RFC 2543
SIP: Session Initiation Protocol
Network Working Group M. Handley Request for Comments: 2543 ACIRI Category: Standards Track H. Schulzrinne Columbia U. E. Schooler Cal Tech J. Rosenberg Bell Labs March 1999 SIP: Session Initiation Protocol Status of this Memo This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited. Copyright Notice Copyright (C) The Internet Society (1999). All Rights Reserved. IESG Note The IESG intends to charter, in the near future, one or more working groups to produce standards for "name lookup", where such names would include electronic mail addresses and telephone numbers, and the result of such a lookup would be a list of attributes and characteristics of the user or terminal associated with the name. Groups which are in need of a "name lookup" protocol should follow the development of these new working groups rather than using SIP for this function. In addition it is anticipated that SIP will migrate towards using such protocols, and SIP implementors are advised to monitor these efforts.Abstract
The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. Members in a session can communicate via multicast or via a mesh of unicast relations, or a combination of these.
SIP invitations used to create sessions carry session descriptions which allow participants to agree on a set of compatible media types. SIP supports user mobility by proxying and redirecting requests to the user's current location. Users can register their current location. SIP is not tied to any particular conference control protocol. SIP is designed to be independent of the lower-layer transport protocol and can be extended with additional capabilities.Table of Contents
1 Introduction ........................................ 7 1.1 Overview of SIP Functionality ....................... 7 1.2 Terminology ......................................... 8 1.3 Definitions ......................................... 9 1.4 Overview of SIP Operation ........................... 12 1.4.1 SIP Addressing ...................................... 12 1.4.2 Locating a SIP Server ............................... 13 1.4.3 SIP Transaction ..................................... 14 1.4.4 SIP Invitation ...................................... 15 1.4.5 Locating a User ..................................... 17 1.4.6 Changing an Existing Session ........................ 18 1.4.7 Registration Services ............................... 18 1.5 Protocol Properties ................................. 18 1.5.1 Minimal State ....................................... 18 1.5.2 Lower-Layer-Protocol Neutral ........................ 18 1.5.3 Text-Based .......................................... 20 2 SIP Uniform Resource Locators ....................... 20 3 SIP Message Overview ................................ 24 4 Request ............................................. 26 4.1 Request-Line ........................................ 26 4.2 Methods ............................................. 27 4.2.1 INVITE .............................................. 28 4.2.2 ACK ................................................. 29 4.2.3 OPTIONS ............................................. 29 4.2.4 BYE ................................................. 30 4.2.5 CANCEL .............................................. 30 4.2.6 REGISTER ............................................ 31 4.3 Request-URI ......................................... 34 4.3.1 SIP Version ......................................... 35 4.4 Option Tags ......................................... 35 4.4.1 Registering New Option Tags with IANA ............... 35 5 Response ............................................ 36 5.1 Status-Line ......................................... 36 5.1.1 Status Codes and Reason Phrases ..................... 37 6 Header Field Definitions ............................ 39 6.1 General Header Fields ............................... 41 6.2 Entity Header Fields ................................ 42 6.3 Request Header Fields ............................... 43
6.4 Response Header Fields .............................. 43 6.5 End-to-end and Hop-by-hop Headers ................... 43 6.6 Header Field Format ................................. 43 6.7 Accept .............................................. 44 6.8 Accept-Encoding ..................................... 44 6.9 Accept-Language ..................................... 45 6.10 Allow ............................................... 45 6.11 Authorization ....................................... 45 6.12 Call-ID ............................................. 46 6.13 Contact ............................................. 47 6.14 Content-Encoding .................................... 50 6.15 Content-Length ...................................... 51 6.16 Content-Type ........................................ 51 6.17 CSeq ................................................ 52 6.18 Date ................................................ 53 6.19 Encryption .......................................... 54 6.20 Expires ............................................. 55 6.21 From ................................................ 56 6.22 Hide ................................................ 57 6.23 Max-Forwards ........................................ 59 6.24 Organization ........................................ 59 6.25 Priority ............................................ 60 6.26 Proxy-Authenticate .................................. 60 6.27 Proxy-Authorization ................................. 61 6.28 Proxy-Require ....................................... 61 6.29 Record-Route ........................................ 62 6.30 Require ............................................. 63 6.31 Response-Key ........................................ 63 6.32 Retry-After ......................................... 64 6.33 Route ............................................... 65 6.34 Server .............................................. 65 6.35 Subject ............................................. 65 6.36 Timestamp ........................................... 66 6.37 To .................................................. 66 6.38 Unsupported ......................................... 68 6.39 User-Agent .......................................... 68 6.40 Via ................................................. 68 6.40.1 Requests ............................................ 68 6.40.2 Receiver-tagged Via Header Fields ................... 69 6.40.3 Responses ........................................... 70 6.40.4 User Agent and Redirect Servers ..................... 70 6.40.5 Syntax .............................................. 71 6.41 Warning ............................................. 72 6.42 WWW-Authenticate .................................... 74 7 Status Code Definitions ............................. 75 7.1 Informational 1xx ................................... 75 7.1.1 100 Trying .......................................... 75 7.1.2 180 Ringing ......................................... 75
7.1.3 181 Call Is Being Forwarded ......................... 75 7.1.4 182 Queued .......................................... 76 7.2 Successful 2xx ...................................... 76 7.2.1 200 OK .............................................. 76 7.3 Redirection 3xx ..................................... 76 7.3.1 300 Multiple Choices ................................ 77 7.3.2 301 Moved Permanently ............................... 77 7.3.3 302 Moved Temporarily ............................... 77 7.3.4 305 Use Proxy ....................................... 77 7.3.5 380 Alternative Service ............................. 78 7.4 Request Failure 4xx ................................. 78 7.4.1 400 Bad Request ..................................... 78 7.4.2 401 Unauthorized .................................... 78 7.4.3 402 Payment Required ................................ 78 7.4.4 403 Forbidden ....................................... 78 7.4.5 404 Not Found ....................................... 78 7.4.6 405 Method Not Allowed .............................. 78 7.4.7 406 Not Acceptable .................................. 79 7.4.8 407 Proxy Authentication Required ................... 79 7.4.9 408 Request Timeout ................................. 79 7.4.10 409 Conflict ........................................ 79 7.4.11 410 Gone ............................................ 79 7.4.12 411 Length Required ................................. 79 7.4.13 413 Request Entity Too Large ........................ 80 7.4.14 414 Request-URI Too Long ............................ 80 7.4.15 415 Unsupported Media Type .......................... 80 7.4.16 420 Bad Extension ................................... 80 7.4.17 480 Temporarily Unavailable ......................... 80 7.4.18 481 Call Leg/Transaction Does Not Exist ............. 81 7.4.19 482 Loop Detected ................................... 81 7.4.20 483 Too Many Hops ................................... 81 7.4.21 484 Address Incomplete .............................. 81 7.4.22 485 Ambiguous ....................................... 81 7.4.23 486 Busy Here ....................................... 82 7.5 Server Failure 5xx .................................. 82 7.5.1 500 Server Internal Error ........................... 82 7.5.2 501 Not Implemented ................................. 82 7.5.3 502 Bad Gateway ..................................... 82 7.5.4 503 Service Unavailable ............................. 83 7.5.5 504 Gateway Time-out ................................ 83 7.5.6 505 Version Not Supported ........................... 83 7.6 Global Failures 6xx ................................. 83 7.6.1 600 Busy Everywhere ................................. 83 7.6.2 603 Decline ......................................... 84 7.6.3 604 Does Not Exist Anywhere ......................... 84 7.6.4 606 Not Acceptable .................................. 84 8 SIP Message Body .................................... 84 8.1 Body Inclusion ...................................... 84
8.2 Message Body Type ................................... 85 8.3 Message Body Length ................................. 85 9 Compact Form ........................................ 85 10 Behavior of SIP Clients and Servers ................. 86 10.1 General Remarks ..................................... 86 10.1.1 Requests ............................................ 86 10.1.2 Responses ........................................... 87 10.2 Source Addresses, Destination Addresses and Connections ......................................... 88 10.2.1 Unicast UDP ......................................... 88 10.2.2 Multicast UDP ....................................... 88 10.3 TCP ................................................. 89 10.4 Reliability for BYE, CANCEL, OPTIONS, REGISTER Requests ............................................ 90 10.4.1 UDP ................................................. 90 10.4.2 TCP ................................................. 91 10.5 Reliability for INVITE Requests ..................... 91 10.5.1 UDP ................................................. 92 10.5.2 TCP ................................................. 95 10.6 Reliability for ACK Requests ........................ 95 10.7 ICMP Handling ....................................... 95 11 Behavior of SIP User Agents ......................... 95 11.1 Caller Issues Initial INVITE Request ................ 96 11.2 Callee Issues Response .............................. 96 11.3 Caller Receives Response to Initial Request ......... 96 11.4 Caller or Callee Generate Subsequent Requests ....... 97 11.5 Receiving Subsequent Requests ....................... 97 12 Behavior of SIP Proxy and Redirect Servers .......... 97 12.1 Redirect Server ..................................... 97 12.2 User Agent Server ................................... 98 12.3 Proxy Server ........................................ 98 12.3.1 Proxying Requests ................................... 98 12.3.2 Proxying Responses .................................. 99 12.3.3 Stateless Proxy: Proxying Responses ................. 99 12.3.4 Stateful Proxy: Receiving Requests .................. 99 12.3.5 Stateful Proxy: Receiving ACKs ...................... 99 12.3.6 Stateful Proxy: Receiving Responses ................. 100 12.3.7 Stateless, Non-Forking Proxy ........................ 100 12.4 Forking Proxy ....................................... 100 13 Security Considerations ............................. 104 13.1 Confidentiality and Privacy: Encryption ............. 104 13.1.1 End-to-End Encryption ............................... 104 13.1.2 Privacy of SIP Responses ............................ 107 13.1.3 Encryption by Proxies ............................... 108 13.1.4 Hop-by-Hop Encryption ............................... 108 13.1.5 Via field encryption ................................ 108 13.2 Message Integrity and Access Control: Authentication ...................................... 109
13.2.1 Trusting responses .................................. 112 13.3 Callee Privacy ...................................... 113 13.4 Known Security Problems ............................. 113 14 SIP Authentication using HTTP Basic and Digest Schemes ............................................. 113 14.1 Framework ........................................... 113 14.2 Basic Authentication ................................ 114 14.3 Digest Authentication ............................... 114 14.4 Proxy-Authentication ................................ 115 15 SIP Security Using PGP .............................. 115 15.1 PGP Authentication Scheme ........................... 115 15.1.1 The WWW-Authenticate Response Header ................ 116 15.1.2 The Authorization Request Header .................... 117 15.2 PGP Encryption Scheme ............................... 118 15.3 Response-Key Header Field for PGP ................... 119 16 Examples ............................................ 119 16.1 Registration ........................................ 119 16.2 Invitation to a Multicast Conference ................ 121 16.2.1 Request ............................................. 121 16.2.2 Response ............................................ 122 16.3 Two-party Call ...................................... 123 16.4 Terminating a Call .................................. 125 16.5 Forking Proxy ....................................... 126 16.6 Redirects ........................................... 130 16.7 Negotiation ......................................... 131 16.8 OPTIONS Request ..................................... 132 A. Minimal Implementation .............................. 134 A.1 Client .............................................. 134 A.2 Server .............................................. 135 A.3 Header Processing ................................... 135 B. Usage of the Session Description Protocol (SDP)...... 136 B.1 Configuring Media Streams ........................... 136 B.2 Setting SDP Values for Unicast ...................... 138 B.3 Multicast Operation ................................. 139 B.4 Delayed Media Streams ............................... 139 B.5 Putting Media Streams on Hold ....................... 139 B.6 Subject and SDP "s=" Line ........................... 140 B.7 The SDP "o=" Line ................................... 140 C. Summary of Augmented BNF ............................ 141 C.1 Basic Rules ......................................... 143 D. Using SRV DNS Records ............................... 146 E. IANA Considerations ................................. 148 F. Acknowledgments ..................................... 149 G. Authors' Addresses .................................. 149 H. Bibliography ........................................ 150 I. Full Copyright Statement ............................ 153
1 Introduction
1.1 Overview of SIP Functionality
The Session Initiation Protocol (SIP) is an application-layer control protocol that can establish, modify and terminate multimedia sessions or calls. These multimedia sessions include multimedia conferences, distance learning, Internet telephony and similar applications. SIP can invite both persons and "robots", such as a media storage service. SIP can invite parties to both unicast and multicast sessions; the initiator does not necessarily have to be a member of the session to which it is inviting. Media and participants can be added to an existing session. SIP can be used to initiate sessions as well as invite members to sessions that have been advertised and established by other means. Sessions can be advertised using multicast protocols such as SAP, electronic mail, news groups, web pages or directories (LDAP), among others. SIP transparently supports name mapping and redirection services, allowing the implementation of ISDN and Intelligent Network telephony subscriber services. These facilities also enable personal mobility. In the parlance of telecommunications intelligent network services, this is defined as: "Personal mobility is the ability of end users to originate and receive calls and access subscribed telecommunication services on any terminal in any location, and the ability of the network to identify end users as they move. Personal mobility is based on the use of a unique personal identity (i.e., personal number)." [1]. Personal mobility complements terminal mobility, i.e., the ability to maintain communications when moving a single end system from one subnet to another. SIP supports five facets of establishing and terminating multimedia communications: User location: determination of the end system to be used for communication; User capabilities: determination of the media and media parameters to be used; User availability: determination of the willingness of the called party to engage in communications; Call setup: "ringing", establishment of call parameters at both called and calling party;
Call handling: including transfer and termination of calls. SIP can also initiate multi-party calls using a multipoint control unit (MCU) or fully-meshed interconnection instead of multicast. Internet telephony gateways that connect Public Switched Telephone Network (PSTN) parties can also use SIP to set up calls between them. SIP is designed as part of the overall IETF multimedia data and control architecture currently incorporating protocols such as RSVP (RFC 2205 [2]) for reserving network resources, the real-time transport protocol (RTP) (RFC 1889 [3]) for transporting real-time data and providing QOS feedback, the real-time streaming protocol (RTSP) (RFC 2326 [4]) for controlling delivery of streaming media, the session announcement protocol (SAP) [5] for advertising multimedia sessions via multicast and the session description protocol (SDP) (RFC 2327 [6]) for describing multimedia sessions. However, the functionality and operation of SIP does not depend on any of these protocols. SIP can also be used in conjunction with other call setup and signaling protocols. In that mode, an end system uses SIP exchanges to determine the appropriate end system address and protocol from a given address that is protocol-independent. For example, SIP could be used to determine that the party can be reached via H.323 [7], obtain the H.245 [8] gateway and user address and then use H.225.0 [9] to establish the call. In another example, SIP might be used to determine that the callee is reachable via the PSTN and indicate the phone number to be called, possibly suggesting an Internet-to-PSTN gateway to be used. SIP does not offer conference control services such as floor control or voting and does not prescribe how a conference is to be managed, but SIP can be used to introduce conference control protocols. SIP does not allocate multicast addresses. SIP can invite users to sessions with and without resource reservation. SIP does not reserve resources, but can convey to the invited system the information necessary to do this.1.2 Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as described in RFC 2119 [10] and indicate requirement levels for compliant SIP implementations.
1.3 Definitions
This specification uses a number of terms to refer to the roles played by participants in SIP communications. The definitions of client, server and proxy are similar to those used by the Hypertext Transport Protocol (HTTP) (RFC 2068 [11]). The terms and generic syntax of URI and URL are defined in RFC 2396 [12]. The following terms have special significance for SIP. Call: A call consists of all participants in a conference invited by a common source. A SIP call is identified by a globally unique call-id (Section 6.12). Thus, if a user is, for example, invited to the same multicast session by several people, each of these invitations will be a unique call. A point-to-point Internet telephony conversation maps into a single SIP call. In a multiparty conference unit (MCU) based call-in conference, each participant uses a separate call to invite himself to the MCU. Call leg: A call leg is identified by the combination of Call-ID, To and From. Client: An application program that sends SIP requests. Clients may or may not interact directly with a human user. User agents and proxies contain clients (and servers). Conference: A multimedia session (see below), identified by a common session description. A conference can have zero or more members and includes the cases of a multicast conference, a full-mesh conference and a two-party "telephone call", as well as combinations of these. Any number of calls can be used to create a conference. Downstream: Requests sent in the direction from the caller to the callee (i.e., user agent client to user agent server). Final response: A response that terminates a SIP transaction, as opposed to a provisional response that does not. All 2xx, 3xx, 4xx, 5xx and 6xx responses are final. Initiator, calling party, caller: The party initiating a conference invitation. Note that the calling party does not have to be the same as the one creating the conference. Invitation: A request sent to a user (or service) requesting participation in a session. A successful SIP invitation consists of two transactions: an INVITE request followed by an ACK request.
Invitee, invited user, called party, callee: The person or service
that the calling party is trying to invite to a conference.
Isomorphic request or response: Two requests or responses are defined
to be isomorphic for the purposes of this document if they have
the same values for the Call-ID, To, From and CSeq header
fields. In addition, isomorphic requests have to have the same
Request-URI.
Location server: See location service.
Location service: A location service is used by a SIP redirect or
proxy server to obtain information about a callee's possible
location(s). Location services are offered by location servers.
Location servers MAY be co-located with a SIP server, but the
manner in which a SIP server requests location services is
beyond the scope of this document.
Parallel search: In a parallel search, a proxy issues several
requests to possible user locations upon receiving an incoming
request. Rather than issuing one request and then waiting for
the final response before issuing the next request as in a
sequential search , a parallel search issues requests without
waiting for the result of previous requests.
Provisional response: A response used by the server to indicate
progress, but that does not terminate a SIP transaction. 1xx
responses are provisional, other responses are considered final.
Proxy, proxy server: An intermediary program that acts as both a
server and a client for the purpose of making requests on behalf
of other clients. Requests are serviced internally or by passing
them on, possibly after translation, to other servers. A proxy
interprets, and, if necessary, rewrites a request message before
forwarding it.
Redirect server: A redirect server is a server that accepts a SIP
request, maps the address into zero or more new addresses and
returns these addresses to the client. Unlike a proxy server ,
it does not initiate its own SIP request. Unlike a user agent
server , it does not accept calls.
Registrar: A registrar is a server that accepts REGISTER requests. A
registrar is typically co-located with a proxy or redirect
server and MAY offer location services.
Ringback: Ringback is the signaling tone produced by the calling
client's application indicating that a called party is being
alerted (ringing).
Server: A server is an application program that accepts requests in
order to service requests and sends back responses to those
requests. Servers are either proxy, redirect or user agent
servers or registrars.
Session: From the SDP specification: "A multimedia session is a set
of multimedia senders and receivers and the data streams flowing
from senders to receivers. A multimedia conference is an example
of a multimedia session." (RFC 2327 [6]) (A session as defined
for SDP can comprise one or more RTP sessions.) As defined, a
callee can be invited several times, by different calls, to the
same session. If SDP is used, a session is defined by the
concatenation of the user name , session id , network type ,
address type and address elements in the origin field.
(SIP) transaction: A SIP transaction occurs between a client and a
server and comprises all messages from the first request sent
from the client to the server up to a final (non-1xx) response
sent from the server to the client. A transaction is identified
by the CSeq sequence number (Section 6.17) within a single call
leg. The ACK request has the same CSeq number as the
corresponding INVITE request, but comprises a transaction of its
own.
Upstream: Responses sent in the direction from the user agent server
to the user agent client.
URL-encoded: A character string encoded according to RFC 1738,
Section 2.2 [13].
User agent client (UAC), calling user agent: A user agent client is a
client application that initiates the SIP request.
User agent server (UAS), called user agent: A user agent server is a
server application that contacts the user when a SIP request is
received and that returns a response on behalf of the user. The
response accepts, rejects or redirects the request.
User agent (UA): An application which contains both a user agent
client and user agent server.
An application program MAY be capable of acting both as a client and
a server. For example, a typical multimedia conference control
application would act as a user agent client to initiate calls or to
invite others to conferences and as a user agent server to accept
invitations. The properties of the different SIP server types are
summarized in Table 1.
property redirect proxy user agent registrar
server server server
__________________________________________________________________
also acts as a SIP client no yes no no
returns 1xx status yes yes yes yes
returns 2xx status no yes yes yes
returns 3xx status yes yes yes yes
returns 4xx status yes yes yes yes
returns 5xx status yes yes yes yes
returns 6xx status no yes yes yes
inserts Via header no yes no no
accepts ACK yes yes yes no
Table 1: Properties of the different SIP server types
1.4 Overview of SIP Operation
This section explains the basic protocol functionality and operation.
Callers and callees are identified by SIP addresses, described in
Section 1.4.1. When making a SIP call, a caller first locates the
appropriate server (Section 1.4.2) and then sends a SIP request
(Section 1.4.3). The most common SIP operation is the invitation
(Section 1.4.4). Instead of directly reaching the intended callee, a
SIP request may be redirected or may trigger a chain of new SIP
requests by proxies (Section 1.4.5). Users can register their
location(s) with SIP servers (Section 4.2.6).
1.4.1 SIP Addressing
The "objects" addressed by SIP are users at hosts, identified by a
SIP URL. The SIP URL takes a form similar to a mailto or telnet URL,
i.e., user@host. The user part is a user name or a telephone number.
The host part is either a domain name or a numeric network address.
See section 2 for a detailed discussion of SIP URL's.
A user's SIP address can be obtained out-of-band, can be learned via
existing media agents, can be included in some mailers' message
headers, or can be recorded during previous invitation interactions.
In many cases, a user's SIP URL can be guessed from their email
address.
A SIP URL address can designate an individual (possibly located at one of several end systems), the first available person from a group of individuals or a whole group. The form of the address, for example, sip:sales@example.com , is not sufficient, in general, to determine the intent of the caller. If a user or service chooses to be reachable at an address that is guessable from the person's name and organizational affiliation, the traditional method of ensuring privacy by having an unlisted "phone" number is compromised. However, unlike traditional telephony, SIP offers authentication and access control mechanisms and can avail itself of lower-layer security mechanisms, so that client software can reject unauthorized or undesired call attempts.1.4.2 Locating a SIP Server
When a client wishes to send a request, the client either sends it to a locally configured SIP proxy server (as in HTTP), independent of the Request-URI, or sends it to the IP address and port corresponding to the Request-URI. For the latter case, the client must determine the protocol, port and IP address of a server to which to send the request. A client SHOULD follow the steps below to obtain this information, but MAY follow the alternative, optional procedure defined in Appendix D. At each step, unless stated otherwise, the client SHOULD try to contact a server at the port number listed in the Request-URI. If no port number is present in the Request-URI, the client uses port 5060. If the Request-URI specifies a protocol (TCP or UDP), the client contacts the server using that protocol. If no protocol is specified, the client tries UDP (if UDP is supported). If the attempt fails, or if the client doesn't support UDP but supports TCP, it then tries TCP. A client SHOULD be able to interpret explicit network notifications (such as ICMP messages) which indicate that a server is not reachable, rather than relying solely on timeouts. (For socket-based programs: For TCP, connect() returns ECONNREFUSED if the client could not connect to a server at that address. For UDP, the socket needs to be bound to the destination address using connect() rather than sendto() or similar so that a second write() fails with ECONNREFUSED if there is no server listening) If the client finds the server is not reachable at a particular address, it SHOULD behave as if it had received a 400-class error response to that request. The client tries to find one or more addresses for the SIP server by querying DNS. The procedure is as follows:
1. If the host portion of the Request-URI is an IP address,
the client contacts the server at the given address.
Otherwise, the client proceeds to the next step.
2. The client queries the DNS server for address records for
the host portion of the Request-URI. If the DNS server
returns no address records, the client stops, as it has
been unable to locate a server. By address record, we mean
A RR's, AAAA RR's, or other similar address records, chosen
according to the client's network protocol capabilities.
There are no mandatory rules on how to select a host name
for a SIP server. Users are encouraged to name their SIP
servers using the sip.domainname (i.e., sip.example.com)
convention, as specified in RFC 2219 [16]. Users may only
know an email address instead of a full SIP URL for a
callee, however. In that case, implementations may be able
to increase the likelihood of reaching a SIP server for
that domain by constructing a SIP URL from that email
address by prefixing the host name with "sip.". In the
future, this mechanism is likely to become unnecessary as
better DNS techniques, such as the one in Appendix D,
become widely available.
A client MAY cache a successful DNS query result. A successful query
is one which contained records in the answer, and a server was
contacted at one of the addresses from the answer. When the client
wishes to send a request to the same host, it MUST start the search
as if it had just received this answer from the name server. The
client MUST follow the procedures in RFC1035 [15] regarding DNS cache
invalidation when the DNS time-to-live expires.
1.4.3 SIP Transaction
Once the host part has been resolved to a SIP server, the client
sends one or more SIP requests to that server and receives one or
more responses from the server. A request (and its retransmissions)
together with the responses triggered by that request make up a SIP
transaction. All responses to a request contain the same values in
the Call-ID, CSeq, To, and From fields (with the possible addition of
a tag in the To field (section 6.37)). This allows responses to be
matched with requests. The ACK request following an INVITE is not
part of the transaction since it may traverse a different set of
hosts.
If TCP is used, request and responses within a single SIP transaction are carried over the same TCP connection (see Section 10). Several SIP requests from the same client to the same server MAY use the same TCP connection or MAY use a new connection for each request. If the client sent the request via unicast UDP, the response is sent to the address contained in the next Via header field (Section 6.40) of the response. If the request is sent via multicast UDP, the response is directed to the same multicast address and destination port. For UDP, reliability is achieved using retransmission (Section 10). The SIP message format and operation is independent of the transport protocol.1.4.4 SIP Invitation
A successful SIP invitation consists of two requests, INVITE followed by ACK. The INVITE (Section 4.2.1) request asks the callee to join a particular conference or establish a two-party conversation. After the callee has agreed to participate in the call, the caller confirms that it has received that response by sending an ACK (Section 4.2.2) request. If the caller no longer wants to participate in the call, it sends a BYE request instead of an ACK. The INVITE request typically contains a session description, for example written in SDP (RFC 2327 [6]) format, that provides the called party with enough information to join the session. For multicast sessions, the session description enumerates the media types and formats that are allowed to be distributed to that session. For a unicast session, the session description enumerates the media types and formats that the caller is willing to use and where it wishes the media data to be sent. In either case, if the callee wishes to accept the call, it responds to the invitation by returning a similar description listing the media it wishes to use. For a multicast session, the callee SHOULD only return a session description if it is unable to receive the media indicated in the caller's description or wants to receive data via unicast. The protocol exchanges for the INVITE method are shown in Fig. 1 for a proxy server and in Fig. 2 for a redirect server. (Note that the messages shown in the figures have been abbreviated slightly.) In Fig. 1, the proxy server accepts the INVITE request (step 1), contacts the location service with all or parts of the address (step 2) and obtains a more precise location (step 3). The proxy server then issues a SIP INVITE request to the address(es) returned by the location service (step 4). The user agent server alerts the user (step 5) and returns a success indication to the proxy server (step
6). The proxy server then returns the success result to the original
caller (step 7). The receipt of this message is confirmed by the
caller using an ACK request, which is forwarded to the callee (steps
8 and 9). Note that an ACK can also be sent directly to the callee,
bypassing the proxy. All requests and responses have the same Call-
ID.
+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@lab :
: 2| 3| :
: | | :
: henning | :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | \/ 4: INVITE 5: ring :
: cz@cs.tu-berlin.de ========================>(~~~~~~)=========>(~~~~~~) :
: <........................( )<.........( ) :
: : 7: 200 OK : ( )6: 200 OK ( ) :
: : : ( work ) ( lab ) :
: : 8: ACK : ( )9: ACK ( ) :
: ========================>(~~~~~~)=========>(~~~~~~) :
+.....................+ +...............................+
====> SIP request
....> SIP response
^
| non-SIP protocols
|
Figure 1: Example of SIP proxy server
The redirect server shown in Fig. 2 accepts the INVITE request (step
1), contacts the location service as before (steps 2 and 3) and,
instead of contacting the newly found address itself, returns the
address to the caller (step 4), which is then acknowledged via an ACK
request (step 5). The caller issues a new request, with the same call-ID but a higher CSeq, to the address returned by the first server (step 6). In the example, the call succeeds (step 7). The caller and callee complete the handshake with an ACK (step 8). The next section discusses what happens if the location service returns more than one possible alternative.1.4.5 Locating a User
A callee may move between a number of different end systems over time. These locations can be dynamically registered with the SIP server (Sections 1.4.7, 4.2.6). A location server MAY also use one or more other protocols, such as finger (RFC 1288 [17]), rwhois (RFC 2167 [18]), LDAP (RFC 1777 [19]), multicast-based protocols [20] or operating-system dependent mechanisms to actively determine the end system where a user might be reachable. A location server MAY return several locations because the user is logged in at several hosts simultaneously or because the location server has (temporarily) inaccurate information. The SIP server combines the results to yield a list of a zero or more locations. The action taken on receiving a list of locations varies with the type of SIP server. A SIP redirect server returns the list to the client as Contact headers (Section 6.13). A SIP proxy server can sequentially or in parallel try the addresses until the call is successful (2xx response) or the callee has declined the call (6xx response). With sequential attempts, a proxy server can implement an "anycast" service. If a proxy server forwards a SIP request, it MUST add itself to the beginning of the list of forwarders noted in the Via (Section 6.40) headers. The Via trace ensures that replies can take the same path back, ensuring correct operation through compliant firewalls and avoiding request loops. On the response path, each host MUST remove its Via, so that routing internal information is hidden from the callee and outside networks. A proxy server MUST check that it does not generate a request to a host listed in the Via sent-by, via- received or via-maddr parameters (Section 6.40). (Note: If a host has several names or network addresses, this does not always work. Thus, each host also checks if it is part of the Via list.) A SIP invitation may traverse more than one SIP proxy server. If one of these "forks" the request, i.e., issues more than one request in response to receiving the invitation request, it is possible that a client is reached, independently, by more than one copy of the
invitation request. Each of these copies bears the same Call-ID. The user agent MUST return the same status response returned in the first response. Duplicate requests are not an error.1.4.6 Changing an Existing Session
In some circumstances, it is desirable to change the parameters of an existing session. This is done by re-issuing the INVITE, using the same Call-ID, but a new or different body or header fields to convey the new information. This re INVITE MUST have a higher CSeq than any previous request from the client to the server. For example, two parties may have been conversing and then want to add a third party, switching to multicast for efficiency. One of the participants invites the third party with the new multicast address and simultaneously sends an INVITE to the second party, with the new multicast session description, but with the old call identifier.1.4.7 Registration Services
The REGISTER request allows a client to let a proxy or redirect server know at which address(es) it can be reached. A client MAY also use it to install call handling features at the server.1.5 Protocol Properties
1.5.1 Minimal State
A single conference session or call involves one or more SIP request-response transactions. Proxy servers do not have to keep state for a particular call, however, they MAY maintain state for a single SIP transaction, as discussed in Section 12. For efficiency, a server MAY cache the results of location service requests.1.5.2 Lower-Layer-Protocol Neutral
SIP makes minimal assumptions about the underlying transport and network-layer protocols. The lower-layer can provide either a packet or a byte stream service, with reliable or unreliable service. In an Internet context, SIP is able to utilize both UDP and TCP as transport protocols, among others. UDP allows the application to more carefully control the timing of messages and their retransmission, to perform parallel searches without requiring TCP connection state for each outstanding request, and to use multicast. Routers can more readily snoop SIP UDP packets. TCP allows easier passage through existing firewalls.
+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@lab :
: 2| 3| :
: | | :
: henning| :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | \/ :
: cz@cs.tu-berlin.de =======================>(~~~~~~) :
: | ^ | <.......................( ) :
: | . | : 4: 302 Moved : ( ) :
: | . | : hgs@lab : ( work ) :
: | . | : : ( ) :
: | . | : 5: ACK : ( ) :
: | . | =======================>(~~~~~~) :
: | . | : : :
+.......|...|.........+ : :
| . | : :
| . | : :
| . | : :
| . | : :
| . | 6: INVITE hgs@lab.cs.columbia.edu (~~~~~~) :
| . ==================================================> ( ) :
| ..................................................... ( ) :
| 7: 200 OK : ( lab ) :
| : ( ) :
| 8: ACK : ( ) :
======================================================> (~~~~~~) :
+...............................+
====> SIP request
....> SIP response
^
| non-SIP protocols
|
Figure 2: Example of SIP redirect server
When TCP is used, SIP can use one or more connections to attempt to contact a user or to modify parameters of an existing conference. Different SIP requests for the same SIP call MAY use different TCP connections or a single persistent connection, as appropriate. For concreteness, this document will only refer to Internet protocols. However, SIP MAY also be used directly with protocols such as ATM AAL5, IPX, frame relay or X.25. The necessary naming conventions are beyond the scope of this document. User agents SHOULD implement both UDP and TCP transport. Proxy, registrar, and redirect servers MUST implement both UDP and TCP transport.1.5.3 Text-Based
SIP is text-based, using ISO 10646 in UTF-8 encoding throughout. This allows easy implementation in languages such as Java, Tcl and Perl, allows easy debugging, and most importantly, makes SIP flexible and extensible. As SIP is used for initiating multimedia conferences rather than delivering media data, it is believed that the additional overhead of using a text-based protocol is not significant.
(page 20 continued on part 2)
