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Network Working Group M. Handley
Request for Comments: 2543 ACIRI
Category: Standards Track H. Schulzrinne
Columbia U.
E. Schooler
Cal Tech
J. Rosenberg
Bell Labs
March 1999
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 (1999). All Rights Reserved.
IESG Note
The IESG intends to charter, in the near future, one or more working
groups to produce standards for "name lookup", where such names would
include electronic mail addresses and telephone numbers, and the
result of such a lookup would be a list of attributes and
characteristics of the user or terminal associated with the name.
Groups which are in need of a "name lookup" protocol should follow
the development of these new working groups rather than using SIP for
this function. In addition it is anticipated that SIP will migrate
towards using such protocols, and SIP implementors are advised to
monitor these efforts.
Abstract
The Session Initiation Protocol (SIP) is an application-layer control
(signaling) protocol for creating, modifying and terminating sessions
with one or more participants. These sessions include Internet
multimedia conferences, Internet telephone calls and multimedia
distribution. Members in a session can communicate via multicast or
via a mesh of unicast relations, or a combination of these.
SIP invitations used to create sessions carry session descriptions
which allow participants to agree on a set of compatible media types.
SIP supports user mobility by proxying and redirecting requests to
the user's current location. Users can register their current
location. SIP is not tied to any particular conference control
protocol. SIP is designed to be independent of the lower-layer
transport protocol and can be extended with additional capabilities.
Table of Contents
1 Introduction ........................................ 7
1.1 Overview of SIP Functionality ....................... 7
1.2 Terminology ......................................... 8
1.3 Definitions ......................................... 9
1.4 Overview of SIP Operation ........................... 12
1.4.1 SIP Addressing ...................................... 12
1.4.2 Locating a SIP Server ............................... 13
1.4.3 SIP Transaction ..................................... 14
1.4.4 SIP Invitation ...................................... 15
1.4.5 Locating a User ..................................... 17
1.4.6 Changing an Existing Session ........................ 18
1.4.7 Registration Services ............................... 18
1.5 Protocol Properties ................................. 18
1.5.1 Minimal State ....................................... 18
1.5.2 Lower-Layer-Protocol Neutral ........................ 18
1.5.3 Text-Based .......................................... 20
2 SIP Uniform Resource Locators ....................... 20
3 SIP Message Overview ................................ 24
4 Request ............................................. 26
4.1 Request-Line ........................................ 26
4.2 Methods ............................................. 27
4.2.1 INVITE .............................................. 28
4.2.2 ACK ................................................. 29
4.2.3 OPTIONS ............................................. 29
4.2.4 BYE ................................................. 30
4.2.5 CANCEL .............................................. 30
4.2.6 REGISTER ............................................ 31
4.3 Request-URI ......................................... 34
4.3.1 SIP Version ......................................... 35
4.4 Option Tags ......................................... 35
4.4.1 Registering New Option Tags with IANA ............... 35
5 Response ............................................ 36
5.1 Status-Line ......................................... 36
5.1.1 Status Codes and Reason Phrases ..................... 37
6 Header Field Definitions ............................ 39
6.1 General Header Fields ............................... 41
6.2 Entity Header Fields ................................ 42
6.3 Request Header Fields ............................... 43
6.4 Response Header Fields .............................. 43
6.5 End-to-end and Hop-by-hop Headers ................... 43
6.6 Header Field Format ................................. 43
6.7 Accept .............................................. 44
6.8 Accept-Encoding ..................................... 44
6.9 Accept-Language ..................................... 45
6.10 Allow ............................................... 45
6.11 Authorization ....................................... 45
6.12 Call-ID ............................................. 46
6.13 Contact ............................................. 47
6.14 Content-Encoding .................................... 50
6.15 Content-Length ...................................... 51
6.16 Content-Type ........................................ 51
6.17 CSeq ................................................ 52
6.18 Date ................................................ 53
6.19 Encryption .......................................... 54
6.20 Expires ............................................. 55
6.21 From ................................................ 56
6.22 Hide ................................................ 57
6.23 Max-Forwards ........................................ 59
6.24 Organization ........................................ 59
6.25 Priority ............................................ 60
6.26 Proxy-Authenticate .................................. 60
6.27 Proxy-Authorization ................................. 61
6.28 Proxy-Require ....................................... 61
6.29 Record-Route ........................................ 62
6.30 Require ............................................. 63
6.31 Response-Key ........................................ 63
6.32 Retry-After ......................................... 64
6.33 Route ............................................... 65
6.34 Server .............................................. 65
6.35 Subject ............................................. 65
6.36 Timestamp ........................................... 66
6.37 To .................................................. 66
6.38 Unsupported ......................................... 68
6.39 User-Agent .......................................... 68
6.40 Via ................................................. 68
6.40.1 Requests ............................................ 68
6.40.2 Receiver-tagged Via Header Fields ................... 69
6.40.3 Responses ........................................... 70
6.40.4 User Agent and Redirect Servers ..................... 70
6.40.5 Syntax .............................................. 71
6.41 Warning ............................................. 72
6.42 WWW-Authenticate .................................... 74
7 Status Code Definitions ............................. 75
7.1 Informational 1xx ................................... 75
7.1.1 100 Trying .......................................... 75
7.1.2 180 Ringing ......................................... 75
7.1.3 181 Call Is Being Forwarded ......................... 75
7.1.4 182 Queued .......................................... 76
7.2 Successful 2xx ...................................... 76
7.2.1 200 OK .............................................. 76
7.3 Redirection 3xx ..................................... 76
7.3.1 300 Multiple Choices ................................ 77
7.3.2 301 Moved Permanently ............................... 77
7.3.3 302 Moved Temporarily ............................... 77
7.3.4 305 Use Proxy ....................................... 77
7.3.5 380 Alternative Service ............................. 78
7.4 Request Failure 4xx ................................. 78
7.4.1 400 Bad Request ..................................... 78
7.4.2 401 Unauthorized .................................... 78
7.4.3 402 Payment Required ................................ 78
7.4.4 403 Forbidden ....................................... 78
7.4.5 404 Not Found ....................................... 78
7.4.6 405 Method Not Allowed .............................. 78
7.4.7 406 Not Acceptable .................................. 79
7.4.8 407 Proxy Authentication Required ................... 79
7.4.9 408 Request Timeout ................................. 79
7.4.10 409 Conflict ........................................ 79
7.4.11 410 Gone ............................................ 79
7.4.12 411 Length Required ................................. 79
7.4.13 413 Request Entity Too Large ........................ 80
7.4.14 414 Request-URI Too Long ............................ 80
7.4.15 415 Unsupported Media Type .......................... 80
7.4.16 420 Bad Extension ................................... 80
7.4.17 480 Temporarily Unavailable ......................... 80
7.4.18 481 Call Leg/Transaction Does Not Exist ............. 81
7.4.19 482 Loop Detected ................................... 81
7.4.20 483 Too Many Hops ................................... 81
7.4.21 484 Address Incomplete .............................. 81
7.4.22 485 Ambiguous ....................................... 81
7.4.23 486 Busy Here ....................................... 82
7.5 Server Failure 5xx .................................. 82
7.5.1 500 Server Internal Error ........................... 82
7.5.2 501 Not Implemented ................................. 82
7.5.3 502 Bad Gateway ..................................... 82
7.5.4 503 Service Unavailable ............................. 83
7.5.5 504 Gateway Time-out ................................ 83
7.5.6 505 Version Not Supported ........................... 83
7.6 Global Failures 6xx ................................. 83
7.6.1 600 Busy Everywhere ................................. 83
7.6.2 603 Decline ......................................... 84
7.6.3 604 Does Not Exist Anywhere ......................... 84
7.6.4 606 Not Acceptable .................................. 84
8 SIP Message Body .................................... 84
8.1 Body Inclusion ...................................... 84
8.2 Message Body Type ................................... 85
8.3 Message Body Length ................................. 85
9 Compact Form ........................................ 85
10 Behavior of SIP Clients and Servers ................. 86
10.1 General Remarks ..................................... 86
10.1.1 Requests ............................................ 86
10.1.2 Responses ........................................... 87
10.2 Source Addresses, Destination Addresses and
Connections ......................................... 88
10.2.1 Unicast UDP ......................................... 88
10.2.2 Multicast UDP ....................................... 88
10.3 TCP ................................................. 89
10.4 Reliability for BYE, CANCEL, OPTIONS, REGISTER
Requests ............................................ 90
10.4.1 UDP ................................................. 90
10.4.2 TCP ................................................. 91
10.5 Reliability for INVITE Requests ..................... 91
10.5.1 UDP ................................................. 92
10.5.2 TCP ................................................. 95
10.6 Reliability for ACK Requests ........................ 95
10.7 ICMP Handling ....................................... 95
11 Behavior of SIP User Agents ......................... 95
11.1 Caller Issues Initial INVITE Request ................ 96
11.2 Callee Issues Response .............................. 96
11.3 Caller Receives Response to Initial Request ......... 96
11.4 Caller or Callee Generate Subsequent Requests ....... 97
11.5 Receiving Subsequent Requests ....................... 97
12 Behavior of SIP Proxy and Redirect Servers .......... 97
12.1 Redirect Server ..................................... 97
12.2 User Agent Server ................................... 98
12.3 Proxy Server ........................................ 98
12.3.1 Proxying Requests ................................... 98
12.3.2 Proxying Responses .................................. 99
12.3.3 Stateless Proxy: Proxying Responses ................. 99
12.3.4 Stateful Proxy: Receiving Requests .................. 99
12.3.5 Stateful Proxy: Receiving ACKs ...................... 99
12.3.6 Stateful Proxy: Receiving Responses ................. 100
12.3.7 Stateless, Non-Forking Proxy ........................ 100
12.4 Forking Proxy ....................................... 100
13 Security Considerations ............................. 104
13.1 Confidentiality and Privacy: Encryption ............. 104
13.1.1 End-to-End Encryption ............................... 104
13.1.2 Privacy of SIP Responses ............................ 107
13.1.3 Encryption by Proxies ............................... 108
13.1.4 Hop-by-Hop Encryption ............................... 108
13.1.5 Via field encryption ................................ 108
13.2 Message Integrity and Access Control:
Authentication ...................................... 109
13.2.1 Trusting responses .................................. 112
13.3 Callee Privacy ...................................... 113
13.4 Known Security Problems ............................. 113
14 SIP Authentication using HTTP Basic and Digest
Schemes ............................................. 113
14.1 Framework ........................................... 113
14.2 Basic Authentication ................................ 114
14.3 Digest Authentication ............................... 114
14.4 Proxy-Authentication ................................ 115
15 SIP Security Using PGP .............................. 115
15.1 PGP Authentication Scheme ........................... 115
15.1.1 The WWW-Authenticate Response Header ................ 116
15.1.2 The Authorization Request Header .................... 117
15.2 PGP Encryption Scheme ............................... 118
15.3 Response-Key Header Field for PGP ................... 119
16 Examples ............................................ 119
16.1 Registration ........................................ 119
16.2 Invitation to a Multicast Conference ................ 121
16.2.1 Request ............................................. 121
16.2.2 Response ............................................ 122
16.3 Two-party Call ...................................... 123
16.4 Terminating a Call .................................. 125
16.5 Forking Proxy ....................................... 126
16.6 Redirects ........................................... 130
16.7 Negotiation ......................................... 131
16.8 OPTIONS Request ..................................... 132
A Minimal Implementation .............................. 134
A.1 Client .............................................. 134
A.2 Server .............................................. 135
A.3 Header Processing ................................... 135
B Usage of the Session Description Protocol (SDP)...... 136
B.1 Configuring Media Streams ........................... 136
B.2 Setting SDP Values for Unicast ...................... 138
B.3 Multicast Operation ................................. 139
B.4 Delayed Media Streams ............................... 139
B.5 Putting Media Streams on Hold ....................... 139
B.6 Subject and SDP "s=" Line ........................... 140
B.7 The SDP "o=" Line ................................... 140
C Summary of Augmented BNF ............................ 141
C.1 Basic Rules ......................................... 143
D Using SRV DNS Records ............................... 146
E IANA Considerations ................................. 148
F Acknowledgments ..................................... 149
G Authors' Addresses .................................. 149
H Bibliography ........................................ 150
I Full Copyright Statement ............................ 153
1 Introduction
1.1 Overview of SIP Functionality
The Session Initiation Protocol (SIP) is an application-layer control
protocol that can establish, modify and terminate multimedia sessions
or calls. These multimedia sessions include multimedia conferences,
distance learning, Internet telephony and similar applications. SIP
can invite both persons and "robots", such as a media storage
service. SIP can invite parties to both unicast and multicast
sessions; the initiator does not necessarily have to be a member of
the session to which it is inviting. Media and participants can be
added to an existing session.
SIP can be used to initiate sessions as well as invite members to
sessions that have been advertised and established by other means.
Sessions can be advertised using multicast protocols such as SAP,
electronic mail, news groups, web pages or directories (LDAP), among
others.
SIP transparently supports name mapping and redirection services,
allowing the implementation of ISDN and Intelligent Network telephony
subscriber services. These facilities also enable personal mobility.
In the parlance of telecommunications intelligent network services,
this is defined as: "Personal mobility is the ability of end users to
originate and receive calls and access subscribed telecommunication
services on any terminal in any location, and the ability of the
network to identify end users as they move. Personal mobility is
based on the use of a unique personal identity (i.e., personal
number)." [1]. Personal mobility complements terminal mobility, i.e.,
the ability to maintain communications when moving a single end
system from one subnet to another.
SIP supports five facets of establishing and terminating multimedia
communications:
User location: determination of the end system to be used for
communication;
User capabilities: determination of the media and media parameters to
be used;
User availability: determination of the willingness of the called
party to engage in communications;
Call setup: "ringing", establishment of call parameters at both
called and calling party;
Call handling: including transfer and termination of calls.
SIP can also initiate multi-party calls using a multipoint control
unit (MCU) or fully-meshed interconnection instead of multicast.
Internet telephony gateways that connect Public Switched Telephone
Network (PSTN) parties can also use SIP to set up calls between them.
SIP is designed as part of the overall IETF multimedia data and
control architecture currently incorporating protocols such as RSVP
(RFC 2205 [2]) for reserving network resources, the real-time
transport protocol (RTP) (RFC 1889 [3]) for transporting real-time
data and providing QOS feedback, the real-time streaming protocol
(RTSP) (RFC 2326 [4]) for controlling delivery of streaming media,
the session announcement protocol (SAP) [5] for advertising
multimedia sessions via multicast and the session description
protocol (SDP) (RFC 2327 [6]) for describing multimedia sessions.
However, the functionality and operation of SIP does not depend on
any of these protocols.
SIP can also be used in conjunction with other call setup and
signaling protocols. In that mode, an end system uses SIP exchanges
to determine the appropriate end system address and protocol from a
given address that is protocol-independent. For example, SIP could be
used to determine that the party can be reached via H.323 [7], obtain
the H.245 [8] gateway and user address and then use H.225.0 [9] to
establish the call.
In another example, SIP might be used to determine that the callee is
reachable via the PSTN and indicate the phone number to be called,
possibly suggesting an Internet-to-PSTN gateway to be used.
SIP does not offer conference control services such as floor control
or voting and does not prescribe how a conference is to be managed,
but SIP can be used to introduce conference control protocols. SIP
does not allocate multicast addresses.
SIP can invite users to sessions with and without resource
reservation. SIP does not reserve resources, but can convey to the
invited system the information necessary to do this.
1.2 Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [10]
and indicate requirement levels for compliant SIP implementations.
1.3 Definitions
This specification uses a number of terms to refer to the roles
played by participants in SIP communications. The definitions of
client, server and proxy are similar to those used by the Hypertext
Transport Protocol (HTTP) (RFC 2068 [11]). The terms and generic
syntax of URI and URL are defined in RFC 2396 [12]. The following
terms have special significance for SIP.
Call: A call consists of all participants in a conference invited by
a common source. A SIP call is identified by a globally unique
call-id (Section 6.12). Thus, if a user is, for example, invited
to the same multicast session by several people, each of these
invitations will be a unique call. A point-to-point Internet
telephony conversation maps into a single SIP call. In a
multiparty conference unit (MCU) based call-in conference, each
participant uses a separate call to invite himself to the MCU.
Call leg: A call leg is identified by the combination of Call-ID, To
and From.
Client: An application program that sends SIP requests. Clients may
or may not interact directly with a human user. User agents and
proxies contain clients (and servers).
Conference: A multimedia session (see below), identified by a common
session description. A conference can have zero or more members
and includes the cases of a multicast conference, a full-mesh
conference and a two-party "telephone call", as well as
combinations of these. Any number of calls can be used to
create a conference.
Downstream: Requests sent in the direction from the caller to the
callee (i.e., 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.
Initiator, calling party, caller: The party initiating a conference
invitation. Note that the calling party does not have to be the
same as the one creating the conference.
Invitation: A request sent to a user (or service) requesting
participation in a session. A successful SIP invitation consists
of two transactions: an INVITE request followed by an ACK
request.
Invitee, invited user, called party, callee: The person or service
that the calling party is trying to invite to a conference.
Isomorphic request or response: Two requests or responses are defined
to be isomorphic for the purposes of this document if they have
the same values for the Call-ID, To, From and CSeq header
fields. In addition, isomorphic requests have to have the same
Request-URI.
Location server: See location service.
Location service: A location service is used by a SIP redirect or
proxy server to obtain information about a callee's possible
location(s). Location services are offered by location servers.
Location servers MAY be co-located with a SIP server, but the
manner in which a SIP server requests location services is
beyond the scope of this document.
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 program that acts as both a
server and a client for the purpose of making requests on behalf
of other clients. Requests are serviced internally or by passing
them on, possibly after translation, to other servers. A proxy
interprets, and, if necessary, rewrites a request message before
forwarding it.
Redirect server: A redirect server is a server that accepts a SIP
request, maps the address into zero or more new addresses and
returns these addresses to the client. Unlike a proxy server ,
it does not initiate its own SIP request. Unlike a user agent
server , it does not accept calls.
Registrar: A registrar is a server that accepts REGISTER requests. A
registrar is typically co-located with a proxy or redirect
server and MAY offer location services.
Ringback: Ringback is the signaling tone produced by the calling
client's application indicating that a called party is being
alerted (ringing).
Server: A server is an application program that accepts requests in
order to service requests and sends back responses to those
requests. Servers are either proxy, redirect or user agent
servers or registrars.
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 [6]) (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 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
sent from the server to the client. A transaction is identified
by the CSeq sequence number (Section 6.17) within a single call
leg. The ACK request has the same CSeq number as the
corresponding INVITE request, but comprises a transaction of its
own.
Upstream: Responses sent in the direction from the user agent server
to the user agent client.
URL-encoded: A character string encoded according to RFC 1738,
Section 2.2 [13].
User agent client (UAC), calling user agent: A user agent client is a
client application that initiates the SIP request.
User agent server (UAS), called user agent: A user agent server is a
server application that contacts the user when a SIP request is
received and that returns a response on behalf of the user. The
response accepts, rejects or redirects the request.
User agent (UA): An application which contains both a user agent
client and user agent server.
An application program MAY be capable of acting both as a client and
a server. For example, a typical multimedia conference control
application would act as a user agent client to initiate calls or to
invite others to conferences and as a user agent server to accept
invitations. The properties of the different SIP server types are
summarized in Table 1.
property redirect proxy user agent registrar
server server server
__________________________________________________________________
also acts as a SIP client no yes no no
returns 1xx status yes yes yes yes
returns 2xx status no yes yes yes
returns 3xx status yes yes yes yes
returns 4xx status yes yes yes yes
returns 5xx status yes yes yes yes
returns 6xx status no yes yes yes
inserts Via header no yes no no
accepts ACK yes yes yes no
Table 1: Properties of the different SIP server types
1.4 Overview of SIP Operation
This section explains the basic protocol functionality and operation.
Callers and callees are identified by SIP addresses, described in
Section 1.4.1. When making a SIP call, a caller first locates the
appropriate server (Section 1.4.2) and then sends a SIP request
(Section 1.4.3). The most common SIP operation is the invitation
(Section 1.4.4). Instead of directly reaching the intended callee, a
SIP request may be redirected or may trigger a chain of new SIP
requests by proxies (Section 1.4.5). Users can register their
location(s) with SIP servers (Section 4.2.6).
1.4.1 SIP Addressing
The "objects" addressed by SIP are users at hosts, identified by a
SIP URL. The SIP URL takes a form similar to a mailto or telnet URL,
i.e., user@host. The user part is a user name or a telephone number.
The host part is either a domain name or a numeric network address.
See section 2 for a detailed discussion of SIP URL's.
A user's SIP address can be obtained out-of-band, can be learned via
existing media agents, can be included in some mailers' message
headers, or can be recorded during previous invitation interactions.
In many cases, a user's SIP URL can be guessed from their email
address.
A SIP URL address can designate an individual (possibly located at
one of several end systems), the first available person from a group
of individuals or a whole group. The form of the address, for
example, sip:sales@example.com , is not sufficient, in general, to
determine the intent of the caller.
If a user or service chooses to be reachable at an address that is
guessable from the person's name and organizational affiliation, the
traditional method of ensuring privacy by having an unlisted "phone"
number is compromised. However, unlike traditional telephony, SIP
offers authentication and access control mechanisms and can avail
itself of lower-layer security mechanisms, so that client software
can reject unauthorized or undesired call attempts.
1.4.2 Locating a SIP Server
When a client wishes to send a request, the client either sends it to
a locally configured SIP proxy server (as in HTTP), independent of
the Request-URI, or sends it to the IP address and port corresponding
to the Request-URI.
For the latter case, the client must determine the protocol, port and
IP address of a server to which to send the request. A client SHOULD
follow the steps below to obtain this information, but MAY follow the
alternative, optional procedure defined in Appendix D. At each step,
unless stated otherwise, the client SHOULD try to contact a server at
the port number listed in the Request-URI. If no port number is
present in the Request-URI, the client uses port 5060. If the
Request-URI specifies a protocol (TCP or UDP), the client contacts
the server using that protocol. If no protocol is specified, the
client tries UDP (if UDP is supported). If the attempt fails, or if
the client doesn't support UDP but supports TCP, it then tries TCP.
A client SHOULD be able to interpret explicit network notifications
(such as ICMP messages) which indicate that a server is not
reachable, rather than relying solely on timeouts. (For socket-based
programs: For TCP, connect() returns ECONNREFUSED if the client could
not connect to a server at that address. For UDP, the socket needs to
be bound to the destination address using connect() rather than
sendto() or similar so that a second write() fails with ECONNREFUSED
if there is no server listening) If the client finds the server is
not reachable at a particular address, it SHOULD behave as if it had
received a 400-class error response to that request.
The client tries to find one or more addresses for the SIP server by
querying DNS. The procedure is as follows:
1. If the host portion of the Request-URI is an IP address,
the client contacts the server at the given address.
Otherwise, the client proceeds to the next step.
2. The client queries the DNS server for address records for
the host portion of the Request-URI. If the DNS server
returns no address records, the client stops, as it has
been unable to locate a server. By address record, we mean
A RR's, AAAA RR's, or other similar address records, chosen
according to the client's network protocol capabilities.
There are no mandatory rules on how to select a host name
for a SIP server. Users are encouraged to name their SIP
servers using the sip.domainname (i.e., sip.example.com)
convention, as specified in RFC 2219 [16]. Users may only
know an email address instead of a full SIP URL for a
callee, however. In that case, implementations may be able
to increase the likelihood of reaching a SIP server for
that domain by constructing a SIP URL from that email
address by prefixing the host name with "sip.". In the
future, this mechanism is likely to become unnecessary as
better DNS techniques, such as the one in Appendix D,
become widely available.
A client MAY cache a successful DNS query result. A successful query
is one which contained records in the answer, and a server was
contacted at one of the addresses from the answer. When the client
wishes to send a request to the same host, it MUST start the search
as if it had just received this answer from the name server. The
client MUST follow the procedures in RFC1035 [15] regarding DNS cache
invalidation when the DNS time-to-live expires.
1.4.3 SIP Transaction
Once the host part has been resolved to a SIP server, the client
sends one or more SIP requests to that server and receives one or
more responses from the server. A request (and its retransmissions)
together with the responses triggered by that request make up a SIP
transaction. All responses to a request contain the same values in
the Call-ID, CSeq, To, and From fields (with the possible addition of
a tag in the To field (section 6.37)). This allows responses to be
matched with requests. The ACK request following an INVITE is not
part of the transaction since it may traverse a different set of
hosts.
If TCP is used, request and responses within a single SIP transaction
are carried over the same TCP connection (see Section 10). Several
SIP requests from the same client to the same server MAY use the same
TCP connection or MAY use a new connection for each request.
If the client sent the request via unicast UDP, the response is sent
to the address contained in the next Via header field (Section 6.40)
of the response. If the request is sent via multicast UDP, the
response is directed to the same multicast address and destination
port. For UDP, reliability is achieved using retransmission (Section
10).
The SIP message format and operation is independent of the transport
protocol.
1.4.4 SIP Invitation
A successful SIP invitation consists of two requests, INVITE followed
by ACK. The INVITE (Section 4.2.1) request asks the callee to join a
particular conference or establish a two-party conversation. After
the callee has agreed to participate in the call, the caller confirms
that it has received that response by sending an ACK (Section 4.2.2)
request. If the caller no longer wants to participate in the call, it
sends a BYE request instead of an ACK.
The INVITE request typically contains a session description, for
example written in SDP (RFC 2327 [6]) format, that provides the
called party with enough information to join the session. For
multicast sessions, the session description enumerates the media
types and formats that are allowed to be distributed to that session.
For a unicast session, the session description enumerates the media
types and formats that the caller is willing to use and where it
wishes the media data to be sent. In either case, if the callee
wishes to accept the call, it responds to the invitation by returning
a similar description listing the media it wishes to use. For a
multicast session, the callee SHOULD only return a session
description if it is unable to receive the media indicated in the
caller's description or wants to receive data via unicast.
The protocol exchanges for the INVITE method are shown in Fig. 1 for
a proxy server and in Fig. 2 for a redirect server. (Note that the
messages shown in the figures have been abbreviated slightly.) In
Fig. 1, the proxy server accepts the INVITE request (step 1),
contacts the location service with all or parts of the address (step
2) and obtains a more precise location (step 3). The proxy server
then issues a SIP INVITE request to the address(es) returned by the
location service (step 4). The user agent server alerts the user
(step 5) and returns a success indication to the proxy server (step
6). The proxy server then returns the success result to the original
caller (step 7). The receipt of this message is confirmed by the
caller using an ACK request, which is forwarded to the callee (steps
8 and 9). Note that an ACK can also be sent directly to the callee,
bypassing the proxy. All requests and responses have the same Call-
ID.
+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@lab :
: 2| 3| :
: | | :
: henning | :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | \/ 4: INVITE 5: ring :
: cz@cs.tu-berlin.de ========================>(~~~~~~)=========>(~~~~~~) :
: <........................( )<.........( ) :
: : 7: 200 OK : ( )6: 200 OK ( ) :
: : : ( work ) ( lab ) :
: : 8: ACK : ( )9: ACK ( ) :
: ========================>(~~~~~~)=========>(~~~~~~) :
+.....................+ +...............................+
====> SIP request
....> SIP response
^
| non-SIP protocols
|
Figure 1: Example of SIP proxy server
The redirect server shown in Fig. 2 accepts the INVITE request (step
1), contacts the location service as before (steps 2 and 3) and,
instead of contacting the newly found address itself, returns the
address to the caller (step 4), which is then acknowledged via an ACK
request (step 5). The caller issues a new request, with the same
call-ID but a higher CSeq, to the address returned by the first
server (step 6). In the example, the call succeeds (step 7). The
caller and callee complete the handshake with an ACK (step 8).
The next section discusses what happens if the location service
returns more than one possible alternative.
1.4.5 Locating a User
A callee may move between a number of different end systems over
time. These locations can be dynamically registered with the SIP
server (Sections 1.4.7, 4.2.6). A location server MAY also use one or
more other protocols, such as finger (RFC 1288 [17]), rwhois (RFC
2167 [18]), LDAP (RFC 1777 [19]), multicast-based protocols [20] or
operating-system dependent mechanisms to actively determine the end
system where a user might be reachable. A location server MAY return
several locations because the user is logged in at several hosts
simultaneously or because the location server has (temporarily)
inaccurate information. The SIP server combines the results to yield
a list of a zero or more locations.
The action taken on receiving a list of locations varies with the
type of SIP server. A SIP redirect server returns the list to the
client as Contact headers (Section 6.13). A SIP proxy server can
sequentially or in parallel try the addresses until the call is
successful (2xx response) or the callee has declined the call (6xx
response). With sequential attempts, a proxy server can implement an
"anycast" service.
If a proxy server forwards a SIP request, it MUST add itself to the
beginning of the list of forwarders noted in the Via (Section 6.40)
headers. The Via trace ensures that replies can take the same path
back, ensuring correct operation through compliant firewalls and
avoiding request loops. On the response path, each host MUST remove
its Via, so that routing internal information is hidden from the
callee and outside networks. A proxy server MUST check that it does
not generate a request to a host listed in the Via sent-by, via-
received or via-maddr parameters (Section 6.40). (Note: If a host has
several names or network addresses, this does not always work. Thus,
each host also checks if it is part of the Via list.)
A SIP invitation may traverse more than one SIP proxy server. If one
of these "forks" the request, i.e., issues more than one request in
response to receiving the invitation request, it is possible that a
client is reached, independently, by more than one copy of the
invitation request. Each of these copies bears the same Call-ID. The
user agent MUST return the same status response returned in the first
response. Duplicate requests are not an error.
1.4.6 Changing an Existing Session
In some circumstances, it is desirable to change the parameters of an
existing session. This is done by re-issuing the INVITE, using the
same Call-ID, but a new or different body or header fields to convey
the new information. This re INVITE MUST have a higher CSeq than any
previous request from the client to the server.
For example, two parties may have been conversing and then want to
add a third party, switching to multicast for efficiency. One of the
participants invites the third party with the new multicast address
and simultaneously sends an INVITE to the second party, with the new
multicast session description, but with the old call identifier.
1.4.7 Registration Services
The REGISTER request allows a client to let a proxy or redirect
server know at which address(es) it can be reached. A client MAY also
use it to install call handling features at the server.
1.5 Protocol Properties
1.5.1 Minimal State
A single conference session or call involves one or more SIP
request-response transactions. Proxy servers do not have to keep
state for a particular call, however, they MAY maintain state for a
single SIP transaction, as discussed in Section 12. For efficiency, a
server MAY cache the results of location service requests.
1.5.2 Lower-Layer-Protocol Neutral
SIP makes minimal assumptions about the underlying transport and
network-layer protocols. The lower-layer can provide either a packet
or a byte stream service, with reliable or unreliable service.
In an Internet context, SIP is able to utilize both UDP and TCP as
transport protocols, among others. UDP allows the application to more
carefully control the timing of messages and their retransmission, to
perform parallel searches without requiring TCP connection state for
each outstanding request, and to use multicast. Routers can more
readily snoop SIP UDP packets. TCP allows easier passage through
existing firewalls.
+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@lab :
: 2| 3| :
: | | :
: henning| :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | \/ :
: cz@cs.tu-berlin.de =======================>(~~~~~~) :
: | ^ | <.......................( ) :
: | . | : 4: 302 Moved : ( ) :
: | . | : hgs@lab : ( work ) :
: | . | : : ( ) :
: | . | : 5: ACK : ( ) :
: | . | =======================>(~~~~~~) :
: | . | : : :
+.......|...|.........+ : :
| . | : :
| . | : :
| . | : :
| . | : :
| . | 6: INVITE hgs@lab.cs.columbia.edu (~~~~~~) :
| . ==================================================> ( ) :
| ..................................................... ( ) :
| 7: 200 OK : ( lab ) :
| : ( ) :
| 8: ACK : ( ) :
======================================================> (~~~~~~) :
+...............................+
====> SIP request
....> SIP response
^
| non-SIP protocols
|
Figure 2: Example of SIP redirect server
When TCP is used, SIP can use one or more connections to attempt to
contact a user or to modify parameters of an existing conference.
Different SIP requests for the same SIP call MAY use different TCP
connections or a single persistent connection, as appropriate.
For concreteness, this document will only refer to Internet
protocols. However, SIP MAY also be used directly with protocols
such as ATM AAL5, IPX, frame relay or X.25. The necessary naming
conventions are beyond the scope of this document. User agents SHOULD
implement both UDP and TCP transport. Proxy, registrar, and redirect
servers MUST implement both UDP and TCP transport.
1.5.3 Text-Based
SIP is text-based, using ISO 10646 in UTF-8 encoding throughout. This
allows easy implementation in languages such as Java, Tcl and Perl,
allows easy debugging, and most importantly, makes SIP flexible and
extensible. As SIP is used for initiating multimedia conferences
rather than delivering media data, it is believed that the additional
overhead of using a text-based protocol is not significant.
2 SIP Uniform Resource Locators
SIP URLs are used within SIP messages to indicate the originator
(From), current destination (Request-URI) and final recipient (To) of
a SIP request, and to specify redirection addresses (Contact). A SIP
URL can also be embedded in web pages or other hyperlinks to indicate
that a particular user or service can be called via SIP. When used as
a hyperlink, the SIP URL indicates the use of the INVITE method.
The SIP URL scheme is defined to allow setting SIP request-header
fields and the SIP message-body.
This corresponds to the use of mailto: URLs. It makes it
possible, for example, to specify the subject, urgency or
media types of calls initiated through a web page or as
part of an email message.
A SIP URL follows the guidelines of RFC 2396 [12] and has the syntax
shown in Fig. 3. The syntax is described using Augmented Backus-Naur
Form (See Section C). Note that reserved characters have to be
escaped and that the "set of characters 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" [12].
SIP-URL = "sip:" [ userinfo "@" ] hostport
url-parameters [ headers ]
userinfo = user [ ":" password ]
user = *( unreserved | escaped
| "&" | "=" | "+" | "$" | "," )
password = *( unreserved | escaped
| "&" | "=" | "+" | "$" | "," )
hostport = host [ ":" port ]
host = hostname | IPv4address
hostname = *( domainlabel "." ) toplabel [ "." ]
domainlabel = alphanum | alphanum *( alphanum | "-" ) alphanum
toplabel = alpha | alpha *( alphanum | "-" ) alphanum
IPv4address = 1*digit "." 1*digit "." 1*digit "." 1*digit
port = *digit
url-parameters = *( ";" url-parameter )
url-parameter = transport-param | user-param | method-param
| ttl-param | maddr-param | other-param
transport-param = "transport=" ( "udp" | "tcp" )
ttl-param = "ttl=" ttl
ttl = 1*3DIGIT ; 0 to 255
maddr-param = "maddr=" host
user-param = "user=" ( "phone" | "ip" )
method-param = "method=" Method
tag-param = "tag=" UUID
UUID = 1*( hex | "-" )
other-param = ( token | ( token "=" ( token | quoted-string )))
headers = "?" header *( "&" header )
header = hname "=" hvalue
hname = 1*uric
hvalue = *uric
uric = reserved | unreserved | escaped
reserved = ";" | "/" | "?" | ":" | "@" | "&" | "=" | "+" |
"$" | ","
digits = 1*DIGIT
Figure 3: SIP URL syntax
The URI character classes referenced above are described in Appendix
C.
The components of the SIP URI have the following meanings.
telephone-subscriber = global-phone-number | local-phone-number
global-phone-number = "+" 1*phonedigit [isdn-subaddress]
[post-dial]
local-phone-number = 1*(phonedigit | dtmf-digit |
pause-character) [isdn-subaddress]
[post-dial]
isdn-subaddress = ";isub=" 1*phonedigit
post-dial = ";postd=" 1*(phonedigit | dtmf-digit
| pause-character)
phonedigit = DIGIT | visual-separator
visual-separator = "-" | "."
pause-character = one-second-pause | wait-for-dial-tone
one-second-pause = "p"
wait-for-dial-tone = "w"
dtmf-digit = "*" | "#" | "A" | "B" | "C" | "D"
Figure 4: SIP URL syntax; telephone subscriber
user: If the host is an Internet telephony gateway, the user field
MAY also encode a telephone number using the notation of
telephone-subscriber (Fig. 4). The telephone number is a special
case of a user name and cannot be distinguished by a BNF. Thus,
a URL parameter, user, is added to distinguish telephone numbers
from user names. The phone identifier is to be used when
connecting to a telephony gateway. Even without this parameter,
recipients of SIP URLs MAY interpret the pre-@ part as a phone
number if local restrictions on the name space for user name
allow it.
password: The SIP scheme MAY use the format "user:password" in the
userinfo field. The use of passwords in the userinfo 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.
host: The mailto: URL and RFC 822 email addresses require that
numeric host addresses ("host numbers") are enclosed in square
brackets (presumably, since host names might be numeric), while
host numbers without brackets are used for all other URLs. The
SIP URL requires the latter form, without brackets.
The issue of IPv6 literal addresses in URLs is being looked at
elsewhere in the IETF. SIP implementers are advised to keep up to
date on that activity.
port: The port number to send a request to. If not present, the
procedures outlined in Section 1.4.2 are used to determine the
port number to send a request to.
URL parameters: SIP URLs can define specific parameters of the
request. URL parameters are added after the host component and
are separated by semi-colons. The transport parameter determines
the the transport mechanism (UDP or TCP). UDP is to be assumed
when no explicit transport parameter is included. The maddr
parameter provides the server address to be contacted for this
user, overriding the address supplied in the host field. This
address is typically a multicast address, but could also be the
address of a backup server. 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. The user parameter was described above. For example, to
specify to call j.doe@big.com using multicast to 239.255.255.1
with a ttl of 15, the following URL would be used:
sip:j.doe@big.com;maddr=239.255.255.1;ttl=15
The transport, maddr, and ttl parameters MUST NOT be used in the From
and To header fields and the Request-URI; they are ignored if
present.
Headers: Headers of the SIP request can be defined with the "?"
mechanism within a SIP URL. The special hname "body" indicates
that the associated hvalue is the message-body of the SIP INVITE
request. Headers MUST NOT be used in the From and To header
fields and the Request-URI; they are ignored if present. hname
and hvalue are encodings of a SIP header name and value,
respectively. All URL reserved characters in the header names
and values MUST be escaped.
Method: The method of the SIP request can be specified with the
method parameter. This parameter MUST NOT be used in the From
and To header fields and the Request-URI; they are ignored if
present.
Table 2 summarizes where the components of the SIP URL can be used
and what default values they assume if not present.
Examples of SIP URLs are:
default Req.-URI To From Contact external
user -- x x x x x
password -- x x x x
host mandatory x x x x x
port 5060 x x x x x
user-param ip x x x x x
method INVITE x x
maddr-param -- x x
ttl-param 1 x x
transp.-param -- x x
headers -- x x
Table 2: Use and default values of URL components for SIP headers,
Request-URI and references
sip:j.doe@big.com
sip:j.doe:secret@big.com;transport=tcp
sip:j.doe@big.com?subject=project
sip:+1-212-555-1212:1234@gateway.com;user=phone
sip:1212@gateway.com
sip:alice@10.1.2.3
sip:alice@example.com
sip:alice%40example.com@gateway.com
sip:alice@registrar.com;method=REGISTER
Within a SIP message, URLs are used to indicate the source and
intended destination of a request, redirection addresses and the
current destination of a request. Normally all these fields will
contain SIP URLs.
SIP URLs are case-insensitive, so that for example the two URLs
sip:j.doe@example.com and SIP:J.Doe@Example.com are equivalent. All
URL parameters are included when comparing SIP URLs for equality.
SIP header fields MAY contain non-SIP URLs. As an example, if a call
from a telephone is relayed to the Internet via SIP, the SIP From
header field might contain a phone URL.
3 SIP Message Overview
SIP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2279 [21]). Senders MUST terminate lines with a
CRLF, but receivers MUST also interpret CR and LF by themselves as
line terminators.
Except for the above difference in character sets, much of the
message syntax is and header fields are 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 2068 [11]).
In addition, we describe SIP in both prose and an augmented Backus-
Naur form (ABNF). See section C for an overview of ABNF.
Note, however, that SIP is not an extension of HTTP.
Unlike HTTP, SIP MAY use UDP. When sent over TCP or UDP, multiple SIP
transactions can be carried in a single TCP connection or UDP
datagram. UDP datagrams, including all headers, SHOULD NOT be larger
than the path maximum transmission unit (MTU) if the MTU is known, or
1500 bytes if the MTU is unknown.
The 1500 bytes accommodates encapsulation within the
"typical" ethernet MTU without IP fragmentation. Recent
studies [22] indicate that an MTU of 1500 bytes is a
reasonable assumption. The next lower common MTU values are
1006 bytes for SLIP and 296 for low-delay PPP (RFC 1191
[23]). Thus, another reasonable value would be a message
size of 950 bytes, to accommodate packet headers within the
SLIP MTU without fragmentation.
A SIP message is either a request from a client to a server, or a
response from a server to a client.
SIP-message = Request | Response
Both Request (section 4) and Response (section 5) messages use the
generic-message format of RFC 822 [24] for transferring entities (the
body of the message). Both types of messages consist of a start-line,
one or more header fields (also known as "headers"), an empty line
(i.e., a line with nothing preceding the carriage-return line-feed
(CRLF)) indicating the end of the header fields, and an optional
message-body. To avoid confusion with similar-named headers in HTTP,
we refer to the headers describing the message body as entity
headers. These components are described in detail in the upcoming
sections.
generic-message = start-line
*message-header
CRLF
[ message-body ]
start-line = Request-Line | ;Section 4.1
Status-Line ;Section 5.1
message-header = ( general-header
| request-header
| response-header
| entity-header )
In the interest of robustness, any leading empty line(s) MUST be
ignored. In other words, if the Request or Response message begins
with one or more CRLF, CR, or LFs, these characters MUST be ignored.
4 Request
The Request message format is shown below:
Request = Request-Line ; Section 4.1
*( general-header
| request-header
| entity-header )
CRLF
[ message-body ] ; Section 8
4.1 Request-Line
The Request-Line begins with a method token, followed by the
Request-URI and the protocol version, and ending with CRLF. The
elements are separated by SP characters. No CR or LF are allowed
except in the final CRLF sequence.
Request-Line = Method SP Request-URI SP SIP-Version CRLF
general-header = Accept ; Section 6.7
| Accept-Encoding ; Section 6.8
| Accept-Language ; Section 6.9
| Call-ID ; Section 6.12
| Contact ; Section 6.13
| CSeq ; Section 6.17
| Date ; Section 6.18
| Encryption ; Section 6.19
| Expires ; Section 6.20
| From ; Section 6.21
| Record-Route ; Section 6.29
| Timestamp ; Section 6.36
| To ; Section 6.37
| Via ; Section 6.40
entity-header = Content-Encoding ; Section 6.14
| Content-Length ; Section 6.15
| Content-Type ; Section 6.16
request-header = Authorization ; Section 6.11
| Contact ; Section 6.13
| Hide ; Section 6.22
| Max-Forwards ; Section 6.23
| Organization ; Section 6.24
| Priority ; Section 6.25
| Proxy-Authorization ; Section 6.27
| Proxy-Require ; Section 6.28
| Route ; Section 6.33
| Require ; Section 6.30
| Response-Key ; Section 6.31
| Subject ; Section 6.35
| User-Agent ; Section 6.39
response-header = Allow ; Section 6.10
| Proxy-Authenticate ; Section 6.26
| Retry-After ; Section 6.32
| Server ; Section 6.34
| Unsupported ; Section 6.38
| Warning ; Section 6.41
| WWW-Authenticate ; Section 6.42
Table 3: SIP headers
4.2 Methods
The methods are defined below. Methods that are not supported by a
proxy or redirect server are treated by that server as if they were
an OPTIONS method and forwarded accordingly. Methods that are not
supported by a user agent server or registrar cause a 501 (Not
Implemented) response to be returned (Section 7). As in HTTP, the
Method token is case-sensitive.
Method = "INVITE" | "ACK" | "OPTIONS" | "BYE"
| "CANCEL" | "REGISTER"
4.2.1 INVITE
The INVITE method indicates that the user or service is being invited
to participate in a session. The message body contains a description
of the session to which the callee is being invited. For two-party
calls, the caller indicates the type of media it is able to receive
and possibly the media it is willing to send as well as their
parameters such as network destination. A success response MUST
indicate in its message body which media the callee wishes to receive
and MAY indicate the media the callee is going to send.
Not all session description formats have the ability to
indicate sending media.
A server MAY automatically respond to an invitation for a conference
the user is already participating in, identified either by the SIP
Call-ID or a globally unique identifier within the session
description, with a 200 (OK) response.
If a user agent receives an INVITE request for an existing call leg
with a higher CSeq sequence number than any previous INVITE for the
same Call-ID, 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. It MUST also
inspect any other header fields for changes. If there is a change,
the user agent MUST update any internal state or information
generated as a result of that header. If the session description has
changed, the user agent server 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.)
This method MUST be supported by SIP proxy, redirect and user agent
servers as well as clients.
4.2.2 ACK
The ACK request confirms that the client has received a final
response to an INVITE request. (ACK is used only with INVITE
requests.) 2xx responses are acknowledged by client user agents, all
other final responses by the first proxy or client user agent to
receive the response. The Via is always initialized to the host that
originates the ACK request, i.e., the client user agent after a 2xx
response or the first proxy to receive a non-2xx final response. The
ACK request is forwarded as the corresponding INVITE request, based
on its Request-URI. See Section 10 for details.
The ACK request MAY contain a message body with the final session
description to be used by the callee. If the ACK message body is
empty, the callee uses the session description in the INVITE request.
A proxy server receiving an ACK request after having sent a 3xx, 4xx,
5xx, or 6xx response must make a determination about whether the ACK
is for it, or for some user agent or proxy server further downstream.
This determination is made by examining the tag in the To field. If
the tag in the ACK To header field matches the tag in the To header
field of the response, and the From, CSeq and Call-ID header fields
in the response match those in the ACK, the ACK is meant for the
proxy server. Otherwise, the ACK SHOULD be proxied downstream as any
other request.
It is possible for a user agent client or proxy server to
receive multiple 3xx, 4xx, 5xx, and 6xx responses to a
request along a single branch. This can happen under
various error conditions, typically when a forking proxy
transitions from stateful to stateless before receiving all
responses. The various responses will all be identical,
except for the tag in the To field, which is different for
each one. It can therefore be used as a means to
disambiguate them.
This method MUST be supported by SIP proxy, redirect and user agent
servers as well as clients.
4.2.3 OPTIONS
The server is being queried as to its capabilities. A server that
believes it can contact the user, such as a user agent where the user
is logged in and has been recently active, MAY respond to this
request with a capability set. A called user agent MAY return a
status reflecting how it would have responded to an invitation, e.g.,
600 (Busy). Such a server SHOULD return an Allow header field
indicating the methods that it supports. Proxy and redirect servers
simply forward the request without indicating their capabilities.
This method MUST be supported by SIP proxy, redirect and user agent
servers, registrars and clients.
4.2.4 BYE
The user agent client uses BYE to indicate to the server that it
wishes to release the call. A BYE request is forwarded like an INVITE
request and MAY be issued by either caller or callee. A party to a
call SHOULD issue a BYE request before releasing a call ("hanging
up"). A party receiving a BYE request MUST cease transmitting media
streams specifically directed at the party issuing the BYE request.
If the INVITE request contained a Contact header, the callee SHOULD
send a BYE request to that address rather than the From address.
This method MUST be supported by proxy servers and SHOULD be
supported by redirect and user agent SIP servers.
4.2.5 CANCEL
The CANCEL request cancels a pending request with the same Call-ID,
To, From and CSeq (sequence number only) header field values, but
does not affect a completed request. (A request is considered
completed if the server has returned a final status response.)
A user agent client or proxy client MAY issue a CANCEL request at any
time. A proxy, in particular, MAY choose to send a CANCEL to
destinations that have not yet returned a final response after it has
received a 2xx or 6xx response for one or more of the parallel-search
requests. A proxy that receives a CANCEL request forwards the request
to all destinations with pending requests.
The Call-ID, To, the numeric part of CSeq and From headers in the
CANCEL request are identical to those in the original request. This
allows a CANCEL request to be matched with the request it cancels.
However, to allow the client to distinguish responses to the CANCEL
from those to the original request, the CSeq Method component is set
to CANCEL. The Via header field is initialized to the proxy issuing
the CANCEL request. (Thus, responses to this CANCEL request only
reach the issuing proxy.)
Once a user agent server has received a CANCEL, it MUST NOT issue a
2xx response for the cancelled original request.
A redirect or user agent server receiving a CANCEL request responds
with a status of 200 (OK) if the transaction exists and a status of
481 (Transaction Does Not Exist) if not, but takes no further action.
In particular, any existing call is unaffected.
The BYE request cannot be used to cancel branches of a
parallel search, since several branches may, through
intermediate proxies, find the same user agent server and
then terminate the call. To terminate a call instead of
just pending searches, the UAC must use BYE instead of or
in addition to CANCEL. While CANCEL can terminate any
pending request other than ACK or CANCEL, it is typically
useful only for INVITE. 200 responses to INVITE and 200
responses to CANCEL are distinguished by the method in the
Cseq header field, so there is no ambiguity.
This method MUST be supported by proxy servers and SHOULD be
supported by all other SIP server types.
4.2.6 REGISTER
A client uses the REGISTER method to register the address listed in
the To header field with a SIP server.
A user agent MAY register with a local server on startup by sending a
REGISTER request to the well-known "all SIP servers" multicast
address "sip.mcast.net" (224.0.1.75). This request SHOULD be scoped
to ensure it is not forwarded beyond the boundaries of the
administrative system. This MAY be done with either TTL or
administrative scopes [25], depending on what is implemented in the
network. SIP user agents MAY listen to that address and use it to
become aware of the location of other local users [20]; however, they
do not respond to the request. A user agent MAY also be configured
with the address of a registrar server to which it sends a REGISTER
request upon startup.
Requests are processed in the order received. Clients SHOULD avoid
sending a new registration (as opposed to a retransmission) until
they have received the response from the server for the previous one.
Clients may register from different locations, by necessity
using different Call-ID values. Thus, the CSeq value cannot
be used to enforce ordering. Since registrations are
additive, ordering is less of a problem than if each
REGISTER request completely replaced all earlier ones.
The meaning of the REGISTER request-header fields is defined as
follows. We define "address-of-record" as the SIP address that the
registry knows the registrand, typically of the form "user@domain"
rather than "user@host". In third-party registration, the entity
issuing the request is different from the entity being registered.
To: The To header field contains the address-of-record whose
registration is to be created or updated.
From: The From header field contains the address-of-record of the
person responsible for the registration. For first-party
registration, it is identical to the To header field value.
Request-URI: The Request-URI names the destination of the
registration request, i.e., the domain of the registrar. The
user name MUST be empty. Generally, the domains in the Request-
URI and the To header field have the same value; however, it is
possible to register as a "visitor", while maintaining one's
name. For example, a traveler sip:alice@acme.com (To) might
register under the Request-URI sip:atlanta.hiayh.org , with the
former as the To header field and the latter as the Request-URI.
The REGISTER request is no longer forwarded once it has reached
the server whose authoritative domain is the one listed in the
Request-URI.
Call-ID: All registrations from a client SHOULD use the same Call-ID
header value, at least within the same reboot cycle.
Cseq: Registrations with the same Call-ID MUST have increasing CSeq
header values. However, the server does not reject out-of-order
requests.
Contact: The request MAY contain a Contact header field; future non-
REGISTER requests for the URI given in the To header field
SHOULD be directed to the address(es) given in the Contact
header.
If the request does not contain a Contact header, the registration
remains unchanged.
This is useful to obtain the current list of registrations
in the response. Registrations using SIP URIs that differ
in one or more of host, port, transport-param or maddr-
param (see Figure 3) from an existing registration are
added to the list of registrations. Other URI types are
compared according to the standard URI equivalency rules
for the URI schema. If the URIs are equivalent to that of
an existing registration, the new registration replaces the
old one if it has a higher q value or, for the same value
of q, if the ttl value is higher. All current registrations
MUST share the same action value. Registrations that have
a different action than current registrations for the same
user MUST be rejected with status of 409 (Conflict).
A proxy server ignores the q parameter when processing non-REGISTER
requests, while a redirect server simply returns that parameter in
its Contact response header field.
Having the proxy server interpret the q parameter is not
sufficient to guide proxy behavior, as it is not clear, for
example, how long it is supposed to wait between trying
addresses.
If the registration is changed while a user agent or proxy server
processes an invitation, the new information SHOULD be used.
This allows a service known as "directed pick-up". In the
telephone network, directed pickup permits a user at a
remote station who hears his own phone ringing to pick up
at that station, dial an access code, and be connected to
the calling user as if he had answered his own phone.
A server MAY choose any duration for the registration lifetime.
Registrations not refreshed after this amount of time SHOULD be
silently discarded. Responses to a registration SHOULD include an
Expires header (Section 6.20) or expires Contact parameters (Section
6.13), indicating the time at which the server will drop the
registration. If none is present, one hour is assumed. Clients MAY
request a registration lifetime by indicating the time in an Expires
header in the request. A server SHOULD NOT use a higher lifetime than
the one requested, but MAY use a lower one. A single address (if
host-independent) MAY be registered from several different clients.
A client cancels an existing registration by sending a REGISTER
request with an expiration time (Expires) of zero seconds for a
particular Contact or the wildcard Contact designated by a "*" for
all registrations. Registrations are matched based on the user, host,
port and maddr parameters.
The server SHOULD return the current list of registrations in the 200
response as Contact header fields.
It is particularly important that REGISTER requests are authenticated
since they allow to redirect future requests (see Section 13.2).
Beyond its use as a simple location service, this method is
needed if there are several SIP servers on a single host.
In that case, only one of the servers can use the default
port number.
Support of this method is RECOMMENDED.
4.3 Request-URI
The Request-URI is a SIP URL as described in Section 2 or a general
URI. It indicates the user or service to which this request is being
addressed. Unlike the To field, the Request-URI MAY be re-written by
proxies.
When used as a Request-URI, a SIP-URL MUST NOT contain the
transport-param, maddr-param, ttl-param, or headers elements. A
server that receives a SIP-URL with these elements removes them
before further processing.
Typically, the UAC sets the Request-URI and To to the same
SIP URL, presumed to remain unchanged over long time
periods. However, if the UAC has cached a more direct path
to the callee, e.g., from the Contact header field of a
response to a previous request, the To would still contain
the long-term, "public" address, while the Request-URI
would be set to the cached address.
Proxy and redirect servers MAY use the information in the Request-URI
and request header fields to handle the request and possibly rewrite
the Request-URI. For example, a request addressed to the generic
address sip:sales@acme.com is proxied to the particular person, e.g.,
sip:bob@ny.acme.com , with the To field remaining as
sip:sales@acme.com. At ny.acme.com , Bob then designates Alice as
the temporary substitute.
The host part of the Request-URI typically agrees with one of the
host names of the receiving server. If it does not, the server SHOULD
proxy the request to the address indicated or return a 404 (Not
Found) response if it is unwilling or unable to do so. For example,
the Request-URI and server host name can disagree in the case of a
firewall proxy that handles outgoing calls. This mode of operation is
similar to that of HTTP proxies.
If a SIP server receives a request with a URI indicating a scheme
other than SIP which that server does not understand, the server MUST
return a 400 (Bad Request) response. It MUST do this even if the To
header field contains a scheme it does understand. This is because
proxies are responsible for processing the Request-URI; the To field
is of end-to-end significance.
4.3.1 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".
4.4 Option Tags
Option tags are unique identifiers used to designate new options in
SIP. These tags are used in Require (Section 6.30) and Unsupported
(Section 6.38) fields.
Syntax:
option-tag = token
See Section C for a definition of token. The creator of a new SIP
option MUST either prefix the option with their reverse domain name
or register the new option with the Internet Assigned Numbers
Authority (IANA). For example, "com.foo.mynewfeature" is an apt name
for a feature whose inventor can be reached at "foo.com". Individual
organizations are then responsible for ensuring that option names
don't collide. Options registered with IANA have the prefix
"org.iana.sip.", options described in RFCs have the prefix
"org.ietf.rfc.N", where N is the RFC number. Option tags are case-
insensitive.
4.4.1 Registering New Option Tags with IANA
When registering a new SIP option, the following information MUST be
provided:
o Name and description of option. The name MAY be of any
length, but SHOULD be no more than twenty characters long. The
name MUST consist of alphanum (See Figure 3) characters only;
o Indication of who has change control over the option (for
example, IETF, ISO, ITU-T, other international standardization
bodies, a consortium or a particular company or group of
companies);
o A reference to a further description, if available, for
example (in order of preference) an RFC, a published paper, a
patent filing, a technical report, documented source code or a
computer manual;
o Contact information (postal and email address);
Registrations should be sent to iana@iana.org
This procedure has been borrowed from RTSP [4] and the RTP
AVP [26].
5 Response
After receiving and interpreting a request message, the recipient
responds with a SIP response message. The response message format is
shown below:
Response = Status-Line ; Section 5.1
*( general-header
| response-header
| entity-header )
CRLF
[ message-body ] ; Section 8
SIP's structure of responses is similar to [H6], but is defined
explicitly here.
5.1 Status-Line
The first line of a Response message is the Status-Line, consisting
of the protocol version (Section 4.3.1) followed by a numeric
Status-Code and its associated textual phrase, with each element
separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence.
Status-Line = SIP-version SP Status-Code SP Reason-Phrase CRLF
5.1.1 Status Codes and Reason Phrases
The Status-Code is a 3-digit integer result code that indicates the
outcome of the attempt to understand and satisfy the 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. The client is not
required to examine or display the Reason-Phrase.
Status-Code = Informational ;Fig. 5
| Success ;Fig. 5
| Redirection ;Fig. 6
| Client-Error ;Fig. 7
| Server-Error ;Fig. 8
| Global-Failure ;Fig. 9
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *<TEXT-UTF8, excluding CR, LF>
We provide an overview of the Status-Code below, and provide full
definitions in Section 7. The first digit of the Status-Code defines
the class of response. The last two digits do not have any
categorization role. SIP/2.0 allows 6 values for the first digit:
1xx: Informational -- 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.
Figures 5 through 9 present the individual values of the numeric
response codes, and an example set of corresponding reason phrases
for SIP/2.0. These reason phrases are only recommended; they may be
replaced by local equivalents without affecting the protocol. Note
that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
codes in the range starting at x80 to avoid conflicts with newly
defined HTTP response codes, and adds a new class, 6xx, of response
codes.
SIP response codes are extensible. SIP applications are not required
to understand the meaning of all registered response codes, though
such understanding is obviously desirable. However, applications MUST
understand the class of any response code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 response code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if a client
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. In
such cases, user agents SHOULD present to the user the message body
returned with the response, since that message body is likely to
include human-readable information which will explain the unusual
status.
Informational = "100" ; Trying
| "180" ; Ringing
| "181" ; Call Is Being Forwarded
| "182" ; Queued
Success = "200" ; OK
Figure 5: Informational and success status codes
Redirection = "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "303" ; See Other
| "305" ; Use Proxy
| "380" ; Alternative Service
Figure 6: Redirection status codes
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
| "409" ; Conflict
| "410" ; Gone
| "411" ; Length Required
| "413" ; Request Entity Too Large
| "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type
| "420" ; Bad Extension
| "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
Figure 7: Client error status codes
Server-Error = "500" ; Internal Server Error
| "501" ; Not Implemented
| "502" ; Bad Gateway
| "503" ; Service Unavailable
| "504" ; Gateway Time-out
| "505" ; SIP Version not supported
Figure 8: Server error status codes
6 Header Field Definitions
SIP header fields are similar to HTTP header fields in both syntax
and semantics. In particular, SIP header fields follow the syntax for
message-header as described in [H4.2]. The rules for extending header
fields over multiple lines, and use of multiple message-header fields
with the same field-name, described in [H4.2] also apply to SIP. The
Global-Failure | "600" ; Busy Everywhere
| "603" ; Decline
| "604" ; Does not exist anywhere
| "606" ; Not Acceptable
Figure 9: Global failure status codes
rules in [H4.2] regarding ordering of header fields apply to SIP,
with the exception of Via fields, see below, whose order matters.
Additionally, header fields which are hop-by-hop MUST appear before
any header fields which are end-to-end. Proxies SHOULD NOT reorder
header fields. Proxies add Via header fields and MAY add other hop-
by-hop header fields. They can modify certain header fields, such as
Max-Forwards (Section 6.23) and "fix up" the Via header fields with
"received" parameters as described in Section 6.40.1. Proxies MUST
NOT alter any fields that are authenticated (see Section 13.2).
The header fields required, optional and not applicable for each
method are listed in Table 4 and Table 5. The table uses "o" to
indicate optional, "m" mandatory and "-" for not applicable. A "*"
indicates that the header fields are needed only if message body is
not empty. See sections 6.15, 6.16 and 8 for details.
The "where" column describes the request and response types with
which the header field can be used. "R" refers to header fields that
can be used in requests (that is, request and general header fields).
"r" designates a response or general-header field as applicable to
all responses, while a list of numeric values indicates the status
codes with which the header field can be used. "g" and "e" designate
general (Section 6.1) and entity header (Section 6.2) fields,
respectively. If a header field is marked "c", it is copied from the
request to the response.
The "enc." column describes whether this message header field MAY be
encrypted end-to-end. A "n" designates fields that MUST NOT be
encrypted, while "c" designates fields that SHOULD be encrypted if
encryption is used.
The "e-e" column has a value of "e" for end-to-end and a value of "h"
for hop-by-hop header fields.
where enc. e-e ACK BYE CAN INV OPT REG
__________________________________________________________
Accept R e - - - o o o
Accept 415 e - - - o o o
Accept-Encoding R e - - - o o o
Accept-Encoding 415 e - - - o o o
Accept-Language R e - o o o o o
Accept-Language 415 e - o o o o o
Allow 200 e - - - - m -
Allow 405 e o o o o o o
Authorization R e o o o o o o
Call-ID gc n e m m m m m m
Contact R e o - - o o o
Contact 1xx e - - - o o -
Contact 2xx e - - - o o o
Contact 3xx e - o - o o o
Contact 485 e - o - o o o
Content-Encoding e e o - - o o o
Content-Length e e o - - o o o
Content-Type e e * - - * * *
CSeq gc n e m m m m m m
Date g e o o o o o o
Encryption g n e o o o o o o
Expires g e - - - o - o
From gc n e m m m m m m
Hide R n h o o o o o o
Max-Forwards R n e o o o o o o
Organization g c h - - - o o o
Table 4: Summary of header fields, A--O
Other header fields can be added as required; a server MUST ignore
header fields not defined in this specification that it does not
understand. A proxy MUST NOT remove or modify header fields not
defined in this specification that it does not understand. A compact
form of these header fields is also defined in Section 9 for use over
UDP when the request has to fit into a single packet and size is an
issue.
Table 6 in Appendix A lists those header fields that different client
and server types MUST be able to parse.
6.1 General Header Fields
General header fields apply to both request and response messages.
The "general-header" field names can be extended reliably only in
combination with a change in the protocol version. However, new or
where enc. e-e ACK BYE CAN INV OPT REG
___________________________________________________________________
Proxy-Authenticate 407 n h o o o o o o
Proxy-Authorization R n h o o o o o o
Proxy-Require R n h o o o o o o
Priority R c e - - - o - -
Require R e o o o o o o
Retry-After R c e - - - - - o
Retry-After 404,480,486 c e o o o o o o
503 c e o o o o o o
600,603 c e o o o o o o
Response-Key R c e - o o o o o
Record-Route R h o o o o o o
Record-Route 2xx h o o o o o o
Route R h o o o o o o
Server r c e o o o o o o
Subject R c e - - - o - -
Timestamp g e o o o o o o
To gc(1) n e m m m m m m
Unsupported 420 e o o o o o o
User-Agent g c e o o o o o o
Via gc(2) n e m m m m m m
Warning r e o o o o o o
WWW-Authenticate 401 c e o o o o o o
Table 5: Summary of header fields, P--Z; (1): copied with possible
addition of tag; (2): UAS removes first Via header field
experimental header fields MAY be given the semantics of general
header fields if all parties in the communication recognize them to
be "general-header" fields. Unrecognized header fields are treated as
"entity-header" fields.
6.2 Entity Header Fields
The "entity-header" fields define meta-information about the
message-body or, if no body is present, about the resource identified
by the request. The term "entity header" is an HTTP 1.1 term where
the response body can contain a transformed version of the message
body. The original message body is referred to as the "entity". We
retain the same terminology for header fields but usually refer to
the "message body" rather then the entity as the two are the same in
SIP.
6.3 Request Header Fields
The "request-header" fields allow the client to pass additional
information about the request, and about the client itself, to the
server. These fields act as request modifiers, with semantics
equivalent to the parameters of a programming language method
invocation.
The "request-header" field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of "request-
header" fields if all parties in the communication recognize them to
be request-header fields. Unrecognized header fields are treated as
"entity-header" fields.
6.4 Response Header Fields
The "response-header" fields allow the server to pass additional
information about the response which cannot be placed in the Status-
Line. These header fields give information about the server and about
further access to the resource identified by the Request-URI.
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of "response-
header" fields if all parties in the communication recognize them to
be "response-header" fields. Unrecognized header fields are treated
as "entity-header" fields.
6.5 End-to-end and Hop-by-hop Headers
End-to-end headers MUST be transmitted unmodified across all proxies,
while hop-by-hop headers MAY be modified or added by proxies.
6.6 Header Field Format
Header fields ("general-header", "request-header", "response-header",
and "entity-header") follow the same generic header format as that
given in Section 3.1 of RFC 822 [24]. Each header field consists of a
name followed by a colon (":") and the field value. Field names are
case-insensitive. The field value MAY be preceded by any amount of
leading white space (LWS), though a single space (SP) is preferred.
Header fields can be extended over multiple lines by preceding each
extra line with at least one SP or horizontal tab (HT). Applications
MUST follow HTTP "common form" when generating these constructs,
since there might exist some implementations that fail to accept
anything beyond the common forms.
message-header = field-name ":" [ field-value ] CRLF
field-name = token
field-value = *( field-content | LWS )
field-content = < the OCTETs making up the field-value
and consisting of either *TEXT-UTF8
or combinations of token,
separators, and quoted-string>
The relative order of header fields with different field names is not
significant. Multiple header fields with the same field-name may be
present in a message if and only if the entire field-value for that
header field is defined as a comma-separated list (i.e., #(values)).
It MUST be possible to combine the multiple header fields into one
"field-name: field-value" pair, without changing the semantics of the
message, by appending each subsequent field-value to the first, each
separated by a comma. The order in which header fields with the same
field-name are received is therefore significant to the
interpretation of the combined field value, and thus a proxy MUST NOT
change the order of these field values when a message is forwarded.
Field names are not case-sensitive, although their values may be.
6.7 Accept
The Accept header follows the syntax defined in [H14.1]. The
semantics are also identical, with the exception that if no Accept
header is present, the server SHOULD assume a default value of
application/sdp.
This request-header field is used only with the INVITE, OPTIONS and
REGISTER request methods to indicate what media types are acceptable
in the response.
Example:
Accept: application/sdp;level=1, application/x-private, text/html
6.8 Accept-Encoding
The Accept-Encoding request-header field is similar to Accept, but
restricts the content-codings [H3.4.1] that are acceptable in the
response. See [H14.3]. The syntax of this header is defined in
[H14.3]. The semantics in SIP are identical to those defined in
[H14.3].
6.9 Accept-Language
The Accept-Language header follows the syntax defined in [H14.4]. The
rules for ordering the languages based on the q parameter apply to
SIP as well. When used in SIP, the Accept-Language request-header
field can be used to allow the client to indicate to the server in
which language it would prefer to receive reason phrases, session
descriptions or status responses carried as message bodies. A proxy
MAY use this field to help select the destination for the call, for
example, a human operator conversant in a language spoken by the
caller.
Example:
Accept-Language: da, en-gb;q=0.8, en;q=0.7
6.10 Allow
The Allow entity-header field lists the set of methods supported by
the resource identified by the Request-URI. The purpose of this field
is strictly to inform the recipient of valid methods associated with
the resource. An Allow header field MUST be present in a 405 (Method
Not Allowed) response and SHOULD be present in an OPTIONS response.
Allow = "Allow" ":" 1#Method
6.11 Authorization
A user agent that wishes to authenticate itself with a server --
usually, but not necessarily, after receiving a 401 response -- MAY
do so by including an Authorization request-header field with the
request. The Authorization field value consists of credentials
containing the authentication information of the user agent for the
realm of the resource being requested.
Section 13.2 overviews the use of the Authorization header, and
section 15 describes the syntax and semantics when used with PGP
based authentication.
6.12 Call-ID
The Call-ID general-header field uniquely identifies a particular
invitation or all registrations of a particular client. Note that a
single multimedia conference can give rise to several calls with
different Call-IDs, e.g., if a user invites a single individual
several times to the same (long-running) conference.
For an INVITE request, a callee user agent server SHOULD NOT alert
the user if the user has responded previously to the Call-ID in the
INVITE request. If the user is already a member of the conference and
the conference parameters contained in the session description have
not changed, a callee user agent server MAY silently accept the call,
regardless of the Call-ID. An invitation for an existing Call-ID or
session can change the parameters of the conference. A client
application MAY decide to simply indicate to the user that the
conference parameters have been changed and accept the invitation
automatically or it MAY require user confirmation.
A user may be invited to the same conference or call using several
different Call-IDs. If desired, the client MAY use identifiers within
the session description to detect this duplication. For example, SDP
contains a session id and version number in the origin (o) field.
The REGISTER and OPTIONS methods use the Call-ID value to
unambiguously match requests and responses. All REGISTER requests
issued by a single client SHOULD use the same Call-ID, at least
within the same boot cycle.
Since the Call-ID is generated by and for SIP, there is no
reason to deal with the complexity of URL-encoding and
case-ignoring string comparison.
Call-ID = ( "Call-ID" | "i" ) ":" local-id "@" host
local-id = 1*uric
"host" SHOULD be either a fully qualified domain name or a globally
routable IP address. If this is the case, the "local-id" SHOULD be an
identifier consisting of URI characters that is unique within "host".
Use of cryptographically random identifiers [27] is RECOMMENDED. If,
however, host is not an FQDN or globally routable IP address (such as
a net 10 address), the local-id MUST be globally unique, as opposed
to unique within host. These rules guarantee overall global
uniqueness of the Call-ID. The value for Call-ID MUST NOT be reused
for a different call. Call-IDs are case-sensitive.
Using cryptographically random identifiers provides some
protection against session hijacking. Call-ID, To and From
are needed to identify a call leg. The distinction between
call and call leg matters in calls with third-party
control.
For systems which have tight bandwidth constraints, many of the
mandatory SIP headers have a compact form, as discussed in Section 9.
These are alternate names for the headers which occupy less space in
the message. In the case of Call-ID, the compact form is i.
For example, both of the following are valid:
Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com
or
i:f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com
6.13 Contact
The Contact general-header field can appear in INVITE, ACK, and
REGISTER requests, and in 1xx, 2xx, 3xx, and 485 responses. In
general, it provides a URL where the user can be reached for further
communications.
INVITE and ACK requests: INVITE and ACK requests MAY contain Contact
headers indicating from which location the request is
originating.
This allows the callee to send future requests, such as
BYE, directly to the caller instead of through a series of
proxies. The Via header is not sufficient since the
desired address may be that of a proxy.
INVITE 2xx responses: A user agent server sending a definitive,
positive response (2xx) MAY insert a Contact response header
field indicating the SIP address under which it is reachable
most directly for future SIP requests, such as ACK, within the
same Call-ID. The Contact header field contains the address of
the server itself or that of a proxy, e.g., if the host is
behind a firewall. The value of this Contact header is copied
into the Request-URI of subsequent requests for this call if the
response did not also contain a Record-Route header. If the
response also contains a Record-Route header field, the address
in the Contact header field is added as the last item in the
Route header field. See Section 6.29 for details.
The Contact value SHOULD NOT be cached across calls, as it
may not represent the most desirable location for a
particular destination address.
INVITE 1xx responses: A UAS sending a provisional response (1xx) MAY
insert a Contact response header. It has the same semantics in a
1xx response as a 2xx INVITE response. Note that CANCEL requests
MUST NOT be sent to that address, but rather follow the same
path as the original request.
REGISTER requests: REGISTER requests MAY contain a Contact header
field indicating at which locations the user is reachable. The
REGISTER request defines a wildcard Contact field, "*", which
MUST only be used with Expires: 0 to remove all registrations
for a particular user. An optional "expires" parameter indicates
the desired expiration time of the registration. If a Contact
entry does not have an "expires" parameter, the Expires header
field is used as the default value. If neither of these
mechanisms is used, SIP URIs are assumed to expire after one
hour. Other URI schemes have no expiration times.
REGISTER 2xx responses: A REGISTER response MAY return all locations
at which the user is currently reachable. An optional "expires"
parameter indicates the expiration time of the registration. If
a Contact entry does not have an "expires" parameter, the value
of the Expires header field indicates the expiration time. If
neither mechanism is used, the expiration time specified in the
request, explicitly or by default, is used.
3xx and 485 responses: The Contact response-header field can be used
with a 3xx or 485 (Ambiguous) response codes to indicate one or
more alternate addresses to try. It can appear in responses to
BYE, INVITE and OPTIONS methods. The Contact header field
contains URIs giving the new locations or user names to try, or
may simply specify additional transport parameters. A 300
(Multiple Choices), 301 (Moved Permanently), 302 (Moved
Temporarily) or 485 (Ambiguous) response SHOULD contain a
Contact field containing URIs of new addresses to be tried. A
301 or 302 response may also give the same location and username
that was being tried 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. The
client copies the "user", "password", "host", "port" and "user-
param" elements of the Contact URI into the Request-URI of the
redirected request and directs the request to the address
specified by the "maddr" and "port" parameters, using the
transport protocol given in the "transport" parameter. If
"maddr" is a multicast address, the value of "ttl" is used as
the time-to-live value.
Note that the Contact header field MAY also refer to a different
entity than the one originally called. For example, a SIP call
connected to GSTN gateway may need to deliver a special information
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
URLs. For example, it could contain URL's for phones, fax, or irc (if
they were defined) or a mailto: (RFC 2368, [28]) URL.
The following parameters are defined. Additional parameters may be
defined in other specifications.
q: The "qvalue" indicates the relative preference among the locations
given. "qvalue" values are decimal numbers from 0 to 1, with
higher values indicating higher preference.
action: The "action" parameter is used only when registering with the
REGISTER request. It indicates whether the client wishes that
the server proxy or redirect future requests intended for the
client. If this parameter is not specified the action taken
depends on server configuration. In its response, the registrar
SHOULD indicate the mode used. This parameter is ignored for
other requests.
expires: The "expires" parameter indicates how long the URI is valid.
The parameter is either a number indicating seconds or a quoted
string containing a SIP-date. If this parameter is not provided,
the value of the Expires header field determines how long the
URI is valid. Implementations MAY treat values larger than
2**32-1 (4294967295 seconds or 136 years) as equivalent to
2**32-1.
Contact = ( "Contact" | "m" ) ":"
("*" | (1# (( name-addr | addr-spec )
[ *( ";" contact-params ) ] [ comment ] )))
name-addr = [ display-name ] "<" addr-spec ">"
addr-spec = SIP-URL | URI
display-name = *token | quoted-string
contact-params = "q" "=" qvalue
| "action" "=" "proxy" | "redirect"
| "expires" "=" delta-seconds | <"> SIP-date <">
| extension-attribute
extension-attribute = extension-name [ "=" extension-value ]
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 current syntax corresponds to that for the To and From
header, which also allows the use of display names.
Example:
Contact: "Mr. Watson" <sip:watson@worcester.bell-telephone.com>
;q=0.7; expires=3600,
"Mr. Watson" <mailto:watson@bell-telephone.com> ;q=0.1
6.14 Content-Encoding
Content-Encoding = ( "Content-Encoding" | "e" ) ":"
1#content-coding
The Content-Encoding entity-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, the content
codings MUST be listed in the order in which they were applied.
All content-coding values are case-insensitive. The Internet Assigned
Numbers Authority (IANA) acts as a registry for content-coding value
tokens. See [3.5] for a definition of the syntax for content-coding.
Clients MAY apply content encodings to the body in requests. If the
server is not capable of decoding the body, or does not recognize any
of the content-coding values, it MUST send a 415 "Unsupported Media
Type" response, listing acceptable encodings in the Accept-Encoding
header. A server MAY apply content encodings to the bodies in
responses. The server MUST only use encodings listed in the Accept-
Encoding header in the request.
6.15 Content-Length
The Content-Length entity-header field indicates the size of the
message-body, in decimal number of octets, sent to the recipient.
Content-Length = ( "Content-Length" | "l" ) ":" 1*DIGIT
An example is
Content-Length: 3495
Applications SHOULD use this field to indicate the size of the
message-body to be transferred, regardless of the media type of the
entity. 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 MUST be set to zero. If a server receives a UDP request
without Content-Length, it MUST assume that the request encompasses
the remainder of the packet. If a server receives a UDP request with
a Content-Length, but the value is larger than the size of the body
sent in the request, the client SHOULD generate a 400 class response.
If there is additional data in the UDP packet after the last byte of
the body has been read, the server MUST treat the remaining data as a
separate message. This allows several messages to be placed in a
single UDP packet.
If a response does not contain a Content-Length, the client assumes
that it encompasses the remainder of the UDP packet or the data until
the TCP connection is closed, as applicable. Section 8 describes how
to determine the length of the message body.
6.16 Content-Type
The Content-Type entity-header field indicates the media type of the
message-body sent to the recipient. The "media-type" element is
defined in [H3.7].
Content-Type = ( "Content-Type" | "c" ) ":" media-type
Examples of this header field are
Content-Type: application/sdp
Content-Type: text/html; charset=ISO-8859-4
6.17 CSeq
Clients MUST add the CSeq (command sequence) general-header field to
every request. A CSeq header field in a request contains the request
method and a single decimal sequence number chosen by the requesting
client, unique within a single value of Call-ID. The sequence number
MUST be expressible as a 32-bit unsigned integer. The initial value
of the sequence number is arbitrary, but MUST be less than 2**31.
Consecutive requests that differ in request method, headers or body,
but have the same Call-ID MUST contain strictly monotonically
increasing and contiguous sequence numbers; sequence numbers do not
wrap around. Retransmissions of the same request carry the same
sequence number, but an INVITE with a different message body or
different header fields (a "re-invitation") acquires a new, higher
sequence number. A server MUST echo the CSeq value from the request
in its response. If the Method value is missing in the received CSeq
header field, the server fills it in appropriately.
The ACK and CANCEL requests MUST contain the same CSeq value as the
INVITE request that it refers to, while a BYE request cancelling an
invitation MUST have a higher sequence number. A BYE request with a
CSeq that is not higher should cause a 400 response to be generated.
A user agent server MUST remember the highest sequence number for any
INVITE request with the same Call-ID value. The server MUST respond
to, and then discard, any INVITE request with a lower sequence
number.
All requests spawned in a parallel search have the same CSeq value as
the request triggering the parallel search.
CSeq = "CSeq" ":" 1*DIGIT Method
Strictly speaking, CSeq header fields are needed for any
SIP request that can be cancelled by a BYE or CANCEL
request or where a client can issue several requests for
the same Call-ID in close succession. Without a sequence
number, the response to an INVITE could be mistaken for the
response to the cancellation (BYE or CANCEL). Also, if the
network duplicates packets or if an ACK is delayed until
the server has sent an additional response, the client
could interpret an old response as the response to a re-
invitation issued shortly thereafter. Using CSeq also makes
it easy for the server to distinguish different versions of
an invitation, without comparing the message body.
The Method value allows the client to distinguish the response to an
INVITE request from that of a CANCEL response. CANCEL requests can be
generated by proxies; if they were to increase the sequence number,
it might conflict with a later request issued by the user agent for
the same call.
With a length of 32 bits, a server 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 to use a time-based initial sequence
number, if the client desires. A client could, for example, choose
the 31 most significant bits of a 32-bit second clock as an initial
sequence number.
Forked requests MUST have the same CSeq as there would be ambiguity
otherwise between these forked requests and later BYE issued by the
client user agent.
Example:
CSeq: 4711 INVITE
6.18 Date
Date is a general-header field. Its syntax is:
SIP-date = rfc1123-date
See [H14.19] for a definition of rfc1123-date. Note that unlike
HTTP/1.1, SIP only supports the most recent RFC1123 [29] formatting
for dates.
The Date header field reflects the time when the request or response
is first sent. Thus, retransmissions have the same Date header field
value as the original.
The Date header field can be used by simple end systems
without a battery-backed clock to acquire a notion of
current time.
6.19 Encryption
The Encryption general-header field specifies that the content has
been encrypted. Section 13 describes the overall SIP security
architecture and algorithms. This header field is intended for end-
to-end encryption of requests and responses. Requests are encrypted
based on the publi