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.
Handley, et al. Standards Track [Page 1]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 2]
RFC 2543 SIP: Session Initiation Protocol March 1999
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
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RFC 2543 SIP: Session Initiation Protocol March 1999
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
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RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 5]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 6]RFC 2543 SIP: Session Initiation Protocol March 1999
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;
Handley, et al. Standards Track [Page 7]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
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RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 9]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 10]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 11]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 12]RFC 2543 SIP: Session Initiation Protocol March 1999
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:
Handley, et al. Standards Track [Page 13]
RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 14]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 15]
RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 16]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 17]
RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 18]
RFC 2543 SIP: Session Initiation Protocol March 1999
+....... 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
Handley, et al. Standards Track [Page 19]RFC 2543 SIP: Session Initiation Protocol March 1999
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].
Handley, et al. Standards Track [Page 20]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 21]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 22]RFC 2543 SIP: Session Initiation Protocol March 1999
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:
Handley, et al. Standards Track [Page 23]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 24]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 25]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 26]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 27]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 28]RFC 2543 SIP: Session Initiation Protocol March 1999
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.,
Handley, et al. Standards Track [Page 29]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 30]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 31]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 32]RFC 2543 SIP: Session Initiation Protocol March 1999
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).
Handley, et al. Standards Track [Page 33]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 34]RFC 2543 SIP: Session Initiation Protocol March 1999
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;
Handley, et al. Standards Track [Page 35]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 36]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 37]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 38]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 39]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 40]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 41]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 42]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 43]RFC 2543 SIP: Session Initiation Protocol March 1999
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].
Handley, et al. Standards Track [Page 44]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 45]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 46]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 47]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 48]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 50]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 51]RFC 2543 SIP: Session Initiation Protocol March 1999
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
Handley, et al. Standards Track [Page 52]RFC 2543 SIP: Session Initiation Protocol March 1999
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.
Handley, et al. Standards Track [Page 53]RFC 2543 SIP: Session Initiation Protocol March 1999
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 public key belonging to the entity named in the To
header field. Responses are encrypted based on the public key
conveyed in the Response-Key header field. Note that the public keys
themselves may not be used for the encryption. This depends on the
particular algorithms used.
For any encrypted message, at least the message body and possibly
other message header fields are encrypted. An application receiving a
request or response containing an Encryption header field decrypts
the body and then concatenates the plaintext to the request line and
headers of the original message. Message headers in the decrypted
part completely replace those with the same field name in the
plaintext part. (Note: If only the body of the message is to be
encrypted, the body has to be prefixed with CRLF to allow proper
concatenation.) Note that the request method and Request-URI cannot
be encrypted.
Encryption only provides privacy; the recipient has no
guarantee that the request or response came from the party
listed in the From message header, only that the sender
used the recipient's public key. However, proxies will not
be able to modify the request or response.
Encryption = "Encryption" ":" encryption-scheme 1*SP
#encryption-params
encryption-scheme = token
encryption-params = token "=" ( token | quoted-string )
The token indicates the form of encryption used; it is
described in section 13.
Handley, et al. Standards Track [Page 54]RFC 2543 SIP: Session Initiation Protocol March 1999
The example in Figure 10 shows a message encrypted with ASCII-armored
PGP that was generated by applying "pgp -ea" to the payload to be
encrypted.
INVITE sip:watson@boston.bell-telephone.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5
From: <sip:a.g.bell@bell-telephone.com>
To: T. A. Watson <sip:watson@bell-telephone.com>
Call-ID: 187602141351@worcester.bell-telephone.com
Content-Length: 885
Encryption: PGP version=2.6.2,encoding=ascii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=bOW+
Figure 10: PGP Encryption Example
Since proxies can base their forwarding decision on any combination
of SIP header fields, there is no guarantee that an encrypted request
"hiding" header fields will reach the same destination as an
otherwise identical un-encrypted request.
6.20 Expires
The Expires entity-header field gives the date and time after which
the message content expires.
This header field is currently defined only for the REGISTER and
INVITE methods. For REGISTER, it is a request and response-header
field. In a REGISTER request, the client indicates how long it wishes
the registration to be valid. In the response, the server indicates
Handley, et al. Standards Track [Page 55]
RFC 2543 SIP: Session Initiation Protocol March 1999
the earliest expiration time of all registrations. The server MAY
choose a shorter time interval than that requested by the client, but
SHOULD NOT choose a longer one.
For INVITE requests, it is a request and response-header field. In a
request, the caller can limit the validity of an invitation, for
example, if a client wants to limit the time duration of a search or
a conference invitation. A user interface MAY take this as a hint to
leave the invitation window on the screen even if the user is not
currently at the workstation. This also limits the duration of a
search. If the request expires before the search completes, the proxy
returns a 408 (Request Timeout) status. In a 302 (Moved Temporarily)
response, a server can advise the client of the maximal duration of
the redirection.
The value of this field can be either a SIP-date or an integer number
of seconds (in decimal), measured from the receipt of the request.
The latter approach is preferable for short durations, as it does not
depend on clients and servers sharing a synchronized clock.
Implementations MAY treat values larger than 2**32-1 (4294967295 or
136 years) as equivalent to 2**32-1.
Expires = "Expires" ":" ( SIP-date | delta-seconds )
Two examples of its use are
Expires: Thu, 01 Dec 1994 16:00:00 GMT
Expires: 5
6.21 From
Requests and responses MUST contain a From general-header field,
indicating the initiator of the request. The From field MAY contain
the "tag" parameter. The server copies the From header field from the
request to the response. The optional "display-name" is meant to be
rendered by a human-user interface. A system SHOULD use the display
name "Anonymous" if the identity of the client is to remain hidden.
The 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.
Handley, et al. Standards Track [Page 56]RFC 2543 SIP: Session Initiation Protocol March 1999
Even if the "display-name" is empty, the "name-addr" form MUST be
used if the "addr-spec" contains a comma, question mark, or
semicolon.
From = ( "From" | "f" ) ":" ( name-addr | addr-spec )
*( ";" addr-params )
addr-params = tag-param
tag-param = "tag=" UUID
UUID = 1*( hex | "-" )
Examples:
From: "A. G. Bell" <sip:agb@bell-telephone.com>
From: sip:+12125551212@server.phone2net.com
From: Anonymous <sip:c8oqz84zk7z@privacy.org>
The "tag" MAY appear in the From field of a request. It MUST be
present when it is possible that two instances of a user sharing a
SIP address can make call invitations with the same Call-ID.
The "tag" value MUST be globally unique and cryptographically random
with at least 32 bits of randomness. A single user maintains the same
tag throughout the call identified by the Call-ID.
Call-ID, To and From are needed to identify a call leg.
The distinction between call and call leg matters in calls
with multiple responses to a forked request. The format is
similar to the equivalent RFC 822 [24] header, but with a
URI instead of just an email address.
6.22 Hide
A client uses the Hide request header field to indicate that it wants
the path comprised of the Via header fields (Section 6.40) to be
hidden from subsequent proxies and user agents. It can take two
forms: Hide: route and Hide: hop. Hide header fields are typically
added by the client user agent, but MAY be added by any proxy along
the path.
Handley, et al. Standards Track [Page 57]RFC 2543 SIP: Session Initiation Protocol March 1999
If a request contains the "Hide: route" header field, all following
proxies SHOULD hide their previous hop. If a request contains the
"Hide: hop" header field, only the next proxy SHOULD hide the
previous hop and then remove the Hide option unless it also wants to
remain anonymous.
A server hides the previous hop by encrypting the "host" and "port"
parts of the top-most Via header field with an algorithm of its
choice. Servers SHOULD add additional "salt" to the "host" and "port"
information prior to encryption to prevent malicious downstream
proxies from guessing earlier parts of the path based on seeing
identical encrypted Via headers. Hidden Via fields are marked with
the "hidden" Via option, as described in Section 6.40.
A server that is capable of hiding Via headers MUST attempt to
decrypt all Via headers marked as "hidden" to perform loop detection.
Servers that are not capable of hiding can ignore hidden Via fields
in their loop detection algorithm.
If hidden headers were not marked, a proxy would have to
decrypt all headers to detect loops, just in case one was
encrypted, as the Hide: Hop option may have been removed
along the way.
A host MUST NOT add such a "Hide: hop" header field unless it can
guarantee it will only send a request for this destination to the
same next hop. The reason for this is that it is possible that the
request will loop back through this same hop from a downstream proxy.
The loop will be detected by the next hop if the choice of next hop
is fixed, but could loop an arbitrary number of times otherwise.
A client requesting "Hide: route" can only rely on keeping the
request path private if it sends the request to a trusted proxy.
Hiding the route of a SIP request is of limited value if the request
results in data packets being exchanged directly between the calling
and called user agent.
The use of Hide header fields is discouraged unless path privacy is
truly needed; Hide fields impose extra processing costs and
restrictions for proxies and can cause requests to generate 482 (Loop
Detected) responses that could otherwise be avoided.
The encryption of Via header fields is described in more detail in
Section 13.
The Hide header field has the following syntax:
Handley, et al. Standards Track [Page 58]RFC 2543 SIP: Session Initiation Protocol March 1999
Hide = "Hide" ":" ( "route" | "hop" )
6.23 Max-Forwards
The Max-Forwards request-header field may be used with any SIP method
to limit the number of proxies or gateways that can forward the
request to the next downstream server. This can also be useful when
the client is attempting to trace a request chain which appears to be
failing or looping in mid-chain.
Max-Forwards = "Max-Forwards" ":" 1*DIGIT
The Max-Forwards value is a decimal integer indicating the remaining
number of times this request message is allowed to be forwarded.
Each proxy or gateway recipient of a request containing a Max-
Forwards header field MUST check and update its value prior to
forwarding the request. If the received value is zero (0), the
recipient MUST NOT forward the request. Instead, for the OPTIONS and
REGISTER methods, it MUST respond as the final recipient. For all
other methods, the server returns 483 (Too many hops).
If the received Max-Forwards value is greater than zero, then the
forwarded message MUST contain an updated Max-Forwards field with a
value decremented by one (1).
Example:
Max-Forwards: 6
6.24 Organization
The Organization general-header field conveys the name of the
organization to which the entity issuing the request or response
belongs. It MAY also be inserted by proxies at the boundary of an
organization.
The field MAY be used by client software to filter calls.
Handley, et al. Standards Track [Page 59]RFC 2543 SIP: Session Initiation Protocol March 1999
Organization = "Organization" ":" *TEXT-UTF8
6.25 Priority
The Priority request-header field indicates the urgency of the
request as perceived by the client.
Priority = "Priority" ":" priority-value
priority-value = "emergency" | "urgent" | "normal"
| "non-urgent"
It is RECOMMENDED that the value of "emergency" only be used when
life, limb or property are in imminent danger.
Examples:
Subject: A tornado is heading our way!
Priority: emergency
Subject: Weekend plans
Priority: non-urgent
These are the values of RFC 2076 [30], with the addition of
"emergency".
6.26 Proxy-Authenticate
The Proxy-Authenticate response-header field MUST be included as part
of a 407 (Proxy Authentication Required) response. The field value
consists of a challenge that indicates the authentication scheme and
parameters applicable to the proxy for this Request-URI.
Unlike its usage within HTTP, the Proxy-Authenticate header MUST be
passed upstream in the response to the UAC. In SIP, only UAC's can
authenticate themselves to proxies.
The syntax for this header is defined in [H14.33]. See 14 for further
details on its usage.
Handley, et al. Standards Track [Page 60]RFC 2543 SIP: Session Initiation Protocol March 1999
A client SHOULD cache the credentials used for a particular proxy
server and realm for the next request to that server. Credentials
are, in general, valid for a specific value of the Request-URI at a
particular proxy server. If a client contacts a proxy server that has
required authentication in the past, but the client does not have
credentials for the particular Request-URI, it MAY attempt to use the
most-recently used credential. The server responds with 401
(Unauthorized) if the client guessed wrong.
This suggested caching behavior is motivated by proxies
restricting phone calls to authenticated users. It seems
likely that in most cases, all destinations require the
same password. Note that end-to-end authentication is
likely to be destination-specific.
6.27 Proxy-Authorization
The Proxy-Authorization request-header field allows the client to
identify itself (or its user) to a proxy which requires
authentication. The Proxy-Authorization field value consists of
credentials containing the authentication information of the user
agent for the proxy and/or realm of the resource being requested.
Unlike Authorization, the Proxy-Authorization header field applies
only to the next outbound proxy that demanded authentication using
the Proxy- Authenticate field. When multiple proxies are used in a
chain, the Proxy-Authorization header field is consumed by the first
outbound proxy that was expecting to receive credentials. A proxy MAY
relay the credentials from the client request to the next proxy if
that is the mechanism by which the proxies cooperatively authenticate
a given request.
See [H14.34] for a definition of the syntax, and section 14 for a
discussion of its usage.
6.28 Proxy-Require
The Proxy-Require header field is used to indicate proxy-sensitive
features that MUST be supported by the proxy. Any Proxy-Require
header field features that are not supported by the proxy MUST be
negatively acknowledged by the proxy to the client if not supported.
Proxy servers treat this field identically to the Require field.
See Section 6.30 for more details on the mechanics of this message
and a usage example.
Handley, et al. Standards Track [Page 61]RFC 2543 SIP: Session Initiation Protocol March 1999
6.29 Record-Route
The Record-Route request and response header field is added to a
request by any proxy that insists on being in the path of subsequent
requests for the same call leg. It contains a globally reachable
Request-URI that identifies the proxy server. Each proxy server adds
its Request-URI to the beginning of the list.
The server copies the Record-Route header field unchanged into the
response. (Record-Route is only relevant for 2xx responses.)
The calling user agent client copies the Record-Route header into a
Route header field of subsequent requests within the same call leg,
reversing the order of requests, so that the first entry is closest
to the user agent client. If the response contained a Contact header
field, the calling user agent adds its content as the last Route
header. Unless this would cause a loop, any client MUST send any
subsequent requests for this call leg to the first Request-URI in the
Route request header field and remove that entry.
The calling user agent MUST NOT use the Record-Route header field in
requests that contain Route header fields.
Some proxies, such as those controlling firewalls or in an
automatic call distribution (ACD) system, need to maintain
call state and thus need to receive any BYE and ACK packets
for the call.
The Record-Route header field has the following syntax:
Record-Route = "Record-Route" ":" 1# name-addr
Proxy servers SHOULD use the "maddr" URL parameter containing their
address to ensure that subsequent requests are guaranteed to reach
exactly the same server.
Example for a request that has traversed the hosts ieee.org and
bell-telephone.com , in that order:
Record-Route: <sip:a.g.bell@bell-telephone.com>,
<sip:a.bell@ieee.org>
Handley, et al. Standards Track [Page 62]RFC 2543 SIP: Session Initiation Protocol March 1999
6.30 Require
The Require request-header field is used by clients to tell user
agent servers about options that the client expects the server to
support in order to properly process the request. If a server does
not understand the option, it MUST respond by returning status code
420 (Bad Extension) and list those options it does not understand in
the Unsupported header.
Require = "Require" ":" 1#option-tag
Example:
C->S: INVITE sip:watson@bell-telephone.com SIP/2.0
Require: com.example.billing
Payment: sheep_skins, conch_shells
S->C: SIP/2.0 420 Bad Extension
Unsupported: com.example.billing
This is to make sure that the client-server interaction
will proceed without delay when all options are understood
by both sides, and only slow down if options are not
understood (as in the example above). For a well-matched
client-server pair, the interaction proceeds quickly,
saving a round-trip often required by negotiation
mechanisms. In addition, it also removes ambiguity when the
client requires features that the server does not
understand. Some features, such as call handling fields,
are only of interest to end systems.
Proxy and redirect servers MUST ignore features that are not
understood. If a particular extension requires that intermediate
devices support it, the extension MUST be tagged in the Proxy-Require
field as well (see Section 6.28).
6.31 Response-Key
The Response-Key request-header field can be used by a client to
request the key that the called user agent SHOULD use to encrypt the
response with. The syntax is:
Handley, et al. Standards Track [Page 63]RFC 2543 SIP: Session Initiation Protocol March 1999
Response-Key = "Response-Key" ":" key-scheme 1*SP #key-param
key-scheme = token
key-param = token "=" ( token | quoted-string )
The "key-scheme" gives the type of encryption to be used for the
response. Section 13 describes security schemes.
If the client insists that the server return an encrypted response,
it includes a
Require: org.ietf.sip.encrypt-response
header field in its request. If the server cannot encrypt for
whatever reason, it MUST follow normal Require header field
procedures and return a 420 (Bad Extension) response. If this Require
header field is not present, a server SHOULD still encrypt if it can.
6.32 Retry-After
The Retry-After general-header field can be used with a 503 (Service
Unavailable) response to indicate how long the service is expected to
be unavailable to the requesting client and with a 404 (Not Found),
600 (Busy), or 603 (Decline) response to indicate when the called
party anticipates being available again. The value of this field can
be either an SIP-date or an integer number of seconds (in decimal)
after the time of the response.
A REGISTER request MAY include this header field when deleting
registrations with "Contact: * ;expires: 0". The Retry-After value
then indicates when the user might again be reachable. The registrar
MAY then include this information in responses to future calls.
An optional comment can be used to indicate additional information
about the time of callback. An optional "duration" parameter
indicates how long the called party will be reachable starting at the
initial time of availability. If no duration parameter is given, the
service is assumed to be available indefinitely.
Retry-After = "Retry-After" ":" ( SIP-date | delta-seconds )
[ comment ] [ ";" "duration" "=" delta-seconds ]
Examples of its use are
Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting)
Handley, et al. Standards Track [Page 64]RFC 2543 SIP: Session Initiation Protocol March 1999
Retry-After: Mon, 01 Jan 9999 00:00:00 GMT
(Dear John: Don't call me back, ever)
Retry-After: Fri, 26 Sep 1997 21:00:00 GMT;duration=3600
Retry-After: 120
In the third example, the callee is reachable for one hour starting
at 21:00 GMT. In the last example, the delay is 2 minutes.
6.33 Route
The Route request-header field determines the route taken by a
request. Each host removes the first entry and then proxies the
request to the host listed in that entry, also using it as the
Request-URI. The operation is further described in Section 6.29.
The Route header field has the following syntax:
Route = "Route" ":" 1# name-addr
6.34 Server
The Server response-header field contains information about the
software used by the user agent server to handle the request. The
syntax for this field is defined in [H14.39].
6.35 Subject
This is intended to provide a summary, or to indicate the nature, of
the call, allowing call filtering without having to parse the session
description. (Also, the session description does not have to use the
same subject indication as the invitation.)
Subject = ( "Subject" | "s" ) ":" *TEXT-UTF8
Example:
Subject: Tune in - they are talking about your work!
Handley, et al. Standards Track [Page 65]RFC 2543 SIP: Session Initiation Protocol March 1999
6.36 Timestamp
The timestamp general-header field describes when the client sent the
request to the server. The value of the timestamp is of significance
only to the client and it MAY use any timescale. The server MUST echo
the exact same value and MAY, if it has accurate information about
this, add a floating point number indicating the number of seconds
that have elapsed since it has received the request. The timestamp is
used by the client to compute the round-trip time to the server so
that it can adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
delay = *(DIGIT) [ "." *(DIGIT) ]
Note that there MUST NOT be any LWS between a DIGIT and the decimal
point.
6.37 To
The To general-header field specifies recipient of the request, with
the same SIP URL syntax as the From field.
To = ( "To" | "t" ) ":" ( name-addr | addr-spec )
*( ";" addr-params )
Requests and responses MUST contain a To general-header field,
indicating the desired recipient of the request. The optional
"display-name" is meant to be rendered by a human-user interface.
The UAS or redirect server copies the To header field into its
response, and MUST add a "tag" parameter if the request contained
more than one Via header field.
If there was more than one Via header field, the request
was handled by at least one proxy server. Since the
receiver cannot know whether any of the proxy servers
forked the request, it is safest to assume that they might
have.
The 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.
Handley, et al. Standards Track [Page 66]RFC 2543 SIP: Session Initiation Protocol March 1999
The "tag" parameter serves as a general mechanism to distinguish
multiple instances of a user identified by a single SIP URL. As
proxies can fork requests, the same request can reach multiple
instances of a user (mobile and home phones, for example). As each
can respond, there needs to be a means to distinguish the responses
from each at the caller. The situation also arises with multicast
requests. The tag in the To header field serves to distinguish
responses at the UAC. It MUST be placed in the To field of the
response by each instance when there is a possibility that the
request was forked at an intermediate proxy. The "tag" MUST be added
by UAS, registrars and redirect servers, but MUST NOT be inserted
into responses forwarded upstream by proxies. The "tag" is added for
all definitive responses for all methods, and MAY be added for
informational responses from a UAS or redirect server. All subsequent
transactions between two entities MUST include the "tag" parameter,
as described in Section 11.
See Section 6.21 for details of the "tag" parameter.
The "tag" parameter in To headers is ignored when matching responses
to requests that did not contain a "tag" in their To header.
A SIP server returns a 400 (Bad Request) response if it receives a
request with a To header field containing a URI with a scheme it does
not recognize.
Even if the "display-name" is empty, the "name-addr" form MUST be
used if the "addr-spec" contains a comma, question mark, or
semicolon.
The following are examples of valid To headers:
To: The Operator <sip:operator@cs.columbia.edu>;tag=287447
To: sip:+12125551212@server.phone2net.com
Call-ID, To and From are needed to identify a call leg.
The distinction between call and call leg matters in calls
with multiple responses from a forked request. The "tag" is
added to the To header field in the response to allow
forking of future requests for the same call by proxies,
while addressing only one of the possibly several
responding user agent servers. It also allows several
instances of the callee to send requests that can be
distinguished.
Handley, et al. Standards Track [Page 67]RFC 2543 SIP: Session Initiation Protocol March 1999
6.38 Unsupported
The Unsupported response-header field lists the features not
supported by the server. See Section 6.30 for a usage example and
motivation.
Syntax:
Unsupported = "Unsupported" ":" 1#option-tag
6.39 User-Agent
The User-Agent general-header field contains information about the
client user agent originating the request. The syntax and semantics
are defined in [H14.42].
6.40 Via
The Via field indicates the path taken by the request so far. This
prevents request looping and ensures replies take the same path as
the requests, which assists in firewall traversal and other unusual
routing situations.
6.40.1 Requests
The client originating the request MUST insert into the request a Via
field containing its host name or network address and, if not the
default port number, the port number at which it wishes to receive
responses. (Note that this port number can differ from the UDP source
port number of the request.) A fully-qualified domain name is
RECOMMENDED. Each subsequent proxy server that sends the request
onwards MUST add its own additional Via field before any existing Via
fields. A proxy that receives a redirection (3xx) response and then
searches recursively, MUST use the same Via headers as on the
original proxied request.
A proxy SHOULD check the top-most Via header field to ensure that it
contains the sender's correct network address, as seen from that
proxy. If the sender's address is incorrect, the proxy MUST add an
additional "received" attribute, as described 6.40.2.
A host behind a network address translator (NAT) or
firewall may not be able to insert a network address into
the Via header that can be reached by the next hop beyond
Handley, et al. Standards Track [Page 68]RFC 2543 SIP: Session Initiation Protocol March 1999
the NAT. Use of the received attribute allows SIP requests
to traverse NAT's which only modify the source IP address.
NAT's which modify port numbers, called Network Address
Port Translator's (NAPT) will not properly pass SIP when
transported on UDP, in which case an application layer
gateway is required. When run over TCP, SIP stands a better
chance of traversing NAT's, since its behavior is similar
to HTTP in this case (but of course on different ports).
A proxy sending a request to a multicast address MUST add the "maddr"
parameter to its Via header field, and SHOULD add the "ttl"
parameter. If a server receives a request which contained an "maddr"
parameter in the topmost Via field, it SHOULD send the response to
the multicast address listed in the "maddr" parameter.
If a proxy server receives a request which contains its own address
in the Via header value, it MUST respond with a 482 (Loop Detected)
status code.
A proxy server MUST NOT forward a request to a multicast group which
already appears in any of the Via headers.
This prevents a malfunctioning proxy server from causing
loops. Also, it cannot be guaranteed that a proxy server
can always detect that the address returned by a location
service refers to a host listed in the Via list, as a
single host may have aliases or several network interfaces.
6.40.2 Receiver-tagged Via Header Fields
Normally, every host that sends or forwards a SIP message adds a Via
field indicating the path traversed. However, it is possible that
Network Address Translators (NATs) changes the source address and
port of the request (e.g., from net-10 to a globally routable
address), in which case the Via header field cannot be relied on to
route replies. To prevent this, a proxy SHOULD check the top-most Via
header field to ensure that it contains the sender's correct network
address, as seen from that proxy. If the sender's address is
incorrect, the proxy MUST add a "received" parameter to the Via
header field inserted by the previous hop. Such a modified Via header
field is known as a receiver-tagged Via header field. An example is:
Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
Via: SIP/2.0/UDP 10.0.0.1:5060 ;received=199.172.136.3
Handley, et al. Standards Track [Page 69]RFC 2543 SIP: Session Initiation Protocol March 1999
In this example, the message originated from 10.0.0.1 and traversed a
NAT with the external address border.ieee.org (199.172.136.3) to
reach erlang.bell-telephone.com. The latter noticed the mismatch,
and added a parameter to the previous hop's Via header field,
containing the address that the packet actually came from. (Note that
the NAT border.ieee.org is not a SIP server.)
6.40.3 Responses
Via header fields in responses are processed by a proxy or UAC
according to the following rules:
1. The first Via header field should indicate the proxy or
client processing this response. If it does not, discard
the message. Otherwise, remove this Via field.
2. If there is no second Via header field, this response is
destined for this client. Otherwise, the processing depends
on whether the Via field contains a "maddr" parameter or is
a receiver-tagged field:
- If the second Via header field contains a "maddr"
parameter, send the response to the multicast address
listed there, using the port indicated in "sent-by", or
port 5060 if none is present. The response SHOULD be sent
using the TTL indicated in the "ttl" parameter, or with a
TTL of 1 if that parameter is not present. For
robustness, responses MUST be sent to the address
indicated in the "maddr" parameter even if it is not a
multicast address.
- If the second Via header field does not contain a "maddr"
parameter and is a receiver-tagged field (Section
6.40.2), send the message to the address in the
"received" parameter, using the port indicated in the
"sent-by" value, or using port 5060 if none is present.
- If neither of the previous cases apply, send the message
to the address indicated by the "sent-by" value in the
second Via header field.
6.40.4 User Agent and Redirect Servers
A UAS or redirect server sends a response based on one of the
following rules:
o If the first Via header field in the request contains a
"maddr" parameter, send the response to the multicast address
Handley, et al. Standards Track [Page 70]RFC 2543 SIP: Session Initiation Protocol March 1999
listed there, using the port indicated in "sent-by", or port
5060 if none is present. The response SHOULD be sent using the
TTL indicated in the "ttl" parameter, or with a TTL of 1 if
that parameter is not present. For robustness, responses MUST
be sent to the address indicated in the "maddr" parameter even
if it is not a multicast address.
o If the address in the "sent-by" value of the first Via field
differs from the source address of the packet, send the
response to the actual packet source address, similar to the
treatment for receiver-tagged Via header fields (Section
6.40.2).
o If neither of these conditions is true, send the response to
the address contained in the "sent-by" value. If the request
was sent using TCP, use the existing TCP connection if
available.
6.40.5 Syntax
The format for a Via header field is shown in Fig. 11. The defaults
for "protocol-name" and "transport" are "SIP" and "UDP",
respectively. The "maddr" parameter, designating the multicast
address, and the "ttl" parameter, designating the time-to-live (TTL)
value, are included only if the request was sent via multicast. The
"received" parameter is added only for receiver-added Via fields
(Section 6.40.2). For reasons of privacy, a client or proxy may wish
to hide its Via information by encrypting it (see Section 6.22). The
"hidden" parameter is included if this header field was hidden by the
upstream proxy (see 6.22). Note that privacy of the proxy relies on
the cooperation of the next hop, as the next-hop proxy will, by
necessity, know the IP address and port number of the source host.
The "branch" parameter is included by every forking proxy. The token
MUST be unique for each distinct request generated when a proxy
forks. CANCEL requests MUST have the same branch value as the
corresponding forked request. When a response arrives at the proxy it
can use the branch value to figure out which branch the response
corresponds to. A proxy which generates a single request (non-
forking) MAY also insert the "branch" parameter. The identifier has
to be unique only within a set of isomorphic requests.
Via: SIP/2.0/UDP first.example.com:4000;ttl=16
;maddr=224.2.0.1 ;branch=a7c6a8dlze (Example)
Via: SIP/2.0/UDP adk8
Handley, et al. Standards Track [Page 71]RFC 2543 SIP: Session Initiation Protocol March 1999
Via = ( "Via" | "v") ":" 1#( sent-protocol sent-by
*( ";" via-params ) [ comment ] )
via-params = via-hidden | via-ttl | via-maddr
| via-received | via-branch
via-hidden = "hidden"
via-ttl = "ttl" "=" ttl
via-maddr = "maddr" "=" maddr
via-received = "received" "=" host
via-branch = "branch" "=" token
sent-protocol = protocol-name "/" protocol-version "/" transport
protocol-name = "SIP" | token
protocol-version = token
transport = "UDP" | "TCP" | token
sent-by = ( host [ ":" port ] ) | ( concealed-host )
concealed-host = token
ttl = 1*3DIGIT ; 0 to 255
Figure 11: Syntax of Via header field
6.41 Warning
The Warning response-header field is used to carry additional
information about the status of a response. Warning headers are sent
with responses and have the following format:
Warning = "Warning" ":" 1#warning-value
warning-value = warn-code SP warn-agent SP warn-text
warn-code = 3DIGIT
warn-agent = ( host [ ":" port ] ) | pseudonym
; the name or pseudonym of the server adding
; the Warning header, for use in debugging
warn-text = quoted-string
A response MAY carry more than one Warning header.
The "warn-text" should be in a natural language that is most likely
to be intelligible to the human user receiving the response. This
decision can be based on any available knowledge, such as the
location of the cache or user, the Accept-Language field in a
request, or the Content-Language field in a response. The default
language is i-default [31].
Handley, et al. Standards Track [Page 72]RFC 2543 SIP: Session Initiation Protocol March 1999
Any server MAY add Warning headers to a response. Proxy servers MUST
place additional Warning headers before any Authorization headers.
Within that constraint, Warning headers MUST be added after any
existing Warning headers not covered by a signature. A proxy server
MUST NOT delete any Warning header field that it received with a
response.
When multiple Warning headers are attached to a response, the user
agent SHOULD display as many of them as possible, in the order that
they appear in the response. If it is not possible to display all of
the warnings, the user agent first displays warnings that appear
early in the response.
The warn-code consists of three digits. A first digit of "3"
indicates warnings specific to SIP.
This is a list of the currently-defined "warn-code"s, each with a
recommended warn-text in English, and a description of its meaning.
Note that these warnings describe failures induced by the session
description.
Warnings 300 through 329 are reserved for indicating problems with
keywords in the session description, 330 through 339 are warnings
related to basic network services requested in the session
description, 370 through 379 are warnings related to quantitative QoS
parameters requested in the session description, and 390 through 399
are miscellaneous warnings that do not fall into one of the above
categories.
300 Incompatible network protocol: One or more network protocols
contained in the session description are not available.
301 Incompatible network address formats: One or more network address
formats contained in the session description are not available.
302 Incompatible transport protocol: One or more transport protocols
described in the session description are not available.
303 Incompatible bandwidth units: One or more bandwidth measurement
units contained in the session description were not understood.
304 Media type not available: One or more media types contained in
the session description are not available.
305 Incompatible media format: One or more media formats contained in
the session description are not available.
Handley, et al. Standards Track [Page 73]RFC 2543 SIP: Session Initiation Protocol March 1999
306 Attribute not understood: One or more of the media attributes in
the session description are not supported.
307 Session description parameter not understood: A parameter other
than those listed above was not understood.
330 Multicast not available: The site where the user is located does
not support multicast.
331 Unicast not available: The site where the user is located does
not support unicast communication (usually due to the presence
of a firewall).
370 Insufficient bandwidth: The bandwidth specified in the session
description or defined by the media exceeds that known to be
available.
399 Miscellaneous warning: The warning text can include arbitrary
information to be presented to a human user, or logged. A system
receiving this warning MUST NOT take any automated action.
1xx and 2xx have been taken by HTTP/1.1.
Additional "warn-code"s, as in the example below, can be defined
through IANA.
Examples:
Warning: 307 isi.edu "Session parameter 'foo' not understood"
Warning: 301 isi.edu "Incompatible network address type 'E.164'"
6.42 WWW-Authenticate
The WWW-Authenticate response-header field MUST be included in 401
(Unauthorized) response messages. The field value consists of at
least one challenge that indicates the authentication scheme(s) and
parameters applicable to the Request-URI. See [H14.46] for a
definition of the syntax, and section 14 for an overview of usage.
The content of the "realm" parameter SHOULD be displayed to the user.
A user agent SHOULD cache the authorization credentials for a given
value of the destination (To header) and "realm" and attempt to re-
use these values on the next request for that destination.
Handley, et al. Standards Track [Page 74]RFC 2543 SIP: Session Initiation Protocol March 1999
In addition to the "basic" and "digest" authentication schemes
defined in the specifications cited above, SIP defines a new scheme,
PGP (RFC 2015, [32]), Section 15. Other schemes, such as S/MIME, are
for further study.
7 Status Code Definitions
The response codes are consistent with, and extend, HTTP/1.1 response
codes. Not all HTTP/1.1 response codes are appropriate, and only
those that are appropriate are given here. Other HTTP/1.1 response
codes SHOULD NOT be used. Response codes not defined by HTTP/1.1 have
codes x80 upwards to avoid clashes with future HTTP response codes.
Also, SIP defines a new class, 6xx. The default behavior for unknown
response codes is given for each category of codes.
7.1 Informational 1xx
Informational responses indicate that the server or proxy contacted
is performing some further action and does not yet have a definitive
response. The client SHOULD wait for a further response from the
server, and the server SHOULD send such a response without further
prompting. A server SHOULD send a 1xx response if it expects to take
more than 200 ms to obtain a final response. A server MAY issue zero
or more 1xx responses, with no restriction on their ordering or
uniqueness. Note that 1xx responses are not transmitted reliably,
that is, they do not cause the client to send an ACK. Servers are
free to retransmit informational responses and clients can inquire
about the current state of call processing by re-sending the request.
7.1.1 100 Trying
Some unspecified action is being taken on behalf of this call (e.g.,
a database is being consulted), but the user has not yet been
located.
7.1.2 180 Ringing
The called user agent has located a possible location where the user
has registered recently and is trying to alert the user.
7.1.3 181 Call Is Being Forwarded
A proxy server MAY use this status code to indicate that the call is
being forwarded to a different set of destinations.
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7.1.4 182 Queued
The called party is temporarily unavailable, but the callee has
decided to queue the call rather than reject it. When the callee
becomes available, it will return the appropriate final status
response. The reason phrase MAY give further details about the status
of the call, e.g., "5 calls queued; expected waiting time is 15
minutes". The server MAY issue several 182 responses to update the
caller about the status of the queued call.
7.2 Successful 2xx
The request was successful and MUST terminate a search.
7.2.1 200 OK
The request has succeeded. The information returned with the response
depends on the method used in the request, for example:
BYE: The call has been terminated. The message body is empty.
CANCEL: The search has been cancelled. The message body is empty.
INVITE: The callee has agreed to participate; the message body
indicates the callee's capabilities.
OPTIONS: The callee has agreed to share its capabilities, included in
the message body.
REGISTER: The registration has succeeded. The client treats the
message body according to its Content-Type.
7.3 Redirection 3xx
3xx responses give information about the user's new location, or
about alternative services that might be able to satisfy the call.
They SHOULD terminate an existing search, and MAY cause the initiator
to begin a new search if appropriate.
Any redirection (3xx) response MUST NOT suggest any of the addresses
in the Via (Section 6.40) path of the request in the Contact header
field. (Addresses match if their host and port number match.)
To avoid forwarding loops, a user agent client or proxy MUST check
whether the address returned by a redirect server equals an address
tried earlier.
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7.3.1 300 Multiple Choices
The address in the request resolved to several choices, each with its
own specific location, and the user (or user agent) can select a
preferred communication end point and redirect its request to that
location.
The response SHOULD include an entity containing a list of resource
characteristics and location(s) from which the user or user agent can
choose the one most appropriate, if allowed by the Accept request
header. The entity format is specified by the media type given in the
Content-Type header field. The choices SHOULD also be listed as
Contact fields (Section 6.13). Unlike HTTP, the SIP response MAY
contain several Contact fields or a list of addresses in a Contact
field. User agents MAY use the Contact header field value for
automatic redirection or MAY ask the user to confirm a choice.
However, this specification does not define any standard for such
automatic selection.
This status response is appropriate if the callee can be
reached at several different locations and the server
cannot or prefers not to proxy the request.
7.3.2 301 Moved Permanently
The user can no longer be found at the address in the Request-URI and
the requesting client SHOULD retry at the new address given by the
Contact header field (Section 6.13). The caller SHOULD update any
local directories, address books and user location caches with this
new value and redirect future requests to the address(es) listed.
7.3.3 302 Moved Temporarily
The requesting client SHOULD retry the request at the new address(es)
given by the Contact header field (Section 6.13). The duration of
the redirection can be indicated through an Expires (Section 6.20)
header. If there is no explicit expiration time, the address is only
valid for this call and MUST NOT be cached for future calls.
7.3.4 305 Use Proxy
The requested resource MUST be accessed through the proxy given by
the Contact field. The Contact field gives the URI of the proxy. The
recipient is expected to repeat this single request via the proxy.
305 responses MUST only be generated by user agent servers.
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7.3.5 380 Alternative Service
The call was not successful, but alternative services are possible.
The alternative services are described in the message body of the
response. Formats for such bodies are not defined here, and may be
the subject of future standardization.
7.4 Request Failure 4xx
4xx responses are definite failure responses from a particular
server. The client SHOULD NOT retry the same request without
modification (e.g., adding appropriate authorization). However, the
same request to a different server might be successful.
7.4.1 400 Bad Request
The request could not be understood due to malformed syntax.
7.4.2 401 Unauthorized
The request requires user authentication.
7.4.3 402 Payment Required
Reserved for future use.
7.4.4 403 Forbidden
The server understood the request, but is refusing to fulfill it.
Authorization will not help, and the request SHOULD NOT be repeated.
7.4.5 404 Not Found
The server has definitive information that the user does not exist at
the domain specified in the Request-URI. This status is also returned
if the domain in the Request-URI does not match any of the domains
handled by the recipient of the request.
7.4.6 405 Method Not Allowed
The method specified in the Request-Line is not allowed for the
address identified by the Request-URI. The response MUST include an
Allow header field containing a list of valid methods for the
indicated address.
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7.4.7 406 Not Acceptable
The resource identified by the request is only capable of generating
response entities which have content characteristics not acceptable
according to the accept headers sent in the request.
7.4.8 407 Proxy Authentication Required
This code is similar to 401 (Unauthorized), but indicates that the
client MUST first authenticate itself with the proxy. The proxy MUST
return a Proxy-Authenticate header field (section 6.26) containing a
challenge applicable to the proxy for the requested resource. The
client MAY repeat the request with a suitable Proxy-Authorization
header field (section 6.27). SIP access authentication is explained
in section 13.2 and 14.
This status code is used for applications where access to the
communication channel (e.g., a telephony gateway) rather than the
callee requires authentication.
7.4.9 408 Request Timeout
The server could not produce a response, e.g., a user location,
within the time indicated in the Expires request-header field. The
client MAY repeat the request without modifications at any later
time.
7.4.10 409 Conflict
The request could not be completed due to a conflict with the current
state of the resource. This response is returned if the action
parameter in a REGISTER request conflicts with existing
registrations.
7.4.11 410 Gone
The requested resource is no longer available at the server and no
forwarding address is known. This condition is expected to be
considered permanent. If the server does not know, or has no facility
to determine, whether or not the condition is permanent, the status
code 404 (Not Found) SHOULD be used instead.
7.4.12 411 Length Required
The server refuses to accept the request without a defined Content-
Length. The client MAY repeat the request if it adds a valid
Content-Length header field containing the length of the message-body
in the request message.
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7.4.13 413 Request Entity Too Large
The server is refusing to process a request because the request
entity is larger than the server is willing or able to process. The
server MAY close the connection to prevent the client from continuing
the request.
If the condition is temporary, the server SHOULD include a Retry-
After header field to indicate that it is temporary and after what
time the client MAY try again.
7.4.14 414 Request-URI Too Long
The server is refusing to service the request because the Request-URI
is longer than the server is willing to interpret.
7.4.15 415 Unsupported Media Type
The server is refusing to service the request because the message
body of the request is in a format not supported by the requested
resource for the requested method. The server SHOULD return a list of
acceptable formats using the Accept, Accept-Encoding and Accept-
Language header fields.
7.4.16 420 Bad Extension
The server did not understand the protocol extension specified in a
Require (Section 6.30) header field.
7.4.17 480 Temporarily Unavailable
The callee's end system was contacted successfully but the callee is
currently unavailable (e.g., not logged in or logged in in such a
manner as to preclude communication with the callee). The response
MAY indicate a better time to call in the Retry-After header. The
user could also be available elsewhere (unbeknownst to this host),
thus, this response does not terminate any searches. The reason
phrase SHOULD indicate a more precise cause as to why the callee is
unavailable. This value SHOULD be setable by the user agent. Status
486 (Busy Here) MAY be used to more precisely indicate a particular
reason for the call failure.
This status is also returned by a redirect server that recognizes the
user identified by the Request-URI, but does not currently have a
valid forwarding location for that user.
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7.4.18 481 Call Leg/Transaction Does Not Exist
This status is returned under two conditions: The server received a
BYE request that does not match any existing call leg or the server
received a CANCEL request that does not match any existing
transaction. (A server simply discards an ACK referring to an unknown
transaction.)
7.4.19 482 Loop Detected
The server received a request with a Via (Section 6.40) path
containing itself.
7.4.20 483 Too Many Hops
The server received a request that contains more Via entries (hops)
(Section 6.40) than allowed by the Max-Forwards (Section 6.23) header
field.
7.4.21 484 Address Incomplete
The server received a request with a To (Section 6.37) address or
Request-URI that was incomplete. Additional information SHOULD be
provided.
This status code allows overlapped dialing. With overlapped
dialing, the client does not know the length of the dialing
string. It sends strings of increasing lengths, prompting
the user for more input, until it no longer receives a 484
status response.
7.4.22 485 Ambiguous
The callee address provided in the request was ambiguous. The
response MAY contain a listing of possible unambiguous addresses in
Contact headers.
Revealing alternatives can infringe on privacy concerns of the user
or the organization. It MUST be possible to configure a server to
respond with status 404 (Not Found) or to suppress the listing of
possible choices if the request address was ambiguous.
Example response to a request with the URL lee@example.com :
485 Ambiguous SIP/2.0
Contact: Carol Lee <sip:carol.lee@example.com>
Handley, et al. Standards Track [Page 81]RFC 2543 SIP: Session Initiation Protocol March 1999
Contact: Ping Lee <sip:p.lee@example.com>
Contact: Lee M. Foote <sip:lee.foote@example.com>
Some email and voice mail systems provide this
functionality. A status code separate from 3xx is used
since the semantics are different: for 300, it is assumed
that the same person or service will be reached by the
choices provided. While an automated choice or sequential
search makes sense for a 3xx response, user intervention is
required for a 485 response.
7.4.23 486 Busy Here
The callee's end system was contacted successfully but the callee is
currently not willing or able to take additional calls. The response
MAY indicate a better time to call in the Retry-After header. The
user could also be available elsewhere, such as through a voice mail
service, thus, this response does not terminate any searches. Status
600 (Busy Everywhere) SHOULD be used if the client knows that no
other end system will be able to accept this call.
7.5 Server Failure 5xx
5xx responses are failure responses given when a server itself has
erred. They are not definitive failures, and MUST NOT terminate a
search if other possible locations remain untried.
7.5.1 500 Server Internal Error
The server encountered an unexpected condition that prevented it from
fulfilling the request. The client MAY display the specific error
condition, and MAY retry the request after several seconds.
7.5.2 501 Not Implemented
The server does not support the functionality required to fulfill the
request. This is the appropriate response when the server does not
recognize the request method and is not capable of supporting it for
any user.
7.5.3 502 Bad Gateway
The server, while acting as a gateway or proxy, received an invalid
response from the downstream server it accessed in attempting to
fulfill the request.
Handley, et al. Standards Track [Page 82]RFC 2543 SIP: Session Initiation Protocol March 1999
7.5.4 503 Service Unavailable
The server is currently unable to handle the request due to a
temporary overloading or maintenance of the server. The implication
is that this is a temporary condition which will be alleviated after
some delay. If known, the length of the delay MAY be indicated in a
Retry-After header. If no Retry-After is given, the client MUST
handle the response as it would for a 500 response.
Note: The existence of the 503 status code does not imply that a
server has to use it when becoming overloaded. Some servers MAY wish
to simply refuse the connection.
7.5.5 504 Gateway Time-out
The server, while acting as a gateway, did not receive a timely
response from the server (e.g., a location server) it accessed in
attempting to complete the request.
7.5.6 505 Version Not Supported
The server does not support, or refuses to support, the SIP protocol
version that was used in the request message. The server is
indicating that it is unable or unwilling to complete the request
using the same major version as the client, other than with this
error message. The response MAY contain an entity describing why that
version is not supported and what other protocols are supported by
that server. The format for such an entity is not defined here and
may be the subject of future standardization.
7.6 Global Failures 6xx
6xx responses indicate that a server has definitive information about
a particular user, not just the particular instance indicated in the
Request-URI. All further searches for this user are doomed to failure
and pending searches SHOULD be terminated.
7.6.1 600 Busy Everywhere
The callee's end system was contacted successfully but the callee is
busy and does not wish to take the call at this time. The response
MAY indicate a better time to call in the Retry-After header. If the
callee does not wish to reveal the reason for declining the call, the
callee uses status code 603 (Decline) instead. This status response
is returned only if the client knows that no other end point (such as
a voice mail system) will answer the request. Otherwise, 486 (Busy
Here) should be returned.
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7.6.2 603 Decline
The callee's machine was successfully contacted but the user
explicitly does not wish to or cannot participate. The response MAY
indicate a better time to call in the Retry-After header.
7.6.3 604 Does Not Exist Anywhere
The server has authoritative information that the user indicated in
the To request field does not exist anywhere. Searching for the user
elsewhere will not yield any results.
7.6.4 606 Not Acceptable
The user's agent was contacted successfully but some aspects of the
session description such as the requested media, bandwidth, or
addressing style were not acceptable.
A 606 (Not Acceptable) response means that the user wishes to
communicate, but cannot adequately support the session described. The
606 (Not Acceptable) response MAY contain a list of reasons in a
Warning header field describing why the session described cannot be
supported. Reasons are listed in Section 6.41. It is hoped that
negotiation will not frequently be needed, and when a new user is
being invited to join an already existing conference, negotiation may
not be possible. It is up to the invitation initiator to decide
whether or not to act on a 606 (Not Acceptable) response.
8 SIP Message Body
8.1 Body Inclusion
Requests MAY contain message bodies unless otherwise noted. Within
this specification, the BYE request MUST NOT contain a message body.
For ACK, INVITE and OPTIONS, the message body is always a session
description. The use of message bodies for REGISTER requests is for
further study.
For response messages, the request method and the response status
code determine the type and interpretation of any message body. All
responses MAY include a body. Message bodies for 1xx responses
contain advisory information about the progress of the request. 2xx
responses to INVITE requests contain session descriptions. In 3xx
responses, the message body MAY contain the description of
alternative destinations or services, as described in Section 7.3.
For responses with status 400 or greater, the message body MAY
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contain additional, human-readable information about the reasons for
failure. It is RECOMMENDED that information in 1xx and 300 and
greater responses be of type text/plain or text/html
8.2 Message Body Type
The Internet media type of the message body MUST be given by the
Content-Type header field. If the body has undergone any encoding
(such as compression) then this MUST be indicated by the Content-
Encoding header field, otherwise Content-Encoding MUST be omitted. If
applicable, the character set of the message body is indicated as
part of the Content-Type header-field value.
8.3 Message Body Length
The body length in bytes SHOULD be given by the Content-Length header
field. Section 6.15 describes the behavior in detail.
The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
(Note: The chunked encoding modifies the body of a message in order
to transfer it as a series of chunks, each with its own size
indicator.)
9 Compact Form
When SIP is carried over UDP with authentication and a complex
session description, it may be possible that the size of a request or
response is larger than the MTU. To address this problem, a more
compact form of SIP is also defined by using abbreviations for the
common header fields listed below:
short field name long field name note
c Content-Type
e Content-Encoding
f From
i Call-ID
m Contact from "moved"
l Content-Length
s Subject
t To
v Via
Thus, the message in section 16.2 could also be written:
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INVITE sip:schooler@vlsi.caltech.edu SIP/2.0
v:SIP/2.0/UDP 131.215.131.131;maddr=239.128.16.254;ttl=16
v:SIP/2.0/UDP 128.16.64.19
f:sip:mjh@isi.edu
t:sip:schooler@cs.caltech.edu
i:62729-27@128.16.64.19
c:application/sdp
CSeq: 4711 INVITE
l:187
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
Clients MAY mix short field names and long field names within the
same request. Servers MUST accept both short and long field names for
requests. Proxies MAY change header fields between their long and
short forms, but this MUST NOT be done to fields following an
Authorization header.
10 Behavior of SIP Clients and Servers
10.1 General Remarks
SIP is defined so it can use either UDP (unicast or multicast) or TCP
as a transport protocol; it provides its own reliability mechanism.
10.1.1 Requests
Servers discard isomorphic requests, but first retransmit the
appropriate response. (SIP requests are said to be idempotent , i.e.,
receiving more than one copy of a request does not change the server
state.)
After receiving a CANCEL request from an upstream client, a stateful
proxy server MAY send a CANCEL on all branches where it has not yet
received a final response.
When a user agent receives a request, it checks the Call-ID against
those of in-progress calls. If the Call-ID was found, it compares the
tag value of To with the user's tag and rejects the request if the
Handley, et al. Standards Track [Page 86]RFC 2543 SIP: Session Initiation Protocol March 1999
two do not match. If the From header, including any tag value,
matches the value for an existing call leg, the server compares the
CSeq header field value. If less than or equal to the current
sequence number, the request is a retransmission. Otherwise, it is a
new request. If the From header does not match an existing call leg,
a new call leg is created.
If the Call-ID was not found, a new call leg is created, with entries
for the To, From and Call-ID headers. In this case, the To header
field should not have contained a tag. The server returns a response
containing the same To value, but with a unique tag added. The tag
MAY be omitted if the request contained only one Via header field.
10.1.2 Responses
A server MAY issue one or more provisional responses at any time
before sending a final response. If a stateful proxy, user agent
server, redirect server or registrar cannot respond to a request with
a final response within 200 ms, it SHOULD issue a provisional (1xx)
response as soon as possible. Stateless proxies MUST NOT issue
provisional responses on their own.
Responses are mapped to requests by the matching To, From, Call-ID,
CSeq headers and the branch parameter of the first Via header.
Responses terminate request retransmissions even if they have Via
headers that cause them to be delivered to an upstream client.
A stateful proxy may receive a response that it does not have state
for, that is, where it has no a record of an associated request. If
the Via header field indicates that the upstream server used TCP, the
proxy actively opens a TCP connection to that address. Thus, proxies
have to be prepared to receive responses on the incoming side of
passive TCP connections, even though most responses will arrive on
the incoming side of an active connection. (An active connection is a
TCP connection initiated by the proxy, a passive connection is one
accepted by the proxy, but initiated by another entity.)
100 responses SHOULD NOT be forwarded, other 1xx responses MAY be
forwarded, possibly after the server eliminates responses with status
codes that had already been sent earlier. 2xx responses are forwarded
according to the Via header. Once a stateful proxy has received a 2xx
response, it MUST NOT forward non-2xx final responses. Responses
with status 300 and higher are retransmitted by each stateful proxy
until the next upstream proxy sends an ACK (see below for timing
details) or CANCEL.
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RFC 2543 SIP: Session Initiation Protocol March 1999
A stateful proxy SHOULD maintain state for at least 32 seconds after
the receipt of the first definitive non-200 response, in order to
handle retransmissions of the response.
The 32 second window is given by the maximum retransmission
duration of 200-class responses using the default timers,
in case the ACK is lost somewhere on the way to the called
user agent or the next stateful proxy.
10.2 Source Addresses, Destination Addresses and Connections
10.2.1 Unicast UDP
Responses are returned to the address listed in the Via header field
(Section 6.40), not the source address of the request.
Recall that responses are not generated by the next-hop
stateless server, but generated by either a proxy server or
the user agent server. Thus, the stateless proxy can only
use the Via header field to forward the response.
10.2.2 Multicast UDP
Requests MAY be multicast; multicast requests likely feature a host-
independent Request-URI. 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.
A client receiving a multicast query does not have to check whether
the host part of the Request-URI matches its own host or domain name.
If the request was received via multicast, the response is also
returned via multicast. Responses to multicast requests are multicast
with the same TTL as the request, where the TTL is derived from the
ttl parameter in the Via header (Section 6.40).
To avoid response implosion, servers MUST NOT answer multicast
requests with a status code other than 2xx or 6xx. The server delays
its response by a random interval uniformly distributed between zero
and one second. Servers MAY suppress responses if they hear a lower-
numbered or 6xx response from another group member prior to sending.
Servers do not respond to CANCEL requests received via multicast to
avoid request implosion. A proxy or UAC SHOULD send a CANCEL on
receiving the first 2xx or 6xx response to a multicast request.
Handley, et al. Standards Track [Page 88]RFC 2543 SIP: Session Initiation Protocol March 1999
Server response suppression is a MAY since it requires a
server to violate some basic message processing rules. Lets
say A sends a multicast request, and it is received by B,C,
and D. B sends a 200 response. The topmost Via field in the
response will contain the address of A. C will also receive
this response, and could use it to suppress its own
response. However, C would normally not examine this
response, as the topmost Via is not its own. Normally, a
response received with an incorrect topmost Via MUST be
dropped, but not in this case. To distinguish this packet
from a misrouted or multicast looped packet is fairly
complex, and for this reason the procedure is a MAY. The
CANCEL, instead, provides a simpler and more standard way
to perform response suppression. It is for this reason that
the use of CANCEL here is a SHOULD
10.3 TCP
A single TCP connection can serve one or more SIP transactions. A
transaction contains zero or more provisional responses followed by
one or more final responses. (Typically, transactions contain exactly
one final response, but there are exceptional circumstances, where,
for example, multiple 200 responses can be generated.)
The client SHOULD keep the connection open at least until the first
final response arrives. If the client closes or resets the TCP
connection prior to receiving the first final response, the server
treats this action as equivalent to a CANCEL request.
This behavior makes it less likely that malfunctioning
clients cause a proxy server to keep connection state
indefinitely.
The server SHOULD NOT close the TCP connection until it has sent its
final response, at which point it MAY close the TCP connection if it
wishes to. However, normally it is the client's responsibility to
close the connection.
If the server leaves the connection open, and if the client so
desires it MAY re-use the connection for further SIP requests or for
requests from the same family of protocols (such as HTTP or stream
control commands).
Handley, et al. Standards Track [Page 89]RFC 2543 SIP: Session Initiation Protocol March 1999
If a server needs to return a response to a client and no longer has
a connection open to that client, it MAY open a connection to the
address listed in the Via header. Thus, a proxy or user agent MUST be
prepared to receive both requests and responses on a "passive"
connection.
10.4 Reliability for BYE, CANCEL, OPTIONS, REGISTER Requests
10.4.1 UDP
A SIP client using UDP SHOULD retransmit a BYE, CANCEL, OPTIONS, or
REGISTER request with an exponential backoff, starting at a T1 second
interval, doubling the interval for each packet, and capping off at a
T2 second interval. This means that after the first packet is sent,
the second is sent T1 seconds later, the next 2*T1 seconds after
that, the next 4*T1 seconds after that, and so on, until the interval
hits T2. Subsequent retransmissions are spaced by T2 seconds. If the
client receives a provisional response, it continues to retransmit
the request, but with an interval of T2 seconds. Retransmissions
cease when the client has sent a total of eleven packets, or receives
a definitive response. Default values for T1 and T2 are 500 ms and 4
s, respectively. Clients MAY use larger values, but SHOULD NOT use
smaller ones. Servers retransmit the response upon receipt of a
request retransmission. After the server sends a final response, it
cannot be sure the client has received the response, and thus SHOULD
cache the results for at least 10*T2 seconds to avoid having to, for
example, contact the user or location server again upon receiving a
request retransmission.
Use of the exponential backoff is for congestion control
purposes. However, the back-off must cap off, since request
retransmissions are used to trigger response
retransmissions at the server. Without a cap, the loss of a
single response could significantly increase transaction
latencies.
The value of the initial retransmission timer is smaller than that
that for TCP since it is expected that network paths suitable for
interactive communications have round-trip times smaller than 500 ms.
For congestion control purposes, the retransmission count has to be
bounded. Given that most transactions are expected to consist of one
request and a few responses, round-trip time estimation is not likely
to be very useful. If RTT estimation is desired to more quickly
discover a missing final response, each request retransmission needs
to be labeled with its own Timestamp (Section 6.36), returned in the
response. The server caches the result until it can be sure that the
client will not retransmit the same request again.
Handley, et al. Standards Track [Page 90]RFC 2543 SIP: Session Initiation Protocol March 1999
Each server in a proxy chain generates its own final response to a
CANCEL request. The server responds immediately upon receipt of the
CANCEL request rather than waiting until it has received final
responses from the CANCEL requests it generates.
BYE and OPTIONS final responses are generated by redirect and user
agent servers; REGISTER final responses are generated by registrars.
Note that in contrast to the reliability mechanism described in
Section 10.5, responses to these requests are not retransmitted
periodically and not acknowledged via ACK.
10.4.2 TCP
Clients using TCP do not need to retransmit requests.
10.5 Reliability for INVITE Requests
Special considerations apply for the INVITE method.
1. After receiving an invitation, considerable time can elapse
before the server can determine the outcome. For example,
if the called party is "rung" or extensive searches are
performed, delays between the request and a definitive
response can reach several tens of seconds. If either
caller or callee are automated servers not directly
controlled by a human being, a call attempt could be
unbounded in time.
2. If a telephony user interface is modeled or if we need to
interface to the PSTN, the caller's user interface will
provide "ringback", a signal that the callee is being
alerted. (The status response 180 (Ringing) MAY be used to
initiate ringback.) Once the callee picks up, the caller
needs to know so that it can enable the voice path and stop
ringback. The callee's response to the invitation could get
lost. Unless the response is transmitted reliably, the
caller will continue to hear ringback while the callee
assumes that the call exists.
3. The client has to be able to terminate an on-going request,
e.g., because it is no longer willing to wait for the
connection or search to succeed. The server will have to
wait several retransmission intervals to interpret the lack
of request retransmissions as the end of a call. If the
call succeeds shortly after the caller has given up, the
callee will "pick up the phone" and not be "connected".
Handley, et al. Standards Track [Page 91]RFC 2543 SIP: Session Initiation Protocol March 1999
10.5.1 UDP
For UDP, A SIP client SHOULD retransmit a SIP INVITE request with an
interval that starts at T1 seconds, and doubles after each packet
transmission. The client ceases retransmissions if it receives a
provisional or definitive response, or once it has sent a total of 7
request packets.
A server which transmits a provisional response should retransmit it
upon reception of a duplicate request. A server which transmits a
final response should retransmit it with an interval that starts at
T1 seconds, and doubles for each subsequent packet. Response
retransmissions cease when any one of the following occurs:
1. An ACK request for the same transaction is received;
2. a BYE request for the same call leg is received;
3. a CANCEL request for the same call leg is received and the
final response status was equal or greater to 300;
4. the response has been transmitted 7 times.
Only the user agent client generates an ACK for 2xx final responses,
If the response contained a Contact header field, the ACK MAY be sent
to the address listed in that Contact header field. If the response
did not contain a Contact header, the client uses the same To header
field and Request-URI as for the INVITE request and sends the ACK to
the same destination as the original INVITE request. ACKs for final
responses other than 2xx are sent to the same server that the
original request was sent to, using the same Request-URI as the
original request. Note, however, that the To header field in the ACK
is copied from the response being acknowledged, not the request, and
thus MAY additionally contain the tag parameter. Also note than
unlike 2xx final responses, a proxy generates an ACK for non-2xx
final responses.
The ACK request MUST NOT be acknowledged to prevent a response-ACK
feedback loop. Fig. 12 and 13 show the client and server state
diagram for invitations.
The mechanism in Sec. 10.4 would not work well for INVITE
because of the long delays between INVITE and a final
response. If the 200 response were to get lost, the callee
would believe the call to exist, but the voice path would
Handley, et al. Standards Track [Page 92]RFC 2543 SIP: Session Initiation Protocol March 1999
+===========+
* *
...........>* Initial *<;;;;;;;;;;
: 7 INVITE * * ;
: sent +===========+ ;
: | ;
: | - ;
: | INVITE ;
: | ;
: v ;
: ************* ;
: T1*2^n <--* * ;
: INVITE -->* Calling *--------+ ;
: * * | ;
: ************* | ;
: : | | ;
:.............: | 1xx xxx | ;
| - ACK | ;
| | ;
v | ;
************* | ;
* * | ;
* Ringing *<->1xx | ;
* * | ;
************* | ;
| | ;
|<-------------+ ;
| ;
v ;
************* ;
xxx <--* * ;
ACK -->* Completed * ;
* * ;
************* ;
; 32s (for proxy);
;;;;;;;;;;;;;;;;;;
event (xxx=status)
message
Figure 12: State transition diagram of client for INVITE method
Handley, et al. Standards Track [Page 93]RFC 2543 SIP: Session Initiation Protocol March 1999
7 pkts sent +===============+
+-------------->* *
| * Initial *<...............
|;;;;;;;;;;;;;;>* * :
|; +===============+ :
|; CANCEL ! :
|; 200 ! INVITE :
|; ! 1xx :
|; ! :
|; v :
|; ***************** BYE :
|; INVITE -->* * 200 :
|; 1xx <--* Call proceed. *..............>:
|; * * :
|;;;;;;;;;;;;;;;***************** :
|; ! ! :
|: ! ! :
|; failure ! ! picks up :
|; >= 300 ! ! 200 :
|; +-------+ +-------+ :
|; v v :
|; *********** *********** :
|;INVITE<* *<T1*2^n->* *>INVITE :
|;status>* failure *>status<-* success *<status :
|; * * * * :
|;;;;;;;;*********** *********** :
| ! : | | ! : :
| ! : | | ! : :
+-------------!-:-+------------+ ! : :
! :.................!..:.........>:
! ! BYE :
+---------+---------+ 200 :
event ! ACK :
message sent v :
***************** :
V---* * :
ACK * Confirmed * :
|-->* * :
***************** .
:......................>:
Figure 13: State transition diagram of server for INVITE method
Handley, et al. Standards Track [Page 94]RFC 2543 SIP: Session Initiation Protocol March 1999
be dead since the caller does not know that the callee has
picked up. Thus, the INVITE retransmission interval would
have to be on the order of a second or two to limit the
duration of this state confusion. Retransmitting the
response with an exponential back-off helps ensure that the
response is received, without placing an undue burden on
the network.
10.5.2 TCP
A user agent using TCP MUST NOT retransmit requests, but uses the
same algorithm as for UDP (Section 10.5.1) to retransmit responses
until it receives an ACK.
It is necessary to retransmit 2xx responses as their
reliability is assured end-to-end only. If the chain of
proxies has a UDP link in the middle, it could lose the
response, with no possibility of recovery. For simplicity,
we also retransmit non-2xx responses, although that is not
strictly necessary.
10.6 Reliability for ACK Requests
The ACK request does not generate responses. It is only generated
when a response to an INVITE request arrives (see Section 10.5). This
behavior is independent of the transport protocol. Note that the ACK
request MAY take a different path than the original INVITE request,
and MAY even cause a new TCP connection to be opened in order to send
it.
10.7 ICMP Handling
Handling of ICMP messages in the case of UDP messages is
straightforward. For requests, a host, network, port, or protocol
unreachable error SHOULD be treated as if a 400-class response was
received. For responses, these errors SHOULD cause the server to
cease retransmitting the response.
Source quench ICMP messages SHOULD be ignored. TTL exceeded errors
SHOULD be ignored. Parameter problem errors SHOULD be treated as if a
400-class response was received.
11 Behavior of SIP User Agents
This section describes the rules for user agent client and servers
for generating and processing requests and responses.
Handley, et al. Standards Track [Page 95]RFC 2543 SIP: Session Initiation Protocol March 1999
11.1 Caller Issues Initial INVITE Request
When a user agent client desires to initiate a call, it formulates an
INVITE request. The To field in the request contains the address of
the callee. The Request-URI contains the same address. The From field
contains the address of the caller. If the From address can appear
in requests generated by other user agent clients for the same call,
the caller MUST insert the tag parameter in the From field. A UAC MAY
optionally add a Contact header containing an address where it would
like to be contacted for transactions from the callee back to the
caller.
11.2 Callee Issues Response
When the initial INVITE request is received at the callee, the callee
can accept, redirect, or reject the call. In all of these cases, it
formulates a response. The response MUST copy the To, From, Call-ID,
CSeq and Via fields from the request. Additionally, the responding
UAS MUST add the tag parameter to the To field in the response if the
request contained more than one Via header field. Since a request
from a UAC may fork and arrive at multiple hosts, the tag parameter
serves to distinguish, at the UAC, multiple responses from different
UAS's. The UAS MAY add a Contact header field in the response. It
contains an address where the callee would like to be contacted for
subsequent transactions, including the ACK for the current INVITE.
The UAS stores the values of the To and From field, including any
tags. These become the local and remote addresses of the call leg,
respectively.
11.3 Caller Receives Response to Initial Request
Multiple responses may arrive at the UAC for a single INVITE request,
due to a forking proxy. Each response is distinguished by the "tag"
parameter in the To header field, and each represents a distinct call
leg. The caller MAY choose to acknowledge or terminate the call with
each responding UAS. To acknowledge, it sends an ACK request, and to
terminate it sends a BYE request. The To header field in the ACK or
BYE MUST be the same as the To field in the 200 response, including
any tag. The From header field MUST be the same as the From header
field in the 200 (OK) response, including any tag. The Request-URI of
the ACK or BYE request MAY be set to whatever address was found in
the Contact header field in the 200 (OK) response, if present.
Alternately, a UAC may copy the address from the To header field into
the Request-URI. The UAC also notes the value of the To and From
header fields in each response. For each call leg, the To header
field becomes the remote address, and the From header field becomes
the local address.
Handley, et al. Standards Track [Page 96]
RFC 2543 SIP: Session Initiation Protocol March 1999
11.4 Caller or Callee Generate Subsequent Requests
Once the call has been established, either the caller or callee MAY
generate INVITE or BYE requests to change or terminate the call.
Regardless of whether the caller or callee is generating the new
request, the header fields in the request are set as follows. For the
desired call leg, the To header field is set to the remote address,
and the From header field is set to the local address (both including
any tags). The Contact header field MAY be different than the Contact
header field sent in a previous response or request. The Request-URI
MAY be set to the value of the Contact header field received in a
previous request or response from the remote party, or to the value
of the remote address.
11.5 Receiving Subsequent Requests
When a request is received subsequently, the following checks are
made:
1. If the Call-ID is new, the request is for a new call,
regardless of the values of the To and From header fields.
2. If the Call-ID exists, the request is for an existing call.
If the To, From, Call-ID, and CSeq values exactly match
(including tags) those of any requests received previously,
the request is a retransmission.
3. If there was no match to the previous step, the To and From
fields are compared against existing call leg local and
remote addresses. If there is a match, and the CSeq in the
request is higher than the last CSeq received on that leg,
the request is a new transaction for an existing call leg.
12 Behavior of SIP Proxy and Redirect Servers
This section describes behavior of SIP redirect and proxy servers in
detail. Proxy servers can "fork" connections, i.e., a single incoming
request spawns several outgoing (client) requests.
12.1 Redirect Server
A redirect server does not issue any SIP requests of its own. After
receiving a request other than CANCEL, the server gathers the list of
alternative locations and returns a final response of class 3xx or it
refuses the request. For well-formed CANCEL requests, it SHOULD
return a 2xx response. This response ends the SIP transaction. The
Handley, et al. Standards Track [Page 97]RFC 2543 SIP: Session Initiation Protocol March 1999
redirect server maintains transaction state for the whole SIP
transaction. It is up to the client to detect forwarding loops
between redirect servers.
12.2 User Agent Server
User agent servers behave similarly to redirect servers, except that
they also accept requests and can return a response of class 2xx.
12.3 Proxy Server
This section outlines processing rules for proxy servers. A proxy
server can either be stateful or stateless. When stateful, a proxy
remembers the incoming request which generated outgoing requests, and
the outgoing requests. A stateless proxy forgets all information once
an outgoing request is generated. A forking proxy SHOULD be stateful.
Proxies that accept TCP connections MUST be stateful.
Otherwise, if the proxy were to lose a request, the TCP
client would never retransmit it.
A stateful proxy SHOULD NOT become stateless until after it sends a
definitive response upstream, and at least 32 seconds after it
received a definitive response.
A stateful proxy acts as a virtual UAS/UAC. It implements the server
state machine when receiving requests, and the client state machine
for generating outgoing requests, with the exception of receiving a
2xx response to an INVITE. Instead of generating an ACK, the 2xx
response is always forwarded upstream towards the caller.
Furthermore, ACK's for 200 responses to INVITE's are always proxied
downstream towards the UAS, as they would be for a stateless proxy.
A stateless proxy does not act as a virtual UAS/UAC (as this would
require state). Rather, a stateless proxy forwards every request it
receives downstream, and every response it receives upstream.
12.3.1 Proxying Requests
To prevent loops, a server MUST check if its own address is already
contained in the Via header field of the incoming request.
The To, From, Call-ID, and Contact tags are copied exactly from the
original request. The proxy SHOULD change the Request-URI to indicate
the server where it intends to send the request.
Handley, et al. Standards Track [Page 98]RFC 2543 SIP: Session Initiation Protocol March 1999
A proxy server always inserts a Via header field containing its own
address into those requests that are caused by an incoming request.
Each proxy MUST insert a "branch" parameter (Section 6.40).
12.3.2 Proxying Responses
A proxy only processes a response if the topmost Via field matches
one of its addresses. A response with a non-matching top Via field
MUST be dropped.
12.3.3 Stateless Proxy: Proxying Responses
A stateless proxy removes its own Via field, and checks the address
in the next Via field. In the case of UDP, the response is sent to
the address listed in the "maddr" tag if present, otherwise to the
"received" tag if present, and finally to the address in the "sent-
by" field. A proxy MUST remain stateful when handling requests
received via TCP.
A stateless proxy MUST NOT generate its own provisional responses.
12.3.4 Stateful Proxy: Receiving Requests
When a stateful proxy receives a request, it checks the To, From
(including tags), Call-ID and CSeq against existing request records.
If the tuple exists, the request is a retransmission. The provisional
or final response sent previously is retransmitted, as per the server
state machine. If the tuple does not exist, the request corresponds
to a new transaction, and the request should be proxied.
A stateful proxy server MAY generate its own provisional (1xx)
responses.
12.3.5 Stateful Proxy: Receiving ACKs
When an ACK request is received, it is either processed locally or
proxied. To make this determination, the To, From, CSeq and Call-ID
fields are compared against those in previous requests. If there is
no match, the ACK request is proxied as if it were an INVITE request.
If there is a match, and if the server had ever sent a 200 response
upstream, the ACK is proxied. If the server had never sent any
responses upstream, the ACK is also proxied. If the server had sent a
3xx, 4xx, 5xx or 6xx response, but no 2xx response, the ACK is
processed locally if the tag in the To field of the ACK matches the
tag sent by the proxy in the response.
Handley, et al. Standards Track [Page 99]
RFC 2543 SIP: Session Initiation Protocol March 1999
12.3.6 Stateful Proxy: Receiving Responses
When a proxy server receives a response that has passed the Via
checks, the proxy server checks the To (without the tag), From
(including the tag), Call-ID and CSeq against values seen in previous
requests. If there is no match, the response is forwarded upstream to
the address listed in the Via field. If there is a match, the
"branch" tag in the Via field is examined. If it matches a known
branch identifier, the response is for the given branch, and
processed by the virtual client for the given branch. Otherwise, the
response is dropped.
A stateful proxy should obey the rules in Section 12.4 to determine
if the response should be proxied upstream. If it is to be proxied,
the same rules for stateless proxies above are followed, with the
following addition for TCP. If a request was received via TCP
(indicated by the protocol in the top Via header), the proxy checks
to see if it has a connection currently open to that address. If so,
the response is sent on that connection. Otherwise, a new TCP
connection is opened to the address and port in the Via field, and
the response is sent there. Note that this implies that a UAC or
proxy MUST be prepared to receive responses on the incoming side of a
TCP connection. Definitive non 200-class responses MUST be
retransmitted by the proxy, even over a TCP connection.
12.3.7 Stateless, Non-Forking Proxy
Proxies in this category issue at most a single unicast request for
each incoming SIP request, that is, they do not "fork" requests.
However, servers MAY choose to always operate in a mode that allows
issuing of several requests, as described in Section 12.4.
The server can forward the request and any responses. It does not
have to maintain any state for the SIP transaction. Reliability is
assured by the next redirect or stateful proxy server in the server
chain.
A proxy server SHOULD cache the result of any address translations
and the response to speed forwarding of retransmissions. After the
cache entry has been expired, the server cannot tell whether an
incoming request is actually a retransmission of an older request.
The server will treat it as a new request and commence another
search.
12.4 Forking Proxy
The server MUST respond to the request immediately with a 100
(Trying) response.
Handley, et al. Standards Track [Page 100]
RFC 2543 SIP: Session Initiation Protocol March 1999
Successful responses to an INVITE request MAY contain a Contact
header field so that the following ACK or BYE bypasses the proxy
search mechanism. If the proxy requires future requests to be routed
through it, it adds a Record-Route header to the request (Section
6.29).
The following C-code describes the behavior of a proxy server issuing
several requests in response to an incoming INVITE request. The
function request(r, a, b) sends a SIP request of type r to address a,
with branch id b. await_response() waits until a response is received
and returns the response. close(a) closes the TCP connection to
client with address a. response(r) sends a response to the client.
ismulticast() returns 1 if the location is a multicast address and
zero otherwise. The variable timeleft indicates the amount of time
left until the maximum response time has expired. The variable
recurse indicates whether the server will recursively try addresses
returned through a 3xx response. A server MAY decide to recursively
try only certain addresses, e.g., those which are within the same
domain as the proxy server. Thus, an initial multicast request can
trigger additional unicast requests.
/* request type */
typedef enum {INVITE, ACK, BYE, OPTIONS, CANCEL, REGISTER} Method;
process_request(Method R, int N, address_t address[])
{
struct {
int branch; /* branch id */
int done; /* has responded */
} outgoing[];
int done[]; /* address has responded */
char *location[]; /* list of locations */
int heard = 0; /* number of sites heard from */
int class; /* class of status code */
int timeleft = 120; /* sample timeout value */
int loc = 0; /* number of locations */
struct { /* response */
int status; /* response: CANCEL=-1 */
int locations; /* number of redirect locations */
char *location[]; /* redirect locations */
address_t a; /* address of respondent */
int branch; /* branch identifier */
} r, best; /* response, best response */
int i;
best.status = 1000;
for (i = 0; i < N; i++) {
Handley, et al. Standards Track [Page 101]RFC 2543 SIP: Session Initiation Protocol March 1999
request(R, address[i], i);
outgoing[i].done = 0;
outgoing[i].branch = i;
}
while (timeleft > 0 && heard < N) {
r = await_response();
class = r.status / 100;
/* If final response, mark branch as done. */
if (class >= 2) {
heard++;
for (i = 0; i < N; i++) {
if (r.branch == outgoing[i].branch) {
outgoing[i].done = 1;
break;
}
}
}
/* CANCEL: respond, fork and wait for responses */
else if (class < 0) {
best.status = 200;
response(best);
for (i = 0; i < N; i++) {
if (!outgoing[i].done)
request(CANCEL, address[i], outgoing[i].branch);
}
best.status = -1;
}
/* Send an ACK */
if (class != 2) {
if (R == INVITE) request(ACK, r.a, r.branch);
}
if (class == 2) {
if (r.status < best.status) best = r;
break;
}
else if (class == 3) {
/* A server MAY optionally recurse. The server MUST check
* whether it has tried this location before and whether
* the location is part of the Via path of the incoming
* request. This check is omitted here for brevity.
* Multicast locations MUST NOT be returned to the client if
* the server is not recursing.
Handley, et al. Standards Track [Page 102]RFC 2543 SIP: Session Initiation Protocol March 1999
*/
if (recurse) {
multicast = 0;
N += r.locations;
for (i = 0; i < r.locations; i++) {
request(R, r.location[i]);
}
} else if (!ismulticast(r.location)) {
best = r;
}
}
else if (class == 4) {
if (best.status >= 400) best = r;
}
else if (class == 5) {
if (best.status >= 500) best = r;
}
else if (class == 6) {
best = r;
break;
}
}
/* We haven't heard anything useful from anybody. */
if (best.status == 1000) {
best.status = 404;
}
if (best.status/100 != 3) loc = 0;
response(best);
}
Responses are processed as follows. The process completes (and state
can be freed) when all requests have been answered by final status
responses (for unicast) or 60 seconds have elapsed (for multicast). A
proxy MAY send a CANCEL to all branches and return a 408 (Timeout) to
the client after 60 seconds or more.
1xx: The proxy MAY forward the response upstream towards the client.
2xx: The proxy MUST forward the response upstream towards the client,
without sending an ACK downstream. After receiving a 2xx, the
server MAY terminate all other pending requests by sending a
CANCEL request and closing the TCP connection, if applicable.
(Terminating pending requests is advisable as searches consume
resources. Also, INVITE requests could "ring" on a number of
workstations if the callee is currently logged in more than
once.)
Handley, et al. Standards Track [Page 103]RFC 2543 SIP: Session Initiation Protocol March 1999
3xx: The proxy MUST send an ACK and MAY recurse on the listed Contact
addresses. Otherwise, the lowest-numbered response is returned
if there were no 2xx responses.
Location lists are not merged as that would prevent
forwarding of authenticated responses. Also, responses can
have message bodies, so that merging is not feasible.
4xx, 5xx: The proxy MUST send an ACK and remember the response if it
has a lower status code than any previous 4xx and 5xx responses.
On completion, the lowest-numbered response is returned if there
were no 2xx or 3xx responses.
6xx: The proxy MUST forward the response to the client and send an
ACK. Other pending requests MAY be terminated with CANCEL as
described for 2xx responses.
A proxy server forwards any response for Call-IDs for which it does
not have a pending transaction according to the response's Via
header. User agent servers respond to BYE requests for unknown call
legs with status code 481 (Transaction Does Not Exist); they drop ACK
requests with unknown call legs silently.
Special considerations apply for choosing forwarding destinations for
ACK and BYE requests. In most cases, these requests will bypass
proxies and reach the desired party directly, keeping proxies from
having to make forwarding decisions.
A proxy MAY maintain call state for a period of its choosing. If a
proxy still has list of destinations that it forwarded the last
INVITE to, it SHOULD direct ACK requests only to those downstream
servers.
13 Security Considerations
13.1 Confidentiality and Privacy: Encryption
13.1.1 End-to-End Encryption
SIP requests and responses can contain sensitive information about
the communication patterns and communication content of individuals.
The SIP message body MAY also contain encryption keys for the session
itself. SIP supports three complementary forms of encryption to
protect privacy:
o End-to-end encryption of the SIP message body and certain
sensitive header fields;
Handley, et al. Standards Track [Page 104]RFC 2543 SIP: Session Initiation Protocol March 1999
o hop-by-hop encryption to prevent eavesdropping that tracks
who is calling whom;
o hop-by-hop encryption of Via fields to hide the route a
request has taken.
Not all of the SIP request or response can be encrypted end-to-end
because header fields such as To and Via need to be visible to
proxies so that the SIP request can be routed correctly. Hop-by-hop
encryption encrypts the entire SIP request or response on the wire so
that packet sniffers or other eavesdroppers cannot see who is calling
whom. Hop-by-hop encryption can also encrypt requests and responses
that have been end-to-end encrypted. Note that proxies can still see
who is calling whom, and this information is also deducible by
performing a network traffic analysis, so this provides a very
limited but still worthwhile degree of protection.
SIP Via fields are used to route a response back along the path taken
by the request and to prevent infinite request loops. However, the
information given by them can also provide useful information to an
attacker. Section 6.22 describes how a sender can request that Via
fields be encrypted by cooperating proxies without compromising the
purpose of the Via field.
End-to-end encryption relies on keys shared by the two user agents
involved in the request. Typically, the message is sent encrypted
with the public key of the recipient, so that only that recipient can
read the message. All implementations SHOULD support PGP-based
encryption [33] and MAY implement other schemes.
A SIP request (or response) is end-to-end encrypted by splitting the
message to be sent into a part to be encrypted and a short header
that will remain in the clear. Some parts of the SIP message, namely
the request line, the response line and certain header fields marked
with "n" in the "enc." column in Table 4 and 5 need to be read and
returned by proxies and thus MUST NOT be encrypted end-to-end.
Possibly sensitive information that needs to be made available as
plaintext include destination address (To) and the forwarding path
(Via) of the call. The Authorization header field MUST remain in the
clear if it contains a digital signature as the signature is
generated after encryption, but MAY be encrypted if it contains
"basic" or "digest" authentication. The From header field SHOULD
normally remain in the clear, but MAY be encrypted if required, in
which case some proxies MAY return a 401 (Unauthorized) status if
they require a From field.
Handley, et al. Standards Track [Page 105]RFC 2543 SIP: Session Initiation Protocol March 1999
Other header fields MAY be encrypted or MAY travel in the clear as
desired by the sender. The Subject, Allow and Content-Type header
fields will typically be encrypted. The Accept, Accept-Language,
Date, Expires, Priority, Require, Call-ID, Cseq, and Timestamp header
fields will remain in the clear.
All fields that will remain in the clear MUST precede those that will
be encrypted. The message is encrypted starting with the first
character of the first header field that will be encrypted and
continuing through to the end of the message body. If no header
fields are to be encrypted, encrypting starts with the second CRLF
pair after the last header field, as shown below. Carriage return and
line feed characters have been made visible as "$", and the encrypted
part of the message is outlined.
INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
Via: SIP/2.0/UDP 169.130.12.5$
To: T. A. Watson <sip:watson@bell-telephone.com>$
From: A. Bell <sip:a.g.bell@bell-telephone.com>$
Encryption: PGP version=5.0$
Content-Length: 224$
Call-ID: 187602141351@worcester.bell-telephone.com$
CSeq: 488$
$
*******************************************************
* Subject: Mr. Watson, come here.$ *
* Content-Type: application/sdp$ *
* $ *
* v=0$ *
* o=bell 53655765 2353687637 IN IP4 128.3.4.5$ *
* c=IN IP4 135.180.144.94$ *
* m=audio 3456 RTP/AVP 0 3 4 5$ *
*******************************************************
An Encryption header field MUST be added to indicate the encryption
mechanism used. A Content-Length field is added that indicates the
length of the encrypted body. The encrypted body is preceded by a
blank line as a normal SIP message body would be.
Upon receipt by the called user agent possessing the correct
decryption key, the message body as indicated by the Content-Length
field is decrypted, and the now-decrypted body is appended to the
clear-text header fields. There is no need for an additional
Content-Length header field within the encrypted body because the
length of the actual message body is unambiguous after decryption.
Handley, et al. Standards Track [Page 106]RFC 2543 SIP: Session Initiation Protocol March 1999
Had no SIP header fields required encryption, the message would have
been as below. Note that the encrypted body MUST then include a blank
line (start with CRLF) to disambiguate between any possible SIP
header fields that might have been present and the SIP message body.
INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
Via: SIP/2.0/UDP 169.130.12.5$
To: T. A. Watson <sip:watson@bell-telephone.com>$
From: A. Bell <a.g.bell@bell-telephone.com>$
Encryption: PGP version=5.0$
Content-Type: application/sdp$
Content-Length: 107$
$
*************************************************
* $ *
* v=0$ *
* o=bell 53655765 2353687637 IN IP4 128.3.4.5$ *
* c=IN IP4 135.180.144.94$ *
* m=audio 3456 RTP/AVP 0 3 4 5$ *
*************************************************
13.1.2 Privacy of SIP Responses
SIP requests can be sent securely using end-to-end encryption and
authentication to a called user agent that sends an insecure
response. This is allowed by the SIP security model, but is not a
good idea. However, unless the correct behavior is explicit, it
would not always be possible for the called user agent to infer what
a reasonable behavior was. Thus when end-to-end encryption is used by
the request originator, the encryption key to be used for the
response SHOULD be specified in the request. If this were not done,
it might be possible for the called user agent to incorrectly infer
an appropriate key to use in the response. Thus, to prevent key-
guessing becoming an acceptable strategy, we specify that a called
user agent receiving a request that does not specify a key to be used
for the response SHOULD send that response unencrypted.
Any SIP header fields that were encrypted in a request SHOULD also be
encrypted in an encrypted response. Contact response fields MAY be
encrypted if the information they contain is sensitive, or MAY be
left in the clear to permit proxies more scope for localized
searches.
Handley, et al. Standards Track [Page 107]RFC 2543 SIP: Session Initiation Protocol March 1999
13.1.3 Encryption by Proxies
Normally, proxies are not allowed to alter end-to-end header fields
and message bodies. Proxies MAY, however, encrypt an unsigned request
or response with the key of the call recipient.
Proxies need to encrypt a SIP request if the end system
cannot perform encryption or to enforce organizational
security policies.
13.1.4 Hop-by-Hop Encryption
SIP requests and responses MAY also be protected by security
mechanisms at the transport or network layer. No particular mechanism
is defined or recommended here. Two possibilities are IPSEC [34] or
TLS [35]. The use of a particular mechanism will generally need to be
specified out of band, through manual configuration, for example.
13.1.5 Via field encryption
When Via header fields are to be hidden, a proxy that receives a
request containing an appropriate "Hide: hop" header field (as
specified in section 6.22) SHOULD encrypt the header field. As only
the proxy that encrypts the field will decrypt it, the algorithm
chosen is entirely up to the proxy implementor. Two methods satisfy
these requirements:
o The server keeps a cache of Via header fields and the
associated To header field, and replaces the Via header field
with an index into the cache. On the reverse path, take the
Via header field from the cache rather than the message.
This is insufficient to prevent message looping, and so an
additional ID MUST be added so that the proxy can detect loops.
This SHOULD NOT normally be the address of the proxy as the goal
is to hide the route, so instead a sufficiently large random
number SHOULD be used by the proxy and maintained in the cache.
It is possible for replies to get directed to the wrong
originator if the cache entry gets reused, so great care needs
to be taken to ensure this does not happen.
o The server MAY use a secret key to encrypt the Via field, a
timestamp and an appropriate checksum in any such message with
the same secret key. The checksum is needed to detect whether
successful decoding has occurred, and the timestamp is
Handley, et al. Standards Track [Page 108]RFC 2543 SIP: Session Initiation Protocol March 1999
required to prevent possible replay attacks and to ensure that
no two requests from the same previous hop have the same
encrypted Via field. This is the preferred solution.
13.2 Message Integrity and Access Control: Authentication
Protective measures need to be taken to prevent an active attacker
from modifying and replaying SIP requests and responses. The same
cryptographic measures that are used to ensure the authenticity of
the SIP message also serve to authenticate the originator of the
message. However, the "basic" and "digest" authentication mechanism
offer authentication only, without message integrity.
Transport-layer or network-layer authentication MAY be used for hop-
by-hop authentication. SIP also extends the HTTP WWW-Authenticate
(Section 6.42) and Authorization (Section 6.11) header field and
their Proxy counterparts to include cryptographically strong
signatures. SIP also supports the HTTP "basic" and "digest" schemes
(see Section 14) and other HTTP authentication schemes to be defined
that offer a rudimentary mechanism of ascertaining the identity of
the caller.
Since SIP requests are often sent to parties with which no
prior communication relationship has existed, we do not
specify authentication based on shared secrets.
SIP requests MAY be authenticated using the Authorization header
field to include a digital signature of certain header fields, the
request method and version number and the payload, none of which are
modified between client and called user agent. The Authorization
header field is used in requests to authenticate the request
originator end-to-end to proxies and the called user agent, and in
responses to authenticate the called user agent or proxies returning
their own failure codes. If required, hop-by-hop authentication can
be provided, for example, by the IPSEC Authentication Header.
SIP does not dictate which digital signature scheme is used for
authentication, but does define how to provide authentication using
PGP in Section 15. As indicated above, SIP implementations MAY also
use "basic" and "digest" authentication and other authentication
mechanisms defined for HTTP. Note that "basic" authentication has
severe security limitations. The following does not apply to these
schemes.
To cryptographically sign a SIP request, the order of the SIP header
fields is important. When an Authorization header field is present,
it indicates that all header fields following the Authorization
Handley, et al. Standards Track [Page 109]RFC 2543 SIP: Session Initiation Protocol March 1999
header field have been included in the signature. Therefore, hop-
by-hop header fields which MUST or SHOULD be modified by proxies MUST
precede the Authorization header field as they will generally be
modified or added-to by proxy servers. Hop-by-hop header fields
which MAY be modified by a proxy MAY appear before or after the
Authorization header. When they appear before, they MAY be modified
by a proxy. When they appear after, they MUST NOT be modified by a
proxy. To sign a request, a client constructs a message from the
request method (in upper case) followed, without LWS, by the SIP
version number, followed, again without LWS, by the request headers
to be signed and the message body. The message thus constructed is
then signed.
For example, if the SIP request is to be:
INVITE sip:watson@boston.bell-telephone.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5
Authorization: PGP version=5.0, signature=...
From: A. Bell <sip:a.g.bell@bell-telephone.com>
To: T. A. Watson <sip:watson@bell-telephone.com>
Call-ID: 187602141351@worcester.bell-telephone.com
Subject: Mr. Watson, come here.
Content-Type: application/sdp
Content-Length: ...
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5
Then the data block that is signed is:
INVITESIP/2.0From: A. Bell <sip:a.g.bell@bell-telephone.com>
To: T. A. Watson <sip:watson@bell-telephone.com>
Call-ID: 187602141351@worcester.bell-telephone.com
Subject: Mr. Watson, come here.
Content-Type: application/sdp
Content-Length: ...
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5
Handley, et al. Standards Track [Page 110]
RFC 2543 SIP: Session Initiation Protocol March 1999
Clients wishing to authenticate requests MUST construct the portion
of the message below the Authorization header using a canonical form.
This allows a proxy to parse the message, take it apart, and
reconstruct it, without causing an authentication failure due to
extra white space, for example. Canonical form consists of the
following rules:
o No short form header fields
o Header field names are capitalized as shown in this document
o No white space between the header name and the colon
o A single space after the colon
o Line termination with a CRLF
o No line folding
o No comma separated lists of header values; each must appear
as a separate header
o Only a single SP between tokens, between tokens and quoted
strings, and between quoted strings; no SP after last token or
quoted string
o No LWS between tokens and separators, except as described
above for after the colon in header fields
Note that if a message is encrypted and authenticated using a digital
signature, when the message is generated encryption is performed
before the digital signature is generated. On receipt, the digital
signature is checked before decryption.
A client MAY require that a server sign its response by including a
Require: org.ietf.sip.signed-response request header field. The
client indicates the desired authentication method via the WWW-
Authenticate header.
The correct behavior in handling unauthenticated responses to a
request that requires authenticated responses is described in section
13.2.1.
Handley, et al. Standards Track [Page 111]RFC 2543 SIP: Session Initiation Protocol March 1999
13.2.1 Trusting responses
There is the possibility that an eavesdropper listens to requests and
then injects unauthenticated responses that terminate, redirect or
otherwise interfere with a call. (Even encrypted requests contain
enough information to fake a response.)
Clients need to be particularly careful with 3xx redirection
responses. Thus a client receiving, for example, a 301 (Moved
Permanently) which was not authenticated when the public key of the
called user agent is known to the client, and authentication was
requested in the request SHOULD be treated as suspicious. The correct
behavior in such a case would be for the called-user to form a dated
response containing the Contact field to be used, to sign it, and
give this signed stub response to the proxy that will provide the
redirection. Thus the response can be authenticated correctly. A
client SHOULD NOT automatically redirect such a request to the new
location without alerting the user to the authentication failure
before doing so.
Another problem might be responses such as 6xx failure responses
which would simply terminate a search, or "4xx" and "5xx" response
failures.
If TCP is being used, a proxy SHOULD treat 4xx and 5xx responses as
valid, as they will not terminate a search. However, fake 6xx
responses from a rogue proxy terminate a search incorrectly. 6xx
responses SHOULD be authenticated if requested by the client, and
failure to do so SHOULD cause such a client to ignore the 6xx
response and continue a search.
With UDP, the same problem with 6xx responses exists, but also an
active eavesdropper can generate 4xx and 5xx responses that might
cause a proxy or client to believe a failure occurred when in fact it
did not. Typically 4xx and 5xx responses will not be signed by the
called user agent, and so there is no simple way to detect these
rogue responses. This problem is best prevented by using hop-by-hop
encryption of the SIP request, which removes any additional problems
that UDP might have over TCP.
These attacks are prevented by having the client require response
authentication and dropping unauthenticated responses. A server user
agent that cannot perform response authentication responds using the
normal Require response of 420 (Bad Extension).
Handley, et al. Standards Track [Page 112]
RFC 2543 SIP: Session Initiation Protocol March 1999
13.3 Callee Privacy
User location and SIP-initiated calls can violate a callee's privacy.
An implementation SHOULD be able to restrict, on a per-user basis,
what kind of location and availability information is given out to
certain classes of callers.
13.4 Known Security Problems
With either TCP or UDP, a denial of service attack exists by a rogue
proxy sending 6xx responses. Although a client SHOULD choose to
ignore such responses if it requested authentication, a proxy cannot
do so. It is obliged to forward the 6xx response back to the client.
The client can then ignore the response, but if it repeats the
request it will probably reach the same rogue proxy again, and the
process will repeat.
14 SIP Authentication using HTTP Basic and Digest Schemes
SIP implementations MAY use HTTP's basic and digest authentication
mechanisms to provide a rudimentary form of security. This section
overviews usage of these mechanisms in SIP. The basic operation is
almost completely identical to that for HTTP [36]. This section
outlines this operation, pointing to [36] for details, and noting the
differences when used in SIP.
14.1 Framework
The framework for SIP authentication parallels that for HTTP [36]. In
particular, the BNF for auth-scheme, auth-param, challenge, realm,
realm-value, and credentials is identical. The 401 response is used
by user agent servers in SIP to challenge the authorization of a user
agent client. Additionally, registrars and redirect servers MAY make
use of 401 responses for authorization, but proxies MUST NOT, and
instead MAY use the 407 response. The requirements for inclusion of
the Proxy-Authenticate, Proxy-Authorization, WWW-Authenticate, and
Authorization in the various messages is identical to [36].
Since SIP does not have the concept of a canonical root URL, the
notion of protections spaces are interpreted differently for SIP. The
realm is a protection domain for all SIP URIs with the same value for
the userinfo, host and port part of the SIP Request-URI. For example:
INVITE sip:alice.wonderland@example.com SIP/2.0
WWW-Authenticate: Basic realm="business"
Handley, et al. Standards Track [Page 113]RFC 2543 SIP: Session Initiation Protocol March 1999
and
INVITE sip:aw@example.com SIP/2.0
WWW-Authenticate: Basic realm="business"
define different protection realms according to this rule.
When a UAC resubmits a request with its credentials after receiving a
401 or 407 response, it MUST increment the CSeq header field as it
would normally do when sending an updated request.
14.2 Basic Authentication
The rules for basic authentication follow those defined in [36], but
with the words "origin server" replaced with "user agent server,
redirect server , or registrar".
Since SIP URIs are not hierarchical, the paragraph in [36] that
states that "all paths at or deeper than the depth of the last
symbolic element in the path field of the Request-URI also are within
the protection space specified by the Basic realm value of the
current challenge" does not apply for SIP. SIP clients MAY
preemptively send the corresponding Authorization header with
requests for SIP URIs within the same protection realm (as defined
above) without receipt of another challenge from the server.
14.3 Digest Authentication
The rules for digest authentication follow those defined in [36],
with "HTTP 1.1" replaced by "SIP/2.0" in addition to the following
differences:
1. The URI included in the challenge has the following BNF:
URI = SIP-URL
2. The BNF for digest-uri-value is:
digest-uri-value = Request-URI ; a defined in Section
4.3
Handley, et al. Standards Track [Page 114]RFC 2543 SIP: Session Initiation Protocol March 1999
3. The example procedure for choosing a nonce based on Etag
does not work for SIP.
4. The Authentication-Info and Proxy-Authentication-Info
fields are not used in SIP.
5. The text in [36] regarding cache operation does not apply
to SIP.
6. [36] requires that a server check that the URI in the
request line, and the URI included in the Authorization
header, point to the same resource. In a SIP context, these
two URI's may actually refer to different users, due to
forwarding at some proxy. Therefore, in SIP, a server MAY
check that the request-uri in the Authorization header
corresponds to a user that the server is willing to accept
forwarded or direct calls for.
14.4 Proxy-Authentication
The use of the Proxy-Authentication and Proxy-Authorization parallel
that as described in [36], with one difference. Proxies MUST NOT add
the Proxy-Authorization header. 407 responses MUST be forwarded
upstream towards the client following the procedures for any other
response. It is the client's responsibility to add the Proxy-
Authorization header containing credentials for the proxy which has
asked for authentication.
If a proxy were to resubmit a request with a Proxy-
Authorization header field, it would need to increment the
CSeq in the new request. However, this would mean that the
UAC which submitted the original request would discard a
response from the UAS, as the CSeq value would be
different.
See sections 6.26 and 6.27 for additional information on usage of
these fields as they apply to SIP.
15 SIP Security Using PGP
15.1 PGP Authentication Scheme
The "pgp" authentication scheme is based on the model that the client
authenticates itself with a request signed with the client's private
key. The server can then ascertain the origin of the request if it
has access to the public key, preferably signed by a trusted third
party.
Handley, et al. Standards Track [Page 115]RFC 2543 SIP: Session Initiation Protocol March 1999
15.1.1 The WWW-Authenticate Response Header
WWW-Authenticate = "WWW-Authenticate" ":" "pgp" pgp-challenge
pgp-challenge = * (";" pgp-params )
pgp-params = realm | pgp-version | pgp-algorithm | nonce
realm = "realm" "=" realm-value
realm-value = quoted-string
pgp-version = "version" "="
<"> digit *( "." digit ) *letter <">
pgp-algorithm = "algorithm" "=" ( "md5" | "sha1" | token )
nonce = "nonce" "=" nonce-value
nonce-value = quoted-string
The meanings of the values of the parameters used above are as
follows:
realm: A string to be displayed to users so they know which identity
to use. This string SHOULD contain at least the name of the host
performing the authentication and MAY additionally indicate the
collection of users who might have access. An example might be "
Users with call-out privileges ".
pgp-algorithm: The value of this parameter indicates the PGP message
integrity check (MIC) to be used to produce the signature. If
this not present it is assumed to be "md5". The currently
defined values are "md5" for the MD5 checksum, and "sha1" for
the SHA.1 algorithm.
pgp-version: The version of PGP that the client MUST use. Common
values are "2.6.2" and "5.0". The default is 5.0.
nonce: A server-specified data string which should be uniquely
generated each time a 401 response is made. It is RECOMMENDED
that this string be base64 or hexadecimal data. Specifically,
since the string is passed in the header lines as a quoted
string, the double-quote character is not allowed. The contents
of the nonce are implementation dependent. The quality of the
implementation depends on a good choice. Since the nonce is used
only to prevent replay attacks and is signed, a time stamp in
units convenient to the server is sufficient.
Handley, et al. Standards Track [Page 116]RFC 2543 SIP: Session Initiation Protocol March 1999
Replay attacks within the duration of the call setup are of
limited interest, so that timestamps with a resolution of a
few seconds are often should be sufficient. In that case,
the server does not have to keep a record of the nonces.
Example:
WWW-Authenticate: pgp ;version="5.0"
;realm="Your Startrek identity, please" ;algorithm=md5
;nonce="913082051"
15.1.2 The Authorization Request Header
The client is expected to retry the request, passing an Authorization
header line, which is defined as follows.
Authorization = "Authorization" ":" "pgp" *( ";" pgp-response )
pgp-response = realm | pgp-version | pgp-signature
| signed-by | nonce
pgp-signature = "signature" "=" quoted-string
signed-by = "signed-by" "=" <"> URI <">
The client MUST increment the CSeq header before resubmitting the
request. The signature MUST correspond to the From header of the
request unless the signed-by parameter is provided.
pgp-signature: The PGP ASCII-armored signature [33], as it appears
between the "BEGIN PGP MESSAGE" and "END PGP MESSAGE"
delimiters, without the version indication. The signature is
included without any linebreaks.
The signature is computed across the nonce (if present), request
method, request version and header fields following the Authorization
header and the message body, in the same order as they appear in the
message. The request method and version are prepended to the header
fields without any white space. The signature is computed across the
headers as sent, and the terminating CRLF. The CRLF following the
Authorization header is NOT included in the signature.
A server MAY be configured not to generate nonces only if replay
attacks are not a concern.
Handley, et al. Standards Track [Page 117]RFC 2543 SIP: Session Initiation Protocol March 1999
Not generating nonces avoids the additional set of request,
401 response and possibly ACK messages and reduces delay by
one round-trip time.
Using the ASCII-armored version is about 25% less space-
efficient than including the binary signature, but it is
significantly easier for the receiver to piece together.
Versions of the PGP program always include the full
(compressed) signed text in their output unless ASCII-
armored mode ( -sta ) is specified. Typical signatures are
about 200 bytes long. -- The PGP signature mechanism allows
the client to simply pass the request to an external PGP
program. This relies on the requirement that proxy servers
are not allowed to reorder or change header fields.
realm: The realm is copied from the corresponding WWW-Authenticate
header field parameter.
signed-by: If and only if the request was not signed by the entity
listed in the From header, the signed-by header indicates the
name of the signing entity, expressed as a URI.
Receivers of signed SIP messages SHOULD discard any end-to-end header
fields above the Authorization header, as they may have been
maliciously added en route by a proxy.
Example:
Authorization: pgp version="5.0"
;realm="Your Startrek identity, please"
;nonce="913082051"
;signature="iQB1AwUBNNJiUaYBnHmiiQh1AQFYsgL/Wt3dk6TWK81/b0gcNDf
VAUGU4rhEBW972IPxFSOZ94L1qhCLInTPaqhHFw1cb3lB01rA0RhpV4t5yCdUt
SRYBSkOK29o5e1KlFeW23EzYPVUm2TlDAhbcjbMdfC+KLFX
=aIrx"
15.2 PGP Encryption Scheme
The PGP encryption scheme uses the following syntax:
Encryption = "Encryption" ":" "pgp" pgp-eparams
pgp-eparams = 1# ( pgp-version | pgp-encoding )
pgp-encoding = "encoding" "=" "ascii" | token
Handley, et al. Standards Track [Page 118]RFC 2543 SIP: Session Initiation Protocol March 1999
encoding: Describes the encoding or "armor" used by PGP. The value
"ascii" refers to the standard PGP ASCII armor, without the
lines containing "BEGIN PGP MESSAGE" and "END PGP MESSAGE" and
without the version identifier. By default, the encrypted part
is included as binary.
Example:
Encryption: pgp version="2.6.2", encoding="ascii"
15.3 Response-Key Header Field for PGP
Response-Key = "Response-Key" ":" "pgp" pgp-eparams
pgp-eparams = 1# ( pgp-version | pgp-encoding | pgp-key)
pgp-key = "key" "=" quoted-string
If ASCII encoding has been requested via the encoding parameter, the
key parameter contains the user's public key as extracted from the
pgp key ring with the "pgp -kxa user ".
Example:
Response-Key: pgp version="2.6.2", encoding="ascii",
key="mQBtAzNWHNYAAAEDAL7QvAdK2utY05wuUG+ItYK5tCF8HNJM60sU4rLaV+eUnkMk
mOmJWtc2wXcZx1XaXb2lkydTQOesrUR75IwNXBuZXPEIMThEa5WLsT7VLme7njnx
sE86SgWmAZx5ookIdQAFEbQxSGVubmluZyBTY2h1bHpyaW5uZSA8c2NodWx6cmlu
bmVAY3MuY29sdW1iaWEuZWR1Pg==
=+y19"
16 Examples
In the following examples, we often omit the message body and the
corresponding Content-Length and Content-Type headers for brevity.
16.1 Registration
A user at host saturn.bell-tel.com registers on start-up, via
multicast, with the local SIP server named bell-tel.com. In the
example, the user agent on saturn expects to receive SIP requests on
UDP port 3890.
Handley, et al. Standards Track [Page 119]RFC 2543 SIP: Session Initiation Protocol March 1999
C->S: REGISTER sip:bell-tel.com SIP/2.0
Via: SIP/2.0/UDP saturn.bell-tel.com
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com
Call-ID: 70710@saturn.bell-tel.com
CSeq: 1 REGISTER
Contact: <sip:watson@saturn.bell-tel.com:3890;transport=udp>
Expires: 7200
The registration expires after two hours. Any future invitations for
watson@bell-tel.com arriving at sip.bell-tel.com will now be
redirected to watson@saturn.bell-tel.com, UDP port 3890.
If Watson wants to be reached elsewhere, say, an on-line service he
uses while traveling, he updates his reservation after first
cancelling any existing locations:
C->S: REGISTER sip:bell-tel.com SIP/2.0
Via: SIP/2.0/UDP saturn.bell-tel.com
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com
Call-ID: 70710@saturn.bell-tel.com
CSeq: 2 REGISTER
Contact: *
Expires: 0
C->S: REGISTER sip:bell-tel.com SIP/2.0
Via: SIP/2.0/UDP saturn.bell-tel.com
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com
Call-ID: 70710@saturn.bell-tel.com
CSeq: 3 REGISTER
Contact: sip:tawatson@example.com
Now, the server will forward any request for Watson to the server at
example.com, using the Request-URI tawatson@example.com. For the
server at example.com to reach Watson, he will need to send a
REGISTER there, or inform the server of his current location through
some other means.
It is possible to use third-party registration. Here, the secretary
jon.diligent registers his boss, T. Watson:
Handley, et al. Standards Track [Page 120]RFC 2543 SIP: Session Initiation Protocol March 1999
C->S: REGISTER sip:bell-tel.com SIP/2.0
Via: SIP/2.0/UDP pluto.bell-tel.com
From: sip:jon.diligent@bell-tel.com
To: sip:watson@bell-tel.com
Call-ID: 17320@pluto.bell-tel.com
CSeq: 1 REGISTER
Contact: sip:tawatson@example.com
The request could be sent to either the registrar at bell-tel.com or
the server at example.com. In the latter case, the server at
example.com would proxy the request to the address indicated in the
Request-URI. Then, Max-Forwards header could be used to restrict the
registration to that server.
16.2 Invitation to a Multicast Conference
The first example invites schooler@vlsi.cs.caltech.edu to a multicast
session. All examples use the Session Description Protocol (SDP) (RFC
2327 [6]) as the session description format.
16.2.1 Request
C->S: INVITE sip:schooler@cs.caltech.edu SIP/2.0
Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
;maddr=239.128.16.254;ttl=16
Via: SIP/2.0/UDP north.east.isi.edu
From: Mark Handley <sip:mjh@isi.edu>
To: Eve Schooler <sip:schooler@caltech.edu>
Call-ID: 2963313058@north.east.isi.edu
CSeq: 1 INVITE
Subject: SIP will be discussed, too
Content-Type: application/sdp
Content-Length: 187
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
Handley, et al. Standards Track [Page 121]RFC 2543 SIP: Session Initiation Protocol March 1999
The From request header above states that the request was initiated
by mjh@isi.edu and addressed to schooler@caltech.edu (From header
fields). The Via fields list the hosts along the path from invitation
initiator (the last element of the list) towards the callee. In the
example above, the message was last multicast to the administratively
scoped group 239.128.16.254 with a ttl of 16 from the host
csvax.cs.caltech.edu. The second Via header field indicates that it
was originally sent from the host north.east.isi.edu. The Request-URI
indicates that the request is currently being being addressed to
schooler@cs.caltech.edu, the local address that csvax looked up for
the callee.
In this case, the session description is using the Session
Description Protocol (SDP), as stated in the Content-Type header.
The header is terminated by an empty line and is followed by a
message body containing the session description.
16.2.2 Response
The called user agent, directly or indirectly through proxy servers,
indicates that it is alerting ("ringing") the called party:
S->C: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
;maddr=239.128.16.254;ttl=16
Via: SIP/2.0/UDP north.east.isi.edu
From: Mark Handley <sip:mjh@isi.edu>
To: Eve Schooler <sip:schooler@caltech.edu> ;tag=9883472
Call-ID: 2963313058@north.east.isi.edu
CSeq: 1 INVITE
A sample response to the invitation is given below. The first line of
the response states the SIP version number, that it is a 200 (OK)
response, which means the request was successful. The Via headers are
taken from the request, and entries are removed hop by hop as the
response retraces the path of the request. A new authentication field
MAY be added by the invited user's agent if required. The Call-ID is
taken directly from the original request, along with the remaining
fields of the request message. The original sense of From field is
preserved (i.e., it is the session initiator).
In addition, the Contact header gives details of the host where the
user was located, or alternatively the relevant proxy contact point
which should be reachable from the caller's host.
Handley, et al. Standards Track [Page 122]RFC 2543 SIP: Session Initiation Protocol March 1999
S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
;maddr=239.128.16.254;ttl=16
Via: SIP/2.0/UDP north.east.isi.edu
From: Mark Handley <sip:mjh@isi.edu>
To: Eve Schooler <sip:schooler@caltech.edu> ;tag=9883472
Call-ID: 2963313058@north.east.isi.edu
CSeq: 1 INVITE
Contact: sip:es@jove.cs.caltech.edu
The caller confirms the invitation by sending an ACK request to the
location named in the Contact header:
C->S: ACK sip:es@jove.cs.caltech.edu SIP/2.0
Via: SIP/2.0/UDP north.east.isi.edu
From: Mark Handley <sip:mjh@isi.edu>
To: Eve Schooler <sip:schooler@caltech.edu> ;tag=9883472
Call-ID: 2963313058@north.east.isi.edu
CSeq: 1 ACK
16.3 Two-party Call
For two-party Internet phone calls, the response must contain a
description of where to send the data. In the example below, Bell
calls Watson. Bell indicates that he can receive RTP audio codings 0
(PCMU), 3 (GSM), 4 (G.723) and 5 (DVI4).
C->S: INVITE sip:watson@boston.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com>
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 INVITE
Subject: Mr. Watson, come here.
Content-Type: application/sdp
Content-Length: ...
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
s=Mr. Watson, come here.
c=IN IP4 kton.bell-tel.com
m=audio 3456 RTP/AVP 0 3 4 5
Handley, et al. Standards Track [Page 123]RFC 2543 SIP: Session Initiation Protocol March 1999
S->C: SIP/2.0 100 Trying
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 INVITE
Content-Length: 0
S->C: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 INVITE
Content-Length: 0
S->C: SIP/2.0 182 Queued, 2 callers ahead
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 INVITE
Content-Length: 0
S->C: SIP/2.0 182 Queued, 1 caller ahead
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 INVITE
Content-Length: 0
S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: <sip:watson@bell-tel.com> ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 INVITE
Contact: sip:watson@boston.bell-tel.com
Content-Type: application/sdp
Content-Length: ...
v=0
o=watson 4858949 4858949 IN IP4 192.1.2.3
s=I'm on my way
c=IN IP4 boston.bell-tel.com
m=audio 5004 RTP/AVP 0 3
Handley, et al. Standards Track [Page 124]RFC 2543 SIP: Session Initiation Protocol March 1999
The example illustrates the use of informational status responses.
Here, the reception of the call is confirmed immediately (100), then,
possibly after some database mapping delay, the call rings (180) and
is then queued, with periodic status updates.
Watson can only receive PCMU and GSM. Note that Watson's list of
codecs may or may not be a subset of the one offered by Bell, as each
party indicates the data types it is willing to receive. Watson will
send audio data to port 3456 at c.bell-tel.com, Bell will send to
port 5004 at boston.bell-tel.com.
By default, the media session is one RTP session. Watson will receive
RTCP packets on port 5005, while Bell will receive them on port 3457.
Since the two sides have agreed on the set of media, Bell confirms
the call without enclosing another session description:
C->S: ACK sip:watson@boston.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 1 ACK
16.4 Terminating a Call
To terminate a call, caller or callee can send a BYE request:
C->S: BYE sip:watson@boston.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP kton.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. A. Watson <sip:watson@bell-tel.com> ;tag=37462311
Call-ID: 3298420296@kton.bell-tel.com
CSeq: 2 BYE
If the callee wants to abort the call, it simply reverses the To and
From fields. Note that it is unlikely that a BYE from the callee will
traverse the same proxies as the original INVITE.
Handley, et al. Standards Track [Page 125]RFC 2543 SIP: Session Initiation Protocol March 1999
16.5 Forking Proxy
In this example, Bell (a.g.bell@bell-tel.com) (C), currently seated
at host c.bell-tel.com wants to call Watson (t.watson@ieee.org). At
the time of the call, Watson is logged in at two workstations,
t.watson@x.bell-tel.com (X) and watson@y.bell-tel.com (Y), and has
registered with the IEEE proxy server (P) called sip.ieee.org. The
IEEE server also has a registration for the home machine of Watson,
at watson@h.bell-tel.com (H), as well as a permanent registration at
watson@acm.org (A). For brevity, the examples omit the session
description and Via header fields.
Bell's user agent sends the invitation to the SIP server for the
ieee.org domain:
C->P: INVITE sip:t.watson@ieee.org SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
The SIP server at ieee.org tries the four addresses in parallel. It
sends the following message to the home machine:
P->H: INVITE sip:watson@h.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=1
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
This request immediately yields a 404 (Not Found) response, since
Watson is not currently logged in at home:
H->P: SIP/2.0 404 Not Found
Via: SIP/2.0/UDP sip.ieee.org ;branch=1
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>;tag=87454273
Handley, et al. Standards Track [Page 126]RFC 2543 SIP: Session Initiation Protocol March 1999
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
The proxy ACKs the response so that host H can stop retransmitting
it:
P->H: ACK sip:watson@h.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=1
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>;tag=87454273
Call-ID: 31415@c.bell-tel.com
CSeq: 1 ACK
Also, P attempts to reach Watson through the ACM server:
P->A: INVITE sip:watson@acm.org SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=2
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
In parallel, the next attempt proceeds, with an INVITE to X and Y:
P->X: INVITE sip:t.watson@x.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=3
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
P->Y: INVITE sip:watson@y.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=4
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Handley, et al. Standards Track [Page 127]RFC 2543 SIP: Session Initiation Protocol March 1999
As it happens, both Watson at X and a colleague in the other lab at
host Y hear the phones ringing and pick up. Both X and Y return 200s
via the proxy to Bell.
X->P: SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.ieee.org ;branch=3
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org> ;tag=192137601
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Contact: sip:t.watson@x.bell-tel.com
Y->P: SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.ieee.org ;branch=4
Via: SIP/2.0/UDP c.bell-tel.com
Contact: sip:t.watson@y.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org> ;tag=35253448
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Both responses are forwarded to Bell, using the Via information. At
this point, the ACM server is still searching its database. P can now
cancel this attempt:
P->A: CANCEL sip:watson@acm.org SIP/2.0
Via: SIP/2.0/UDP sip.ieee.org ;branch=2
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq: 1 CANCEL
The ACM server gladly stops its neural-network database search and
responds with a 200. The 200 will not travel any further, since P is
the last Via stop.
A->P: SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.ieee.org ;branch=2
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Handley, et al. Standards Track [Page 128]RFC 2543 SIP: Session Initiation Protocol March 1999
Call-ID: 31415@c.bell-tel.com
CSeq: 1 CANCEL
Bell gets the two 200 responses from X and Y in short order. Bell's
reaction now depends on his software. He can either send an ACK to
both if human intelligence is needed to determine who he wants to
talk to or he can automatically reject one of the two calls. Here, he
acknowledges both, separately and directly to the final destination:
C->X: ACK sip:t.watson@x.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>;tag=192137601
Call-ID: 31415@c.bell-tel.com
CSeq: 1 ACK
C->Y: ACK sip:watson@y.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>;tag=35253448
Call-ID: 31415@c.bell-tel.com
CSeq: 1 ACK
After a brief discussion between Bell with X and Y, it becomes clear
that Watson is at X. (Note that this is not a three-way call; only
Bell can talk to X and Y, but X and Y cannot talk to each other.)
Thus, Bell sends a BYE to Y, which is replied to:
C->Y: BYE sip:watson@y.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>;tag=35253448
Call-ID: 31415@c.bell-tel.com
CSeq: 2 BYE
Y->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>;tag=35253448
Call-ID: 31415@c.bell-tel.com
CSeq: 2 BYE
Handley, et al. Standards Track [Page 129]RFC 2543 SIP: Session Initiation Protocol March 1999
16.6 Redirects
Replies with status codes 301 (Moved Permanently) or 302 (Moved
Temporarily) specify another location using the Contact field.
Continuing our earlier example, the server P at ieee.org decides to
redirect rather than proxy the request:
P->C: SIP/2.0 302 Moved temporarily
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>;tag=72538263
Call-ID: 31415@c.bell-tel.com
CSeq: 1 INVITE
Contact: sip:watson@h.bell-tel.com,
sip:watson@acm.org, sip:t.watson@x.bell-tel.com,
sip:watson@y.bell-tel.com
CSeq: 1 INVITE
As another example, assume Alice (A) wants to delegate her calls to
Bob (B) while she is on vacation until July 29th, 1998. Any calls
meant for her will reach Bob with Alice's To field, indicating to him
what role he is to play. Charlie (C) calls Alice (A), whose server
returns:
A->C: SIP/2.0 302 Moved temporarily
From: Charlie <sip:charlie@caller.com>
To: Alice <sip:alice@anywhere.com> ;tag=2332462
Call-ID: 27182@caller.com
Contact: sip:bob@anywhere.com
Expires: Wed, 29 Jul 1998 9:00:00 GMT
CSeq: 1 INVITE
Charlie then sends the following request to the SIP server of the
anywhere.com domain. Note that the server at anywhere.com forwards
the request to Bob based on the Request-URI.
C->B: INVITE sip:bob@anywhere.com SIP/2.0
From: sip:charlie@caller.com
To: sip:alice@anywhere.com
Call-ID: 27182@caller.com
CSeq: 2 INVITE
Handley, et al. Standards Track [Page 130]RFC 2543 SIP: Session Initiation Protocol March 1999
In the third redirection example, we assume that all outgoing
requests are directed through a local firewall F at caller.com, with
Charlie again inviting Alice:
C->F: INVITE sip:alice@anywhere.com SIP/2.0
From: sip:charlie@caller.com
To: Alice <sip:alice@anywhere.com>
Call-ID: 27182@caller.com
CSeq: 1 INVITE
The local firewall at caller.com happens to be overloaded and thus
redirects the call from Charlie to a secondary server S:
F->C: SIP/2.0 302 Moved temporarily
From: sip:charlie@caller.com
To: Alice <sip:alice@anywhere.com>
Call-ID: 27182@caller.com
CSeq: 1 INVITE
Contact: <sip:alice@anywhere.com:5080;maddr=spare.caller.com>
Based on this response, Charlie directs the same invitation to the
secondary server spare.caller.com at port 5080, but maintains the
same Request-URI as before:
C->S: INVITE sip:alice@anywhere.com SIP/2.0
From: sip:charlie@caller.com
To: Alice <sip:alice@anywhere.com>
Call-ID: 27182@caller.com
CSeq: 2 INVITE
16.7 Negotiation
An example of a 606 (Not Acceptable) response is:
S->C: SIP/2.0 606 Not Acceptable
From: sip:mjh@isi.edu
To: <sip:schooler@cs.caltech.edu> ;tag=7434264
Call-ID: 14142@north.east.isi.edu
Handley, et al. Standards Track [Page 131]RFC 2543 SIP: Session Initiation Protocol March 1999
CSeq: 1 INVITE
Contact: sip:mjh@north.east.isi.edu
Warning: 370 "Insufficient bandwidth (only have ISDN)",
305 "Incompatible media format",
330 "Multicast not available"
Content-Type: application/sdp
Content-Length: 50
v=0
s=Let's talk
b=CT:128
c=IN IP4 north.east.isi.edu
m=audio 3456 RTP/AVP 5 0 7
m=video 2232 RTP/AVP 31
In this example, the original request specified a bandwidth that was
higher than the access link could support, requested multicast, and
requested a set of media encodings. The response states that only 128
kb/s is available and that (only) DVI, PCM or LPC audio could be
supported in order of preference.
The response also states that multicast is not available. In such a
case, it might be appropriate to set up a transcoding gateway and
re-invite the user.
16.8 OPTIONS Request
A caller Alice can use an OPTIONS request to find out the
capabilities of a potential callee Bob, without "ringing" the
designated address. Bob returns a description indicating that he is
capable of receiving audio encodings PCM Ulaw (payload type 0), 1016
(payload type 1), GSM (payload type 3), and SX7300/8000 (dynamic
payload type 99), and video encodings H.261 (payload type 31) and
H.263 (payload type 34).
C->S: OPTIONS sip:bob@example.com SIP/2.0
From: Alice <sip:alice@anywhere.org>
To: Bob <sip:bob@example.com>
Call-ID: 6378@host.anywhere.org
CSeq: 1 OPTIONS
Accept: application/sdp
S->C: SIP/2.0 200 OK
From: Alice <sip:alice@anywhere.org>
To: Bob <sip:bob@example.com> ;tag=376364382
Handley, et al. Standards Track [Page 132]RFC 2543 SIP: Session Initiation Protocol March 1999
Call-ID: 6378@host.anywhere.org
Content-Length: 81
Content-Type: application/sdp
v=0
m=audio 0 RTP/AVP 0 1 3 99
m=video 0 RTP/AVP 31 34
a=rtpmap:99 SX7300/8000
Handley, et al. Standards Track [Page 133]RFC 2543 SIP: Session Initiation Protocol March 1999
A Minimal Implementation
A.1 Client
All clients MUST be able to generate the INVITE and ACK requests.
Clients MUST generate and parse the Call-ID, Content-Length,
Content-Type, CSeq, From and To headers. Clients MUST also parse the
Require header. A minimal implementation MUST understand SDP (RFC
2327, [6]). It MUST be able to recognize the status code classes 1
through 6 and act accordingly.
The following capability sets build on top of the minimal
implementation described in the previous paragraph. In general, each
capability listed below builds on the ones above it:
Basic: A basic implementation adds support for the BYE method to
allow the interruption of a pending call attempt. It includes a
User-Agent header in its requests and indicates its preferred
language in the Accept-Language header.
Redirection: To support call forwarding, a client needs to be able to
understand the Contact header, but only the SIP-URL part, not
the parameters.
Firewall-friendly: A firewall-friendly client understands the Route
and Record-Route header fields and can be configured to use a
local proxy for all outgoing requests.
Negotiation: A client MUST be able to request the OPTIONS method and
understand the 380 (Alternative Service) status and the Contact
parameters to participate in terminal and media negotiation. It
SHOULD be able to parse the Warning response header to provide
useful feedback to the caller.
Authentication: If a client wishes to invite callees that require
caller authentication, it MUST be able to recognize the 401
(Unauthorized) status code, MUST be able to generate the
Authorization request header and MUST understand the WWW-
Authenticate response header.
If a client wishes to use proxies that require caller authentication,
it MUST be able to recognize the 407 (Proxy Authentication Required)
status code, MUST be able to generate the Proxy-Authorization request
header and understand the Proxy-Authenticate response header.
Handley, et al. Standards Track [Page 134]RFC 2543 SIP: Session Initiation Protocol March 1999
A.2 Server
A minimally compliant server implementation MUST understand the
INVITE, ACK, OPTIONS and BYE requests. A proxy server MUST also
understand CANCEL. It MUST parse and generate, as appropriate, the
Call-ID, Content-Length, Content-Type, CSeq, Expires, From, Max-
Forwards, Require, To and Via headers. It MUST echo the CSeq and
Timestamp headers in the response. It SHOULD include the Server
header in its responses.
A.3 Header Processing
Table 6 lists the headers that different implementations support. UAC
refers to a user-agent client (calling user agent), UAS to a user-
agent server (called user-agent).
The fields in the table have the following meaning. Type is as in
Table 4 and 5. "-" indicates the field is not meaningful to this
system (although it might be generated by it). "m" indicates the
field MUST be understood. "b" indicates the field SHOULD be
understood by a Basic implementation. "r" indicates the field SHOULD
be understood if the system claims to understand redirection. "a"
indicates the field SHOULD be understood if the system claims to
support authentication. "e" indicates the field SHOULD be understood
if the system claims to support encryption. "o" indicates support of
the field is purely optional. Headers whose support is optional for
all implementations are not shown.
Handley, et al. Standards Track [Page 135]
RFC 2543 SIP: Session Initiation Protocol March 1999
type UAC proxy UAS registrar
_____________________________________________________
Accept R - o m m
Accept-Encoding R - - m m
Accept-Language R - b b b
Allow 405 o - - -
Authorization R a o a a
Call-ID g m m m m
Content-Encoding g m - m m
Content-Length g m m m m
Content-Type g m - m m
CSeq g m m m m
Encryption g e - e e
Expires g - o o m
From g m o m m
Hide R - m - -
Contact R - - - m
Contact r r r - -
Max-Forwards R - b - -
Proxy-Authenticate 407 a - - -
Proxy-Authorization R - a - -
Proxy-Require R - m - -
Require R m - m m
Response-Key R - - e e
Route R - m - -
Timestamp g o o m m
To g m m m m
Unsupported r b b - -
User-Agent g b - b -
Via g m m m m
WWW-Authenticate 401 a - - -
Table 6: Header Field Processing Requirements
B Usage of the Session Description Protocol (SDP)
This section describes the use of the Session Description Protocol
(SDP) (RFC 2327 [6]).
B.1 Configuring Media Streams
The caller and callee align their media descriptions so that the nth
media stream ("m=" line) in the caller's session description
corresponds to the nth media stream in the callee's description.
Handley, et al. Standards Track [Page 136]RFC 2543 SIP: Session Initiation Protocol March 1999
All media descriptions SHOULD contain "a=rtpmap" mappings from RTP
payload types to encodings.
This allows easier migration away from static payload
types.
If the callee wants to neither send nor receive a stream offered by
the caller, the callee sets the port number of that stream to zero in
its media description.
There currently is no other way than port zero for the
callee to refuse a bidirectional stream offered by the
caller. Both caller and callee need to be aware what media
tools are to be started.
For example, assume that the caller Alice has included the following
description in her INVITE request. It includes an audio stream and
two bidirectional video streams, using H.261 (payload type 31) and
MPEG (payload type 32).
v=0
o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
c=IN IP4 host.anywhere.com
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 51372 RTP/AVP 31
a=rtpmap:31 H261/90000
m=video 53000 RTP/AVP 32
a=rtpmap:32 MPV/90000
The callee, Bob, does not want to receive or send the first video
stream, so it returns the media description below:
v=0
o=bob 2890844730 2890844730 IN IP4 host.example.com
c=IN IP4 host.example.com
m=audio 47920 RTP/AVP 0 1
a=rtpmap:0 PCMU/8000
a=rtpmap:1 1016/8000
m=video 0 RTP/AVP 31
m=video 53000 RTP/AVP 32
a=rtpmap:32 MPV/90000
Handley, et al. Standards Track [Page 137]RFC 2543 SIP: Session Initiation Protocol March 1999
B.2 Setting SDP Values for Unicast
If a session description from a caller contains a media stream which
is listed as send (receive) only, it means that the caller is only
willing to send (receive) this stream, not receive (send). The same
is true for the callee.
For receive-only and send-or-receive streams, the port number and
address in the session description indicate where the media stream
should be sent to by the recipient of the session description, either
caller or callee. For send-only streams, the address and port number
have no significance and SHOULD be set to zero.
The list of payload types for each media stream conveys two pieces of
information, namely the set of codecs that the caller or callee is
capable of sending or receiving, and the RTP payload type numbers
used to identify those codecs. For receive-only or send-and-receive
media streams, a caller SHOULD list all of the codecs it is capable
of supporting in the session description in an INVITE or ACK. For
send-only streams, the caller SHOULD indicate only those it wishes to
send for this session. For receive-only streams, the payload type
numbers indicate the value of the payload type field in RTP packets
the caller is expecting to receive for that codec type. For send-only
streams, the payload type numbers indicate the value of the payload
type field in RTP packets the caller is planning to send for that
codec type. For send-and-receive streams, the payload type numbers
indicate the value of the payload type field the caller expects to
both send and receive.
If a media stream is listed as receive-only by the caller, the callee
lists, in the response, those codecs it intends to use from among the
ones listed in the request. If a media stream is listed as send-only
by the caller, the callee lists, in the response, those codecs it is
willing to receive among the ones listed in the the request. If the
media stream is listed as both send and receive, the callee lists
those codecs it is capable of sending or receiving among the ones
listed by the caller in the INVITE. The actual payload type numbers
in the callee's session description corresponding to a particular
codec MUST be the same as the caller's session description.
If caller and callee have no media formats in common for a particular
stream, the callee MUST return a session description containing the
particular "m=" line, but with the port number set to zero, and no
payload types listed.
If there are no media formats in common for all streams, the callee
SHOULD return a 400 response, with a 304 Warning header field.
Handley, et al. Standards Track [Page 138]
RFC 2543 SIP: Session Initiation Protocol March 1999
B.3 Multicast Operation
The interpretation of send-only and receive-only for multicast media
sessions differs from that for unicast sessions. For multicast,
send-only means that the recipient of the session description (caller
or callee) SHOULD only send media streams to the address and port
indicated. Receive-only means that the recipient of the session
description SHOULD only receive media on the address and port
indicated.
For multicast, receive and send multicast addresses are the same and
all parties use the same port numbers to receive media data. If the
session description provided by the caller is acceptable to the
callee, the callee can choose not to include a session description or
MAY echo the description in the response.
A callee MAY, in the response, return a session description with some
of the payload types removed, or port numbers set to zero (but no
other value). This indicates to the caller that the callee does not
support the given stream or media types which were removed. A callee
MUST NOT change whether a given stream is send-only, receive-only, or
send-and-receive.
If a callee does not support multicast at all, it SHOULD return a 400
status response and include a 330 Warning.
B.4 Delayed Media Streams
In some cases, a caller may not know the set of media formats which
it can support at the time it would like to issue an invitation. This
is the case when the caller is actually a gateway to another protocol
which performs media format negotiation after call setup. When this
occurs, a caller MAY issue an INVITE with a session description that
contains no media lines. The callee SHOULD interpret this to mean
that the caller wishes to participate in a multimedia session
described by the session description, but that the media streams are
not yet known. The callee SHOULD return a session description
indicating the streams and media formats it is willing to support,
however. The caller MAY update the session description either in the
ACK request or in a re-INVITE at a later time, once the streams are
known.
B.5 Putting Media Streams on Hold
If a party in a call wants to put the other party "on hold", i.e.,
request that it temporarily stops sending one or more media streams,
a party re-invites the other by sending an INVITE request with a
modified session description. The session description is the same as
Handley, et al. Standards Track [Page 139]
RFC 2543 SIP: Session Initiation Protocol March 1999
in the original invitation (or response), but the "c" destination
addresses for the media streams to be put on hold are set to zero
(0.0.0.0).
B.6 Subject and SDP "s=" Line
The SDP "s=" line and the SIP Subject header field have different
meanings when inviting to a multicast session. The session
description line describes the subject of the multicast session,
while the SIP Subject header field describes the reason for the
invitation. The example in Section 16.2 illustrates this point. For
invitations to two-party sessions, the SDP "s=" line MAY be left
empty.
B.7 The SDP "o=" Line
The "o=" line is not strictly necessary for two-party sessions, but
MUST be present to allow re-use of SDP-based tools.
Handley, et al. Standards Track [Page 140]
RFC 2543 SIP: Session Initiation Protocol March 1999
C Summary of Augmented BNF
All of the mechanisms specified in this document are described in
both prose and an augmented Backus-Naur Form (BNF) similar to that
used by RFC 822 [9]. Implementors will need to be familiar with the
notation in order to understand this specification. The augmented BNF
includes the following constructs:
name = definition
The name of a rule is simply the name itself (without any enclosing
"<" and ">") and is separated from its definition by the equal "="
character. White space is only significant in that indentation of
continuation lines is used to indicate a rule definition that spans
more than one line. Certain basic rules are in uppercase, such as SP,
LWS, HT, CRLF, DIGIT, ALPHA, etc. Angle brackets are used within
definitions whenever their presence will facilitate discerning the
use of rule names.
"literal"
Quotation marks surround literal text. Unless stated otherwise, the
text is case-insensitive.
rule1 | rule2
Elements separated by a bar ("|") are alternatives, e.g., "yes | no"
will accept yes or no.
(rule1 rule2)
Elements enclosed in parentheses are treated as a single element.
Thus, "(elem (foo | bar) elem)" allows the token sequences "elem foo
elem" and "elem bar elem".
Handley, et al. Standards Track [Page 141]RFC 2543 SIP: Session Initiation Protocol March 1999
*rule
The character "*" preceding an element indicates repetition. The full
form is "<n>*<m>element" indicating at least <n> and at most <m>
occurrences of element. Default values are 0 and infinity so that
"*(element)" allows any number, including zero; "1*element" requires
at least one; and "1*2element" allows one or two.
[rule]
Square brackets enclose optional elements; "[foo bar]" is equivalent
to "*1(foo bar)".
N rule
Specific repetition: "<n>(element)" is equivalent to
"<n>*<n>(element)"; that is, exactly <n> occurrences of (element).
Thus 2DIGIT is a 2-digit number, and 3ALPHA is a string of three
alphabetic characters.
#rule
A construct "#" is defined, similar to "*", for defining lists of
elements. The full form is "<n>#<m> element" indicating at least <n>
and at most <m> elements, each separated by one or more commas (",")
and OPTIONAL linear white space (LWS). This makes the usual form of
lists very easy; a rule such as
( *LWS element *( *LWS "," *LWS element ))
can be shown as 1# element. Wherever this construct is used, null
elements are allowed, but do not contribute to the count of elements
present. That is, "(element), , (element)" is permitted, but counts
as only two elements. Therefore, where at least one element is
required, at least one non-null element MUST be present. Default
values are 0 and infinity so that "#element" allows any number,
including zero; "1#element" requires at least one; and "1#2element"
allows one or two.
Handley, et al. Standards Track [Page 142]RFC 2543 SIP: Session Initiation Protocol March 1999
; comment
A semi-colon, set off some distance to the right of rule text, starts
a comment that continues to the end of line. This is a simple way of
including useful notes in parallel with the specifications.
implied *LWS
The grammar described by this specification is word-based. Except
where noted otherwise, linear white space (LWS) can be included
between any two adjacent words (token or quoted-string), and between
adjacent tokens and separators, without changing the interpretation
of a field. At least one delimiter (LWS and/or separators) MUST exist
between any two tokens (for the definition of "token" below), since
they would otherwise be interpreted as a single token.
C.1 Basic Rules
The following rules are used throughout this specification to
describe basic parsing constructs. The US-ASCII coded character set
is defined by ANSI X3.4-1986.
OCTET = <any 8-bit sequence of data>
CHAR = <any US-ASCII character (octets 0 - 127)>
upalpha = "A" | "B" | "C" | "D" | "E" | "F" | "G" | "H" | "I" |
"J" | "K" | "L" | "M" | "N" | "O" | "P" | "Q" | "R" |
"S" | "T" | "U" | "V" | "W" | "X" | "Y" | "Z"
lowalpha = "a" | "b" | "c" | "d" | "e" | "f" | "g" | "h" | "i" |
"j" | "k" | "l" | "m" | "n" | "o" | "p" | "q" | "r" |
"s" | "t" | "u" | "v" | "w" | "x" | "y" | "z"
alpha = lowalpha | upalpha
digit = "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" |
"8" | "9"
alphanum = alpha | digit
CTL = <any US-ASCII control character
(octets 0 -- 31) and DEL (127)>
CR = %d13 ; US-ASCII CR, carriage return character
LF = %d10 ; US-ASCII LF, line feed character
SP = %d32 ; US-ASCII SP, space character
HT = %d09 ; US-ASCII HT, horizontal tab character
CRLF = CR LF ; typically the end of a line
The following are defined in RFC 2396 [12] for the SIP URI:
Handley, et al. Standards Track [Page 143]RFC 2543 SIP: Session Initiation Protocol March 1999
unreserved = alphanum | mark
mark = "-" | "_" | "." | "!" | "~" | "*" | "'"
| "(" | ")"
escaped = "%" hex hex
SIP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value or forwarding the message downstream.
LWS = [CRLF] 1*( SP | HT ) ; linear whitespace
The TEXT-UTF8 rule is only used for descriptive field contents and
values that are not intended to be interpreted by the message parser.
Words of *TEXT-UTF8 contain characters from the UTF-8 character set
(RFC 2279 [21]). In this regard, SIP differs from HTTP, which uses
the ISO 8859-1 character set.
TEXT-UTF8 = <any UTF-8 character encoding, except CTLs,
but including LWS>
A CRLF is allowed in the definition of TEXT-UTF8 only as part of a
header field continuation. It is expected that the folding LWS will
be replaced with a single SP before interpretation of the TEXT-UTF8
value.
Hexadecimal numeric characters are used in several protocol elements.
hex = "A" | "B" | "C" | "D" | "E" | "F"
| "a" | "b" | "c" | "d" | "e" | "f" | digit
Many SIP header field values consist of words separated by LWS or
special characters. These special characters MUST be in a quoted
string to be used within a parameter value.
Handley, et al. Standards Track [Page 144]RFC 2543 SIP: Session Initiation Protocol March 1999
token = 1*< any CHAR except CTL's or separators>
separators = "(" | ")" | "<" | ">" | "@" |
"," | ";" | ":" | "\" | <"> |
"/" | "[" | "]" | "?" | "=" |
"{" | "}" | SP | HT
Comments can be included in some SIP header fields by surrounding the
comment text with parentheses. Comments are only allowed in fields
containing "comment" as part of their field value definition. In all
other fields, parentheses are considered part of the field value.
comment = "(" *(ctext | quoted-pair | comment) ")"
ctext = < any TEXT-UTF8 excluding "(" and ")">
A string of text is parsed as a single word if it is quoted using
double-quote marks.
quoted-string = ( <"> *(qdtext | quoted-pair ) <"> )
qdtext = <any TEXT-UTF8 except <">>
The backslash character ("\") MAY be used as a single-character
quoting mechanism only within quoted-string and comment constructs.
quoted-pair = " \ " CHAR
Handley, et al. Standards Track [Page 145]RFC 2543 SIP: Session Initiation Protocol March 1999
D Using SRV DNS Records
The following procedure is experimental and relies on DNS SRV records
(RFC 2052 [14]). The steps listed below are used in place of the two
steps in section 1.4.2.
If a step elicits no addresses, the client continues to the next
step. However if a step elicits one or more addresses, but no SIP
server at any of those addresses responds, then the client concludes
the server is down and doesn't continue on to the next step.
When SRV records are to be used, the protocol to use when querying
for the SRV record is "sip". SRV records contain port numbers for
servers, in addition to IP addresses; the client always uses this
port number when contacting the SIP server. Otherwise, the port
number in the SIP URI is used, if present. If there is no port number
in the URI, the default port, 5060, is used.
1. If the host portion of the Request-URI is an IP address,
the client contacts the server at the given address. If the
host portion of the Request-URI is not an IP address, the
client proceeds to the next step.
2. The Request-URI is examined. If it contains an explicit
port number, the next two steps are skipped.
3. The Request-URI is examined. If it does not specify a
protocol (TCP or UDP), the client queries the name server
for SRV records for both UDP (if supported by the client)
and TCP (if supported by the client) SIP servers. The
format of these queries is defined in RFC 2052 [14]. The
results of the query or queries are merged together and
ordered based on priority. Then, the searching technique
outlined in RFC 2052 [14] is used to select servers in
order. If DNS doesn't return any records, the user goes to
the last step. Otherwise, the user attempts to contact
each server in the order listed. If no server is
contacted, the user gives up.
4. If the Request-URI specifies a protocol (TCP or UDP) that
is supported by the client, the client queries the name
server for SRV records for SIP servers of that protocol
type only. If the client does not support the protocol
specified in the Request-URI, it gives up. The searching
technique outlined in RFC 2052 [14] is used to select
servers from the DNS response in order. If DNS doesn't
Handley, et al. Standards Track [Page 146]RFC 2543 SIP: Session Initiation Protocol March 1999
return any records, the user goes to the last step.
Otherwise, the user attempts to contact each server in the
order listed. If no server is contacted, the user gives up.
5. The client queries the name server for address records for
the host portion of the Request-URI. If there were 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 their most modern equivalent.
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 starts the search as if
it had just received this answer from the name server. The server
uses the procedures specified in RFC1035 [15] regarding cache
invalidation when the time-to-live of the DNS result expires. If the
client does not find a SIP server among the addresses listed in the
cached answer, it starts the search at the beginning of the sequence
described above.
For example, consider a client that wishes to send a SIP request. The
Request-URI for the destination is sip:user@company.com. The client
only supports UDP. It would follow these steps:
1. The host portion is not an IP address, so the client goes
to step 2 above.
2. The client does a DNS query of QNAME="sip.udp.company.com",
QCLASS=IN, QTYPE=SRV. Since it doesn't support TCP, it
omits the TCP query. There were no addresses in the DNS
response, so the client goes to the next step.
3. The client does a DNS query for A records for
"company.com". An address is found, so that client attempts
to contact a server at that address at port 5060.
Handley, et al. Standards Track [Page 147]RFC 2543 SIP: Session Initiation Protocol March 1999
E IANA Considerations
Section 4.4 describes a name space and mechanism for registering SIP
options.
Section 6.41 describes the name space for registering SIP warn-codes.
Handley, et al. Standards Track [Page 148]
RFC 2543 SIP: Session Initiation Protocol March 1999
F Acknowledgments
We wish to thank the members of the IETF MMUSIC WG for their comments
and suggestions. Detailed comments were provided by Anders
Kristensen, Jim Buller, Dave Devanathan, Yaron Goland, Christian
Huitema, Gadi Karmi, Jonathan Lennox, Keith Moore, Vern Paxson, Moshe
J. Sambol, and Eric Tremblay.
This work is based, inter alia, on [37,38].
G Authors' Addresses
Mark Handley
AT&T Center for Internet Research at ISCI (ACIRI)
1947 Center St., Suite 600
Berkeley, CA 94704-119
USA
Email: mjh@aciri.org
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
Email: schulzrinne@cs.columbia.edu
Eve Schooler
Computer Science Department 256-80
California Institute of Technology
Pasadena, CA 91125
USA
Email: schooler@cs.caltech.edu
Jonathan Rosenberg
Lucent Technologies, Bell Laboratories
Rm. 4C-526
101 Crawfords Corner Road
Holmdel, NJ 07733
USA
Email: jdrosen@bell-labs.com
Handley, et al. Standards Track [Page 149]
RFC 2543 SIP: Session Initiation Protocol March 1999
H Bibliography
[1] Pandya, R., "Emerging mobile and personal communication systems,"
IEEE Communications Magazine , vol. 33, pp. 44--52, June 1995.
[2] Braden, B., Zhang, L., Berson, S., Herzog, S. and S. Jamin,
"Resource ReSerVation protocol (RSVP) -- version 1 functional
specification", RFC 2205, October 1997.
[3] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP:
a transport protocol for real-time applications", RFC 1889,
Internet Engineering Task Force, Jan. 1996.
[4] Schulzrinne, H., Lanphier, R. and A. Rao, "Real time streaming
protocol (RTSP)", RFC 2326, April 1998.
[5] Handley, M., "SAP: Session announcement protocol," Internet
Draft, Internet Engineering Task Force, Nov. 1996. Work in
progress.
[6] Handley, M. and V. Jacobson, "SDP: session description protocol",
RFC 2327, April 1998.
[7] International Telecommunication Union, "Visual telephone systems
and equipment for local area networks which provide a non-
guaranteed quality of service," Recommendation H.323,
Telecommunication Standardization Sector of ITU, Geneva,
Switzerland, May 1996.
[8] International Telecommunication Union, "Control protocol for
multimedia communication," Recommendation H.245,
Telecommunication Standardization Sector of ITU, Geneva,
Switzerland, Feb. 1998.
[9] International Telecommunication Union, "Media stream
packetization and synchronization on non-guaranteed quality of
service LANs," Recommendation H.225.0, Telecommunication
Standardization Sector of ITU, Geneva, Switzerland, Nov. 1996.
[10] Bradner, S., "Key words for use in RFCs to indicate requirement
levels", BCP 14, RFC 2119, Mardch 1997.
[11] Fielding, R., Gettys, J., Mogul, J., Nielsen, H. and T.
Berners-Lee, "Hypertext transfer protocol -- HTTP/1.1", RFC
2068, January 1997.
[12] Berners-Lee, T., Fielding, R. and L. Masinter, "Uniform resource
identifiers (URI): generic syntax", RFC 2396, August 1998.
Handley, et al. Standards Track [Page 150]RFC 2543 SIP: Session Initiation Protocol March 1999
[13] Berners-Lee, T., Masinter, L. and M. McCahill, "Uniform resource
locators (URL)", RFC 1738, December 1994.
[14] Gulbrandsen, A. and P. Vixie, "A DNS RR for specifying the
location of services (DNS SRV)", RFC 2052, October 1996.
[15] Mockapetris, P., "Domain names - implementation and
specification", STD 13, RFC 1035, Noveberm 1997.
[16] Hamilton, M. and R. Wright, "Use of DNS aliases for network
services", RFC 2219, October 1997.
[17] Zimmerman, D., "The finger user information protocol", RFC 1288,
December 1991.
[18] Williamson, S., Kosters, M., Blacka, D., Singh, J. and K.
Zeilstra, "Referral whois (rwhois) protocol V1.5", RFC 2167,
June 1997.
[19] Yeong, W., Howes, T. and S. Kille, "Lightweight directory access
protocol", RFC 1777, March 1995.
[20] Schooler, E., "A multicast user directory service for
synchronous rendezvous," Master's Thesis CS-TR-96-18, Department
of Computer Science, California Institute of Technology,
Pasadena, California, Aug. 1996.
[21] Yergeau, F., "UTF-8, a transformation format of ISO 10646", RFC
2279, January 1998.
[22] Stevens, W., TCP/IP illustrated: the protocols , vol. 1.
Reading, Massachusetts: Addison-Wesley, 1994.
[23] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
November 1990.
[24] Crocker, D., "Standard for the format of ARPA internet text
messages", RFC STD 11, RFC 822, August 1982.
[25] Meyer, D., "Administratively scoped IP multicast", RFC 2365,
July 1998.
[26] Schulzrinne, H., "RTP profile for audio and video conferences
with minimal control", RFC 1890, January 1996
[27] Eastlake, D., Crocker, S. and J. Schiller, "Randomness
recommendations for security", RFC 1750, December 1994.
Handley, et al. Standards Track [Page 151]RFC 2543 SIP: Session Initiation Protocol March 1999
[28] Hoffman, P., Masinter, L. and J. Zawinski, "The mailto URL
scheme", RFC 2368, July 1998.
[29] Braden, B., "Requirements for internet hosts - application and
support", STD 3, RFC 1123, October 1989.
[30] Palme, J., "Common internet message headers", RFC 2076, February
1997.
[31] Alvestrand, H., "IETF policy on character sets and languages",
RFC 2277, January 1998.
[32] Elkins, M., "MIME security with pretty good privacy (PGP)", RFC
2015, October 1996.
[33] Atkins, D., Stallings, W. and P. Zimmermann, "PGP message
exchange formats", RFC 1991, August 1996.
[34] Atkinson, R., "Security architecture for the internet protocol",
RFC 2401, November 1998.
[35] Allen, C. and T. Dierks, "The TLS protocol version 1.0," RFC
2246, January 1999.
[36] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A. and L. Stewart, "HTTP authentication:
Basic and digest access authentication," Internet Draft,
Internet Engineering Task Force, Sept. 1998. Work in progress.
[37] Schooler, E., "Case study: multimedia conference control in a
packet-switched teleconferencing system," Journal of
Internetworking: Research and Experience , vol. 4, pp. 99--120,
June 1993. ISI reprint series ISI/RS-93-359.
[38] Schulzrinne, H., "Personal mobility for multimedia services in
the Internet," in European Workshop on Interactive Distributed
Multimedia Systems and Services (IDMS) , (Berlin, Germany), Mar.
1996.
Handley, et al. Standards Track [Page 152]RFC 2543 SIP: Session Initiation Protocol March 1999
Full Copyright Statement
Copyright (C) The Internet Society (1999). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Handley, et al. Standards Track [Page 153]
