2.0
    Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds
    Max-Forwards: 70
    To: Bob <sip:bob@biloxi.com>
    From: Alice <sip:alice@atlanta.com>;tag=1928301774
    Call-ID: a84b4c76e66710@pc33.atlanta.com
    CSeq: 314159 INVITE
    Contact: <sip:alice@pc33.atlanta.com>
    Content-Type: application/sdp
    Content-Length: 142
    (Alice's SDP not shown)
 The first line of the text-encoded message contains the method name
 (INVITE).  The lines that follow are a list of header fields.  This
 example contains a minimum required set.  The header fields are
 briefly described below:

Rosenberg, et. al. Standards Track [Page 12] RFC 3261 SIP: Session Initiation Protocol June 2002

 Via contains the address (pc33.atlanta.com) at which Alice is
 expecting to receive responses to this request.  It also contains a
 branch parameter that identifies this transaction.
 To contains a display name (Bob) and a SIP or SIPS URI
 (sip:bob@biloxi.com) towards which the request was originally
 directed.  Display names are described in RFC 2822 [3].
 From also contains a display name (Alice) and a SIP or SIPS URI
 (sip:alice@atlanta.com) that indicate the originator of the request.
 This header field also has a tag parameter containing a random string
 (1928301774) that was added to the URI by the softphone.  It is used
 for identification purposes.
 Call-ID contains a globally unique identifier for this call,
 generated by the combination of a random string and the softphone's
 host name or IP address.  The combination of the To tag, From tag,
 and Call-ID completely defines a peer-to-peer SIP relationship
 between Alice and Bob and is referred to as a dialog.
 CSeq or Command Sequence contains an integer and a method name.  The
 CSeq number is incremented for each new request within a dialog and
 is a traditional sequence number.
 Contact contains a SIP or SIPS URI that represents a direct route to
 contact Alice, usually composed of a username at a fully qualified
 domain name (FQDN).  While an FQDN is preferred, many end systems do
 not have registered domain names, so IP addresses are permitted.
 While the Via header field tells other elements where to send the
 response, the Contact header field tells other elements where to send
 future requests.
 Max-Forwards serves to limit the number of hops a request can make on
 the way to its destination.  It consists of an integer that is
 decremented by one at each hop.
 Content-Type contains a description of the message body (not shown).
 Content-Length contains an octet (byte) count of the message body.
 The complete set of SIP header fields is defined in Section 20.
 The details of the session, such as the type of media, codec, or
 sampling rate, are not described using SIP.  Rather, the body of a
 SIP message contains a description of the session, encoded in some
 other protocol format.  One such format is the Session Description
 Protocol (SDP) (RFC 2327 [1]).  This SDP message (not shown in the

Rosenberg, et. al. Standards Track [Page 13] RFC 3261 SIP: Session Initiation Protocol June 2002

 example) is carried by the SIP message in a way that is analogous to
 a document attachment being carried by an email message, or a web
 page being carried in an HTTP message.
 Since the softphone does not know the location of Bob or the SIP
 server in the biloxi.com domain, the softphone sends the INVITE to
 the SIP server that serves Alice's domain, atlanta.com.  The address
 of the atlanta.com SIP server could have been configured in Alice's
 softphone, or it could have been discovered by DHCP, for example.
 The atlanta.com SIP server is a type of SIP server known as a proxy
 server.  A proxy server receives SIP requests and forwards them on
 behalf of the requestor.  In this example, the proxy server receives
 the INVITE request and sends a 100 (Trying) response back to Alice's
 softphone.  The 100 (Trying) response indicates that the INVITE has
 been received and that the proxy is working on her behalf to route
 the INVITE to the destination.  Responses in SIP use a three-digit
 code followed by a descriptive phrase.  This response contains the
 same To, From, Call-ID, CSeq and branch parameter in the Via as the
 INVITE, which allows Alice's softphone to correlate this response to
 the sent INVITE.  The atlanta.com proxy server locates the proxy
 server at biloxi.com, possibly by performing a particular type of DNS
 (Domain Name Service) lookup to find the SIP server that serves the
 biloxi.com domain.  This is described in [4].  As a result, it
 obtains the IP address of the biloxi.com proxy server and forwards,
 or proxies, the INVITE request there.  Before forwarding the request,
 the atlanta.com proxy server adds an additional Via header field
 value that contains its own address (the INVITE already contains
 Alice's address in the first Via).  The biloxi.com proxy server
 receives the INVITE and responds with a 100 (Trying) response back to
 the atlanta.com proxy server to indicate that it has received the
 INVITE and is processing the request.  The proxy server consults a
 database, generically called a location service, that contains the
 current IP address of Bob.  (We shall see in the next section how
 this database can be populated.)  The biloxi.com proxy server adds
 another Via header field value with its own address to the INVITE and
 proxies it to Bob's SIP phone.
 Bob's SIP phone receives the INVITE and alerts Bob to the incoming
 call from Alice so that Bob can decide whether to answer the call,
 that is, Bob's phone rings.  Bob's SIP phone indicates this in a 180
 (Ringing) response, which is routed back through the two proxies in
 the reverse direction.  Each proxy uses the Via header field to
 determine where to send the response and removes its own address from
 the top.  As a result, although DNS and location service lookups were
 required to route the initial INVITE, the 180 (Ringing) response can
 be returned to the caller without lookups or without state being

Rosenberg, et. al. Standards Track [Page 14] RFC 3261 SIP: Session Initiation Protocol June 2002

 maintained in the proxies.  This also has the desirable property that
 each proxy that sees the INVITE will also see all responses to the
 INVITE.
 When Alice's softphone receives the 180 (Ringing) response, it passes
 this information to Alice, perhaps using an audio ringback tone or by
 displaying a message on Alice's screen.
 In this example, Bob decides to answer the call.  When he picks up
 the handset, his SIP phone sends a 200 (OK) response to indicate that
 the call has been answered.  The 200 (OK) contains a message body
 with the SDP media description of the type of session that Bob is
 willing to establish with Alice.  As a result, there is a two-phase
 exchange of SDP messages: Alice sent one to Bob, and Bob sent one
 back to Alice.  This two-phase exchange provides basic negotiation
 capabilities and is based on a simple offer/answer model of SDP
 exchange.  If Bob did not wish to answer the call or was busy on
 another call, an error response would have been sent instead of the
 200 (OK), which would have resulted in no media session being
 established.  The complete list of SIP response codes is in Section
 21.  The 200 (OK) (message F9 in Figure 1) might look like this as
 Bob sends it out:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP server10.biloxi.com
       ;branch=z9hG4bKnashds8;received=192.0.2.3
    Via: SIP/2.0/UDP bigbox3.site3.atlanta.com
       ;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2
    Via: SIP/2.0/UDP pc33.atlanta.com
       ;branch=z9hG4bK776asdhds ;received=192.0.2.1
    To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
    From: Alice <sip:alice@atlanta.com>;tag=1928301774
    Call-ID: a84b4c76e66710@pc33.atlanta.com
    CSeq: 314159 INVITE
    Contact: <sip:bob@192.0.2.4>
    Content-Type: application/sdp
    Content-Length: 131
    (Bob's SDP not shown)
 The first line of the response contains the response code (200) and
 the reason phrase (OK).  The remaining lines contain header fields.
 The Via, To, From, Call-ID, and CSeq header fields are copied from
 the INVITE request.  (There are three Via header field values - one
 added by Alice's SIP phone, one added by the atlanta.com proxy, and
 one added by the biloxi.com proxy.)  Bob's SIP phone has added a tag
 parameter to the To header field.  This tag will be incorporated by
 both endpoints into the dialog and will be included in all future

Rosenberg, et. al. Standards Track [Page 15] RFC 3261 SIP: Session Initiation Protocol June 2002

 requests and responses in this call.  The Contact header field
 contains a URI at which Bob can be directly reached at his SIP phone.
 The Content-Type and Content-Length refer to the message body (not
 shown) that contains Bob's SDP media information.
 In addition to DNS and location service lookups shown in this
 example, proxy servers can make flexible "routing decisions" to
 decide where to send a request.  For example, if Bob's SIP phone
 returned a 486 (Busy Here) response, the biloxi.com proxy server
 could proxy the INVITE to Bob's voicemail server.  A proxy server can
 also send an INVITE to a number of locations at the same time.  This
 type of parallel search is known as forking.
 In this case, the 200 (OK) is routed back through the two proxies and
 is received by Alice's softphone, which then stops the ringback tone
 and indicates that the call has been answered.  Finally, Alice's
 softphone sends an acknowledgement message, ACK, to Bob's SIP phone
 to confirm the reception of the final response (200 (OK)).  In this
 example, the ACK is sent directly from Alice's softphone to Bob's SIP
 phone, bypassing the two proxies.  This occurs because the endpoints
 have learned each other's address from the Contact header fields
 through the INVITE/200 (OK) exchange, which was not known when the
 initial INVITE was sent.  The lookups performed by the two proxies
 are no longer needed, so the proxies drop out of the call flow.  This
 completes the INVITE/200/ACK three-way handshake used to establish
 SIP sessions.  Full details on session setup are in Section 13.
 Alice and Bob's media session has now begun, and they send media
 packets using the format to which they agreed in the exchange of SDP.
 In general, the end-to-end media packets take a different path from
 the SIP signaling messages.
 During the session, either Alice or Bob may decide to change the
 characteristics of the media session.  This is accomplished by
 sending a re-INVITE containing a new media description.  This re-
 INVITE references the existing dialog so that the other party knows
 that it is to modify an existing session instead of establishing a
 new session.  The other party sends a 200 (OK) to accept the change.
 The requestor responds to the 200 (OK) with an ACK.  If the other
 party does not accept the change, he sends an error response such as
 488 (Not Acceptable Here), which also receives an ACK.  However, the
 failure of the re-INVITE does not cause the existing call to fail -
 the session continues using the previously negotiated
 characteristics.  Full details on session modification are in Section
 14.

Rosenberg, et. al. Standards Track [Page 16] RFC 3261 SIP: Session Initiation Protocol June 2002

 At the end of the call, Bob disconnects (hangs up) first and
 generates a BYE message.  This BYE is routed directly to Alice's
 softphone, again bypassing the proxies.  Alice confirms receipt of
 the BYE with a 200 (OK) response, which terminates the session and
 the BYE transaction.  No ACK is sent - an ACK is only sent in
 response to a response to an INVITE request.  The reasons for this
 special handling for INVITE will be discussed later, but relate to
 the reliability mechanisms in SIP, the length of time it can take for
 a ringing phone to be answered, and forking.  For this reason,
 request handling in SIP is often classified as either INVITE or non-
 INVITE, referring to all other methods besides INVITE.  Full details
 on session termination are in Section 15.
 Section 24.2 describes the messages shown in Figure 1 in full.
 In some cases, it may be useful for proxies in the SIP signaling path
 to see all the messaging between the endpoints for the duration of
 the session.  For example, if the biloxi.com proxy server wished to
 remain in the SIP messaging path beyond the initial INVITE, it would
 add to the INVITE a required routing header field known as Record-
 Route that contained a URI resolving to the hostname or IP address of
 the proxy.  This information would be received by both Bob's SIP
 phone and (due to the Record-Route header field being passed back in
 the 200 (OK)) Alice's softphone and stored for the duration of the
 dialog.  The biloxi.com proxy server would then receive and proxy the
 ACK, BYE, and 200 (OK) to the BYE.  Each proxy can independently
 decide to receive subsequent messages, and those messages will pass
 through all proxies that elect to receive it.  This capability is
 frequently used for proxies that are providing mid-call features.
 Registration is another common operation in SIP.  Registration is one
 way that the biloxi.com server can learn the current location of Bob.
 Upon initialization, and at periodic intervals, Bob's SIP phone sends
 REGISTER messages to a server in the biloxi.com domain known as a SIP
 registrar.  The REGISTER messages associate Bob's SIP or SIPS URI
 (sip:bob@biloxi.com) with the machine into which he is currently
 logged (conveyed as a SIP or SIPS URI in the Contact header field).
 The registrar writes this association, also called a binding, to a
 database, called the location service, where it can be used by the
 proxy in the biloxi.com domain.  Often, a registrar server for a
 domain is co-located with the proxy for that domain.  It is an
 important concept that the distinction between types of SIP servers
 is logical, not physical.
 Bob is not limited to registering from a single device.  For example,
 both his SIP phone at home and the one in the office could send
 registrations.  This information is stored together in the location

Rosenberg, et. al. Standards Track [Page 17] RFC 3261 SIP: Session Initiation Protocol June 2002

 service and allows a proxy to perform various types of searches to
 locate Bob.  Similarly, more than one user can be registered on a
 single device at the same time.
 The location service is just an abstract concept.  It generally
 contains information that allows a proxy to input a URI and receive a
 set of zero or more URIs that tell the proxy where to send the
 request.  Registrations are one way to create this information, but
 not the only way.  Arbitrary mapping functions can be configured at
 the discretion of the administrator.
 Finally, it is important to note that in SIP, registration is used
 for routing incoming SIP requests and has no role in authorizing
 outgoing requests.  Authorization and authentication are handled in
 SIP either on a request-by-request basis with a challenge/response
 mechanism, or by using a lower layer scheme as discussed in Section
 26.
 The complete set of SIP message details for this registration example
 is in Section 24.1.
 Additional operations in SIP, such as querying for the capabilities
 of a SIP server or client using OPTIONS, or canceling a pending
 request using CANCEL, will be introduced in later sections.

5 Structure of the Protocol

 SIP is structured as a layered protocol, which means that its
 behavior is described in terms of a set of fairly independent
 processing stages with only a loose coupling between each stage.  The
 protocol behavior is described as layers for the purpose of
 presentation, allowing the description of functions common across
 elements in a single section.  It does not dictate an implementation
 in any way.  When we say that an element "contains" a layer, we mean
 it is compliant to the set of rules defined by that layer.
 Not every element specified by the protocol contains every layer.
 Furthermore, the elements specified by SIP are logical elements, not
 physical ones.  A physical realization can choose to act as different
 logical elements, perhaps even on a transaction-by-transaction basis.
 The lowest layer of SIP is its syntax and encoding.  Its encoding is
 specified using an augmented Backus-Naur Form grammar (BNF).  The
 complete BNF is specified in Section 25; an overview of a SIP
 message's structure can be found in Section 7.

Rosenberg, et. al. Standards Track [Page 18] RFC 3261 SIP: Session Initiation Protocol June 2002

 The second layer is the transport layer.  It defines how a client
 sends requests and receives responses and how a server receives
 requests and sends responses over the network.  All SIP elements
 contain a transport layer.  The transport layer is described in
 Section 18.
 The third layer is the transaction layer.  Transactions are a
 fundamental component of SIP.  A transaction is a request sent by a
 client transaction (using the transport layer) to a server
 transaction, along with all responses to that request sent from the
 server transaction back to the client.  The transaction layer handles
 application-layer retransmissions, matching of responses to requests,
 and application-layer timeouts.  Any task that a user agent client
 (UAC) accomplishes takes place using a series of transactions.
 Discussion of transactions can be found in Section 17.  User agents
 contain a transaction layer, as do stateful proxies.  Stateless
 proxies do not contain a transaction layer.  The transaction layer
 has a client component (referred to as a client transaction) and a
 server component (referred to as a server transaction), each of which
 are represented by a finite state machine that is constructed to
 process a particular request.
 The layer above the transaction layer is called the transaction user
 (TU).  Each of the SIP entities, except the stateless proxy, is a
 transaction user.  When a TU wishes to send a request, it creates a
 client transaction instance and passes it the request along with the
 destination IP address, port, and transport to which to send the
 request.  A TU that creates a client transaction can also cancel it.
 When a client cancels a transaction, it requests that the server stop
 further processing, revert to the state that existed before the
 transaction was initiated, and generate a specific error response to
 that transaction.  This is done with a CANCEL request, which
 constitutes its own transaction, but references the transaction to be
 cancelled (Section 9).
 The SIP elements, that is, user agent clients and servers, stateless
 and stateful proxies and registrars, contain a core that
 distinguishes them from each other.  Cores, except for the stateless
 proxy, are transaction users.  While the behavior of the UAC and UAS
 cores depends on the method, there are some common rules for all
 methods (Section 8).  For a UAC, these rules govern the construction
 of a request; for a UAS, they govern the processing of a request and
 generating a response.  Since registrations play an important role in
 SIP, a UAS that handles a REGISTER is given the special name
 registrar.  Section 10 describes UAC and UAS core behavior for the
 REGISTER method.  Section 11 describes UAC and UAS core behavior for
 the OPTIONS method, used for determining the capabilities of a UA.

Rosenberg, et. al. Standards Track [Page 19] RFC 3261 SIP: Session Initiation Protocol June 2002

 Certain other requests are sent within a dialog.  A dialog is a
 peer-to-peer SIP relationship between two user agents that persists
 for some time.  The dialog facilitates sequencing of messages and
 proper routing of requests between the user agents.  The INVITE
 method is the only way defined in this specification to establish a
 dialog.  When a UAC sends a request that is within the context of a
 dialog, it follows the common UAC rules as discussed in Section 8 but
 also the rules for mid-dialog requests.  Section 12 discusses dialogs
 and presents the procedures for their construction and maintenance,
 in addition to construction of requests within a dialog.
 The most important method in SIP is the INVITE method, which is used
 to establish a session between participants.  A session is a
 collection of participants, and streams of media between them, for
 the purposes of communication.  Section 13 discusses how sessions are
 initiated, resulting in one or more SIP dialogs.  Section 14
 discusses how characteristics of that session are modified through
 the use of an INVITE request within a dialog.  Finally, section 15
 discusses how a session is terminated.
 The procedures of Sections 8, 10, 11, 12, 13, 14, and 15 deal
 entirely with the UA core (Section 9 describes cancellation, which
 applies to both UA core and proxy core).  Section 16 discusses the
 proxy element, which facilitates routing of messages between user
 agents.

6 Definitions

 The following terms have special significance for SIP.
    Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
       that points to a domain with a location service that can map
       the URI to another URI where the user might be available.
       Typically, the location service is populated through
       registrations.  An AOR is frequently thought of as the "public
       address" of the user.
    Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a
       logical entity that receives a request and processes it as a
       user agent server (UAS).  In order to determine how the request
       should be answered, it acts as a user agent client (UAC) and
       generates requests.  Unlike a proxy server, it maintains dialog
       state and must participate in all requests sent on the dialogs
       it has established.  Since it is a concatenation of a UAC and
       UAS, no explicit definitions are needed for its behavior.

Rosenberg, et. al. Standards Track [Page 20] RFC 3261 SIP: Session Initiation Protocol June 2002

    Call: A call is an informal term that refers to some communication
       between peers, generally set up for the purposes of a
       multimedia conversation.
    Call Leg: Another name for a dialog [31]; no longer used in this
       specification.
    Call Stateful: A proxy is call stateful if it retains state for a
       dialog from the initiating INVITE to the terminating BYE
       request.  A call stateful proxy is always transaction stateful,
       but the converse is not necessarily true.
    Client: A client is any network element that sends SIP requests
       and receives SIP responses.  Clients may or may not interact
       directly with a human user.  User agent clients and proxies are
       clients.
    Conference: A multimedia session (see below) that contains
       multiple participants.
    Core: Core designates the functions specific to a particular type
       of SIP entity, i.e., specific to either a stateful or stateless
       proxy, a user agent or registrar.  All cores, except those for
       the stateless proxy, are transaction users.
    Dialog: A dialog is a peer-to-peer SIP relationship between two
       UAs that persists for some time.  A dialog is established by
       SIP messages, such as a 2xx response to an INVITE request.  A
       dialog is identified by a call identifier, local tag, and a
       remote tag.  A dialog was formerly known as a call leg in RFC
       2543.
    Downstream: A direction of message forwarding within a transaction
       that refers to the direction that requests flow from the user
       agent client to user agent server.
    Final Response: A response that terminates a SIP transaction, as
       opposed to a provisional response that does not.  All 2xx, 3xx,
       4xx, 5xx and 6xx responses are final.
    Header: A header is a component of a SIP message that conveys
       information about the message.  It is structured as a sequence
       of header fields.
    Header Field: A header field is a component of the SIP message
       header.  A header field can appear as one or more header field
       rows. Header field rows consist of a header field name and zero
       or more header field values. Multiple header field values on a

Rosenberg, et. al. Standards Track [Page 21] RFC 3261 SIP: Session Initiation Protocol June 2002

       given header field row are separated by commas. Some header
       fields can only have a single header field value, and as a
       result, always appear as a single header field row.
    Header Field Value: A header field value is a single value; a
       header field consists of zero or more header field values.
    Home Domain: The domain providing service to a SIP user.
       Typically, this is the domain present in the URI in the
       address-of-record of a registration.
    Informational Response: Same as a provisional response.
    Initiator, Calling Party, Caller: The party initiating a session
       (and dialog) with an INVITE request.  A caller retains this
       role from the time it sends the initial INVITE that established
       a dialog until the termination of that dialog.
    Invitation: An INVITE request.
    Invitee, Invited User, Called Party, Callee: The party that
       receives an INVITE request for the purpose of establishing a
       new session.  A callee retains this role from the time it
       receives the INVITE until the termination of the dialog
       established by that INVITE.
    Location Service: A location service is used by a SIP redirect or
       proxy server to obtain information about a callee's possible
       location(s).  It contains a list of bindings of address-of-
       record keys to zero or more contact addresses.  The bindings
       can be created and removed in many ways; this specification
       defines a REGISTER method that updates the bindings.
    Loop: A request that arrives at a proxy, is forwarded, and later
       arrives back at the same proxy.  When it arrives the second
       time, its Request-URI is identical to the first time, and other
       header fields that affect proxy operation are unchanged, so
       that the proxy would make the same processing decision on the
       request it made the first time.  Looped requests are errors,
       and the procedures for detecting them and handling them are
       described by the protocol.
    Loose Routing: A proxy is said to be loose routing if it follows
       the procedures defined in this specification for processing of
       the Route header field.  These procedures separate the
       destination of the request (present in the Request-URI) from

Rosenberg, et. al. Standards Track [Page 22] RFC 3261 SIP: Session Initiation Protocol June 2002

       the set of proxies that need to be visited along the way
       (present in the Route header field).  A proxy compliant to
       these mechanisms is also known as a loose router.
    Message: Data sent between SIP elements as part of the protocol.
       SIP messages are either requests or responses.
    Method: The method is the primary function that a request is meant
       to invoke on a server.  The method is carried in the request
       message itself.  Example methods are INVITE and BYE.
    Outbound Proxy: A proxy that receives requests from a client, even
       though it may not be the server resolved by the Request-URI.
       Typically, a UA is manually configured with an outbound proxy,
       or can learn about one through auto-configuration protocols.
    Parallel Search: In a parallel search, a proxy issues several
       requests to possible user locations upon receiving an incoming
       request.  Rather than issuing one request and then waiting for
       the final response before issuing the next request as in a
       sequential search, a parallel search issues requests without
       waiting for the result of previous requests.
    Provisional Response: A response used by the server to indicate
       progress, but that does not terminate a SIP transaction.  1xx
       responses are provisional, other responses are considered
       final.
    Proxy, Proxy Server: An intermediary entity that acts as both a
       server and a client for the purpose of making requests on
       behalf of other clients.  A proxy server primarily plays the
       role of routing, which means its job is to ensure that a
       request is sent to another entity "closer" to the targeted
       user.  Proxies are also useful for enforcing policy (for
       example, making sure a user is allowed to make a call).  A
       proxy interprets, and, if necessary, rewrites specific parts of
       a request message before forwarding it.
    Recursion: A client recurses on a 3xx response when it generates a
       new request to one or more of the URIs in the Contact header
       field in the response.
    Redirect Server: A redirect server is a user agent server that
       generates 3xx responses to requests it receives, directing the
       client to contact an alternate set of URIs.

Rosenberg, et. al. Standards Track [Page 23] RFC 3261 SIP: Session Initiation Protocol June 2002

    Registrar: A registrar is a server that accepts REGISTER requests
       and places the information it receives in those requests into
       the location service for the domain it handles.
    Regular Transaction: A regular transaction is any transaction with
       a method other than INVITE, ACK, or CANCEL.
    Request: A SIP message sent from a client to a server, for the
       purpose of invoking a particular operation.
    Response: A SIP message sent from a server to a client, for
       indicating the status of a request sent from the client to the
       server.
    Ringback: Ringback is the signaling tone produced by the calling
       party's application indicating that a called party is being
       alerted (ringing).
    Route Set: A route set is a collection of ordered SIP or SIPS URI
       which represent a list of proxies that must be traversed when
       sending a particular request.  A route set can be learned,
       through headers like Record-Route, or it can be configured.
    Server: A server is a network element that receives requests in
       order to service them and sends back responses to those
       requests.  Examples of servers are proxies, user agent servers,
       redirect servers, and registrars.
    Sequential Search: In a sequential search, a proxy server attempts
       each contact address in sequence, proceeding to the next one
       only after the previous has generated a final response.  A 2xx
       or 6xx class final response always terminates a sequential
       search.
    Session: From the SDP specification: "A multimedia session is a
       set of multimedia senders and receivers and the data streams
       flowing from senders to receivers.  A multimedia conference is
       an example of a multimedia session." (RFC 2327 [1]) (A session
       as defined for SDP can comprise one or more RTP sessions.)  As
       defined, a callee can be invited several times, by different
       calls, to the same session.  If SDP is used, a session is
       defined by the concatenation of the SDP user name, session id,
       network type, address type, and address elements in the origin
       field.
    SIP Transaction: A SIP transaction occurs between a client and a
       server and comprises all messages from the first request sent
       from the client to the server up to a final (non-1xx) response

Rosenberg, et. al. Standards Track [Page 24] RFC 3261 SIP: Session Initiation Protocol June 2002

       sent from the server to the client.  If the request is INVITE
       and the final response is a non-2xx, the transaction also
       includes an ACK to the response.  The ACK for a 2xx response to
       an INVITE request is a separate transaction.
    Spiral: A spiral is a SIP request that is routed to a proxy,
       forwarded onwards, and arrives once again at that proxy, but
       this time differs in a way that will result in a different
       processing decision than the original request.  Typically, this
       means that the request's Request-URI differs from its previous
       arrival.  A spiral is not an error condition, unlike a loop.  A
       typical cause for this is call forwarding.  A user calls
       joe@example.com.  The example.com proxy forwards it to Joe's
       PC, which in turn, forwards it to bob@example.com.  This
       request is proxied back to the example.com proxy.  However,
       this is not a loop.  Since the request is targeted at a
       different user, it is considered a spiral, and is a valid
       condition.
    Stateful Proxy: A logical entity that maintains the client and
       server transaction state machines defined by this specification
       during the processing of a request, also known as a transaction
       stateful proxy.  The behavior of a stateful proxy is further
       defined in Section 16.  A (transaction) stateful proxy is not
       the same as a call stateful proxy.
    Stateless Proxy: A logical entity that does not maintain the
       client or server transaction state machines defined in this
       specification when it processes requests.  A stateless proxy
       forwards every request it receives downstream and every
       response it receives upstream.
    Strict Routing: A proxy is said to be strict routing if it follows
       the Route processing rules of RFC 2543 and many prior work in
       progress versions of this RFC.  That rule caused proxies to
       destroy the contents of the Request-URI when a Route header
       field was present.  Strict routing behavior is not used in this
       specification, in favor of a loose routing behavior.  Proxies
       that perform strict routing are also known as strict routers.
    Target Refresh Request: A target refresh request sent within a
       dialog is defined as a request that can modify the remote
       target of the dialog.
    Transaction User (TU): The layer of protocol processing that
       resides above the transaction layer.  Transaction users include
       the UAC core, UAS core, and proxy core.

Rosenberg, et. al. Standards Track [Page 25] RFC 3261 SIP: Session Initiation Protocol June 2002

    Upstream: A direction of message forwarding within a transaction
       that refers to the direction that responses flow from the user
       agent server back to the user agent client.
    URL-encoded: A character string encoded according to RFC 2396,
       Section 2.4 [5].
    User Agent Client (UAC): A user agent client is a logical entity
       that creates a new request, and then uses the client
       transaction state machinery to send it.  The role of UAC lasts
       only for the duration of that transaction.  In other words, if
       a piece of software initiates a request, it acts as a UAC for
       the duration of that transaction.  If it receives a request
       later, it assumes the role of a user agent server for the
       processing of that transaction.
    UAC Core: The set of processing functions required of a UAC that
       reside above the transaction and transport layers.
    User Agent Server (UAS): A user agent server is a logical entity
       that generates a response to a SIP request.  The response
       accepts, rejects, or redirects the request.  This role lasts
       only for the duration of that transaction.  In other words, if
       a piece of software responds to a request, it acts as a UAS for
       the duration of that transaction.  If it generates a request
       later, it assumes the role of a user agent client for the
       processing of that transaction.
    UAS Core: The set of processing functions required at a UAS that
       resides above the transaction and transport layers.
    User Agent (UA): A logical entity that can act as both a user
       agent client and user agent server.
 The role of UAC and UAS, as well as proxy and redirect servers, are
 defined on a transaction-by-transaction basis.  For example, the user
 agent initiating a call acts as a UAC when sending the initial INVITE
 request and as a UAS when receiving a BYE request from the callee.
 Similarly, the same software can act as a proxy server for one
 request and as a redirect server for the next request.
 Proxy, location, and registrar servers defined above are logical
 entities; implementations MAY combine them into a single application.

7 SIP Messages

 SIP is a text-based protocol and uses the UTF-8 charset (RFC 2279
 [7]).

Rosenberg, et. al. Standards Track [Page 26] RFC 3261 SIP: Session Initiation Protocol June 2002

 A SIP message is either a request from a client to a server, or a
 response from a server to a client.
 Both Request (section 7.1) and Response (section 7.2) messages use
 the basic format of RFC 2822 [3], even though the syntax differs in
 character set and syntax specifics.  (SIP allows header fields that
 would not be valid RFC 2822 header fields, for example.)  Both types
 of messages consist of a start-line, one or more header fields, an
 empty line indicating the end of the header fields, and an optional
 message-body.
       generic-message  =  start-line
                           *message-header
                           CRLF
                           [ message-body ]
       start-line       =  Request-Line / Status-Line
 The start-line, each message-header line, and the empty line MUST be
 terminated by a carriage-return line-feed sequence (CRLF).  Note that
 the empty line MUST be present even if the message-body is not.
 Except for the above difference in character sets, much of SIP's
 message and header field syntax is identical to HTTP/1.1.  Rather
 than repeating the syntax and semantics here, we use [HX.Y] to refer
 to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [8]).
 However, SIP is not an extension of HTTP.

7.1 Requests

 SIP requests are distinguished by having a Request-Line for a start-
 line.  A Request-Line contains a method name, a Request-URI, and the
 protocol version separated by a single space (SP) character.
 The Request-Line ends with CRLF.  No CR or LF are allowed except in
 the end-of-line CRLF sequence.  No linear whitespace (LWS) is allowed
 in any of the elements.
       Request-Line  =  Method SP Request-URI SP SIP-Version CRLF
    Method: This specification defines six methods: REGISTER for
         registering contact information, INVITE, ACK, and CANCEL for
         setting up sessions, BYE for terminating sessions, and
         OPTIONS for querying servers about their capabilities.  SIP
         extensions, documented in standards track RFCs, may define
         additional methods.

Rosenberg, et. al. Standards Track [Page 27] RFC 3261 SIP: Session Initiation Protocol June 2002

    Request-URI: The Request-URI is a SIP or SIPS URI as described in
         Section 19.1 or a general URI (RFC 2396 [5]).  It indicates
         the user or service to which this request is being addressed.
         The Request-URI MUST NOT contain unescaped spaces or control
         characters and MUST NOT be enclosed in "<>".
         SIP elements MAY support Request-URIs with schemes other than
         "sip" and "sips", for example the "tel" URI scheme of RFC
         2806 [9].  SIP elements MAY translate non-SIP URIs using any
         mechanism at their disposal, resulting in SIP URI, SIPS URI,
         or some other scheme.
    SIP-Version: Both request and response messages include the
         version of SIP in use, and follow [H3.1] (with HTTP replaced
         by SIP, and HTTP/1.1 replaced by SIP/2.0) regarding version
         ordering, compliance requirements, and upgrading of version
         numbers.  To be compliant with this specification,
         applications sending SIP messages MUST include a SIP-Version
         of "SIP/2.0".  The SIP-Version string is case-insensitive,
         but implementations MUST send upper-case.
         Unlike HTTP/1.1, SIP treats the version number as a literal
         string.  In practice, this should make no difference.

7.2 Responses

 SIP responses are distinguished from requests by having a Status-Line
 as their start-line.  A Status-Line consists of the protocol version
 followed by a numeric Status-Code and its associated textual phrase,
 with each element separated by a single SP character.
 No CR or LF is allowed except in the final CRLF sequence.
    Status-Line  =  SIP-Version SP Status-Code SP Reason-Phrase CRLF
 The Status-Code is a 3-digit integer result code that indicates the
 outcome of an attempt to understand and satisfy a request.  The
 Reason-Phrase is intended to give a short textual description of the
 Status-Code.  The Status-Code is intended for use by automata,
 whereas the Reason-Phrase is intended for the human user.  A client
 is not required to examine or display the Reason-Phrase.
 While this specification suggests specific wording for the reason
 phrase, implementations MAY choose other text, for example, in the
 language indicated in the Accept-Language header field of the
 request.

Rosenberg, et. al. Standards Track [Page 28] RFC 3261 SIP: Session Initiation Protocol June 2002

 The first digit of the Status-Code defines the class of response.
 The last two digits do not have any categorization role.  For this
 reason, any response with a status code between 100 and 199 is
 referred to as a "1xx response", any response with a status code
 between 200 and 299 as a "2xx response", and so on.  SIP/2.0 allows
 six values for the first digit:
    1xx: Provisional -- request received, continuing to process the
         request;
    2xx: Success -- the action was successfully received, understood,
         and accepted;
    3xx: Redirection -- further action needs to be taken in order to
         complete the request;
    4xx: Client Error -- the request contains bad syntax or cannot be
         fulfilled at this server;
    5xx: Server Error -- the server failed to fulfill an apparently
         valid request;
    6xx: Global Failure -- the request cannot be fulfilled at any
         server.
 Section 21 defines these classes and describes the individual codes.

7.3 Header Fields

 SIP header fields are similar to HTTP header fields in both syntax
 and semantics.  In particular, SIP header fields follow the [H4.2]
 definitions of syntax for the message-header and the rules for
 extending header fields over multiple lines.  However, the latter is
 specified in HTTP with implicit whitespace and folding.  This
 specification conforms to RFC 2234 [10] and uses only explicit
 whitespace and folding as an integral part of the grammar.
 [H4.2] also specifies that multiple header fields of the same field
 name whose value is a comma-separated list can be combined into one
 header field.  That applies to SIP as well, but the specific rule is
 different because of the different grammars.  Specifically, any SIP
 header whose grammar is of the form
    header  =  "header-name" HCOLON header-value *(COMMA header-value)
 allows for combining header fields of the same name into a comma-
 separated list.  The Contact header field allows a comma-separated
 list unless the header field value is "*".

Rosenberg, et. al. Standards Track [Page 29] RFC 3261 SIP: Session Initiation Protocol June 2002

7.3.1 Header Field Format

 Header fields follow the same generic header format as that given in
 Section 2.2 of RFC 2822 [3].  Each header field consists of a field
 name followed by a colon (":") and the field value.
    field-name: field-value
 The formal grammar for a message-header specified in Section 25
 allows for an arbitrary amount of whitespace on either side of the
 colon; however, implementations should avoid spaces between the field
 name and the colon and use a single space (SP) between the colon and
 the field-value.
    Subject:            lunch
    Subject      :      lunch
    Subject            :lunch
    Subject: lunch
 Thus, the above are all valid and equivalent, but the last is the
 preferred form.
 Header fields can be extended over multiple lines by preceding each
 extra line with at least one SP or horizontal tab (HT).  The line
 break and the whitespace at the beginning of the next line are
 treated as a single SP character.  Thus, the following are
 equivalent:
    Subject: I know you're there, pick up the phone and talk to me!
    Subject: I know you're there,
             pick up the phone
             and talk to me!
 The relative order of header fields with different field names is not
 significant.  However, it is RECOMMENDED that header fields which are
 needed for proxy processing (Via, Route, Record-Route, Proxy-Require,
 Max-Forwards, and Proxy-Authorization, for example) appear towards
 the top of the message to facilitate rapid parsing.  The relative
 order of header field rows with the same field name is important.
 Multiple header field rows with the same field-name MAY be present in
 a message if and only if the entire field-value for that header field
 is defined as a comma-separated list (that is, if follows the grammar
 defined in Section 7.3).  It MUST be possible to combine the multiple
 header field rows into one "field-name: field-value" pair, without
 changing the semantics of the message, by appending each subsequent
 field-value to the first, each separated by a comma.  The exceptions
 to this rule are the WWW-Authenticate, Authorization, Proxy-
 Authenticate, and Proxy-Authorization header fields.  Multiple header

Rosenberg, et. al. Standards Track [Page 30] RFC 3261 SIP: Session Initiation Protocol June 2002

 field rows with these names MAY be present in a message, but since
 their grammar does not follow the general form listed in Section 7.3,
 they MUST NOT be combined into a single header field row.
 Implementations MUST be able to process multiple header field rows
 with the same name in any combination of the single-value-per-line or
 comma-separated value forms.
 The following groups of header field rows are valid and equivalent:
    Route: <sip:alice@atlanta.com>
    Subject: Lunch
    Route: <sip:bob@biloxi.com>
    Route: <sip:carol@chicago.com>
    Route: <sip:alice@atlanta.com>, <sip:bob@biloxi.com>
    Route: <sip:carol@chicago.com>
    Subject: Lunch
    Subject: Lunch
    Route: <sip:alice@atlanta.com>, <sip:bob@biloxi.com>,
           <sip:carol@chicago.com>
 Each of the following blocks is valid but not equivalent to the
 others:
    Route: <sip:alice@atlanta.com>
    Route: <sip:bob@biloxi.com>
    Route: <sip:carol@chicago.com>
    Route: <sip:bob@biloxi.com>
    Route: <sip:alice@atlanta.com>
    Route: <sip:carol@chicago.com>
    Route: <sip:alice@atlanta.com>,<sip:carol@chicago.com>,
           <sip:bob@biloxi.com>
 The format of a header field-value is defined per header-name.  It
 will always be either an opaque sequence of TEXT-UTF8 octets, or a
 combination of whitespace, tokens, separators, and quoted strings.
 Many existing header fields will adhere to the general form of a
 value followed by a semi-colon separated sequence of parameter-name,
 parameter-value pairs:
       field-name: field-value *(;parameter-name=parameter-value)

Rosenberg, et. al. Standards Track [Page 31] RFC 3261 SIP: Session Initiation Protocol June 2002

 Even though an arbitrary number of parameter pairs may be attached to
 a header field value, any given parameter-name MUST NOT appear more
 than once.
 When comparing header fields, field names are always case-
 insensitive.  Unless otherwise stated in the definition of a
 particular header field, field values, parameter names, and parameter
 values are case-insensitive.  Tokens are always case-insensitive.
 Unless specified otherwise, values expressed as quoted strings are
 case-sensitive.  For example,
    Contact: <sip:alice@atlanta.com>;expires=3600
 is equivalent to
    CONTACT: <sip:alice@atlanta.com>;ExPiReS=3600
 and
    Content-Disposition: session;handling=optional
 is equivalent to
    content-disposition: Session;HANDLING=OPTIONAL
 The following two header fields are not equivalent:
    Warning: 370 devnull "Choose a bigger pipe"
    Warning: 370 devnull "CHOOSE A BIGGER PIPE"

7.3.2 Header Field Classification

 Some header fields only make sense in requests or responses.  These
 are called request header fields and response header fields,
 respectively.  If a header field appears in a message not matching
 its category (such as a request header field in a response), it MUST
 be ignored.  Section 20 defines the classification of each header
 field.

7.3.3 Compact Form

 SIP provides a mechanism to represent common header field names in an
 abbreviated form.  This may be useful when messages would otherwise
 become too large to be carried on the transport available to it
 (exceeding the maximum transmission unit (MTU) when using UDP, for
 example).  These compact forms are defined in Section 20.  A compact
 form MAY be substituted for the longer form of a header field name at
 any time without changing the semantics of the message.  A header

Rosenberg, et. al. Standards Track [Page 32] RFC 3261 SIP: Session Initiation Protocol June 2002

 field name MAY appear in both long and short forms within the same
 message.  Implementations MUST accept both the long and short forms
 of each header name.

7.4 Bodies

 Requests, including new requests defined in extensions to this
 specification, MAY contain message bodies unless otherwise noted.
 The interpretation of the body depends on the request method.
 For response messages, the request method and the response status
 code determine the type and interpretation of any message body.  All
 responses MAY include a body.

7.4.1 Message Body Type

 The Internet media type of the message body MUST be given by the
 Content-Type header field.  If the body has undergone any encoding
 such as compression, then this MUST be indicated by the Content-
 Encoding header field; otherwise, Content-Encoding MUST be omitted.
 If applicable, the character set of the message body is indicated as
 part of the Content-Type header-field value.
 The "multipart" MIME type defined in RFC 2046 [11] MAY be used within
 the body of the message.  Implementations that send requests
 containing multipart message bodies MUST send a session description
 as a non-multipart message body if the remote implementation requests
 this through an Accept header field that does not contain multipart.
 SIP messages MAY contain binary bodies or body parts. When no
 explicit charset parameter is provided by the sender, media subtypes
 of the "text" type are defined to have a default charset value of
 "UTF-8".

7.4.2 Message Body Length

 The body length in bytes is provided by the Content-Length header
 field.  Section 20.14 describes the necessary contents of this header
 field in detail.
 The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
 (Note: The chunked encoding modifies the body of a message in order
 to transfer it as a series of chunks, each with its own size
 indicator.)

Rosenberg, et. al. Standards Track [Page 33] RFC 3261 SIP: Session Initiation Protocol June 2002

7.5 Framing SIP Messages

 Unlike HTTP, SIP implementations can use UDP or other unreliable
 datagram protocols.  Each such datagram carries one request or
 response.  See Section 18 on constraints on usage of unreliable
 transports.
 Implementations processing SIP messages over stream-oriented
 transports MUST ignore any CRLF appearing before the start-line
 [H4.1].
    The Content-Length header field value is used to locate the end of
    each SIP message in a stream.  It will always be present when SIP
    messages are sent over stream-oriented transports.

8 General User Agent Behavior

 A user agent represents an end system.  It contains a user agent
 client (UAC), which generates requests, and a user agent server
 (UAS), which responds to them.  A UAC is capable of generating a
 request based on some external stimulus (the user clicking a button,
 or a signal on a PSTN line) and processing a response.  A UAS is
 capable of receiving a request and generating a response based on
 user input, external stimulus, the result of a program execution, or
 some other mechanism.
 When a UAC sends a request, the request passes through some number of
 proxy servers, which forward the request towards the UAS. When the
 UAS generates a response, the response is forwarded towards the UAC.
 UAC and UAS procedures depend strongly on two factors.  First, based
 on whether the request or response is inside or outside of a dialog,
 and second, based on the method of a request.  Dialogs are discussed
 thoroughly in Section 12; they represent a peer-to-peer relationship
 between user agents and are established by specific SIP methods, such
 as INVITE.
 In this section, we discuss the method-independent rules for UAC and
 UAS behavior when processing requests that are outside of a dialog.
 This includes, of course, the requests which themselves establish a
 dialog.
 Security procedures for requests and responses outside of a dialog
 are described in Section 26.  Specifically, mechanisms exist for the
 UAS and UAC to mutually authenticate.  A limited set of privacy
 features are also supported through encryption of bodies using
 S/MIME.

Rosenberg, et. al. Standards Track [Page 34] RFC 3261 SIP: Session Initiation Protocol June 2002

8.1 UAC Behavior

 This section covers UAC behavior outside of a dialog.

8.1.1 Generating the Request

 A valid SIP request formulated by a UAC MUST, at a minimum, contain
 the following header fields: To, From, CSeq, Call-ID, Max-Forwards,
 and Via; all of these header fields are mandatory in all SIP
 requests.  These six header fields are the fundamental building
 blocks of a SIP message, as they jointly provide for most of the
 critical message routing services including the addressing of
 messages, the routing of responses, limiting message propagation,
 ordering of messages, and the unique identification of transactions.
 These header fields are in addition to the mandatory request line,
 which contains the method, Request-URI, and SIP version.
 Examples of requests sent outside of a dialog include an INVITE to
 establish a session (Section 13) and an OPTIONS to query for
 capabilities (Section 11).

8.1.1.1 Request-URI

 The initial Request-URI of the message SHOULD be set to the value of
 the URI in the To field.  One notable exception is the REGISTER
 method; behavior for setting the Request-URI of REGISTER is given in
 Section 10.  It may also be undesirable for privacy reasons or
 convenience to set these fields to the same value (especially if the
 originating UA expects that the Request-URI will be changed during
 transit).
 In some special circumstances, the presence of a pre-existing route
 set can affect the Request-URI of the message.  A pre-existing route
 set is an ordered set of URIs that identify a chain of servers, to
 which a UAC will send outgoing requests that are outside of a dialog.
 Commonly, they are configured on the UA by a user or service provider
 manually, or through some other non-SIP mechanism.  When a provider
 wishes to configure a UA with an outbound proxy, it is RECOMMENDED
 that this be done by providing it with a pre-existing route set with
 a single URI, that of the outbound proxy.
 When a pre-existing route set is present, the procedures for
 populating the Request-URI and Route header field detailed in Section
 12.2.1.1 MUST be followed (even though there is no dialog), using the
 desired Request-URI as the remote target URI.

Rosenberg, et. al. Standards Track [Page 35] RFC 3261 SIP: Session Initiation Protocol June 2002

8.1.1.2 To

 The To header field first and foremost specifies the desired
 "logical" recipient of the request, or the address-of-record of the
 user or resource that is the target of this request.  This may or may
 not be the ultimate recipient of the request.  The To header field
 MAY contain a SIP or SIPS URI, but it may also make use of other URI
 schemes (the tel URL (RFC 2806 [9]), for example) when appropriate.
 All SIP implementations MUST support the SIP URI scheme.  Any
 implementation that supports TLS MUST support the SIPS URI scheme.
 The To header field allows for a display name.
 A UAC may learn how to populate the To header field for a particular
 request in a number of ways.  Usually the user will suggest the To
 header field through a human interface, perhaps inputting the URI
 manually or selecting it from some sort of address book.  Frequently,
 the user will not enter a complete URI, but rather a string of digits
 or letters (for example, "bob").  It is at the discretion of the UA
 to choose how to interpret this input.  Using the string to form the
 user part of a SIP URI implies that the UA wishes the name to be
 resolved in the domain to the right-hand side (RHS) of the at-sign in
 the SIP URI (for instance, sip:bob@example.com).  Using the string to
 form the user part of a SIPS URI implies that the UA wishes to
 communicate securely, and that the name is to be resolved in the
 domain to the RHS of the at-sign.  The RHS will frequently be the
 home domain of the requestor, which allows for the home domain to
 process the outgoing request.  This is useful for features like
 "speed dial" that require interpretation of the user part in the home
 domain.  The tel URL may be used when the UA does not wish to specify
 the domain that should interpret a telephone number that has been
 input by the user.  Rather, each domain through which the request
 passes would be given that opportunity.  As an example, a user in an
 airport might log in and send requests through an outbound proxy in
 the airport.  If they enter "411" (this is the phone number for local
 directory assistance in the United States), that needs to be
 interpreted and processed by the outbound proxy in the airport, not
 the user's home domain.  In this case, tel:411 would be the right
 choice.
 A request outside of a dialog MUST NOT contain a To tag; the tag in
 the To field of a request identifies the peer of the dialog.  Since
 no dialog is established, no tag is present.
 For further information on the To header field, see Section 20.39.
 The following is an example of a valid To header field:
    To: Carol <sip:carol@chicago.com>

Rosenberg, et. al. Standards Track [Page 36] RFC 3261 SIP: Session Initiation Protocol June 2002

8.1.1.3 From

 The From header field indicates the logical identity of the initiator
 of the request, possibly the user's address-of-record.  Like the To
 header field, it contains a URI and optionally a display name.  It is
 used by SIP elements to determine which processing rules to apply to
 a request (for example, automatic call rejection).  As such, it is
 very important that the From URI not contain IP addresses or the FQDN
 of the host on which the UA is running, since these are not logical
 names.
 The From header field allows for a display name.  A UAC SHOULD use
 the display name "Anonymous", along with a syntactically correct, but
 otherwise meaningless URI (like sip:thisis@anonymous.invalid), if the
 identity of the client is to remain hidden.
 Usually, the value that populates the From header field in requests
 generated by a particular UA is pre-provisioned by the user or by the
 administrators of the user's local domain.  If a particular UA is
 used by multiple users, it might have switchable profiles that
 include a URI corresponding to the identity of the profiled user.
 Recipients of requests can authenticate the originator of a request
 in order to ascertain that they are who their From header field
 claims they are (see Section 22 for more on authentication).
 The From field MUST contain a new "tag" parameter, chosen by the UAC.
 See Section 19.3 for details on choosing a tag.
 For further information on the From header field, see Section 20.20.
 Examples:
    From: "Bob" <sips:bob@biloxi.com> ;tag=a48s
    From: sip:+12125551212@phone2net.com;tag=887s
    From: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8

8.1.1.4 Call-ID

 The Call-ID header field acts as a unique identifier to group
 together a series of messages.  It MUST be the same for all requests
 and responses sent by either UA in a dialog.  It SHOULD be the same
 in each registration from a UA.
 In a new request created by a UAC outside of any dialog, the Call-ID
 header field MUST be selected by the UAC as a globally unique
 identifier over space and time unless overridden by method-specific
 behavior.  All SIP UAs must have a means to guarantee that the Call-
 ID header fields they produce will not be inadvertently generated by
 any other UA.  Note that when requests are retried after certain

Rosenberg, et. al. Standards Track [Page 37] RFC 3261 SIP: Session Initiation Protocol June 2002

 failure responses that solicit an amendment to a request (for
 example, a challenge for authentication), these retried requests are
 not considered new requests, and therefore do not need new Call-ID
 header fields; see Section 8.1.3.5.
 Use of cryptographically random identifiers (RFC 1750 [12]) in the
 generation of Call-IDs is RECOMMENDED.  Implementations MAY use the
 form "localid@host".  Call-IDs are case-sensitive and are simply
 compared byte-by-byte.
    Using cryptographically random identifiers provides some
    protection against session hijacking and reduces the likelihood of
    unintentional Call-ID collisions.
 No provisioning or human interface is required for the selection of
 the Call-ID header field value for a request.
 For further information on the Call-ID header field, see Section
 20.8.
 Example:
    Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com

8.1.1.5 CSeq

 The CSeq header field serves as a way to identify and order
 transactions.  It consists of a sequence number and a method.  The
 method MUST match that of the request.  For non-REGISTER requests
 outside of a dialog, the sequence number value is arbitrary.  The
 sequence number value MUST be expressible as a 32-bit unsigned
 integer and MUST be less than 2**31.  As long as it follows the above
 guidelines, a client may use any mechanism it would like to select
 CSeq header field values.
 Section 12.2.1.1 discusses construction of the CSeq for requests
 within a dialog.
 Example:
    CSeq: 4711 INVITE

Rosenberg, et. al. Standards Track [Page 38] RFC 3261 SIP: Session Initiation Protocol June 2002

8.1.1.6 Max-Forwards

 The Max-Forwards header field serves to limit the number of hops a
 request can transit on the way to its destination.  It consists of an
 integer that is decremented by one at each hop.  If the Max-Forwards
 value reaches 0 before the request reaches its destination, it will
 be rejected with a 483(Too Many Hops) error response.
 A UAC MUST insert a Max-Forwards header field into each request it
 originates with a value that SHOULD be 70.  This number was chosen to
 be sufficiently large to guarantee that a request would not be
 dropped in any SIP network when there were no loops, but not so large
 as to consume proxy resources when a loop does occur.  Lower values
 should be used with caution and only in networks where topologies are
 known by the UA.

8.1.1.7 Via

 The Via header field indicates the transport used for the transaction
 and identifies the location where the response is to be sent.  A Via
 header field value is added only after the transport that will be
 used to reach the next hop has been selected (which may involve the
 usage of the procedures in [4]).
 When the UAC creates a request, it MUST insert a Via into that
 request.  The protocol name and protocol version in the header field
 MUST be SIP and 2.0, respectively.  The Via header field value MUST
 contain a branch parameter.  This parameter is used to identify the
 transaction created by that request.  This parameter is used by both
 the client and the server.
 The branch parameter value MUST be unique across space and time for
 all requests sent by the UA.  The exceptions to this rule are CANCEL
 and ACK for non-2xx responses.  As discussed below, a CANCEL request
 will have the same value of the branch parameter as the request it
 cancels.  As discussed in Section 17.1.1.3, an ACK for a non-2xx
 response will also have the same branch ID as the INVITE whose
 response it acknowledges.
    The uniqueness property of the branch ID parameter, to facilitate
    its use as a transaction ID, was not part of RFC 2543.
 The branch ID inserted by an element compliant with this
 specification MUST always begin with the characters "z9hG4bK".  These
 7 characters are used as a magic cookie (7 is deemed sufficient to
 ensure that an older RFC 2543 implementation would not pick such a
 value), so that servers receiving the request can determine that the
 branch ID was constructed in the fashion described by this

Rosenberg, et. al. Standards Track [Page 39] RFC 3261 SIP: Session Initiation Protocol June 2002

 specification (that is, globally unique).  Beyond this requirement,
 the precise format of the branch token is implementation-defined.
 The Via header maddr, ttl, and sent-by components will be set when
 the request is processed by the transport layer (Section 18).
 Via processing for proxies is described in Section 16.6 Item 8 and
 Section 16.7 Item 3.

8.1.1.8 Contact

 The Contact header field provides a SIP or SIPS URI that can be used
 to contact that specific instance of the UA for subsequent requests.
 The Contact header field MUST be present and contain exactly one SIP
 or SIPS URI in any request that can result in the establishment of a
 dialog.  For the methods defined in this specification, that includes
 only the INVITE request.  For these requests, the scope of the
 Contact is global.  That is, the Contact header field value contains
 the URI at which the UA would like to receive requests, and this URI
 MUST be valid even if used in subsequent requests outside of any
 dialogs.
 If the Request-URI or top Route header field value contains a SIPS
 URI, the Contact header field MUST contain a SIPS URI as well.
 For further information on the Contact header field, see Section
 20.10.

8.1.1.9 Supported and Require

 If the UAC supports extensions to SIP that can be applied by the
 server to the response, the UAC SHOULD include a Supported header
 field in the request listing the option tags (Section 19.2) for those
 extensions.
 The option tags listed MUST only refer to extensions defined in
 standards-track RFCs.  This is to prevent servers from insisting that
 clients implement non-standard, vendor-defined features in order to
 receive service.  Extensions defined by experimental and
 informational RFCs are explicitly excluded from usage with the
 Supported header field in a request, since they too are often used to
 document vendor-defined extensions.
 If the UAC wishes to insist that a UAS understand an extension that
 the UAC will apply to the request in order to process the request, it
 MUST insert a Require header field into the request listing the
 option tag for that extension.  If the UAC wishes to apply an
 extension to the request and insist that any proxies that are

Rosenberg, et. al. Standards Track [Page 40] RFC 3261 SIP: Session Initiation Protocol June 2002

 traversed understand that extension, it MUST insert a Proxy-Require
 header field into the request listing the option tag for that
 extension.
 As with the Supported header field, the option tags in the Require
 and Proxy-Require header fields MUST only refer to extensions defined
 in standards-track RFCs.

8.1.1.10 Additional Message Components

 After a new request has been created, and the header fields described
 above have been properly constructed, any additional optional header
 fields are added, as are any header fields specific to the method.
 SIP requests MAY contain a MIME-encoded message-body.  Regardless of
 the type of body that a request contains, certain header fields must
 be formulated to characterize the contents of the body.  For further
 information on these header fields, see Sections 20.11 through 20.15.

8.1.2 Sending the Request

 The destination for the request is then computed.  Unless there is
 local policy specifying otherwise, the destination MUST be determined
 by applying the DNS procedures described in [4] as follows.  If the
 first element in the route set indicated a strict router (resulting
 in forming the request as described in Section 12.2.1.1), the
 procedures MUST be applied to the Request-URI of the request.
 Otherwise, the procedures are applied to the first Route header field
 value in the request (if one exists), or to the request's Request-URI
 if there is no Route header field present.  These procedures yield an
 ordered set of address, port, and transports to attempt.  Independent
 of which URI is used as input to the procedures of [4], if the
 Request-URI specifies a SIPS resource, the UAC MUST follow the
 procedures of [4] as if the input URI were a SIPS URI.
 Local policy MAY specify an alternate set of destinations to attempt.
 If the Request-URI contains a SIPS URI, any alternate destinations
 MUST be contacted with TLS.  Beyond that, there are no restrictions
 on the alternate destinations if the request contains no Route header
 field.  This provides a simple alternative to a pre-existing route
 set as a way to specify an outbound proxy.  However, that approach
 for configuring an outbound proxy is NOT RECOMMENDED; a pre-existing
 route set with a single URI SHOULD be used instead.  If the request
 contains a Route header field, the request SHOULD be sent to the
 locations derived from its topmost value, but MAY be sent to any
 server that the UA is certain will honor the Route and Request-URI
 policies specified in this document (as opposed to those in RFC
 2543).  In particular, a UAC configured with an outbound proxy SHOULD

Rosenberg, et. al. Standards Track [Page 41] RFC 3261 SIP: Session Initiation Protocol June 2002

 attempt to send the request to the location indicated in the first
 Route header field value instead of adopting the policy of sending
 all messages to the outbound proxy.
    This ensures that outbound proxies that do not add Record-Route
    header field values will drop out of the path of subsequent
    requests.  It allows endpoints that cannot resolve the first Route
    URI to delegate that task to an outbound proxy.
 The UAC SHOULD follow the procedures defined in [4] for stateful
 elements, trying each address until a server is contacted.  Each try
 constitutes a new transaction, and therefore each carries a different
 topmost Via header field value with a new branch parameter.
 Furthermore, the transport value in the Via header field is set to
 whatever transport was determined for the target server.

8.1.3 Processing Responses

 Responses are first processed by the transport layer and then passed
 up to the transaction layer.  The transaction layer performs its
 processing and then passes the response up to the TU.  The majority
 of response processing in the TU is method specific.  However, there
 are some general behaviors independent of the method.

8.1.3.1 Transaction Layer Errors

 In some cases, the response returned by the transaction layer will
 not be a SIP message, but rather a transaction layer error.  When a
 timeout error is received from the transaction layer, it MUST be
 treated as if a 408 (Request Timeout) status code has been received.
 If a fatal transport error is reported by the transport layer
 (generally, due to fatal ICMP errors in UDP or connection failures in
 TCP), the condition MUST be treated as a 503 (Service Unavailable)
 status code.

8.1.3.2 Unrecognized Responses

 A UAC MUST treat any final response it does not recognize as being
 equivalent to the x00 response code of that class, and MUST be able
 to process the x00 response code for all classes.  For example, if a
 UAC receives an unrecognized response code of 431, it can safely
 assume that there was something wrong with its request and treat the
 response as if it had received a 400 (Bad Request) response code.  A
 UAC MUST treat any provisional response different than 100 that it
 does not recognize as 183 (Session Progress).  A UAC MUST be able to
 process 100 and 183 responses.

Rosenberg, et. al. Standards Track [Page 42] RFC 3261 SIP: Session Initiation Protocol June 2002

8.1.3.3 Vias

 If more than one Via header field value is present in a response, the
 UAC SHOULD discard the message.
    The presence of additional Via header field values that precede
    the originator of the request suggests that the message was
    misrouted or possibly corrupted.

8.1.3.4 Processing 3xx Responses

 Upon receipt of a redirection response (for example, a 301 response
 status code), clients SHOULD use the URI(s) in the Contact header
 field to formulate one or more new requests based on the redirected
 request.  This process is similar to that of a proxy recursing on a
 3xx class response as detailed in Sections 16.5 and 16.6.  A client
 starts with an initial target set containing exactly one URI, the
 Request-URI of the original request.  If a client wishes to formulate
 new requests based on a 3xx class response to that request, it places
 the URIs to try into the target set.  Subject to the restrictions in
 this specification, a client can choose which Contact URIs it places
 into the target set.  As with proxy recursion, a client processing
 3xx class responses MUST NOT add any given URI to the target set more
 than once.  If the original request had a SIPS URI in the Request-
 URI, the client MAY choose to recurse to a non-SIPS URI, but SHOULD
 inform the user of the redirection to an insecure URI.
    Any new request may receive 3xx responses themselves containing
    the original URI as a contact.  Two locations can be configured to
    redirect to each other.  Placing any given URI in the target set
    only once prevents infinite redirection loops.
 As the target set grows, the client MAY generate new requests to the
 URIs in any order.  A common mechanism is to order the set by the "q"
 parameter value from the Contact header field value.  Requests to the
 URIs MAY be generated serially or in parallel.  One approach is to
 process groups of decreasing q-values serially and process the URIs
 in each q-value group in parallel.  Another is to perform only serial
 processing in decreasing q-value order, arbitrarily choosing between
 contacts of equal q-value.
 If contacting an address in the list results in a failure, as defined
 in the next paragraph, the element moves to the next address in the
 list, until the list is exhausted.  If the list is exhausted, then
 the request has failed.

Rosenberg, et. al. Standards Track [Page 43] RFC 3261 SIP: Session Initiation Protocol June 2002

 Failures SHOULD be detected through failure response codes (codes
 greater than 399); for network errors the client transaction will
 report any transport layer failures to the transaction user.  Note
 that some response codes (detailed in 8.1.3.5) indicate that the
 request can be retried; requests that are reattempted should not be
 considered failures.
 When a failure for a particular contact address is received, the
 client SHOULD try the next contact address.  This will involve
 creating a new client transaction to deliver a new request.
 In order to create a request based on a contact address in a 3xx
 response, a UAC MUST copy the entire URI from the target set into the
 Request-URI, except for the "method-param" and "header" URI
 parameters (see Section 19.1.1 for a definition of these parameters).
 It uses the "header" parameters to create header field values for the
 new request, overwriting header field values associated with the
 redirected request in accordance with the guidelines in Section
 19.1.5.
 Note that in some instances, header fields that have been
 communicated in the contact address may instead append to existing
 request header fields in the original redirected request.  As a
 general rule, if the header field can accept a comma-separated list
 of values, then the new header field value MAY be appended to any
 existing values in the original redirected request.  If the header
 field does not accept multiple values, the value in the original
 redirected request MAY be overwritten by the header field value
 communicated in the contact address.  For example, if a contact
 address is returned with the following value:
    sip:user@host?Subject=foo&Call-Info=<http://www.foo.com>
 Then any Subject header field in the original redirected request is
 overwritten, but the HTTP URL is merely appended to any existing
 Call-Info header field values.
 It is RECOMMENDED that the UAC reuse the same To, From, and Call-ID
 used in the original redirected request, but the UAC MAY also choose
 to update the Call-ID header field value for new requests, for
 example.
 Finally, once the new request has been constructed, it is sent using
 a new client transaction, and therefore MUST have a new branch ID in
 the top Via field as discussed in Section 8.1.1.7.

Rosenberg, et. al. Standards Track [Page 44] RFC 3261 SIP: Session Initiation Protocol June 2002

 In all other respects, requests sent upon receipt of a redirect
 response SHOULD re-use the header fields and bodies of the original
 request.
 In some instances, Contact header field values may be cached at UAC
 temporarily or permanently depending on the status code received and
 the presence of an expiration interval; see Sections 21.3.2 and
 21.3.3.

8.1.3.5 Processing 4xx Responses

 Certain 4xx response codes require specific UA processing,
 independent of the method.
 If a 401 (Unauthorized) or 407 (Proxy Authentication Required)
 response is received, the UAC SHOULD follow the authorization
 procedures of Section 22.2 and Section 22.3 to retry the request with
 credentials.
 If a 413 (Request Entity Too Large) response is received (Section
 21.4.11), the request contained a body that was longer than the UAS
 was willing to accept.  If possible, the UAC SHOULD retry the
 request, either omitting the body or using one of a smaller length.
 If a 415 (Unsupported Media Type) response is received (Section
 21.4.13), the request contained media types not supported by the UAS.
 The UAC SHOULD retry sending the request, this time only using
 content with types listed in the Accept header field in the response,
 with encodings listed in the Accept-Encoding header field in the
 response, and with languages listed in the Accept-Language in the
 response.
 If a 416 (Unsupported URI Scheme) response is received (Section
 21.4.14), the Request-URI used a URI scheme not supported by the
 server.  The client SHOULD retry the request, this time, using a SIP
 URI.
 If a 420 (Bad Extension) response is received (Section 21.4.15), the
 request contained a Require or Proxy-Require header field listing an
 option-tag for a feature not supported by a proxy or UAS.  The UAC
 SHOULD retry the request, this time omitting any extensions listed in
 the Unsupported header field in the response.
 In all of the above cases, the request is retried by creating a new
 request with the appropriate modifications.  This new request
 constitutes a new transaction and SHOULD have the same value of the
 Call-ID, To, and From of the previous request, but the CSeq should
 contain a new sequence number that is one higher than the previous.

Rosenberg, et. al. Standards Track [Page 45] RFC 3261 SIP: Session Initiation Protocol June 2002

 With other 4xx responses, including those yet to be defined, a retry
 may or may not be possible depending on the method and the use case.

8.2 UAS Behavior

 When a request outside of a dialog is processed by a UAS, there is a
 set of processing rules that are followed, independent of the method.
 Section 12 gives guidance on how a UAS can tell whether a request is
 inside or outside of a dialog.
 Note that request processing is atomic.  If a request is accepted,
 all state changes associated with it MUST be performed.  If it is
 rejected, all state changes MUST NOT be performed.
 UASs SHOULD process the requests in the order of the steps that
 follow in this section (that is, starting with authentication, then
 inspecting the method, the header fields, and so on throughout the
 remainder of this section).

8.2.1 Method Inspection

 Once a request is authenticated (or authentication is skipped), the
 UAS MUST inspect the method of the request.  If the UAS recognizes
 but does not support the method of a request, it MUST generate a 405
 (Method Not Allowed) response.  Procedures for generating responses
 are described in Section 8.2.6.  The UAS MUST also add an Allow
 header field to the 405 (Method Not Allowed) response.  The Allow
 header field MUST list the set of methods supported by the UAS
 generating the message.  The Allow header field is presented in
 Section 20.5.
 If the method is one supported by the server, processing continues.

8.2.2 Header Inspection

 If a UAS does not understand a header field in a request (that is,
 the header field is not defined in this specification or in any
 supported extension), the server MUST ignore that header field and
 continue processing the message.  A UAS SHOULD ignore any malformed
 header fields that are not necessary for processing requests.

8.2.2.1 To and Request-URI

 The To header field identifies the original recipient of the request
 designated by the user identified in the From field.  The original
 recipient may or may not be the UAS processing the request, due to
 call forwarding or other proxy operations.  A UAS MAY apply any
 policy it wishes to determine whether to accept requests when the To

Rosenberg, et. al. Standards Track [Page 46] RFC 3261 SIP: Session Initiation Protocol June 2002

 header field is not the identity of the UAS.  However, it is
 RECOMMENDED that a UAS accept requests even if they do not recognize
 the URI scheme (for example, a tel: URI) in the To header field, or
 if the To header field does not address a known or current user of
 this UAS.  If, on the other hand, the UAS decides to reject the
 request, it SHOULD generate a response with a 403 (Forbidden) status
 code and pass it to the server transaction for transmission.
 However, the Request-URI identifies the UAS that is to process the
 request.  If the Request-URI uses a scheme not supported by the UAS,
 it SHOULD reject the request with a 416 (Unsupported URI Scheme)
 response.  If the Request-URI does not identify an address that the
 UAS is willing to accept requests for, it SHOULD reject the request
 with a 404 (Not Found) response.  Typically, a UA that uses the
 REGISTER method to bind its address-of-record to a specific contact
 address will see requests whose Request-URI equals that contact
 address.  Other potential sources of received Request-URIs include
 the Contact header fields of requests and responses sent by the UA
 that establish or refresh dialogs.

8.2.2.2 Merged Requests

 If the request has no tag in the To header field, the UAS core MUST
 check the request against ongoing transactions.  If the From tag,
 Call-ID, and CSeq exactly match those associated with an ongoing
 transaction, but the request does not match that transaction (based
 on the matching rules in Section 17.2.3), the UAS core SHOULD
 generate a 482 (Loop Detected) response and pass it to the server
 transaction.
    The same request has arrived at the UAS more than once, following
    different paths, most likely due to forking.  The UAS processes
    the first such request received and responds with a 482 (Loop
    Detected) to the rest of them.

8.2.2.3 Require

 Assuming the UAS decides that it is the proper element to process the
 request, it examines the Require header field, if present.
 The Require header field is used by a UAC to tell a UAS about SIP
 extensions that the UAC expects the UAS to support in order to
 process the request properly.  Its format is described in Section
 20.32.  If a UAS does not understand an option-tag listed in a
 Require header field, it MUST respond by generating a response with
 status code 420 (Bad Extension).  The UAS MUST add an Unsupported
 header field, and list in it those options it does not understand
 amongst those in the Require header field of the request.

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 Note that Require and Proxy-Require MUST NOT be used in a SIP CANCEL
 request, or in an ACK request sent for a non-2xx response.  These
 header fields MUST be ignored if they are present in these requests.
 An ACK request for a 2xx response MUST contain only those Require and
 Proxy-Require values that were present in the initial request.
 Example:
    UAC->UAS:   INVITE sip:watson@bell-telephone.com SIP/2.0
                Require: 100rel
    UAS->UAC:   SIP/2.0 420 Bad Extension
                Unsupported: 100rel
    This behavior ensures 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
    example 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 ambiguity
    when the client requires features that the server does not
    understand.  Some features, such as call handling fields, are only
    of interest to end systems.

8.2.3 Content Processing

 Assuming the UAS understands any extensions required by the client,
 the UAS examines the body of the message, and the header fields that
 describe it.  If there are any bodies whose type (indicated by the
 Content-Type), language (indicated by the Content-Language) or
 encoding (indicated by the Content-Encoding) are not understood, and
 that body part is not optional (as indicated by the Content-
 Disposition header field), the UAS MUST reject the request with a 415
 (Unsupported Media Type) response.  The response MUST contain an
 Accept header field listing the types of all bodies it understands,
 in the event the request contained bodies of types not supported by
 the UAS.  If the request contained content encodings not understood
 by the UAS, the response MUST contain an Accept-Encoding header field
 listing the encodings understood by the UAS.  If the request
 contained content with languages not understood by the UAS, the
 response MUST contain an Accept-Language header field indicating the
 languages understood by the UAS.  Beyond these checks, body handling
 depends on the method and type.  For further information on the
 processing of content-specific header fields, see Section 7.4 as well
 as Section 20.11 through 20.15.

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8.2.4 Applying Extensions

 A UAS that wishes to apply some extension when generating the
 response MUST NOT do so unless support for that extension is
 indicated in the Supported header field in the request.  If the
 desired extension is not supported, the server SHOULD rely only on
 baseline SIP and any other extensions supported by the client.  In
 rare circumstances, where the server cannot process the request
 without the extension, the server MAY send a 421 (Extension Required)
 response.  This response indicates that the proper response cannot be
 generated without support of a specific extension.  The needed
 extension(s) MUST be included in a Require header field in the
 response.  This behavior is NOT RECOMMENDED, as it will generally
 break interoperability.
 Any extensions applied to a non-421 response MUST be listed in a
 Require header field included in the response.  Of course, the server
 MUST NOT apply extensions not listed in the Supported header field in
 the request.  As a result of this, the Require header field in a
 response will only ever contain option tags defined in standards-
 track RFCs.

8.2.5 Processing the Request

 Assuming all of the checks in the previous subsections are passed,
 the UAS processing becomes method-specific.  Section 10 covers the
 REGISTER request, Section 11 covers the OPTIONS request, Section 13
 covers the INVITE request, and Section 15 covers the BYE request.

8.2.6 Generating the Response

 When a UAS wishes to construct a response to a request, it follows
 the general procedures detailed in the following subsections.
 Additional behaviors specific to the response code in question, which
 are not detailed in this section, may also be required.
 Once all procedures associated with the creation of a response have
 been completed, the UAS hands the response back to the server
 transaction from which it received the request.

8.2.6.1 Sending a Provisional Response

 One largely non-method-specific guideline for the generation of
 responses is that UASs SHOULD NOT issue a provisional response for a
 non-INVITE request.  Rather, UASs SHOULD generate a final response to
 a non-INVITE request as soon as possible.

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 When a 100 (Trying) response is generated, any Timestamp header field
 present in the request MUST be copied into this 100 (Trying)
 response.  If there is a delay in generating the response, the UAS
 SHOULD add a delay value into the Timestamp value in the response.
 This value MUST contain the difference between the time of sending of
 the response and receipt of the request, measured in seconds.

8.2.6.2 Headers and Tags

 The From field of the response MUST equal the From header field of
 the request.  The Call-ID header field of the response MUST equal the
 Call-ID header field of the request.  The CSeq header field of the
 response MUST equal the CSeq field of the request.  The Via header
 field values in the response MUST equal the Via header field values
 in the request and MUST maintain the same ordering.
 If a request contained a To tag in the request, the To header field
 in the response MUST equal that of the request.  However, if the To
 header field in the request did not contain a tag, the URI in the To
 header field in the response MUST equal the URI in the To header
 field; additionally, the UAS MUST add a tag to the To header field in
 the response (with the exception of the 100 (Trying) response, in
 which a tag MAY be present).  This serves to identify the UAS that is
 responding, possibly resulting in a component of a dialog ID.  The
 same tag MUST be used for all responses to that request, both final
 and provisional (again excepting the 100 (Trying)).  Procedures for
 the generation of tags are defined in Section 19.3.

8.2.7 Stateless UAS Behavior

 A stateless UAS is a UAS that does not maintain transaction state.
 It replies to requests normally, but discards any state that would
 ordinarily be retained by a UAS after a response has been sent.  If a
 stateless UAS receives a retransmission of a request, it regenerates
 the response and resends it, just as if it were replying to the first
 instance of the request. A UAS cannot be stateless unless the request
 processing for that method would always result in the same response
 if the requests are identical. This rules out stateless registrars,
 for example.  Stateless UASs do not use a transaction layer; they
 receive requests directly from the transport layer and send responses
 directly to the transport layer.
 The stateless UAS role is needed primarily to handle unauthenticated
 requests for which a challenge response is issued.  If
 unauthenticated requests were handled statefully, then malicious
 floods of unauthenticated requests could create massive amounts of

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 transaction state that might slow or completely halt call processing
 in a UAS, effectively creating a denial of service condition; for
 more information see Section 26.1.5.
 The most important behaviors of a stateless UAS are the following:
    o  A stateless UAS MUST NOT send provisional (1xx) responses.
    o  A stateless UAS MUST NOT retransmit responses.
    o  A stateless UAS MUST ignore ACK requests.
    o  A stateless UAS MUST ignore CANCEL requests.
    o  To header tags MUST be generated for responses in a stateless
       manner - in a manner that will generate the same tag for the
       same request consistently.  For information on tag construction
       see Section 19.3.
 In all other respects, a stateless UAS behaves in the same manner as
 a stateful UAS.  A UAS can operate in either a stateful or stateless
 mode for each new request.

8.3 Redirect Servers

 In some architectures it may be desirable to reduce the processing
 load on proxy servers that are responsible for routing requests, and
 improve signaling path robustness, by relying on redirection.
 Redirection allows servers to push routing information for a request
 back in a response to the client, thereby taking themselves out of
 the loop of further messaging for this transaction while still aiding
 in locating the target of the request.  When the originator of the
 request receives the redirection, it will send a new request based on
 the URI(s) it has received.  By propagating URIs from the core of the
 network to its edges, redirection allows for considerable network
 scalability.
 A redirect server is logically constituted of a server transaction
 layer and a transaction user that has access to a location service of
 some kind (see Section 10 for more on registrars and location
 services).  This location service is effectively a database
 containing mappings between a single URI and a set of one or more
 alternative locations at which the target of that URI can be found.
 A redirect server does not issue any SIP requests of its own.  After
 receiving a request other than CANCEL, the server either refuses the
 request or gathers the list of alternative locations from the

Rosenberg, et. al. Standards Track [Page 51] RFC 3261 SIP: Session Initiation Protocol June 2002

 location service and returns a final response of class 3xx.  For
 well-formed CANCEL requests, it SHOULD return a 2xx response.  This
 response ends the SIP transaction.  The redirect server maintains
 transaction state for an entire SIP transaction.  It is the
 responsibility of clients to detect forwarding loops between redirect
 servers.
 When a redirect server returns a 3xx response to a request, it
 populates the list of (one or more) alternative locations into the
 Contact header field.  An "expires" parameter to the Contact header
 field values may also be supplied to indicate the lifetime of the
 Contact data.
 The Contact header field contains URIs giving the new locations or
 user names to try, or may simply specify additional transport
 parameters.  A 301 (Moved Permanently) or 302 (Moved Temporarily)
 response may also give the same location and username that was
 targeted by the initial request but specify additional transport
 parameters such as a different server or multicast address to try, or
 a change of SIP transport from UDP to TCP or vice versa.
 However, redirect servers MUST NOT redirect a request to a URI equal
 to the one in the Request-URI; instead, provided that the URI does
 not point to itself, the server MAY proxy the request to the
 destination URI, or MAY reject it with a 404.
    If a client is using an outbound proxy, and that proxy actually
    redirects requests, a potential arises for infinite redirection
    loops.
 Note that a Contact header field value MAY also refer to a different
 resource than the one originally called.  For example, a SIP call
 connected to PSTN gateway may need to deliver a special informational
 announcement such as "The number you have dialed has been changed."
 A Contact response header field can contain any suitable URI
 indicating where the called party can be reached, not limited to SIP
 URIs.  For example, it could contain URIs for phones, fax, or irc (if
 they were defined) or a mailto:  (RFC 2368 [32]) URL.  Section 26.4.4
 discusses implications and limitations of redirecting a SIPS URI to a
 non-SIPS URI.
 The "expires" parameter of a Contact header field value indicates how
 long the URI is valid.  The value of the parameter is a number
 indicating seconds.  If this parameter is not provided, the value of
 the Expires header field determines how long the URI is valid.
 Malformed values SHOULD be treated as equivalent to 3600.

Rosenberg, et. al. Standards Track [Page 52] RFC 3261 SIP: Session Initiation Protocol June 2002

    This provides a modest level of backwards compatibility with RFC
    2543, which allowed absolute times in this header field.  If an
    absolute time is received, it will be treated as malformed, and
    then default to 3600.
 Redirect servers MUST ignore features that are not understood
 (including unrecognized header fields, any unknown option tags in
 Require, or even method names) and proceed with the redirection of
 the request in question.

9 Canceling a Request

 The previous section has discussed general UA behavior for generating
 requests and processing responses for requests of all methods.  In
 this section, we discuss a general purpose method, called CANCEL.
 The CANCEL request, as the name implies, is used to cancel a previous
 request sent by a client.  Specifically, it asks the UAS to cease
 processing the request and to generate an error response to that
 request.  CANCEL has no effect on a request to which a UAS has
 already given a final response.  Because of this, it is most useful
 to CANCEL requests to which it can take a server long time to
 respond.  For this reason, CANCEL is best for INVITE requests, which
 can take a long time to generate a response.  In that usage, a UAS
 that receives a CANCEL request for an INVITE, but has not yet sent a
 final response, would "stop ringing", and then respond to the INVITE
 with a specific error response (a 487).
 CANCEL requests can be constructed and sent by both proxies and user
 agent clients.  Section 15 discusses under what conditions a UAC
 would CANCEL an INVITE request, and Section 16.10 discusses proxy
 usage of CANCEL.
 A stateful proxy responds to a CANCEL, rather than simply forwarding
 a response it would receive from a downstream element.  For that
 reason, CANCEL is referred to as a "hop-by-hop" request, since it is
 responded to at each stateful proxy hop.

9.1 Client Behavior

 A CANCEL request SHOULD NOT be sent to cancel a request other than
 INVITE.
    Since requests other than INVITE are responded to immediately,
    sending a CANCEL for a non-INVITE request would always create a
    race condition.

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 The following procedures are used to construct a CANCEL request.  The
 Request-URI, Call-ID, To, the numeric part of CSeq, and From header
 fields in the CANCEL request MUST be identical to those in the
 request being cancelled, including tags.  A CANCEL constructed by a
 client MUST have only a single Via header field value matching the
 top Via value in the request being cancelled.  Using the same values
 for these header fields allows the CANCEL to be matched with the
 request it cancels (Section 9.2 indicates how such matching occurs).
 However, the method part of the CSeq header field MUST have a value
 of CANCEL.  This allows it to be identified and processed as a
 transaction in its own right (See Section 17).
 If the request being cancelled contains a Route header field, the
 CANCEL request MUST include that Route header field's values.
    This is needed so that stateless proxies are able to route CANCEL
    requests properly.
 The CANCEL request MUST NOT contain any Require or Proxy-Require
 header fields.
 Once the CANCEL is constructed, the client SHOULD check whether it
 has received any response (provisional or final) for the request
 being cancelled (herein referred to as the "original request").
 If no provisional response has been received, the CANCEL request MUST
 NOT be sent; rather, the client MUST wait for the arrival of a
 provisional response before sending the request.  If the original
 request has generated a final response, the CANCEL SHOULD NOT be
 sent, as it is an effective no-op, since CANCEL has no effect on
 requests that have already generated a final response.  When the
 client decides to send the CANCEL, it creates a client transaction
 for the CANCEL and passes it the CANCEL request along with the
 destination address, port, and transport.  The destination address,
 port, and transport for the CANCEL MUST be identical to those used to
 send the original request.
    If it was allowed to send the CANCEL before receiving a response
    for the previous request, the server could receive the CANCEL
    before the original request.
 Note that both the transaction corresponding to the original request
 and the CANCEL transaction will complete independently.  However, a
 UAC canceling a request cannot rely on receiving a 487 (Request
 Terminated) response for the original request, as an RFC 2543-
 compliant UAS will not generate such a response.  If there is no
 final response for the original request in 64*T1 seconds (T1 is

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 defined in Section 17.1.1.1), the client SHOULD then consider the
 original transaction cancelled and SHOULD destroy the client
 transaction handling the original request.

9.2 Server Behavior

 The CANCEL method requests that the TU at the server side cancel a
 pending transaction.  The TU determines the transaction to be
 cancelled by taking the CANCEL request, and then assuming that the
 request method is anything but CANCEL or ACK and applying the
 transaction matching procedures of Section 17.2.3.  The matching
 transaction is the one to be cancelled.
 The processing of a CANCEL request at a server depends on the type of
 server.  A stateless proxy will forward it, a stateful proxy might
 respond to it and generate some CANCEL requests of its own, and a UAS
 will respond to it.  See Section 16.10 for proxy treatment of CANCEL.
 A UAS first processes the CANCEL request according to the general UAS
 processing described in Section 8.2.  However, since CANCEL requests
 are hop-by-hop and cannot be resubmitted, they cannot be challenged
 by the server in order to get proper credentials in an Authorization
 header field.  Note also that CANCEL requests do not contain a
 Require header field.
 If the UAS did not find a matching transaction for the CANCEL
 according to the procedure above, it SHOULD respond to the CANCEL
 with a 481 (Call Leg/Transaction Does Not Exist).  If the transaction
 for the original request still exists, the behavior of the UAS on
 receiving a CANCEL request depends on whether it has already sent a
 final response for the original request.  If it has, the CANCEL
 request has no effect on the processing of the original request, no
 effect on any session state, and no effect on the responses generated
 for the original request.  If the UAS has not issued a final response
 for the original request, its behavior depends on the method of the
 original request.  If the original request was an INVITE, the UAS
 SHOULD immediately respond to the INVITE with a 487 (Request
 Terminated).  A CANCEL request has no impact on the processing of
 transactions with any other method defined in this specification.
 Regardless of the method of the original request, as long as the
 CANCEL matched an existing transaction, the UAS answers the CANCEL
 request itself with a 200 (OK) response.  This response is
 constructed following the procedures described in Section 8.2.6
 noting that the To tag of the response to the CANCEL and the To tag
 in the response to the original request SHOULD be the same.  The
 response to CANCEL is passed to the server transaction for
 transmission.

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10 Registrations

10.1 Overview

 SIP offers a discovery capability.  If a user wants to initiate a
 session with another user, SIP must discover the current host(s) at
 which the destination user is reachable.  This discovery process is
 frequently accomplished by SIP network elements such as proxy servers
 and redirect servers which are responsible for receiving a request,
 determining where to send it based on knowledge of the location of
 the user, and then sending it there.  To do this, SIP network
 elements consult an abstract service known as a location service,
 which provides address bindings for a particular domain.  These
 address bindings map an incoming SIP or SIPS URI, sip:bob@biloxi.com,
 for example, to one or more URIs that are somehow "closer" to the
 desired user, sip:bob@engineering.biloxi.com, for example.
 Ultimately, a proxy will consult a location service that maps a
 received URI to the user agent(s) at which the desired recipient is
 currently residing.
 Registration creates bindings in a location service for a particular
 domain that associates an address-of-record URI with one or more
 contact addresses.  Thus, when a proxy for that domain receives a
 request whose Request-URI matches the address-of-record, the proxy
 will forward the request to the contact addresses registered to that
 address-of-record.  Generally, it only makes sense to register an
 address-of-record at a domain's location service when requests for
 that address-of-record would be routed to that domain.  In most
 cases, this means that the domain of the registration will need to
 match the domain in the URI of the address-of-record.
 There are many ways by which the contents of the location service can
 be established.  One way is administratively.  In the above example,
 Bob is known to be a member of the engineering department through
 access to a corporate database.  However, SIP provides a mechanism
 for a UA to create a binding explicitly.  This mechanism is known as
 registration.
 Registration entails sending a REGISTER request to a special type of
 UAS known as a registrar.  A registrar acts as the front end to the
 location service for a domain, reading and writing mappings based on
 the contents of REGISTER requests.  This location service is then
 typically consulted by a proxy server that is responsible for routing
 requests for that domain.
 An illustration of the overall registration process is given in
 Figure 2.  Note that the registrar and proxy server are logical roles
 that can be played by a single device in a network; for purposes of

Rosenberg, et. al. Standards Track [Page 56] RFC 3261 SIP: Session Initiation Protocol June 2002

 clarity the two are separated in this illustration.  Also note that
 UAs may send requests through a proxy server in order to reach a
 registrar if the two are separate elements.
 SIP does not mandate a particular mechanism for implementing the
 location service.  The only requirement is that a registrar for some
 domain MUST be able to read and write data to the location service,
 and a proxy or a redirect server for that domain MUST be capable of
 reading that same data.  A registrar MAY be co-located with a
 particular SIP proxy server for the same domain.

10.2 Constructing the REGISTER Request

 REGISTER requests add, remove, and query bindings.  A REGISTER
 request can add a new binding between an address-of-record and one or
 more contact addresses.  Registration on behalf of a particular
 address-of-record can be performed by a suitably authorized third
 party.  A client can also remove previous bindings or query to
 determine which bindings are currently in place for an address-of-
 record.
 Except as noted, the construction of the REGISTER request and the
 behavior of clients sending a REGISTER request is identical to the
 general UAC behavior described in Section 8.1 and Section 17.1.
 A REGISTER request does not establish a dialog.  A UAC MAY include a
 Route header field in a REGISTER request based on a pre-existing
 route set as described in Section 8.1.  The Record-Route header field
 has no meaning in REGISTER requests or responses, and MUST be ignored
 if present.  In particular, the UAC MUST NOT create a new route set
 based on the presence or absence of a Record-Route header field in
 any response to a REGISTER request.
 The following header fields, except Contact, MUST be included in a
 REGISTER request.  A Contact header field MAY be included:
    Request-URI: The Request-URI names the domain of the location
         service for which the registration is meant (for example,
         "sip:chicago.com").  The "userinfo" and "@" components of the
         SIP URI MUST NOT be present.
    To: The To header field contains the address of record whose
         registration is to be created, queried, or modified.  The To
         header field and the Request-URI field typically differ, as
         the former contains a user name.  This address-of-record MUST
         be a SIP URI or SIPS URI.

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    From: The From header field contains the address-of-record of the
         person responsible for the registration.  The value is the
         same as the To header field unless the request is a third-
         party registration.
    Call-ID: All registrations from a UAC SHOULD use the same Call-ID
         header field value for registrations sent to a particular
         registrar.
         If the same client were to use different Call-ID values, a
         registrar could not detect whether a delayed REGISTER request
         might have arrived out of order.
    CSeq: The CSeq value guarantees proper ordering of REGISTER
         requests.  A UA MUST increment the CSeq value by one for each
         REGISTER request with the same Call-ID.
    Contact: REGISTER requests MAY contain a Contact header field with
         zero or more values containing address bindings.
 UAs MUST NOT send a new registration (that is, containing new Contact
 header field values, as opposed to a retransmission) until they have
 received a final response from the registrar for the previous one or
 the previous REGISTER request has timed out.

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                                               bob
                                             +----+
                                             | UA |
                                             |    |
                                             +----+
                                                |
                                                |3)INVITE
                                                |   carol@chicago.com
       chicago.com        +--------+            V
       +---------+ 2)Store|Location|4)Query +-----+
       |Registrar|=======>| Service|<=======|Proxy|sip.chicago.com
       +---------+        +--------+=======>+-----+
             A                      5)Resp      |
             |                                  |
             |                                  |
   1)REGISTER|                                  |
             |                                  |
          +----+                                |
          | UA |<-------------------------------+
 cube2214a|    |                            6)INVITE
          +----+                    carol@cube2214a.chicago.com
           carol
                    Figure 2: REGISTER example
    The following Contact header parameters have a special meaning in
         REGISTER requests:
    action: The "action" parameter from RFC 2543 has been deprecated.
         UACs SHOULD NOT use the "action" parameter.
    expires: The "expires" parameter indicates how long the UA would
         like the binding to be valid.  The value is a number
         indicating seconds.  If this parameter is not provided, the
         value of the Expires header field is used instead.
         Implementations MAY treat values larger than 2**32-1
         (4294967295 seconds or 136 years) as equivalent to 2**32-1.
         Malformed values SHOULD be treated as equivalent to 3600.

10.2.1 Adding Bindings

 The REGISTER request sent to a registrar includes the contact
 address(es) to which SIP requests for the address-of-record should be
 forwarded.  The address-of-record is included in the To header field
 of the REGISTER request.

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 The Contact header field values of the request typically consist of
 SIP or SIPS URIs that identify particular SIP endpoints (for example,
 "sip:carol@cube2214a.chicago.com"), but they MAY use any URI scheme.
 A SIP UA can choose to register telephone numbers (with the tel URL,
 RFC 2806 [9]) or email addresses (with a mailto URL, RFC 2368 [32])
 as Contacts for an address-of-record, for example.
 For example, Carol, with address-of-record "sip:carol@chicago.com",
 would register with the SIP registrar of the domain chicago.com.  Her
 registrations would then be used by a proxy server in the chicago.com
 domain to route requests for Carol's address-of-record to her SIP
 endpoint.
 Once a client has established bindings at a registrar, it MAY send
 subsequent registrations containing new bindings or modifications to
 existing bindings as necessary.  The 2xx response to the REGISTER
 request will contain, in a Contact header field, a complete list of
 bindings that have been registered for this address-of-record at this
 registrar.
 If the address-of-record in the To header field of a REGISTER request
 is a SIPS URI, then any Contact header field values in the request
 SHOULD also be SIPS URIs.  Clients should only register non-SIPS URIs
 under a SIPS address-of-record when the security of the resource
 represented by the contact address is guaranteed by other means.
 This may be applicable to URIs that invoke protocols other than SIP,
 or SIP devices secured by protocols other than TLS.
 Registrations do not need to update all bindings.  Typically, a UA
 only updates its own contact addresses.

10.2.1.1 Setting the Expiration Interval of Contact Addresses

 When a client sends a REGISTER request, it MAY suggest an expiration
 interval that indicates how long the client would like the
 registration to be valid.  (As described in Section 10.3, the
 registrar selects the actual time interval based on its local
 policy.)
 There are two ways in which a client can suggest an expiration
 interval for a binding: through an Expires header field or an
 "expires" Contact header parameter.  The latter allows expiration
 intervals to be suggested on a per-binding basis when more than one
 binding is given in a single REGISTER request, whereas the former
 suggests an expiration interval for all Contact header field values
 that do not contain the "expires" parameter.

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 If neither mechanism for expressing a suggested expiration time is
 present in a REGISTER, the client is indicating its desire for the
 server to choose.

10.2.1.2 Preferences among Contact Addresses

 If more than one Contact is sent in a REGISTER request, the
 registering UA intends to associate all of the URIs in these Contact
 header field values with the address-of-record present in the To
 field.  This list can be prioritized with the "q" parameter in the
 Contact header field.  The "q" parameter indicates a relative
 preference for the particular Contact header field value compared to
 other bindings for this address-of-record.  Section 16.6 describes
 how a proxy server uses this preference indication.

10.2.2 Removing Bindings

 Registrations are soft state and expire unless refreshed, but can
 also be explicitly removed.  A client can attempt to influence the
 expiration interval selected by the registrar as described in Section
 10.2.1.  A UA requests the immediate removal of a binding by
 specifying an expiration interval of "0" for that contact address in
 a REGISTER request.  UAs SHOULD support this mechanism so that
 bindings can be removed before their expiration interval has passed.
 The REGISTER-specific Contact header field value of "*" applies to
 all registrations, but it MUST NOT be used unless the Expires header
 field is present with a value of "0".
    Use of the "*" Contact header field value allows a registering UA
    to remove all bindings associated with an address-of-record
    without knowing their precise values.

10.2.3 Fetching Bindings

 A success response to any REGISTER request contains the complete list
 of existing bindings, regardless of whether the request contained a
 Contact header field.  If no Contact header field is present in a
 REGISTER request, the list of bindings is left unchanged.

10.2.4 Refreshing Bindings

 Each UA is responsible for refreshing the bindings that it has
 previously established.  A UA SHOULD NOT refresh bindings set up by
 other UAs.

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 The 200 (OK) response from the registrar contains a list of Contact
 fields enumerating all current bindings.  The UA compares each
 contact address to see if it created the contact address, using
 comparison rules in Section 19.1.4.  If so, it updates the expiration
 time interval according to the expires parameter or, if absent, the
 Expires field value.  The UA then issues a REGISTER request for each
 of its bindings before the expiration interval has elapsed.  It MAY
 combine several updates into one REGISTER request.
 A UA SHOULD use the same Call-ID for all registrations during a
 single boot cycle.  Registration refreshes SHOULD be sent to the same
 network address as the original registration, unless redirected.

10.2.5 Setting the Internal Clock

 If the response for a REGISTER request contains a Date header field,
 the client MAY use this header field to learn the current time in
 order to set any internal clocks.

10.2.6 Discovering a Registrar

 UAs can use three ways to determine the address to which to send
 registrations:  by configuration, using the address-of-record, and
 multicast.  A UA can be configured, in ways beyond the scope of this
 specification, with a registrar address.  If there is no configured
 registrar address, the UA SHOULD use the host part of the address-
 of-record as the Request-URI and address the request there, using the
 normal SIP server location mechanisms [4].  For example, the UA for
 the user "sip:carol@chicago.com" addresses the REGISTER request to
 "sip:chicago.com".
 Finally, a UA can be configured to use multicast.  Multicast
 registrations are addressed to the well-known "all SIP servers"
 multicast address "sip.mcast.net" (224.0.1.75 for IPv4).  No well-
 known IPv6 multicast address has been allocated; such an allocation
 will be documented separately when needed.  SIP UAs MAY listen to
 that address and use it to become aware of the location of other
 local users (see [33]); however, they do not respond to the request.
    Multicast registration may be inappropriate in some environments,
    for example, if multiple businesses share the same local area
    network.

10.2.7 Transmitting a Request

 Once the REGISTER method has been constructed, and the destination of
 the message identified, UACs follow the procedures described in
 Section 8.1.2 to hand off the REGISTER to the transaction layer.

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 If the transaction layer returns a timeout error because the REGISTER
 yielded no response, the UAC SHOULD NOT immediately re-attempt a
 registration to the same registrar.
    An immediate re-attempt is likely to also timeout.  Waiting some
    reasonable time interval for the conditions causing the timeout to
    be corrected reduces unnecessary load on the network.  No specific
    interval is mandated.

10.2.8 Error Responses

 If a UA receives a 423 (Interval Too Brief) response, it MAY retry
 the registration after making the expiration interval of all contact
 addresses in the REGISTER request equal to or greater than the
 expiration interval within the Min-Expires header field of the 423
 (Interval Too Brief) response.

10.3 Processing REGISTER Requests

 A registrar is a UAS that responds to REGISTER requests and maintains
 a list of bindings that are accessible to proxy servers and redirect
 servers within its administrative domain.  A registrar handles
 requests according to Section 8.2 and Section 17.2, but it accepts
 only REGISTER requests.  A registrar MUST not generate 6xx responses.
 A registrar MAY redirect REGISTER requests as appropriate.  One
 common usage would be for a registrar listening on a multicast
 interface to redirect multicast REGISTER requests to its own unicast
 interface with a 302 (Moved Temporarily) response.
 Registrars MUST ignore the Record-Route header field if it is
 included in a REGISTER request.  Registrars MUST NOT include a
 Record-Route header field in any response to a REGISTER request.
    A registrar might receive a request that traversed a proxy which
    treats REGISTER as an unknown request and which added a Record-
    Route header field value.
 A registrar has to know (for example, through configuration) the set
 of domain(s) for which it maintains bindings.  REGISTER requests MUST
 be processed by a registrar in the order that they are received.
 REGISTER requests MUST also be processed atomically, meaning that a
 particular REGISTER request is either processed completely or not at
 all.  Each REGISTER message MUST be processed independently of any
 other registration or binding changes.

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 When receiving a REGISTER request, a registrar follows these steps:
    1. The registrar inspects the Request-URI to determine whether it
       has access to bindings for the domain identified in the
       Request-URI.  If not, and if the server also acts as a proxy
       server, the server SHOULD forward the request to the addressed
       domain, following the general behavior for proxying messages
       described in Section 16.
    2. To guarantee that the registrar supports any necessary
       extensions, the registrar MUST process the Require header field
       values as described for UASs in Section 8.2.2.
    3. A registrar SHOULD authenticate the UAC.  Mechanisms for the
       authentication of SIP user agents are described in Section 22.
       Registration behavior in no way overrides the generic
       authentication framework for SIP.  If no authentication
       mechanism is available, the registrar MAY take the From address
       as the asserted identity of the originator of the request.
    4. The registrar SHOULD determine if the authenticated user is
       authorized to modify registrations for this address-of-record.
       For example, a registrar might consult an authorization
       database that maps user names to a list of addresses-of-record
       for which that user has authorization to modify bindings.  If
       the authenticated user is not authorized to modify bindings,
       the registrar MUST return a 403 (Forbidden) and skip the
       remaining steps.
       In architectures that support third-party registration, one
       entity may be responsible for updating the registrations
       associated with multiple addresses-of-record.
    5. The registrar extracts the address-of-record from the To header
       field of the request.  If the address-of-record is not valid
       for the domain in the Request-URI, the registrar MUST send a
       404 (Not Found) response and skip the remaining steps.  The URI
       MUST then be converted to a canonical form.  To do that, all
       URI parameters MUST be removed (including the user-param), and
       any escaped characters MUST be converted to their unescaped
       form.  The result serves as an index into the list of bindings.

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    6. The registrar checks whether the request contains the Contact
       header field.  If not, it skips to the last step.  If the
       Contact header field is present, the registrar checks if there
       is one Contact field value that contains the special value "*"
       and an Expires field.  If the request has additional Contact
       fields or an expiration time other than zero, the request is
       invalid, and the server MUST return a 400 (Invalid Request) and
       skip the remaining steps.  If not, the registrar checks whether
       the Call-ID agrees with the value stored for each binding.  If
       not, it MUST remove the binding.  If it does agree, it MUST
       remove the binding only if the CSeq in the request is higher
       than the value stored for that binding.  Otherwise, the update
       MUST be aborted and the request fails.
    7. The registrar now processes each contact address in the Contact
       header field in turn.  For each address, it determines the
       expiration interval as follows:
  1. If the field value has an "expires" parameter, that value

MUST be taken as the requested expiration.

  1. If there is no such parameter, but the request has an

Expires header field, that value MUST be taken as the

          requested expiration.
  1. If there is neither, a locally-configured default value MUST

be taken as the requested expiration.

       The registrar MAY choose an expiration less than the requested
       expiration interval.  If and only if the requested expiration
       interval is greater than zero AND smaller than one hour AND
       less than a registrar-configured minimum, the registrar MAY
       reject the registration with a response of 423 (Interval Too
       Brief).  This response MUST contain a Min-Expires header field
       that states the minimum expiration interval the registrar is
       willing to honor.  It then skips the remaining steps.
       Allowing the registrar to set the registration interval
       protects it against excessively frequent registration refreshes
       while limiting the state that it needs to maintain and
       decreasing the likelihood of registrations going stale.  The
       expiration interval of a registration is frequently used in the
       creation of services.  An example is a follow-me service, where
       the user may only be available at a terminal for a brief
       period.  Therefore, registrars should accept brief
       registrations; a request should only be rejected if the
       interval is so short that the refreshes would degrade registrar
       performance.

Rosenberg, et. al. Standards Track [Page 65] RFC 3261 SIP: Session Initiation Protocol June 2002

       For each address, the registrar then searches the list of
       current bindings using the URI comparison rules.  If the
       binding does not exist, it is tentatively added.  If the
       binding does exist, the registrar checks the Call-ID value.  If
       the Call-ID value in the existing binding differs from the
       Call-ID value in the request, the binding MUST be removed if
       the expiration time is zero and updated otherwise.  If they are
       the same, the registrar compares the CSeq value.  If the value
       is higher than that of the existing binding, it MUST update or
       remove the binding as above.  If not, the update MUST be
       aborted and the request fails.
       This algorithm ensures that out-of-order requests from the same
       UA are ignored.
       Each binding record records the Call-ID and CSeq values from
       the request.
       The binding updates MUST be committed (that is, made visible to
       the proxy or redirect server) if and only if all binding
       updates and additions succeed.  If any one of them fails (for
       example, because the back-end database commit failed), the
       request MUST fail with a 500 (Server Error) response and all
       tentative binding updates MUST be removed.
    8. The registrar returns a 200 (OK) response.  The response MUST
       contain Contact header field values enumerating all current
       bindings.  Each Contact value MUST feature an "expires"
       parameter indicating its expiration interval chosen by the
       registrar.  The response SHOULD include a Date header field.

11 Querying for Capabilities

 The SIP method OPTIONS allows a UA to query another UA or a proxy
 server as to its capabilities.  This allows a client to discover
 information about the supported methods, content types, extensions,
 codecs, etc. without "ringing" the other party.  For example, before
 a client inserts a Require header field into an INVITE listing an
 option that it is not certain the destination UAS supports, the
 client can query the destination UAS with an OPTIONS to see if this
 option is returned in a Supported header field.  All UAs MUST support
 the OPTIONS method.
 The target of the OPTIONS request is identified by the Request-URI,
 which could identify another UA or a SIP server.  If the OPTIONS is
 addressed to a proxy server, the Request-URI is set without a user
 part, similar to the way a Request-URI is set for a REGISTER request.

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 Alternatively, a server receiving an OPTIONS request with a Max-
 Forwards header field value of 0 MAY respond to the request
 regardless of the Request-URI.
    This behavior is common with HTTP/1.1.  This behavior can be used
    as a "traceroute" functionality to check the capabilities of
    individual hop servers by sending a series of OPTIONS requests
    with incremented Max-Forwards values.
 As is the case for general UA behavior, the transaction layer can
 return a timeout error if the OPTIONS yields no response.  This may
 indicate that the target is unreachable and hence unavailable.
 An OPTIONS request MAY be sent as part of an established dialog to
 query the peer on capabilities that may be utilized later in the
 dialog.

11.1 Construction of OPTIONS Request

 An OPTIONS request is constructed using the standard rules for a SIP
 request as discussed in Section 8.1.1.
 A Contact header field MAY be present in an OPTIONS.
 An Accept header field SHOULD be included to indicate the type of
 message body the UAC wishes to receive in the response.  Typically,
 this is set to a format that is used to describe the media
 capabilities of a UA, such as SDP (application/sdp).
 The response to an OPTIONS request is assumed to be scoped to the
 Request-URI in the original request.  However, only when an OPTIONS
 is sent as part of an established dialog is it guaranteed that future
 requests will be received by the server that generated the OPTIONS
 response.
 Example OPTIONS request:
    OPTIONS sip:carol@chicago.com SIP/2.0
    Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
    Max-Forwards: 70
    To: <sip:carol@chicago.com>
    From: Alice <sip:alice@atlanta.com>;tag=1928301774
    Call-ID: a84b4c76e66710
    CSeq: 63104 OPTIONS
    Contact: <sip:alice@pc33.atlanta.com>
    Accept: application/sdp
    Content-Length: 0

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11.2 Processing of OPTIONS Request

 The response to an OPTIONS is constructed using the standard rules
 for a SIP response as discussed in Section 8.2.6.  The response code
 chosen MUST be the same that would have been chosen had the request
 been an INVITE.  That is, a 200 (OK) would be returned if the UAS is
 ready to accept a call, a 486 (Busy Here) would be returned if the
 UAS is busy, etc.  This allows an OPTIONS request to be used to
 determine the basic state of a UAS, which can be an indication of
 whether the UAS will accept an INVITE request.
 An OPTIONS request received within a dialog generates a 200 (OK)
 response that is identical to one constructed outside a dialog and
 does not have any impact on the dialog.
 This use of OPTIONS has limitations due to the differences in proxy
 handling of OPTIONS and INVITE requests.  While a forked INVITE can
 result in multiple 200 (OK) responses being returned, a forked
 OPTIONS will only result in a single 200 (OK) response, since it is
 treated by proxies using the non-INVITE handling.  See Section 16.7
 for the normative details.
 If the response to an OPTIONS is generated by a proxy server, the
 proxy returns a 200 (OK), listing the capabilities of the server.
 The response does not contain a message body.
 Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
 fields SHOULD be present in a 200 (OK) response to an OPTIONS
 request.  If the response is generated by a proxy, the Allow header
 field SHOULD be omitted as it is ambiguous since a proxy is method
 agnostic.  Contact header fields MAY be present in a 200 (OK)
 response and have the same semantics as in a 3xx response.  That is,
 they may list a set of alternative names and methods of reaching the
 user.  A Warning header field MAY be present.
 A message body MAY be sent, the type of which is determined by the
 Accept header field in the OPTIONS request (application/sdp is the
 default if the Accept header field is not present).  If the types
 include one that can describe media capabilities, the UAS SHOULD
 include a body in the response for that purpose.  Details on the
 construction of such a body in the case of application/sdp are
 described in [13].

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 Example OPTIONS response generated by a UAS (corresponding to the
 request in Section 11.1):
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
     ;received=192.0.2.4
    To: <sip:carol@chicago.com>;tag=93810874
    From: Alice <sip:alice@atlanta.com>;tag=1928301774
    Call-ID: a84b4c76e66710
    CSeq: 63104 OPTIONS
    Contact: <sip:carol@chicago.com>
    Contact: <mailto:carol@chicago.com>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
    Accept: application/sdp
    Accept-Encoding: gzip
    Accept-Language: en
    Supported: foo
    Content-Type: application/sdp
    Content-Length: 274
    (SDP not shown)

12 Dialogs

 A key concept for a user agent is that of a dialog.  A dialog
 represents a peer-to-peer SIP relationship between two user agents
 that persists for some time.  The dialog facilitates sequencing of
 messages between the user agents and proper routing of requests
 between both of them.  The dialog represents a context in which to
 interpret SIP messages.  Section 8 discussed method independent UA
 processing for requests and responses outside of a dialog.  This
 section discusses how those requests and responses are used to
 construct a dialog, and then how subsequent requests and responses
 are sent within a dialog.
 A dialog is identified at each UA with a dialog ID, which consists of
 a Call-ID value, a local tag and a remote tag.  The dialog ID at each
 UA involved in the dialog is not the same.  Specifically, the local
 tag at one UA is identical to the remote tag at the peer UA.  The
 tags are opaque tokens that facilitate the generation of unique
 dialog IDs.
 A dialog ID is also associated with all responses and with any
 request that contains a tag in the To field.  The rules for computing
 the dialog ID of a message depend on whether the SIP element is a UAC
 or UAS.  For a UAC, the Call-ID value of the dialog ID is set to the
 Call-ID of the message, the remote tag is set to the tag in the To
 field of the message, and the local tag is set to the tag in the From

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 field of the message (these rules apply to both requests and
 responses).  As one would expect for a UAS, the Call-ID value of the
 dialog ID is set to the Call-ID of the message, the remote tag is set
 to the tag in the From field of the message, and the local tag is set
 to the tag in the To field of the message.
 A dialog contains certain pieces of state needed for further message
 transmissions within the dialog.  This state consists of the dialog
 ID, a local sequence number (used to order requests from the UA to
 its peer), a remote sequence number (used to order requests from its
 peer to the UA), a local URI, a remote URI, remote target, a boolean
 flag called "secure", and a route set, which is an ordered list of
 URIs.  The route set is the list of servers that need to be traversed
 to send a request to the peer.  A dialog can also be in the "early"
 state, which occurs when it is created with a provisional response,
 and then transition to the "confirmed" state when a 2xx final
 response arrives.  For other responses, or if no response arrives at
 all on that dialog, the early dialog terminates.

12.1 Creation of a Dialog

 Dialogs are created through the generation of non-failure responses
 to requests with specific methods.  Within this specification, only
 2xx and 101-199 responses with a To tag, where the request was
 INVITE, will establish a dialog.  A dialog established by a non-final
 response to a request is in the "early" state and it is called an
 early dialog.  Extensions MAY define other means for creating
 dialogs.  Section 13 gives more details that are specific to the
 INVITE method.  Here, we describe the process for creation of dialog
 state that is not dependent on the method.
 UAs MUST assign values to the dialog ID components as described
 below.

12.1.1 UAS behavior

 When a UAS responds to a request with a response that establishes a
 dialog (such as a 2xx to INVITE), the UAS MUST copy all Record-Route
 header field values from the request into the response (including the
 URIs, URI parameters, and any Record-Route header field parameters,
 whether they are known or unknown to the UAS) and MUST maintain the
 order of those values.  The UAS MUST add a Contact header field to
 the response.  The Contact header field contains an address where the
 UAS would like to be contacted for subsequent requests in the dialog
 (which includes the ACK for a 2xx response in the case of an INVITE).
 Generally, the host portion of this URI is the IP address or FQDN of
 the host.  The URI provided in the Contact header field MUST be a SIP
 or SIPS URI.  If the request that initiated the dialog contained a

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 SIPS URI in the Request-URI or in the top Record-Route header field
 value, if there was any, or the Contact header field if there was no
 Record-Route header field, the Contact header field in the response
 MUST be a SIPS URI.  The URI SHOULD have global scope (that is, the
 same URI can be used in messages outside this dialog).  The same way,
 the scope of the URI in the Contact header field of the INVITE is not
 limited to this dialog either.  It can therefore be used in messages
 to the UAC even outside this dialog.
 The UAS then constructs the state of the dialog.  This state MUST be
 maintained for the duration of the dialog.
 If the request arrived over TLS, and the Request-URI contained a SIPS
 URI, the "secure" flag is set to TRUE.
 The route set MUST be set to the list of URIs in the Record-Route
 header field from the request, taken in order and preserving all URI
 parameters.  If no Record-Route header field is present in the
 request, the route set MUST be set to the empty set.  This route set,
 even if empty, overrides any pre-existing route set for future
 requests in this dialog.  The remote target MUST be set to the URI
 from the Contact header field of the request.
 The remote sequence number MUST be set to the value of the sequence
 number in the CSeq header field of the request.  The local sequence
 number MUST be empty.  The call identifier component of the dialog ID
 MUST be set to the value of the Call-ID in the request.  The local
 tag component of the dialog ID MUST be set to the tag in the To field
 in the response to the request (which always includes a tag), and the
 remote tag component of the dialog ID MUST be set to the tag from the
 From field in the request.  A UAS MUST be prepared to receive a
 request without a tag in the From field, in which case the tag is
 considered to have a value of null.
    This is to maintain backwards compatibility with RFC 2543, which
    did not mandate From tags.
 The remote URI MUST be set to the URI in the From field, and the
 local URI MUST be set to the URI in the To field.

12.1.2 UAC Behavior

 When a UAC sends a request that can establish a dialog (such as an
 INVITE) it MUST provide a SIP or SIPS URI with global scope (i.e.,
 the same SIP URI can be used in messages outside this dialog) in the
 Contact header field of the request.  If the request has a Request-
 URI or a topmost Route header field value with a SIPS URI, the
 Contact header field MUST contain a SIPS URI.

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 When a UAC receives a response that establishes a dialog, it
 constructs the state of the dialog.  This state MUST be maintained
 for the duration of the dialog.
 If the request was sent over TLS, and the Request-URI contained a
 SIPS URI, the "secure" flag is set to TRUE.
 The route set MUST be set to the list of URIs in the Record-Route
 header field from the response, taken in reverse order and preserving
 all URI parameters.  If no Record-Route header field is present in
 the response, the route set MUST be set to the empty set.  This route
 set, even if empty, overrides any pre-existing route set for future
 requests in this dialog.  The remote target MUST be set to the URI
 from the Contact header field of the response.
 The local sequence number MUST be set to the value of the sequence
 number in the CSeq header field of the request.  The remote sequence
 number MUST be empty (it is established when the remote UA sends a
 request within the dialog).  The call identifier component of the
 dialog ID MUST be set to the value of the Call-ID in the request.
 The local tag component of the dialog ID MUST be set to the tag in
 the From field in the request, and the remote tag component of the
 dialog ID MUST be set to the tag in the To field of the response.  A
 UAC MUST be prepared to receive a response without a tag in the To
 field, in which case the tag is considered to have a value of null.
    This is to maintain backwards compatibility with RFC 2543, which
    did not mandate To tags.
 The remote URI MUST be set to the URI in the To field, and the local
 URI MUST be set to the URI in the From field.

12.2 Requests within a Dialog

 Once a dialog has been established between two UAs, either of them
 MAY initiate new transactions as needed within the dialog.  The UA
 sending the request will take the UAC role for the transaction.  The
 UA receiving the request will take the UAS role.  Note that these may
 be different roles than the UAs held during the transaction that
 established the dialog.
 Requests within a dialog MAY contain Record-Route and Contact header
 fields.  However, these requests do not cause the dialog's route set
 to be modified, although they may modify the remote target URI.
 Specifically, requests that are not target refresh requests do not
 modify the dialog's remote target URI, and requests that are target
 refresh requests do.  For dialogs that have been established with an

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 INVITE, the only target refresh request defined is re-INVITE (see
 Section 14).  Other extensions may define different target refresh
 requests for dialogs established in other ways.
    Note that an ACK is NOT a target refresh request.
 Target refresh requests only update the dialog's remote target URI,
 and not the route set formed from the Record-Route.  Updating the
 latter would introduce severe backwards compatibility problems with
 RFC 2543-compliant systems.

12.2.1 UAC Behavior

12.2.1.1 Generating the Request

 A request within a dialog is constructed by using many of the
 components of the state stored as part of the dialog.
 The URI in the To field of the request MUST be set to the remote URI
 from the dialog state.  The tag in the To header field of the request
 MUST be set to the remote tag of the dialog ID.  The From URI of the
 request MUST be set to the local URI from the dialog state.  The tag
 in the From header field of the request MUST be set to the local tag
 of the dialog ID.  If the value of the remote or local tags is null,
 the tag parameter MUST be omitted from the To or From header fields,
 respectively.
    Usage of the URI from the To and From fields in the original
    request within subsequent requests is done for backwards
    compatibility with RFC 2543, which used the URI for dialog
    identification.  In this specification, only the tags are used for
    dialog identification.  It is expected that mandatory reflection
    of the original To and From URI in mid-dialog requests will be
    deprecated in a subsequent revision of this specification.
 The Call-ID of the request MUST be set to the Call-ID of the dialog.
 Requests within a dialog MUST contain strictly monotonically
 increasing and contiguous CSeq sequence numbers (increasing-by-one)
 in each direction (excepting ACK and CANCEL of course, whose numbers
 equal the requests being acknowledged or cancelled).  Therefore, if
 the local sequence number is not empty, the value of the local
 sequence number MUST be incremented by one, and this value MUST be
 placed into the CSeq header field.  If the local sequence number is
 empty, an initial value MUST be chosen using the guidelines of
 Section 8.1.1.5.  The method field in the CSeq header field value
 MUST match the method of the request.

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    With a length of 32 bits, a client could generate, within a single
    call, one request a second for about 136 years before needing to
    wrap around.  The initial value of the sequence number is chosen
    so that subsequent requests within the same call will not wrap
    around.  A non-zero initial value allows clients to use a time-
    based initial sequence number.  A client could, for example,
    choose the 31 most significant bits of a 32-bit second clock as an
    initial sequence number.
 The UAC uses the remote target and route set to build the Request-URI
 and Route header field of the request.
 If the route set is empty, the UAC MUST place the remote target URI
 into the Request-URI.  The UAC MUST NOT add a Route header field to
 the request.
 If the route set is not empty, and the first URI in the route set
 contains the lr parameter (see Section 19.1.1), the UAC MUST place
 the remote target URI into the Request-URI and MUST include a Route
 header field containing the route set values in order, including all
 parameters.
 If the route set is not empty, and its first URI does not contain the
 lr parameter, the UAC MUST place the first URI from the route set
 into the Request-URI, stripping any parameters that are not allowed
 in a Request-URI.  The UAC MUST add a Route header field containing
 the remainder of the route set values in order, including all
 parameters.  The UAC MUST then place the remote target URI into the
 Route header field as the last value.
 For example, if the remote target is sip:user@remoteua and the route
 set contains:
    <sip:proxy1>,<sip:proxy2>,<sip:proxy3;lr>,<sip:proxy4>
 The request will be formed with the following Request-URI and Route
 header field:
 METHOD sip:proxy1
 Route: <sip:proxy2>,<sip:proxy3;lr>,<sip:proxy4>,<sip:user@remoteua>
    If the first URI of the route set does not contain the lr
    parameter, the proxy indicated does not understand the routing
    mechanisms described in this document and will act as specified in
    RFC 2543, replacing the Request-URI with the first Route header
    field value it receives while forwarding the message.  Placing the
    Request-URI at the end of the Route header field preserves the

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    information in that Request-URI across the strict router (it will
    be returned to the Request-URI when the request reaches a loose-
    router).
 A UAC SHOULD include a Contact header field in any target refresh
 requests within a dialog, and unless there is a need to change it,
 the URI SHOULD be the same as used in previous requests within the
 dialog.  If the "secure" flag is true, that URI MUST be a SIPS URI.
 As discussed in Section 12.2.2, a Contact header field in a target
 refresh request updates the remote target URI.  This allows a UA to
 provide a new contact address, should its address change during the
 duration of the dialog.
 However, requests that are not target refresh requests do not affect
 the remote target URI for the dialog.
 The rest of the request is formed as described in Section 8.1.1.
 Once the request has been constructed, the address of the server is
 computed and the request is sent, using the same procedures for
 requests outside of a dialog (Section 8.1.2).
    The procedures in Section 8.1.2 will normally result in the
    request being sent to the address indicated by the topmost Route
    header field value or the Request-URI if no Route header field is
    present.  Subject to certain restrictions, they allow the request
    to be sent to an alternate address (such as a default outbound
    proxy not represented in the route set).

12.2.1.2 Processing the Responses

 The UAC will receive responses to the request from the transaction
 layer.  If the client transaction returns a timeout, this is treated
 as a 408 (Request Timeout) response.
 The behavior of a UAC that receives a 3xx response for a request sent
 within a dialog is the same as if the request had been sent outside a
 dialog.  This behavior is described in Section 8.1.3.4.
    Note, however, that when the UAC tries alternative locations, it
    still uses the route set for the dialog to build the Route header
    of the request.
 When a UAC receives a 2xx response to a target refresh request, it
 MUST replace the dialog's remote target URI with the URI from the
 Contact header field in that response, if present.

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 If the response for a request within a dialog is a 481
 (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the UAC
 SHOULD terminate the dialog.  A UAC SHOULD also terminate a dialog if
 no response at all is received for the request (the client
 transaction would inform the TU about the timeout.)
    For INVITE initiated dialogs, terminating the dialog consists of
    sending a BYE.

12.2.2 UAS Behavior

 Requests sent within a dialog, as any other requests, are atomic.  If
 a particular request is accepted by the UAS, all the state changes
 associated with it are performed.  If the request is rejected, none
 of the state changes are performed.
    Note that some requests, such as INVITEs, affect several pieces of
    state.
 The UAS will receive the request from the transaction layer.  If the
 request has a tag in the To header field, the UAS core computes the
 dialog identifier corresponding to the request and compares it with
 existing dialogs.  If there is a match, this is a mid-dialog request.
 In that case, the UAS first applies the same processing rules for
 requests outside of a dialog, discussed in Section 8.2.
 If the request has a tag in the To header field, but the dialog
 identifier does not match any existing dialogs, the UAS may have
 crashed and restarted, or it may have received a request for a
 different (possibly failed) UAS (the UASs can construct the To tags
 so that a UAS can identify that the tag was for a UAS for which it is
 providing recovery).  Another possibility is that the incoming
 request has been simply misrouted.  Based on the To tag, the UAS MAY
 either accept or reject the request.  Accepting the request for
 acceptable To tags provides robustness, so that dialogs can persist
 even through crashes.  UAs wishing to support this capability must
 take into consideration some issues such as choosing monotonically
 increasing CSeq sequence numbers even across reboots, reconstructing
 the route set, and accepting out-of-range RTP timestamps and sequence
 numbers.
 If the UAS wishes to reject the request because it does not wish to
 recreate the dialog, it MUST respond to the request with a 481
 (Call/Transaction Does Not Exist) status code and pass that to the
 server transaction.

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 Requests that do not change in any way the state of a dialog may be
 received within a dialog (for example, an OPTIONS request).  They are
 processed as if they had been received outside the dialog.
 If the remote sequence number is empty, it MUST be set to the value
 of the sequence number in the CSeq header field value in the request.
 If the remote sequence number was not empty, but the sequence number
 of the request is lower than the remote sequence number, the request
 is out of order and MUST be rejected with a 500 (Server Internal
 Error) response.  If the remote sequence number was not empty, and
 the sequence number of the request is greater than the remote
 sequence number, the request is in order.  It is possible for the
 CSeq sequence number to be higher than the remote sequence number by
 more than one.  This is not an error condition, and a UAS SHOULD be
 prepared to receive and process requests with CSeq values more than
 one higher than the previous received request.  The UAS MUST then set
 the remote sequence number to the value of the sequence number in the
 CSeq header field value in the request.
    If a proxy challenges a request generated by the UAC, the UAC has
    to resubmit the request with credentials.  The resubmitted request
    will have a new CSeq number.  The UAS will never see the first
    request, and thus, it will notice a gap in the CSeq number space.
    Such a gap does not represent any error condition.
 When a UAS receives a target refresh request, it MUST replace the
 dialog's remote target URI with the URI from the Contact header field
 in that request, if present.

12.3 Termination of a Dialog

 Independent of the method, if a request outside of a dialog generates
 a non-2xx final response, any early dialogs created through
 provisional responses to that request are terminated.  The mechanism
 for terminating confirmed dialogs is method specific.  In this
 specification, the BYE method terminates a session and the dialog
 associated with it.  See Section 15 for details.

13 Initiating a Session

13.1 Overview

 When a user agent client desires to initiate a session (for example,
 audio, video, or a game), it formulates an INVITE request.  The
 INVITE request asks a server to establish a session.  This request
 may be forwarded by proxies, eventually arriving at one or more UAS
 that can potentially accept the invitation.  These UASs will
 frequently need to query the user about whether to accept the

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 invitation.  After some time, those UASs can accept the invitation
 (meaning the session is to be established) by sending a 2xx response.
 If the invitation is not accepted, a 3xx, 4xx, 5xx or 6xx response is
 sent, depending on the reason for the rejection.  Before sending a
 final response, the UAS can also send provisional responses (1xx) to
 advise the UAC of progress in contacting the called user.
 After possibly receiving one or more provisional responses, the UAC
 will get one or more 2xx responses or one non-2xx final response.
 Because of the protracted amount of time it can take to receive final
 responses to INVITE, the reliability mechanisms for INVITE
 transactions differ from those of other requests (like OPTIONS).
 Once it receives a final response, the UAC needs to send an ACK for
 every final response it receives.  The procedure for sending this ACK
 depends on the type of response.  For final responses between 300 and
 699, the ACK processing is done in the transaction layer and follows
 one set of rules (See Section 17).  For 2xx responses, the ACK is
 generated by the UAC core.
 A 2xx response to an INVITE establishes a session, and it also
 creates a dialog between the UA that issued the INVITE and the UA
 that generated the 2xx response.  Therefore, when multiple 2xx
 responses are received from different remote UAs (because the INVITE
 forked), each 2xx establishes a different dialog.  All these dialogs
 are part of the same call.
 This section provides details on the establishment of a session using
 INVITE.  A UA that supports INVITE MUST also support ACK, CANCEL and
 BYE.

13.2 UAC Processing

13.2.1 Creating the Initial INVITE

 Since the initial INVITE represents a request outside of a dialog,
 its construction follows the procedures of Section 8.1.1.  Additional
 processing is required for the specific case of INVITE.
 An Allow header field (Section 20.5) SHOULD be present in the INVITE.
 It indicates what methods can be invoked within a dialog, on the UA
 sending the INVITE, for the duration of the dialog.  For example, a
 UA capable of receiving INFO requests within a dialog [34] SHOULD
 include an Allow header field listing the INFO method.
 A Supported header field (Section 20.37) SHOULD be present in the
 INVITE.  It enumerates all the extensions understood by the UAC.

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 An Accept (Section 20.1) header field MAY be present in the INVITE.
 It indicates which Content-Types are acceptable to the UA, in both
 the response received by it, and in any subsequent requests sent to
 it within dialogs established by the INVITE.  The Accept header field
 is especially useful for indicating support of various session
 description formats.
 The UAC MAY add an Expires header field (Section 20.19) to limit the
 validity of the invitation.  If the time indicated in the Expires
 header field is reached and no final answer for the INVITE has been
 received, the UAC core SHOULD generate a CANCEL request for the
 INVITE, as per Section 9.
 A UAC MAY also find it useful to add, among others, Subject (Section
 20.36), Organization (Section 20.25) and User-Agent (Section 20.41)
 header fields.  They all contain information related to the INVITE.
 The UAC MAY choose to add a message body to the INVITE.  Section
 8.1.1.10 deals with how to construct the header fields -- Content-
 Type among others -- needed to describe the message body.
 There are special rules for message bodies that contain a session
 description - their corresponding Content-Disposition is "session".
 SIP uses an offer/answer model where one UA sends a session
 description, called the offer, which contains a proposed description
 of the session.  The offer indicates the desired communications means
 (audio, video, games), parameters of those means (such as codec
 types) and addresses for receiving media from the answerer.  The
 other UA responds with another session description, called the
 answer, which indicates which communications means are accepted, the
 parameters that apply to those means, and addresses for receiving
 media from the offerer. An offer/answer exchange is within the
 context of a dialog, so that if a SIP INVITE results in multiple
 dialogs, each is a separate offer/answer exchange.  The offer/answer
 model defines restrictions on when offers and answers can be made
 (for example, you cannot make a new offer while one is in progress).
 This results in restrictions on where the offers and answers can
 appear in SIP messages.  In this specification, offers and answers
 can only appear in INVITE requests and responses, and ACK.  The usage
 of offers and answers is further restricted.  For the initial INVITE
 transaction, the rules are:
    o  The initial offer MUST be in either an INVITE or, if not there,
       in the first reliable non-failure message from the UAS back to
       the UAC.  In this specification, that is the final 2xx
       response.

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    o  If the initial offer is in an INVITE, the answer MUST be in a
       reliable non-failure message from UAS back to UAC which is
       correlated to that INVITE.  For this specification, that is
       only the final 2xx response to that INVITE.  That same exact
       answer MAY also be placed in any provisional responses sent
       prior to the answer.  The UAC MUST treat the first session
       description it receives as the answer, and MUST ignore any
       session descriptions in subsequent responses to the initial
       INVITE.
    o  If the initial offer is in the first reliable non-failure
       message from the UAS back to UAC, the answer MUST be in the
       acknowledgement for that message (in this specification, ACK
       for a 2xx response).
    o  After having sent or received an answer to the first offer, the
       UAC MAY generate subsequent offers in requests based on rules
       specified for that method, but only if it has received answers
       to any previous offers, and has not sent any offers to which it
       hasn't gotten an answer.
    o  Once the UAS has sent or received an answer to the initial
       offer, it MUST NOT generate subsequent offers in any responses
       to the initial INVITE.  This means that a UAS based on this
       specification alone can never generate subsequent offers until
       completion of the initial transaction.
 Concretely, the above rules specify two exchanges for UAs compliant
 to this specification alone - the offer is in the INVITE, and the
 answer in the 2xx (and possibly in a 1xx as well, with the same
 value), or the offer is in the 2xx, and the answer is in the ACK.
 All user agents that support INVITE MUST support these two exchanges.
 The Session Description Protocol (SDP) (RFC 2327 [1]) MUST be
 supported by all user agents as a means to describe sessions, and its
 usage for constructing offers and answers MUST follow the procedures
 defined in [13].
 The restrictions of the offer-answer model just described only apply
 to bodies whose Content-Disposition header field value is "session".
 Therefore, it is possible that both the INVITE and the ACK contain a
 body message (for example, the INVITE carries a photo (Content-
 Disposition: render) and the ACK a session description (Content-
 Disposition: session)).
 If the Content-Disposition header field is missing, bodies of
 Content-Type application/sdp imply the disposition "session", while
 other content types imply "render".

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 Once the INVITE has been created, the UAC follows the procedures
 defined for sending requests outside of a dialog (Section 8).  This
 results in the construction of a client transaction that will
 ultimately send the request and deliver responses to the UAC.

13.2.2 Processing INVITE Responses

 Once the INVITE has been passed to the INVITE client transaction, the
 UAC waits for responses for the INVITE.  If the INVITE client
 transaction returns a timeout rather than a response the TU acts as
 if a 408 (Request Timeout) response had been received, as described
 in Section 8.1.3.

13.2.2.1 1xx Responses

 Zero, one or multiple provisional responses may arrive before one or
 more final responses are received.  Provisional responses for an
 INVITE request can create "early dialogs".  If a provisional response
 has a tag in the To field, and if the dialog ID of the response does
 not match an existing dialog, one is constructed using the procedures
 defined in Section 12.1.2.
 The early dialog will only be needed if the UAC needs to send a
 request to its peer within the dialog before the initial INVITE
 transaction completes.  Header fields present in a provisional
 response are applicable as long as the dialog is in the early state
 (for example, an Allow header field in a provisional response
 contains the methods that can be used in the dialog while this is in
 the early state).

13.2.2.2 3xx Responses

 A 3xx response may contain one or more Contact header field values
 providing new addresses where the callee might be reachable.
 Depending on the status code of the 3xx response (see Section 21.3),
 the UAC MAY choose to try those new addresses.

13.2.2.3 4xx, 5xx and 6xx Responses

 A single non-2xx final response may be received for the INVITE.  4xx,
 5xx and 6xx responses may contain a Contact header field value
 indicating the location where additional information about the error
 can be found.  Subsequent final responses (which would only arrive
 under error conditions) MUST be ignored.
 All early dialogs are considered terminated upon reception of the
 non-2xx final response.

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 After having received the non-2xx final response the UAC core
 considers the INVITE transaction completed.  The INVITE client
 transaction handles the generation of ACKs for the response (see
 Section 17).

13.2.2.4 2xx Responses

 Multiple 2xx responses may arrive at the UAC for a single INVITE
 request due to a forking proxy.  Each response is distinguished by
 the tag parameter in the To header field, and each represents a
 distinct dialog, with a distinct dialog identifier.
 If the dialog identifier in the 2xx response matches the dialog
 identifier of an existing dialog, the dialog MUST be transitioned to
 the "confirmed" state, and the route set for the dialog MUST be
 recomputed based on the 2xx response using the procedures of Section
 12.2.1.2.  Otherwise, a new dialog in the "confirmed" state MUST be
 constructed using the procedures of Section 12.1.2.
    Note that the only piece of state that is recomputed is the route
    set.  Other pieces of state such as the highest sequence numbers
    (remote and local) sent within the dialog are not recomputed.  The
    route set only is recomputed for backwards compatibility.  RFC
    2543 did not mandate mirroring of the Record-Route header field in
    a 1xx, only 2xx.  However, we cannot update the entire state of
    the dialog, since mid-dialog requests may have been sent within
    the early dialog, modifying the sequence numbers, for example.
 The UAC core MUST generate an ACK request for each 2xx received from
 the transaction layer.  The header fields of the ACK are constructed
 in the same way as for any request sent within a dialog (see Section
 12) with the exception of the CSeq and the header fields related to
 authentication.  The sequence number of the CSeq header field MUST be
 the same as the INVITE being acknowledged, but the CSeq method MUST
 be ACK.  The ACK MUST contain the same credentials as the INVITE.  If
 the 2xx contains an offer (based on the rules above), the ACK MUST
 carry an answer in its body.  If the offer in the 2xx response is not
 acceptable, the UAC core MUST generate a valid answer in the ACK and
 then send a BYE immediately.
 Once the ACK has been constructed, the procedures of [4] are used to
 determine the destination address, port and transport.  However, the
 request is passed to the transport layer directly for transmission,
 rather than a client transaction.  This is because the UAC core
 handles retransmissions of the ACK, not the transaction layer.  The
 ACK MUST be passed to the client transport every time a
 retransmission of the 2xx final response that triggered the ACK
 arrives.

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 The UAC core considers the INVITE transaction completed 64*T1 seconds
 after the reception of the first 2xx response.  At this point all the
 early dialogs that have not transitioned to established dialogs are
 terminated.  Once the INVITE transaction is considered completed by
 the UAC core, no more new 2xx responses are expected to arrive.
 If, after acknowledging any 2xx response to an INVITE, the UAC does
 not want to continue with that dialog, then the UAC MUST terminate
 the dialog by sending a BYE request as described in Section 15.

13.3 UAS Processing

13.3.1 Processing of the INVITE

 The UAS core will receive INVITE requests from the transaction layer.
 It first performs the request processing procedures of Section 8.2,
 which are applied for both requests inside and outside of a dialog.
 Assuming these processing states are completed without generating a
 response, the UAS core performs the additional processing steps:
    1. If the request is an INVITE that contains an Expires header
       field, the UAS core sets a timer for the number of seconds
       indicated in the header field value.  When the timer fires, the
       invitation is considered to be expired.  If the invitation
       expires before the UAS has generated a final response, a 487
       (Request Terminated) response SHOULD be generated.
    2. If the request is a mid-dialog request, the method-independent
       processing described in Section 12.2.2 is first applied.  It
       might also modify the session; Section 14 provides details.
    3. If the request has a tag in the To header field but the dialog
       identifier does not match any of the existing dialogs, the UAS
       may have crashed and restarted, or may have received a request
       for a different (possibly failed) UAS.  Section 12.2.2 provides
       guidelines to achieve a robust behavior under such a situation.
 Processing from here forward assumes that the INVITE is outside of a
 dialog, and is thus for the purposes of establishing a new session.
 The INVITE may contain a session description, in which case the UAS
 is being presented with an offer for that session.  It is possible
 that the user is already a participant in that session, even though
 the INVITE is outside of a dialog.  This can happen when a user is
 invited to the same multicast conference by multiple other
 participants.  If desired, the UAS MAY use identifiers within the
 session description to detect this duplication.  For example, SDP

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 contains a session id and version number in the origin (o) field.  If
 the user is already a member of the session, and the session
 parameters contained in the session description have not changed, the
 UAS MAY silently accept the INVITE (that is, send a 2xx response
 without prompting the user).
 If the INVITE does not contain a session description, the UAS is
 being asked to participate in a session, and the UAC has asked that
 the UAS provide the offer of the session.  It MUST provide the offer
 in its first non-failure reliable message back to the UAC.  In this
 specification, that is a 2xx response to the INVITE.
 The UAS can indicate progress, accept, redirect, or reject the
 invitation.  In all of these cases, it formulates a response using
 the procedures described in Section 8.2.6.

13.3.1.1 Progress

 If the UAS is not able to answer the invitation immediately, it can
 choose to indicate some kind of progress to the UAC (for example, an
 indication that a phone is ringing).  This is accomplished with a
 provisional response between 101 and 199.  These provisional
 responses establish early dialogs and therefore follow the procedures
 of Section 12.1.1 in addition to those of Section 8.2.6.  A UAS MAY
 send as many provisional responses as it likes.  Each of these MUST
 indicate the same dialog ID.  However, these will not be delivered
 reliably.
 If the UAS desires an extended period of time to answer the INVITE,
 it will need to ask for an "extension" in order to prevent proxies
 from canceling the transaction.  A proxy has the option of canceling
 a transaction when there is a gap of 3 minutes between responses in a
 transaction.  To prevent cancellation, the UAS MUST send a non-100
 provisional response at every minute, to handle the possibility of
 lost provisional responses.
    An INVITE transaction can go on for extended durations when the
    user is placed on hold, or when interworking with PSTN systems
    which allow communications to take place without answering the
    call.  The latter is common in Interactive Voice Response (IVR)
    systems.

13.3.1.2 The INVITE is Redirected

 If the UAS decides to redirect the call, a 3xx response is sent.  A
 300 (Multiple Choices), 301 (Moved Permanently) or 302 (Moved
 Temporarily) response SHOULD contain a Contact header field

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 containing one or more URIs of new addresses to be tried.  The
 response is passed to the INVITE server transaction, which will deal
 with its retransmissions.

13.3.1.3 The INVITE is Rejected

 A common scenario occurs when the callee is currently not willing or
 able to take additional calls at this end system.  A 486 (Busy Here)
 SHOULD be returned in such a scenario.  If the UAS knows that no
 other end system will be able to accept this call, a 600 (Busy
 Everywhere) response SHOULD be sent instead.  However, it is unlikely
 that a UAS will be able to know this in general, and thus this
 response will not usually be used.  The response is passed to the
 INVITE server transaction, which will deal with its retransmissions.
 A UAS rejecting an offer contained in an INVITE SHOULD return a 488
 (Not Acceptable Here) response.  Such a response SHOULD include a
 Warning header field value explaining why the offer was rejected.

13.3.1.4 The INVITE is Accepted

 The UAS core generates a 2xx response.  This response establishes a
 dialog, and therefore follows the procedures of Section 12.1.1 in
 addition to those of Section 8.2.6.
 A 2xx response to an INVITE SHOULD contain the Allow header field and
 the Supported header field, and MAY contain the Accept header field.
 Including these header fields allows the UAC to determine the
 features and extensions supported by the UAS for the duration of the
 call, without probing.
 If the INVITE request contained an offer, and the UAS had not yet
 sent an answer, the 2xx MUST contain an answer.  If the INVITE did
 not contain an offer, the 2xx MUST contain an offer if the UAS had
 not yet sent an offer.
 Once the response has been constructed, it is passed to the INVITE
 server transaction.  Note, however, that the INVITE server
 transaction will be destroyed as soon as it receives this final
 response and passes it to the transport.  Therefore, it is necessary
 to periodically pass the response directly to the transport until the
 ACK arrives.  The 2xx response is passed to the transport with an
 interval that starts at T1 seconds and doubles for each
 retransmission until it reaches T2 seconds (T1 and T2 are defined in
 Section 17).  Response retransmissions cease when an ACK request for
 the response is received.  This is independent of whatever transport
 protocols are used to send the response.

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    Since 2xx is retransmitted end-to-end, there may be hops between
    UAS and UAC that are UDP.  To ensure reliable delivery across
    these hops, the response is retransmitted periodically even if the
    transport at the UAS is reliable.
 If the server retransmits the 2xx response for 64*T1 seconds without
 receiving an ACK, the dialog is confirmed, but the session SHOULD be
 terminated.  This is accomplished with a BYE, as described in Section
 15.

14 Modifying an Existing Session

 A successful INVITE request (see Section 13) establishes both a
 dialog between two user agents and a session using the offer-answer
 model.  Section 12 explains how to modify an existing dialog using a
 target refresh request (for example, changing the remote target URI
 of the dialog).  This section describes how to modify the actual
 session.  This modification can involve changing addresses or ports,
 adding a media stream, deleting a media stream, and so on.  This is
 accomplished by sending a new INVITE request within the same dialog
 that established the session.  An INVITE request sent within an
 existing dialog is known as a re-INVITE.
    Note that a single re-INVITE can modify the dialog and the
    parameters of the session at the same time.
 Either the caller or callee can modify an existing session.
 The behavior of a UA on detection of media failure is a matter of
 local policy.  However, automated generation of re-INVITE or BYE is
 NOT RECOMMENDED to avoid flooding the network with traffic when there
 is congestion.  In any case, if these messages are sent
 automatically, they SHOULD be sent after some randomized interval.
    Note that the paragraph above refers to automatically generated
    BYEs and re-INVITEs.  If the user hangs up upon media failure, the
    UA would send a BYE request as usual.

14.1 UAC Behavior

 The same offer-answer model that applies to session descriptions in
 INVITEs (Section 13.2.1) applies to re-INVITEs.  As a result, a UAC
 that wants to add a media stream, for example, will create a new
 offer that contains this media stream, and send that in an INVITE
 request to its peer.  It is important to note that the full
 description of the session, not just the change, is sent.  This
 supports stateless session processing in various elements, and
 supports failover and recovery capabilities.  Of course, a UAC MAY

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 send a re-INVITE with no session description, in which case the first
 reliable non-failure response to the re-INVITE will contain the offer
 (in this specification, that is a 2xx response).
 If the session description format has the capability for version
 numbers, the offerer SHOULD indicate that the version of the session
 description has changed.
 The To, From, Call-ID, CSeq, and Request-URI of a re-INVITE are set
 following the same rules as for regular requests within an existing
 dialog, described in Section 12.
 A UAC MAY choose not to add an Alert-Info header field or a body with
 Content-Disposition "alert" to re-INVITEs because UASs do not
 typically alert the user upon reception of a re-INVITE.
 Unlike an INVITE, which can fork, a re-INVITE will never fork, and
 therefore, only ever generate a single final response.  The reason a
 re-INVITE will never fork is that the Request-URI identifies the
 target as the UA instance it established the dialog with, rather than
 identifying an address-of-record for the user.
 Note that a UAC MUST NOT initiate a new INVITE transaction within a
 dialog while another INVITE transaction is in progress in either
 direction.
    1. If there is an ongoing INVITE client transaction, the TU MUST
       wait until the transaction reaches the completed or terminated
       state before initiating the new INVITE.
    2. If there is an ongoing INVITE server transaction, the TU MUST
       wait until the transaction reaches the confirmed or terminated
       state before initiating the new INVITE.
 However, a UA MAY initiate a regular transaction while an INVITE
 transaction is in progress.  A UA MAY also initiate an INVITE
 transaction while a regular transaction is in progress.
 If a UA receives a non-2xx final response to a re-INVITE, the session
 parameters MUST remain unchanged, as if no re-INVITE had been issued.
 Note that, as stated in Section 12.2.1.2, if the non-2xx final
 response is a 481 (Call/Transaction Does Not Exist), or a 408
 (Request Timeout), or no response at all is received for the re-
 INVITE (that is, a timeout is returned by the INVITE client
 transaction), the UAC will terminate the dialog.

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 If a UAC receives a 491 response to a re-INVITE, it SHOULD start a
 timer with a value T chosen as follows:
    1. If the UAC is the owner of the Call-ID of the dialog ID
       (meaning it generated the value), T has a randomly chosen value
       between 2.1 and 4 seconds in units of 10 ms.
    2. If the UAC is not the owner of the Call-ID of the dialog ID, T
       has a randomly chosen value of between 0 and 2 seconds in units
       of 10 ms.
 When the timer fires, the UAC SHOULD attempt the re-INVITE once more,
 if it still desires for that session modification to take place.  For
 example, if the call was already hung up with a BYE, the re-INVITE
 would not take place.
 The rules for transmitting a re-INVITE and for generating an ACK for
 a 2xx response to re-INVITE are the same as for the initial INVITE
 (Section 13.2.1).

14.2 UAS Behavior

 Section 13.3.1 describes the procedure for distinguishing incoming
 re-INVITEs from incoming initial INVITEs and handling a re-INVITE for
 an existing dialog.
 A UAS that receives a second INVITE before it sends the final
 response to a first INVITE with a lower CSeq sequence number on the
 same dialog MUST return a 500 (Server Internal Error) response to the
 second INVITE and MUST include a Retry-After header field with a
 randomly chosen value of between 0 and 10 seconds.
 A UAS that receives an INVITE on a dialog while an INVITE it had sent
 on that dialog is in progress MUST return a 491 (Request Pending)
 response to the received INVITE.
 If a UA receives a re-INVITE for an existing dialog, it MUST check
 any version identifiers in the session description or, if there are
 no version identifiers, the content of the session description to see
 if it has changed.  If the session description has changed, the UAS
 MUST adjust the session parameters accordingly, possibly after asking
 the user for confirmation.
    Versioning of the session description can be used to accommodate
    the capabilities of new arrivals to a conference, add or delete
    media, or change from a unicast to a multicast conference.

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 If the new session description is not acceptable, the UAS can reject
 it by returning a 488 (Not Acceptable Here) response for the re-
 INVITE.  This response SHOULD include a Warning header field.
 If a UAS generates a 2xx response and never receives an ACK, it
 SHOULD generate a BYE to terminate the dialog.
 A UAS MAY choose not to generate 180 (Ringing) responses for a re-
 INVITE because UACs do not typically render this information to the
 user.  For the same reason, UASs MAY choose not to use an Alert-Info
 header field or a body with Content-Disposition "alert" in responses
 to a re-INVITE.
 A UAS providing an offer in a 2xx (because the INVITE did not contain
 an offer) SHOULD construct the offer as if the UAS were making a
 brand new call, subject to the constraints of sending an offer that
 updates an existing session, as described in [13] in the case of SDP.
 Specifically, this means that it SHOULD include as many media formats
 and media types that the UA is willing to support.  The UAS MUST
 ensure that the session description overlaps with its previous
 session description in media formats, transports, or other parameters
 that require support from the peer.  This is to avoid the need for
 the peer to reject the session description.  If, however, it is
 unacceptable to the UAC, the UAC SHOULD generate an answer with a
 valid session description, and then send a BYE to terminate the
 session.

15 Terminating a Session

 This section describes the procedures for terminating a session
 established by SIP.  The state of the session and the state of the
 dialog are very closely related.  When a session is initiated with an
 INVITE, each 1xx or 2xx response from a distinct UAS creates a
 dialog, and if that response completes the offer/answer exchange, it
 also creates a session.  As a result, each session is "associated"
 with a single dialog - the one which resulted in its creation.  If an
 initial INVITE generates a non-2xx final response, that terminates
 all sessions (if any) and all dialogs (if any) that were created
 through responses to the request.  By virtue of completing the
 transaction, a non-2xx final response also prevents further sessions
 from being created as a result of the INVITE.  The BYE request is
 used to terminate a specific session or attempted session.  In this
 case, the specific session is the one with the peer UA on the other
 side of the dialog.  When a BYE is received on a dialog, any session
 associated with that dialog SHOULD terminate.  A UA MUST NOT send a
 BYE outside of a dialog.  The caller's UA MAY send a BYE for either
 confirmed or early dialogs, and the callee's UA MAY send a BYE on
 confirmed dialogs, but MUST NOT send a BYE on early dialogs.

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 However, the callee's UA MUST NOT send a BYE on a confirmed dialog
 until it has received an ACK for its 2xx response or until the server
 transaction times out.  If no SIP extensions have defined other
 application layer states associated with the dialog, the BYE also
 terminates the dialog.
 The impact of a non-2xx final response to INVITE on dialogs and
 sessions makes the use of CANCEL attractive.  The CANCEL attempts to
 force a non-2xx response to the INVITE (in particular, a 487).
 Therefore, if a UAC wishes to give up on its call attempt entirely,
 it can send a CANCEL.  If the INVITE results in 2xx final response(s)
 to the INVITE, this means that a UAS accepted the invitation while
 the CANCEL was in progress.  The UAC MAY continue with the sessions
 established by any 2xx responses, or MAY terminate them with BYE.
    The notion of "hanging up" is not well defined within SIP.  It is
    specific to a particular, albeit common, user interface.
    Typically, when the user hangs up, it indicates a desire to
    terminate the attempt to establish a session, and to terminate any
    sessions already created.  For the caller's UA, this would imply a
    CANCEL request if the initial INVITE has not generated a final
    response, and a BYE to all confirmed dialogs after a final
    response.  For the callee's UA, it would typically imply a BYE;
    presumably, when the user picked up the phone, a 2xx was
    generated, and so hanging up would result in a BYE after the ACK
    is received.  This does not mean a user cannot hang up before
    receipt of the ACK, it just means that the software in his phone
    needs to maintain state for a short while in order to clean up
    properly.  If the particular UI allows for the user to reject a
    call before its answered, a 403 (Forbidden) is a good way to
    express that.  As per the rules above, a BYE can't be sent.

15.1 Terminating a Session with a BYE Request

15.1.1 UAC Behavior

 A BYE request is constructed as would any other request within a
 dialog, as described in Section 12.
 Once the BYE is constructed, the UAC core creates a new non-INVITE
 client transaction, and passes it the BYE request.  The UAC MUST
 consider the session terminated (and therefore stop sending or
 listening for media) as soon as the BYE request is passed to the
 client transaction.  If the response for the BYE is a 481
 (Call/Transaction Does Not Exist) or a 408 (Request Timeout) or no

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 response at all is received for the BYE (that is, a timeout is
 returned by the client transaction), the UAC MUST consider the
 session and the dialog terminated.

15.1.2 UAS Behavior

 A UAS first processes the BYE request according to the general UAS
 processing described in Section 8.2.  A UAS core receiving a BYE
 request checks if it matches an existing dialog.  If the BYE does not
 match an existing dialog, the UAS core SHOULD generate a 481
 (Call/Transaction Does Not Exist) response and pass that to the
 server transaction.
    This rule means that a BYE sent without tags by a UAC will be
    rejected.  This is a change from RFC 2543, which allowed BYE
    without tags.
 A UAS core receiving a BYE request for an existing dialog MUST follow
 the procedures of Section 12.2.2 to process the request.  Once done,
 the UAS SHOULD terminate the session (and therefore stop sending and
 listening for media).  The only case where it can elect not to are
 multicast sessions, where participation is possible even if the other
 participant in the dialog has terminated its involvement in the
 session.  Whether or not it ends its participation on the session,
 the UAS core MUST generate a 2xx response to the BYE, and MUST pass
 that to the server transaction for transmission.
 The UAS MUST still respond to any pending requests received for that
 dialog.  It is RECOMMENDED that a 487 (Request Terminated) response
 be generated to those pending requests.

16 Proxy Behavior

16.1 Overview

 SIP proxies are elements that route SIP requests to user agent
 servers and SIP responses to user agent clients.  A request may
 traverse several proxies on its way to a UAS.  Each will make routing
 decisions, modifying the request before forwarding it to the next
 element.  Responses will route through the same set of proxies
 traversed by the request in the reverse order.
 Being a proxy is a logical role for a SIP element.  When a request
 arrives, an element that can play the role of a proxy first decides
 if it needs to respond to the request on its own.  For instance, the
 request may be malformed or the element may need credentials from the
 client before acting as a proxy.  The element MAY respond with any

Rosenberg, et. al. Standards Track [Page 91] RFC 3261 SIP: Session Initiation Protocol June 2002

 appropriate error code.  When responding directly to a request, the
 element is playing the role of a UAS and MUST behave as described in
 Section 8.2.
 A proxy can operate in either a stateful or stateless mode for each
 new request.  When stateless, a proxy acts as a simple forwarding
 element.  It forwards each request downstream to a single element
 determined by making a targeting and routing decision based on the
 request.  It simply forwards every response it receives upstream.  A
 stateless proxy discards information about a message once the message
 has been forwarded.  A stateful proxy remembers information
 (specifically, transaction state) about each incoming request and any
 requests it sends as a result of processing the incoming request.  It
 uses this information to affect the processing of future messages
 associated with that request.  A stateful proxy MAY choose to "fork"
 a request, routing it to multiple destinations.  Any request that is
 forwarded to more than one location MUST be handled statefully.
 In some circumstances, a proxy MAY forward requests using stateful
 transports (such as TCP) without being transaction-stateful.  For
 instance, a proxy MAY forward a request from one TCP connection to
 another transaction statelessly as long as it places enough
 information in the message to be able to forward the response down
 the same connection the request arrived on.  Requests forwarded
 between different types of transports where the proxy's TU must take
 an active role in ensuring reliable delivery on one of the transports
 MUST be forwarded transaction statefully.
 A stateful proxy MAY transition to stateless operation at any time
 during the processing of a request, so long as it did not do anything
 that would otherwise prevent it from being stateless initially
 (forking, for example, or generation of a 100 response).  When
 performing such a transition, all state is simply discarded.  The
 proxy SHOULD NOT initiate a CANCEL request.
 Much of the processing involved when acting statelessly or statefully
 for a request is identical.  The next several subsections are written
 from the point of view of a stateful proxy.  The last section calls
 out those places where a stateless proxy behaves differently.

16.2 Stateful Proxy

 When stateful, a proxy is purely a SIP transaction processing engine.
 Its behavior is modeled here in terms of the server and client
 transactions defined in Section 17.  A stateful proxy has a server
 transaction associated with one or more client transactions by a
 higher layer proxy processing component (see figure 3), known as a
 proxy core.  An incoming request is processed by a server

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 transaction.  Requests from the server transaction are passed to a
 proxy core.  The proxy core determines where to route the request,
 choosing one or more next-hop locations.  An outgoing request for
 each next-hop location is processed by its own associated client
 transaction.  The proxy core collects the responses from the client
 transactions and uses them to send responses to the server
 transaction.
 A stateful proxy creates a new server transaction for each new
 request received.  Any retransmissions of the request will then be
 handled by that server transaction per Section 17.  The proxy core
 MUST behave as a UAS with respect to sending an immediate provisional
 on that server transaction (such as 100 Trying) as described in
 Section 8.2.6.  Thus, a stateful proxy SHOULD NOT generate 100
 (Trying) responses to non-INVITE requests.
 This is a model of proxy behavior, not of software.  An
 implementation is free to take any approach that replicates the
 external behavior this model defines.
 For all new requests, including any with unknown methods, an element
 intending to proxy the request MUST:
    1. Validate the request (Section 16.3)
    2. Preprocess routing information (Section 16.4)
    3. Determine target(s) for the request (Section 16.5)
          +--------------------+
          |                    | +---+
          |                    | | C |
          |                    | | T |
          |                    | +---+
    +---+ |       Proxy        | +---+   CT = Client Transaction
    | S | |  "Higher" Layer    | | C |
    | T | |                    | | T |   ST = Server Transaction
    +---+ |                    | +---+
          |                    | +---+
          |                    | | C |
          |                    | | T |
          |                    | +---+
          +--------------------+
             Figure 3: Stateful Proxy Model

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    4. Forward the request to each target (Section 16.6)
    5. Process all responses (Section 16.7)

16.3 Request Validation

 Before an element can proxy a request, it MUST verify the message's
 validity.  A valid message must pass the following checks:
    1. Reasonable Syntax
    2. URI scheme
    3. Max-Forwards
    4. (Optional) Loop Detection
    5. Proxy-Require
    6. Proxy-Authorization
 If any of these checks fail, the element MUST behave as a user agent
 server (see Section 8.2) and respond with an error code.
 Notice that a proxy is not required to detect merged requests and
 MUST NOT treat merged requests as an error condition.  The endpoints
 receiving the requests will resolve the merge as described in Section
 8.2.2.2.
 1. Reasonable syntax check
    The request MUST be well-formed enough to be handled with a server
    transaction.  Any components involved in the remainder of these
    Request Validation steps or the Request Forwarding section MUST be
    well-formed.  Any other components, well-formed or not, SHOULD be
    ignored and remain unchanged when the message is forwarded.  For
    instance, an element would not reject a request because of a
    malformed Date header field.  Likewise, a proxy would not remove a
    malformed Date header field before forwarding a request.
    This protocol is designed to be extended.  Future extensions may
    define new methods and header fields at any time.  An element MUST
    NOT refuse to proxy a request because it contains a method or
    header field it does not know about.

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 2. URI scheme check
    If the Request-URI has a URI whose scheme is not understood by the
    proxy, the proxy SHOULD reject the request with a 416 (Unsupported
    URI Scheme) response.
 3. Max-Forwards check
    The Max-Forwards header field (Section 20.22) is used to limit the
    number of elements a SIP request can traverse.
    If the request does not contain a Max-Forwards header field, this
    check is passed.
    If the request contains a Max-Forwards header field with a field
    value greater than zero, the check is passed.
    If the request contains a Max-Forwards header field with a field
    value of zero (0), the element MUST NOT forward the request.  If
    the request was for OPTIONS, the element MAY act as the final
    recipient and respond per Section 11.  Otherwise, the element MUST
    return a 483 (Too many hops) response.
 4. Optional Loop Detection check
    An element MAY check for forwarding loops before forwarding a
    request.  If the request contains a Via header field with a sent-
    by value that equals a value placed into previous requests by the
    proxy, the request has been forwarded by this element before.  The
    request has either looped or is legitimately spiraling through the
    element.  To determine if the request has looped, the element MAY
    perform the branch parameter calculation described in Step 8 of
    Section 16.6 on this message and compare it to the parameter
    received in that Via header field.  If the parameters match, the
    request has looped.  If they differ, the request is spiraling, and
    processing continues.  If a loop is detected, the element MAY
    return a 482 (Loop Detected) response.
 5. Proxy-Require check
    Future extensions to this protocol may introduce features that
    require special handling by proxies.  Endpoints will include a
    Proxy-Require header field in requests that use these features,
    telling the proxy not to process the request unless the feature is
    understood.

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    If the request contains a Proxy-Require header field (Section
    20.29) with one or more option-tags this element does not
    understand, the element MUST return a 420 (Bad Extension)
    response.  The response MUST include an Unsupported (Section
    20.40) header field listing those option-tags the element did not
    understand.
 6. Proxy-Authorization check
    If an element requires credentials before forwarding a request,
    the request MUST be inspected as described in Section 22.3.  That
    section also defines what the element must do if the inspection
    fails.

16.4 Route Information Preprocessing

 The proxy MUST inspect the Request-URI of the request.  If the
 Request-URI of the request contains a value this proxy previously
 placed into a Record-Route header field (see Section 16.6 item 4),
 the proxy MUST replace the Request-URI in the request with the last
 value from the Route header field, and remove that value from the
 Route header field.  The proxy MUST then proceed as if it received
 this modified request.
    This will only happen when the element sending the request to the
    proxy (which may have been an endpoint) is a strict router.  This
    rewrite on receive is necessary to enable backwards compatibility
    with those elements.  It also allows elements following this
    specification to preserve the Request-URI through strict-routing
    proxies (see Section 12.2.1.1).
    This requirement does not obligate a proxy to keep state in order
    to detect URIs it previously placed in Record-Route header fields.
    Instead, a proxy need only place enough information in those URIs
    to recognize them as values it provided when they later appear.
 If the Request-URI contains a maddr parameter, the proxy MUST check
 to see if its value is in the set of addresses or domains the proxy
 is configured to be responsible for.  If the Request-URI has a maddr
 parameter with a value the proxy is responsible for, and the request
 was received using the port and transport indicated (explicitly or by
 default) in the Request-URI, the proxy MUST strip the maddr and any
 non-default port or transport parameter and continue processing as if
 those values had not been present in the request.

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    A request may arrive with a maddr matching the proxy, but on a
    port or transport different from that indicated in the URI.  Such
    a request needs to be forwarded to the proxy using the indicated
    port and transport.
 If the first value in the Route header field indicates this proxy,
 the proxy MUST remove that value from the request.

16.5 Determining Request Targets

 Next, the proxy calculates the target(s) of the request.  The set of
 targets will either be predetermined by the contents of the request
 or will be obtained from an abstract location service.  Each target
 in the set is represented as a URI.
 If the Request-URI of the request contains an maddr parameter, the
 Request-URI MUST be placed into the target set as the only target
 URI, and the proxy MUST proceed to Section 16.6.
 If the domain of the Request-URI indicates a domain this element is
 not responsible for, the Request-URI MUST be placed into the target
 set as the only target, and the element MUST proceed to the task of
 Request Forwarding (Section 16.6).
    There are many circumstances in which a proxy might receive a
    request for a domain it is not responsible for.  A firewall proxy
    handling outgoing calls (the way HTTP proxies handle outgoing
    requests) is an example of where this is likely to occur.
 If the target set for the request has not been predetermined as
 described above, this implies that the element is responsible for the
 domain in the Request-URI, and the element MAY use whatever mechanism
 it desires to determine where to send the request.  Any of these
 mechanisms can be modeled as accessing an abstract Location Service.
 This may consist of obtaining information from a location service
 created by a SIP Registrar, reading a database, consulting a presence
 server, utilizing other protocols, or simply performing an
 algorithmic substitution on the Request-URI.  When accessing the
 location service constructed by a registrar, the Request-URI MUST
 first be canonicalized as described in Section 10.3 before being used
 as an index.  The output of these mechanisms is used to construct the
 target set.
 If the Request-URI does not provide sufficient information for the
 proxy to determine the target set, it SHOULD return a 485 (Ambiguous)
 response.  This response SHOULD contain a Contact header field
 containing URIs of new addresses to be tried.  For example, an INVITE

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 to sip:John.Smith@company.com may be ambiguous at a proxy whose
 location service has multiple John Smiths listed.  See Section
 21.4.23 for details.
 Any information in or about the request or the current environment of
 the element MAY be used in the construction of the target set.  For
 instance, different sets may be constructed depending on contents or
 the presence of header fields and bodies, the time of day of the
 request's arrival, the interface on which the request arrived,
 failure of previous requests, or even the element's current level of
 utilization.
 As potential targets are located through these services, their URIs
 are added to the target set.  Targets can only be placed in the
 target set once.  If a target URI is already present in the set
 (based on the definition of equality for the URI type), it MUST NOT
 be added again.
 A proxy MUST NOT add additional targets to the target set if the
 Request-URI of the original request does not indicate a resource this
 proxy is responsible for.
    A proxy can only change the Request-URI of a request during
    forwarding if it is responsible for that URI.  If the proxy is not
    responsible for that URI, it will not recurse on 3xx or 416
    responses as described below.
 If the Request-URI of the original request indicates a resource this
 proxy is responsible for, the proxy MAY continue to add targets to
 the set after beginning Request Forwarding.  It MAY use any
 information obtained during that processing to determine new targets.
 For instance, a proxy may choose to incorporate contacts obtained in
 a redirect response (3xx) into the target set.  If a proxy uses a
 dynamic source of information while building the target set (for
 instance, if it consults a SIP Registrar), it SHOULD monitor that
 source for the duration of processing the request.  New locations
 SHOULD be added to the target set as they become available.  As
 above, any given URI MUST NOT be added to the set more than once.
    Allowing a URI to be added to the set only once reduces
    unnecessary network traffic, and in the case of incorporating
    contacts from redirect requests prevents infinite recursion.
 For example, a trivial location service is a "no-op", where the
 target URI is equal to the incoming request URI.  The request is sent
 to a specific next hop proxy for further processing.  During request

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 forwarding of Section 16.6, Item 6, the identity of that next hop,
 expressed as a SIP or SIPS URI, is inserted as the top-most Route
 header field value into the request.
 If the Request-URI indicates a resource at this proxy that does not
 exist, the proxy MUST return a 404 (Not Found) response.
 If the target set remains empty after applying all of the above, the
 proxy MUST return an error response, which SHOULD be the 480
 (Temporarily Unavailable) response.

16.6 Request Forwarding

 As soon as the target set is non-empty, a proxy MAY begin forwarding
 the request.  A stateful proxy MAY process the set in any order.  It
 MAY process multiple targets serially, allowing each client
 transaction to complete before starting the next.  It MAY start
 client transactions with every target in parallel.  It also MAY
 arbitrarily divide the set into groups, processing the groups
 serially and processing the targets in each group in parallel.
 A common ordering mechanism is to use the qvalue parameter of targets
 obtained from Contact header fields (see Section 20.10).  Targets are
 processed from highest qvalue to lowest.  Targets with equal qvalues
 may be processed in parallel.
 A stateful proxy must have a mechanism to maintain the target set as
 responses are received and associate the responses to each forwarded
 request with the original request.  For the purposes of this model,
 this mechanism is a "response context" created by the proxy layer
 before forwarding the first request.
 For each target, the proxy forwards the request following these
 steps:
    1.  Make a copy of the received request
    2.  Update the Request-URI
    3.  Update the Max-Forwards header field
    4.  Optionally add a Record-route header field value
    5.  Optionally add additional header fields
    6.  Postprocess routing information
    7.  Determine the next-hop address, port, and transport

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    8.  Add a Via header field value
    9.  Add a Content-Length header field if necessary
    10. Forward the new request
    11. Set timer C
 Each of these steps is detailed below:
    1. Copy request
       The proxy starts with a copy of the received request.  The copy
       MUST initially contain all of the header fields from the
       received request.  Fields not detailed in the processing
       described below MUST NOT be removed.  The copy SHOULD maintain
       the ordering of the header fields as in the received request.
       The proxy MUST NOT reorder field values with a common field
       name (See Section 7.3.1).  The proxy MUST NOT add to, modify,
       or remove the message body.
       An actual implementation need not perform a copy; the primary
       requirement is that the processing for each next hop begin with
       the same request.
    2. Request-URI
       The Request-URI in the copy's start line MUST be replaced with
       the URI for this target.  If the URI contains any parameters
       not allowed in a Request-URI, they MUST be removed.
       This is the essence of a proxy's role.  This is the mechanism
       through which a proxy routes a request toward its destination.
       In some circumstances, the received Request-URI is placed into
       the target set without being modified.  For that target, the
       replacement above is effectively a no-op.
    3. Max-Forwards
       If the copy contains a Max-Forwards header field, the proxy
       MUST decrement its value by one (1).
       If the copy does not contain a Max-Forwards header field, the
       proxy MUST add one with a field value, which SHOULD be 70.
       Some existing UAs will not provide a Max-Forwards header field
       in a request.

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    4. Record-Route
       If this proxy wishes to remain on the path of future requests
       in a dialog created by this request (assuming the request
       creates a dialog), it MUST insert a Record-Route header field
       value into the copy before any existing Record-Route header
       field values, even if a Route header field is already present.
       Requests establishing a dialog may contain a preloaded Route
       header field.
       If this request is already part of a dialog, the proxy SHOULD
       insert a Record-Route header field value if it wishes to remain
       on the path of future requests in the dialog.  In normal
       endpoint operation as described in Section 12, these Record-
       Route header field values will not have any effect on the route
       sets used by the endpoints.
       The proxy will remain on the path if it chooses to not insert a
       Record-Route header field value into requests that are already
       part of a dialog.  However, it would be removed from the path
       when an endpoint that has failed reconstitutes the dialog.
       A proxy MAY insert a Record-Route header field value into any
       request.  If the request does not initiate a dialog, the
       endpoints will ignore the value.  See Section 12 for details on
       how endpoints use the Record-Route header field values to
       construct Route header fields.
       Each proxy in the path of a request chooses whether to add a
       Record-Route header field value independently - the presence of
       a Record-Route header field in a request does not obligate this
       proxy to add a value.
       The URI placed in the Record-Route header field value MUST be a
       SIP or SIPS URI.  This URI MUST contain an lr parameter (see
       Section 19.1.1).  This URI MAY be different for each
       destination the request is forwarded to.  The URI SHOULD NOT
       contain the transport parameter unless the proxy has knowledge
       (such as in a private network) that the next downstream element
       that will be in the path of subsequent requests supports that
       transport.
       The URI this proxy provides will be used by some other element
       to make a routing decision.  This proxy, in general, has no way
       of knowing the capabilities of that element, so it must
       restrict itself to the mandatory elements of a SIP
       implementation: SIP URIs and either the TCP or UDP transports.

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       The URI placed in the Record-Route header field MUST resolve to
       the element inserting it (or a suitable stand-in) when the
       server location procedures of [4] are applied to it, so that
       subsequent requests reach the same SIP element.  If the
       Request-URI contains a SIPS URI, or the topmost Route header
       field value (after the post processing of bullet 6) contains a
       SIPS URI, the URI placed into the Record-Route header field
       MUST be a SIPS URI.  Furthermore, if the request was not
       received over TLS, the proxy MUST insert a Record-Route header
       field.  In a similar fashion, a proxy that receives a request
       over TLS, but generates a request without a SIPS URI in the
       Request-URI or topmost Route header field value (after the post
       processing of bullet 6), MUST insert a Record-Route header
       field that is not a SIPS URI.
       A proxy at a security perimeter must remain on the perimeter
       throughout the dialog.
       If the URI placed in the Record-Route header field needs to be
       rewritten when it passes back through in a response, the URI
       MUST be distinct enough to locate at that time.  (The request
       may spiral through this proxy, resulting in more than one
       Record-Route header field value being added).  Item 8 of
       Section 16.7 recommends a mechanism to make the URI
       sufficiently distinct.
       The proxy MAY include parameters in the Record-Route header
       field value.  These will be echoed in some responses to the
       request such as the 200 (OK) responses to INVITE.  Such
       parameters may be useful for keeping state in the message
       rather than the proxy.
       If a proxy needs to be in the path of any type of dialog (such
       as one straddling a firewall), it SHOULD add a Record-Route
       header field value to every request with a method it does not
       understand since that method may have dialog semantics.
       The URI a proxy places into a Record-Route header field is only
       valid for the lifetime of any dialog created by the transaction
       in which it occurs.  A dialog-stateful proxy, for example, MAY
       refuse to accept future requests with that value in the
       Request-URI after the dialog has terminated.  Non-dialog-
       stateful proxies, of course, have no concept of when the dialog
       has terminated, but they MAY encode enough information in the
       value to compare it against the dialog identifier of future
       requests and MAY reject requests not matching that information.
       Endpoints MUST NOT use a URI obtained from a Record-Route
       header field outside the dialog in which it was provided.  See

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       Section 12 for more information on an endpoint's use of
       Record-Route header fields.
       Record-routing may be required by certain services where the
       proxy needs to observe all messages in a dialog.  However, it
       slows down processing and impairs scalability and thus proxies
       should only record-route if required for a particular service.
       The Record-Route process is designed to work for any SIP
       request that initiates a dialog.  INVITE is the only such
       request in this specification, but extensions to the protocol
       MAY define others.
    5. Add Additional Header Fields
       The proxy MAY add any other appropriate header fields to the
       copy at this point.
    6. Postprocess routing information
       A proxy MAY have a local policy that mandates that a request
       visit a specific set of proxies before being delivered to the
       destination.  A proxy MUST ensure that all such proxies are
       loose routers.  Generally, this can only be known with
       certainty if the proxies are within the same administrative
       domain.  This set of proxies is represented by a set of URIs
       (each of which contains the lr parameter).  This set MUST be
       pushed into the Route header field of the copy ahead of any
       existing values, if present.  If the Route header field is
       absent, it MUST be added, containing that list of URIs.
       If the proxy has a local policy that mandates that the request
       visit one specific proxy, an alternative to pushing a Route
       value into the Route header field is to bypass the forwarding
       logic of item 10 below, and instead just send the request to
       the address, port, and transport for that specific proxy.  If
       the request has a Route header field, this alternative MUST NOT
       be used unless it is known that next hop proxy is a loose
       router.  Otherwise, this approach MAY be used, but the Route
       insertion mechanism above is preferred for its robustness,
       flexibility, generality and consistency of operation.
       Furthermore, if the Request-URI contains a SIPS URI, TLS MUST
       be used to communicate with that proxy.
       If the copy contains a Route header field, the proxy MUST
       inspect the URI in its first value.  If that URI does not
       contain an lr parameter, the proxy MUST modify the copy as
       follows:

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  1. The proxy MUST place the Request-URI into the Route header

field as the last value.

  1. The proxy MUST then place the first Route header field value

into the Request-URI and remove that value from the Route

          header field.
       Appending the Request-URI to the Route header field is part of
       a mechanism used to pass the information in that Request-URI
       through strict-routing elements.  "Popping" the first Route
       header field value into the Request-URI formats the message the
       way a strict-routing element expects to receive it (with its
       own URI in the Request-URI and the next location to visit in
       the first Route header field value).
    7. Determine Next-Hop Address, Port, and Transport
       The proxy MAY have a local policy to send the request to a
       specific IP address, port, and transport, independent of the
       values of the Route and Request-URI.  Such a policy MUST NOT be
       used if the proxy is not certain that the IP address, port, and
       transport correspond to a server that is a loose router.
       However, this mechanism for sending the request through a
       specific next hop is NOT RECOMMENDED; instead a Route header
       field should be used for that purpose as described above.
       In the absence of such an overriding mechanism, the proxy
       applies the procedures listed in [4] as follows to determine
       where to send the request.  If the proxy has reformatted the
       request to send to a strict-routing element as described in
       step 6 above, the proxy MUST apply those procedures to the
       Request-URI of the request.  Otherwise, the proxy MUST apply
       the procedures to the first value in the Route header field, if
       present, else the Request-URI.  The procedures will produce an
       ordered set of (address, port, transport) tuples.
       Independently of which URI is being used as input to the
       procedures of [4], if the Request-URI specifies a SIPS
       resource, the proxy MUST follow the procedures of [4] as if the
       input URI were a SIPS URI.
       As described in [4], the proxy MUST attempt to deliver the
       message to the first tuple in that set, and proceed through the
       set in order until the delivery attempt succeeds.
       For each tuple attempted, the proxy MUST format the message as
       appropriate for the tuple and send the request using a new
       client transaction as detailed in steps 8 through 10.

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       Since each attempt uses a new client transaction, it represents
       a new branch.  Thus, the branch parameter provided with the Via
       header field inserted in step 8 MUST be different for each
       attempt.
       If the client transaction reports failure to send the request
       or a timeout from its state machine, the proxy continues to the
       next address in that ordered set.  If the ordered set is
       exhausted, the request cannot be forwarded to this element in
       the target set.  The proxy does not need to place anything in
       the response context, but otherwise acts as if this element of
       the target set returned a 408 (Request Timeout) final response.
    8. Add a Via header field value
       The proxy MUST insert a Via header field value into the copy
       before the existing Via header field values.  The construction
       of this value follows the same guidelines of Section 8.1.1.7.
       This implies that the proxy will compute its own branch
       parameter, which will be globally unique for that branch, and
       contain the requisite magic cookie. Note that this implies that
       the branch parameter will be different for different instances
       of a spiraled or looped request through a proxy.
       Proxies choosing to detect loops have an additional constraint
       in the value they use for construction of the branch parameter.
       A proxy choosing to detect loops SHOULD create a branch
       parameter separable into two parts by the implementation.  The
       first part MUST satisfy the constraints of Section 8.1.1.7 as
       described above.  The second is used to perform loop detection
       and distinguish loops from spirals.
       Loop detection is performed by verifying that, when a request
       returns to a proxy, those fields having an impact on the
       processing of the request have not changed.  The value placed
       in this part of the branch parameter SHOULD reflect all of
       those fields (including any Route, Proxy-Require and Proxy-
       Authorization header fields).  This is to ensure that if the
       request is routed back to the proxy and one of those fields
       changes, it is treated as a spiral and not a loop (see Section
       16.3).  A common way to create this value is to compute a
       cryptographic hash of the To tag, From tag, Call-ID header
       field, the Request-URI of the request received (before
       translation), the topmost Via header, and the sequence number
       from the CSeq header field, in addition to any Proxy-Require
       and Proxy-Authorization header fields that may be present.  The

Rosenberg, et. al. Standards Track [Page 105] RFC 3261 SIP: Session Initiation Protocol June 2002

       algorithm used to compute the hash is implementation-dependent,
       but MD5 (RFC 1321 [35]), expressed in hexadecimal, is a
       reasonable choice.  (Base64 is not permissible for a token.)
       If a proxy wishes to detect loops, the "branch" parameter it
       supplies MUST depend on all information affecting processing of
       a request, including the incoming Request-URI and any header
       fields affecting the request's admission or routing.  This is
       necessary to distinguish looped requests from requests whose
       routing parameters have changed before returning to this
       server.
       The request method MUST NOT be included in the calculation of
       the branch parameter.  In particular, CANCEL and ACK requests
       (for non-2xx responses) MUST have the same branch value as the
       corresponding request they cancel or acknowledge.  The branch
       parameter is used in correlating those requests at the server
       handling them (see Sections 17.2.3 and 9.2).
    9. Add a Content-Length header field if necessary
       If the request will be sent to the next hop using a stream-
       based transport and the copy contains no Content-Length header
       field, the proxy MUST insert one with the correct value for the
       body of the request (see Section 20.14).
    10. Forward Request
       A stateful proxy MUST create a new client transaction for this
       request as described in Section 17.1 and instructs the
       transaction to send the request using the address, port and
       transport determined in step 7.
    11. Set timer C
       In order to handle the case where an INVITE request never
       generates a final response, the TU uses a timer which is called
       timer C.  Timer C MUST be set for each client transaction when
       an INVITE request is proxied.  The timer MUST be larger than 3
       minutes.  Section 16.7 bullet 2 discusses how this timer is
       updated with provisional responses, and Section 16.8 discusses
       processing when it fires.

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16.7 Response Processing

 When a response is received by an element, it first tries to locate a
 client transaction (Section 17.1.3) matching the response.  If none
 is found, the element MUST process the response (even if it is an
 informational response) as a stateless proxy (described below).  If a
 match is found, the response is handed to the client transaction.
    Forwarding responses for which a client transaction (or more
    generally any knowledge of having sent an associated request) is
    not found improves robustness.  In particular, it ensures that
    "late" 2xx responses to INVITE requests are forwarded properly.
 As client transactions pass responses to the proxy layer, the
 following processing MUST take place:
    1.  Find the appropriate response context
    2.  Update timer C for provisional responses
    3.  Remove the topmost Via
    4.  Add the response to the response context
    5.  Check to see if this response should be forwarded immediately
    6.  When necessary, choose the best final response from the
        response context
 If no final response has been forwarded after every client
 transaction associated with the response context has been terminated,
 the proxy must choose and forward the "best" response from those it
 has seen so far.
 The following processing MUST be performed on each response that is
 forwarded.  It is likely that more than one response to each request
 will be forwarded: at least each provisional and one final response.
    7.  Aggregate authorization header field values if necessary
    8.  Optionally rewrite Record-Route header field values
    9.  Forward the response
    10. Generate any necessary CANCEL requests

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 Each of the above steps are detailed below:
    1.  Find Context
       The proxy locates the "response context" it created before
       forwarding the original request using the key described in
       Section 16.6.  The remaining processing steps take place in
       this context.
    2.  Update timer C for provisional responses
       For an INVITE transaction, if the response is a provisional
       response with status codes 101 to 199 inclusive (i.e., anything
       but 100), the proxy MUST reset timer C for that client
       transaction.  The timer MAY be reset to a different value, but
       this value MUST be greater than 3 minutes.
    3.  Via
       The proxy removes the topmost Via header field value from the
       response.
       If no Via header field values remain in the response, the
       response was meant for this element and MUST NOT be forwarded.
       The remainder of the processing described in this section is
       not performed on this message, the UAC processing rules
       described in Section 8.1.3 are followed instead (transport
       layer processing has already occurred).
       This will happen, for instance, when the element generates
       CANCEL requests as described in Section 10.
    4.  Add response to context
       Final responses received are stored in the response context
       until a final response is generated on the server transaction
       associated with this context.  The response may be a candidate
       for the best final response to be returned on that server
       transaction.  Information from this response may be needed in
       forming the best response, even if this response is not chosen.
       If the proxy chooses to recurse on any contacts in a 3xx
       response by adding them to the target set, it MUST remove them
       from the response before adding the response to the response
       context.  However, a proxy SHOULD NOT recurse to a non-SIPS URI
       if the Request-URI of the original request was a SIPS URI.  If

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       the proxy recurses on all of the contacts in a 3xx response,
       the proxy SHOULD NOT add the resulting contactless response to
       the response context.
       Removing the contact before adding the response to the response
       context prevents the next element upstream from retrying a
       location this proxy has already attempted.
       3xx responses may contain a mixture of SIP, SIPS, and non-SIP
       URIs.  A proxy may choose to recurse on the SIP and SIPS URIs
       and place the remainder into the response context to be
       returned, potentially in the final response.
       If a proxy receives a 416 (Unsupported URI Scheme) response to
       a request whose Request-URI scheme was not SIP, but the scheme
       in the original received request was SIP or SIPS (that is, the
       proxy changed the scheme from SIP or SIPS to something else
       when it proxied a request), the proxy SHOULD add a new URI to
       the target set.  This URI SHOULD be a SIP URI version of the
       non-SIP URI that was just tried.  In the case of the tel URL,
       this is accomplished by placing the telephone-subscriber part
       of the tel URL into the user part of the SIP URI, and setting
       the hostpart to the domain where the prior request was sent.
       See Section 19.1.6 for more detail on forming SIP URIs from tel
       URLs.
       As with a 3xx response, if a proxy "recurses" on the 416 by
       trying a SIP or SIPS URI instead, the 416 response SHOULD NOT
       be added to the response context.
    5.  Check response for forwarding
       Until a final response has been sent on the server transaction,
       the following responses MUST be forwarded immediately:
  1. Any provisional response other than 100 (Trying)
  1. Any 2xx response
       If a 6xx response is received, it is not immediately forwarded,
       but the stateful proxy SHOULD cancel all client pending
       transactions as described in Section 10, and it MUST NOT create
       any new branches in this context.
       This is a change from RFC 2543, which mandated that the proxy
       was to forward the 6xx response immediately.  For an INVITE
       transaction, this approach had the problem that a 2xx response
       could arrive on another branch, in which case the proxy would

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       have to forward the 2xx.  The result was that the UAC could
       receive a 6xx response followed by a 2xx response, which should
       never be allowed to happen.  Under the new rules, upon
       receiving a 6xx, a proxy will issue a CANCEL request, which
       will generally result in 487 responses from all outstanding
       client transactions, and then at that point the 6xx is
       forwarded upstream.
       After a final response has been sent on the server transaction,
       the following responses MUST be forwarded immediately:
  1. Any 2xx response to an INVITE request
       A stateful proxy MUST NOT immediately forward any other
       responses.  In particular, a stateful proxy MUST NOT forward
       any 100 (Trying) response.  Those responses that are candidates
       for forwarding later as the "best" response have been gathered
       as described in step "Add Response to Context".
       Any response chosen for immediate forwarding MUST be processed
       as described in steps "Aggregate Authorization Header Field
       Values" through "Record-Route".
       This step, combined with the next, ensures that a stateful
       proxy will forward exactly one final response to a non-INVITE
       request, and either exactly one non-2xx response or one or more
       2xx responses to an INVITE request.
    6.  Choosing the best response
       A stateful proxy MUST send a final response to a response
       context's server transaction if no final responses have been
       immediately forwarded by the above rules and all client
       transactions in this response context have been terminated.
       The stateful proxy MUST choose the "best" final response among
       those received and stored in the response context.
       If there are no final responses in the context, the proxy MUST
       send a 408 (Request Timeout) response to the server
       transaction.
       Otherwise, the proxy MUST forward a response from the responses
       stored in the response context.  It MUST choose from the 6xx
       class responses if any exist in the context.  If no 6xx class
       responses are present, the proxy SHOULD choose from the lowest
       response class stored in the response context.  The proxy MAY
       select any response within that chosen class.  The proxy SHOULD

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       give preference to responses that provide information affecting
       resubmission of this request, such as 401, 407, 415, 420, and
       484 if the 4xx class is chosen.
       A proxy which receives a 503 (Service Unavailable) response
       SHOULD NOT forward it upstream unless it can determine that any
       subsequent requests it might proxy will also generate a 503.
       In other words, forwarding a 503 means that the proxy knows it
       cannot service any requests, not just the one for the Request-
       URI in the request which generated the 503.  If the only
       response that was received is a 503, the proxy SHOULD generate
       a 500 response and forward that upstream.
       The forwarded response MUST be processed as described in steps
       "Aggregate Authorization Header Field Values" through "Record-
       Route".
       For example, if a proxy forwarded a request to 4 locations, and
       received 503, 407, 501, and 404 responses, it may choose to
       forward the 407 (Proxy Authentication Required) response.
       1xx and 2xx responses may be involved in the establishment of
       dialogs.  When a request does not contain a To tag, the To tag
       in the response is used by the UAC to distinguish multiple
       responses to a dialog creating request.  A proxy MUST NOT
       insert a tag into the To header field of a 1xx or 2xx response
       if the request did not contain one.  A proxy MUST NOT modify
       the tag in the To header field of a 1xx or 2xx response.
       Since a proxy may not insert a tag into the To header field of
       a 1xx response to a request that did not contain one, it cannot
       issue non-100 provisional responses on its own.  However, it
       can branch the request to a UAS sharing the same element as the
       proxy.  This UAS can return its own provisional responses,
       entering into an early dialog with the initiator of the
       request.  The UAS does not have to be a discreet process from
       the proxy.  It could be a virtual UAS implemented in the same
       code space as the proxy.
       3-6xx responses are delivered hop-by-hop.  When issuing a 3-6xx
       response, the element is effectively acting as a UAS, issuing
       its own response, usually based on the responses received from
       downstream elements.  An element SHOULD preserve the To tag
       when simply forwarding a 3-6xx response to a request that did
       not contain a To tag.
       A proxy MUST NOT modify the To tag in any forwarded response to
       a request that contains a To tag.

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       While it makes no difference to the upstream elements if the
       proxy replaced the To tag in a forwarded 3-6xx response,
       preserving the original tag may assist with debugging.
       When the proxy is aggregating information from several
       responses, choosing a To tag from among them is arbitrary, and
       generating a new To tag may make debugging easier.  This
       happens, for instance, when combining 401 (Unauthorized) and
       407 (Proxy Authentication Required) challenges, or combining
       Contact values from unencrypted and unauthenticated 3xx
       responses.
    7.  Aggregate Authorization Header Field Values
       If the selected response is a 401 (Unauthorized) or 407 (Proxy
       Authentication Required), the proxy MUST collect any WWW-
       Authenticate and Proxy-Authenticate header field values from
       all other 401 (Unauthorized) and 407 (Proxy Authentication
       Required) responses received so far in this response context
       and add them to this response without modification before
       forwarding.  The resulting 401 (Unauthorized) or 407 (Proxy
       Authentication Required) response could have several WWW-
       Authenticate AND Proxy-Authenticate header field values.
       This is necessary because any or all of the destinations the
       request was forwarded to may have requested credentials.  The
       client needs to receive all of those challenges and supply
       credentials for each of them when it retries the request.
       Motivation for this behavior is provided in Section 26.
    8.  Record-Route
       If the selected response contains a Record-Route header field
       value originally provided by this proxy, the proxy MAY choose
       to rewrite the value before forwarding the response.  This
       allows the proxy to provide different URIs for itself to the
       next upstream and downstream elements.  A proxy may choose to
       use this mechanism for any reason.  For instance, it is useful
       for multi-homed hosts.
       If the proxy received the request over TLS, and sent it out
       over a non-TLS connection, the proxy MUST rewrite the URI in
       the Record-Route header field to be a SIPS URI.  If the proxy
       received the request over a non-TLS connection, and sent it out
       over TLS, the proxy MUST rewrite the URI in the Record-Route
       header field to be a SIP URI.

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       The new URI provided by the proxy MUST satisfy the same
       constraints on URIs placed in Record-Route header fields in
       requests (see Step 4 of Section 16.6) with the following
       modifications:
       The URI SHOULD NOT contain the transport parameter unless the
       proxy has knowledge that the next upstream (as opposed to
       downstream) element that will be in the path of subsequent
       requests supports that transport.
       When a proxy does decide to modify the Record-Route header
       field in the response, one of the operations it performs is
       locating the Record-Route value that it had inserted.  If the
       request spiraled, and the proxy inserted a Record-Route value
       in each iteration of the spiral, locating the correct value in
       the response (which must be the proper iteration in the reverse
       direction) is tricky.  The rules above recommend that a proxy
       wishing to rewrite Record-Route header field values insert
       sufficiently distinct URIs into the Record-Route header field
       so that the right one may be selected for rewriting.  A
       RECOMMENDED mechanism to achieve this is for the proxy to
       append a unique identifier for the proxy instance to the user
       portion of the URI.
       When the response arrives, the proxy modifies the first
       Record-Route whose identifier matches the proxy instance.  The
       modification results in a URI without this piece of data
       appended to the user portion of the URI.  Upon the next
       iteration, the same algorithm (find the topmost Record-Route
       header field value with the parameter) will correctly extract
       the next Record-Route header field value inserted by that
       proxy.
       Not every response to a request to which a proxy adds a
       Record-Route header field value will contain a Record-Route
       header field.  If the response does contain a Record-Route
       header field, it will contain the value the proxy added.
    9.  Forward response
       After performing the processing described in steps "Aggregate
       Authorization Header Field Values" through "Record-Route", the
       proxy MAY perform any feature specific manipulations on the
       selected response.  The proxy MUST NOT add to, modify, or
       remove the message body.  Unless otherwise specified, the proxy
       MUST NOT remove any header field values other than the Via
       header field value discussed in Section 16.7 Item 3.  In
       particular, the proxy MUST NOT remove any "received" parameter

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       it may have added to the next Via header field value while
       processing the request associated with this response.  The
       proxy MUST pass the response to the server transaction
       associated with the response context.  This will result in the
       response being sent to the location now indicated in the
       topmost Via header field value.  If the server transaction is
       no longer available to handle the transmission, the element
       MUST forward the response statelessly by sending it to the
       server transport.  The server transaction might indicate
       failure to send the response or signal a timeout in its state
       machine.  These errors would be logged for diagnostic purposes
       as appropriate, but the protocol requires no remedial action
       from the proxy.
       The proxy MUST maintain the response context until all of its
       associated transactions have been terminated, even after
       forwarding a final response.
    10. Generate CANCELs
       If the forwarded response was a final response, the proxy MUST
       generate a CANCEL request for all pending client transactions
       associated with this response context.  A proxy SHOULD also
       generate a CANCEL request for all pending client transactions
       associated with this response context when it receives a 6xx
       response.  A pending client transaction is one that has
       received a provisional response, but no final response (it is
       in the proceeding state) and has not had an associated CANCEL
       generated for it.  Generating CANCEL requests is described in
       Section 9.1.
       The requirement to CANCEL pending client transactions upon
       forwarding a final response does not guarantee that an endpoint
       will not receive multiple 200 (OK) responses to an INVITE.  200
       (OK) responses on more than one branch may be generated before
       the CANCEL requests can be sent and processed.  Further, it is
       reasonable to expect that a future extension may override this
       requirement to issue CANCEL requests.

16.8 Processing Timer C

 If timer C should fire, the proxy MUST either reset the timer with
 any value it chooses, or terminate the client transaction.  If the
 client transaction has received a provisional response, the proxy
 MUST generate a CANCEL request matching that transaction.  If the
 client transaction has not received a provisional response, the proxy
 MUST behave as if the transaction received a 408 (Request Timeout)
 response.

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 Allowing the proxy to reset the timer allows the proxy to dynamically
 extend the transaction's lifetime based on current conditions (such
 as utilization) when the timer fires.

16.9 Handling Transport Errors

 If the transport layer notifies a proxy of an error when it tries to
 forward a request (see Section 18.4), the proxy MUST behave as if the
 forwarded request received a 503 (Service Unavailable) response.
 If the proxy is notified of an error when forwarding a response, it
 drops the response.  The proxy SHOULD NOT cancel any outstanding
 client transactions associated with this response context due to this
 notification.
    If a proxy cancels its outstanding client transactions, a single
    malicious or misbehaving client can cause all transactions to fail
    through its Via header field.

16.10 CANCEL Processing

 A stateful proxy MAY generate a CANCEL to any other request it has
 generated at any time (subject to receiving a provisional response to
 that request as described in section 9.1).  A proxy MUST cancel any
 pending client transactions associated with a response context when
 it receives a matching CANCEL request.
 A stateful proxy MAY generate CANCEL requests for pending INVITE
 client transactions based on the period specified in the INVITE's
 Expires header field elapsing.  However, this is generally
 unnecessary since the endpoints involved will take care of signaling
 the end of the transaction.
 While a CANCEL request is handled in a stateful proxy by its own
 server transaction, a new response context is not created for it.
 Instead, the proxy layer searches its existing response contexts for
 the server transaction handling the request associated with this
 CANCEL.  If a matching response context is found, the element MUST
 immediately return a 200 (OK) response to the CANCEL request.  In
 this case, the element is acting as a user agent server as defined in
 Section 8.2.  Furthermore, the element MUST generate CANCEL requests
 for all pending client transactions in the context as described in
 Section 16.7 step 10.
 If a response context is not found, the element does not have any
 knowledge of the request to apply the CANCEL to.  It MUST statelessly
 forward the CANCEL request (it may have statelessly forwarded the
 associated request previously).

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16.11 Stateless Proxy

 When acting statelessly, a proxy is a simple message forwarder.  Much
 of the processing performed when acting statelessly is the same as
 when behaving statefully.  The differences are detailed here.
 A stateless proxy does not have any notion of a transaction, or of
 the response context used to describe stateful proxy behavior.
 Instead, the stateless proxy takes messages, both requests and
 responses, directly from the transport layer (See section 18).  As a
 result, stateless proxies do not retransmit messages on their own.
 They do, however, forward all retransmissions they receive (they do
 not have the ability to distinguish a retransmission from the
 original message).  Furthermore, when handling a request statelessly,
 an element MUST NOT generate its own 100 (Trying) or any other
 provisional response.
 A stateless proxy MUST validate a request as described in Section
 16.3
 A stateless proxy MUST follow the request processing steps described
 in Sections 16.4 through 16.5 with the following exception:
    o  A stateless proxy MUST choose one and only one target from the
       target set.  This choice MUST only rely on fields in the
       message and time-invariant properties of the server.  In
       particular, a retransmitted request MUST be forwarded to the
       same destination each time it is processed.  Furthermore,
       CANCEL and non-Routed ACK requests MUST generate the same
       choice as their associated INVITE.
 A stateless proxy MUST follow the request processing steps described
 in Section 16.6 with the following exceptions:
    o  The requirement for unique branch IDs across space and time
       applies to stateless proxies as well.  However, a stateless
       proxy cannot simply use a random number generator to compute
       the first component of the branch ID, as described in Section
       16.6 bullet 8.  This is because retransmissions of a request
       need to have the same value, and a stateless proxy cannot tell
       a retransmission from the original request.  Therefore, the
       component of the branch parameter that makes it unique MUST be
       the same each time a retransmitted request is forwarded.  Thus
       for a stateless proxy, the branch parameter MUST be computed as
       a combinatoric function of message parameters which are
       invariant on retransmission.

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       The stateless proxy MAY use any technique it likes to guarantee
       uniqueness of its branch IDs across transactions.  However, the
       following procedure is RECOMMENDED.  The proxy examines the
       branch ID in the topmost Via header field of the received
       request.  If it begins with the magic cookie, the first
       component of the branch ID of the outgoing request is computed
       as a hash of the received branch ID.  Otherwise, the first
       component of the branch ID is computed as a hash of the topmost
       Via, the tag in the To header field, the tag in the From header
       field, the Call-ID header field, the CSeq number (but not
       method), and the Request-URI from the received request.  One of
       these fields will always vary across two different
       transactions.
    o  All other message transformations specified in Section 16.6
       MUST result in the same transformation of a retransmitted
       request.  In particular, if the proxy inserts a Record-Route
       value or pushes URIs into the Route header field, it MUST place
       the same values in retransmissions of the request.  As for the
       Via branch parameter, this implies that the transformations
       MUST be based on time-invariant configuration or
       retransmission-invariant properties of the request.
    o  A stateless proxy determines where to forward the request as
       described for stateful proxies in Section 16.6 Item 10.  The
       request is sent directly to the transport layer instead of
       through a client transaction.
       Since a stateless proxy must forward retransmitted requests to
       the same destination and add identical branch parameters to
       each of them, it can only use information from the message
       itself and time-invariant configuration data for those
       calculations.  If the configuration state is not time-invariant
       (for example, if a routing table is updated) any requests that
       could be affected by the change may not be forwarded
       statelessly during an interval equal to the transaction timeout
       window before or after the change.  The method of processing
       the affected requests in that interval is an implementation
       decision.  A common solution is to forward them transaction
       statefully.
 Stateless proxies MUST NOT perform special processing for CANCEL
 requests.  They are processed by the above rules as any other
 requests.  In particular, a stateless proxy applies the same Route
 header field processing to CANCEL requests that it applies to any
 other request.

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 Response processing as described in Section 16.7 does not apply to a
 proxy behaving statelessly.  When a response arrives at a stateless
 proxy, the proxy MUST inspect the sent-by value in the first
 (topmost) Via header field value.  If that address matches the proxy,
 (it equals a value this proxy has inserted into previous requests)
 the proxy MUST remove that header field value from the response and
 forward the result to the location indicated in the next Via header
 field value.  The proxy MUST NOT add to, modify, or remove the
 message body.  Unless specified otherwise, the proxy MUST NOT remove
 any other header field values.  If the address does not match the
 proxy, the message MUST be silently discarded.

16.12 Summary of Proxy Route Processing

 In the absence of local policy to the contrary, the processing a
 proxy performs on a request containing a Route header field can be
 summarized in the following steps.
    1.  The proxy will inspect the Request-URI.  If it indicates a
        resource owned by this proxy, the proxy will replace it with
        the results of running a location service.  Otherwise, the
        proxy will not change the Request-URI.
    2.  The proxy will inspect the URI in the topmost Route header
        field value.  If it indicates this proxy, the proxy removes it
        from the Route header field (this route node has been
        reached).
    3.  The proxy will forward the request to the resource indicated
        by the URI in the topmost Route header field value or in the
        Request-URI if no Route header field is present.  The proxy
        determines the address, port and transport to use when
        forwarding the request by applying the procedures in [4] to
        that URI.
 If no strict-routing elements are encountered on the path of the
 request, the Request-URI will always indicate the target of the
 request.

16.12.1 Examples

16.12.1.1 Basic SIP Trapezoid

 This scenario is the basic SIP trapezoid, U1 -> P1 -> P2 -> U2, with
 both proxies record-routing.  Here is the flow.

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 U1 sends:
    INVITE sip:callee@domain.com SIP/2.0
    Contact: sip:caller@u1.example.com
 to P1.  P1 is an outbound proxy.  P1 is not responsible for
 domain.com, so it looks it up in DNS and sends it there.  It also
 adds a Record-Route header field value:
    INVITE sip:callee@domain.com SIP/2.0
    Contact: sip:caller@u1.example.com
    Record-Route: <sip:p1.example.com;lr>
 P2 gets this.  It is responsible for domain.com so it runs a location
 service and rewrites the Request-URI.  It also adds a Record-Route
 header field value.  There is no Route header field, so it resolves
 the new Request-URI to determine where to send the request:
    INVITE sip:callee@u2.domain.com SIP/2.0
    Contact: sip:caller@u1.example.com
    Record-Route: <sip:p2.domain.com;lr>
    Record-Route: <sip:p1.example.com;lr>
 The callee at u2.domain.com gets this and responds with a 200 OK:
    SIP/2.0 200 OK
    Contact: sip:callee@u2.domain.com
    Record-Route: <sip:p2.domain.com;lr>
    Record-Route: <sip:p1.example.com;lr>
 The callee at u2 also sets its dialog state's remote target URI to
 sip:caller@u1.example.com and its route set to:
    (<sip:p2.domain.com;lr>,<sip:p1.example.com;lr>)
 This is forwarded by P2 to P1 to U1 as normal.  Now, U1 sets its
 dialog state's remote target URI to sip:callee@u2.domain.com and its
 route set to:
    (<sip:p1.example.com;lr>,<sip:p2.domain.com;lr>)
 Since all the route set elements contain the lr parameter, U1
 constructs the following BYE request:
    BYE sip:callee@u2.domain.com SIP/2.0
    Route: <sip:p1.example.com;lr>,<sip:p2.domain.com;lr>

Rosenberg, et. al. Standards Track [Page 119] RFC 3261 SIP: Session Initiation Protocol June 2002

 As any other element (including proxies) would do, it resolves the
 URI in the topmost Route header field value using DNS to determine
 where to send the request.  This goes to P1.  P1 notices that it is
 not responsible for the resource indicated in the Request-URI so it
 doesn't change it.  It does see that it is the first value in the
 Route header field, so it removes that value, and forwards the
 request to P2:
    BYE sip:callee@u2.domain.com SIP/2.0
    Route: <sip:p2.domain.com;lr>
 P2 also notices it is not responsible for the resource indicated by
 the Request-URI (it is responsible for domain.com, not
 u2.domain.com), so it doesn't change it.  It does see itself in the
 first Route header field value, so it removes it and forwards the
 following to u2.domain.com based on a DNS lookup against the
 Request-URI:
    BYE sip:callee@u2.domain.com SIP/2.0

16.12.1.2 Traversing a Strict-Routing Proxy

 In this scenario, a dialog is established across four proxies, each
 of which adds Record-Route header field values.  The third proxy
 implements the strict-routing procedures specified in RFC 2543 and
 many works in progress.
    U1->P1->P2->P3->P4->U2
 The INVITE arriving at U2 contains:
    INVITE sip:callee@u2.domain.com SIP/2.0
    Contact: sip:caller@u1.example.com
    Record-Route: <sip:p4.domain.com;lr>
    Record-Route: <sip:p3.middle.com>
    Record-Route: <sip:p2.example.com;lr>
    Record-Route: <sip:p1.example.com;lr>
 Which U2 responds to with a 200 OK.  Later, U2 sends the following
 BYE request to P4 based on the first Route header field value.
    BYE sip:caller@u1.example.com SIP/2.0
    Route: <sip:p4.domain.com;lr>
    Route: <sip:p3.middle.com>
    Route: <sip:p2.example.com;lr>
    Route: <sip:p1.example.com;lr>

Rosenberg, et. al. Standards Track [Page 120] RFC 3261 SIP: Session Initiation Protocol June 2002

 P4 is not responsible for the resource indicated in the Request-URI
 so it will leave it alone.  It notices that it is the element in the
 first Route header field value so it removes it.  It then prepares to
 send the request based on the now first Route header field value of
 sip:p3.middle.com, but it notices that this URI does not contain the
 lr parameter, so before sending, it reformats the request to be:
    BYE sip:p3.middle.com SIP/2.0
    Route: <sip:p2.example.com;lr>
    Route: <sip:p1.example.com;lr>
    Route: <sip:caller@u1.example.com>
 P3 is a strict router, so it forwards the following to P2:
    BYE sip:p2.example.com;lr SIP/2.0
    Route: <sip:p1.example.com;lr>
    Route: <sip:caller@u1.example.com>
 P2 sees the request-URI is a value it placed into a Record-Route
 header field, so before further processing, it rewrites the request
 to be:
    BYE sip:caller@u1.example.com SIP/2.0
    Route: <sip:p1.example.com;lr>
 P2 is not responsible for u1.example.com, so it sends the request to
 P1 based on the resolution of the Route header field value.
 P1 notices itself in the topmost Route header field value, so it
 removes it, resulting in:
    BYE sip:caller@u1.example.com SIP/2.0
 Since P1 is not responsible for u1.example.com and there is no Route
 header field, P1 will forward the request to u1.example.com based on
 the Request-URI.

16.12.1.3 Rewriting Record-Route Header Field Values

 In this scenario, U1 and U2 are in different private namespaces and
 they enter a dialog through a proxy P1, which acts as a gateway
 between the namespaces.
    U1->P1->U2

Rosenberg, et. al. Standards Track [Page 121] RFC 3261 SIP: Session Initiation Protocol June 2002

 U1 sends:
    INVITE sip:callee@gateway.leftprivatespace.com SIP/2.0
    Contact: <sip:caller@u1.leftprivatespace.com>
 P1 uses its location service and sends the following to U2:
    INVITE sip:callee@rightprivatespace.com SIP/2.0
    Contact: <sip:caller@u1.leftprivatespace.com>
    Record-Route: <sip:gateway.rightprivatespace.com;lr>
 U2 sends this 200 (OK) back to P1:
    SIP/2.0 200 OK
    Contact: <sip:callee@u2.rightprivatespace.com>
    Record-Route: <sip:gateway.rightprivatespace.com;lr>
 P1 rewrites its Record-Route header parameter to provide a value that
 U1 will find useful, and sends the following to U1:
    SIP/2.0 200 OK
    Contact: <sip:callee@u2.rightprivatespace.com>
    Record-Route: <sip:gateway.leftprivatespace.com;lr>
 Later, U1 sends the following BYE request to P1:
    BYE sip:callee@u2.rightprivatespace.com SIP/2.0
    Route: <sip:gateway.leftprivatespace.com;lr>
 which P1 forwards to U2 as:
    BYE sip:callee@u2.rightprivatespace.com SIP/2.0

17 Transactions

 SIP is a transactional protocol: interactions between components take
 place in a series of independent message exchanges.  Specifically, a
 SIP transaction consists of a single request and any responses to
 that request, which include zero or more provisional responses and
 one or more final responses.  In the case of a transaction where the
 request was an INVITE (known as an INVITE transaction), the
 transaction also includes the ACK only if the final response was not
 a 2xx response.  If the response was a 2xx, the ACK is not considered
 part of the transaction.
    The reason for this separation is rooted in the importance of
    delivering all 200 (OK) responses to an INVITE to the UAC.  To
    deliver them all to the UAC, the UAS alone takes responsibility

Rosenberg, et. al. Standards Track [Page 122] RFC 3261 SIP: Session Initiation Protocol June 2002

    for retransmitting them (see Section 13.3.1.4), and the UAC alone
    takes responsibility for acknowledging them with ACK (see Section
    13.2.2.4).  Since this ACK is retransmitted only by the UAC, it is
    effectively considered its own transaction.
 Transactions have a client side and a server side.  The client side
 is known as a client transaction and the server side as a server
 transaction.  The client transaction sends the request, and the
 server transaction sends the response.  The client and server
 transactions are logical functions that are embedded in any number of
 elements.  Specifically, they exist within user agents and stateful
 proxy servers.  Consider the example in Section 4.  In this example,
 the UAC executes the client transaction, and its outbound proxy
 executes the server transaction.  The outbound proxy also executes a
 client transaction, which sends the request to a server transaction
 in the inbound proxy.  That proxy also executes a client transaction,
 which in turn sends the request to a server transaction in the UAS.
 This is shown in Figure 4.
 +---------+        +---------+        +---------+        +---------+
 |      +-+|Request |+-+   +-+|Request |+-+   +-+|Request |+-+      |
 |      |C||------->||S|   |C||------->||S|   |C||------->||S|      |
 |      |l||        ||e|   |l||        ||e|   |l||        ||e|      |
 |      |i||        ||r|   |i||        ||r|   |i||        ||r|      |
 |      |e||        ||v|   |e||        ||v|   |e||        ||v|      |
 |      |n||        ||e|   |n||        ||e|   |n||        ||e|      |
 |      |t||        ||r|   |t||        ||r|   |t||        ||r|      |
 |      | ||        || |   | ||        || |   | ||        || |      |
 |      |T||        ||T|   |T||        ||T|   |T||        ||T|      |
 |      |r||        ||r|   |r||        ||r|   |r||        ||r|      |
 |      |a||        ||a|   |a||        ||a|   |a||        ||a|      |
 |      |n||        ||n|   |n||        ||n|   |n||        ||n|      |
 |      |s||Response||s|   |s||Response||s|   |s||Response||s|      |
 |      +-+|<-------|+-+   +-+|<-------|+-+   +-+|<-------|+-+      |
 +---------+        +---------+        +---------+        +---------+
    UAC               Outbound           Inbound              UAS
                      Proxy               Proxy
                Figure 4: Transaction relationships
 A stateless proxy does not contain a client or server transaction.
 The transaction exists between the UA or stateful proxy on one side,
 and the UA or stateful proxy on the other side.  As far as SIP
 transactions are concerned, stateless proxies are effectively
 transparent.  The purpose of the client transaction is to receive a
 request from the element in which the client is embedded (call this
 element the "Transaction User" or TU; it can be a UA or a stateful
 proxy), and reliably deliver the request to a server transaction.

Rosenberg, et. al. Standards Track [Page 123] RFC 3261 SIP: Session Initiation Protocol June 2002

 The client transaction is also responsible for receiving responses
 and delivering them to the TU, filtering out any response
 retransmissions or disallowed responses (such as a response to ACK).
 Additionally, in the case of an INVITE request, the client
 transaction is responsible for generating the ACK request for any
 final response accepting a 2xx response.
 Similarly, the purpose of the server transaction is to receive
 requests from the transport layer and deliver them to the TU.  The
 server transaction filters any request retransmissions from the
 network.  The server transaction accepts responses from the TU and
 delivers them to the transport layer for transmission over the
 network.  In the case of an INVITE transaction, it absorbs the ACK
 request for any final response excepting a 2xx response.
 The 2xx response and its ACK receive special treatment.  This
 response is retransmitted only by a UAS, and its ACK generated only
 by the UAC.  This end-to-end treatment is needed so that a caller
 knows the entire set of users that have accepted the call.  Because
 of this special handling, retransmissions of the 2xx response are
 handled by the UA core, not the transaction layer.  Similarly,
 generation of the ACK for the 2xx is handled by the UA core.  Each
 proxy along the path merely forwards each 2xx response to INVITE and
 its corresponding ACK.

17.1 Client Transaction

 The client transaction provides its functionality through the
 maintenance of a state machine.
 The TU communicates with the client transaction through a simple
 interface.  When the TU wishes to initiate a new transaction, it
 creates a client transaction and passes it the SIP request to send
 and an IP address, port, and transport to which to send it.  The
 client transaction begins execution of its state machine.  Valid
 responses are passed up to the TU from the client transaction.
 There are two types of client transaction state machines, depending
 on the method of the request passed by the TU.  One handles client
 transactions for INVITE requests.  This type of machine is referred
 to as an INVITE client transaction.  Another type handles client
 transactions for all requests except INVITE and ACK.  This is
 referred to as a non-INVITE client transaction.  There is no client
 transaction for ACK.  If the TU wishes to send an ACK, it passes one
 directly to the transport layer for transmission.

Rosenberg, et. al. Standards Track [Page 124] RFC 3261 SIP: Session Initiation Protocol June 2002

 The INVITE transaction is different from those of other methods
 because of its extended duration.  Normally, human input is required
 in order to respond to an INVITE.  The long delays expected for
 sending a response argue for a three-way handshake.  On the other
 hand, requests of other methods are expected to complete rapidly.
 Because of the non-INVITE transaction's reliance on a two-way
 handshake, TUs SHOULD respond immediately to non-INVITE requests.

17.1.1 INVITE Client Transaction

17.1.1.1 Overview of INVITE Transaction

 The INVITE transaction consists of a three-way handshake.  The client
 transaction sends an INVITE, the server transaction sends responses,
 and the client transaction sends an ACK.  For unreliable transports
 (such as UDP), the client transaction retransmits requests at an
 interval that starts at T1 seconds and doubles after every
 retransmission.  T1 is an estimate of the round-trip time (RTT), and
 it defaults to 500 ms.  Nearly all of the transaction timers
 described here scale with T1, and changing T1 adjusts their values.
 The request is not retransmitted over reliable transports.  After
 receiving a 1xx response, any retransmissions cease altogether, and
 the client waits for further responses.  The server transaction can
 send additional 1xx responses, which are not transmitted reliably by
 the server transaction.  Eventually, the server transaction decides
 to send a final response.  For unreliable transports, that response
 is retransmitted periodically, and for reliable transports, it is
 sent once.  For each final response that is received at the client
 transaction, the client transaction sends an ACK, the purpose of
 which is to quench retransmissions of the response.

17.1.1.2 Formal Description

 The state machine for the INVITE client transaction is shown in
 Figure 5.  The initial state, "calling", MUST be entered when the TU
 initiates a new client transaction with an INVITE request.  The
 client transaction MUST pass the request to the transport layer for
 transmission (see Section 18).  If an unreliable transport is being
 used, the client transaction MUST start timer A with a value of T1.
 If a reliable transport is being used, the client transaction SHOULD
 NOT start timer A (Timer A controls request retransmissions).  For
 any transport, the client transaction MUST start timer B with a value
 of 64*T1 seconds (Timer B controls transaction timeouts).
 When timer A fires, the client transaction MUST retransmit the
 request by passing it to the transport layer, and MUST reset the
 timer with a value of 2*T1.  The formal definition of retransmit

Rosenberg, et. al. Standards Track [Page 125] RFC 3261 SIP: Session Initiation Protocol June 2002

 within the context of the transaction layer is to take the message
 previously sent to the transport layer and pass it to the transport
 layer once more.
 When timer A fires 2*T1 seconds later, the request MUST be
 retransmitted again (assuming the client transaction is still in this
 state).  This process MUST continue so that the request is
 retransmitted with intervals that double after each transmission.
 These retransmissions SHOULD only be done while the client
 transaction is in the "calling" state.
 The default value for T1 is 500 ms.  T1 is an estimate of the RTT
 between the client and server transactions.  Elements MAY (though it
 is NOT RECOMMENDED) use smaller values of T1 within closed, private
 networks that do not permit general Internet connection.  T1 MAY be
 chosen larger, and this is RECOMMENDED if it is known in advance
 (such as on high latency access links) that the RTT is larger.
 Whatever the value of T1, the exponential backoffs on retransmissions
 described in this section MUST be used.
 If the client transaction is still in the "Calling" state when timer
 B fires, the client transaction SHOULD inform the TU that a timeout
 has occurred.  The client transaction MUST NOT generate an ACK.  The
 value of 64*T1 is equal to the amount of time required to send seven
 requests in the case of an unreliable transport.
 If the client transaction receives a provisional response while in
 the "Calling" state, it transitions to the "Proceeding" state. In the
 "Proceeding" state, the client transaction SHOULD NOT retransmit the
 request any longer. Furthermore, the provisional response MUST be
 passed to the TU.  Any further provisional responses MUST be passed
 up to the TU while in the "Proceeding" state.
 When in either the "Calling" or "Proceeding" states, reception of a
 response with status code from 300-699 MUST cause the client
 transaction to transition to "Completed".  The client transaction
 MUST pass the received response up to the TU, and the client
 transaction MUST generate an ACK request, even if the transport is
 reliable (guidelines for constructing the ACK from the response are
 given in Section 17.1.1.3) and then pass the ACK to the transport
 layer for transmission.  The ACK MUST be sent to the same address,
 port, and transport to which the original request was sent.  The
 client transaction SHOULD start timer D when it enters the
 "Completed" state, with a value of at least 32 seconds for unreliable
 transports, and a value of zero seconds for reliable transports.
 Timer D reflects the amount of time that the server transaction can
 remain in the "Completed" state when unreliable transports are used.
 This is equal to Timer H in the INVITE server transaction, whose

Rosenberg, et. al. Standards Track [Page 126] RFC 3261 SIP: Session Initiation Protocol June 2002

 default is 64*T1.  However, the client transaction does not know the
 value of T1 in use by the server transaction, so an absolute minimum
 of 32s is used instead of basing Timer D on T1.
 Any retransmissions of the final response that are received while in
 the "Completed" state MUST cause the ACK to be re-passed to the
 transport layer for retransmission, but the newly received response
 MUST NOT be passed up to the TU.  A retransmission of the response is
 defined as any response which would match the same client transaction
 based on the rules of Section 17.1.3.

Rosenberg, et. al. Standards Track [Page 127] RFC 3261 SIP: Session Initiation Protocol June 2002

                             |INVITE from TU
           Timer A fires     |INVITE sent
           Reset A,          V                      Timer B fires
           INVITE sent +-----------+                or Transport Err.
             +---------|           |---------------+inform TU
             |         |  Calling  |               |
             +-------->|           |-------------->|
                       +-----------+ 2xx           |
                          |  |       2xx to TU     |
                          |  |1xx                  |
  300-699 +---------------+  |1xx to TU            |
 ACK sent |                  |                     |

resp. to TU | 1xx V |

          |  1xx to TU  -----------+               |
          |  +---------|           |               |
          |  |         |Proceeding |-------------->|
          |  +-------->|           | 2xx           |
          |            +-----------+ 2xx to TU     |
          |       300-699    |                     |
          |       ACK sent,  |                     |
          |       resp. to TU|                     |
          |                  |                     |      NOTE:
          |  300-699         V                     |
          |  ACK sent  +-----------+Transport Err. |  transitions
          |  +---------|           |Inform TU      |  labeled with
          |  |         | Completed |-------------->|  the event
          |  +-------->|           |               |  over the action
          |            +-----------+               |  to take
          |              ^   |                     |
          |              |   | Timer D fires       |
          +--------------+   | -                   |
                             |                     |
                             V                     |
                       +-----------+               |
                       |           |               |
                       | Terminated|<--------------+
                       |           |
                       +-----------+
               Figure 5: INVITE client transaction
 If timer D fires while the client transaction is in the "Completed"
 state, the client transaction MUST move to the terminated state.
 When in either the "Calling" or "Proceeding" states, reception of a
 2xx response MUST cause the client transaction to enter the
 "Terminated" state, and the response MUST be passed up to the TU.
 The handling of this response depends on whether the TU is a proxy

Rosenberg, et. al. Standards Track [Page 128] RFC 3261 SIP: Session Initiation Protocol June 2002

 core or a UAC core.  A UAC core will handle generation of the ACK for
 this response, while a proxy core will always forward the 200 (OK)
 upstream.  The differing treatment of 200 (OK) between proxy and UAC
 is the reason that handling of it does not take place in the
 transaction layer.
 The client transaction MUST be destroyed the instant it enters the
 "Terminated" state.  This is actually necessary to guarantee correct
 operation.  The reason is that 2xx responses to an INVITE are treated
 differently; each one is forwarded by proxies, and the ACK handling
 in a UAC is different.  Thus, each 2xx needs to be passed to a proxy
 core (so that it can be forwarded) and to a UAC core (so it can be
 acknowledged).  No transaction layer processing takes place.
 Whenever a response is received by the transport, if the transport
 layer finds no matching client transaction (using the rules of
 Section 17.1.3), the response is passed directly to the core.  Since
 the matching client transaction is destroyed by the first 2xx,
 subsequent 2xx will find no match and therefore be passed to the
 core.

17.1.1.3 Construction of the ACK Request

 This section specifies the construction of ACK requests sent within
 the client transaction.  A UAC core that generates an ACK for 2xx
 MUST instead follow the rules described in Section 13.
 The ACK request constructed by the client transaction MUST contain
 values for the Call-ID, From, and Request-URI that are equal to the
 values of those header fields in the request passed to the transport
 by the client transaction (call this the "original request").  The To
 header field in the ACK MUST equal the To header field in the
 response being acknowledged, and therefore will usually differ from
 the To header field in the original request by the addition of the
 tag parameter.  The ACK MUST contain a single Via header field, and
 this MUST be equal to the top Via header field of the original
 request.  The CSeq header field in the ACK MUST contain the same
 value for the sequence number as was present in the original request,
 but the method parameter MUST be equal to "ACK".

Rosenberg, et. al. Standards Track [Page 129] RFC 3261 SIP: Session Initiation Protocol June 2002

 If the INVITE request whose response is being acknowledged had Route
 header fields, those header fields MUST appear in the ACK.  This is
 to ensure that the ACK can be routed properly through any downstream
 stateless proxies.
 Although any request MAY contain a body, a body in an ACK is special
 since the request cannot be rejected if the body is not understood.
 Therefore, placement of bodies in ACK for non-2xx is NOT RECOMMENDED,
 but if done, the body types are restricted to any that appeared in
 the INVITE, assuming that the response to the INVITE was not 415.  If
 it was, the body in the ACK MAY be any type listed in the Accept
 header field in the 415.
 For example, consider the following request:
 INVITE sip:bob@biloxi.com SIP/2.0
 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKkjshdyff
 To: Bob <sip:bob@biloxi.com>
 From: Alice <sip:alice@atlanta.com>;tag=88sja8x
 Max-Forwards: 70
 Call-ID: 987asjd97y7atg
 CSeq: 986759 INVITE
 The ACK request for a non-2xx final response to this request would
 look like this:
 ACK sip:bob@biloxi.com SIP/2.0
 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKkjshdyff
 To: Bob <sip:bob@biloxi.com>;tag=99sa0xk
 From: Alice <sip:alice@atlanta.com>;tag=88sja8x
 Max-Forwards: 70
 Call-ID: 987asjd97y7atg
 CSeq: 986759 ACK

17.1.2 Non-INVITE Client Transaction

17.1.2.1 Overview of the non-INVITE Transaction

 Non-INVITE transactions do not make use of ACK.  They are simple
 request-response interactions.  For unreliable transports, requests
 are retransmitted at an interval which starts at T1 and doubles until
 it hits T2.  If a provisional response is received, retransmissions
 continue for unreliable transports, but at an interval of T2.  The
 server transaction retransmits the last response it sent, which can
 be a provisional or final response, only when a retransmission of the
 request is received.  This is why request retransmissions need to
 continue even after a provisional response; they are to ensure
 reliable delivery of the final response.

Rosenberg, et. al. Standards Track [Page 130] RFC 3261 SIP: Session Initiation Protocol June 2002

 Unlike an INVITE transaction, a non-INVITE transaction has no special
 handling for the 2xx response.  The result is that only a single 2xx
 response to a non-INVITE is ever delivered to a UAC.

17.1.2.2 Formal Description

 The state machine for the non-INVITE client transaction is shown in
 Figure 6.  It is very similar to the state machine for INVITE.
 The "Trying" state is entered when the TU initiates a new client
 transaction with a request.  When entering this state, the client
 transaction SHOULD set timer F to fire in 64*T1 seconds.  The request
 MUST be passed to the transport layer for transmission.  If an
 unreliable transport is in use, the client transaction MUST set timer
 E to fire in T1 seconds.  If timer E fires while still in this state,
 the timer is reset, but this time with a value of MIN(2*T1, T2).
 When the timer fires again, it is reset to a MIN(4*T1, T2).  This
 process continues so that retransmissions occur with an exponentially
 increasing interval that caps at T2.  The default value of T2 is 4s,
 and it represents the amount of time a non-INVITE server transaction
 will take to respond to a request, if it does not respond
 immediately.  For the default values of T1 and T2, this results in
 intervals of 500 ms, 1 s, 2 s, 4 s, 4 s, 4 s, etc.
 If Timer F fires while the client transaction is still in the
 "Trying" state, the client transaction SHOULD inform the TU about the
 timeout, and then it SHOULD enter the "Terminated" state.  If a
 provisional response is received while in the "Trying" state, the
 response MUST be passed to the TU, and then the client transaction
 SHOULD move to the "Proceeding" state.  If a final response (status
 codes 200-699) is received while in the "Trying" state, the response
 MUST be passed to the TU, and the client transaction MUST transition
 to the "Completed" state.
 If Timer E fires while in the "Proceeding" state, the request MUST be
 passed to the transport layer for retransmission, and Timer E MUST be
 reset with a value of T2 seconds.  If timer F fires while in the
 "Proceeding" state, the TU MUST be informed of a timeout, and the
 client transaction MUST transition to the terminated state.  If a
 final response (status codes 200-699) is received while in the
 "Proceeding" state, the response MUST be passed to the TU, and the
 client transaction MUST transition to the "Completed" state.
 Once the client transaction enters the "Completed" state, it MUST set
 Timer K to fire in T4 seconds for unreliable transports, and zero
 seconds for reliable transports.  The "Completed" state exists to
 buffer any additional response retransmissions that may be received
 (which is why the client transaction remains there only for

Rosenberg, et. al. Standards Track [Page 131] RFC 3261 SIP: Session Initiation Protocol June 2002

 unreliable transports).  T4 represents the amount of time the network
 will take to clear messages between client and server transactions.
 The default value of T4 is 5s.  A response is a retransmission when
 it matches the same transaction, using the rules specified in Section
 17.1.3.  If Timer K fires while in this state, the client transaction
 MUST transition to the "Terminated" state.
 Once the transaction is in the terminated state, it MUST be destroyed
 immediately.

17.1.3 Matching Responses to Client Transactions

 When the transport layer in the client receives a response, it has to
 determine which client transaction will handle the response, so that
 the processing of Sections 17.1.1 and 17.1.2 can take place.  The
 branch parameter in the top Via header field is used for this
 purpose.  A response matches a client transaction under two
 conditions:
    1.  If the response has the same value of the branch parameter in
        the top Via header field as the branch parameter in the top
        Via header field of the request that created the transaction.
    2.  If the method parameter in the CSeq header field matches the
        method of the request that created the transaction.  The
        method is needed since a CANCEL request constitutes a
        different transaction, but shares the same value of the branch
        parameter.
 If a request is sent via multicast, it is possible that it will
 generate multiple responses from different servers.  These responses
 will all have the same branch parameter in the topmost Via, but vary
 in the To tag.  The first response received, based on the rules
 above, will be used, and others will be viewed as retransmissions.
 That is not an error; multicast SIP provides only a rudimentary
 "single-hop-discovery-like" service that is limited to processing a
 single response.  See Section 18.1.1 for details.

Rosenberg, et. al. Standards Track [Page 132] RFC 3261 SIP: Session Initiation Protocol June 2002

17.1.4 Handling Transport Errors

                                 |Request from TU
                                 |send request
             Timer E             V
             send request  +-----------+
                 +---------|           |-------------------+
                 |         |  Trying   |  Timer F          |
                 +-------->|           |  or Transport Err.|
                           +-----------+  inform TU        |
              200-699         |  |                         |
              resp. to TU     |  |1xx                      |
              +---------------+  |resp. to TU              |
              |                  |                         |
              |   Timer E        V       Timer F           |
              |   send req +-----------+ or Transport Err. |
              |  +---------|           | inform TU         |
              |  |         |Proceeding |------------------>|
              |  +-------->|           |-----+             |
              |            +-----------+     |1xx          |
              |              |      ^        |resp to TU   |
              | 200-699      |      +--------+             |
              | resp. to TU  |                             |
              |              |                             |
              |              V                             |
              |            +-----------+                   |
              |            |           |                   |
              |            | Completed |                   |
              |            |           |                   |
              |            +-----------+                   |
              |              ^   |                         |
              |              |   | Timer K                 |
              +--------------+   | -                       |
                                 |                         |
                                 V                         |
           NOTE:           +-----------+                   |
                           |           |                   |
       transitions         | Terminated|<------------------+
       labeled with        |           |
       the event           +-----------+
       over the action
       to take
               Figure 6: non-INVITE client transaction
 When the client transaction sends a request to the transport layer to
 be sent, the following procedures are followed if the transport layer
 indicates a failure.

Rosenberg, et. al. Standards Track [Page 133] RFC 3261 SIP: Session Initiation Protocol June 2002

 The client transaction SHOULD inform the TU that a transport failure
 has occurred, and the client transaction SHOULD transition directly
 to the "Terminated" state.  The TU will handle the failover
 mechanisms described in [4].

17.2 Server Transaction

 The server transaction is responsible for the delivery of requests to
 the TU and the reliable transmission of responses.  It accomplishes
 this through a state machine.  Server transactions are created by the
 core when a request is received, and transaction handling is desired
 for that request (this is not always the case).
 As with the client transactions, the state machine depends on whether
 the received request is an INVITE request.

17.2.1 INVITE Server Transaction

 The state diagram for the INVITE server transaction is shown in
 Figure 7.
 When a server transaction is constructed for a request, it enters the
 "Proceeding" state.  The server transaction MUST generate a 100
 (Trying) response unless it knows that the TU will generate a
 provisional or final response within 200 ms, in which case it MAY
 generate a 100 (Trying) response.  This provisional response is
 needed to quench request retransmissions rapidly in order to avoid
 network congestion.  The 100 (Trying) response is constructed
 according to the procedures in Section 8.2.6, except that the
 insertion of tags in the To header field of the response (when none
 was present in the request) is downgraded from MAY to SHOULD NOT.
 The request MUST be passed to the TU.
 The TU passes any number of provisional responses to the server
 transaction.  So long as the server transaction is in the
 "Proceeding" state, each of these MUST be passed to the transport
 layer for transmission.  They are not sent reliably by the
 transaction layer (they are not retransmitted by it) and do not cause
 a change in the state of the server transaction.  If a request
 retransmission is received while in the "Proceeding" state, the most
 recent provisional response that was received from the TU MUST be
 passed to the transport layer for retransmission.  A request is a
 retransmission if it matches the same server transaction based on the
 rules of Section 17.2.3.
 If, while in the "Proceeding" state, the TU passes a 2xx response to
 the server transaction, the server transaction MUST pass this
 response to the transport layer for transmission.  It is not

Rosenberg, et. al. Standards Track [Page 134] RFC 3261 SIP: Session Initiation Protocol June 2002

 retransmitted by the server transaction; retransmissions of 2xx
 responses are handled by the TU.  The server transaction MUST then
 transition to the "Terminated" state.
 While in the "Proceeding" state, if the TU passes a response with
 status code from 300 to 699 to the server transaction, the response
 MUST be passed to the transport layer for transmission, and the state
 machine MUST enter the "Completed" state.  For unreliable transports,
 timer G is set to fire in T1 seconds, and is not set to fire for
 reliable transports.
    This is a change from RFC 2543, where responses were always
    retransmitted, even over reliable transports.
 When the "Completed" state is entered, timer H MUST be set to fire in
 64*T1 seconds for all transports.  Timer H determines when the server
 transaction abandons retransmitting the response.  Its value is
 chosen to equal Timer B, the amount of time a client transaction will
 continue to retry sending a request.  If timer G fires, the response
 is passed to the transport layer once more for retransmission, and
 timer G is set to fire in MIN(2*T1, T2) seconds.  From then on, when
 timer G fires, the response is passed to the transport again for
 transmission, and timer G is reset with a value that doubles, unless
 that value exceeds T2, in which case it is reset with the value of
 T2.  This is identical to the retransmit behavior for requests in the
 "Trying" state of the non-INVITE client transaction.  Furthermore,
 while in the "Completed" state, if a request retransmission is
 received, the server SHOULD pass the response to the transport for
 retransmission.
 If an ACK is received while the server transaction is in the
 "Completed" state, the server transaction MUST transition to the
 "Confirmed" state.  As Timer G is ignored in this state, any
 retransmissions of the response will cease.
 If timer H fires while in the "Completed" state, it implies that the
 ACK was never received.  In this case, the server transaction MUST
 transition to the "Terminated" state, and MUST indicate to the TU
 that a transaction failure has occurred.

Rosenberg, et. al. Standards Track [Page 135] RFC 3261 SIP: Session Initiation Protocol June 2002

                             |INVITE
                             |pass INV to TU
          INVITE             V send 100 if TU won't in 200ms
          send response+-----------+
              +--------|           |--------+101-199 from TU
              |        | Proceeding|        |send response
              +------->|           |<-------+
                       |           |          Transport Err.
                       |           |          Inform TU
                       |           |--------------->+
                       +-----------+                |
          300-699 from TU |     |2xx from TU        |
          send response   |     |send response      |
                          |     +------------------>+
                          |                         |
          INVITE          V          Timer G fires  |
          send response+-----------+ send response  |
              +--------|           |--------+       |
              |        | Completed |        |       |
              +------->|           |<-------+       |
                       +-----------+                |
                          |     |                   |
                      ACK |     |                   |
                      -   |     +------------------>+
                          |        Timer H fires    |
                          V        or Transport Err.|
                       +-----------+  Inform TU     |
                       |           |                |
                       | Confirmed |                |
                       |           |                |
                       +-----------+                |
                             |                      |
                             |Timer I fires         |
                             |-                     |
                             |                      |
                             V                      |
                       +-----------+                |
                       |           |                |
                       | Terminated|<---------------+
                       |           |
                       +-----------+
            Figure 7: INVITE server transaction

Rosenberg, et. al. Standards Track [Page 136] RFC 3261 SIP: Session Initiation Protocol June 2002

 The purpose of the "Confirmed" state is to absorb any additional ACK
 messages that arrive, triggered from retransmissions of the final
 response.  When this state is entered, timer I is set to fire in T4
 seconds for unreliable transports, and zero seconds for reliable
 transports.  Once timer I fires, the server MUST transition to the
 "Terminated" state.
 Once the transaction is in the "Terminated" state, it MUST be
 destroyed immediately.  As with client transactions, this is needed
 to ensure reliability of the 2xx responses to INVITE.

17.2.2 Non-INVITE Server Transaction

 The state machine for the non-INVITE server transaction is shown in
 Figure 8.
 The state machine is initialized in the "Trying" state and is passed
 a request other than INVITE or ACK when initialized.  This request is
 passed up to the TU.  Once in the "Trying" state, any further request
 retransmissions are discarded.  A request is a retransmission if it
 matches the same server transaction, using the rules specified in
 Section 17.2.3.
 While in the "Trying" state, if the TU passes a provisional response
 to the server transaction, the server transaction MUST enter the
 "Proceeding" state.  The response MUST be passed to the transport
 layer for transmission.  Any further provisional responses that are
 received from the TU while in the "Proceeding" state MUST be passed
 to the transport layer for transmission.  If a retransmission of the
 request is received while in the "Proceeding" state, the most
 recently sent provisional response MUST be passed to the transport
 layer for retransmission.  If the TU passes a final response (status
 codes 200-699) to the server while in the "Proceeding" state, the
 transaction MUST enter the "Completed" state, and the response MUST
 be passed to the transport layer for transmission.
 When the server transaction enters the "Completed" state, it MUST set
 Timer J to fire in 64*T1 seconds for unreliable transports, and zero
 seconds for reliable transports.  While in the "Completed" state, the
 server transaction MUST pass the final response to the transport
 layer for retransmission whenever a retransmission of the request is
 received.  Any other final responses passed by the TU to the server
 transaction MUST be discarded while in the "Completed" state.  The
 server transaction remains in this state until Timer J fires, at
 which point it MUST transition to the "Terminated" state.
 The server transaction MUST be destroyed the instant it enters the
 "Terminated" state.

Rosenberg, et. al. Standards Track [Page 137] RFC 3261 SIP: Session Initiation Protocol June 2002

17.2.3 Matching Requests to Server Transactions

 When a request is received from the network by the server, it has to
 be matched to an existing transaction.  This is accomplished in the
 following manner.
 The branch parameter in the topmost Via header field of the request
 is examined.  If it is present and begins with the magic cookie
 "z9hG4bK", the request was generated by a client transaction
 compliant to this specification.  Therefore, the branch parameter
 will be unique across all transactions sent by that client.  The
 request matches a transaction if:
    1. the branch parameter in the request is equal to the one in the
       top Via header field of the request that created the
       transaction, and
    2. the sent-by value in the top Via of the request is equal to the
       one in the request that created the transaction, and
    3. the method of the request matches the one that created the
       transaction, except for ACK, where the method of the request
       that created the transaction is INVITE.
 This matching rule applies to both INVITE and non-INVITE transactions
 alike.
    The sent-by value is used as part of the matching process because
    there could be accidental or malicious duplication of branch
    parameters from different clients.
 If the branch parameter in the top Via header field is not present,
 or does not contain the magic cookie, the following procedures are
 used.  These exist to handle backwards compatibility with RFC 2543
 compliant implementations.
 The INVITE request matches a transaction if the Request-URI, To tag,
 From tag, Call-ID, CSeq, and top Via header field match those of the
 INVITE request which created the transaction.  In this case, the
 INVITE is a retransmission of the original one that created the
 transaction.  The ACK request matches a transaction if the Request-
 URI, From tag, Call-ID, CSeq number (not the method), and top Via
 header field match those of the INVITE request which created the
 transaction, and the To tag of the ACK matches the To tag of the
 response sent by the server transaction.  Matching is done based on
 the matching rules defined for each of those header fields.
 Inclusion of the tag in the To header field in the ACK matching
 process helps disambiguate ACK for 2xx from ACK for other responses

Rosenberg, et. al. Standards Track [Page 138] RFC 3261 SIP: Session Initiation Protocol June 2002

 at a proxy, which may have forwarded both responses (This can occur
 in unusual conditions.  Specifically, when a proxy forked a request,
 and then crashes, the responses may be delivered to another proxy,
 which might end up forwarding multiple responses upstream).  An ACK
 request that matches an INVITE transaction matched by a previous ACK
 is considered a retransmission of that previous ACK.

Rosenberg, et. al. Standards Track [Page 139] RFC 3261 SIP: Session Initiation Protocol June 2002

                                |Request received
                                |pass to TU
                                V
                          +-----------+
                          |           |
                          | Trying    |-------------+
                          |           |             |
                          +-----------+             |200-699 from TU
                                |                   |send response
                                |1xx from TU        |
                                |send response      |
                                |                   |
             Request            V      1xx from TU  |
             send response+-----------+send response|
                 +--------|           |--------+    |
                 |        | Proceeding|        |    |
                 +------->|           |<-------+    |
          +<--------------|           |             |
          |Trnsprt Err    +-----------+             |
          |Inform TU            |                   |
          |                     |                   |
          |                     |200-699 from TU    |
          |                     |send response      |
          |  Request            V                   |
          |  send response+-----------+             |
          |      +--------|           |             |
          |      |        | Completed |<------------+
          |      +------->|           |
          +<--------------|           |
          |Trnsprt Err    +-----------+
          |Inform TU            |
          |                     |Timer J fires
          |                     |-
          |                     |
          |                     V
          |               +-----------+
          |               |           |
          +-------------->| Terminated|
                          |           |
                          +-----------+
              Figure 8: non-INVITE server transaction
 For all other request methods, a request is matched to a transaction
 if the Request-URI, To tag, From tag, Call-ID, CSeq (including the
 method), and top Via header field match those of the request that
 created the transaction.  Matching is done based on the matching

Rosenberg, et. al. Standards Track [Page 140] RFC 3261 SIP: Session Initiation Protocol June 2002

 rules defined for each of those header fields.  When a non-INVITE
 request matches an existing transaction, it is a retransmission of
 the request that created that transaction.
 Because the matching rules include the Request-URI, the server cannot
 match a response to a transaction.  When the TU passes a response to
 the server transaction, it must pass it to the specific server
 transaction for which the response is targeted.

17.2.4 Handling Transport Errors

 When the server transaction sends a response to the transport layer
 to be sent, the following procedures are followed if the transport
 layer indicates a failure.
 First, the procedures in [4] are followed, which attempt to deliver
 the response to a backup.  If those should all fail, based on the
 definition of failure in [4], the server transaction SHOULD inform
 the TU that a failure has occurred, and SHOULD transition to the
 terminated state.

18 Transport

 The transport layer is responsible for the actual transmission of
 requests and responses over network transports.  This includes
 determination of the connection to use for a request or response in
 the case of connection-oriented transports.
 The transport layer is responsible for managing persistent
 connections for transport protocols like TCP and SCTP, or TLS over
 those, including ones opened to the transport layer.  This includes
 connections opened by the client or server transports, so that
 connections are shared between client and server transport functions.
 These connections are indexed by the tuple formed from the address,
 port, and transport protocol at the far end of the connection.  When
 a connection is opened by the transport layer, this index is set to
 the destination IP, port and transport.  When the connection is
 accepted by the transport layer, this index is set to the source IP
 address, port number, and transport.  Note that, because the source
 port is often ephemeral, but it cannot be known whether it is
 ephemeral or selected through procedures in [4], connections accepted
 by the transport layer will frequently not be reused.  The result is
 that two proxies in a "peering" relationship using a connection-
 oriented transport frequently will have two connections in use, one
 for transactions initiated in each direction.

Rosenberg, et. al. Standards Track [Page 141] RFC 3261 SIP: Session Initiation Protocol June 2002

 It is RECOMMENDED that connections be kept open for some
 implementation-defined duration after the last message was sent or
 received over that connection.  This duration SHOULD at least equal
 the longest amount of time the element would need in order to bring a
 transaction from instantiation to the terminated state.  This is to
 make it likely that transactions are completed over the same
 connection on which they are initiated (for example, request,
 response, and in the case of INVITE, ACK for non-2xx responses).
 This usually means at least 64*T1 (see Section 17.1.1.1 for a
 definition of T1).  However, it could be larger in an element that
 has a TU using a large value for timer C (bullet 11 of Section 16.6),
 for example.
 All SIP elements MUST implement UDP and TCP.  SIP elements MAY
 implement other protocols.
    Making TCP mandatory for the UA is a substantial change from RFC
    2543.  It has arisen out of the need to handle larger messages,
    which MUST use TCP, as discussed below.  Thus, even if an element
    never sends large messages, it may receive one and needs to be
    able to handle them.

18.1 Clients

18.1.1 Sending Requests

 The client side of the transport layer is responsible for sending the
 request and receiving responses.  The user of the transport layer
 passes the client transport the request, an IP address, port,
 transport, and possibly TTL for multicast destinations.
 If a request is within 200 bytes of the path MTU, or if it is larger
 than 1300 bytes and the path MTU is unknown, the request MUST be sent
 using an RFC 2914 [43] congestion controlled transport protocol, such
 as TCP. If this causes a change in the transport protocol from the
 one indicated in the top Via, the value in the top Via MUST be
 changed.  This prevents fragmentation of messages over UDP and
 provides congestion control for larger messages.  However,
 implementations MUST be able to handle messages up to the maximum
 datagram packet size.  For UDP, this size is 65,535 bytes, including
 IP and UDP headers.
    The 200 byte "buffer" between the message size and the MTU
    accommodates the fact that the response in SIP can be larger than
    the request.  This happens due to the addition of Record-Route
    header field values to the responses to INVITE, for example.  With
    the extra buffer, the response can be about 170 bytes larger than
    the request, and still not be fragmented on IPv4 (about 30 bytes

Rosenberg, et. al. Standards Track [Page 142] RFC 3261 SIP: Session Initiation Protocol June 2002

    is consumed by IP/UDP, assuming no IPSec).  1300 is chosen when
    path MTU is not known, based on the assumption of a 1500 byte
    Ethernet MTU.
 If an element sends a request over TCP because of these message size
 constraints, and that request would have otherwise been sent over
 UDP, if the attempt to establish the connection generates either an
 ICMP Protocol Not Supported, or results in a TCP reset, the element
 SHOULD retry the request, using UDP.  This is only to provide
 backwards compatibility with RFC 2543 compliant implementations that
 do not support TCP.  It is anticipated that this behavior will be
 deprecated in a future revision of this specification.
 A client that sends a request to a multicast address MUST add the
 "maddr" parameter to its Via header field value containing the
 destination multicast address, and for IPv4, SHOULD add the "ttl"
 parameter with a value of 1.  Usage of IPv6 multicast is not defined
 in this specification, and will be a subject of future
 standardization when the need arises.
 These rules result in a purposeful limitation of multicast in SIP.
 Its primary function is to provide a "single-hop-discovery-like"
 service, delivering a request to a group of homogeneous servers,
 where it is only required to process the response from any one of
 them.  This functionality is most useful for registrations.  In fact,
 based on the transaction processing rules in Section 17.1.3, the
 client transaction will accept the first response, and view any
 others as retransmissions because they all contain the same Via
 branch identifier.
 Before a request is sent, the client transport MUST insert a value of
 the "sent-by" field into the Via header field.  This field contains
 an IP address or host name, and port.  The usage of an FQDN is
 RECOMMENDED.  This field is used for sending responses under certain
 conditions, described below.  If the port is absent, the default
 value depends on the transport.  It is 5060 for UDP, TCP and SCTP,
 5061 for TLS.
 For reliable transports, the response is normally sent on the
 connection on which the request was received.  Therefore, the client
 transport MUST be prepared to receive the response on the same
 connection used to send the request.  Under error conditions, the
 server may attempt to open a new connection to send the response.  To
 handle this case, the transport layer MUST also be prepared to
 receive an incoming connection on the source IP address from which
 the request was sent and port number in the "sent-by" field.  It also

Rosenberg, et. al. Standards Track [Page 143] RFC 3261 SIP: Session Initiation Protocol June 2002

 MUST be prepared to receive incoming connections on any address and
 port that would be selected by a server based on the procedures
 described in Section 5 of [4].
 For unreliable unicast transports, the client transport MUST be
 prepared to receive responses on the source IP address from which the
 request is sent (as responses are sent back to the source address)
 and the port number in the "sent-by" field.  Furthermore, as with
 reliable transports, in certain cases the response will be sent
 elsewhere.  The client MUST be prepared to receive responses on any
 address and port that would be selected by a server based on the
 procedures described in Section 5 of [4].
 For multicast, the client transport MUST be prepared to receive
 responses on the same multicast group and port to which the request
 is sent (that is, it needs to be a member of the multicast group it
 sent the request to.)
 If a request is destined to an IP address, port, and transport to
 which an existing connection is open, it is RECOMMENDED that this
 connection be used to send the request, but another connection MAY be
 opened and used.
 If a request is sent using multicast, it is sent to the group
 address, port, and TTL provided by the transport user.  If a request
 is sent using unicast unreliable transports, it is sent to the IP
 address and port provided by the transport user.

18.1.2 Receiving Responses

 When a response is received, the client transport examines the top
 Via header field value.  If the value of the "sent-by" parameter in
 that header field value does not correspond to a value that the
 client transport is configured to insert into requests, the response
 MUST be silently discarded.
 If there are any client transactions in existence, the client
 transport uses the matching procedures of Section 17.1.3 to attempt
 to match the response to an existing transaction.  If there is a
 match, the response MUST be passed to that transaction.  Otherwise,
 the response MUST be passed to the core (whether it be stateless
 proxy, stateful proxy, or UA) for further processing.  Handling of
 these "stray" responses is dependent on the core (a proxy will
 forward them, while a UA will discard, for example).

Rosenberg, et. al. Standards Track [Page 144] RFC 3261 SIP: Session Initiation Protocol June 2002

18.2 Servers

18.2.1 Receiving Requests

 A server SHOULD be prepared to receive requests on any IP address,
 port and transport combination that can be the result of a DNS lookup
 on a SIP or SIPS URI [4] that is handed out for the purposes of
 communicating with that server.  In this context, "handing out"
 includes placing a URI in a Contact header field in a REGISTER
 request or a redirect response, or in a Record-Route header field in
 a request or response.  A URI can also be "handed out" by placing it
 on a web page or business card.  It is also RECOMMENDED that a server
 listen for requests on the default SIP ports (5060 for TCP and UDP,
 5061 for TLS over TCP) on all public interfaces.  The typical
 exception would be private networks, or when multiple server
 instances are running on the same host.  For any port and interface
 that a server listens on for UDP, it MUST listen on that same port
 and interface for TCP.  This is because a message may need to be sent
 using TCP, rather than UDP, if it is too large.  As a result, the
 converse is not true.  A server need not listen for UDP on a
 particular address and port just because it is listening on that same
 address and port for TCP.  There may, of course, be other reasons why
 a server needs to listen for UDP on a particular address and port.
 When the server transport receives a request over any transport, it
 MUST examine the value of the "sent-by" parameter in the top Via
 header field value.  If the host portion of the "sent-by" parameter
 contains a domain name, or if it contains an IP address that differs
 from the packet source address, the server MUST add a "received"
 parameter to that Via header field value.  This parameter MUST
 contain the source address from which the packet was received.  This
 is to assist the server transport layer in sending the response,
 since it must be sent to the source IP address from which the request
 came.
 Consider a request received by the server transport which looks like,
 in part:
    INVITE sip:bob@Biloxi.com SIP/2.0
    Via: SIP/2.0/UDP bobspc.biloxi.com:5060
 The request is received with a source IP address of 192.0.2.4.
 Before passing the request up, the transport adds a "received"
 parameter, so that the request would look like, in part:
    INVITE sip:bob@Biloxi.com SIP/2.0
    Via: SIP/2.0/UDP bobspc.biloxi.com:5060;received=192.0.2.4

Rosenberg, et. al. Standards Track [Page 145] RFC 3261 SIP: Session Initiation Protocol June 2002

 Next, the server transport attempts to match the request to a server
 transaction.  It does so using the matching rules described in
 Section 17.2.3.  If a matching server transaction is found, the
 request is passed to that transaction for processing.  If no match is
 found, the request is passed to the core, which may decide to
 construct a new server transaction for that request.  Note that when
 a UAS core sends a 2xx response to INVITE, the server transaction is
 destroyed.  This means that when the ACK arrives, there will be no
 matching server transaction, and based on this rule, the ACK is
 passed to the UAS core, where it is processed.

18.2.2 Sending Responses

 The server transport uses the value of the top Via header field in
 order to determine where to send a response.  It MUST follow the
 following process:
    o  If the "sent-protocol" is a reliable transport protocol such as
       TCP or SCTP, or TLS over those, the response MUST be sent using
       the existing connection to the source of the original request
       that created the transaction, if that connection is still open.
       This requires the server transport to maintain an association
       between server transactions and transport connections.  If that
       connection is no longer open, the server SHOULD open a
       connection to the IP address in the "received" parameter, if
       present, using the port in the "sent-by" value, or the default
       port for that transport, if no port is specified.  If that
       connection attempt fails, the server SHOULD use the procedures
       in [4] for servers in order to determine the IP address and
       port to open the connection and send the response to.
    o  Otherwise, if the Via header field value contains a "maddr"
       parameter, the response MUST be forwarded to the address listed
       there, using the port indicated in "sent-by", or port 5060 if
       none is present.  If the address is a multicast address, the
       response SHOULD be sent using the TTL indicated in the "ttl"
       parameter, or with a TTL of 1 if that parameter is not present.
    o  Otherwise (for unreliable unicast transports), if the top Via
       has a "received" parameter, the response MUST be sent to the
       address in the "received" parameter, using the port indicated
       in the "sent-by" value, or using port 5060 if none is specified
       explicitly.  If this fails, for example, elicits an ICMP "port
       unreachable" response, the procedures of Section 5 of [4]
       SHOULD be used to determine where to send the response.

Rosenberg, et. al. Standards Track [Page 146] RFC 3261 SIP: Session Initiation Protocol June 2002

    o  Otherwise, if it is not receiver-tagged, the response MUST be
       sent to the address indicated by the "sent-by" value, using the
       procedures in Section 5 of [4].

18.3 Framing

 In the case of message-oriented transports (such as UDP), if the
 message has a Content-Length header field, the message body is
 assumed to contain that many bytes.  If there are additional bytes in
 the transport packet beyond the end of the body, they MUST be
 discarded.  If the transport packet ends before the end of the
 message body, this is considered an error.  If the message is a
 response, it MUST be discarded.  If the message is a request, the
 element SHOULD generate a 400 (Bad Request) response.  If the message
 has no Content-Length header field, the message body is assumed to
 end at the end of the transport packet.
 In the case of stream-oriented transports such as TCP, the Content-
 Length header field indicates the size of the body.  The Content-
 Length header field MUST be used with stream oriented transports.

18.4 Error Handling

 Error handling is independent of whether the message was a request or
 response.
 If the transport user asks for a message to be sent over an
 unreliable transport, and the result is an ICMP error, the behavior
 depends on the type of ICMP error.  Host, network, port or protocol
 unreachable errors, or parameter problem errors SHOULD cause the
 transport layer to inform the transport user of a failure in sending.
 Source quench and TTL exceeded ICMP errors SHOULD be ignored.
 If the transport user asks for a request to be sent over a reliable
 transport, and the result is a connection failure, the transport
 layer SHOULD inform the transport user of a failure in sending.

19 Common Message Components

 There are certain components of SIP messages that appear in various
 places within SIP messages (and sometimes, outside of them) that
 merit separate discussion.

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19.1 SIP and SIPS Uniform Resource Indicators

 A SIP or SIPS URI identifies a communications resource.  Like all
 URIs, SIP and SIPS URIs may be placed in web pages, email messages,
 or printed literature.  They contain sufficient information to
 initiate and maintain a communication session with the resource.
 Examples of communications resources include the following:
    o  a user of an online service
    o  an appearance on a multi-line phone
    o  a mailbox on a messaging system
    o  a PSTN number at a gateway service
    o  a group (such as "sales" or "helpdesk") in an organization
 A SIPS URI specifies that the resource be contacted securely.  This
 means, in particular, that TLS is to be used between the UAC and the
 domain that owns the URI.  From there, secure communications are used
 to reach the user, where the specific security mechanism depends on
 the policy of the domain.  Any resource described by a SIP URI can be
 "upgraded" to a SIPS URI by just changing the scheme, if it is
 desired to communicate with that resource securely.

19.1.1 SIP and SIPS URI Components

 The "sip:" and "sips:" schemes follow the guidelines in RFC 2396 [5].
 They use a form similar to the mailto URL, allowing the specification
 of SIP request-header fields and the SIP message-body.  This makes it
 possible to specify the subject, media type, or urgency of sessions
 initiated by using a URI on a web page or in an email message.  The
 formal syntax for a SIP or SIPS URI is presented in Section 25.  Its
 general form, in the case of a SIP URI, is:
    sip:user:password@host:port;uri-parameters?headers
 The format for a SIPS URI is the same, except that the scheme is
 "sips" instead of sip.  These tokens, and some of the tokens in their
 expansions, have the following meanings:
    user: The identifier of a particular resource at the host being
       addressed.  The term "host" in this context frequently refers
       to a domain.  The "userinfo" of a URI consists of this user
       field, the password field, and the @ sign following them.  The
       userinfo part of a URI is optional and MAY be absent when the

Rosenberg, et. al. Standards Track [Page 148] RFC 3261 SIP: Session Initiation Protocol June 2002

       destination host does not have a notion of users or when the
       host itself is the resource being identified.  If the @ sign is
       present in a SIP or SIPS URI, the user field MUST NOT be empty.
       If the host being addressed can process telephone numbers, for
       instance, an Internet telephony gateway, a telephone-
       subscriber field defined in RFC 2806 [9] MAY be used to
       populate the user field.  There are special escaping rules for
       encoding telephone-subscriber fields in SIP and SIPS URIs
       described in Section 19.1.2.
    password: A password associated with the user.  While the SIP and
       SIPS URI syntax allows this field to be present, its use is NOT
       RECOMMENDED, because the passing of authentication information
       in clear text (such as URIs) has proven to be a security risk
       in almost every case where it has been used.  For instance,
       transporting a PIN number in this field exposes the PIN.
       Note that the password field is just an extension of the user
       portion.  Implementations not wishing to give special
       significance to the password portion of the field MAY simply
       treat "user:password" as a single string.
    host: The host providing the SIP resource.  The host part contains
       either a fully-qualified domain name or numeric IPv4 or IPv6
       address.  Using the fully-qualified domain name form is
       RECOMMENDED whenever possible.
    port: The port number where the request is to be sent.
    URI parameters: Parameters affecting a request constructed from
       the URI.
       URI parameters are added after the hostport component and are
       separated by semi-colons.
       URI parameters take the form:
          parameter-name "=" parameter-value
       Even though an arbitrary number of URI parameters may be
       included in a URI, any given parameter-name MUST NOT appear
       more than once.
       This extensible mechanism includes the transport, maddr, ttl,
       user, method and lr parameters.

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       The transport parameter determines the transport mechanism to
       be used for sending SIP messages, as specified in [4].  SIP can
       use any network transport protocol.  Parameter names are
       defined for UDP (RFC 768 [14]), TCP (RFC 761 [15]), and SCTP
       (RFC 2960 [16]).  For a SIPS URI, the transport parameter MUST
       indicate a reliable transport.
       The maddr parameter indicates the server address to be
       contacted for this user, overriding any address derived from
       the host field.  When an maddr parameter is present, the port
       and transport components of the URI apply to the address
       indicated in the maddr parameter value.  [4] describes the
       proper interpretation of the transport, maddr, and hostport in
       order to obtain the destination address, port, and transport
       for sending a request.
       The maddr field has been used as a simple form of loose source
       routing.  It allows a URI to specify a proxy that must be
       traversed en-route to the destination.  Continuing to use the
       maddr parameter this way is strongly discouraged (the
       mechanisms that enable it are deprecated).  Implementations
       should instead use the Route mechanism described in this
       document, establishing a pre-existing route set if necessary
       (see Section 8.1.1.1).  This provides a full URI to describe
       the node to be traversed.
       The ttl parameter determines the time-to-live value of the UDP
       multicast packet and MUST only be used if maddr is a multicast
       address and the transport protocol is UDP.  For example, to
       specify a call to alice@atlanta.com using multicast to
       239.255.255.1 with a ttl of 15, the following URI would be
       used:
          sip:alice@atlanta.com;maddr=239.255.255.1;ttl=15
       The set of valid telephone-subscriber strings is a subset of
       valid user strings.  The user URI parameter exists to
       distinguish telephone numbers from user names that happen to
       look like telephone numbers.  If the user string contains a
       telephone number formatted as a telephone-subscriber, the user
       parameter value "phone" SHOULD be present.  Even without this
       parameter, recipients of SIP and SIPS URIs MAY interpret the
       pre-@ part as a telephone number if local restrictions on the
       name space for user name allow it.
       The method of the SIP request constructed from the URI can be
       specified with the method parameter.

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       The lr parameter, when present, indicates that the element
       responsible for this resource implements the routing mechanisms
       specified in this document.  This parameter will be used in the
       URIs proxies place into Record-Route header field values, and
       may appear in the URIs in a pre-existing route set.
       This parameter is used to achieve backwards compatibility with
       systems implementing the strict-routing mechanisms of RFC 2543
       and the rfc2543bis drafts up to bis-05.  An element preparing
       to send a request based on a URI not containing this parameter
       can assume the receiving element implements strict-routing and
       reformat the message to preserve the information in the
       Request-URI.
       Since the uri-parameter mechanism is extensible, SIP elements
       MUST silently ignore any uri-parameters that they do not
       understand.
    Headers: Header fields to be included in a request constructed
       from the URI.
       Headers fields in the SIP request can be specified with the "?"
       mechanism within a URI.  The header names and values are
       encoded in ampersand separated hname = hvalue pairs.  The
       special hname "body" indicates that the associated hvalue is
       the message-body of the SIP request.
 Table 1 summarizes the use of SIP and SIPS URI components based on
 the context in which the URI appears.  The external column describes
 URIs appearing anywhere outside of a SIP message, for instance on a
 web page or business card.  Entries marked "m" are mandatory, those
 marked "o" are optional, and those marked "-" are not allowed.
 Elements processing URIs SHOULD ignore any disallowed components if
 they are present.  The second column indicates the default value of
 an optional element if it is not present.  "--" indicates that the
 element is either not optional, or has no default value.
 URIs in Contact header fields have different restrictions depending
 on the context in which the header field appears.  One set applies to
 messages that establish and maintain dialogs (INVITE and its 200 (OK)
 response).  The other applies to registration and redirection
 messages (REGISTER, its 200 (OK) response, and 3xx class responses to
 any method).

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19.1.2 Character Escaping Requirements

                                                     dialog
                                        reg./redir. Contact/
            default  Req.-URI  To  From  Contact   R-R/Route  external

user – o o o o o o password – o o o o o o host – m m m m m m port (1) o - - o o o user-param ip o o o o o o method INVITE - - - - - o maddr-param – o - - o o o ttl-param 1 o - - o - o transp.-param (2) o - - o o o lr-param – o - - - o o other-param – o o o o o o headers – - - - o - o

 (1): The default port value is transport and scheme dependent.  The
 default  is  5060  for  sip: using UDP, TCP, or SCTP.  The default is
 5061 for sip: using TLS over TCP and sips: over TCP.
 (2): The default transport is scheme dependent.  For sip:, it is UDP.
 For sips:, it is TCP.
 Table 1: Use and default values of URI components for SIP header
 field values, Request-URI and references
 SIP follows the requirements and guidelines of RFC 2396 [5] when
 defining the set of characters that must be escaped in a SIP URI, and
 uses its ""%" HEX HEX" mechanism for escaping.  From RFC 2396 [5]:
    The set of characters actually reserved within any given URI
    component is defined by that component.  In general, a character
    is reserved if the semantics of the URI changes if the character
    is replaced with its escaped US-ASCII encoding [5].  Excluded US-
    ASCII characters (RFC 2396 [5]), such as space and control
    characters and characters used as URI delimiters, also MUST be
    escaped.  URIs MUST NOT contain unescaped space and control
    characters.
 For each component, the set of valid BNF expansions defines exactly
 which characters may appear unescaped.  All other characters MUST be
 escaped.
 For example, "@" is not in the set of characters in the user
 component, so the user "j@s0n" must have at least the @ sign encoded,
 as in "j%40s0n".

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 Expanding the hname and hvalue tokens in Section 25 show that all URI
 reserved characters in header field names and values MUST be escaped.
 The telephone-subscriber subset of the user component has special
 escaping considerations.  The set of characters not reserved in the
 RFC 2806 [9] description of telephone-subscriber contains a number of
 characters in various syntax elements that need to be escaped when
 used in SIP URIs.  Any characters occurring in a telephone-subscriber
 that do not appear in an expansion of the BNF for the user rule MUST
 be escaped.
 Note that character escaping is not allowed in the host component of
 a SIP or SIPS URI (the % character is not valid in its expansion).
 This is likely to change in the future as requirements for
 Internationalized Domain Names are finalized.  Current
 implementations MUST NOT attempt to improve robustness by treating
 received escaped characters in the host component as literally
 equivalent to their unescaped counterpart.  The behavior required to
 meet the requirements of IDN may be significantly different.

19.1.3 Example SIP and SIPS URIs

 sip:alice@atlanta.com
 sip:alice:secretword@atlanta.com;transport=tcp
 sips:alice@atlanta.com?subject=project%20x&priority=urgent
 sip:+1-212-555-1212:1234@gateway.com;user=phone
 sips:1212@gateway.com
 sip:alice@192.0.2.4
 sip:atlanta.com;method=REGISTER?to=alice%40atlanta.com
 sip:alice;day=tuesday@atlanta.com
 The last sample URI above has a user field value of
 "alice;day=tuesday".  The escaping rules defined above allow a
 semicolon to appear unescaped in this field.  For the purposes of
 this protocol, the field is opaque.  The structure of that value is
 only useful to the SIP element responsible for the resource.

19.1.4 URI Comparison

 Some operations in this specification require determining whether two
 SIP or SIPS URIs are equivalent.  In this specification, registrars
 need to compare bindings in Contact URIs in REGISTER requests (see
 Section 10.3.).  SIP and SIPS URIs are compared for equality
 according to the following rules:
    o  A SIP and SIPS URI are never equivalent.

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    o  Comparison of the userinfo of SIP and SIPS URIs is case-
       sensitive.  This includes userinfo containing passwords or
       formatted as telephone-subscribers.  Comparison of all other
       components of the URI is case-insensitive unless explicitly
       defined otherwise.
    o  The ordering of parameters and header fields is not significant
       in comparing SIP and SIPS URIs.
    o  Characters other than those in the "reserved" set (see RFC 2396
       [5]) are equivalent to their ""%" HEX HEX" encoding.
    o  An IP address that is the result of a DNS lookup of a host name
       does not match that host name.
    o  For two URIs to be equal, the user, password, host, and port
       components must match.
       A URI omitting the user component will not match a URI that
       includes one.  A URI omitting the password component will not
       match a URI that includes one.
       A URI omitting any component with a default value will not
       match a URI explicitly containing that component with its
       default value.  For instance, a URI omitting the optional port
       component will not match a URI explicitly declaring port 5060.
       The same is true for the transport-parameter, ttl-parameter,
       user-parameter, and method components.
          Defining sip:user@host to not be equivalent to
          sip:user@host:5060 is a change from RFC 2543.  When deriving
          addresses from URIs, equivalent addresses are expected from
          equivalent URIs.  The URI sip:user@host:5060 will always
          resolve to port 5060.  The URI sip:user@host may resolve to
          other ports through the DNS SRV mechanisms detailed in [4].
    o  URI uri-parameter components are compared as follows:
  1. Any uri-parameter appearing in both URIs must match.
  1. A user, ttl, or method uri-parameter appearing in only one

URI never matches, even if it contains the default value.

  1. A URI that includes an maddr parameter will not match a URI

that contains no maddr parameter.

  1. All other uri-parameters appearing in only one URI are

ignored when comparing the URIs.

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    o  URI header components are never ignored.  Any present header
       component MUST be present in both URIs and match for the URIs
       to match.  The matching rules are defined for each header field
       in Section 20.
 The URIs within each of the following sets are equivalent:
 sip:%61lice@atlanta.com;transport=TCP
 sip:alice@AtLanTa.CoM;Transport=tcp
 sip:carol@chicago.com
 sip:carol@chicago.com;newparam=5
 sip:carol@chicago.com;security=on
 sip:biloxi.com;transport=tcp;method=REGISTER?to=sip:bob%40biloxi.com
 sip:biloxi.com;method=REGISTER;transport=tcp?to=sip:bob%40biloxi.com
 sip:alice@atlanta.com?subject=project%20x&priority=urgent
 sip:alice@atlanta.com?priority=urgent&subject=project%20x
 The URIs within each of the following sets are not equivalent:
 SIP:ALICE@AtLanTa.CoM;Transport=udp             (different usernames)
 sip:alice@AtLanTa.CoM;Transport=UDP
 sip:bob@biloxi.com                   (can resolve to different ports)
 sip:bob@biloxi.com:5060
 sip:bob@biloxi.com              (can resolve to different transports)
 sip:bob@biloxi.com;transport=udp
 sip:bob@biloxi.com     (can resolve to different port and transports)
 sip:bob@biloxi.com:6000;transport=tcp
 sip:carol@chicago.com                    (different header component)
 sip:carol@chicago.com?Subject=next%20meeting
 sip:bob@phone21.boxesbybob.com   (even though that's what
 sip:bob@192.0.2.4                 phone21.boxesbybob.com resolves to)
 Note that equality is not transitive:
    o  sip:carol@chicago.com and sip:carol@chicago.com;security=on are
       equivalent
    o  sip:carol@chicago.com and sip:carol@chicago.com;security=off
       are equivalent

Rosenberg, et. al. Standards Track [Page 155] RFC 3261 SIP: Session Initiation Protocol June 2002

    o  sip:carol@chicago.com;security=on and
       sip:carol@chicago.com;security=off are not equivalent

19.1.5 Forming Requests from a URI

 An implementation needs to take care when forming requests directly
 from a URI.  URIs from business cards, web pages, and even from
 sources inside the protocol such as registered contacts may contain
 inappropriate header fields or body parts.
 An implementation MUST include any provided transport, maddr, ttl, or
 user parameter in the Request-URI of the formed request.  If the URI
 contains a method parameter, its value MUST be used as the method of
 the request.  The method parameter MUST NOT be placed in the
 Request-URI.  Unknown URI parameters MUST be placed in the message's
 Request-URI.
 An implementation SHOULD treat the presence of any headers or body
 parts in the URI as a desire to include them in the message, and
 choose to honor the request on a per-component basis.
 An implementation SHOULD NOT honor these obviously dangerous header
 fields: From, Call-ID, CSeq, Via, and Record-Route.
 An implementation SHOULD NOT honor any requested Route header field
 values in order to not be used as an unwitting agent in malicious
 attacks.
 An implementation SHOULD NOT honor requests to include header fields
 that may cause it to falsely advertise its location or capabilities.
 These include: Accept, Accept-Encoding, Accept-Language, Allow,
 Contact (in its dialog usage), Organization, Supported, and User-
 Agent.
 An implementation SHOULD verify the accuracy of any requested
 descriptive header fields, including: Content-Disposition, Content-
 Encoding, Content-Language, Content-Length, Content-Type, Date,
 Mime-Version, and Timestamp.
 If the request formed from constructing a message from a given URI is
 not a valid SIP request, the URI is invalid.  An implementation MUST
 NOT proceed with transmitting the request.  It should instead pursue
 the course of action due an invalid URI in the context it occurs.
    The constructed request can be invalid in many ways.  These
    include, but are not limited to, syntax error in header fields,
    invalid combinations of URI parameters, or an incorrect
    description of the message body.

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 Sending a request formed from a given URI may require capabilities
 unavailable to the implementation.  The URI might indicate use of an
 unimplemented transport or extension, for example.  An implementation
 SHOULD refuse to send these requests rather than modifying them to
 match their capabilities.  An implementation MUST NOT send a request
 requiring an extension that it does not support.
    For example, such a request can be formed through the presence of
    a Require header parameter or a method URI parameter with an
    unknown or explicitly unsupported value.

19.1.6 Relating SIP URIs and tel URLs

 When a tel URL (RFC 2806 [9]) is converted to a SIP or SIPS URI, the
 entire telephone-subscriber portion of the tel URL, including any
 parameters, is placed into the userinfo part of the SIP or SIPS URI.
 Thus, tel:+358-555-1234567;postd=pp22 becomes
    sip:+358-555-1234567;postd=pp22@foo.com;user=phone
 or
    sips:+358-555-1234567;postd=pp22@foo.com;user=phone
 not
    sip:+358-555-1234567@foo.com;postd=pp22;user=phone
 or
    sips:+358-555-1234567@foo.com;postd=pp22;user=phone
 In general, equivalent "tel" URLs converted to SIP or SIPS URIs in
 this fashion may not produce equivalent SIP or SIPS URIs.  The
 userinfo of SIP and SIPS URIs are compared as a case-sensitive
 string.  Variance in case-insensitive portions of tel URLs and
 reordering of tel URL parameters does not affect tel URL equivalence,
 but does affect the equivalence of SIP URIs formed from them.
 For example,
    tel:+358-555-1234567;postd=pp22
    tel:+358-555-1234567;POSTD=PP22
 are equivalent, while
    sip:+358-555-1234567;postd=pp22@foo.com;user=phone
    sip:+358-555-1234567;POSTD=PP22@foo.com;user=phone

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 are not.
 Likewise,
    tel:+358-555-1234567;postd=pp22;isub=1411
    tel:+358-555-1234567;isub=1411;postd=pp22
 are equivalent, while
    sip:+358-555-1234567;postd=pp22;isub=1411@foo.com;user=phone
    sip:+358-555-1234567;isub=1411;postd=pp22@foo.com;user=phone
 are not.
 To mitigate this problem, elements constructing telephone-subscriber
 fields to place in the userinfo part of a SIP or SIPS URI SHOULD fold
 any case-insensitive portion of telephone-subscriber to lower case,
 and order the telephone-subscriber parameters lexically by parameter
 name, excepting isdn-subaddress and post-dial, which occur first and
 in that order.  (All components of a tel URL except for future-
 extension parameters are defined to be compared case-insensitive.)
 Following this suggestion, both
    tel:+358-555-1234567;postd=pp22
    tel:+358-555-1234567;POSTD=PP22
    become
      sip:+358-555-1234567;postd=pp22@foo.com;user=phone
 and both
      tel:+358-555-1234567;tsp=a.b;phone-context=5
      tel:+358-555-1234567;phone-context=5;tsp=a.b
    become
      sip:+358-555-1234567;phone-context=5;tsp=a.b@foo.com;user=phone

19.2 Option Tags

 Option tags are unique identifiers used to designate new options
 (extensions) in SIP.  These tags are used in Require (Section 20.32),
 Proxy-Require (Section 20.29), Supported (Section 20.37) and
 Unsupported (Section 20.40) header fields.  Note that these options
 appear as parameters in those header fields in an option-tag = token
 form (see Section 25 for the definition of token).

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 Option tags are defined in standards track RFCs.  This is a change
 from past practice, and is instituted to ensure continuing multi-
 vendor interoperability (see discussion in Section 20.32 and Section
 20.37).  An IANA registry of option tags is used to ensure easy
 reference.

19.3 Tags

 The "tag" parameter is used in the To and From header fields of SIP
 messages.  It serves as a general mechanism to identify a dialog,
 which is the combination of the Call-ID along with two tags, one from
 each participant in the dialog.  When a UA sends a request outside of
 a dialog, it contains a From tag only, providing "half" of the dialog
 ID.  The dialog is completed from the response(s), each of which
 contributes the second half in the To header field.  The forking of
 SIP requests means that multiple dialogs can be established from a
 single request.  This also explains the need for the two-sided dialog
 identifier; without a contribution from the recipients, the
 originator could not disambiguate the multiple dialogs established
 from a single request.
 When a tag is generated by a UA for insertion into a request or
 response, it MUST be globally unique and cryptographically random
 with at least 32 bits of randomness.  A property of this selection
 requirement is that a UA will place a different tag into the From
 header of an INVITE than it would place into the To header of the
 response to the same INVITE.  This is needed in order for a UA to
 invite itself to a session, a common case for "hairpinning" of calls
 in PSTN gateways.  Similarly, two INVITEs for different calls will
 have different From tags, and two responses for different calls will
 have different To tags.
 Besides the requirement for global uniqueness, the algorithm for
 generating a tag is implementation-specific.  Tags are helpful in
 fault tolerant systems, where a dialog is to be recovered on an
 alternate server after a failure.  A UAS can select the tag in such a
 way that a backup can recognize a request as part of a dialog on the
 failed server, and therefore determine that it should attempt to
 recover the dialog and any other state associated with it.

20 Header Fields

 The general syntax for header fields is covered in Section 7.3.  This
 section lists the full set of header fields along with notes on
 syntax, meaning, and usage.  Throughout this section, we use [HX.Y]
 to refer to Section X.Y of the current HTTP/1.1 specification RFC
 2616 [8].  Examples of each header field are given.

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 Information about header fields in relation to methods and proxy
 processing is summarized in Tables 2 and 3.
 The "where" column describes the request and response types in which
 the header field can be used.  Values in this column are:
    R: header field may only appear in requests;
    r: header field may only appear in responses;
    2xx, 4xx, etc.: A numerical value or range indicates response
         codes with which the header field can be used;
    c: header field is copied from the request to the response.
    An empty entry in the "where" column indicates that the header
         field may be present in all requests and responses.
 The "proxy" column describes the operations a proxy may perform on a
 header field:
    a: A proxy can add or concatenate the header field if not present.
    m: A proxy can modify an existing header field value.
    d: A proxy can delete a header field value.
    r: A proxy must be able to read the header field, and thus this
         header field cannot be encrypted.
 The next six columns relate to the presence of a header field in a
 method:
    c: Conditional; requirements on the header field depend on the
         context of the message.
    m: The header field is mandatory.
    m*: The header field SHOULD be sent, but clients/servers need to
         be prepared to receive messages without that header field.
    o: The header field is optional.
    t: The header field SHOULD be sent, but clients/servers need to be
         prepared to receive messages without that header field.
         If a stream-based protocol (such as TCP) is used as a
         transport, then the header field MUST be sent.

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  • : The header field is required if the message body is not empty.

See Sections 20.14, 20.15 and 7.4 for details.

  1. : The header field is not applicable.
 "Optional" means that an element MAY include the header field in a
 request or response, and a UA MAY ignore the header field if present
 in the request or response (The exception to this rule is the Require
 header field discussed in 20.32).  A "mandatory" header field MUST be
 present in a request, and MUST be understood by the UAS receiving the
 request.  A mandatory response header field MUST be present in the
 response, and the header field MUST be understood by the UAC
 processing the response.  "Not applicable" means that the header
 field MUST NOT be present in a request.  If one is placed in a
 request by mistake, it MUST be ignored by the UAS receiving the
 request.  Similarly, a header field labeled "not applicable" for a
 response means that the UAS MUST NOT place the header field in the
 response, and the UAC MUST ignore the header field in the response.
 A UA SHOULD ignore extension header parameters that are not
 understood.
 A compact form of some common header field names is also defined for
 use when overall message size is an issue.
 The Contact, From, and To header fields contain a URI.  If the URI
 contains a comma, question mark or semicolon, the URI MUST be
 enclosed in angle brackets (< and >).  Any URI parameters are
 contained within these brackets.  If the URI is not enclosed in angle
 brackets, any semicolon-delimited parameters are header-parameters,
 not URI parameters.

20.1 Accept

 The Accept header field follows the syntax defined in [H14.1].  The
 semantics are also identical, with the exception that if no Accept
 header field is present, the server SHOULD assume a default value of
 application/sdp.
 An empty Accept header field means that no formats are acceptable.

Rosenberg, et. al. Standards Track [Page 161] RFC 3261 SIP: Session Initiation Protocol June 2002

 Example:
    Header field          where   proxy ACK BYE CAN INV OPT REG
    ___________________________________________________________
    Accept                  R            -   o   -   o   m*  o
    Accept                 2xx           -   -   -   o   m*  o
    Accept                 415           -   c   -   c   c   c
    Accept-Encoding         R            -   o   -   o   o   o
    Accept-Encoding        2xx           -   -   -   o   m*  o
    Accept-Encoding        415           -   c   -   c   c   c
    Accept-Language         R            -   o   -   o   o   o
    Accept-Language        2xx           -   -   -   o   m*  o
    Accept-Language        415           -   c   -   c   c   c
    Alert-Info              R      ar    -   -   -   o   -   -
    Alert-Info             180     ar    -   -   -   o   -   -
    Allow                   R            -   o   -   o   o   o
    Allow                  2xx           -   o   -   m*  m*  o
    Allow                   r            -   o   -   o   o   o
    Allow                  405           -   m   -   m   m   m
    Authentication-Info    2xx           -   o   -   o   o   o
    Authorization           R            o   o   o   o   o   o
    Call-ID                 c       r    m   m   m   m   m   m
    Call-Info                      ar    -   -   -   o   o   o
    Contact                 R            o   -   -   m   o   o
    Contact                1xx           -   -   -   o   -   -
    Contact                2xx           -   -   -   m   o   o
    Contact                3xx      d    -   o   -   o   o   o
    Contact                485           -   o   -   o   o   o
    Content-Disposition                  o   o   -   o   o   o
    Content-Encoding                     o   o   -   o   o   o
    Content-Language                     o   o   -   o   o   o
    Content-Length                 ar    t   t   t   t   t   t
    Content-Type                         *   *   -   *   *   *
    CSeq                    c       r    m   m   m   m   m   m
    Date                            a    o   o   o   o   o   o
    Error-Info           300-699    a    -   o   o   o   o   o
    Expires                              -   -   -   o   -   o
    From                    c       r    m   m   m   m   m   m
    In-Reply-To             R            -   -   -   o   -   -
    Max-Forwards            R      amr   m   m   m   m   m   m
    Min-Expires            423           -   -   -   -   -   m
    MIME-Version                         o   o   -   o   o   o
    Organization                   ar    -   -   -   o   o   o
           Table 2: Summary of header fields, A--O

Rosenberg, et. al. Standards Track [Page 162] RFC 3261 SIP: Session Initiation Protocol June 2002

 Header field              where       proxy ACK BYE CAN INV OPT REG
 ___________________________________________________________________
 Priority                    R          ar    -   -   -   o   -   -
 Proxy-Authenticate         407         ar    -   m   -   m   m   m
 Proxy-Authenticate         401         ar    -   o   o   o   o   o
 Proxy-Authorization         R          dr    o   o   -   o   o   o
 Proxy-Require               R          ar    -   o   -   o   o   o
 Record-Route                R          ar    o   o   o   o   o   -
 Record-Route             2xx,18x       mr    -   o   o   o   o   -
 Reply-To                                     -   -   -   o   -   -
 Require                                ar    -   c   -   c   c   c
 Retry-After          404,413,480,486         -   o   o   o   o   o
                          500,503             -   o   o   o   o   o
                          600,603             -   o   o   o   o   o
 Route                       R          adr   c   c   c   c   c   c
 Server                      r                -   o   o   o   o   o
 Subject                     R                -   -   -   o   -   -
 Supported                   R                -   o   o   m*  o   o
 Supported                  2xx               -   o   o   m*  m*  o
 Timestamp                                    o   o   o   o   o   o
 To                        c(1)          r    m   m   m   m   m   m
 Unsupported                420               -   m   -   m   m   m
 User-Agent                                   o   o   o   o   o   o
 Via                         R          amr   m   m   m   m   m   m
 Via                        rc          dr    m   m   m   m   m   m
 Warning                     r                -   o   o   o   o   o
 WWW-Authenticate           401         ar    -   m   -   m   m   m
 WWW-Authenticate           407         ar    -   o   -   o   o   o
 Table 3: Summary of header fields, P--Z; (1): copied with possible
 addition of tag
    Accept: application/sdp;level=1, application/x-private, text/html

20.2 Accept-Encoding

 The Accept-Encoding header field is similar to Accept, but restricts
 the content-codings [H3.5] that are acceptable in the response.  See
 [H14.3].  The semantics in SIP are identical to those defined in
 [H14.3].
 An empty Accept-Encoding header field is permissible.  It is
 equivalent to Accept-Encoding: identity, that is, only the identity
 encoding, meaning no encoding, is permissible.
 If no Accept-Encoding header field is present, the server SHOULD
 assume a default value of identity.

Rosenberg, et. al. Standards Track [Page 163] RFC 3261 SIP: Session Initiation Protocol June 2002

 This differs slightly from the HTTP definition, which indicates that
 when not present, any encoding can be used, but the identity encoding
 is preferred.
 Example:
    Accept-Encoding: gzip

20.3 Accept-Language

 The Accept-Language header field is used in requests to indicate the
 preferred languages for reason phrases, session descriptions, or
 status responses carried as message bodies in the response.  If no
 Accept-Language header field is present, the server SHOULD assume all
 languages are acceptable to the client.
 The Accept-Language header field follows the syntax defined in
 [H14.4].  The rules for ordering the languages based on the "q"
 parameter apply to SIP as well.
 Example:
    Accept-Language: da, en-gb;q=0.8, en;q=0.7

20.4 Alert-Info

 When present in an INVITE request, the Alert-Info header field
 specifies an alternative ring tone to the UAS.  When present in a 180
 (Ringing) response, the Alert-Info header field specifies an
 alternative ringback tone to the UAC.  A typical usage is for a proxy
 to insert this header field to provide a distinctive ring feature.
 The Alert-Info header field can introduce security risks.  These
 risks and the ways to handle them are discussed in Section 20.9,
 which discusses the Call-Info header field since the risks are
 identical.
 In addition, a user SHOULD be able to disable this feature
 selectively.
    This helps prevent disruptions that could result from the use of
    this header field by untrusted elements.
 Example:
    Alert-Info: <http://www.example.com/sounds/moo.wav>

Rosenberg, et. al. Standards Track [Page 164] RFC 3261 SIP: Session Initiation Protocol June 2002

20.5 Allow

 The Allow header field lists the set of methods supported by the UA
 generating the message.
 All methods, including ACK and CANCEL, understood by the UA MUST be
 included in the list of methods in the Allow header field, when
 present.  The absence of an Allow header field MUST NOT be
 interpreted to mean that the UA sending the message supports no
 methods.   Rather, it implies that the UA is not providing any
 information on what methods it supports.
 Supplying an Allow header field in responses to methods other than
 OPTIONS reduces the number of messages needed.
 Example:
    Allow: INVITE, ACK, OPTIONS, CANCEL, BYE

20.6 Authentication-Info

 The Authentication-Info header field provides for mutual
 authentication with HTTP Digest.  A UAS MAY include this header field
 in a 2xx response to a request that was successfully authenticated
 using digest based on the Authorization header field.
 Syntax and semantics follow those specified in RFC 2617 [17].
 Example:
    Authentication-Info: nextnonce="47364c23432d2e131a5fb210812c"

20.7 Authorization

 The Authorization header field contains authentication credentials of
 a UA.  Section 22.2 overviews the use of the Authorization header
 field, and Section 22.4 describes the syntax and semantics when used
 with HTTP authentication.
 This header field, along with Proxy-Authorization, breaks the general
 rules about multiple header field values.  Although not a comma-
 separated list, this header field name may be present multiple times,
 and MUST NOT be combined into a single header line using the usual
 rules described in Section 7.3.

Rosenberg, et. al. Standards Track [Page 165] RFC 3261 SIP: Session Initiation Protocol June 2002

 In the example below, there are no quotes around the Digest
 parameter:
    Authorization: Digest username="Alice", realm="atlanta.com",
     nonce="84a4cc6f3082121f32b42a2187831a9e",
     response="7587245234b3434cc3412213e5f113a5432"

20.8 Call-ID

 The Call-ID header field uniquely identifies a particular invitation
 or all registrations of a particular client.  A single multimedia
 conference can give rise to several calls with different Call-IDs,
 for example, if a user invites a single individual several times to
 the same (long-running) conference.  Call-IDs are case-sensitive and
 are simply compared byte-by-byte.
 The compact form of the Call-ID header field is i.
 Examples:
    Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@biloxi.com
    i:f81d4fae-7dec-11d0-a765-00a0c91e6bf6@192.0.2.4

20.9 Call-Info

 The Call-Info header field provides additional information about the
 caller or callee, depending on whether it is found in a request or
 response.  The purpose of the URI is described by the "purpose"
 parameter.  The "icon" parameter designates an image suitable as an
 iconic representation of the caller or callee.  The "info" parameter
 describes the caller or callee in general, for example, through a web
 page.  The "card" parameter provides a business card, for example, in
 vCard [36] or LDIF [37] formats.  Additional tokens can be registered
 using IANA and the procedures in Section 27.
 Use of the Call-Info header field can pose a security risk.  If a
 callee fetches the URIs provided by a malicious caller, the callee
 may be at risk for displaying inappropriate or offensive content,
 dangerous or illegal content, and so on.  Therefore, it is
 RECOMMENDED that a UA only render the information in the Call-Info
 header field if it can verify the authenticity of the element that
 originated the header field and trusts that element.  This need not
 be the peer UA; a proxy can insert this header field into requests.
 Example:
 Call-Info: <http://wwww.example.com/alice/photo.jpg> ;purpose=icon,
   <http://www.example.com/alice/> ;purpose=info

Rosenberg, et. al. Standards Track [Page 166] RFC 3261 SIP: Session Initiation Protocol June 2002

20.10 Contact

 A Contact header field value provides a URI whose meaning depends on
 the type of request or response it is in.
 A Contact header field value can contain a display name, a URI with
 URI parameters, and header parameters.
 This document defines the Contact parameters "q" and "expires".
 These parameters are only used when the Contact is present in a
 REGISTER request or response, or in a 3xx response.  Additional
 parameters may be defined in other specifications.
 When the header field value contains a display name, the URI
 including all URI parameters is enclosed in "<" and ">".  If no "<"
 and ">" are present, all parameters after the URI are header
 parameters, not URI parameters.  The display name can be tokens, or a
 quoted string, if a larger character set is desired.
 Even if the "display-name" is empty, the "name-addr" form MUST be
 used if the "addr-spec" contains a comma, semicolon, or question
 mark.  There may or may not be LWS between the display-name and the
 "<".
 These rules for parsing a display name, URI and URI parameters, and
 header parameters also apply for the header fields To and From.
    The Contact header field has a role similar to the Location header
    field in HTTP.  However, the HTTP header field only allows one
    address, unquoted.  Since URIs can contain commas and semicolons
    as reserved characters, they can be mistaken for header or
    parameter delimiters, respectively.
 The compact form of the Contact header field is m (for "moved").
 Examples:
    Contact: "Mr. Watson" <sip:watson@worcester.bell-telephone.com>
       ;q=0.7; expires=3600,
       "Mr. Watson" <mailto:watson@bell-telephone.com> ;q=0.1
    m: <sips:bob@192.0.2.4>;expires=60

Rosenberg, et. al. Standards Track [Page 167] RFC 3261 SIP: Session Initiation Protocol June 2002

20.11 Content-Disposition

 The Content-Disposition header field describes how the message body
 or, for multipart messages, a message body part is to be interpreted
 by the UAC or UAS.  This SIP header field extends the MIME Content-
 Type (RFC 2183 [18]).
 Several new "disposition-types" of the Content-Disposition header are
 defined by SIP.  The value "session" indicates that the body part
 describes a session, for either calls or early (pre-call) media.  The
 value "render" indicates that the body part should be displayed or
 otherwise rendered to the user.  Note that the value "render" is used
 rather than "inline" to avoid the connotation that the MIME body is
 displayed as a part of the rendering of the entire message (since the
 MIME bodies of SIP messages oftentimes are not displayed to users).
 For backward-compatibility, if the Content-Disposition header field
 is missing, the server SHOULD assume bodies of Content-Type
 application/sdp are the disposition "session", while other content
 types are "render".
 The disposition type "icon" indicates that the body part contains an
 image suitable as an iconic representation of the caller or callee
 that could be rendered informationally by a user agent when a message
 has been received, or persistently while a dialog takes place.  The
 value "alert" indicates that the body part contains information, such
 as an audio clip, that should be rendered by the user agent in an
 attempt to alert the user to the receipt of a request, generally a
 request that initiates a dialog; this alerting body could for example
 be rendered as a ring tone for a phone call after a 180 Ringing
 provisional response has been sent.
 Any MIME body with a "disposition-type" that renders content to the
 user should only be processed when a message has been properly
 authenticated.
 The handling parameter, handling-param, describes how the UAS should
 react if it receives a message body whose content type or disposition
 type it does not understand.  The parameter has defined values of
 "optional" and "required".  If the handling parameter is missing, the
 value "required" SHOULD be assumed.  The handling parameter is
 described in RFC 3204 [19].
 If this header field is missing, the MIME type determines the default
 content disposition.  If there is none, "render" is assumed.
 Example:
    Content-Disposition: session

Rosenberg, et. al. Standards Track [Page 168] RFC 3261 SIP: Session Initiation Protocol June 2002

20.12 Content-Encoding

 The Content-Encoding header field is used as a modifier to the
 "media-type".  When present, its value indicates what additional
 content codings have been applied to the entity-body, and thus what
 decoding mechanisms MUST be applied in order to obtain the media-type
 referenced by the Content-Type header field.  Content-Encoding is
 primarily used to allow a body to be compressed without losing the
 identity of its underlying media type.
 If multiple encodings have been applied to an entity-body, the
 content codings MUST be listed in the order in which they were
 applied.
 All content-coding values are case-insensitive.  IANA acts as a
 registry for content-coding value tokens.  See [H3.5] for a
 definition of the syntax for content-coding.
 Clients MAY apply content encodings to the body in requests.  A
 server MAY apply content encodings to the bodies in responses.  The
 server MUST only use encodings listed in the Accept-Encoding header
 field in the request.
 The compact form of the Content-Encoding header field is e.
 Examples:
    Content-Encoding: gzip
    e: tar

20.13 Content-Language

 See [H14.12]. Example:
    Content-Language: fr

20.14 Content-Length

 The Content-Length header field indicates the size of the message-
 body, in decimal number of octets, sent to the recipient.
 Applications SHOULD use this field to indicate the size of the
 message-body to be transferred, regardless of the media type of the
 entity.  If a stream-based protocol (such as TCP) is used as
 transport, the header field MUST be used.
 The size of the message-body does not include the CRLF separating
 header fields and body.  Any Content-Length greater than or equal to
 zero is a valid value.  If no body is present in a message, then the
 Content-Length header field value MUST be set to zero.

Rosenberg, et. al. Standards Track [Page 169] RFC 3261 SIP: Session Initiation Protocol June 2002

    The ability to omit Content-Length simplifies the creation of
    cgi-like scripts that dynamically generate responses.
 The compact form of the header field is l.
 Examples:
    Content-Length: 349
    l: 173

20.15 Content-Type

 The Content-Type header field indicates the media type of the
 message-body sent to the recipient.  The "media-type" element is
 defined in [H3.7].  The Content-Type header field MUST be present if
 the body is not empty.  If the body is empty, and a Content-Type
 header field is present, it indicates that the body of the specific
 type has zero length (for example, an empty audio file).
 The compact form of the header field is c.
 Examples:
    Content-Type: application/sdp
    c: text/html; charset=ISO-8859-4

20.16 CSeq

 A CSeq header field in a request contains a single decimal sequence
 number and the request method.  The sequence number MUST be
 expressible as a 32-bit unsigned integer.  The method part of CSeq is
 case-sensitive.  The CSeq header field serves to order transactions
 within a dialog, to provide a means to uniquely identify
 transactions, and to differentiate between new requests and request
 retransmissions.  Two CSeq header fields are considered equal if the
 sequence number and the request method are identical.  Example:
    CSeq: 4711 INVITE

20.17 Date

 The Date header field contains the date and time.  Unlike HTTP/1.1,
 SIP only supports the most recent RFC 1123 [20] format for dates.  As
 in [H3.3], SIP restricts the time zone in SIP-date to "GMT", while
 RFC 1123 allows any time zone.  An RFC 1123 date is case-sensitive.
 The Date header field reflects the time when the request or response
 is first sent.

Rosenberg, et. al. Standards Track [Page 170] RFC 3261 SIP: Session Initiation Protocol June 2002

    The Date header field can be used by simple end systems without a
    battery-backed clock to acquire a notion of current time.
    However, in its GMT form, it requires clients to know their offset
    from GMT.
 Example:
    Date: Sat, 13 Nov 2010 23:29:00 GMT

20.18 Error-Info

 The Error-Info header field provides a pointer to additional
 information about the error status response.
    SIP UACs have user interface capabilities ranging from pop-up
    windows and audio on PC softclients to audio-only on "black"
    phones or endpoints connected via gateways.  Rather than forcing a
    server generating an error to choose between sending an error
    status code with a detailed reason phrase and playing an audio
    recording, the Error-Info header field allows both to be sent.
    The UAC then has the choice of which error indicator to render to
    the caller.
 A UAC MAY treat a SIP or SIPS URI in an Error-Info header field as if
 it were a Contact in a redirect and generate a new INVITE, resulting
 in a recorded announcement session being established.  A non-SIP URI
 MAY be rendered to the user.
 Examples:
    SIP/2.0 404 The number you have dialed is not in service
    Error-Info: <sip:not-in-service-recording@atlanta.com>

20.19 Expires

 The Expires header field gives the relative time after which the
 message (or content) expires.
 The precise meaning of this is method dependent.
 The expiration time in an INVITE does not affect the duration of the
 actual session that may result from the invitation.  Session
 description protocols may offer the ability to express time limits on
 the session duration, however.
 The value of this field is an integral number of seconds (in decimal)
 between 0 and (2**32)-1, measured from the receipt of the request.

Rosenberg, et. al. Standards Track [Page 171] RFC 3261 SIP: Session Initiation Protocol June 2002

 Example:
    Expires: 5

20.20 From

 The From header field indicates the initiator of the request.  This
 may be different from the initiator of the dialog.  Requests sent by
 the callee to the caller use the callee's address in the From header
 field.
 The optional "display-name" is meant to be rendered by a human user
 interface.  A system SHOULD use the display name "Anonymous" if the
 identity of the client is to remain hidden.  Even if the "display-
 name" is empty, the "name-addr" form MUST be used if the "addr-spec"
 contains a comma, question mark, or semicolon.  Syntax issues are
 discussed in Section 7.3.1.
 Two From header fields are equivalent if their URIs match, and their
 parameters match. Extension parameters in one header field, not
 present in the other are ignored for the purposes of comparison. This
 means that the display name and presence or absence of angle brackets
 do not affect matching.
 See Section 20.10 for the rules for parsing a display name, URI and
 URI parameters, and header field parameters.
 The compact form of the From header field is f.
 Examples:
    From: "A. G. Bell" <sip:agb@bell-telephone.com> ;tag=a48s
    From: sip:+12125551212@server.phone2net.com;tag=887s
    f: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8

20.21 In-Reply-To

 The In-Reply-To header field enumerates the Call-IDs that this call
 references or returns.  These Call-IDs may have been cached by the
 client then included in this header field in a return call.
    This allows automatic call distribution systems to route return
    calls to the originator of the first call.  This also allows
    callees to filter calls, so that only return calls for calls they
    originated will be accepted.  This field is not a substitute for
    request authentication.

Rosenberg, et. al. Standards Track [Page 172] RFC 3261 SIP: Session Initiation Protocol June 2002

 Example:
    In-Reply-To: 70710@saturn.bell-tel.com, 17320@saturn.bell-tel.com

20.22 Max-Forwards

 The Max-Forwards header field must be used with any SIP method to
 limit the number of proxies or gateways that can forward the request
 to the next downstream server.  This can also be useful when the
 client is attempting to trace a request chain that appears to be
 failing or looping in mid-chain.
 The Max-Forwards value is an integer in the range 0-255 indicating
 the remaining number of times this request message is allowed to be
 forwarded.  This count is decremented by each server that forwards
 the request.  The recommended initial value is 70.
 This header field should be inserted by elements that can not
 otherwise guarantee loop detection.  For example, a B2BUA should
 insert a Max-Forwards header field.
 Example:
    Max-Forwards: 6

20.23 Min-Expires

 The Min-Expires header field conveys the minimum refresh interval
 supported for soft-state elements managed by that server.  This
 includes Contact header fields that are stored by a registrar.  The
 header field contains a decimal integer number of seconds from 0 to
 (2**32)-1.  The use of the header field in a 423 (Interval Too Brief)
 response is described in Sections 10.2.8, 10.3, and 21.4.17.
 Example:
    Min-Expires: 60

20.24 MIME-Version

 See [H19.4.1].
 Example:
    MIME-Version: 1.0

Rosenberg, et. al. Standards Track [Page 173] RFC 3261 SIP: Session Initiation Protocol June 2002

20.25 Organization

 The Organization header field conveys the name of the organization to
 which the SIP element issuing the request or response belongs.
    The field MAY be used by client software to filter calls.
 Example:
    Organization: Boxes by Bob

20.26 Priority

 The Priority header field indicates the urgency of the request as
 perceived by the client.  The Priority header field describes the
 priority that the SIP request should have to the receiving human or
 its agent.  For example, it may be factored into decisions about call
 routing and acceptance.  For these decisions, a message containing no
 Priority header field SHOULD be treated as if it specified a Priority
 of "normal".  The Priority header field does not influence the use of
 communications resources such as packet forwarding priority in
 routers or access to circuits in PSTN gateways.  The header field can
 have the values "non-urgent", "normal", "urgent", and "emergency",
 but additional values can be defined elsewhere.  It is RECOMMENDED
 that the value of "emergency" only be used when life, limb, or
 property are in imminent danger.  Otherwise, there are no semantics
 defined for this header field.
    These are the values of RFC 2076 [38], with the addition of
    "emergency".
 Examples:
    Subject: A tornado is heading our way!
    Priority: emergency
 or
    Subject: Weekend plans
    Priority: non-urgent

20.27 Proxy-Authenticate

 A Proxy-Authenticate header field value contains an authentication
 challenge.
 The use of this header field is defined in [H14.33].  See Section
 22.3 for further details on its usage.

Rosenberg, et. al. Standards Track [Page 174] RFC 3261 SIP: Session Initiation Protocol June 2002

 Example:
    Proxy-Authenticate: Digest realm="atlanta.com",
     domain="sip:ss1.carrier.com", qop="auth",
     nonce="f84f1cec41e6cbe5aea9c8e88d359",
     opaque="", stale=FALSE, algorithm=MD5

20.28 Proxy-Authorization

 The Proxy-Authorization header field allows the client to identify
 itself (or its user) to a proxy that requires authentication.  A
 Proxy-Authorization field value consists of credentials containing
 the authentication information of the user agent for the proxy and/or
 realm of the resource being requested.
 See Section 22.3 for a definition of the usage of this header field.
 This header field, along with Authorization, breaks the general rules
 about multiple header field names.  Although not a comma-separated
 list, this header field name may be present multiple times, and MUST
 NOT be combined into a single header line using the usual rules
 described in Section 7.3.1.
 Example:
 Proxy-Authorization: Digest username="Alice", realm="atlanta.com",
    nonce="c60f3082ee1212b402a21831ae",
    response="245f23415f11432b3434341c022"

20.29 Proxy-Require

 The Proxy-Require header field is used to indicate proxy-sensitive
 features that must be supported by the proxy.  See Section 20.32 for
 more details on the mechanics of this message and a usage example.
 Example:
    Proxy-Require: foo

20.30 Record-Route

 The Record-Route header field is inserted by proxies in a request to
 force future requests in the dialog to be routed through the proxy.
 Examples of its use with the Route header field are described in
 Sections 16.12.1.

Rosenberg, et. al. Standards Track [Page 175] RFC 3261 SIP: Session Initiation Protocol June 2002

 Example:
    Record-Route: <sip:server10.biloxi.com;lr>,
                  <sip:bigbox3.site3.atlanta.com;lr>

20.31 Reply-To

 The Reply-To header field contains a logical return URI that may be
 different from the From header field.  For example, the URI MAY be
 used to return missed calls or unestablished sessions.  If the user
 wished to remain anonymous, the header field SHOULD either be omitted
 from the request or populated in such a way that does not reveal any
 private information.
 Even if the "display-name" is empty, the "name-addr" form MUST be
 used if the "addr-spec" contains a comma, question mark, or
 semicolon.  Syntax issues are discussed in Section 7.3.1.
 Example:
    Reply-To: Bob <sip:bob@biloxi.com>

20.32 Require

 The Require header field is used by UACs to tell UASs about options
 that the UAC expects the UAS to support in order to process the
 request.  Although an optional header field, the Require MUST NOT be
 ignored if it is present.
 The Require header field contains a list of option tags, described in
 Section 19.2.  Each option tag defines a SIP extension that MUST be
 understood to process the request.  Frequently, this is used to
 indicate that a specific set of extension header fields need to be
 understood.  A UAC compliant to this specification MUST only include
 option tags corresponding to standards-track RFCs.
 Example:
    Require: 100rel

20.33 Retry-After

 The Retry-After header field can be used with a 500 (Server Internal
 Error) or 503 (Service Unavailable) response to indicate how long the
 service is expected to be unavailable to the requesting client and
 with a 404 (Not Found), 413 (Request Entity Too Large), 480
 (Temporarily Unavailable), 486 (Busy Here), 600 (Busy), or 603

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 (Decline) response to indicate when the called party anticipates
 being available again.  The value of this field is a positive integer
 number of seconds (in decimal) after the time of the response.
 An optional comment can be used to indicate additional information
 about the time of callback.  An optional "duration" parameter
 indicates how long the called party will be reachable starting at the
 initial time of availability.  If no duration parameter is given, the
 service is assumed to be available indefinitely.
 Examples:
    Retry-After: 18000;duration=3600
    Retry-After: 120 (I'm in a meeting)

20.34 Route

 The Route header field is used to force routing for a request through
 the listed set of proxies.  Examples of the use of the Route header
 field are in Section 16.12.1.
 Example:
    Route: <sip:bigbox3.site3.atlanta.com;lr>,
           <sip:server10.biloxi.com;lr>

20.35 Server

 The Server header field contains information about the software used
 by the UAS to handle the request.
 Revealing the specific software version of the server might allow the
 server to become more vulnerable to attacks against software that is
 known to contain security holes.  Implementers SHOULD make the Server
 header field a configurable option.
 Example:
    Server: HomeServer v2

20.36 Subject

 The Subject header field provides a summary or indicates the nature
 of the call, allowing call filtering without having to parse the
 session description.  The session description does not have to use
 the same subject indication as the invitation.
 The compact form of the Subject header field is s.

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 Example:
    Subject: Need more boxes
    s: Tech Support

20.37 Supported

 The Supported header field enumerates all the extensions supported by
 the UAC or UAS.
 The Supported header field contains a list of option tags, described
 in Section 19.2, that are understood by the UAC or UAS.  A UA
 compliant to this specification MUST only include option tags
 corresponding to standards-track RFCs.  If empty, it means that no
 extensions are supported.
 The compact form of the Supported header field is k.
 Example:
    Supported: 100rel

20.38 Timestamp

 The Timestamp header field describes when the UAC sent the request to
 the UAS.
 See Section 8.2.6 for details on how to generate a response to a
 request that contains the header field.  Although there is no
 normative behavior defined here that makes use of the header, it
 allows for extensions or SIP applications to obtain RTT estimates.
 Example:
    Timestamp: 54

20.39 To

 The To header field specifies the logical recipient of the request.
 The optional "display-name" is meant to be rendered by a human-user
 interface.  The "tag" parameter serves as a general mechanism for
 dialog identification.
 See Section 19.3 for details of the "tag" parameter.

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 Comparison of To header fields for equality is identical to
 comparison of From header fields.  See Section 20.10 for the rules
 for parsing a display name, URI and URI parameters, and header field
 parameters.
 The compact form of the To header field is t.
 The following are examples of valid To header fields:
    To: The Operator <sip:operator@cs.columbia.edu>;tag=287447
    t: sip:+12125551212@server.phone2net.com

20.40 Unsupported

 The Unsupported header field lists the features not supported by the
 UAS.  See Section 20.32 for motivation.
 Example:
    Unsupported: foo

20.41 User-Agent

 The User-Agent header field contains information about the UAC
 originating the request.  The semantics of this header field are
 defined in [H14.43].
 Revealing the specific software version of the user agent might allow
 the user agent to become more vulnerable to attacks against software
 that is known to contain security holes.  Implementers SHOULD make
 the User-Agent header field a configurable option.
 Example:
    User-Agent: Softphone Beta1.5

20.42 Via

 The Via header field indicates the path taken by the request so far
 and indicates the path that should be followed in routing responses.
 The branch ID parameter in the Via header field values serves as a
 transaction identifier, and is used by proxies to detect loops.
 A Via header field value contains the transport protocol used to send
 the message, the client's host name or network address, and possibly
 the port number at which it wishes to receive responses.  A Via
 header field value can also contain parameters such as "maddr",
 "ttl", "received", and "branch", whose meaning and use are described

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 in other sections.  For implementations compliant to this
 specification, the value of the branch parameter MUST start with the
 magic cookie "z9hG4bK", as discussed in Section 8.1.1.7.
 Transport protocols defined here are "UDP", "TCP", "TLS", and "SCTP".
 "TLS" means TLS over TCP.  When a request is sent to a SIPS URI, the
 protocol still indicates "SIP", and the transport protocol is TLS.

Via: SIP/2.0/UDP erlang.bell-telephone.com:5060;branch=z9hG4bK87asdks7 Via: SIP/2.0/UDP 192.0.2.1:5060 ;received=192.0.2.207

   ;branch=z9hG4bK77asjd
 The compact form of the Via header field is v.
 In this example, the message originated from a multi-homed host with
 two addresses, 192.0.2.1 and 192.0.2.207.  The sender guessed wrong
 as to which network interface would be used.  Erlang.bell-
 telephone.com noticed the mismatch and added a parameter to the
 previous hop's Via header field value, containing the address that
 the packet actually came from.
 The host or network address and port number are not required to
 follow the SIP URI syntax.  Specifically, LWS on either side of the
 ":" or "/" is allowed, as shown here:
    Via: SIP / 2.0 / UDP first.example.com: 4000;ttl=16
      ;maddr=224.2.0.1 ;branch=z9hG4bKa7c6a8dlze.1
 Even though this specification mandates that the branch parameter be
 present in all requests, the BNF for the header field indicates that
 it is optional.  This allows interoperation with RFC 2543 elements,
 which did not have to insert the branch parameter.
 Two Via header fields are equal if their sent-protocol and sent-by
 fields are equal, both have the same set of parameters, and the
 values of all parameters are equal.

20.43 Warning

 The Warning header field is used to carry additional information
 about the status of a response.  Warning header field values are sent
 with responses and contain a three-digit warning code, host name, and
 warning text.
 The "warn-text" should be in a natural language that is most likely
 to be intelligible to the human user receiving the response.  This
 decision can be based on any available knowledge, such as the
 location of the user, the Accept-Language field in a request, or the

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 Content-Language field in a response.  The default language is i-
 default [21].
 The currently-defined "warn-code"s are listed below, with a
 recommended warn-text in English and a description of their meaning.
 These warnings describe failures induced by the session description.
 The first digit of warning codes beginning with "3" indicates
 warnings specific to SIP.  Warnings 300 through 329 are reserved for
 indicating problems with keywords in the session description, 330
 through 339 are warnings related to basic network services requested
 in the session description, 370 through 379 are warnings related to
 quantitative QoS parameters requested in the session description, and
 390 through 399 are miscellaneous warnings that do not fall into one
 of the above categories.
    300 Incompatible network protocol: One or more network protocols
        contained in the session description are not available.
    301 Incompatible network address formats: One or more network
        address formats contained in the session description are not
        available.
    302 Incompatible transport protocol: One or more transport
        protocols described in the session description are not
        available.
    303 Incompatible bandwidth units: One or more bandwidth
        measurement units contained in the session description were
        not understood.
    304 Media type not available: One or more media types contained in
        the session description are not available.
    305 Incompatible media format: One or more media formats contained
        in the session description are not available.
    306 Attribute not understood: One or more of the media attributes
        in the session description are not supported.
    307 Session description parameter not understood: A parameter
        other than those listed above was not understood.
    330 Multicast not available: The site where the user is located
        does not support multicast.
    331 Unicast not available: The site where the user is located does
        not support unicast communication (usually due to the presence
        of a firewall).

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    370 Insufficient bandwidth: The bandwidth specified in the session
        description or defined by the media exceeds that known to be
        available.
    399 Miscellaneous warning: The warning text can include arbitrary
        information to be presented to a human user or logged.  A
        system receiving this warning MUST NOT take any automated
        action.
           1xx and 2xx have been taken by HTTP/1.1.
 Additional "warn-code"s can be defined through IANA, as defined in
 Section 27.2.
 Examples:
    Warning: 307 isi.edu "Session parameter 'foo' not understood"
    Warning: 301 isi.edu "Incompatible network address type 'E.164'"

20.44 WWW-Authenticate

 A WWW-Authenticate header field value contains an authentication
 challenge.  See Section 22.2 for further details on its usage.
 Example:
    WWW-Authenticate: Digest realm="atlanta.com",
      domain="sip:boxesbybob.com", qop="auth",
      nonce="f84f1cec41e6cbe5aea9c8e88d359",
      opaque="", stale=FALSE, algorithm=MD5

21 Response Codes

 The response codes are consistent with, and extend, HTTP/1.1 response
 codes.  Not all HTTP/1.1 response codes are appropriate, and only
 those that are appropriate are given here.  Other HTTP/1.1 response
 codes SHOULD NOT be used.  Also, SIP defines a new class, 6xx.

21.1 Provisional 1xx

 Provisional responses, also known as informational responses,
 indicate that the server contacted is performing some further action
 and does not yet have a definitive response.  A server sends a 1xx
 response if it expects to take more than 200 ms to obtain a final
 response.  Note that 1xx responses are not transmitted reliably.
 They never cause the client to send an ACK.  Provisional (1xx)
 responses MAY contain message bodies, including session descriptions.

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21.1.1 100 Trying

 This response indicates that the request has been received by the
 next-hop server and that some unspecified action is being taken on
 behalf of this call (for example, a database is being consulted).
 This response, like all other provisional responses, stops
 retransmissions of an INVITE by a UAC.  The 100 (Trying) response is
 different from other provisional responses, in that it is never
 forwarded upstream by a stateful proxy.

21.1.2 180 Ringing

 The UA receiving the INVITE is trying to alert the user.  This
 response MAY be used to initiate local ringback.

21.1.3 181 Call Is Being Forwarded

 A server MAY use this status code to indicate that the call is being
 forwarded to a different set of destinations.

21.1.4 182 Queued

 The called party is temporarily unavailable, but the server has
 decided to queue the call rather than reject it.  When the callee
 becomes available, it will return the appropriate final status
 response.  The reason phrase MAY give further details about the
 status of the call, for example, "5 calls queued; expected waiting
 time is 15 minutes".  The server MAY issue several 182 (Queued)
 responses to update the caller about the status of the queued call.

21.1.5 183 Session Progress

 The 183 (Session Progress) response is used to convey information
 about the progress of the call that is not otherwise classified.  The
 Reason-Phrase, header fields, or message body MAY be used to convey
 more details about the call progress.

21.2 Successful 2xx

 The request was successful.

21.2.1 200 OK

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

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21.3 Redirection 3xx

 3xx responses give information about the user's new location, or
 about alternative services that might be able to satisfy the call.

21.3.1 300 Multiple Choices

 The address in the request resolved to several choices, each with its
 own specific location, and the user (or UA) can select a preferred
 communication end point and redirect its request to that location.
 The response MAY include a message body containing a list of resource
 characteristics and location(s) from which the user or UA can choose
 the one most appropriate, if allowed by the Accept request header
 field.  However, no MIME types have been defined for this message
 body.
 The choices SHOULD also be listed as Contact fields (Section 20.10).
 Unlike HTTP, the SIP response MAY contain several Contact fields or a
 list of addresses in a Contact field.  UAs MAY use the Contact header
 field value for automatic redirection or MAY ask the user to confirm
 a choice.  However, this specification does not define any standard
 for such automatic selection.
    This status response is appropriate if the callee can be reached
    at several different locations and the server cannot or prefers
    not to proxy the request.

21.3.2 301 Moved Permanently

 The user can no longer be found at the address in the Request-URI,
 and the requesting client SHOULD retry at the new address given by
 the Contact header field (Section 20.10).  The requestor SHOULD
 update any local directories, address books, and user location caches
 with this new value and redirect future requests to the address(es)
 listed.

21.3.3 302 Moved Temporarily

 The requesting client SHOULD retry the request at the new address(es)
 given by the Contact header field (Section 20.10).  The Request-URI
 of the new request uses the value of the Contact header field in the
 response.

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 The duration of the validity of the Contact URI can be indicated
 through an Expires (Section 20.19) header field or an expires
 parameter in the Contact header field.  Both proxies and UAs MAY
 cache this URI for the duration of the expiration time.  If there is
 no explicit expiration time, the address is only valid once for
 recursing, and MUST NOT be cached for future transactions.
 If the URI cached from the Contact header field fails, the Request-
 URI from the redirected request MAY be tried again a single time.
    The temporary URI may have become out-of-date sooner than the
    expiration time, and a new temporary URI may be available.

21.3.4 305 Use Proxy

 The requested resource MUST be accessed through the proxy given by
 the Contact field.  The Contact field gives the URI of the proxy.
 The recipient is expected to repeat this single request via the
 proxy.  305 (Use Proxy) responses MUST only be generated by UASs.

21.3.5 380 Alternative Service

 The call was not successful, but alternative services are possible.
 The alternative services are described in the message body of the
 response.  Formats for such bodies are not defined here, and may be
 the subject of future standardization.

21.4 Request Failure 4xx

 4xx responses are definite failure responses from a particular
 server.  The client SHOULD NOT retry the same request without
 modification (for example, adding appropriate authorization).
 However, the same request to a different server might be successful.

21.4.1 400 Bad Request

 The request could not be understood due to malformed syntax.  The
 Reason-Phrase SHOULD identify the syntax problem in more detail, for
 example, "Missing Call-ID header field".

21.4.2 401 Unauthorized

 The request requires user authentication.  This response is issued by
 UASs and registrars, while 407 (Proxy Authentication Required) is
 used by proxy servers.

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21.4.3 402 Payment Required

 Reserved for future use.

21.4.4 403 Forbidden

 The server understood the request, but is refusing to fulfill it.
 Authorization will not help, and the request SHOULD NOT be repeated.

21.4.5 404 Not Found

 The server has definitive information that the user does not exist at
 the domain specified in the Request-URI.  This status is also
 returned if the domain in the Request-URI does not match any of the
 domains handled by the recipient of the request.

21.4.6 405 Method Not Allowed

 The method specified in the Request-Line is understood, but not
 allowed for the address identified by the Request-URI.
 The response MUST include an Allow header field containing a list of
 valid methods for the indicated address.

21.4.7 406 Not Acceptable

 The resource identified by the request is only capable of generating
 response entities that have content characteristics not acceptable
 according to the Accept header field sent in the request.

21.4.8 407 Proxy Authentication Required

 This code is similar to 401 (Unauthorized), but indicates that the
 client MUST first authenticate itself with the proxy.  SIP access
 authentication is explained in Sections 26 and 22.3.
 This status code can be used for applications where access to the
 communication channel (for example, a telephony gateway) rather than
 the callee requires authentication.

21.4.9 408 Request Timeout

 The server could not produce a response within a suitable amount of
 time, for example, if it could not determine the location of the user
 in time.  The client MAY repeat the request without modifications at
 any later time.

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21.4.10 410 Gone

 The requested resource is no longer available at the server and no
 forwarding address is known.  This condition is expected to be
 considered permanent.  If the server does not know, or has no
 facility to determine, whether or not the condition is permanent, the
 status code 404 (Not Found) SHOULD be used instead.

21.4.11 413 Request Entity Too Large

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

21.4.12 414 Request-URI Too Long

 The server is refusing to service the request because the Request-URI
 is longer than the server is willing to interpret.

21.4.13 415 Unsupported Media Type

 The server is refusing to service the request because the message
 body of the request is in a format not supported by the server for
 the requested method.  The server MUST return a list of acceptable
 formats using the Accept, Accept-Encoding, or Accept-Language header
 field, depending on the specific problem with the content.  UAC
 processing of this response is described in Section 8.1.3.5.

21.4.14 416 Unsupported URI Scheme

 The server cannot process the request because the scheme of the URI
 in the Request-URI is unknown to the server.  Client processing of
 this response is described in Section 8.1.3.5.

21.4.15 420 Bad Extension

 The server did not understand the protocol extension specified in a
 Proxy-Require (Section 20.29) or Require (Section 20.32) header
 field.  The server MUST include a list of the unsupported extensions
 in an Unsupported header field in the response.  UAC processing of
 this response is described in Section 8.1.3.5.

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21.4.16 421 Extension Required

 The UAS needs a particular extension to process the request, but this
 extension is not listed in a Supported header field in the request.
 Responses with this status code MUST contain a Require header field
 listing the required extensions.
 A UAS SHOULD NOT use this response unless it truly cannot provide any
 useful service to the client.  Instead, if a desirable extension is
 not listed in the Supported header field, servers SHOULD process the
 request using baseline SIP capabilities and any extensions supported
 by the client.

21.4.17 423 Interval Too Brief

 The server is rejecting the request because the expiration time of
 the resource refreshed by the request is too short.  This response
 can be used by a registrar to reject a registration whose Contact
 header field expiration time was too small.  The use of this response
 and the related Min-Expires header field are described in Sections
 10.2.8, 10.3, and 20.23.

21.4.18 480 Temporarily Unavailable

 The callee's end system was contacted successfully but the callee is
 currently unavailable (for example, is not logged in, logged in but
 in a state that precludes communication with the callee, or has
 activated the "do not disturb" feature).  The response MAY indicate a
 better time to call in the Retry-After header field.  The user could
 also be available elsewhere (unbeknownst to this server).  The reason
 phrase SHOULD indicate a more precise cause as to why the callee is
 unavailable.  This value SHOULD be settable by the UA.  Status 486
 (Busy Here) MAY be used to more precisely indicate a particular
 reason for the call failure.
 This status is also returned by a redirect or proxy server that
 recognizes the user identified by the Request-URI, but does not
 currently have a valid forwarding location for that user.

21.4.19 481 Call/Transaction Does Not Exist

 This status indicates that the UAS received a request that does not
 match any existing dialog or transaction.

21.4.20 482 Loop Detected

 The server has detected a loop (Section 16.3 Item 4).

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21.4.21 483 Too Many Hops

 The server received a request that contains a Max-Forwards (Section
 20.22) header field with the value zero.

21.4.22 484 Address Incomplete

 The server received a request with a Request-URI that was incomplete.
 Additional information SHOULD be provided in the reason phrase.
    This status code allows overlapped dialing.  With overlapped
    dialing, the client does not know the length of the dialing
    string.  It sends strings of increasing lengths, prompting the
    user for more input, until it no longer receives a 484 (Address
    Incomplete) status response.

21.4.23 485 Ambiguous

 The Request-URI was ambiguous.  The response MAY contain a listing of
 possible unambiguous addresses in Contact header fields.  Revealing
 alternatives can infringe on privacy of the user or the organization.
 It MUST be possible to configure a server to respond with status 404
 (Not Found) or to suppress the listing of possible choices for
 ambiguous Request-URIs.
 Example response to a request with the Request-URI
 sip:lee@example.com:
    SIP/2.0 485 Ambiguous
    Contact: Carol Lee <sip:carol.lee@example.com>
    Contact: Ping Lee <sip:p.lee@example.com>
    Contact: Lee M. Foote <sips:lee.foote@example.com>
    Some email and voice mail systems provide this functionality.  A
    status code separate from 3xx is used since the semantics are
    different: for 300, it is assumed that the same person or service
    will be reached by the choices provided.  While an automated
    choice or sequential search makes sense for a 3xx response, user
    intervention is required for a 485 (Ambiguous) response.

21.4.24 486 Busy Here

 The callee's end system was contacted successfully, but the callee is
 currently not willing or able to take additional calls at this end
 system.  The response MAY indicate a better time to call in the
 Retry-After header field.  The user could also be available

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 elsewhere, such as through a voice mail service.  Status 600 (Busy
 Everywhere) SHOULD be used if the client knows that no other end
 system will be able to accept this call.

21.4.25 487 Request Terminated

 The request was terminated by a BYE or CANCEL request.  This response
 is never returned for a CANCEL request itself.

21.4.26 488 Not Acceptable Here

 The response has the same meaning as 606 (Not Acceptable), but only
 applies to the specific resource addressed by the Request-URI and the
 request may succeed elsewhere.
 A message body containing a description of media capabilities MAY be
 present in the response, which is formatted according to the Accept
 header field in the INVITE (or application/sdp if not present), the
 same as a message body in a 200 (OK) response to an OPTIONS request.

21.4.27 491 Request Pending

 The request was received by a UAS that had a pending request within
 the same dialog.  Section 14.2 describes how such "glare" situations
 are resolved.

21.4.28 493 Undecipherable

 The request was received by a UAS that contained an encrypted MIME
 body for which the recipient does not possess or will not provide an
 appropriate decryption key.  This response MAY have a single body
 containing an appropriate public key that should be used to encrypt
 MIME bodies sent to this UA.  Details of the usage of this response
 code can be found in Section 23.2.

21.5 Server Failure 5xx

 5xx responses are failure responses given when a server itself has
 erred.

21.5.1 500 Server Internal Error

 The server encountered an unexpected condition that prevented it from
 fulfilling the request.  The client MAY display the specific error
 condition and MAY retry the request after several seconds.
 If the condition is temporary, the server MAY indicate when the
 client may retry the request using the Retry-After header field.

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21.5.2 501 Not Implemented

 The server does not support the functionality required to fulfill the
 request.  This is the appropriate response when a UAS does not
 recognize the request method and is not capable of supporting it for
 any user.  (Proxies forward all requests regardless of method.)
 Note that a 405 (Method Not Allowed) is sent when the server
 recognizes the request method, but that method is not allowed or
 supported.

21.5.3 502 Bad Gateway

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

21.5.4 503 Service Unavailable

 The server is temporarily unable to process the request due to a
 temporary overloading or maintenance of the server.  The server MAY
 indicate when the client should retry the request in a Retry-After
 header field.  If no Retry-After is given, the client MUST act as if
 it had received a 500 (Server Internal Error) response.
 A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD
 attempt to forward the request to an alternate server.  It SHOULD NOT
 forward any other requests to that server for the duration specified
 in the Retry-After header field, if present.
 Servers MAY refuse the connection or drop the request instead of
 responding with 503 (Service Unavailable).

21.5.5 504 Server Time-out

 The server did not receive a timely response from an external server
 it accessed in attempting to process the request.  408 (Request
 Timeout) should be used instead if there was no response within the
 period specified in the Expires header field from the upstream
 server.

21.5.6 505 Version Not Supported

 The server does not support, or refuses to support, the SIP protocol
 version that was used in the request.  The server is indicating that
 it is unable or unwilling to complete the request using the same
 major version as the client, other than with this error message.

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21.5.7 513 Message Too Large

 The server was unable to process the request since the message length
 exceeded its capabilities.

21.6 Global Failures 6xx

 6xx responses indicate that a server has definitive information about
 a particular user, not just the particular instance indicated in the
 Request-URI.

21.6.1 600 Busy Everywhere

 The callee's end system was contacted successfully but the callee is
 busy and does not wish to take the call at this time.  The response
 MAY indicate a better time to call in the Retry-After header field.
 If the callee does not wish to reveal the reason for declining the
 call, the callee uses status code 603 (Decline) instead.  This status
 response is returned only if the client knows that no other end point
 (such as a voice mail system) will answer the request.  Otherwise,
 486 (Busy Here) should be returned.

21.6.2 603 Decline

 The callee's machine was successfully contacted but the user
 explicitly does not wish to or cannot participate.  The response MAY
 indicate a better time to call in the Retry-After header field.  This
 status response is returned only if the client knows that no other
 end point will answer the request.

21.6.3 604 Does Not Exist Anywhere

 The server has authoritative information that the user indicated in
 the Request-URI does not exist anywhere.

21.6.4 606 Not Acceptable

 The user's agent was contacted successfully but some aspects of the
 session description such as the requested media, bandwidth, or
 addressing style were not acceptable.
 A 606 (Not Acceptable) response means that the user wishes to
 communicate, but cannot adequately support the session described.
 The 606 (Not Acceptable) response MAY contain a list of reasons in a
 Warning header field describing why the session described cannot be
 supported.  Warning reason codes are listed in Section 20.43.

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 A message body containing a description of media capabilities MAY be
 present in the response, which is formatted according to the Accept
 header field in the INVITE (or application/sdp if not present), the
 same as a message body in a 200 (OK) response to an OPTIONS request.
 It is hoped that negotiation will not frequently be needed, and when
 a new user is being invited to join an already existing conference,
 negotiation may not be possible.  It is up to the invitation
 initiator to decide whether or not to act on a 606 (Not Acceptable)
 response.
 This status response is returned only if the client knows that no
 other end point will answer the request.

22 Usage of HTTP Authentication

 SIP provides a stateless, challenge-based mechanism for
 authentication that is based on authentication in HTTP.  Any time
 that a proxy server or UA receives a request (with the exceptions
 given in Section 22.1), it MAY challenge the initiator of the request
 to provide assurance of its identity.  Once the originator has been
 identified, the recipient of the request SHOULD ascertain whether or
 not this user is authorized to make the request in question.  No
 authorization systems are recommended or discussed in this document.
 The "Digest" authentication mechanism described in this section
 provides message authentication and replay protection only, without
 message integrity or confidentiality.  Protective measures above and
 beyond those provided by Digest need to be taken to prevent active
 attackers from modifying SIP requests and responses.
 Note that due to its weak security, the usage of "Basic"
 authentication has been deprecated.  Servers MUST NOT accept
 credentials using the "Basic" authorization scheme, and servers also
 MUST NOT challenge with "Basic".  This is a change from RFC 2543.

22.1 Framework

 The framework for SIP authentication closely parallels that of HTTP
 (RFC 2617 [17]).  In particular, the BNF for auth-scheme, auth-param,
 challenge, realm, realm-value, and credentials is identical (although
 the usage of "Basic" as a scheme is not permitted).  In SIP, a UAS
 uses the 401 (Unauthorized) response to challenge the identity of a
 UAC.  Additionally, registrars and redirect servers MAY make use of
 401 (Unauthorized) responses for authentication, but proxies MUST
 NOT, and instead MAY use the 407 (Proxy Authentication Required)

Rosenberg, et. al. Standards Track [Page 193] RFC 3261 SIP: Session Initiation Protocol June 2002

 response.  The requirements for inclusion of the Proxy-Authenticate,
 Proxy-Authorization, WWW-Authenticate, and Authorization in the
 various messages are identical to those described in RFC 2617 [17].
 Since SIP does not have the concept of a canonical root URL, the
 notion of protection spaces is interpreted differently in SIP.  The
 realm string alone defines the protection domain.  This is a change
 from RFC 2543, in which the Request-URI and the realm together
 defined the protection domain.
    This previous definition of protection domain caused some amount
    of confusion since the Request-URI sent by the UAC and the
    Request-URI received by the challenging server might be different,
    and indeed the final form of the Request-URI might not be known to
    the UAC.  Also, the previous definition depended on the presence
    of a SIP URI in the Request-URI and seemed to rule out alternative
    URI schemes (for example, the tel URL).
 Operators of user agents or proxy servers that will authenticate
 received requests MUST adhere to the following guidelines for
 creation of a realm string for their server:
    o  Realm strings MUST be globally unique.  It is RECOMMENDED that
       a realm string contain a hostname or domain name, following the
       recommendation in Section 3.2.1 of RFC 2617 [17].
    o  Realm strings SHOULD present a human-readable identifier that
       can be rendered to a user.
 For example:
    INVITE sip:bob@biloxi.com SIP/2.0
    Authorization: Digest realm="biloxi.com", <...>
 Generally, SIP authentication is meaningful for a specific realm, a
 protection domain.  Thus, for Digest authentication, each such
 protection domain has its own set of usernames and passwords.  If a
 server does not require authentication for a particular request, it
 MAY accept a default username, "anonymous", which has no password
 (password of "").  Similarly, UACs representing many users, such as
 PSTN gateways, MAY have their own device-specific username and
 password, rather than accounts for particular users, for their realm.
 While a server can legitimately challenge most SIP requests, there
 are two requests defined by this document that require special
 handling for authentication: ACK and CANCEL.

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 Under an authentication scheme that uses responses to carry values
 used to compute nonces (such as Digest), some problems come up for
 any requests that take no response, including ACK.  For this reason,
 any credentials in the INVITE that were accepted by a server MUST be
 accepted by that server for the ACK.  UACs creating an ACK message
 will duplicate all of the Authorization and Proxy-Authorization
 header field values that appeared in the INVITE to which the ACK
 corresponds.  Servers MUST NOT attempt to challenge an ACK.
 Although the CANCEL method does take a response (a 2xx), servers MUST
 NOT attempt to challenge CANCEL requests since these requests cannot
 be resubmitted.  Generally, a CANCEL request SHOULD be accepted by a
 server if it comes from the same hop that sent the request being
 canceled (provided that some sort of transport or network layer
 security association, as described in Section 26.2.1, is in place).
 When a UAC receives a challenge, it SHOULD render to the user the
 contents of the "realm" parameter in the challenge (which appears in
 either a WWW-Authenticate header field or Proxy-Authenticate header
 field) if the UAC device does not already know of a credential for
 the realm in question.  A service provider that pre-configures UAs
 with credentials for its realm should be aware that users will not
 have the opportunity to present their own credentials for this realm
 when challenged at a pre-configured device.
 Finally, note that even if a UAC can locate credentials that are
 associated with the proper realm, the potential exists that these
 credentials may no longer be valid or that the challenging server
 will not accept these credentials for whatever reason (especially
 when "anonymous" with no password is submitted).  In this instance a
 server may repeat its challenge, or it may respond with a 403
 Forbidden.  A UAC MUST NOT re-attempt requests with the credentials
 that have just been rejected (though the request may be retried if
 the nonce was stale).

22.2 User-to-User Authentication

 When a UAS receives a request from a UAC, the UAS MAY authenticate
 the originator before the request is processed.  If no credentials
 (in the Authorization header field) are provided in the request, the
 UAS can challenge the originator to provide credentials by rejecting
 the request with a 401 (Unauthorized) status code.
 The WWW-Authenticate response-header field MUST be included in 401
 (Unauthorized) response messages.  The field value consists of at
 least one challenge that indicates the authentication scheme(s) and
 parameters applicable to the realm.

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 An example of the WWW-Authenticate header field in a 401 challenge
 is:
    WWW-Authenticate: Digest
            realm="biloxi.com",
            qop="auth,auth-int",
            nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
            opaque="5ccc069c403ebaf9f0171e9517f40e41"
 When the originating UAC receives the 401 (Unauthorized), it SHOULD,
 if it is able, re-originate the request with the proper credentials.
 The UAC may require input from the originating user before
 proceeding.  Once authentication credentials have been supplied
 (either directly by the user, or discovered in an internal keyring),
 UAs SHOULD cache the credentials for a given value of the To header
 field and "realm" and attempt to re-use these values on the next
 request for that destination.  UAs MAY cache credentials in any way
 they would like.
 If no credentials for a realm can be located, UACs MAY attempt to
 retry the request with a username of "anonymous" and no password (a
 password of "").
 Once credentials have been located, any UA that wishes to
 authenticate itself with a UAS or registrar -- usually, but not
 necessarily, after receiving a 401 (Unauthorized) response -- MAY do
 so by including an Authorization header field with the request.  The
 Authorization field value consists of credentials containing the
 authentication information of the UA for the realm of the resource
 being requested as well as parameters required in support of
 authentication and replay protection.
 An example of the Authorization header field is:
    Authorization: Digest username="bob",
            realm="biloxi.com",
            nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
            uri="sip:bob@biloxi.com",
            qop=auth,
            nc=00000001,
            cnonce="0a4f113b",
            response="6629fae49393a05397450978507c4ef1",
            opaque="5ccc069c403ebaf9f0171e9517f40e41"
 When a UAC resubmits a request with its credentials after receiving a
 401 (Unauthorized) or 407 (Proxy Authentication Required) response,
 it MUST increment the CSeq header field value as it would normally
 when sending an updated request.

Rosenberg, et. al. Standards Track [Page 196] RFC 3261 SIP: Session Initiation Protocol June 2002

22.3 Proxy-to-User Authentication

 Similarly, when a UAC sends a request to a proxy server, the proxy
 server MAY authenticate the originator before the request is
 processed.  If no credentials (in the Proxy-Authorization header
 field) are provided in the request, the proxy can challenge the
 originator to provide credentials by rejecting the request with a 407
 (Proxy Authentication Required) status code.  The proxy MUST populate
 the 407 (Proxy Authentication Required) message with a Proxy-
 Authenticate header field value applicable to the proxy for the
 requested resource.
 The use of Proxy-Authenticate and Proxy-Authorization parallel that
 described in [17], with one difference.  Proxies MUST NOT add values
 to the Proxy-Authorization header field.  All 407 (Proxy
 Authentication Required) responses MUST be forwarded upstream toward
 the UAC following the procedures for any other response.  It is the
 UAC's responsibility to add the Proxy-Authorization header field
 value containing credentials for the realm of the proxy that has
 asked for authentication.
    If a proxy were to resubmit a request adding a Proxy-Authorization
    header field value, it would need to increment the CSeq in the new
    request.  However, this would cause the UAC that submitted the
    original request to discard a response from the UAS, as the CSeq
    value would be different.
 When the originating UAC receives the 407 (Proxy Authentication
 Required) it SHOULD, if it is able, re-originate the request with the
 proper credentials.  It should follow the same procedures for the
 display of the "realm" parameter that are given above for responding
 to 401.
 If no credentials for a realm can be located, UACs MAY attempt to
 retry the request with a username of "anonymous" and no password (a
 password of "").
 The UAC SHOULD also cache the credentials used in the re-originated
 request.
 The following rule is RECOMMENDED for proxy credential caching:
 If a UA receives a Proxy-Authenticate header field value in a 401/407
 response to a request with a particular Call-ID, it should
 incorporate credentials for that realm in all subsequent requests
 that contain the same Call-ID.  These credentials MUST NOT be cached
 across dialogs; however, if a UA is configured with the realm of its
 local outbound proxy, when one exists, then the UA MAY cache

Rosenberg, et. al. Standards Track [Page 197] RFC 3261 SIP: Session Initiation Protocol June 2002

 credentials for that realm across dialogs.  Note that this does mean
 a future request in a dialog could contain credentials that are not
 needed by any proxy along the Route header path.
 Any UA that wishes to authenticate itself to a proxy server --
 usually, but not necessarily, after receiving a 407 (Proxy
 Authentication Required) response -- MAY do so by including a Proxy-
 Authorization header field value with the request.  The Proxy-
 Authorization request-header field allows the client to identify
 itself (or its user) to a proxy that requires authentication.  The
 Proxy-Authorization header field value consists of credentials
 containing the authentication information of the UA for the proxy
 and/or realm of the resource being requested.
 A Proxy-Authorization header field value applies only to the proxy
 whose realm is identified in the "realm" parameter (this proxy may
 previously have demanded authentication using the Proxy-Authenticate
 field).  When multiple proxies are used in a chain, a Proxy-
 Authorization header field value MUST NOT be consumed by any proxy
 whose realm does not match the "realm" parameter specified in that
 value.
 Note that if an authentication scheme that does not support realms is
 used in the Proxy-Authorization header field, a proxy server MUST
 attempt to parse all Proxy-Authorization header field values to
 determine whether one of them has what the proxy server considers to
 be valid credentials.  Because this is potentially very time-
 consuming in large networks, proxy servers SHOULD use an
 authentication scheme that supports realms in the Proxy-Authorization
 header field.
 If a request is forked (as described in Section 16.7), various proxy
 servers and/or UAs may wish to challenge the UAC.  In this case, the
 forking proxy server is responsible for aggregating these challenges
 into a single response.  Each WWW-Authenticate and Proxy-Authenticate
 value received in responses to the forked request MUST be placed into
 the single response that is sent by the forking proxy to the UA; the
 ordering of these header field values is not significant.
    When a proxy server issues a challenge in response to a request,
    it will not proxy the request until the UAC has retried the
    request with valid credentials.  A forking proxy may forward a
    request simultaneously to multiple proxy servers that require
    authentication, each of which in turn will not forward the request
    until the originating UAC has authenticated itself in their
    respective realm.  If the UAC does not provide credentials for

Rosenberg, et. al. Standards Track [Page 198] RFC 3261 SIP: Session Initiation Protocol June 2002

    each challenge, the proxy servers that issued the challenges will
    not forward requests to the UA where the destination user might be
    located, and therefore, the virtues of forking are largely lost.
 When resubmitting its request in response to a 401 (Unauthorized) or
 407 (Proxy Authentication Required) that contains multiple
 challenges, a UAC MAY include an Authorization value for each WWW-
 Authenticate value and a Proxy-Authorization value for each Proxy-
 Authenticate value for which the UAC wishes to supply a credential.
 As noted above, multiple credentials in a request SHOULD be
 differentiated by the "realm" parameter.
 It is possible for multiple challenges associated with the same realm
 to appear in the same 401 (Unauthorized) or 407 (Proxy Authentication
 Required).  This can occur, for example, when multiple proxies within
 the same administrative domain, which use a common realm, are reached
 by a forking request.  When it retries a request, a UAC MAY therefore
 supply multiple credentials in Authorization or Proxy-Authorization
 header fields with the same "realm" parameter value.  The same
 credentials SHOULD be used for the same realm.

22.4 The Digest Authentication Scheme

 This section describes the modifications and clarifications required
 to apply the HTTP Digest authentication scheme to SIP.  The SIP
 scheme usage is almost completely identical to that for HTTP [17].
 Since RFC 2543 is based on HTTP Digest as defined in RFC 2069 [39],
 SIP servers supporting RFC 2617 MUST ensure they are backwards
 compatible with RFC 2069.  Procedures for this backwards
 compatibility are specified in RFC 2617.  Note, however, that SIP
 servers MUST NOT accept or request Basic authentication.
 The rules for Digest authentication follow those defined in [17],
 with "HTTP/1.1" replaced by "SIP/2.0" in addition to the following
 differences:
    1.  The URI included in the challenge has the following BNF:
        URI  =  SIP-URI / SIPS-URI
    2.  The BNF in RFC 2617 has an error in that the 'uri' parameter
        of the Authorization header field for HTTP Digest

Rosenberg, et. al. Standards Track [Page 199] RFC 3261 SIP: Session Initiation Protocol June 2002

        authentication is not enclosed in quotation marks.  (The
        example in Section 3.5 of RFC 2617 is correct.)  For SIP, the
        'uri' MUST be enclosed in quotation marks.
    3.  The BNF for digest-uri-value is:
        digest-uri-value  =  Request-URI ; as defined in Section 25
    4.  The example procedure for choosing a nonce based on Etag does
        not work for SIP.
    5.  The text in RFC 2617 [17] regarding cache operation does not
        apply to SIP.
    6.  RFC 2617 [17] requires that a server check that the URI in the
        request line and the URI included in the Authorization header
        field point to the same resource.  In a SIP context, these two
        URIs may refer to different users, due to forwarding at some
        proxy.  Therefore, in SIP, a server MAY check that the
        Request-URI in the Authorization header field value
        corresponds to a user for whom the server is willing to accept
        forwarded or direct requests, but it is not necessarily a
        failure if the two fields are not equivalent.
    7.  As a clarification to the calculation of the A2 value for
        message integrity assurance in the Digest authentication
        scheme, implementers should assume, when the entity-body is
        empty (that is, when SIP messages have no body) that the hash
        of the entity-body resolves to the MD5 hash of an empty
        string, or:
           H(entity-body) = MD5("") =
        "d41d8cd98f00b204e9800998ecf8427e"
    8.  RFC 2617 notes that a cnonce value MUST NOT be sent in an
        Authorization (and by extension Proxy-Authorization) header
        field if no qop directive has been sent.  Therefore, any
        algorithms that have a dependency on the cnonce (including
        "MD5-Sess") require that the qop directive be sent.  Use of
        the "qop" parameter is optional in RFC 2617 for the purposes
        of backwards compatibility with RFC 2069; since RFC 2543 was
        based on RFC 2069, the "qop" parameter must unfortunately
        remain optional for clients and servers to receive.  However,
        servers MUST always send a "qop" parameter in WWW-Authenticate
        and Proxy-Authenticate header field values.  If a client
        receives a "qop" parameter in a challenge header field, it
        MUST send the "qop" parameter in any resulting authorization
        header field.

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 RFC 2543 did not allow usage of the Authentication-Info header field
 (it effectively used RFC 2069).  However, we now allow usage of this
 header field, since it provides integrity checks over the bodies and
 provides mutual authentication.  RFC 2617 [17] defines mechanisms for
 backwards compatibility using the qop attribute in the request.
 These mechanisms MUST be used by a server to determine if the client
 supports the new mechanisms in RFC 2617 that were not specified in
 RFC 2069.

23 S/MIME

 SIP messages carry MIME bodies and the MIME standard includes
 mechanisms for securing MIME contents to ensure both integrity and
 confidentiality (including the 'multipart/signed' and
 'application/pkcs7-mime' MIME types, see RFC 1847 [22], RFC 2630 [23]
 and RFC 2633 [24]).  Implementers should note, however, that there
 may be rare network intermediaries (not typical proxy servers) that
 rely on viewing or modifying the bodies of SIP messages (especially
 SDP), and that secure MIME may prevent these sorts of intermediaries
 from functioning.
    This applies particularly to certain types of firewalls.
    The PGP mechanism for encrypting the header fields and bodies of
    SIP messages described in RFC 2543 has been deprecated.

23.1 S/MIME Certificates

 The certificates that are used to identify an end-user for the
 purposes of S/MIME differ from those used by servers in one important
 respect - rather than asserting that the identity of the holder
 corresponds to a particular hostname, these certificates assert that
 the holder is identified by an end-user address.  This address is
 composed of the concatenation of the "userinfo" "@" and "domainname"
 portions of a SIP or SIPS URI (in other words, an email address of
 the form "bob@biloxi.com"), most commonly corresponding to a user's
 address-of-record.
 These certificates are also associated with keys that are used to
 sign or encrypt bodies of SIP messages.  Bodies are signed with the
 private key of the sender (who may include their public key with the
 message as appropriate), but bodies are encrypted with the public key
 of the intended recipient.  Obviously, senders must have
 foreknowledge of the public key of recipients in order to encrypt
 message bodies.  Public keys can be stored within a UA on a virtual
 keyring.

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 Each user agent that supports S/MIME MUST contain a keyring
 specifically for end-users' certificates.  This keyring should map
 between addresses of record and corresponding certificates.  Over
 time, users SHOULD use the same certificate when they populate the
 originating URI of signaling (the From header field) with the same
 address-of-record.
 Any mechanisms depending on the existence of end-user certificates
 are seriously limited in that there is virtually no consolidated
 authority today that provides certificates for end-user applications.
 However, users SHOULD acquire certificates from known public
 certificate authorities.  As an alternative, users MAY create self-
 signed certificates.  The implications of self-signed certificates
 are explored further in Section 26.4.2.  Implementations may also use
 pre-configured certificates in deployments in which a previous trust
 relationship exists between all SIP entities.
 Above and beyond the problem of acquiring an end-user certificate,
 there are few well-known centralized directories that distribute
 end-user certificates.  However, the holder of a certificate SHOULD
 publish their certificate in any public directories as appropriate.
 Similarly, UACs SHOULD support a mechanism for importing (manually or
 automatically) certificates discovered in public directories
 corresponding to the target URIs of SIP requests.

23.2 S/MIME Key Exchange

 SIP itself can also be used as a means to distribute public keys in
 the following manner.
 Whenever the CMS SignedData message is used in S/MIME for SIP, it
 MUST contain the certificate bearing the public key necessary to
 verify the signature.
 When a UAC sends a request containing an S/MIME body that initiates a
 dialog, or sends a non-INVITE request outside the context of a
 dialog, the UAC SHOULD structure the body as an S/MIME
 'multipart/signed' CMS SignedData body.  If the desired CMS service
 is EnvelopedData (and the public key of the target user is known),
 the UAC SHOULD send the EnvelopedData message encapsulated within a
 SignedData message.
 When a UAS receives a request containing an S/MIME CMS body that
 includes a certificate, the UAS SHOULD first validate the
 certificate, if possible, with any available root certificates for
 certificate authorities.  The UAS SHOULD also determine the subject
 of the certificate (for S/MIME, the SubjectAltName will contain the
 appropriate identity) and compare this value to the From header field

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 of the request.  If the certificate cannot be verified, because it is
 self-signed, or signed by no known authority, or if it is verifiable
 but its subject does not correspond to the From header field of
 request, the UAS MUST notify its user of the status of the
 certificate (including the subject of the certificate, its signer,
 and any key fingerprint information) and request explicit permission
 before proceeding.  If the certificate was successfully verified and
 the subject of the certificate corresponds to the From header field
 of the SIP request, or if the user (after notification) explicitly
 authorizes the use of the certificate, the UAS SHOULD add this
 certificate to a local keyring, indexed by the address-of-record of
 the holder of the certificate.
 When a UAS sends a response containing an S/MIME body that answers
 the first request in a dialog, or a response to a non-INVITE request
 outside the context of a dialog, the UAS SHOULD structure the body as
 an S/MIME 'multipart/signed' CMS SignedData body.  If the desired CMS
 service is EnvelopedData, the UAS SHOULD send the EnvelopedData
 message encapsulated within a SignedData message.
 When a UAC receives a response containing an S/MIME CMS body that
 includes a certificate, the UAC SHOULD first validate the
 certificate, if possible, with any appropriate root certificate.  The
 UAC SHOULD also determine the subject of the certificate and compare
 this value to the To field of the response; although the two may very
 well be different, and this is not necessarily indicative of a
 security breach.  If the certificate cannot be verified because it is
 self-signed, or signed by no known authority, the UAC MUST notify its
 user of the status of the certificate (including the subject of the
 certificate, its signator, and any key fingerprint information) and
 request explicit permission before proceeding.  If the certificate
 was successfully verified, and the subject of the certificate
 corresponds to the To header field in the response, or if the user
 (after notification) explicitly authorizes the use of the
 certificate, the UAC SHOULD add this certificate to a local keyring,
 indexed by the address-of-record of the holder of the certificate.
 If the UAC had not transmitted its own certificate to the UAS in any
 previous transaction, it SHOULD use a CMS SignedData body for its
 next request or response.
 On future occasions, when the UA receives requests or responses that
 contain a From header field corresponding to a value in its keyring,
 the UA SHOULD compare the certificate offered in these messages with
 the existing certificate in its keyring.  If there is a discrepancy,
 the UA MUST notify its user of a change of the certificate
 (preferably in terms that indicate that this is a potential security
 breach) and acquire the user's permission before continuing to

Rosenberg, et. al. Standards Track [Page 203] RFC 3261 SIP: Session Initiation Protocol June 2002

 process the signaling.  If the user authorizes this certificate, it
 SHOULD be added to the keyring alongside any previous value(s) for
 this address-of-record.
 Note well however, that this key exchange mechanism does not
 guarantee the secure exchange of keys when self-signed certificates,
 or certificates signed by an obscure authority, are used - it is
 vulnerable to well-known attacks.  In the opinion of the authors,
 however, the security it provides is proverbially better than
 nothing; it is in fact comparable to the widely used SSH application.
 These limitations are explored in greater detail in Section 26.4.2.
 If a UA receives an S/MIME body that has been encrypted with a public
 key unknown to the recipient, it MUST reject the request with a 493
 (Undecipherable) response.  This response SHOULD contain a valid
 certificate for the respondent (corresponding, if possible, to any
 address of record given in the To header field of the rejected
 request) within a MIME body with a 'certs-only' "smime-type"
 parameter.
 A 493 (Undecipherable) sent without any certificate indicates that
 the respondent cannot or will not utilize S/MIME encrypted messages,
 though they may still support S/MIME signatures.
 Note that a user agent that receives a request containing an S/MIME
 body that is not optional (with a Content-Disposition header
 "handling" parameter of "required") MUST reject the request with a
 415 Unsupported Media Type response if the MIME type is not
 understood.  A user agent that receives such a response when S/MIME
 is sent SHOULD notify its user that the remote device does not
 support S/MIME, and it MAY subsequently resend the request without
 S/MIME, if appropriate; however, this 415 response may constitute a
 downgrade attack.
 If a user agent sends an S/MIME body in a request, but receives a
 response that contains a MIME body that is not secured, the UAC
 SHOULD notify its user that the session could not be secured.
 However, if a user agent that supports S/MIME receives a request with
 an unsecured body, it SHOULD NOT respond with a secured body, but if
 it expects S/MIME from the sender (for example, because the sender's
 From header field value corresponds to an identity on its keychain),
 the UAS SHOULD notify its user that the session could not be secured.
 A number of conditions that arise in the previous text call for the
 notification of the user when an anomalous certificate-management
 event occurs.  Users might well ask what they should do under these
 circumstances.  First and foremost, an unexpected change in a
 certificate, or an absence of security when security is expected, are

Rosenberg, et. al. Standards Track [Page 204] RFC 3261 SIP: Session Initiation Protocol June 2002

 causes for caution but not necessarily indications that an attack is
 in progress.  Users might abort any connection attempt or refuse a
 connection request they have received; in telephony parlance, they
 could hang up and call back.  Users may wish to find an alternate
 means to contact the other party and confirm that their key has
 legitimately changed.  Note that users are sometimes compelled to
 change their certificates, for example when they suspect that the
 secrecy of their private key has been compromised.  When their
 private key is no longer private, users must legitimately generate a
 new key and re-establish trust with any users that held their old
 key.
 Finally, if during the course of a dialog a UA receives a certificate
 in a CMS SignedData message that does not correspond with the
 certificates previously exchanged during a dialog, the UA MUST notify
 its user of the change, preferably in terms that indicate that this
 is a potential security breach.

23.3 Securing MIME bodies

 There are two types of secure MIME bodies that are of interest to
 SIP: use of these bodies should follow the S/MIME specification [24]
 with a few variations.
    o  "multipart/signed" MUST be used only with CMS detached
       signatures.
          This allows backwards compatibility with non-S/MIME-
          compliant recipients.
    o  S/MIME bodies SHOULD have a Content-Disposition header field,
       and the value of the "handling" parameter SHOULD be "required."
    o  If a UAC has no certificate on its keyring associated with the
       address-of-record to which it wants to send a request, it
       cannot send an encrypted "application/pkcs7-mime" MIME message.
       UACs MAY send an initial request such as an OPTIONS message
       with a CMS detached signature in order to solicit the
       certificate of the remote side (the signature SHOULD be over a
       "message/sip" body of the type described in Section 23.4).
          Note that future standardization work on S/MIME may define
          non-certificate based keys.
    o  Senders of S/MIME bodies SHOULD use the "SMIMECapabilities"
       (see Section 2.5.2 of [24]) attribute to express their
       capabilities and preferences for further communications.  Note
       especially that senders MAY use the "preferSignedData"

Rosenberg, et. al. Standards Track [Page 205] RFC 3261 SIP: Session Initiation Protocol June 2002

       capability to encourage receivers to respond with CMS
       SignedData messages (for example, when sending an OPTIONS
       request as described above).
    o  S/MIME implementations MUST at a minimum support SHA1 as a
       digital signature algorithm, and 3DES as an encryption
       algorithm.  All other signature and encryption algorithms MAY
       be supported.  Implementations can negotiate support for these
       algorithms with the "SMIMECapabilities" attribute.
    o  Each S/MIME body in a SIP message SHOULD be signed with only
       one certificate.  If a UA receives a message with multiple
       signatures, the outermost signature should be treated as the
       single certificate for this body.  Parallel signatures SHOULD
       NOT be used.
       The following is an example of an encrypted S/MIME SDP body
       within a SIP message:
      INVITE sip:bob@biloxi.com SIP/2.0
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
      To: Bob <sip:bob@biloxi.com>
      From: Alice <sip:alice@atlanta.com>;tag=1928301774
      Call-ID: a84b4c76e66710
      CSeq: 314159 INVITE
      Max-Forwards: 70
      Contact: <sip:alice@pc33.atlanta.com>
      Content-Type: application/pkcs7-mime; smime-type=enveloped-data;
           name=smime.p7m
      Content-Disposition: attachment; filename=smime.p7m
         handling=required
  • * Content-Type: application/sdp * * * * v=0 * * o=alice 53655765 2353687637 IN IP4 pc33.atlanta.com * * s=- * * t=0 0 * * c=IN IP4 pc33.atlanta.com * * m=audio 3456 RTP/AVP 0 1 3 99 * * a=rtpmap:0 PCMU/8000 * *

Rosenberg, et. al. Standards Track [Page 206] RFC 3261 SIP: Session Initiation Protocol June 2002

23.4 SIP Header Privacy and Integrity using S/MIME: Tunneling SIP

 As a means of providing some degree of end-to-end authentication,
 integrity or confidentiality for SIP header fields, S/MIME can
 encapsulate entire SIP messages within MIME bodies of type
 "message/sip" and then apply MIME security to these bodies in the
 same manner as typical SIP bodies.  These encapsulated SIP requests
 and responses do not constitute a separate dialog or transaction,
 they are a copy of the "outer" message that is used to verify
 integrity or to supply additional information.
 If a UAS receives a request that contains a tunneled "message/sip"
 S/MIME body, it SHOULD include a tunneled "message/sip" body in the
 response with the same smime-type.
 Any traditional MIME bodies (such as SDP) SHOULD be attached to the
 "inner" message so that they can also benefit from S/MIME security.
 Note that "message/sip" bodies can be sent as a part of a MIME
 "multipart/mixed" body if any unsecured MIME types should also be
 transmitted in a request.

23.4.1 Integrity and Confidentiality Properties of SIP Headers

 When the S/MIME integrity or confidentiality mechanisms are used,
 there may be discrepancies between the values in the "inner" message
 and values in the "outer" message.  The rules for handling any such
 differences for all of the header fields described in this document
 are given in this section.
 Note that for the purposes of loose timestamping, all SIP messages
 that tunnel "message/sip" SHOULD contain a Date header in both the
 "inner" and "outer" headers.

23.4.1.1 Integrity

 Whenever integrity checks are performed, the integrity of a header
 field should be determined by matching the value of the header field
 in the signed body with that in the "outer" messages using the
 comparison rules of SIP as described in 20.
 Header fields that can be legitimately modified by proxy servers are:
 Request-URI, Via, Record-Route, Route, Max-Forwards, and Proxy-
 Authorization.  If these header fields are not intact end-to-end,
 implementations SHOULD NOT consider this a breach of security.
 Changes to any other header fields defined in this document
 constitute an integrity violation; users MUST be notified of a
 discrepancy.

Rosenberg, et. al. Standards Track [Page 207] RFC 3261 SIP: Session Initiation Protocol June 2002

23.4.1.2 Confidentiality

 When messages are encrypted, header fields may be included in the
 encrypted body that are not present in the "outer" message.
 Some header fields must always have a plaintext version because they
 are required header fields in requests and responses - these include:
 To, From, Call-ID, CSeq, Contact.  While it is probably not useful to
 provide an encrypted alternative for the Call-ID, CSeq, or Contact,
 providing an alternative to the information in the "outer" To or From
 is permitted.  Note that the values in an encrypted body are not used
 for the purposes of identifying transactions or dialogs - they are
 merely informational.  If the From header field in an encrypted body
 differs from the value in the "outer" message, the value within the
 encrypted body SHOULD be displayed to the user, but MUST NOT be used
 in the "outer" header fields of any future messages.
 Primarily, a user agent will want to encrypt header fields that have
 an end-to-end semantic, including: Subject, Reply-To, Organization,
 Accept, Accept-Encoding, Accept-Language, Alert-Info, Error-Info,
 Authentication-Info, Expires, In-Reply-To, Require, Supported,
 Unsupported, Retry-After, User-Agent, Server, and Warning.  If any of
 these header fields are present in an encrypted body, they should be
 used instead of any "outer" header fields, whether this entails
 displaying the header field values to users or setting internal
 states in the UA.  They SHOULD NOT however be used in the "outer"
 headers of any future messages.
 If present, the Date header field MUST always be the same in the
 "inner" and "outer" headers.
 Since MIME bodies are attached to the "inner" message,
 implementations will usually encrypt MIME-specific header fields,
 including: MIME-Version, Content-Type, Content-Length, Content-
 Language, Content-Encoding and Content-Disposition.  The "outer"
 message will have the proper MIME header fields for S/MIME bodies.
 These header fields (and any MIME bodies they preface) should be
 treated as normal MIME header fields and bodies received in a SIP
 message.
 It is not particularly useful to encrypt the following header fields:
 Min-Expires, Timestamp, Authorization, Priority, and WWW-
 Authenticate.  This category also includes those header fields that
 can be changed by proxy servers (described in the preceding section).
 UAs SHOULD never include these in an "inner" message if they are not

Rosenberg, et. al. Standards Track [Page 208] RFC 3261 SIP: Session Initiation Protocol June 2002

 included in the "outer" message.  UAs that receive any of these
 header fields in an encrypted body SHOULD ignore the encrypted
 values.
 Note that extensions to SIP may define additional header fields; the
 authors of these extensions should describe the integrity and
 confidentiality properties of such header fields.  If a SIP UA
 encounters an unknown header field with an integrity violation, it
 MUST ignore the header field.

23.4.2 Tunneling Integrity and Authentication

 Tunneling SIP messages within S/MIME bodies can provide integrity for
 SIP header fields if the header fields that the sender wishes to
 secure are replicated in a "message/sip" MIME body signed with a CMS
 detached signature.
 Provided that the "message/sip" body contains at least the
 fundamental dialog identifiers (To, From, Call-ID, CSeq), then a
 signed MIME body can provide limited authentication.  At the very
 least, if the certificate used to sign the body is unknown to the
 recipient and cannot be verified, the signature can be used to
 ascertain that a later request in a dialog was transmitted by the
 same certificate-holder that initiated the dialog.  If the recipient
 of the signed MIME body has some stronger incentive to trust the
 certificate (they were able to validate it, they acquired it from a
 trusted repository, or they have used it frequently) then the
 signature can be taken as a stronger assertion of the identity of the
 subject of the certificate.
 In order to eliminate possible confusions about the addition or
 subtraction of entire header fields, senders SHOULD replicate all
 header fields from the request within the signed body.  Any message
 bodies that require integrity protection MUST be attached to the
 "inner" message.
 If a Date header is present in a message with a signed body, the
 recipient SHOULD compare the header field value with its own internal
 clock, if applicable.  If a significant time discrepancy is detected
 (on the order of an hour or more), the user agent SHOULD alert the
 user to the anomaly, and note that it is a potential security breach.
 If an integrity violation in a message is detected by its recipient,
 the message MAY be rejected with a 403 (Forbidden) response if it is
 a request, or any existing dialog MAY be terminated.  UAs SHOULD
 notify users of this circumstance and request explicit guidance on
 how to proceed.

Rosenberg, et. al. Standards Track [Page 209] RFC 3261 SIP: Session Initiation Protocol June 2002

 The following is an example of the use of a tunneled "message/sip"
 body:
    INVITE sip:bob@biloxi.com SIP/2.0
    Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
    To: Bob <sip:bob@biloxi.com>
    From: Alice <sip:alice@atlanta.com>;tag=1928301774
    Call-ID: a84b4c76e66710
    CSeq: 314159 INVITE
    Max-Forwards: 70
    Date: Thu, 21 Feb 2002 13:02:03 GMT
    Contact: <sip:alice@pc33.atlanta.com>
    Content-Type: multipart/signed;
      protocol="application/pkcs7-signature";
      micalg=sha1; boundary=boundary42
    Content-Length: 568
  1. -boundary42

Content-Type: message/sip

    INVITE sip:bob@biloxi.com SIP/2.0
    Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
    To: Bob <bob@biloxi.com>
    From: Alice <alice@atlanta.com>;tag=1928301774
    Call-ID: a84b4c76e66710
    CSeq: 314159 INVITE
    Max-Forwards: 70
    Date: Thu, 21 Feb 2002 13:02:03 GMT
    Contact: <sip:alice@pc33.atlanta.com>
    Content-Type: application/sdp
    Content-Length: 147
    v=0
    o=UserA 2890844526 2890844526 IN IP4 here.com
    s=Session SDP
    c=IN IP4 pc33.atlanta.com
    t=0 0
    m=audio 49172 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
  1. -boundary42

Content-Type: application/pkcs7-signature; name=smime.p7s

    Content-Transfer-Encoding: base64
    Content-Disposition: attachment; filename=smime.p7s;
       handling=required

Rosenberg, et. al. Standards Track [Page 210] RFC 3261 SIP: Session Initiation Protocol June 2002

    ghyHhHUujhJhjH77n8HHGTrfvbnj756tbB9HG4VQpfyF467GhIGfHfYT6
    4VQpfyF467GhIGfHfYT6jH77n8HHGghyHhHUujhJh756tbB9HGTrfvbnj
    n8HHGTrfvhJhjH776tbB9HG4VQbnj7567GhIGfHfYT6ghyHhHUujpfyF4
    7GhIGfHfYT64VQbnj756
  1. -boundary42-

23.4.3 Tunneling Encryption

 It may also be desirable to use this mechanism to encrypt a
 "message/sip" MIME body within a CMS EnvelopedData message S/MIME
 body, but in practice, most header fields are of at least some use to
 the network; the general use of encryption with S/MIME is to secure
 message bodies like SDP rather than message headers.  Some
 informational header fields, such as the Subject or Organization
 could perhaps warrant end-to-end security.  Headers defined by future
 SIP applications might also require obfuscation.
 Another possible application of encrypting header fields is selective
 anonymity.  A request could be constructed with a From header field
 that contains no personal information (for example,
 sip:anonymous@anonymizer.invalid).  However, a second From header
 field containing the genuine address-of-record of the originator
 could be encrypted within a "message/sip" MIME body where it will
 only be visible to the endpoints of a dialog.
    Note that if this mechanism is used for anonymity, the From header
    field will no longer be usable by the recipient of a message as an
    index to their certificate keychain for retrieving the proper
    S/MIME key to associated with the sender.  The message must first
    be decrypted, and the "inner" From header field MUST be used as an
    index.
 In order to provide end-to-end integrity, encrypted "message/sip"
 MIME bodies SHOULD be signed by the sender.  This creates a
 "multipart/signed" MIME body that contains an encrypted body and a
 signature, both of type "application/pkcs7-mime".

Rosenberg, et. al. Standards Track [Page 211] RFC 3261 SIP: Session Initiation Protocol June 2002

 In the following example, of an encrypted and signed message, the
 text boxed in asterisks ("*") is encrypted:
      INVITE sip:bob@biloxi.com SIP/2.0
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
      To: Bob <sip:bob@biloxi.com>
      From: Anonymous <sip:anonymous@atlanta.com>;tag=1928301774
      Call-ID: a84b4c76e66710
      CSeq: 314159 INVITE
      Max-Forwards: 70
      Date: Thu, 21 Feb 2002 13:02:03 GMT
      Contact: <sip:pc33.atlanta.com>
      Content-Type: multipart/signed;
        protocol="application/pkcs7-signature";
        micalg=sha1; boundary=boundary42
      Content-Length: 568
  1. -boundary42

Content-Type: application/pkcs7-mime; smime-type=enveloped-data;

           name=smime.p7m
      Content-Transfer-Encoding: base64
      Content-Disposition: attachment; filename=smime.p7m
         handling=required
      Content-Length: 231
  • * Content-Type: message/sip * * * * INVITE sip:bob@biloxi.com SIP/2.0 * * Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 * * To: Bob bob@biloxi.com * * From: Alice alice@atlanta.com;tag=1928301774 * * Call-ID: a84b4c76e66710 * * CSeq: 314159 INVITE * * Max-Forwards: 70 * * Date: Thu, 21 Feb 2002 13:02:03 GMT * * Contact: <sip:alice@pc33.atlanta.com> * * * * Content-Type: application/sdp * * * * v=0 * * o=alice 53655765 2353687637 IN IP4 pc33.atlanta.com * * s=Session SDP * * t=0 0 * * c=IN IP4 pc33.atlanta.com * * m=audio 3456 RTP/AVP 0 1 3 99 * * a=rtpmap:0 PCMU/8000 * *

Rosenberg, et. al. Standards Track [Page 212] RFC 3261 SIP: Session Initiation Protocol June 2002

  1. -boundary42

Content-Type: application/pkcs7-signature; name=smime.p7s

      Content-Transfer-Encoding: base64
      Content-Disposition: attachment; filename=smime.p7s;
         handling=required
      ghyHhHUujhJhjH77n8HHGTrfvbnj756tbB9HG4VQpfyF467GhIGfHfYT6
      4VQpfyF467GhIGfHfYT6jH77n8HHGghyHhHUujhJh756tbB9HGTrfvbnj
      n8HHGTrfvhJhjH776tbB9HG4VQbnj7567GhIGfHfYT6ghyHhHUujpfyF4
      7GhIGfHfYT64VQbnj756
  1. -boundary42-

24 Examples

 In the following examples, we often omit the message body and the
 corresponding Content-Length and Content-Type header fields for
 brevity.

24.1 Registration

 Bob registers on start-up.  The message flow is shown in Figure 9.
 Note that the authentication usually required for registration is not
 shown for simplicity.
                biloxi.com         Bob's
                 registrar       softphone
                    |                |
                    |   REGISTER F1  |
                    |<---------------|
                    |    200 OK F2   |
                    |--------------->|
                Figure 9: SIP Registration Example
 F1 REGISTER Bob -> Registrar
     REGISTER sip:registrar.biloxi.com SIP/2.0
     Via: SIP/2.0/UDP bobspc.biloxi.com:5060;branch=z9hG4bKnashds7
     Max-Forwards: 70
     To: Bob <sip:bob@biloxi.com>
     From: Bob <sip:bob@biloxi.com>;tag=456248
     Call-ID: 843817637684230@998sdasdh09
     CSeq: 1826 REGISTER
     Contact: <sip:bob@192.0.2.4>
     Expires: 7200
     Content-Length: 0

Rosenberg, et. al. Standards Track [Page 213] RFC 3261 SIP: Session Initiation Protocol June 2002

 The registration expires after two hours.  The registrar responds
 with a 200 OK:
 F2 200 OK Registrar -> Bob
      SIP/2.0 200 OK
      Via: SIP/2.0/UDP bobspc.biloxi.com:5060;branch=z9hG4bKnashds7
       ;received=192.0.2.4
      To: Bob <sip:bob@biloxi.com>;tag=2493k59kd
      From: Bob <sip:bob@biloxi.com>;tag=456248
      Call-ID: 843817637684230@998sdasdh09
      CSeq: 1826 REGISTER
      Contact: <sip:bob@192.0.2.4>
      Expires: 7200
      Content-Length: 0

24.2 Session Setup

 This example contains the full details of the example session setup
 in Section 4.  The message flow is shown in Figure 1.  Note that
 these flows show the minimum required set of header fields - some
 other header fields such as Allow and Supported would normally be
 present.

F1 INVITE Alice → atlanta.com proxy

INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 Max-Forwards: 70 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142

(Alice's SDP not shown)

Rosenberg, et. al. Standards Track [Page 214] RFC 3261 SIP: Session Initiation Protocol June 2002

F2 100 Trying atlanta.com proxy → Alice

SIP/2.0 100 Trying Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Content-Length: 0

F3 INVITE atlanta.com proxy → biloxi.com proxy

INVITE sip:bob@biloxi.com SIP/2.0 Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 Max-Forwards: 69 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142

(Alice's SDP not shown)

F4 100 Trying biloxi.com proxy → atlanta.com proxy

SIP/2.0 100 Trying Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 ;received=192.0.2.2 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Content-Length: 0

Rosenberg, et. al. Standards Track [Page 215] RFC 3261 SIP: Session Initiation Protocol June 2002

F5 INVITE biloxi.com proxy → Bob

INVITE sip:bob@192.0.2.4 SIP/2.0 Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1 Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 ;received=192.0.2.2 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 Max-Forwards: 68 To: Bob <sip:bob@biloxi.com> From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: <sip:alice@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142

(Alice's SDP not shown)

F6 180 Ringing Bob → biloxi.com proxy

SIP/2.0 180 Ringing Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1 ;received=192.0.2.3 Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 ;received=192.0.2.2 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 To: Bob <sip:bob@biloxi.com>;tag=a6c85cf From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 Contact: <sip:bob@192.0.2.4> CSeq: 314159 INVITE Content-Length: 0

F7 180 Ringing biloxi.com proxy → atlanta.com proxy

SIP/2.0 180 Ringing Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 ;received=192.0.2.2 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 To: Bob <sip:bob@biloxi.com>;tag=a6c85cf From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 Contact: <sip:bob@192.0.2.4> CSeq: 314159 INVITE Content-Length: 0

Rosenberg, et. al. Standards Track [Page 216] RFC 3261 SIP: Session Initiation Protocol June 2002

F8 180 Ringing atlanta.com proxy → Alice

SIP/2.0 180 Ringing Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 To: Bob <sip:bob@biloxi.com>;tag=a6c85cf From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 Contact: <sip:bob@192.0.2.4> CSeq: 314159 INVITE Content-Length: 0

F9 200 OK Bob → biloxi.com proxy

SIP/2.0 200 OK Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1 ;received=192.0.2.3 Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 ;received=192.0.2.2 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 To: Bob <sip:bob@biloxi.com>;tag=a6c85cf From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: <sip:bob@192.0.2.4> Content-Type: application/sdp Content-Length: 131

(Bob's SDP not shown)

F10 200 OK biloxi.com proxy → atlanta.com proxy

SIP/2.0 200 OK Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1 ;received=192.0.2.2 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 To: Bob <sip:bob@biloxi.com>;tag=a6c85cf From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: <sip:bob@192.0.2.4> Content-Type: application/sdp Content-Length: 131

(Bob's SDP not shown)

Rosenberg, et. al. Standards Track [Page 217] RFC 3261 SIP: Session Initiation Protocol June 2002

F11 200 OK atlanta.com proxy → Alice

SIP/2.0 200 OK Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 ;received=192.0.2.1 To: Bob <sip:bob@biloxi.com>;tag=a6c85cf From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 INVITE Contact: <sip:bob@192.0.2.4> Content-Type: application/sdp Content-Length: 131

(Bob's SDP not shown)

F12 ACK Alice → Bob

ACK sip:bob@192.0.2.4 SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds9 Max-Forwards: 70 To: Bob <sip:bob@biloxi.com>;tag=a6c85cf From: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314159 ACK Content-Length: 0

 The media session between Alice and Bob is now established.
 Bob hangs up first.  Note that Bob's SIP phone maintains its own CSeq
 numbering space, which, in this example, begins with 231.  Since Bob
 is making the request, the To and From URIs and tags have been
 swapped.

F13 BYE Bob → Alice

BYE sip:alice@pc33.atlanta.com SIP/2.0 Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10 Max-Forwards: 70 From: Bob <sip:bob@biloxi.com>;tag=a6c85cf To: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 231 BYE Content-Length: 0

Rosenberg, et. al. Standards Track [Page 218] RFC 3261 SIP: Session Initiation Protocol June 2002

F14 200 OK Alice → Bob

SIP/2.0 200 OK Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10 From: Bob <sip:bob@biloxi.com>;tag=a6c85cf To: Alice <sip:alice@atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 231 BYE Content-Length: 0

 The SIP Call Flows document [40] contains further examples of SIP
 messages.

25 Augmented BNF for the SIP Protocol

 All of the mechanisms specified in this document are described in
 both prose and an augmented Backus-Naur Form (BNF) defined in RFC
 2234 [10].  Section 6.1 of RFC 2234 defines a set of core rules that
 are used by this specification, and not repeated here.  Implementers
 need to be familiar with the notation and content of RFC 2234 in
 order to understand this specification.  Certain basic rules are in
 uppercase, such as SP, LWS, HTAB, CRLF, DIGIT, ALPHA, etc.  Angle
 brackets are used within definitions to clarify the use of rule
 names.
 The use of square brackets is redundant syntactically.  It is used as
 a semantic hint that the specific parameter is optional to use.

25.1 Basic Rules

 The following rules are used throughout this specification to
 describe basic parsing constructs.  The US-ASCII coded character set
 is defined by ANSI X3.4-1986.
    alphanum  =  ALPHA / DIGIT

Rosenberg, et. al. Standards Track [Page 219] RFC 3261 SIP: Session Initiation Protocol June 2002

 Several rules are incorporated from RFC 2396 [5] but are updated to
 make them compliant with RFC 2234 [10].  These include:
    reserved    =  ";" / "/" / "?" / ":" / "@" / "&" / "=" / "+"
                   / "$" / ","
    unreserved  =  alphanum / mark
    mark        =  "-" / "_" / "." / "!" / "~" / "*" / "'"
                   / "(" / ")"
    escaped     =  "%" HEXDIG HEXDIG
 SIP header field values can be folded onto multiple lines if the
 continuation line begins with a space or horizontal tab.  All linear
 white space, including folding, has the same semantics as SP.  A
 recipient MAY replace any linear white space with a single SP before
 interpreting the field value or forwarding the message downstream.
 This is intended to behave exactly as HTTP/1.1 as described in RFC
 2616 [8].  The SWS construct is used when linear white space is
 optional, generally between tokens and separators.
    LWS  =  [*WSP CRLF] 1*WSP ; linear whitespace
    SWS  =  [LWS] ; sep whitespace
 To separate the header name from the rest of value, a colon is used,
 which, by the above rule, allows whitespace before, but no line
 break, and whitespace after, including a linebreak.  The HCOLON
 defines this construct.
    HCOLON  =  *( SP / HTAB ) ":" SWS
 The TEXT-UTF8 rule is only used for descriptive field contents and
 values that are not intended to be interpreted by the message parser.
 Words of *TEXT-UTF8 contain characters from the UTF-8 charset (RFC
 2279 [7]).  The TEXT-UTF8-TRIM rule is used for descriptive field
 contents that are n t quoted strings, where leading and trailing LWS
 is not meaningful.  In this regard, SIP differs from HTTP, which uses
 the ISO 8859-1 character set.
    TEXT-UTF8-TRIM  =  1*TEXT-UTF8char *(*LWS TEXT-UTF8char)
    TEXT-UTF8char   =  %x21-7E / UTF8-NONASCII
    UTF8-NONASCII   =  %xC0-DF 1UTF8-CONT
                    /  %xE0-EF 2UTF8-CONT
                    /  %xF0-F7 3UTF8-CONT
                    /  %xF8-Fb 4UTF8-CONT
                    /  %xFC-FD 5UTF8-CONT
    UTF8-CONT       =  %x80-BF

Rosenberg, et. al. Standards Track [Page 220] RFC 3261 SIP: Session Initiation Protocol June 2002

 A CRLF is allowed in the definition of TEXT-UTF8-TRIM only as part of
 a header field continuation.  It is expected that the folding LWS
 will be replaced with a single SP before interpretation of the TEXT-
 UTF8-TRIM value.
 Hexadecimal numeric characters are used in several protocol elements.
 Some elements (authentication) force hex alphas to be lower case.
    LHEX  =  DIGIT / %x61-66 ;lowercase a-f
 Many SIP header field values consist of words separated by LWS or
 special characters.  Unless otherwise stated, tokens are case-
 insensitive.  These special characters MUST be in a quoted string to
 be used within a parameter value.  The word construct is used in
 Call-ID to allow most separators to be used.
    token       =  1*(alphanum / "-" / "." / "!" / "%" / "*"
                   / "_" / "+" / "`" / "'" / "~" )
    separators  =  "(" / ")" / "<" / ">" / "@" /
                   "," / ";" / ":" / "\" / DQUOTE /
                   "/" / "[" / "]" / "?" / "=" /
                   "{" / "}" / SP / HTAB
    word        =  1*(alphanum / "-" / "." / "!" / "%" / "*" /
                   "_" / "+" / "`" / "'" / "~" /
                   "(" / ")" / "<" / ">" /
                   ":" / "\" / DQUOTE /
                   "/" / "[" / "]" / "?" /
                   "{" / "}" )
 When tokens are used or separators are used between elements,
 whitespace is often allowed before or after these characters:
    STAR    =  SWS "*" SWS ; asterisk
    SLASH   =  SWS "/" SWS ; slash
    EQUAL   =  SWS "=" SWS ; equal
    LPAREN  =  SWS "(" SWS ; left parenthesis
    RPAREN  =  SWS ")" SWS ; right parenthesis
    RAQUOT  =  ">" SWS ; right angle quote
    LAQUOT  =  SWS "<"; left angle quote
    COMMA   =  SWS "," SWS ; comma
    SEMI    =  SWS ";" SWS ; semicolon
    COLON   =  SWS ":" SWS ; colon
    LDQUOT  =  SWS DQUOTE; open double quotation mark
    RDQUOT  =  DQUOTE SWS ; close double quotation mark

Rosenberg, et. al. Standards Track [Page 221] RFC 3261 SIP: Session Initiation Protocol June 2002

 Comments can be included in some SIP header fields by surrounding the
 comment text with parentheses.  Comments are only allowed in fields
 containing "comment" as part of their field value definition.  In all
 other fields, parentheses are considered part of the field value.
    comment  =  LPAREN *(ctext / quoted-pair / comment) RPAREN
    ctext    =  %x21-27 / %x2A-5B / %x5D-7E / UTF8-NONASCII
                / LWS
 ctext includes all chars except left and right parens and backslash.
 A string of text is parsed as a single word if it is quoted using
 double-quote marks.  In quoted strings, quotation marks (") and
 backslashes (\) need to be escaped.
    quoted-string  =  SWS DQUOTE *(qdtext / quoted-pair ) DQUOTE
    qdtext         =  LWS / %x21 / %x23-5B / %x5D-7E
                      / UTF8-NONASCII
 The backslash character ("\") MAY be used as a single-character
 quoting mechanism only within quoted-string and comment constructs.
 Unlike HTTP/1.1, the characters CR and LF cannot be escaped by this
 mechanism to avoid conflict with line folding and header separation.

quoted-pair = "\" (%x00-09 / %x0B-0C

              / %x0E-7F)

SIP-URI = "sip:" [ userinfo ] hostport

                  uri-parameters [ headers ]

SIPS-URI = "sips:" [ userinfo ] hostport

                  uri-parameters [ headers ]

userinfo = ( user / telephone-subscriber ) [ ":" password ] "@" user = 1*( unreserved / escaped / user-unreserved ) user-unreserved = "&" / "=" / "+" / "$" / "," / ";" / "?" / "/" password = *( unreserved / escaped /

                  "&" / "=" / "+" / "$" / "," )

hostport = host [ ":" port ] host = hostname / IPv4address / IPv6reference hostname = *( domainlabel "." ) toplabel [ "." ] domainlabel = alphanum

                  / alphanum *( alphanum / "-" ) alphanum

toplabel = ALPHA / ALPHA *( alphanum / "-" ) alphanum

Rosenberg, et. al. Standards Track [Page 222] RFC 3261 SIP: Session Initiation Protocol June 2002

IPv4address = 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT IPv6reference = "[" IPv6address "]" IPv6address = hexpart [ ":" IPv4address ] hexpart = hexseq / hexseq "::" [ hexseq ] / "::" [ hexseq ] hexseq = hex4 *( ":" hex4) hex4 = 1*4HEXDIG port = 1*DIGIT

 The BNF for telephone-subscriber can be found in RFC 2806 [9].  Note,
 however, that any characters allowed there that are not allowed in
 the user part of the SIP URI MUST be escaped.

uri-parameters = *( ";" uri-parameter) uri-parameter = transport-param / user-param / method-param

                   / ttl-param / maddr-param / lr-param / other-param

transport-param = "transport="

                   ( "udp" / "tcp" / "sctp" / "tls"
                   / other-transport)

other-transport = token user-param = "user=" ( "phone" / "ip" / other-user) other-user = token method-param = "method=" Method ttl-param = "ttl=" ttl maddr-param = "maddr=" host lr-param = "lr" other-param = pname [ "=" pvalue ] pname = 1*paramchar pvalue = 1*paramchar paramchar = param-unreserved / unreserved / escaped param-unreserved = "[" / "]" / "/" / ":" / "&" / "+" / "$"

headers = "?" header *( "&" header ) header = hname "=" hvalue hname = 1*( hnv-unreserved / unreserved / escaped ) hvalue = *( hnv-unreserved / unreserved / escaped ) hnv-unreserved = "[" / "]" / "/" / "?" / ":" / "+" / "$"

SIP-message = Request / Response Request = Request-Line

  • ( message-header )

CRLF

                [ message-body ]

Request-Line = Method SP Request-URI SP SIP-Version CRLF Request-URI = SIP-URI / SIPS-URI / absoluteURI absoluteURI = scheme ":" ( hier-part / opaque-part ) hier-part = ( net-path / abs-path ) [ "?" query ] net-path = "" authority [ abs-path ] abs-path = "/" path-segments Rosenberg, et. al. Standards Track [Page 223] RFC 3261 SIP: Session Initiation Protocol June 2002 opaque-part = uric-no-slash *uric uric = reserved / unreserved / escaped uric-no-slash = unreserved / escaped / ";" / "?" / ":" / "@" / "&" / "=" / "+" / "$" / "," path-segments = segment *( "/" segment ) segment = *pchar *( ";" param ) param = *pchar pchar = unreserved / escaped / ":" / "@" / "&" / "=" / "+" / "$" / "," scheme = ALPHA *( ALPHA / DIGIT / "+" / "-" / "." ) authority = srvr / reg-name srvr = [ [ userinfo "@" ] hostport ] reg-name = 1*( unreserved / escaped / "$" / "," / ";" / ":" / "@" / "&" / "=" / "+" ) query = *uric SIP-Version = "SIP" "/" 1*DIGIT "." 1*DIGIT message-header = (Accept / Accept-Encoding / Accept-Language / Alert-Info / Allow / Authentication-Info / Authorization / Call-ID / Call-Info / Contact / Content-Disposition / Content-Encoding / Content-Language / Content-Length / Content-Type / CSeq / Date / Error-Info / Expires / From / In-Reply-To / Max-Forwards / MIME-Version / Min-Expires / Organization / Priority / Proxy-Authenticate / Proxy-Authorization / Proxy-Require / Record-Route / Reply-To Rosenberg, et. al. Standards Track [Page 224] RFC 3261 SIP: Session Initiation Protocol June 2002 / Require / Retry-After / Route / Server / Subject / Supported / Timestamp / To / Unsupported / User-Agent / Via / Warning / WWW-Authenticate / extension-header) CRLF INVITEm = %x49.4E.56.49.54.45 ; INVITE in caps ACKm = %x41.43.4B ; ACK in caps OPTIONSm = %x4F.50.54.49.4F.4E.53 ; OPTIONS in caps BYEm = %x42.59.45 ; BYE in caps CANCELm = %x43.41.4E.43.45.4C ; CANCEL in caps REGISTERm = %x52.45.47.49.53.54.45.52 ; REGISTER in caps Method = INVITEm / ACKm / OPTIONSm / BYEm / CANCELm / REGISTERm / extension-method extension-method = token Response = Status-Line *( message-header ) CRLF [ message-body ] Status-Line = SIP-Version SP Status-Code SP Reason-Phrase CRLF Status-Code = Informational / Redirection / Success / Client-Error / Server-Error / Global-Failure / extension-code extension-code = 3DIGIT Reason-Phrase = *(reserved / unreserved / escaped / UTF8-NONASCII / UTF8-CONT / SP / HTAB) Informational = "100" ; Trying / "180" ; Ringing / "181" ; Call Is Being Forwarded / "182" ; Queued / "183" ; Session Progress Rosenberg, et. al. Standards Track [Page 225] RFC 3261 SIP: Session Initiation Protocol June 2002 Success = "200" ; OK Redirection = "300" ; Multiple Choices / "301" ; Moved Permanently / "302" ; Moved Temporarily / "305" ; Use Proxy / "380" ; Alternative Service Client-Error = "400" ; Bad Request / "401" ; Unauthorized / "402" ; Payment Required / "403" ; Forbidden / "404" ; Not Found / "405" ; Method Not Allowed / "406" ; Not Acceptable / "407" ; Proxy Authentication Required / "408" ; Request Timeout / "410" ; Gone / "413" ; Request Entity Too Large / "414" ; Request-URI Too Large / "415" ; Unsupported Media Type / "416" ; Unsupported URI Scheme / "420" ; Bad Extension / "421" ; Extension Required / "423" ; Interval Too Brief / "480" ; Temporarily not available / "481" ; Call Leg/Transaction Does Not Exist / "482" ; Loop Detected / "483" ; Too Many Hops / "484" ; Address Incomplete / "485" ; Ambiguous / "486" ; Busy Here / "487" ; Request Terminated / "488" ; Not Acceptable Here / "491" ; Request Pending / "493" ; Undecipherable Server-Error = "500" ; Internal Server Error / "501" ; Not Implemented / "502" ; Bad Gateway / "503" ; Service Unavailable / "504" ; Server Time-out / "505" ; SIP Version not supported / "513" ; Message Too Large Rosenberg, et. al. Standards Track [Page 226] RFC 3261 SIP: Session Initiation Protocol June 2002 Global-Failure = "600" ; Busy Everywhere / "603" ; Decline / "604" ; Does not exist anywhere / "606" ; Not Acceptable Accept = "Accept" HCOLON [ accept-range *(COMMA accept-range) ] accept-range = media-range *(SEMI accept-param) media-range = ( "*/*" / ( m-type SLASH "*" ) / ( m-type SLASH m-subtype ) ) *( SEMI m-parameter ) accept-param = ("q" EQUAL qvalue) / generic-param qvalue = ( "0" [ "." 0*3DIGIT ] ) / ( "1" [ "." 0*3("0") ] ) generic-param = token [ EQUAL gen-value ] gen-value = token / host / quoted-string Accept-Encoding = "Accept-Encoding" HCOLON [ encoding *(COMMA encoding) ] encoding = codings *(SEMI accept-param) codings = content-coding / "*" content-coding = token Accept-Language = "Accept-Language" HCOLON [ language *(COMMA language) ] language = language-range *(SEMI accept-param) language-range = ( ( 1*8ALPHA *( "-" 1*8ALPHA ) ) / "*" ) Alert-Info = "Alert-Info" HCOLON alert-param *(COMMA alert-param) alert-param = LAQUOT absoluteURI RAQUOT *( SEMI generic-param ) Allow = "Allow" HCOLON [Method *(COMMA Method)] Authorization = "Authorization" HCOLON credentials credentials = ("Digest" LWS digest-response) / other-response digest-response = dig-resp *(COMMA dig-resp) dig-resp = username / realm / nonce / digest-uri / dresponse / algorithm / cnonce / opaque / message-qop / nonce-count / auth-param username = "username" EQUAL username-value username-value = quoted-string digest-uri = "uri" EQUAL LDQUOT digest-uri-value RDQUOT digest-uri-value = rquest-uri ; Equal to request-uri as specified by HTTP/1.1 message-qop = "qop" EQUAL qop-value Rosenberg, et. al. Standards Track [Page 227] RFC 3261 SIP: Session Initiation Protocol June 2002 cnonce = "cnonce" EQUAL cnonce-value cnonce-value = nonce-value nonce-count = "nc" EQUAL nc-value nc-value = 8LHEX dresponse = "response" EQUAL request-digest request-digest = LDQUOT 32LHEX RDQUOT auth-param = auth-param-name EQUAL ( token / quoted-string ) auth-param-name = token other-response = auth-scheme LWS auth-param *(COMMA auth-param) auth-scheme = token Authentication-Info = "Authentication-Info" HCOLON ainfo *(COMMA ainfo) ainfo = nextnonce / message-qop / response-auth / cnonce / nonce-count nextnonce = "nextnonce" EQUAL nonce-value response-auth = "rspauth" EQUAL response-digest response-digest = LDQUOT *LHEX RDQUOT Call-ID = ( "Call-ID" / "i" ) HCOLON callid callid = word [ "@" word ] Call-Info = "Call-Info" HCOLON info *(COMMA info) info = LAQUOT absoluteURI RAQUOT *( SEMI info-param) info-param = ( "purpose" EQUAL ( "icon" / "info" / "card" / token ) ) / generic-param Contact = ("Contact" / "m" ) HCOLON ( STAR / (contact-param *(COMMA contact-param))) contact-param = (name-addr / addr-spec) *(SEMI contact-params) name-addr = [ display-name ] LAQUOT addr-spec RAQUOT addr-spec = SIP-URI / SIPS-URI / absoluteURI display-name = *(token LWS)/ quoted-string contact-params = c-p-q / c-p-expires / contact-extension c-p-q = "q" EQUAL qvalue c-p-expires = "expires" EQUAL delta-seconds contact-extension = generic-param delta-seconds = 1*DIGIT Content-Disposition = "Content-Disposition" HCOLON disp-type *( SEMI disp-param ) disp-type = "render" / "session" / "icon" / "alert" / disp-extension-token Rosenberg, et. al. Standards Track [Page 228] RFC 3261 SIP: Session Initiation Protocol June 2002 disp-param = handling-param / generic-param handling-param = "handling" EQUAL ( "optional" / "required" / other-handling ) other-handling = token disp-extension-token = token Content-Encoding = ( "Content-Encoding" / "e" ) HCOLON content-coding *(COMMA content-coding) Content-Language = "Content-Language" HCOLON language-tag *(COMMA language-tag) language-tag = primary-tag *( "-" subtag ) primary-tag = 1*8ALPHA subtag = 1*8ALPHA Content-Length = ( "Content-Length" / "l" ) HCOLON 1*DIGIT Content-Type = ( "Content-Type" / "c" ) HCOLON media-type media-type = m-type SLASH m-subtype *(SEMI m-parameter) m-type = discrete-type / composite-type discrete-type = "text" / "image" / "audio" / "video" / "application" / extension-token composite-type = "message" / "multipart" / extension-token extension-token = ietf-token / x-token ietf-token = token x-token = "x-" token m-subtype = extension-token / iana-token iana-token = token m-parameter = m-attribute EQUAL m-value m-attribute = token m-value = token / quoted-string CSeq = "CSeq" HCOLON 1*DIGIT LWS Method Date = "Date" HCOLON SIP-date SIP-date = rfc1123-date rfc1123-date = wkday "," SP date1 SP time SP "GMT" date1 = 2DIGIT SP month SP 4DIGIT ; day month year (e.g., 02 Jun 1982) time = 2DIGIT ":" 2DIGIT ":" 2DIGIT ; 00:00:00 - 23:59:59 wkday = "Mon" / "Tue" / "Wed" / "Thu" / "Fri" / "Sat" / "Sun" month = "Jan" / "Feb" / "Mar" / "Apr" / "May" / "Jun" / "Jul" / "Aug" / "Sep" / "Oct" / "Nov" / "Dec" Error-Info = "Error-Info" HCOLON error-uri *(COMMA error-uri) Rosenberg, et. al. Standards Track [Page 229] RFC 3261 SIP: Session Initiation Protocol June 2002 error-uri = LAQUOT absoluteURI RAQUOT *( SEMI generic-param ) Expires = "Expires" HCOLON delta-seconds From = ( "From" / "f" ) HCOLON from-spec from-spec = ( name-addr / addr-spec ) *( SEMI from-param ) from-param = tag-param / generic-param tag-param = "tag" EQUAL token In-Reply-To = "In-Reply-To" HCOLON callid *(COMMA callid) Max-Forwards = "Max-Forwards" HCOLON 1*DIGIT MIME-Version = "MIME-Version" HCOLON 1*DIGIT "." 1*DIGIT Min-Expires = "Min-Expires" HCOLON delta-seconds Organization = "Organization" HCOLON [TEXT-UTF8-TRIM] Priority = "Priority" HCOLON priority-value priority-value = "emergency" / "urgent" / "normal" / "non-urgent" / other-priority other-priority = token Proxy-Authenticate = "Proxy-Authenticate" HCOLON challenge challenge = ("Digest" LWS digest-cln *(COMMA digest-cln)) / other-challenge other-challenge = auth-scheme LWS auth-param *(COMMA auth-param) digest-cln = realm / domain / nonce / opaque / stale / algorithm / qop-options / auth-param realm = "realm" EQUAL realm-value realm-value = quoted-string domain = "domain" EQUAL LDQUOT URI *( 1*SP URI ) RDQUOT URI = absoluteURI / abs-path nonce = "nonce" EQUAL nonce-value nonce-value = quoted-string opaque = "opaque" EQUAL quoted-string stale = "stale" EQUAL ( "true" / "false" ) algorithm = "algorithm" EQUAL ( "MD5" / "MD5-sess" / token ) qop-options = "qop" EQUAL LDQUOT qop-value *("," qop-value) RDQUOT qop-value = "auth" / "auth-int" / token Proxy-Authorization = "Proxy-Authorization" HCOLON credentials Rosenberg, et. al. Standards Track [Page 230] RFC 3261 SIP: Session Initiation Protocol June 2002 Proxy-Require = "Proxy-Require" HCOLON option-tag *(COMMA option-tag) option-tag = token Record-Route = "Record-Route" HCOLON rec-route *(COMMA rec-route) rec-route = name-addr *( SEMI rr-param ) rr-param = generic-param Reply-To = "Reply-To" HCOLON rplyto-spec rplyto-spec = ( name-addr / addr-spec ) *( SEMI rplyto-param ) rplyto-param = generic-param Require = "Require" HCOLON option-tag *(COMMA option-tag) Retry-After = "Retry-After" HCOLON delta-seconds [ comment ] *( SEMI retry-param ) retry-param = ("duration" EQUAL delta-seconds) / generic-param Route = "Route" HCOLON route-param *(COMMA route-param) route-param = name-addr *( SEMI rr-param ) Server = "Server" HCOLON server-val *(LWS server-val) server-val = product / comment product = token [SLASH product-version] product-version = token Subject = ( "Subject" / "s" ) HCOLON [TEXT-UTF8-TRIM] Supported = ( "Supported" / "k" ) HCOLON [option-tag *(COMMA option-tag)] Timestamp = "Timestamp" HCOLON 1*(DIGIT) [ "." *(DIGIT) ] [ LWS delay ] delay = *(DIGIT) [ "." *(DIGIT) ] To = ( "To" / "t" ) HCOLON ( name-addr / addr-spec ) *( SEMI to-param ) to-param = tag-param / generic-param Unsupported = "Unsupported" HCOLON option-tag *(COMMA option-tag) User-Agent = "User-Agent" HCOLON server-val *(LWS server-val) Rosenberg, et. al. Standards Track [Page 231] RFC 3261 SIP: Session Initiation Protocol June 2002 Via = ( "Via" / "v" ) HCOLON via-parm *(COMMA via-parm) via-parm = sent-protocol LWS sent-by *( SEMI via-params ) via-params = via-ttl / via-maddr / via-received / via-branch / via-extension via-ttl = "ttl" EQUAL ttl via-maddr = "maddr" EQUAL host via-received = "received" EQUAL (IPv4address / IPv6address) via-branch = "branch" EQUAL token via-extension = generic-param sent-protocol = protocol-name SLASH protocol-version SLASH transport protocol-name = "SIP" / token protocol-version = token transport = "UDP" / "TCP" / "TLS" / "SCTP" / other-transport sent-by = host [ COLON port ] ttl = 1*3DIGIT ; 0 to 255 Warning = "Warning" HCOLON warning-value *(COMMA warning-value) warning-value = warn-code SP warn-agent SP warn-text warn-code = 3DIGIT warn-agent = hostport / pseudonym ; the name or pseudonym of the server adding ; the Warning header, for use in debugging warn-text = quoted-string pseudonym = token WWW-Authenticate = "WWW-Authenticate" HCOLON challenge extension-header = header-name HCOLON header-value header-name = token header-value = *(TEXT-UTF8char / UTF8-CONT / LWS) message-body = *OCTET 26 Security Considerations: Threat Model and Security Usage Recommendations SIP is not an easy protocol to secure. Its use of intermediaries, its multi-faceted trust relationships, its expected usage between elements with no trust at all, and its user-to-user operation make security far from trivial. Security solutions are needed that are deployable today, without extensive coordination, in a wide variety of environments and usages. In order to meet these diverse needs, several distinct mechanisms applicable to different aspects and usages of SIP will be required. Rosenberg, et. al. Standards Track [Page 232] RFC 3261 SIP: Session Initiation Protocol June 2002 Note that the security of SIP signaling itself has no bearing on the security of protocols used in concert with SIP such as RTP, or with the security implications of any specific bodies SIP might carry (although MIME security plays a substantial role in securing SIP). Any media associated with a session can be encrypted end-to-end independently of any associated SIP signaling. Media encryption is outside the scope of this document. The considerations that follow first examine a set of classic threat models that broadly identify the security needs of SIP. The set of security services required to address these threats is then detailed, followed by an explanation of several security mechanisms that can be used to provide these services. Next, the requirements for implementers of SIP are enumerated, along with exemplary deployments in which these security mechanisms could be used to improve the security of SIP. Some notes on privacy conclude this section. 26.1 Attacks and Threat Models This section details some threats that should be common to most deployments of SIP. These threats have been chosen specifically to illustrate each of the security services that SIP requires. The following examples by no means provide an exhaustive list of the threats against SIP; rather, these are "classic" threats that demonstrate the need for particular security services that can potentially prevent whole categories of threats. These attacks assume an environment in which attackers can potentially read any packet on the network - it is anticipated that SIP will frequently be used on the public Internet. Attackers on the network may be able to modify packets (perhaps at some compromised intermediary). Attackers may wish to steal services, eavesdrop on communications, or disrupt sessions. 26.1.1 Registration Hijacking The SIP registration mechanism allows a user agent to identify itself to a registrar as a device at which a user (designated by an address of record) is located. A registrar assesses the identity asserted in the From header field of a REGISTER message to determine whether this request can modify the contact addresses associated with the address-of-record in the To header field. While these two fields are frequently the same, there are many valid deployments in which a third-party may register contacts on a user's behalf. Rosenberg, et. al. Standards Track [Page 233] RFC 3261 SIP: Session Initiation Protocol June 2002 The From header field of a SIP request, however, can be modified arbitrarily by the owner of a UA, and this opens the door to malicious registrations. An attacker that successfully impersonates a party authorized to change contacts associated with an address-of- record could, for example, de-register all existing contacts for a URI and then register their own device as the appropriate contact address, thereby directing all requests for the affected user to the attacker's device. This threat belongs to a family of threats that rely on the absence of cryptographic assurance of a request's originator. Any SIP UAS that represents a valuable service (a gateway that interworks SIP requests with traditional telephone calls, for example) might want to control access to its resources by authenticating requests that it receives. Even end-user UAs, for example SIP phones, have an interest in ascertaining the identities of originators of requests. This threat demonstrates the need for security services that enable SIP entities to authenticate the originators of requests. 26.1.2 Impersonating a Server The domain to which a request is destined is generally specified in the Request-URI. UAs commonly contact a server in this domain directly in order to deliver a request. However, there is always a possibility that an attacker could impersonate the remote server, and that the UA's request could be intercepted by some other party. For example, consider a case in which a redirect server at one domain, chicago.com, impersonates a redirect server at another domain, biloxi.com. A user agent sends a request to biloxi.com, but the redirect server at chicago.com answers with a forged response that has appropriate SIP header fields for a response from biloxi.com. The forged contact addresses in the redirection response could direct the originating UA to inappropriate or insecure resources, or simply prevent requests for biloxi.com from succeeding. This family of threats has a vast membership, many of which are critical. As a converse to the registration hijacking threat, consider the case in which a registration sent to biloxi.com is intercepted by chicago.com, which replies to the intercepted registration with a forged 301 (Moved Permanently) response. This response might seem to come from biloxi.com yet designate chicago.com as the appropriate registrar. All future REGISTER requests from the originating UA would then go to chicago.com. Prevention of this threat requires a means by which UAs can authenticate the servers to whom they send requests. Rosenberg, et. al. Standards Track [Page 234] RFC 3261 SIP: Session Initiation Protocol June 2002 26.1.3 Tampering with Message Bodies As a matter of course, SIP UAs route requests through trusted proxy servers. Regardless of how that trust is established (authentication of proxies is discussed elsewhere in this section), a UA may trust a proxy server to route a request, but not to inspect or possibly modify the bodies contained in that request. Consider a UA that is using SIP message bodies to communicate session encryption keys for a media session. Although it trusts the proxy server of the domain it is contacting to deliver signaling properly, it may not want the administrators of that domain to be capable of decrypting any subsequent media session. Worse yet, if the proxy server were actively malicious, it could modify the session key, either acting as a man-in-the-middle, or perhaps changing the security characteristics requested by the originating UA. This family of threats applies not only to session keys, but to most conceivable forms of content carried end-to-end in SIP. These might include MIME bodies that should be rendered to the user, SDP, or encapsulated telephony signals, among others. Attackers might attempt to modify SDP bodies, for example, in order to point RTP media streams to a wiretapping device in order to eavesdrop on subsequent voice communications. Also note that some header fields in SIP are meaningful end-to-end, for example, Subject. UAs might be protective of these header fields as well as bodies (a malicious intermediary changing the Subject header field might make an important request appear to be spam, for example). However, since many header fields are legitimately inspected or altered by proxy servers as a request is routed, not all header fields should be secured end-to-end. For these reasons, the UA might want to secure SIP message bodies, and in some limited cases header fields, end-to-end. The security services required for bodies include confidentiality, integrity, and authentication. These end-to-end services should be independent of the means used to secure interactions with intermediaries such as proxy servers. 26.1.4 Tearing Down Sessions Once a dialog has been established by initial messaging, subsequent requests can be sent that modify the state of the dialog and/or session. It is critical that principals in a session can be certain that such requests are not forged by attackers. Rosenberg, et. al. Standards Track [Page 235] RFC 3261 SIP: Session Initiation Protocol June 2002 Consider a case in which a third-party attacker captures some initial messages in a dialog shared by two parties in order to learn the parameters of the session (To tag, From tag, and so forth) and then inserts a BYE request into the session. The attacker could opt to forge the request such that it seemed to come from either participant. Once the BYE is received by its target, the session will be torn down prematurely. Similar mid-session threats include the transmission of forged re- INVITEs that alter the session (possibly to reduce session security or redirect media streams as part of a wiretapping attack). The most effective countermeasure to this threat is the authentication of the sender of the BYE. In this instance, the recipient needs only know that the BYE came from the same party with whom the corresponding dialog was established (as opposed to ascertaining the absolute identity of the sender). Also, if the attacker is unable to learn the parameters of the session due to confidentiality, it would not be possible to forge the BYE. However, some intermediaries (like proxy servers) will need to inspect those parameters as the session is established. 26.1.5 Denial of Service and Amplification Denial-of-service attacks focus on rendering a particular network element unavailable, usually by directing an excessive amount of network traffic at its interfaces. A distributed denial-of-service attack allows one network user to cause multiple network hosts to flood a target host with a large amount of network traffic. In many architectures, SIP proxy servers face the public Internet in order to accept requests from worldwide IP endpoints. SIP creates a number of potential opportunities for distributed denial-of-service attacks that must be recognized and addressed by the implementers and operators of SIP systems. Attackers can create bogus requests that contain a falsified source IP address and a corresponding Via header field that identify a targeted host as the originator of the request and then send this request to a large number of SIP network elements, thereby using hapless SIP UAs or proxies to generate denial-of-service traffic aimed at the target. Similarly, attackers might use falsified Route header field values in a request that identify the target host and then send such messages to forking proxies that will amplify messaging sent to the target. Rosenberg, et. al. Standards Track [Page 236] RFC 3261 SIP: Session Initiation Protocol June 2002 Record-Route could be used to similar effect when the attacker is certain that the SIP dialog initiated by the request will result in numerous transactions originating in the backwards direction. A number of denial-of-service attacks open up if REGISTER requests are not properly authenticated and authorized by registrars. Attackers could de-register some or all users in an administrative domain, thereby preventing these users from being invited to new sessions. An attacker could also register a large number of contacts designating the same host for a given address-of-record in order to use the registrar and any associated proxy servers as amplifiers in a denial-of-service attack. Attackers might also attempt to deplete available memory and disk resources of a registrar by registering huge numbers of bindings. The use of multicast to transmit SIP requests can greatly increase the potential for denial-of-service attacks. These problems demonstrate a general need to define architectures that minimize the risks of denial-of-service, and the need to be mindful in recommendations for security mechanisms of this class of attacks. 26.2 Security Mechanisms From the threats described above, we gather that the fundamental security services required for the SIP protocol are: preserving the confidentiality and integrity of messaging, preventing replay attacks or message spoofing, providing for the authentication and privacy of the participants in a session, and preventing denial-of-service attacks. Bodies within SIP messages separately require the security services of confidentiality, integrity, and authentication. Rather than defining new security mechanisms specific to SIP, SIP reuses wherever possible existing security models derived from the HTTP and SMTP space. Full encryption of messages provides the best means to preserve the confidentiality of signaling - it can also guarantee that messages are not modified by any malicious intermediaries. However, SIP requests and responses cannot be naively encrypted end-to-end in their entirety because message fields such as the Request-URI, Route, and Via need to be visible to proxies in most network architectures so that SIP requests are routed correctly. Note that proxy servers need to modify some features of messages as well (such as adding Via header field values) in order for SIP to function. Proxy servers must therefore be trusted, to some degree, by SIP UAs. To this purpose, low-layer security mechanisms for SIP are recommended, which Rosenberg, et. al. Standards Track [Page 237] RFC 3261 SIP: Session Initiation Protocol June 2002 encrypt the entire SIP requests or responses on the wire on a hop- by-hop basis, and that allow endpoints to verify the identity of proxy servers to whom they send requests. SIP entities also have a need to identify one another in a secure fashion. When a SIP endpoint asserts the identity of its user to a peer UA or to a proxy server, that identity should in some way be verifiable. A cryptographic authentication mechanism is provided in SIP to address this requirement. An independent security mechanism for SIP message bodies supplies an alternative means of end-to-end mutual authentication, as well as providing a limit on the degree to which user agents must trust intermediaries. 26.2.1 Transport and Network Layer Security Transport or network layer security encrypts signaling traffic, guaranteeing message confidentiality and integrity. Oftentimes, certificates are used in the establishment of lower-layer security, and these certificates can also be used to provide a means of authentication in many architectures. Two popular alternatives for providing security at the transport and network layer are, respectively, TLS [25] and IPSec [26]. IPSec is a set of network-layer protocol tools that collectively can be used as a secure replacement for traditional IP (Internet Protocol). IPSec is most commonly used in architectures in which a set of hosts or administrative domains have an existing trust relationship with one another. IPSec is usually implemented at the operating system level in a host, or on a security gateway that provides confidentiality and integrity for all traffic it receives from a particular interface (as in a VPN architecture). IPSec can also be used on a hop-by-hop basis. In many architectures IPSec does not require integration with SIP applications; IPSec is perhaps best suited to deployments in which adding security directly to SIP hosts would be arduous. UAs that have a pre-shared keying relationship with their first-hop proxy server are also good candidates to use IPSec. Any deployment of IPSec for SIP would require an IPSec profile describing the protocol tools that would be required to secure SIP. No such profile is given in this document. Rosenberg, et. al. Standards Track [Page 238] RFC 3261 SIP: Session Initiation Protocol June 2002 TLS provides transport-layer security over connection-oriented protocols (for the purposes of this document, TCP); "tls" (signifying TLS over TCP) can be specified as the desired transport protocol within a Via header field value or a SIP-URI. TLS is most suited to architectures in which hop-by-hop security is required between hosts with no pre-existing trust association. For example, Alice trusts her local proxy server, which after a certificate exchange decides to trust Bob's local proxy server, which Bob trusts, hence Bob and Alice can communicate securely. TLS must be tightly coupled with a SIP application. Note that transport mechanisms are specified on a hop-by-hop basis in SIP, thus a UA that sends requests over TLS to a proxy server has no assurance that TLS will be used end-to-end. The TLS_RSA_WITH_AES_128_CBC_SHA ciphersuite [6] MUST be supported at a minimum by implementers when TLS is used in a SIP application. For purposes of backwards compatibility, proxy servers, redirect servers, and registrars SHOULD support TLS_RSA_WITH_3DES_EDE_CBC_SHA. Implementers MAY also support any other ciphersuite. 26.2.2 SIPS URI Scheme The SIPS URI scheme adheres to the syntax of the SIP URI (described in 19), although the scheme string is "sips" rather than "sip". The semantics of SIPS are very different from the SIP URI, however. SIPS allows resources to specify that they should be reached securely. A SIPS URI can be used as an address-of-record for a particular user - the URI by which the user is canonically known (on their business cards, in the From header field of their requests, in the To header field of REGISTER requests). When used as the Request-URI of a request, the SIPS scheme signifies that each hop over which the request is forwarded, until the request reaches the SIP entity responsible for the domain portion of the Request-URI, must be secured with TLS; once it reaches the domain in question it is handled in accordance with local security and routing policy, quite possibly using TLS for any last hop to a UAS. When used by the originator of a request (as would be the case if they employed a SIPS URI as the address-of-record of the target), SIPS dictates that the entire request path to the target domain be so secured. The SIPS scheme is applicable to many of the other ways in which SIP URIs are used in SIP today in addition to the Request-URI, including in addresses-of-record, contact addresses (the contents of Contact headers, including those of REGISTER methods), and Route headers. In each instance, the SIPS URI scheme allows these existing fields to Rosenberg, et. al. Standards Track [Page 239] RFC 3261 SIP: Session Initiation Protocol June 2002 designate secure resources. The manner in which a SIPS URI is dereferenced in any of these contexts has its own security properties which are detailed in [4]. The use of SIPS in particular entails that mutual TLS authentication SHOULD be employed, as SHOULD the ciphersuite TLS_RSA_WITH_AES_128_CBC_SHA. Certificates received in the authentication process SHOULD be validated with root certificates held by the client; failure to validate a certificate SHOULD result in the failure of the request. Note that in the SIPS URI scheme, transport is independent of TLS, and thus "sips:alice@atlanta.com;transport=tcp" and "sips:alice@atlanta.com;transport=sctp" are both valid (although note that UDP is not a valid transport for SIPS). The use of "transport=tls" has consequently been deprecated, partly because it was specific to a single hop of the request. This is a change since RFC 2543. Users that distribute a SIPS URI as an address-of-record may elect to operate devices that refuse requests over insecure transports. 26.2.3 HTTP Authentication SIP provides a challenge capability, based on HTTP authentication, that relies on the 401 and 407 response codes as well as header fields for carrying challenges and credentials. Without significant modification, the reuse of the HTTP Digest authentication scheme in SIP allows for replay protection and one-way authentication. The usage of Digest authentication in SIP is detailed in Section 22. 26.2.4 S/MIME As is discussed above, encrypting entire SIP messages end-to-end for the purpose of confidentiality is not appropriate because network intermediaries (like proxy servers) need to view certain header fields in order to route messages correctly, and if these intermediaries are excluded from security associations, then SIP messages will essentially be non-routable. However, S/MIME allows SIP UAs to encrypt MIME bodies within SIP, securing these bodies end-to-end without affecting message headers. S/MIME can provide end-to-end confidentiality and integrity for message bodies, as well as mutual authentication. It is also possible to use S/MIME to provide a form of integrity and confidentiality for SIP header fields through SIP message tunneling. Rosenberg, et. al. Standards Track [Page 240] RFC 3261 SIP: Session Initiation Protocol June 2002 The usage of S/MIME in SIP is detailed in Section 23. 26.3 Implementing Security Mechanisms 26.3.1 Requirements for Implementers of SIP Proxy servers, redirect servers, and registrars MUST implement TLS, and MUST support both mutual and one-way authentication. It is strongly RECOMMENDED that UAs be capable initiating TLS; UAs MAY also be capable of acting as a TLS server. Proxy servers, redirect servers, and registrars SHOULD possess a site certificate whose subject corresponds to their canonical hostname. UAs MAY have certificates of their own for mutual authentication with TLS, but no provisions are set forth in this document for their use. All SIP elements that support TLS MUST have a mechanism for validating certificates received during TLS negotiation; this entails possession of one or more root certificates issued by certificate authorities (preferably well-known distributors of site certificates comparable to those that issue root certificates for web browsers). All SIP elements that support TLS MUST also support the SIPS URI scheme. Proxy servers, redirect servers, registrars, and UAs MAY also implement IPSec or other lower-layer security protocols. When a UA attempts to contact a proxy server, redirect server, or registrar, the UAC SHOULD initiate a TLS connection over which it will send SIP messages. In some architectures, UASs MAY receive requests over such TLS connections as well. Proxy servers, redirect servers, registrars, and UAs MUST implement Digest Authorization, encompassing all of the aspects required in 22. Proxy servers, redirect servers, and registrars SHOULD be configured with at least one Digest realm, and at least one "realm" string supported by a given server SHOULD correspond to the server's hostname or domainname. UAs MAY support the signing and encrypting of MIME bodies, and transference of credentials with S/MIME as described in Section 23. If a UA holds one or more root certificates of certificate authorities in order to validate certificates for TLS or IPSec, it SHOULD be capable of reusing these to verify S/MIME certificates, as appropriate. A UA MAY hold root certificates specifically for validating S/MIME certificates. Rosenberg, et. al. Standards Track [Page 241] RFC 3261 SIP: Session Initiation Protocol June 2002 Note that is it anticipated that future security extensions may upgrade the normative strength associated with S/MIME as S/MIME implementations appear and the problem space becomes better understood. 26.3.2 Security Solutions The operation of these security mechanisms in concert can follow the existing web and email security models to some degree. At a high level, UAs authenticate themselves to servers (proxy servers, redirect servers, and registrars) with a Digest username and password; servers authenticate themselves to UAs one hop away, or to another server one hop away (and vice versa), with a site certificate delivered by TLS. On a peer-to-peer level, UAs trust the network to authenticate one another ordinarily; however, S/MIME can also be used to provide direct authentication when the network does not, or if the network itself is not trusted. The following is an illustrative example in which these security mechanisms are used by various UAs and servers to prevent the sorts of threats described in Section 26.1. While implementers and network administrators MAY follow the normative guidelines given in the remainder of this section, these are provided only as example implementations. 26.3.2.1 Registration When a UA comes online and registers with its local administrative domain, it SHOULD establish a TLS connection with its registrar (Section 10 describes how the UA reaches its registrar). The registrar SHOULD offer a certificate to the UA, and the site identified by the certificate MUST correspond with the domain in which the UA intends to register; for example, if the UA intends to register the address-of-record 'alice@atlanta.com', the site certificate must identify a host within the atlanta.com domain (such as sip.atlanta.com). When it receives the TLS Certificate message, the UA SHOULD verify the certificate and inspect the site identified by the certificate. If the certificate is invalid, revoked, or if it does not identify the appropriate party, the UA MUST NOT send the REGISTER message and otherwise proceed with the registration. When a valid certificate has been provided by the registrar, the UA knows that the registrar is not an attacker who might redirect the UA, steal passwords, or attempt any similar attacks. Rosenberg, et. al. Standards Track [Page 242] RFC 3261 SIP: Session Initiation Protocol June 2002 The UA then creates a REGISTER request that SHOULD be addressed to a Request-URI corresponding to the site certificate received from the registrar. When the UA sends the REGISTER request over the existing TLS connection, the registrar SHOULD challenge the request with a 401 (Proxy Authentication Required) response. The "realm" parameter within the Proxy-Authenticate header field of the response SHOULD correspond to the domain previously given by the site certificate. When the UAC receives the challenge, it SHOULD either prompt the user for credentials or take an appropriate credential from a keyring corresponding to the "realm" parameter in the challenge. The username of this credential SHOULD correspond with the "userinfo" portion of the URI in the To header field of the REGISTER request. Once the Digest credentials have been inserted into an appropriate Proxy-Authorization header field, the REGISTER should be resubmitted to the registrar. Since the registrar requires the user agent to authenticate itself, it would be difficult for an attacker to forge REGISTER requests for the user's address-of-record. Also note that since the REGISTER is sent over a confidential TLS connection, attackers will not be able to intercept the REGISTER to record credentials for any possible replay attack. Once the registration has been accepted by the registrar, the UA SHOULD leave this TLS connection open provided that the registrar also acts as the proxy server to which requests are sent for users in this administrative domain. The existing TLS connection will be reused to deliver incoming requests to the UA that has just completed registration. Because the UA has already authenticated the server on the other side of the TLS connection, all requests that come over this connection are known to have passed through the proxy server - attackers cannot create spoofed requests that appear to have been sent through that proxy server. 26.3.2.2 Interdomain Requests Now let's say that Alice's UA would like to initiate a session with a user in a remote administrative domain, namely "bob@biloxi.com". We will also say that the local administrative domain (atlanta.com) has a local outbound proxy. The proxy server that handles inbound requests for an administrative domain MAY also act as a local outbound proxy; for simplicity's sake we'll assume this to be the case for atlanta.com (otherwise the user agent would initiate a new TLS connection to a separate server at this point). Assuming that the client has completed the registration Rosenberg, et. al. Standards Track [Page 243] RFC 3261 SIP: Session Initiation Protocol June 2002 process described in the preceding section, it SHOULD reuse the TLS connection to the local proxy server when it sends an INVITE request to another user. The UA SHOULD reuse cached credentials in the INVITE to avoid prompting the user unnecessarily. When the local outbound proxy server has validated the credentials presented by the UA in the INVITE, it SHOULD inspect the Request-URI to determine how the message should be routed (see [4]). If the "domainname" portion of the Request-URI had corresponded to the local domain (atlanta.com) rather than biloxi.com, then the proxy server would have consulted its location service to determine how best to reach the requested user. Had "alice@atlanta.com" been attempting to contact, say, "alex@atlanta.com", the local proxy would have proxied to the request to the TLS connection Alex had established with the registrar when he registered. Since Alex would receive this request over his authenticated channel, he would be assured that Alice's request had been authorized by the proxy server of the local administrative domain. However, in this instance the Request-URI designates a remote domain. The local outbound proxy server at atlanta.com SHOULD therefore establish a TLS connection with the remote proxy server at biloxi.com. Since both of the participants in this TLS connection are servers that possess site certificates, mutual TLS authentication SHOULD occur. Each side of the connection SHOULD verify and inspect the certificate of the other, noting the domain name that appears in the certificate for comparison with the header fields of SIP messages. The atlanta.com proxy server, for example, SHOULD verify at this stage that the certificate received from the remote side corresponds with the biloxi.com domain. Once it has done so, and TLS negotiation has completed, resulting in a secure channel between the two proxies, the atlanta.com proxy can forward the INVITE request to biloxi.com. The proxy server at biloxi.com SHOULD inspect the certificate of the proxy server at atlanta.com in turn and compare the domain asserted by the certificate with the "domainname" portion of the From header field in the INVITE request. The biloxi proxy MAY have a strict security policy that requires it to reject requests that do not match the administrative domain from which they have been proxied. Such security policies could be instituted to prevent the SIP equivalent of SMTP 'open relays' that are frequently exploited to generate spam. Rosenberg, et. al. Standards Track [Page 244] RFC 3261 SIP: Session Initiation Protocol June 2002 This policy, however, only guarantees that the request came from the domain it ascribes to itself; it does not allow biloxi.com to ascertain how atlanta.com authenticated Alice. Only if biloxi.com has some other way of knowing atlanta.com's authentication policies could it possibly ascertain how Alice proved her identity. biloxi.com might then institute an even stricter policy that forbids requests that come from domains that are not known administratively to share a common authentication policy with biloxi.com. Once the INVITE has been approved by the biloxi proxy, the proxy server SHOULD identify the existing TLS channel, if any, associated with the user targeted by this request (in this case "bob@biloxi.com"). The INVITE should be proxied through this channel to Bob. Since the request is received over a TLS connection that had previously been authenticated as the biloxi proxy, Bob knows that the From header field was not tampered with and that atlanta.com has validated Alice, although not necessarily whether or not to trust Alice's identity. Before they forward the request, both proxy servers SHOULD add a Record-Route header field to the request so that all future requests in this dialog will pass through the proxy servers. The proxy servers can thereby continue to provide security services for the lifetime of this dialog. If the proxy servers do not add themselves to the Record-Route, future messages will pass directly end-to-end between Alice and Bob without any security services (unless the two parties agree on some independent end-to-end security such as S/MIME). In this respect the SIP trapezoid model can provide a nice structure where conventions of agreement between the site proxies can provide a reasonably secure channel between Alice and Bob. An attacker preying on this architecture would, for example, be unable to forge a BYE request and insert it into the signaling stream between Bob and Alice because the attacker has no way of ascertaining the parameters of the session and also because the integrity mechanism transitively protects the traffic between Alice and Bob. 26.3.2.3 Peer-to-Peer Requests Alternatively, consider a UA asserting the identity "carol@chicago.com" that has no local outbound proxy. When Carol wishes to send an INVITE to "bob@biloxi.com", her UA SHOULD initiate a TLS connection with the biloxi proxy directly (using the mechanism described in [4] to determine how to best to reach the given Request-URI). When her UA receives a certificate from the biloxi proxy, it SHOULD be verified normally before she passes her INVITE across the TLS connection. However, Carol has no means of proving Rosenberg, et. al. Standards Track [Page 245] RFC 3261 SIP: Session Initiation Protocol June 2002 her identity to the biloxi proxy, but she does have a CMS-detached signature over a "message/sip" body in the INVITE. It is unlikely in this instance that Carol would have any credentials in the biloxi.com realm, since she has no formal association with biloxi.com. The biloxi proxy MAY also have a strict policy that precludes it from even bothering to challenge requests that do not have biloxi.com in the "domainname" portion of the From header field - it treats these users as unauthenticated. The biloxi proxy has a policy for Bob that all non-authenticated requests should be redirected to the appropriate contact address registered against 'bob@biloxi.com', namely <sip:bob@192.0.2.4>. Carol receives the redirection response over the TLS connection she established with the biloxi proxy, so she trusts the veracity of the contact address. Carol SHOULD then establish a TCP connection with the designated address and send a new INVITE with a Request-URI containing the received contact address (recomputing the signature in the body as the request is readied). Bob receives this INVITE on an insecure interface, but his UA inspects and, in this instance, recognizes the From header field of the request and subsequently matches a locally cached certificate with the one presented in the signature of the body of the INVITE. He replies in similar fashion, authenticating himself to Carol, and a secure dialog begins. Sometimes firewalls or NATs in an administrative domain could preclude the establishment of a direct TCP connection to a UA. In these cases, proxy servers could also potentially relay requests to UAs in a way that has no trust implications (for example, forgoing an existing TLS connection and forwarding the request over cleartext TCP) as local policy dictates. 26.3.2.4 DoS Protection In order to minimize the risk of a denial-of-service attack against architectures using these security solutions, implementers should take note of the following guidelines. When the host on which a SIP proxy server is operating is routable from the public Internet, it SHOULD be deployed in an administrative domain with defensive operational policies (blocking source-routed traffic, preferably filtering ping traffic). Both TLS and IPSec can also make use of bastion hosts at the edges of administrative domains that participate in the security associations to aggregate secure tunnels and sockets. These bastion hosts can also take the brunt of denial-of-service attacks, ensuring that SIP hosts within the administrative domain are not encumbered with superfluous messaging. Rosenberg, et. al. Standards Track [Page 246] RFC 3261 SIP: Session Initiation Protocol June 2002 No matter what security solutions are deployed, floods of messages directed at proxy servers can lock up proxy server resources and prevent desirable traffic from reaching its destination. There is a computational expense associated with processing a SIP transaction at a proxy server, and that expense is greater for stateful proxy servers than it is for stateless proxy servers. Therefore, stateful proxies are more susceptible to flooding than stateless proxy servers. UAs and proxy servers SHOULD challenge questionable requests with only a single 401 (Unauthorized) or 407 (Proxy Authentication Required), forgoing the normal response retransmission algorithm, and thus behaving statelessly towards unauthenticated requests. Retransmitting the 401 (Unauthorized) or 407 (Proxy Authentication Required) status response amplifies the problem of an attacker using a falsified header field value (such as Via) to direct traffic to a third party. In summary, the mutual authentication of proxy servers through mechanisms such as TLS significantly reduces the potential for rogue intermediaries to introduce falsified requests or responses that can deny service. This commensurately makes it harder for attackers to make innocent SIP nodes into agents of amplification. 26.4 Limitations Although these security mechanisms, when applied in a judicious manner, can thwart many threats, there are limitations in the scope of the mechanisms that must be understood by implementers and network operators. 26.4.1 HTTP Digest One of the primary limitations of using HTTP Digest in SIP is that the integrity mechanisms in Digest do not work very well for SIP. Specifically, they offer protection of the Request-URI and the method of a message, but not for any of the header fields that UAs would most likely wish to secure. The existing replay protection mechanisms described in RFC 2617 also have some limitations for SIP. The next-nonce mechanism, for example, does not support pipelined requests. The nonce-count mechanism should be used for replay protection. Another limitation of HTTP Digest is the scope of realms. Digest is valuable when a user wants to authenticate themselves to a resource with which they have a pre-existing association, like a service Rosenberg, et. al. Standards Track [Page 247] RFC 3261 SIP: Session Initiation Protocol June 2002 provider of which the user is a customer (which is quite a common scenario and thus Digest provides an extremely useful function). By way of contrast, the scope of TLS is interdomain or multirealm, since certificates are often globally verifiable, so that the UA can authenticate the server with no pre-existing association. 26.4.2 S/MIME The largest outstanding defect with the S/MIME mechanism is the lack of a prevalent public key infrastructure for end users. If self- signed certificates (or certificates that cannot be verified by one of the participants in a dialog) are used, the SIP-based key exchange mechanism described in Section 23.2 is susceptible to a man-in-the- middle attack with which an attacker can potentially inspect and modify S/MIME bodies. The attacker needs to intercept the first exchange of keys between the two parties in a dialog, remove the existing CMS-detached signatures from the request and response, and insert a different CMS-detached signature containing a certificate supplied by the attacker (but which seems to be a certificate for the proper address-of-record). Each party will think they have exchanged keys with the other, when in fact each has the public key of the attacker. It is important to note that the attacker can only leverage this vulnerability on the first exchange of keys between two parties - on subsequent occasions, the alteration of the key would be noticeable to the UAs. It would also be difficult for the attacker to remain in the path of all future dialogs between the two parties over time (as potentially days, weeks, or years pass). SSH is susceptible to the same man-in-the-middle attack on the first exchange of keys; however, it is widely acknowledged that while SSH is not perfect, it does improve the security of connections. The use of key fingerprints could provide some assistance to SIP, just as it does for SSH. For example, if two parties use SIP to establish a voice communications session, each could read off the fingerprint of the key they received from the other, which could be compared against the original. It would certainly be more difficult for the man-in- the-middle to emulate the voices of the participants than their signaling (a practice that was used with the Clipper chip-based secure telephone). The S/MIME mechanism allows UAs to send encrypted requests without preamble if they possess a certificate for the destination address- of-record on their keyring. However, it is possible that any particular device registered for an address-of-record will not hold the certificate that has been previously employed by the device's current user, and that it will therefore be unable to process an Rosenberg, et. al. Standards Track [Page 248] RFC 3261 SIP: Session Initiation Protocol June 2002 encrypted request properly, which could lead to some avoidable error signaling. This is especially likely when an encrypted request is forked. The keys associated with S/MIME are most useful when associated with a particular user (an address-of-record) rather than a device (a UA). When users move between devices, it may be difficult to transport private keys securely between UAs; how such keys might be acquired by a device is outside the scope of this document. Another, more prosaic difficulty with the S/MIME mechanism is that it can result in very large messages, especially when the SIP tunneling mechanism described in Section 23.4 is used. For that reason, it is RECOMMENDED that TCP should be used as a transport protocol when S/MIME tunneling is employed. 26.4.3 TLS The most commonly voiced concern about TLS is that it cannot run over UDP; TLS requires a connection-oriented underlying transport protocol, which for the purposes of this document means TCP. It may also be arduous for a local outbound proxy server and/or registrar to maintain many simultaneous long-lived TLS connections with numerous UAs. This introduces some valid scalability concerns, especially for intensive ciphersuites. Maintaining redundancy of long-lived TLS connections, especially when a UA is solely responsible for their establishment, could also be cumbersome. TLS only allows SIP entities to authenticate servers to which they are adjacent; TLS offers strictly hop-by-hop security. Neither TLS, nor any other mechanism specified in this document, allows clients to authenticate proxy servers to whom they cannot form a direct TCP connection. 26.4.4 SIPS URIs Actually using TLS on every segment of a request path entails that the terminating UAS must be reachable over TLS (perhaps registering with a SIPS URI as a contact address). This is the preferred use of SIPS. Many valid architectures, however, use TLS to secure part of the request path, but rely on some other mechanism for the final hop to a UAS, for example. Thus SIPS cannot guarantee that TLS usage will be truly end-to-end. Note that since many UAs will not accept incoming TLS connections, even those UAs that do support TLS may be required to maintain persistent TLS connections as described in the TLS limitations section above in order to receive requests over TLS as a UAS. Rosenberg, et. al. Standards Track [Page 249] RFC 3261 SIP: Session Initiation Protocol June 2002 Location services are not required to provide a SIPS binding for a SIPS Request-URI. Although location services are commonly populated by user registrations (as described in Section 10.2.1), various other protocols and interfaces could conceivably supply contact addresses for an AOR, and these tools are free to map SIPS URIs to SIP URIs as appropriate. When queried for bindings, a location service returns its contact addresses without regard for whether it received a request with a SIPS Request-URI. If a redirect server is accessing the location service, it is up to the entity that processes the Contact header field of a redirection to determine the propriety of the contact addresses. Ensuring that TLS will be used for all of the request segments up to the target domain is somewhat complex. It is possible that cryptographically authenticated proxy servers along the way that are non-compliant or compromised may choose to disregard the forwarding rules associated with SIPS (and the general forwarding rules in Section 16.6). Such malicious intermediaries could, for example, retarget a request from a SIPS URI to a SIP URI in an attempt to downgrade security. Alternatively, an intermediary might legitimately retarget a request from a SIP to a SIPS URI. Recipients of a request whose Request-URI uses the SIPS URI scheme thus cannot assume on the basis of the Request-URI alone that SIPS was used for the entire request path (from the client onwards). To address these concerns, it is RECOMMENDED that recipients of a request whose Request-URI contains a SIP or SIPS URI inspect the To header field value to see if it contains a SIPS URI (though note that it does not constitute a breach of security if this URI has the same scheme but is not equivalent to the URI in the To header field). Although clients may choose to populate the Request-URI and To header field of a request differently, when SIPS is used this disparity could be interpreted as a possible security violation, and the request could consequently be rejected by its recipient. Recipients MAY also inspect the Via header chain in order to double-check whether or not TLS was used for the entire request path until the local administrative domain was reached. S/MIME may also be used by the originating UAC to help ensure that the original form of the To header field is carried end-to-end. If the UAS has reason to believe that the scheme of the Request-URI has been improperly modified in transit, the UA SHOULD notify its user of a potential security breach. Rosenberg, et. al. Standards Track [Page 250] RFC 3261 SIP: Session Initiation Protocol June 2002 As a further measure to prevent downgrade attacks, entities that accept only SIPS requests MAY also refuse connections on insecure ports. End users will undoubtedly discern the difference between SIPS and SIP URIs, and they may manually edit them in response to stimuli. This can either benefit or degrade security. For example, if an attacker corrupts a DNS cache, inserting a fake record set that effectively removes all SIPS records for a proxy server, then any SIPS requests that traverse this proxy server may fail. When a user, however, sees that repeated calls to a SIPS AOR are failing, they could on some devices manually convert the scheme from SIPS to SIP and retry. Of course, there are some safeguards against this (if the destination UA is truly paranoid it could refuse all non-SIPS requests), but it is a limitation worth noting. On the bright side, users might also divine that 'SIPS' would be valid even when they are presented only with a SIP URI. 26.5 Privacy SIP messages frequently contain sensitive information about their senders - not just what they have to say, but with whom they communicate, when they communicate and for how long, and from where they participate in sessions. Many applications and their users require that this sort of private information be hidden from any parties that do not need to know it. Note that there are also less direct ways in which private information can be divulged. If a user or service chooses to be reachable at an address that is guessable from the person's name and organizational affiliation (which describes most addresses-of- record), the traditional method of ensuring privacy by having an unlisted "phone number" is compromised. A user location service can infringe on the privacy of the recipient of a session invitation by divulging their specific whereabouts to the caller; an implementation consequently SHOULD be able to restrict, on a per-user basis, what kind of location and availability information is given out to certain classes of callers. This is a whole class of problem that is expected to be studied further in ongoing SIP work. In some cases, users may want to conceal personal information in header fields that convey identity. This can apply not only to the From and related headers representing the originator of the request, but also the To - it may not be appropriate to convey to the final destination a speed-dialing nickname, or an unexpanded identifier for a group of targets, either of which would be removed from the Request-URI as the request is routed, but not changed in the To Rosenberg, et. al. Standards Track [Page 251] RFC 3261 SIP: Session Initiation Protocol June 2002 header field if the two were initially identical. Thus it MAY be desirable for privacy reasons to create a To header field that differs from the Request-URI. 27 IANA Considerations All method names, header field names, status codes, and option tags used in SIP applications are registered with IANA through instructions in an IANA Considerations section in an RFC. The specification instructs the IANA to create four new sub- registries under http://www.iana.org/assignments/sip-parameters: Option Tags, Warning Codes (warn-codes), Methods and Response Codes, added to the sub-registry of Header Fields that is already present there. 27.1 Option Tags This specification establishes the Option Tags sub-registry under http://www.iana.org/assignments/sip-parameters. Option tags are used in header fields such as Require, Supported, Proxy-Require, and Unsupported in support of SIP compatibility mechanisms for extensions (Section 19.2). The option tag itself is a string that is associated with a particular SIP option (that is, an extension). It identifies the option to SIP endpoints. Option tags are registered by the IANA when they are published in standards track RFCs. The IANA Considerations section of the RFC must include the following information, which appears in the IANA registry along with the RFC number of the publication. o Name of the option tag. The name MAY be of any length, but SHOULD be no more than twenty characters long. The name MUST consist of alphanum (Section 25) characters only. o Descriptive text that describes the extension. 27.2 Warn-Codes This specification establishes the Warn-codes sub-registry under http://www.iana.org/assignments/sip-parameters and initiates its population with the warn-codes listed in Section 20.43. Additional warn-codes are registered by RFC publication. Rosenberg, et. al. Standards Track [Page 252] RFC 3261 SIP: Session Initiation Protocol June 2002 The descriptive text for the table of warn-codes is: Warning codes provide information supplemental to the status code in SIP response messages when the failure of the transaction results from a Session Description Protocol (SDP) (RFC 2327 [1]) problem. The "warn-code" consists of three digits. A first digit of "3" indicates warnings specific to SIP. Until a future specification describes uses of warn-codes other than 3xx, only 3xx warn-codes may be registered. Warnings 300 through 329 are reserved for indicating problems with keywords in the session description, 330 through 339 are warnings related to basic network services requested in the session description, 370 through 379 are warnings related to quantitative QoS parameters requested in the session description, and 390 through 399 are miscellaneous warnings that do not fall into one of the above categories. 27.3 Header Field Names This obsoletes the IANA instructions about the header sub-registry under http://www.iana.org/assignments/sip-parameters. The following information needs to be provided in an RFC publication in order to register a new header field name: o The RFC number in which the header is registered; o the name of the header field being registered; o a compact form version for that header field, if one is defined; Some common and widely used header fields MAY be assigned one-letter compact forms (Section 7.3.3). Compact forms can only be assigned after SIP working group review, followed by RFC publication. 27.4 Method and Response Codes This specification establishes the Method and Response-Code sub- registries under http://www.iana.org/assignments/sip-parameters and initiates their population as follows. The initial Methods table is: Rosenberg, et. al. Standards Track [Page 253] RFC 3261 SIP: Session Initiation Protocol June 2002 INVITE [RFC3261] ACK [RFC3261] BYE [RFC3261] CANCEL [RFC3261] REGISTER [RFC3261] OPTIONS [RFC3261] INFO [RFC2976] The response code table is initially populated from Section 21, the portions labeled Informational, Success, Redirection, Client-Error, Server-Error, and Global-Failure. The table has the following format: Type (e.g., Informational) Number Default Reason Phrase [RFC3261] The following information needs to be provided in an RFC publication in order to register a new response code or method: o The RFC number in which the method or response code is registered; o the number of the response code or name of the method being registered; o the default reason phrase for that response code, if applicable; 27.5 The "message/sip" MIME type. This document registers the "message/sip" MIME media type in order to allow SIP messages to be tunneled as bodies within SIP, primarily for end-to-end security purposes. This media type is defined by the following information: Media type name: message Media subtype name: sip Required parameters: none Optional parameters: version version: The SIP-Version number of the enclosed message (e.g., "2.0"). If not present, the version defaults to "2.0". Encoding scheme: SIP messages consist of an 8-bit header optionally followed by a binary MIME data object. As such, SIP messages must be treated as binary. Under normal circumstances SIP messages are transported over binary-capable transports, no special encodings are needed. Rosenberg, et. al. Standards Track [Page 254] RFC 3261 SIP: Session Initiation Protocol June 2002 Security considerations: see below Motivation and examples of this usage as a security mechanism in concert with S/MIME are given in 23.4. 27.6 New Content-Disposition Parameter Registrations This document also registers four new Content-Disposition header "disposition-types": alert, icon, session and render. The authors request that these values be recorded in the IANA registry for Content-Dispositions. Descriptions of these "disposition-types", including motivation and examples, are given in Section 20.11. Short descriptions suitable for the IANA registry are: alert the body is a custom ring tone to alert the user icon the body is displayed as an icon to the user render the body should be displayed to the user session the body describes a communications session, for example, as RFC 2327 SDP body 28 Changes From RFC 2543 This RFC revises RFC 2543. It is mostly backwards compatible with RFC 2543. The changes described here fix many errors discovered in RFC 2543 and provide information on scenarios not detailed in RFC 2543. The protocol has been presented in a more cleanly layered model here. We break the differences into functional behavior that is a substantial change from RFC 2543, which has impact on interoperability or correct operation in some cases, and functional behavior that is different from RFC 2543 but not a potential source of interoperability problems. There have been countless clarifications as well, which are not documented here. 28.1 Major Functional Changes o When a UAC wishes to terminate a call before it has been answered, it sends CANCEL. If the original INVITE still returns a 2xx, the UAC then sends BYE. BYE can only be sent on an existing call leg (now called a dialog in this RFC), whereas it could be sent at any time in RFC 2543. o The SIP BNF was converted to be RFC 2234 compliant. Rosenberg, et. al. Standards Track [Page 255] RFC 3261 SIP: Session Initiation Protocol June 2002 o SIP URL BNF was made more general, allowing a greater set of characters in the user part. Furthermore, comparison rules were simplified to be primarily case-insensitive, and detailed handling of comparison in the presence of parameters was described. The most substantial change is that a URI with a parameter with the default value does not match a URI without that parameter. o Removed Via hiding. It had serious trust issues, since it relied on the next hop to perform the obfuscation process. Instead, Via hiding can be done as a local implementation choice in stateful proxies, and thus is no longer documented. o In RFC 2543, CANCEL and INVITE transactions were intermingled. They are separated now. When a user sends an INVITE and then a CANCEL, the INVITE transaction still terminates normally. A UAS needs to respond to the original INVITE request with a 487 response. o Similarly, CANCEL and BYE transactions were intermingled; RFC 2543 allowed the UAS not to send a response to INVITE when a BYE was received. That is disallowed here. The original INVITE needs a response. o In RFC 2543, UAs needed to support only UDP. In this RFC, UAs need to support both UDP and TCP. o In RFC 2543, a forking proxy only passed up one challenge from downstream elements in the event of multiple challenges. In this RFC, proxies are supposed to collect all challenges and place them into the forwarded response. o In Digest credentials, the URI needs to be quoted; this is unclear from RFC 2617 and RFC 2069 which are both inconsistent on it. o SDP processing has been split off into a separate specification [13], and more fully specified as a formal offer/answer exchange process that is effectively tunneled through SIP. SDP is allowed in INVITE/200 or 200/ACK for baseline SIP implementations; RFC 2543 alluded to the ability to use it in INVITE, 200, and ACK in a single transaction, but this was not well specified. More complex SDP usages are allowed in extensions. Rosenberg, et. al. Standards Track [Page 256] RFC 3261 SIP: Session Initiation Protocol June 2002 o Added full support for IPv6 in URIs and in the Via header field. Support for IPv6 in Via has required that its header field parameters allow the square bracket and colon characters. These characters were previously not permitted. In theory, this could cause interop problems with older implementations. However, we have observed that most implementations accept any non-control ASCII character in these parameters. o DNS SRV procedure is now documented in a separate specification [4]. This procedure uses both SRV and NAPTR resource records and no longer combines data from across SRV records as described in RFC 2543. o Loop detection has been made optional, supplanted by a mandatory usage of Max-Forwards. The loop detection procedure in RFC 2543 had a serious bug which would report "spirals" as an error condition when it was not. The optional loop detection procedure is more fully and correctly specified here. o Usage of tags is now mandatory (they were optional in RFC 2543), as they are now the fundamental building blocks of dialog identification. o Added the Supported header field, allowing for clients to indicate what extensions are supported to a server, which can apply those extensions to the response, and indicate their usage with a Require in the response. o Extension parameters were missing from the BNF for several header fields, and they have been added. o Handling of Route and Record-Route construction was very underspecified in RFC 2543, and also not the right approach. It has been substantially reworked in this specification (and made vastly simpler), and this is arguably the largest change. Backwards compatibility is still provided for deployments that do not use "pre-loaded routes", where the initial request has a set of Route header field values obtained in some way outside of Record-Route. In those situations, the new mechanism is not interoperable. o In RFC 2543, lines in a message could be terminated with CR, LF, or CRLF. This specification only allows CRLF. Rosenberg, et. al. Standards Track [Page 257] RFC 3261 SIP: Session Initiation Protocol June 2002 o Usage of Route in CANCEL and ACK was not well defined in RFC 2543. It is now well specified; if a request had a Route header field, its CANCEL or ACK for a non-2xx response to the request need to carry the same Route header field values. ACKs for 2xx responses use the Route values learned from the Record-Route of the 2xx responses. o RFC 2543 allowed multiple requests in a single UDP packet. This usage has been removed. o Usage of absolute time in the Expires header field and parameter has been removed. It caused interoperability problems in elements that were not time synchronized, a common occurrence. Relative times are used instead. o The branch parameter of the Via header field value is now mandatory for all elements to use. It now plays the role of a unique transaction identifier. This avoids the complex and bug- laden transaction identification rules from RFC 2543. A magic cookie is used in the parameter value to determine if the previous hop has made the parameter globally unique, and comparison falls back to the old rules when it is not present. Thus, interoperability is assured. o In RFC 2543, closure of a TCP connection was made equivalent to a CANCEL. This was nearly impossible to implement (and wrong) for TCP connections between proxies. This has been eliminated, so that there is no coupling between TCP connection state and SIP processing. o RFC 2543 was silent on whether a UA could initiate a new transaction to a peer while another was in progress. That is now specified here. It is allowed for non-INVITE requests, disallowed for INVITE. o PGP was removed. It was not sufficiently specified, and not compatible with the more complete PGP MIME. It was replaced with S/MIME. o Added the "sips" URI scheme for end-to-end TLS. This scheme is not backwards compatible with RFC 2543. Existing elements that receive a request with a SIPS URI scheme in the Request-URI will likely reject the request. This is actually a feature; it ensures that a call to a SIPS URI is only delivered if all path hops can be secured. Rosenberg, et. al. Standards Track [Page 258] RFC 3261 SIP: Session Initiation Protocol June 2002 o Additional security features were added with TLS, and these are described in a much larger and complete security considerations section. o In RFC 2543, a proxy was not required to forward provisional responses from 101 to 199 upstream. This was changed to MUST. This is important, since many subsequent features depend on delivery of all provisional responses from 101 to 199. o Little was said about the 503 response code in RFC 2543. It has since found substantial use in indicating failure or overload conditions in proxies. This requires somewhat special treatment. Specifically, receipt of a 503 should trigger an attempt to contact the next element in the result of a DNS SRV lookup. Also, 503 response is only forwarded upstream by a proxy under certain conditions. o RFC 2543 defined, but did no sufficiently specify, a mechanism for UA authentication of a server. That has been removed. Instead, the mutual authentication procedures of RFC 2617 are allowed. o A UA cannot send a BYE for a call until it has received an ACK for the initial INVITE. This was allowed in RFC 2543 but leads to a potential race condition. o A UA or proxy cannot send CANCEL for a transaction until it gets a provisional response for the request. This was allowed in RFC 2543 but leads to potential race conditions. o The action parameter in registrations has been deprecated. It was insufficient for any useful services, and caused conflicts when application processing was applied in proxies. o RFC 2543 had a number of special cases for multicast. For example, certain responses were suppressed, timers were adjusted, and so on. Multicast now plays a more limited role, and the protocol operation is unaffected by usage of multicast as opposed to unicast. The limitations as a result of that are documented. o Basic authentication has been removed entirely and its usage forbidden. Rosenberg, et. al. Standards Track [Page 259] RFC 3261 SIP: Session Initiation Protocol June 2002 o Proxies no longer forward a 6xx immediately on receiving it. Instead, they CANCEL pending branches immediately. This avoids a potential race condition that would result in a UAC getting a 6xx followed by a 2xx. In all cases except this race condition, the result will be the same - the 6xx is forwarded upstream. o RFC 2543 did not address the problem of request merging. This occurs when a request forks at a proxy and later rejoins at an element. Handling of merging is done only at a UA, and procedures are defined for rejecting all but the first request. 28.2 Minor Functional Changes o Added the Alert-Info, Error-Info, and Call-Info header fields for optional content presentation to users. o Added the Content-Language, Content-Disposition and MIME-Version header fields. o Added a "glare handling" mechanism to deal with the case where both parties send each other a re-INVITE simultaneously. It uses the new 491 (Request Pending) error code. o Added the In-Reply-To and Reply-To header fields for supporting the return of missed calls or messages at a later time. o Added TLS and SCTP as valid SIP transports. o There were a variety of mechanisms described for handling failures at any time during a call; those are now generally unified. BYE is sent to terminate. o RFC 2543 mandated retransmission of INVITE responses over TCP, but noted it was really only needed for 2xx. That was an artifact of insufficient protocol layering. With a more coherent transaction layer defined here, that is no longer needed. Only 2xx responses to INVITEs are retransmitted over TCP. o Client and server transaction machines are now driven based on timeouts rather than retransmit counts. This allows the state machines to be properly specified for TCP and UDP. o The Date header field is used in REGISTER responses to provide a simple means for auto-configuration of dates in user agents. o Allowed a registrar to reject registrations with expirations that are too short in duration. Defined the 423 response code and the Min-Expires for this purpose. Rosenberg, et. al. Standards Track [Page 260] RFC 3261 SIP: Session Initiation Protocol June 2002 29 Normative References [1] Handley, M. and V. Jacobson, "SDP: Session Description Protocol", RFC 2327, April 1998. [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [3] Resnick, P., "Internet Message Format", RFC 2822, April 2001. [4] Rosenberg, J. and H. Schulzrinne, "SIP: Locating SIP Servers", RFC 3263, June 2002. [5] Berners-Lee, T., Fielding, R. and L. Masinter, "Uniform Resource Identifiers (URI): Generic Syntax", RFC 2396, August 1998. [6] Chown, P., "Advanced Encryption Standard (AES) Ciphersuites for Transport Layer Security (TLS)", RFC 3268, June 2002. [7] Yergeau, F., "UTF-8, a transformation format of ISO 10646", RFC 2279, January 1998. [8] Fielding, R., Gettys, J., Mogul, J., Frystyk, H., Masinter, L., Leach, P. and T. Berners-Lee, "Hypertext Transfer Protocol – HTTP/1.1", RFC 2616, June 1999. [9] Vaha-Sipila, A., "URLs for Telephone Calls", RFC 2806, April 2000. [10] Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", RFC 2234, November 1997. [11] Freed, F. and N. Borenstein, "Multipurpose Internet Mail Extensions (MIME) Part Two: Media Types", RFC 2046, November 1996. [12] Eastlake, D., Crocker, S. and J. Schiller, "Randomness Recommendations for Security", RFC 1750, December 1994. [13] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with SDP", RFC 3264, June 2002. [14] Postel, J., "User Datagram Protocol", STD 6, RFC 768, August 1980. [15] Postel, J., "DoD Standard Transmission Control Protocol", RFC 761, January 1980. Rosenberg, et. al. Standards Track [Page 261] RFC 3261 SIP: Session Initiation Protocol June 2002 [16] Stewart, R., Xie, Q., Morneault, K., Sharp, C., Schwarzbauer, H., Taylor, T., Rytina, I., Kalla, M., Zhang, L. and V. Paxson, "Stream Control Transmission Protocol", RFC 2960, October 2000. [17] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., Leach, P., Luotonen, A. and L. Stewart, "HTTP authentication: Basic and Digest Access Authentication", RFC 2617, June 1999. [18] Troost, R., Dorner, S. and K. Moore, "Communicating Presentation Information in Internet Messages: The Content-Disposition Header Field", RFC 2183, August 1997. [19] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, F., Watson, M. and M. Zonoun, "MIME media types for ISUP and QSIG Objects", RFC 3204, December 2001. [20] Braden, R., "Requirements for Internet Hosts - Application and Support", STD 3, RFC 1123, October 1989. [21] Alvestrand, H., "IETF Policy on Character Sets and Languages", BCP 18, RFC 2277, January 1998. [22] Galvin, J., Murphy, S., Crocker, S. and N. Freed, "Security Multiparts for MIME: Multipart/Signed and Multipart/Encrypted", RFC 1847, October 1995. [23] Housley, R., "Cryptographic Message Syntax", RFC 2630, June 1999. [24] Ramsdell B., "S/MIME Version 3 Message Specification", RFC 2633, June 1999. [25] Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", RFC 2246, January 1999. [26] Kent, S. and R. Atkinson, "Security Architecture for the Internet Protocol", RFC 2401, November 1998. 30 Informative References [27] R. Pandya, "Emerging mobile and personal communication systems," IEEE Communications Magazine, Vol. 33, pp. 44–52, June 1995. [28] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", RFC 1889, January 1996. Rosenberg, et. al. Standards Track [Page 262] RFC 3261 SIP: Session Initiation Protocol June 2002 [29] Schulzrinne, H., Rao, R. and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April 1998. [30] Cuervo, F., Greene, N., Rayhan, A., Huitema, C., Rosen, B. and J. Segers, "Megaco Protocol Version 1.0", RFC 3015, November 2000. [31] Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg, "SIP: Session Initiation Protocol", RFC 2543, March 1999. [32] Hoffman, P., Masinter, L. and J. Zawinski, "The mailto URL scheme", RFC 2368, July 1998. [33] E. M. Schooler, "A multicast user directory service for synchronous rendezvous," Master's Thesis CS-TR-96-18, Department of Computer Science, California Institute of Technology, Pasadena, California, Aug. 1996. [34] Donovan, S., "The SIP INFO Method", RFC 2976, October 2000. [35] Rivest, R., "The MD5 Message-Digest Algorithm", RFC 1321, April 1992. [36] Dawson, F. and T. Howes, "vCard MIME Directory Profile", RFC 2426, September 1998. [37] Good, G., "The LDAP Data Interchange Format (LDIF) - Technical Specification", RFC 2849, June 2000. [38] Palme, J., "Common Internet Message Headers", RFC 2076, February 1997. [39] Franks, J., Hallam-Baker, P., Hostetler, J., Leach, P., Luotonen, A., Sink, E. and L. Stewart, "An Extension to HTTP: Digest Access Authentication", RFC 2069, January 1997. [40] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., Willis, D., Rosenberg, J., Summers, K. and H. Schulzrinne, "SIP Call Flow Examples", Work in Progress. [41] E. M. Schooler, "Case study: multimedia conference control in a packet-switched teleconferencing system," Journal of Internetworking: Research and Experience, Vol. 4, pp. 99–120, June 1993. ISI reprint series ISI/RS-93-359. Rosenberg, et. al. Standards Track [Page 263] RFC 3261 SIP: Session Initiation Protocol June 2002 [42] H. Schulzrinne, "Personal mobility for multimedia services in the Internet," in European Workshop on Interactive Distributed Multimedia Systems and Services (IDMS), (Berlin, Germany), Mar. 1996. [43] Floyd, S., "Congestion Control Principles", RFC 2914, September 2000. Rosenberg, et. al. Standards Track [Page 264] RFC 3261 SIP: Session Initiation Protocol June 2002 A Table of Timer Values Table 4 summarizes the meaning and defaults of the various timers used by this specification. Timer Value Section Meaning ———————————————————————- T1 500ms default Section 17.1.1.1 RTT Estimate T2 4s Section 17.1.2.2 The maximum retransmit interval for non-INVITE requests and INVITE responses T4 5s Section 17.1.2.2 Maximum duration a message will remain in the network Timer A initially T1 Section 17.1.1.2 INVITE request retransmit interval, for UDP only Timer B 64*T1 Section 17.1.1.2 INVITE transaction timeout timer Timer C > 3min Section 16.6 proxy INVITE transaction bullet 11 timeout Timer D > 32s for UDP Section 17.1.1.2 Wait time for response 0s for TCP/SCTP retransmits Timer E initially T1 Section 17.1.2.2 non-INVITE request retransmit interval, UDP only Timer F 64*T1 Section 17.1.2.2 non-INVITE transaction timeout timer Timer G initially T1 Section 17.2.1 INVITE response retransmit interval Timer H 64*T1 Section 17.2.1 Wait time for ACK receipt Timer I T4 for UDP Section 17.2.1 Wait time for 0s for TCP/SCTP ACK retransmits Timer J 64*T1 for UDP Section 17.2.2 Wait time for 0s for TCP/SCTP non-INVITE request retransmits Timer K T4 for UDP Section 17.1.2.2 Wait time for 0s for TCP/SCTP response retransmits Table 4: Summary of timers Rosenberg, et. al. Standards Track [Page 265] RFC 3261 SIP: Session Initiation Protocol June 2002 Acknowledgments We wish to thank the members of the IETF MMUSIC and SIP WGs for their comments and suggestions. Detailed comments were provided by Ofir Arkin, Brian Bidulock, Jim Buller, Neil Deason, Dave Devanathan, Keith Drage, Bill Fenner, Cedric Fluckiger, Yaron Goland, John Hearty, Bernie Hoeneisen, Jo Hornsby, Phil Hoffer, Christian Huitema, Hisham Khartabil, Jean Jervis, Gadi Karmi, Peter Kjellerstedt, Anders Kristensen, Jonathan Lennox, Gethin Liddell, Allison Mankin, William Marshall, Rohan Mahy, Keith Moore, Vern Paxson, Bob Penfield, Moshe J. Sambol, Chip Sharp, Igor Slepchin, Eric Tremblay, and Rick Workman. Brian Rosen provided the compiled BNF. Jean Mahoney provided technical writing assistance. This work is based, inter alia, on [41,42]. Rosenberg, et. al. Standards Track [Page 266] RFC 3261 SIP: Session Initiation Protocol June 2002 Authors' Addresses Authors addresses are listed alphabetically for the editors, the writers, and then the original authors of RFC 2543. All listed authors actively contributed large amounts of text to this document. Jonathan Rosenberg dynamicsoft 72 Eagle Rock Ave East Hanover, NJ 07936 USA EMail: jdrosen@dynamicsoft.com Henning Schulzrinne Dept. of Computer Science Columbia University 1214 Amsterdam Avenue New York, NY 10027 USA EMail: schulzrinne@cs.columbia.edu Gonzalo Camarillo Ericsson Advanced Signalling Research Lab. FIN-02420 Jorvas Finland EMail: Gonzalo.Camarillo@ericsson.com Alan Johnston WorldCom 100 South 4th Street St. Louis, MO 63102 USA EMail: alan.johnston@wcom.com Rosenberg, et. al. Standards Track [Page 267] RFC 3261 SIP: Session Initiation Protocol June 2002 Jon Peterson NeuStar, Inc 1800 Sutter Street, Suite 570 Concord, CA 94520 USA EMail: jon.peterson@neustar.com Robert Sparks dynamicsoft, Inc. 5100 Tennyson Parkway Suite 1200 Plano, Texas 75024 USA EMail: rsparks@dynamicsoft.com Mark Handley International Computer Science Institute 1947 Center St, Suite 600 Berkeley, CA 94704 USA EMail: mjh@icir.org Eve Schooler AT&T Labs-Research 75 Willow Road Menlo Park, CA 94025 USA EMail: schooler@research.att.com Rosenberg, et. al. Standards Track [Page 268] RFC 3261 SIP: Session Initiation Protocol June 2002 Full Copyright Statement Copyright (C) The Internet Society (2002). 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. Acknowledgement Funding for the RFC Editor function is currently provided by the Internet Society. Rosenberg, et. al. Standards Track [Page 269]

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