💾 Archived View for gemini.bortzmeyer.org › rfc-mirror › rfc3261.txt captured on 2023-06-14 at 17:51:38.
⬅️ Previous capture (2021-11-30)
-=-=-=-=-=-=-
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.
Rosenberg, et. al. Standards Track [Page 47]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 48]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 49]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 50]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 53]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 54]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 55]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 57]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 58]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 59]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 60]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 61]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 62]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 63]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 64]
RFC 3261 SIP: Session Initiation Protocol June 2002
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:
- If the field value has an "expires" parameter, that value
MUST be taken as the requested expiration.
- If there is no such parameter, but the request has an
Expires header field, that value MUST be taken as the
requested expiration.
- 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.
Rosenberg, et. al. Standards Track [Page 66]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 67]
RFC 3261 SIP: Session Initiation Protocol June 2002
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].
Rosenberg, et. al. Standards Track [Page 68]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 69]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 70]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 71]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 72]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 73]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 74]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 75]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 76]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 77]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 78]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 79]
RFC 3261 SIP: Session Initiation Protocol June 2002
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".
Rosenberg, et. al. Standards Track [Page 80]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 81]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 82]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 83]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 84]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 85]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 86]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 87]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 88]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 89]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 90]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 92]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 93]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 94]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 95]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 96]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 97]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 98]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 99]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 100]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 101]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 102]
RFC 3261 SIP: Session Initiation Protocol June 2002
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:
Rosenberg, et. al. Standards Track [Page 103]
RFC 3261 SIP: Session Initiation Protocol June 2002
- The proxy MUST place the Request-URI into the Route header
field as the last value.
- 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.
Rosenberg, et. al. Standards Track [Page 104]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 106]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 107]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 108]
RFC 3261 SIP: Session Initiation Protocol June 2002
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:
- Any provisional response other than 100 (Trying)
- 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
Rosenberg, et. al. Standards Track [Page 109]
RFC 3261 SIP: Session Initiation Protocol June 2002
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:
- 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
Rosenberg, et. al. Standards Track [Page 110]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 111]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 112]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 113]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 114]
RFC 3261 SIP: Session Initiation Protocol June 2002
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).
Rosenberg, et. al. Standards Track [Page 115]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 116]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 117]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 118]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 147]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 149]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 150]
RFC 3261 SIP: Session Initiation Protocol June 2002
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).
Rosenberg, et. al. Standards Track [Page 151]
RFC 3261 SIP: Session Initiation Protocol June 2002
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".
Rosenberg, et. al. Standards Track [Page 152]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 153]
RFC 3261 SIP: Session Initiation Protocol June 2002
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:
- Any uri-parameter appearing in both URIs must match.
- A user, ttl, or method uri-parameter appearing in only one
URI never matches, even if it contains the default value.
- A URI that includes an maddr parameter will not match a URI
that contains no maddr parameter.
- All other uri-parameters appearing in only one URI are
ignored when comparing the URIs.
Rosenberg, et. al. Standards Track [Page 154]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 156]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 157]
RFC 3261 SIP: Session Initiation Protocol June 2002
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).
Rosenberg, et. al. Standards Track [Page 158]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 159]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 160]
RFC 3261 SIP: Session Initiation Protocol June 2002
*: The header field is required if the message body is not empty.
See Sections 20.14, 20.15 and 7.4 for details.
-: 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
Rosenberg, et. al. Standards Track [Page 176]
RFC 3261 SIP: Session Initiation Protocol June 2002
(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.
Rosenberg, et. al. Standards Track [Page 177]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 178]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 179]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 180]
RFC 3261 SIP: Session Initiation Protocol June 2002
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).
Rosenberg, et. al. Standards Track [Page 181]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 182]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 183]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 184]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 185]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 186]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 187]
RFC 3261 SIP: Session Initiation Protocol June 2002
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).
Rosenberg, et. al. Standards Track [Page 188]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 189]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 190]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 191]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 192]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 194]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 195]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 200]
RFC 3261 SIP: Session Initiation Protocol June 2002
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.
Rosenberg, et. al. Standards Track [Page 201]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
Rosenberg, et. al. Standards Track [Page 202]
RFC 3261 SIP: Session Initiation Protocol June 2002
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
--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
--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
--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
--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
--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
--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]