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Internet Engineering Task Force                                MMUSIC WG
Internet Draft                              Handley/Schulzrinne/Schooler
draft-ietf-mmusic-sip-04.txt                     ISI/Columbia U./Caltech
November 11, 1997
Expires: April 1, 1998


                    SIP: Session Initiation Protocol

STATUS OF THIS MEMO

   This document is an Internet-Draft. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
   and its working groups.  Note that other groups may also distribute
   working documents as Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as ``work in progress''.

   To learn the current status of any Internet-Draft, please check the
   ``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
   Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe),
   munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or
   ftp.isi.edu (US West Coast).

   Distribution of this document is unlimited.

                                 ABSTRACT


         Many styles of multimedia conferencing are likely to co-
         exist on the Internet, and many of them share the need to
         invite users to participate. The Session Initiation
         Protocol (SIP) is a simple protocol designed to enable
         the invitation of users to participate in such multimedia
         sessions. It is not tied to any specific conference
         control scheme. In particular, it aims to enable user
         mobility by relaying and redirecting invitations to a
         user's current location.

         This document is a product of the Multi-party Multimedia
         Session Control (MMUSIC) working group of the Internet
         Engineering Task Force.  Comments are solicited and
         should be addressed to the working group's mailing list
         at confctrl@isi.edu and/or the authors.




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1 Introduction

1.1 Overview of SIP Functionality

   The Session Initiation Protocol (SIP) is an application-layer
   protocol that can establish and control multimedia sessions or calls.
   These multimedia sessions include multimedia conferences, distance
   learning, Internet telephony and similar applications. SIP can invite
   a person to both unicast and multicast sessions; the initiator does
   not necessarily have to be a member of the session it is inviting to.
   Media and participants can be added to an existing session. SIP can
   be used to "call" both persons and "robots", for example, to invite a
   media storage device to record an ongoing conference or to invite a
   video-on-demand server to play a video into a conference. (SIP does
   not directly control these services, however; see RTSP [1].)

   SIP can be used to initiate sessions as well as invite members to
   sessions that have been advertised and established by other means.
   (Sessions may be advertised using multicast protocols such as SAP
   [2], 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. Section 14 discusses these services in detail.

   SIP supports personal mobility 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')." [3].  Personal mobility complements terminal
   mobility, i.e., the ability to maintain communications when moving a
   single end system from one network to another.

   SIP supports some or all of 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;




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   Call setup: "ringing", establishment of call parameters at both
        called and calling party;

   Call handling: including transfer and termination of calls.

   SIP may also be used in conjunction with other call setup and
   signaling protocols. In that mode, an end system uses SIP protocol
   exchanges to determine the appropriate end system address and
   protocol from a given address that is protocol-independent. For
   example, SIP may be used to determine that the party may be reached
   via H.323, obtain the H.245 gateway and user address and then use
   H.225.0 to establish the call [4]. In another example, it may be used
   to determine that the callee is reachable via the public switched
   telephone network (PSTN) and indicate the phone number to be called,
   possibly suggesting an Internet-to-PSTN gateway to be used.

   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 PSTN parties may also use
   SIP to set up calls between them.

   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, leaving this functionality
   to protocols such as SAP [2].

   SIP can invite users to sessions with and without resource
   reservation.  SIP does not reserve resources, but may convey to the
   invited system the information necessary to do this. Quality-of-
   service guarantees, if required, may depend on knowing the full
   membership of the session; this information may or may not be known
   to the agent performing session invitation.

   SIP is designed as part of the overall IETF multimedia data and
   control architecture [5] currently incorporating protocols such as
   RSVP [6] for reserving network resources, the real-time transport
   protocol (RTP) [7] for transporting real-time data and providing QOS
   feedback, the real-time streaming protocol (RTSP) [8] for controlling
   delivery of streaming media, the session announcement protocol (SAP)
   [2] for advertising multimedia sessions via multicast and the session
   description protocol (SDP) [9] for describing multimedia sessions,
   but the functionality and operation of SIP does not depend on any of
   these protocols.

1.2 Terminology



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   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) [11]. The following terms have special
   significance for SIP.

   Call: A call consists of a single invitation attempt from a single
        user. 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 MCU-
        based conference, each participant uses a separate call to
        invite himself to the MCU.

   Client: An application program that establishes connections for the
        purpose of sending requests. Clients may or may not interact
        directly with a human user.

   Final response: A response that terminates a SIP transaction, as
        opposed to a provisional response responses are final.

   Initiator, calling party: The party initiating a conference
        invitation. Note that the calling party does not have to be the
        same as the one creating a 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 a  ACK
        request.

   Invitee, invited user, called party: The person or service that the
        calling party is trying to invite to a conference.

   Location server: See location service

   Location service: A service 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



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        scope of the document.

   Provisional response: A response used by the server to indicate
        progress, but that does not terminate a SIP transaction. All 1xx
        and 6xx 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
        must interpret, and, if necessary, rewrite a request message
        before forwarding it.

   Redirect server: 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.

   Server: 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. An
        application program may act as both server and client.

   Session: "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." [9] (Note: a session as defined here may comprise one
        or more RTP sessions.) Since the word session is used
        differently by protocols relevant to SIP, this document avoids
        the term altogether.

   (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 for a
        single call (identified by a  Call-ID, Section 6.12). There can
        only be one pending transaction between a server and client for
        each call id.

   User agent server, called user agent: The server application that
        contacts the user when a SIP request is received and that
        returns a reply on behalf of the user. The reply may accept,
        reject or redirect the call. (Note: in SIP, user agents can be
        both clients and servers.)

   An application program may be capable of acting both as a client and



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   a server. For example, a typical multimedia conference control
   application would act as a 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
                           server     server      server
   _______________________________________________________
   also acts as client       no        yes          no
   return  1xx status       yes        yes         yes
   return  2xx status        no        yes         yes
   return  3xx status       yes        yes         yes
   return  4xx status       yes        yes         yes
   return  5xx status       yes        yes         yes
   return  6xx status        no        yes         yes
   insert  Via header        no        yes          no
   accept  ACK               no        yes         yes


   Table 1: Properties of the different SIP server types


1.4 Summary 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 trigger a chain of new SIP requests
   by proxies (Section 1.4.5). Users can register with SIP servers
   (Section 4.2.5).

1.4.1 SIP Addressing

   SIP addresses contain a user and host part. The user part is an
   operating-system user name. The host part is either a domain name
   having a DNS A (address) record, or a numeric network address.
   Examples include:

     mjh@metro.isi.edu
     hgs@erlang.cs.columbia.edu
     root@[193.175.132.42]
     root@193.175.132.42



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   A user's 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.

   SIP addresses may contain a moniker (such as a civil name) or user
   name and domain name that may not map into a single host. [1]

   SIP addresses may use any unambiguous user name, including aliases,
   identifying the called party as the user part of the address.  They
   may use a domain name having an MX [12], SRV [13] or A [14] record
   for the host part.  These addresses may have different degrees of
   location- and provider-independence and are often chosen to be
   mnemonic. In many cases, the SIP address can be the same as a user's
   electronic mail address, but this is not required. SIP can thus
   leverage off the domain name system (DNS) to provide a first-stage
   location mechanisms.  Examples of SIP names include

     M.Handley@cs.ucl.ac.uk
     H.G.Schulzrinne@ieee.org
     info@ietf.org



   An 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, e.g.,
   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.

   When used within SIP, SIP addresses are written as SIP URLs (Section
   sec:url), e.g., sip://info@ietf.org as SIP requests and responses may
   also contain non-SIP addresses, e.g., telephone numbers.

1.4.2 Locating a SIP Server
_________________________

  [1] We avoid the term  location-independent  ,  since
the  address  may  indeed refer to a specific location,
e.g., a company department.




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   A SIP client MUST follow the following steps to resolve the host part
   of a callee address. If a client only supports TCP or UDP, but not
   both, the respective address type is omitted. If the SIP address
   contains a port number, that number is to be used, otherwise, the the
   default port number. The default port number for UDP and TCP is the
   same.

        1.   If the SIP address is a numeric IP address, contact a SIP
             server at that address.

        2.   If the SIP address does not contain a port number and if
             there is a SRV DNS resource record [13] of type  sip.udp,
             contact the listed SIP servers in the order of the
             preference values contained in those resource records,
             using UDP as a transport protocol at the port number listed
             in the DNS resource record. [TBD: What if the SIP URL
             contains a port number?]

        3.   If the SIP address does not contain a port number and if
             there is a SRV DNS resource record [13] of type  sip.tcp,
             contact the listed SIP servers in the order of the
             preference value contained in those resource records, using
             TCP as a transport protocol at the port number listed in
             the DNS resource record.

        4.   If there is a DNS MX record [12], contact the hosts listed
             in their order of preference at the default port number
             (TBD).  For each host listed, first try to contact the SIP
             server using UDP, then TCP.

        5.   Finally, check if there is a DNS CNAME or A record for the
             given host and try to contact a SIP server at the one or
             more addresses listed, again trying first UDP, then TCP.

        6.   If all of the above methods fail, the caller MAY contact an
             SMTP server at the user's host and use the SMTP  EXPN
             command to obtain an alternate address and repeat the steps
             above. As a last resort, a client MAY choose to deliver the
             session description to the callee using electronic mail.

   If a server is found using one of the methods below, the other
   methods are not tried. A client SHOULD rely on ICMP "Port
   Unreachable" messages rather than time-outs to determine that a
   server is not reachable at a particular address.

   A client MAY cache the result of the reachability steps for a
   particular address and retry that host address for the next call. If
   the client does not find a SIP server at the cached address, it MUST



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   start the search at the beginning of the sequence.

   Implementation note for socket-based programs: For TCP, connect()
   returns ECONNREFUSED if there is no server at the designated address;
   for UDP, the socket should be bound to the destination address using
   connect() rather than sendto() or similar.


        This sequence is modeled after that described for SMTP,
        where MX records are to be checked before A records [15].

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.

   If TCP is used, request and responses within a single SIP transaction
   are carried over the same TCP connection. Thus, the client SHOULD
   maintain the connection until a final response has been received.
   Several SIP requests from the same client to the same server may use
   the same TCP connection or may open a new connection for each
   request. If the client sent the request sends via unicast UDP, the
   response is sent to the source address of the UDP request.
   (Implementation note: use recvfrom() to obtain the source address and
   port of the request.)  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
   11).


        Need motivation why we ALWAYS want to have a multicast
        return.

   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 call is rejected or otherwise
   unsuccessful, the caller sends a  BYE request instead of an  ACK.



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   The  INVITE request typically contains a session description, for
   example written in SDP 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 may
   be distributed to that session. For unicast session, the session
   description enumerates the media types and formats that the caller is
   willing to receive 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 receive. 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. The caller may ignore the
   session description returned or use it to change the global session
   description.

   The session description may refer to a session start time in the
   future.  Actual transmission of data SHOULD not start until the time
   indicated in the session description.

   The protocol exchanges for the  INVITE method are shown in Fig. 1 for
   a proxy server and in Fig. 2 for a redirect server. 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
   message, which is forwarded to the callee (steps 8 and 9), with a
   response returned (steps 10 and 11). All requests have the same
   Call-ID.


   The redirect server 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). The caller issues a new request, with a new call-ID,
   to the address returned by the first server (step 6). In the example,
   the call succeeds (step 7). The caller and callee complete the
   handshanke with an  ACK (steps 8 and 9).


   The next section discusses what happens if the location service
   returns more than one possible alternative.

1.4.5 Locating a User




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                                            +....... cs.columbia.edu .......+
                                            :                               :
                                            : (~~~~~~~~~~)                  :
                                            : ( location )                  :
                                            : ( service  )                  :
                                            : (~~~~~~~~~~)                  :
                                            :   ^      |                    :
                                            :   |   hgs@play                :
                                            :  2|     3|                    :
                                            :   |      |                    :
                                            : henning  |                    :
   +.. cs.tu-berlin.de ..+ 1: INVITE        :   |      |                    :
   :                     :    henning@cs.col:   |      | 4: INVITE  5: ring :
   : cz@cs.tu-berlin.de ========================> tune  =========> play     :
   :                    <........................       <.........          :
   :                     : 7: 200 OK        :            6: 200 OK          :
   +.....................+                  +...............................+

   ====> SIP request
   ----> non-SIP protocols

   Figure 1: Example of SIP proxy server


   A callee may move between a number of different end systems over
   time.  These locations can be dynamically registered with the SIP
   server (Section 4.2.5) or a location server, typically for a single
   administrative domain, or a location server may use other protocols,
   such as finger [16], rwho, multicast-based protocols or operating-
   system dependent mechanism to actively determine the end system where
   a user might be reachable. The location services yield a list of a
   zero or more possible locations, possibly even sorted in order of
   likelihood of success.

   The location server can be part of the SIP server or the SIP server
   may use a different protocol (e.g., finger [16] or LDAP [17]) to map
   addresses. A single user may be registered at different locations,
   either because she is logged in at several hosts simultaneously or
   because the location server has (temporarily) inaccurate information.

   The action taken on receiving a list of locations varies with the
   type of SIP server. A SIP redirect server simply returns the list to
   the client sending the request as  Location headers (Section 6.18). A
   SIP proxy server can sequentially or in parallel try the addresses



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                                            +....... cs.columbia.edu .......+
                                            :                               :
                                            : (~~~~~~~~~~)                  :
                                            : ( location )                  :
                                            : ( service  )                  :
                                            : (~~~~~~~~~~)                  :
                                            :   ^      |                    :
                                            :   |   hgs@play                :
                                            :  2|     3|                    :
                                            :   |      |                    :
                                            : henning  |                    :
   +.. cs.tu-berlin.de ..+ 1: INVITE        :   |      |                    :
   :                     :    henning@cs.col:   |      |                    :
   : cz@cs.tu-berlin.de =======================>  tune                      :
   :         ^ |        <.......................                            :
   :         . |         : 4: 302 Moved     :                               :
   +...........|.........+    hgs@play      :                               :
             . |                            :                               :
             . | 5: INVITE hgs@play.cs.columbia.edu                6: ring  :
             . ==================================================> play     :
             .....................................................          :
               7: 200 OK                    :                               :
                                            +...............................+

   ====> SIP request
   ----> non-SIP protocols

   Figure 2: Example of SIP redirect server


   until the call is successful (2xx response) or the callee has
   declined the call (60x 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
   end of the list of forwarders noted in the  Via (Section 6.33)
   headers. The  Via trace ensures that replies can take the same path
   back, thus ensuring correct operation through compliant firewalls and
   loop-free requests. On the reply path, each host most remove its Via,
   so that routing internal information is hidden from the callee and
   outside networks. When a multicast request is made, first the host
   making the request, then the multicast address itself are added to
   the path. A proxy server MUST check that it does not generate a
   request to a host listed in the  Via list. (Note: If a host has



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   several names or network addresses, this may 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 appropriate status response, but
   SHOULD NOT alert the user.

   As discussed in Section 1.4.1, a SIP address may designate a group
   rather than an individual. A client indicates using the  Reach
   request header whether it wants to reach the first available
   individual or treat the address as a group, to be invited as a whole.
   The default is to attempt to reach the first available callee.  If
   the address is designated as a group address, a proxy server MUST
   return the list of individuals instead of attempting to connect to
   these. (Otherwise, the proxy cannot report errors, redirections and
   call status individually. For example, some may be contacted
   successfully, while one of the group may be reachable under a
   different address.)

1.4.6 Changing an Existing Session

   In some circumstances, it may be necessary to change the parameters
   of an existing session. 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 the old
   call identifier.

1.4.7 Registration Services

   The  REGISTER and  UNREGISTER requests allow a client to let a proxy
   or redirect server know which address it may be reached under. 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 may involve one or more SIP
   request-response transactions. Proxy server do not have to keep state
   for a particular call, however, they maintain state for a single SIP
   transaction, as discussed in Section 12.



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   For efficiency, a server may cache the results of location service
   requests.

1.5.2 Transport-Protocol Neutral

   SIP is able to utilize both UDP and TCP as transport protocols. 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, and
   given the similar protocol design, allows common servers for SIP,
   HTTP and the Real Time Streaming Protocol (RTSP) [1].

   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.

   Clients SHOULD implement both UDP and TCP transport, servers MUST.

1.5.3 Text-Based

   SIP is text based. This allows easy implementation in languages such
   as 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 and
   recipient of a SIP request, and to specify redirection addresses. A
   SIP URL may also be embedded in web pages or other hyperlinks to
   indicate that a user or service may be called.

   Because interaction with some resources may require message headers
   or message bodies to be specified as well as the SIP address, the sip
   URL scheme is defined to allow setting SIP  request-header fields and
   the SIP  message-body. (This is similar to the  mailto: URL.)

   A SIP URL follows the guidelines of RFC 1630 [18,19] and takes the
   following form:



        SIP-URL            =    short-sip-url | full-sip-url



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        full-sip-url       =    "sip://" ( user | phone ) [ ":" password ]
                                "@" [ host | nhost ]
                                url-parameters [ headers ]
        short-sip-url      =    ( user | phone) [ ":" password ]
                                "@" [ host | nhost ] : port
        user               =    ;  defined in RFC 1738 [20]
        phone              =    "+" DIGIT *( DIGIT | "-" | "." )
        host               =    ;  defined in RFC 1738
        nhost              =    "[" hostnumber "]" | hostnumber
        hostnumber         =    digits "." digits "." digits "." digits
        port               =    *digit
        url-parameters     =    *( ";" url-parameter)
        url-parameter      =    transport-param |
                                ttl-param | maddr-param
        transport-param    =    "transport=" ( "udp" | "tcp" )
        ttl-param          =    "ttl=" ttl
        ttl                =    1*3DIGIT                                        ; 0 to 255
        maddr-param        =    "maddr=" maddr
        maddr              =    ;  dotted decimal multicast address
        headers            =    "?" header *( "                               " header )
        header             =    hname "=" hvalue
        hname              =    *urlc
        hvalue             =    *urlc
        urlc               =    ;  defined in [19]
        digits             =    1*digit


   Thus, a SIP URL can take either a short form or a full form. The
   short form MAY only be used within SIP messages where the scheme
   (SIP) can be assumed. In all other cases, and when parameters are
   required to be specified, the full form MUST be used.

   Note that all URL reserved characters must be encoded. The special
   hname  "body" indicates that the associated  hvalue is the message-
   body of the SIP  INVITE request. Within sip URLs, the characters
   "?",  "=",  "&" are reserved.

   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 allows both
   forms.

   The  password parameter can be used for a basic authentication
   mechanism that takes the place of an unlisted telephone number. Also,
   for Internet telephony gateways, it may serve as a PIN. Including
   just the password in the URL is more convenient than including a
   whole authentication header. This approach may be reasonably secure



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   if the URL is part of a secure web page. Unless the SIP transaction
   is carried over a secure network connection, this carries the same
   security risks as all URL-based passwords and should only be used
   when security requirements are low. In almost all circumstances, use
   of the Authorization (Section 6.10) header is preferred.

   The  phone identifier is to be used when connecting to a telephony
   gateway. The phone number follows the rules for international numbers
   in ITU Recommendation E.123, with only numbers and hyphens allowed.

   Examples of short and full-form SIP URLs are:

     j.doe@big.com
     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
     sip://alice@[10.1.2.3]
     sip://alice@10.1.2.3



   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. When additional parameters are not required, the
   short form SIP URL can be used unambiguously.

   In some circumstances a non-SIP URL may be used in a SIP message. An
   example might be making a call from a telephone which is relayed by a
   gateway onto the internet as a SIP request. In such a case, the
   source of the call is really the telephone number of the caller, and
   so a SIP URL is inappropriate and a phone URL might be used instead.
   Thus where SIP specifies user addresses it allows these addresses to
   be URLs.

   Clearly not all URLs are appropriate to be used in a SIP message as a
   user address. The correct behavior when an unknown scheme is
   encountered by a SIP server is defined in the context of each of the
   header fields that use a SIP URL.

   SIP URLs can define specific parameters of the request, including the
   transport mechanism (UDP or TCP) and the use of multicast to make a
   request. These parameters are added after the  host and are separated
   by semi-colons. 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:




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     sip://j.doe@big.com;maddr=239.255.255.1;ttl=15



   The transport protocol UDP is to be assumed when a multicast address
   is given.

3 SIP Message Overview

   Since much of the message syntax is identical to HTTP/1.1, rather
   than repeating it here we use [HX.Y] to refer to Section X.Y of the
   current HTTP/1.1 specification [11]. In addition, we describe SIP in
   both prose and an augmented Backus-Naur form (BNF) [H2.1] described
   in detail in [21].

   All SIP messages are text-based and use HTTP/1.1 conventions [H4.1],
   except for the additional ability of SIP to 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 normally be larger than the path maximum transmission unit
   (MTU) if the MTU is known, or 1500 bytes if the MTU is unknown.


        The 1400 bytes accommodates lower-layer packet headers
        within the "typical" MTU of around 1500 bytes. There are
        few MTU values around 1 kB; the next value is 1006 bytes
        for SLIP and 296 for low-delay PPP [22]. Recent studies
        [23] indicate that an MTU of 1500 bytes is a reasonable
        assumption. 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  ; SIP messages


   Both  Request (section 4) and  Response (section 5) messages use the
   generic message format of RFC 822 [24] for transferring entities (the
   payload of the message). Both types of message 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 header describing the message body as entity



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   headers.  These components are described in detail in the upcoming
   sections.



        generic-message    =    start-line
                                *message-header
                                CRLF
                                [ message-body ]

        start-line         =    Request-Line |       Section 4.1
                                Status-Line          Section 5.1




        message-header    =    *( general-header
                               | request-header
                               | 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 a  CRLF, the  CRLF should 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.





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        general-header     =     Call-ID                ; Section 6.12
                           |     CSeq                   ; Section 6.26
                           |     Date                   ; Section 6.15
                           |     Expires                ; Section 6.16
                           |     From                   ; Section 6.17
                           |     Via                    ; Section 6.33
        entity-header      =     Content-Length         ; Section 6.13
                           |     Content-Type           ; Section 6.14
        request-header     =     Accept                 ; Section 6.6
                           |     Accept-Language        ; Section 6.7
                           |     Authorization          ; Section 6.10
                           |     Call-Disposition       ; Section 6.11
                           |     Organization           ; Section 6.19
                           |     Priority               ; Section 6.20
                           |     Proxy-Authorization    ; Section 6.22
                           |     Require                ; Section 6.24
                           |     Subject                ; Section 6.28
                           |     To                     ; Section 6.31
                           |     User-Agent             ; Section 6.32
        response-header    =     Location               ; Section 6.18
                           |     Proxy-Authenticate     ; Section 6.21
                           |     Public                 ; Section 6.23
                           |     Retry-After            ; Section 6.25
                           |     Server                 ; Section 6.27
                           |     Unsupported            ; Section 6.29
                           |     Warning                ; Section 6.34
                           |     WWW-Authenticate       ; Section 6.35


   Table 2: SIP headers

        Request-Line  ___   Method SP Request-URI SP SIP-Version CRLF


4.2 Methods

   The methods are defined below. Methods that are not supported by a
   proxy or redirect server SHOULD be treated by that server as if they
   were an  INVITE method and forwarded accordingly.

   Methods that are not supported by a user agent server should cause a
   "501 Not Implemented" response to be returned (Section 7).



        method    =    "INVITE" | "ACK" | "OPTIONS"



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                 |     "BYE" | "REGISTER" | "UNREGISTER"


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 the callee is being invited to. For two-
   party calls, the caller indicates the type of media it is able to
   receive as well as their parameters such as network destination. If
   the session description format allows this, it may also indicate
   "send-only" media. A success response indicates in its message body
   which media the callee wishes to receive.

   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.

   A user agent MUST check any version identifiers in the session
   description to see if it has changed. If the version number has
   changed, the user agent server MUST adjust the session parameters
   accordingly, possibly after asking the user for confirmation.
   (Versioning of the session description may be used to accomodate the
   capabilities of new arrivals to a conference or change from a unicast
   to a multicast conference.)

   This method MUST be supported by a SIP server.

4.2.2  ACK

   ACK request confirms that the client has received a final response to
   an  INVITE request. See Section 11 for details. This method MUST be
   supported by a SIP server and client.

4.2.3  OPTIONS

   The client 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. Support of this method is OPTIONAL.

4.2.4  BYE

   The client indicates to the server that it wishes to abort the call
   attempt. The leaving party can use a  Location header field to
   indicate that the recipient of request should contact the named
   address.  This implements the "call transfer" telephony



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   functionality. A client SHOULD also use this method to indicate to
   the callee that it wishes to abort an on-going call attempt.

        With UDP, the caller has no other way to signal her intent
        to drop the call attempt and the callee side will keep
        "ringing".  When using TCP, a client MAY also close the
        connection to abort a call attempt. Support of this method
        is OPTIONAL.

   Support of this method is OPTIONAL.

4.2.5  REGISTER

   A client uses the  REGISTER method to register the address listed in
   the request line to a SIP server. The host part of the request-URI
   SHOULD correspond to (one of the aliases of) name of the server or to
   the domain that it represents, if location-independent. After
   registration, the server MAY forward incoming SIP requests to the the
   network source address and port from the registration request. A
   server SHOULD silently drop the registration after one hour, unless
   refreshed by the client. A client may request and a server may
   indicate or lower or higher refresh interval and indicate the
   interval through the Expires header (Section 6.16). A single address
   (if host-independent) may be registered from several different
   clients.

   If the request contains a  Location header, requests for the
   request-URI will be directed to the address(es) given.

   Support of this method is OPTIONAL.


        Beyond its use as a simple location service, this method is
        needed if there are several SIP servers on a single host,
        so that some cannot use the default port number. Each such
        server would register with a server for the administrative
        domain.

4.2.6  UNREGISTER

   A client cancels an existing registration established for the
   Request-URI with  REGISTER with the  UNREGISTER method. If it
   unregisters a  Request-URI unknown to the servers, the server returns
   a 200 (OK) response. Support of this method is OPTIONAL.

4.3 Request-URI

   The  Request-URI field is a SIP URL as described in Section 2 or a



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   general URI. It indicates the user or service that this request is
   being addressed to. Unlike the  To field, the  Request-URI field may
   be re-written by proxies. For example, a proxy may perform a lookup
   on the contents of the  To field to resolve a username from a mail
   alias, and then use this username as part of the  Request-URI field
   of requests it generates.

   If a SIP server receives a request contain 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
   field contains a scheme it does understand.

4.3.1 SIP Version

   Both request and response messages include the version of SIP in use,
   and basically follow [H3.1], with HTTP replaced by SIP. To be
   conditionally 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 6.24) and Unsupported
   (Section 6.29) fields.

   Syntax:


        option-tag  ___   1*OCTET   ; LWS must be URL-escaped


   The creator of a new SIP option should either prefix the option with
   a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name
   for a feature whose inventor can be reached at "foo.com"), or
   register the new option with the Internet Assigned Numbers Authority
   (IANA).

4.4.1 Registering New Option Tags with IANA

   When registering a new SIP option, the following information should
   be provided:

        oName and description of option. The name may be of any length,
         but SHOULD be no more than twenty characters long. The name
         should not contain any spaces, control characters or periods.

        oIndication of who has change control over the option (for
         example, IETF, ISO, ITU-T, other international standardization



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         bodies, a consortium or a particular company or group of
         companies);

        oA 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;

        oFor proprietary options, contact information (postal and email
         address);


        Borrowed from RTSP and the RTP AVP.

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


   [H6] applies except that  HTTP-Version is replaced by SIP-Version.
   Also, SIP defines additional response codes and does not use some
   HTTP codes.

5.1 Status-Line

   The first line of a  Response message is the  Status-Line, consisting
   of the protocol version ((Section 4.3.1) followed by a numeric
   Status-Code and its associated textual phrase, with each element
   separated by SP characters. No  CR or LF is allowed except in the
   final  CRLF sequence.



        Status-Line  ___   SIP-version SP Status-Code SP Reason-Phrase CRLF


5.1.1 Status Codes and Reason Phrases



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   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.

   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 process;

   2xx: Success -- the action was successfully received, understood, and
        accepted;

   3xx: Redirection -- further action must 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 is invalid at any server.

   Presented below are the individual values of the numeric response
   codes, and an example set of corresponding reason phrases for
   SIP/2.0. These reason phrases are only recommended; they may be
   replaced by local equivalents without affecting the protocol. Note
   that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
   codes in the range starting at x80 to avoid conflicts with newly
   defined HTTP response codes, and extends these response codes in the
   6xx range.



        Status-Code       =    Informational                  Fig. 3
                         |     Success                        Fig. 3
                         |     Redirection                    Fig. 4
                         |     Client-Error                   Fig. 5
                         |     Server-Error                   Fig. 6
                         |     Global-Failure                 Fig. 7
                         |     extension-code
        extension-code    =    3DIGIT
        Reason-Phrase     =    *<TEXT,  excluding CR, LF>



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        Informational    =    "100"    ;  Trying
                        |     "180"    ;  Ringing
                        |     "181"    ;  Queued




        Success    =    "200"    ;  OK


   Figure 3: 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 4: Redirection status 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 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.

6 Header Field Definitions



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        Client-Error    =    "400"    ;  Bad Request
                       |     "401"    ;  Unauthorized
                       |     "402"    ;  Payment Required
                       |     "403"    ;  Forbidden
                       |     "404"    ;  Not Found
                       |     "405"    ;  Method Not Allowed
                       |     "407"    ;  Proxy Authentication Required
                       |     "408"    ;  Request Timeout
                       |     "409"    ;  Conflict
                       |     "410"    ;  Gone
                       |     "411"    ;  Length Required
                       |     "412"    ;  Precondition Failed
                       |     "413"    ;  Request Message Body Too Large
                       |     "414"    ;  Request-URI Too Large
                       |     "415"    ;  Unsupported Media Type
                       |     "420"    ;  Bad Extension
                       |     "480"    ;  Temporarily not available
                       |     "481"    ;  Invalid Call-ID
                       |     "482"    ;  Loop Detected


   Figure 5: Client error status codes




        Server-Error    =    "500"    ;  Internal Server Error
                       |     "501"    ;  Not Implemented
                       |     "502"    ;  Bad Gateway
                       |     "503"    ;  Service Unavailable
                       |     "504"    ;  Gateway Timeout
                       |     "505"    ;  SIP Version not supported


   Figure 6: Server error status codes


   SIP header fields are similar to HTTP header fields in both syntax
   and semantics [H4.2], [H14]. In general the ordering of the header
   fields is not of importance (with the exception of  Via fields, see
   below), but proxies MUST NOT reorder or otherwise modify header
   fields other than by adding a new  Via field. This allows an
   authentication field to be added after the  Via fields that will not
   be invalidated by proxies.

   The header fields required, optional and not applicable for each



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        Global-Failure   |    "600"    ;  Busy
                         |    "603"    ;  Decline
                         |    "604"    ;  Does not exist anywhere
                         |    "606"    ;  Not Acceptable


   Figure 7: Global failure status Codes


   method are listed in Table 3. The  Content-Type and Content-Length
   headers are required when there is a valid message body (of non-zero
   length) associated with the message (Section 8).


   Other headers may be added as required; a server MAY ignore headers
   that it does not understand. A compact form of these header fields is
   also defined in Section 10 for use over UDP when the request has to
   fit into a single packet and size is an issue.

6.1 General Header Fields

   There are a few header fields that have general applicability for
   both request and response messages. These header fields apply only to
   the message being transmitted.

   General-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 general
   header fields if all parties in the communication recognize them to
   be general-header fields.

6.2 Entity Header Fields

   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 reply
   body may contain a transformed version of the message body. The
   original message body is referred to as the "entity". We retain the
   same terminology for header fields but usually refer to the "message
   body" rather then the entity as the two are the same in SIP.

6.3 Request Header Fields

   The  request-header fields allow the client to pass additional
   information about the request, and about the client itself, to the
   server. These fields act as request modifiers, with semantics



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                             type    ACK   BYE   INV   OPT   REG   UNR
     _________________________________________________________________
     Accept                   R       o     -     o     o     o     o
     Accept-Language          R       o     o     o     o     o     o
     Allow                   405      o     o     o     o     o     o
     Also                     R       -     -     o     -     -     -
     Authorization            R       o     o     o     o     o     o
     Call-Disposition         R       -     o     o     -     -     -
     Call-ID                  g       m     m     m     o     -     -
     Content-Length           g       -     -     *     *     -     -
     Content-Type             g       -     -     *     *     -     -
     CSeq                     g       o     o     o     o     o     o
     Date                     g       o     o     o     o     o     o
     Expires                  g       -     -     o     o     o     -
     From                     R       m     m     m     m     o     o
     Location                 R       -     o     -     -     o     -
     Location                 r       -     -     o     o     -     -
     Organization             R       -     -     o     o     -     -
     Proxy-Authenticate       R       o     o     o     o     o     o
     Proxy-Authorization      R       o     o     o     o     o     o
     Priority                 R       -     -     o     -     -     -
     Public                   r       -     -     -     o     -     -
     Require                  R       o     o     o     o     o     o
     Retry-After           600,603    -     -     o     -     -     -
     Server                   r       o     o     o     o     o     o
     Subject                  R       -     -     o     -     -     -
     Timestamp                g       o     o     o     o     o     o
     To                       g       m     m     m     m     m     m
     Unsupported              r       o     o     o     o     o     o
     User-Agent               R       o     o     o     o     o     o
     Via                      g       m     m     m     m     m     m
     Warning                  r       o     o     o     o     o     o
     WWW-Authenticate        401      o     o     o     o     o     o


   Table 3: Summary of header fields. "o": optional, "m": mandatory,  "-
   ":  not  applicable,  "R': request header, "r": response header, "g":
   general header, "*": needed if message body is not empty.  A  numeric
   value in the "type" column indicates the status code the header field
   is used with.

   equivalent to the parameters on a programming language method
   invocation.

   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


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   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 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 SHOULD follow HTTP "common form"
   when generating these constructs, since there might exist some
   implementations that fail to accept anything beyond the common forms.



        message-header    =    field-name ":" [ field-value ] CRLF
        field-name        =    token
        field-value       =    *( field-content | LWS )
        field-content     =    < the OCTETs  making up the field-value
                                and consisting of either *TEXT or combinations
                                of token, tspecials, and quoted-string>


   The order in which header fields are received is not significant if
   the header fields have different field names. 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,



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   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.

6.6 Accept

   See [H14.1] for syntax. This request header field is used only with
   the OPTIONS and  INVITE request methods to indicate what description
   formats are acceptable in the response.

   Example:


     Accept: application/sdp;level=1, application/x-private



6.7 Accept-Language

   See [H14.4] for syntax. The  Accept-Language request header can be
   used to allow the client to indicate to the server in which language
   it would prefer to receive reason phrases. This may also be used as a
   hint by the proxy as to which destination to connect the call to
   (e.g., for selecting a human operator).

   Example:


     Accept-Language: da, en-gb;q=0.8, en;q=0.7



6.8 Allow

   See [H14.7].

6.9 Also

   The  Also request header advises the callee to send invitations to
   the addresses listed. This supports third-party call initiation
   (Section 13).





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        Also  ___   "Also" ":" 1#( SIP-URL ) [ comment ]


   Example:


     Also: sip://jones@foo.com, sip://mueller@bar.edu



6.10 Authorization

   See [H14.8].

6.11 Call-Disposition

   The  Call-Disposition request header field allows the client to
   indicate how the server is to handle the call. The following options
   can be used singly or in combination:

   all: If the user part of the SIP request address identifies a group
        rather than an individual, the " all" feature indicates that all
        members of the group should be alerted rather than the default
        of locating the first available individual from that group.
        Section 1.4.1 describes the behavior of proxy servers when
        resolving group aliases.

   do-not-forward: The "do-not-forward" request prohibits proxies from
        forwarding the call to another individual (e.g., the call is
        personal or the caller does not want to be shunted to a
        secretary if the line is busy.)

   queue: If the called party is temporarily unreachable, e.g., because
        it is in another call, the caller can indicate that it wants to
        have its call queued rather than rejected immediately. If the
        call is queued, the server returns "181 Queued" (see Section
        7.1.3). A pending call be terminated by a  BYE request (Section
        4.2.4).



        Call-Disposition  ___   "Call-Disposition" ":" 1#( "all" | "do-not-forward"
                           |    "queue" )


   Example:

     Call-Disposition: all, do-not-forward, queue



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        HS: This header is experimental. The name is based on the
        SMTP Content-Disposition header.

6.12 Call-ID

   The  Call-ID general header uniquely identifies a particular
   invitation. Note that a single multimedia conference may give rise to
   several calls with different  Call-IDs, e.g., if a user invites
   several different people. Since the  Call-ID is unique for each
   caller, a user may invited to the same conference using several
   different  Call-IDs. If desired, it must use identifiers within the
   session description to detect this duplication. Calls to different
   callee MUST always use different  Call-IDs unless they are the result
   of a proxy server "forking" a single request.

   The  Call-ID may be any URL-encoded string that can be guaranteed to
   be globally unique for the duration of the request. Using the
   initiator's IP-address, process id, and instance (if more than one
   request is being made simultaneously) satisfies this requirement.

   The form  local-id@host is recommended, where  host is either the
   fully qualified domain name or a globally routable IP address, and
   local-id depends on the application and operating system of the host,
   but is an ID that can be guaranteed to be unique during this session
   initiation request.



        Call-ID  ___   ( "Call-ID" | "i" ) ":" atom "@" host


   Example:

     Call-ID: 9707211351.AA08181@foo.bar.com



6.13 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" ":" 1*DIGIT


   An example is



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     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 MAY be omitted or set to zero.  Section 8 describes how to
   determine the length of the message body.

6.14 Content-Type

   The  Content-Type entity-header field indicates the media type of the
   message-body sent to the recipient.



        Content-Type  ___   "Content-Type" ":" media-type


   An example of the field is

     Content-Type: application/sdp



6.15 Date

   General header field. See [H14.19].


        The  Date header field is useful for simple devices without
        their own clock.

6.16 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 and allows the client to indicate how long the registration
   should be valid; the server uses it to indicate when the client has
   to re-register. The server's choice overrides that of the client. The
   server MAY choose a shorter time interval than that requested by the
   client, but SHOULD not choose a longer one.




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   For  INVITE, it is a request and response-header field. In a request,
   the callee can limit the validity of an invitation. (For example, if
   a client wants to limit how long a search should take at most or when
   a conference being invited to is time-limited. A user interface may
   take this is as a hint to leave the invitation window on the screen
   even if the user is not currently at the workstation.) In a 302
   response, a server can advise the client of the maximal duration of
   the redirection.

   The value of this field can be either an  HTTP-date or an integer
   number of seconds (in decimal), measured from the receipt of the
   request.



        Expires  ___   "Expires" ":" ( HTTP-date | delta-seconds )


   Two example of its use are

     Expires: Thu, 01 Dec 1994 16:00:00 GMT
     Expires: 5



6.17 From

   Requests MUST and responses SHOULD contain a  From header field,
   indicating the invitation initiator. The field MUST be a SIP URL as
   defined in Section 2. Only a single initiator and a single invited
   user are allowed to be specified in a single SIP request.  The sense
   of  To and  From header fields is maintained from request to
   response, i.e., if the  From header is sip://bob@example.edu in the
   request then it is MUST also be sip://bob@example.edu in the response
   to that request.

   The  From field is a URL and not a simple SIP address (Section 1.4.1
   address to allow a gateway to relay a call into a SIP request and
   still produce an appropriate  From field.



        From  ___   ( "From" | "f" ) ":" *1( ( SIP-URL | URL ) [ comment ] )


   Examples:





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     From: agb@bell-telephone.com (A. G. Bell)
     From: +12125551212@server.phone2net.com



6.18 Location

   The  Location response header can be used with a 2xx or 3xx response
   codes to indicate a new location to try. It contains a URL giving the
   new location or username to try, or may simply specify additional
   transport parameters. A "301 Moved Permanently" or "302 Moved
   Temporarily" response SHOULD contain a  Location field containing the
   URL giving a new address to try. A 301 or 302 response may also give
   the same location and username that was being tried but specify
   additional transport parameters such as a multicast address to try or
   a change of SIP transport from UDP to TCP or vice versa.

   A user agent or redirect server sending a definitive, positive
   response (2xx), SHOULD insert a  Location response header indicating
   the SIP address under which it is reachable most directly for future
   SIP requests. This may be the address of the server itself or that of
   a proxy (e.g., if the host is behind a firewall).

   A  Location response header may contain any suitable URL indicating
   where the called party may be reached, not limited to SIP URLs. For
   example, it may contain a phone or fax URL [25], a mailto: URL [26]
   or  irc.

   The following parameters are defined:

   q: The  qvalue indicates the relative preference among the locations
        given.  qvalue values are decimal numbers from 0.0 to 1.0, with
        higher values indicating higher preference.

   class: The class parameter whether this terminal is found in a
        residential or business setting. (A caller may defer a personal
        call if only a business line is available, for example.)

   description: The description field further describes, as text, the
        terminal. It is expected that the user interface will render
        this text.

   duplex: The duplex parameter lists whether the terminal can
        simultaneously send and receive ("full"), alternate between
        sending and receiving ("half"), can only receive ("receive-
        only") or only send ("send-only"). Typically, a caller will
        prefer a full-duplex terminal over a half-duplex terminal and
        these over receive-only or send-only terminals.



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   features: The feature list enumerates additional features of this
        terminal. Values for this field are for further study.

   language: The language parameter lists, in order of preference, the
        languages spoken by the person answering. This feature may be
        used to have a caller automatically select the appropriate
        attendant or customer service representative, without having to
        declare its own language skills.

   media: The media tag lists the media types supported by the terminal.
        Currently, the names defined in SDP may be used [9]: "audio",
        "video", "whiteboard", "text" and "data".

   mobility: The mobility parameter indicates if the terminal is fixed
        or mobile. In some locales, this may affect voice quality or
        charges.

   priority: The priority tag indicates the minimum priority level this
        terminal is to be used for. It can be used for automatically
        restricting the choice of terminals available to the user.

   service: The service tag describes what service is being provided by
        the terminal.



        Location              =    ( "Location" | "m" ) ( SIP-URL | URL )
                                   *( ";" location-params )
        extension-name       =     token
        extension-value      =     *( token | quoted-string | LWS | extension-specials)
        extension-specials   =      < any element of  tspecials except <"> >
        language-tag         =     <  see [H3.10] >
        priority-tag         =     "urgent" | "normal" | "non-urgent"
        service-tag          =     "fax" | "IP" | "PSTN" | "ISDN" | "pager"
        media-tag            =      < see SDP: "audio" | "video" | "email" ...
        feature-list         =     "voice-mail" | "attendant"




        location-params       =    "q"                     "="    qvalue
                              |    "class"                 "="    ( "personal" | "business" )
                              |    "description"           "="    quoted-string
                              |    "duplex"                "="    ( "full" | "half" |
                                                                  "receive-only" | "send-only" )
                              |    "features"              "="    1# feature-list
                              |    "language"              "="    1# language-tag
                              |    "media"                 "="    1# media-tag



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                              |    "mobility"              "="    ( "fixed" | "mobile" )
                              |    "priority"              "="    1# priority-tag
                              |    "service"               "="    1# service-tag
                              |    extension-attributes
        extension-attribute   =    extension-name          "="    extension-value


   Examples:


     Location: sip://watson@worcester.bell-telephone.com ;service=IP,voice-mail
               ;media=audio ;duplex=full ;q=0.7;
     Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed;
               language=en,es,iw ;q=0.5
     Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
               duplex=send-only;media=text; q=0.1; priority=urgent;
               description="For emergencies only"
     Location: mailto:watson@bell-telephone.com
     Location: http://www.bell-telephone.com/~watson



   Attributes which are unknown should be omitted. New tags for class-
   tag and  service-tag can be registered with IANA. The media tag uses
   Internet media types, e.g., audio, video, application/x-wb, etc. This
   is meant for indicating general communication capability, sufficient
   for the caller to choose an appropriate address.

6.19 Organization

   The Organization request-header fields conveys the name of the
   organization to which the callee belongs. It may be inserted by
   proxies at the boundary of an organization and may be used by client
   software to filter calls.

6.20 Priority

   The priority request header signals the urgency of the call to the
   callee.



        Priority          =    "Priority" ":" priority-value
        priority-value    =    "urgent" | "normal" | "non-urgent"


   Example:




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     Subject: A tornado is heading our way!
     Priority: urgent



6.21 Proxy-Authenticate

   See [H14.33].

6.22 Proxy-Authorization

   See [H14.34].

6.23 Public

   See [H14.35].

6.24 Require

   The  Require header is used by clients to query the server about
   options that it may or may not support. The server MUST respond to
   this header 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
   optimally 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.




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        We explored using the W3C's PEP proposal  for this
        functionality. However,  Require,  Proxy-Require, and
        Unsupported allow the addition of extensions with far less
        complexity.

   This field roughly corresponds to the PEP field in the PEP draft.

6.25 Retry-After

   The  Retry-After response 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", "603 Decline" response to indicate when the
   called party may be available again. The value of this field can be
   either an HTTP-date or an integer number of seconds (in decimal)
   after the time of the response. An optional comment can be used to
   indicate additional information about the time of callback. An
   optional duration parameter indicates how long the called party will
   be reachable starting at the initial time of availability.



        Retry-After  ___   "Retry-After" ":" ( HTTP-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)
     Retry-After: Mon,  1 Jan 9999 00:00:00 GMT
       (Dear John: Don't call me back, ever)
     Retry-After: Fri, 26 Sep 1997 21:00:00 GMD;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.

6.26 CSeq

   The  CSeq (command sequence) header field MAY be added by a SIP
   client making a request if it needs to distinguish responses to
   several consecutive requests sent with the same  Call-ID. A  CSeq
   field contains a single decimal sequence number chosen by the
   requesting client. Consecutive different requests made with the same
   Call-ID MUST contain strictly monotonically increasing sequence
   numbers; the sequence space MAY NOT be contiguous. Retransmissions of



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   the same request carry the same sequence number. A server responding
   to a request containing a sequence number MUST echo the sequence
   number back in the response. The  ACK request MUST contain the same
   CSeq value as the  INVITE request that it refers to.



        CSeq = "CSeq" ":" 1*DIGIT


   CSeq header fields are NOT needed for SIP requests using the INVITE
   or  OPTIONS methods but may be needed for future methods.

   Example:


     CSeq: 4711



6.27 Server

   See [H14.39].

6.28 Subject

   This is intended to provide a summary, or indicate the nature, of the
   call, allowing call filtering without having to parse the session
   description. (Also, the session description may not necessarily use
   the same subject indication as the invitation.)



        Subject  ___   ( "Subject" | "s" ) ":" *text


   Example:


     Subject: Tune in - they are talking about your work!



6.29 Unsupported

   The  Unsupported response header lists the features not supported by
   the server.




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   See Section 6.24 for a usage example and motivation.

6.30 Timestamp

   The timestamp general header describes when the client sent the
   request to the server. The value of the timestamp is of significance
   only to the client and 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 has 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) ]


6.31 To

   The  To request header field specifies the invited user, with the
   same SIP URL syntax as the  From field.



        To = ( "To" | "t" ) ":" ( SIP-URL | URL ) [ comment ]


   If a SIP server receives a request destined  To a URL indicating a
   scheme other than SIP and that is unknown to it, the server returns a
   "400 bad request" response.

   Example:

     To: sip://operator@cs.columbia.edu (The Operator)



6.32 User-Agent

   See [H14.42].

6.33 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



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   routing situations.

   The client originating the request MUST insert a  Via field
   containing its address to the request. Each subsequent proxy server
   that sends the request onwards MUST add its own additional  Via
   field, which MUST be added before any existing  Via fields.
   Additionally, if the message goes to a multicast address, an extra
   Via field is added before all the others giving the multicast address
   and TTL.

   If a proxy server receives a request which contains its own address,
   it MUST respond with a "482 Loop Detected" status code. (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.)

   In the return path,  Via fields are processed by a proxy or client
   according to the following rules:

        oIf the first  Via field in the reply received is the client's
         or server's local address, remove the  Via field and process
         the reply.

        oIf the first  Via field in a reply is a multicast address,
         remove that  Via field before sending to the multicast address.

   These rules ensure that a proxy server only has to check the first
   Via field in a reply to see if it needs processing.

   The format for a  Via header is:


        Via                   =    ( "Via" | "v") ":" 1#( sent-protocol sent-by
                                   *( ";" via-params ) [ comment ] )
        via-params            =    "ttl" "=" ttl
        sent-protocol         =    [ protocol-name "/" ] protocol-version
        [ "/" transport ]
        protocol-name         =    "SIP" | token
        protocol-version      =    token
        transport             =    "UDP" | "TCP"
        sent-by               =    host [ ":" port ]
        ttl                   =    1*3DIGIT                                         ; 0 to 255


   The "ttl" parameter is included only if the address is a multicast
   address.



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   Example:


     Via: SIP/2.0/UDP first.example.com:4000



6.34 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        =    2DIGIT
        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 and character set that
   is most likely to be intelligible to the human user receiving the
   response. This decision may be based on any available knowledge, such
   as the location of the cache or user, the  Accept-Language field in a
   request, the  Content-Language field in a response, etc. The default
   language is English.

   Any server may add  Warning headers to a response. New Warning
   headers should be added after any existing  Warning headers. A proxy
   server MUST NOT delete any  Warning header 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 should follow these heuristics:

        oWarnings that appear early in the response take priority over
         those appearing later in the response.

        oWarnings in the user's preferred character set take priority
         over warnings in other character sets but with identical



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         warn-codes and  warn-agents.

   Systems that generate multiple  Warning headers should order them
   with this user agent behavior in mind.

   Example:


     Warning: 606.4 isi.edu Multicast not available
     Warning: 606.2 isi.edu Incompatible protocol (RTP/XXP)



6.35 WWW-Authenticate

   See [H14.46].

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. 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. If UDP transport is being used, the client SHOULD
   periodically re-send the request in case the final response is lost.
   Typically a server should send a "1xx" response if it expects to take
   more than one second to obtain a final reply.

7.1.1 100 Trying

   Some further action is being taken (e.g., the request is being
   forwarded) but the user has not yet been located.

7.1.2 180 Ringing

   The user agent or conference server has located a possible location
   where the user has been recently and is trying to alert them.




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7.1.3 181 Queued

   The called party was temporarily unavailable, but the caller
   indicated via a "Call-Disposition: Queue" directive (Section 6.11) 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 181 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 was successful in contacting the user, and the user has
   agreed to participate.

7.3 Redirection 3xx

   3xx responses give information about the user's new location, or
   about alternative services that may be able to satisfy the call.
   They SHOULD terminate an existing search, and MAY cause the initiator
   to begin a new search if appropriate.

7.3.1 300 Multiple Choices

   The requested resource corresponds to any one of a set of
   representations, each with its own specific location, and agent-
   driven negotiation (i.e., controlled by the SIP client) is being
   provided so that 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. The entity format is specified by
   the media type given in the  Content-Type header field. Depending
   upon the format and the capabilities of the user agent, selection of
   the most appropriate choice may be performed automatically. However,
   this specification does not define any standard for such automatic
   selection.

   The choices SHOULD also be listed as  Location fields (Section 6.18).
   Unlike HTTP, the SIP response may contain several  Location fields.
   User agents MAY use the  Location field value for automatic
   redirection or MAY ask the user to confirm a choice.



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7.3.2 301 Moved Permanently

   The requesting client should retry on the new address given by the
   Location field because the user has permanently moved and the address
   this response is in reply to is no longer a current address for the
   user. A 301 response MUST NOT suggest any of the hosts in the  Via
   (Section 6.33) path of the request as the user's new location.

7.3.3 302 Moved Temporarily

   The requesting client should retry on the new address(es) given by
   the Location header. A 302 response MUST NOT suggest any of the hosts
   in the  Via (Section 6.33) path of the request as the user's new
   location.  The duration of the redirection can be indicated through
   an  Expires (Section 6.16) header.

7.3.4 380 Alternative Service

   The call was not successful, but alternative services are possible.
   The alternative services are described in the message body of the
   response.

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 may 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




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   The server has definitive information that the user does not exist at
   the domain specified in the  Request-URI.

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 containing a list of valid methods for the indicated
   address.

7.4.7 407 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 6.21) 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 6.22). SIP access authentication is explained
   in section [H11].

   This status code should be used for applications where access to the
   communication channel (e.g., a telephony gateway) rather than the
   callee herself requires authentication.

7.4.8 408 Request Timeout

   The client did not produce a request within the time that the server
   was prepared to wait. The client MAY repeat the request without
   modifications at any later time.

7.4.9 420 Bad Extension

   The server did not understand the protocol extension specified with
   strength "must".

7.4.10 480 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 may also be available elsewhere (unbeknownst to this host),
   thus, this response does not terminate any searches. The reason
   phrase SHOULD indicate the more precise cause as to why the callee is
   unavailable.  This value SHOULD be setable by the user agent.

7.4.11 481 Invalid Call-ID




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   The server received a  BYE or  ACK request with a Call-ID value it
   does not recognize.

7.4.12 482 Loop Detected

   The server received a request with a  Via path containing itself.

7.5 Server Failure 5xx

   5xx responses are failure responses given when a server itself has
   erred. They are not definitive failures, and SHOULD 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.

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 upstream server it accessed in attempting to
   fulfill the request.

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 SHOULD
   handle the response as it would for a 500 response.

   Note: The existence of the 503 status code does not imply that a
   server must use it when becoming overloaded. Some servers may wish to
   simply refuse the connection.

7.5.5 504 Gateway Timeout

   The server, while acting as a gateway, did not receive a timely
   response from the upstream server (e.g., a location server) it



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   accessed in attempting to complete the request.

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

   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 should use status code 680 instead.

7.6.2 603 Decline

   The callee's machine was successfully contacted but the user
   explicitly does not wish to 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 profile (the requested media, bandwidth, or addressing style)
   were not acceptable.

   A "606 Not Acceptable" reply means that the user wishes to
   communicate, but cannot adequately support the session described. The
   "606 Not Acceptable" reply MAY contain a list of reasons in a Warning
   header describing why the session described cannot be supported.
   These reasons can be one or more of:

   606.1 Insufficient Bandwidth: The bandwidth specified in the session
        description or defined by the media exceeds that known to be
        available.

   606.2 Incompatible Protocol: One or more protocols described in the
        request are not available.




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   606.3 Incompatible Format: One or more media formats described in the
        request is not available.

   606.4 Multicast not available: The site where the user is located
        does not support multicast.

   606.5 Unicast not available: The site where the user is located does
        not support unicast communication (usually due to the presence
        of a firewall).

   Other reasons are likely to be added later. It is hoped that
   negotiation will not frequently be needed, and when a new user is
   being invited to join a pre-existing lightweight session, negotiation
   may not be possible. It is up to the invitation initiator to decide
   whether or not to act on a "606 Not Acceptable" reply.

8 SIP Message Body

   The session description body gives details of the session the user is
   being invited to join. Its Internet media type MUST be given by the
   Content-type header field, and the body length in bytes MUST be given
   by the  Content-Length 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.

8.1 Body Inclusion

   For a request message, the presence of a body is signaled by the
   inclusion of a  Content-Length header. A body may be included in a
   request only when the request method allows one.

   For response messages, whether or not a body is included is dependent
   on both the request method and the response message's response code.
   All 1xx informational responses MUST NOT include a body. All other
   responses MAY include a payload, although it may be of zero length.

8.2 Message Body Length

   If no body is present in a message, then the  Content-Length header
   MAY be omitted or set to zero. When a body is included, its length in
   bytes is indicated in the  Content-Length header and is determined by
   one of the following:

        1.   Any response message which MUST NOT include a body (such as
             the 1xx responses) is always terminated by the first empty
             line after the header fields, regardless if any  entity-
             header fields are present.



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        2.   Otherwise, a  Content-Length header MUST be present (this
             requirement differs from HTTP/1.1). Its value in bytes
             represents the length of the message body.

   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 Examples

9.1 Invitation to Multimedia Conference

   The first example invites schooler@vlsi.cs.caltech.edu to a multicast
   session.

9.1.1 Request


   C->S: INVITE schooler@vlsi.cs.caltech.edu SIP/2.0
         Via: SIP/2.0/UDP 239.128.16.254 16
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19
         From: mjh@isi.edu (Mark Handley)
         Subject: SIP will be discussed, too
         To: schooler@cs.caltech.edu (Eve Schooler)
         Call-ID: 62729-27@oregon.isi.edu
         Content-type: application/sdp
         CSeq: 4711
         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



   The  Via fields list the hosts along the path from invitation
   initiator (the first element of the list) towards the invitee. 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
   131.215.131.131




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   The request header above states that the request was initiated by
   mjh@isi.edu the host 128.16.64.19 schooler@cs.caltech.edu is being
   invited; the message is currently being routed to
   schooler@vlsi.cs.caltech.edu

   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.

9.1.2 Reply

   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 239.128.16.254 16
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19 1
         From: mjh@isi.edu
         Call-ID: 62729-27@128.16.64.19
         Location: sip://es@jove.cs.caltech.edu
         CSeq: 4711



   A sample reply to the invitation is given below. The first line of
   the reply states the SIP version number, that it is a "200 OK" reply,
   which means the request was successful. The  Via headers are taken
   from the request, and entries are removed hop by hop as the reply
   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  Location 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.


   S->C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP 239.128.16.254 16
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19 1
         From: mjh@isi.edu



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         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19
         Location: sip://es@jove.cs.caltech.edu
         CSeq: 4711



   The caller confirms the invitation by sending a request to the
   location named in the  Location header:


   C->S: ACK schooler@jove.cs.caltech.edu SIP/2.0
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19
         CSeq: 4711



9.2 Two-party Call

   A two-party call proceeds as above. The only difference is

   For two-party Internet phone calls, the response must contain a
   description of where to send data to. 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 watson@boston.bell-telephone.com SIP/2.0
         Via: SIP/2.0/UDP 169.130.12.5
         From: a.g.bell@bell-telephone.com (A. Bell)
         To: watson@bell-telephone.com (T. A. Watson)
         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

   S->C: SIP/2.0 200 OK
         From: a.g.bell@bell-telephone.com (A. Bell)
         To: watson@bell-telephone.com
         Call-ID: 187602141351@worcester.bell-telephone.com
         Location: sip://watson@boston.bell-telephone.com



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         Content-Length: ...

         v=0
         o=watson 4858949 4858949 IN IP4 192.1.2.3
         c=IN IP4 135.180.161.25
         m=audio 5004 RTP/AVP 0 3



   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 135.180.144.94, Bell will send to
   port 5004 at 135.180.161.25.

9.3 Aborting a Call

   If the caller wants to abort a pending call, it sends a  BYE request.


   C->S: BYE schooler@jove.cs.caltech.edu
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19



9.3.1 Redirects

   Replies with status codes "301 Moved Permanently" or "302 Moved
   Temporarily" SHOULD specify another location using the  Location
   field.


   S->C: SIP/2.0 302 Moved temporarily
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19
         Location: sip://239.128.16.254;ttl=16;transport=udp
         Content-length: 0



   In this example, the proxy located at 131.215.131.131 is being
   advised to contact the multicast group 239.128.16.254 with a ttl of
   16 and UDP transport. In normal situations, a server would not



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   suggest a redirect to a local multicast group unless, as in the above
   situation, it knows that the previous proxy or client is within the
   scope of the local group. If the request is redirected to a multicast
   group, a proxy server SHOULD query the multicast address itself
   rather than sending the redirect back towards the client as multicast
   may be scoped; this allows a proxy within the appropriate scope
   regions to make the query.

9.3.2 Alternative Services

   An example of a "350 Alternative Service" reply is:


   S->C: SIP/2.0 350 Alternative Service
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19
         Location: recorder@131.215.131.131
         Content-type: application/sdp
         Content-length: 146

         v=0
         o=mm-server 2523535 0 IN IP4 131.215.131.131
         s=Answering Machine
         i=Leave an audio message
         c=IN IP4 131.215.131.131
         t=0 0
         m=audio 12345 RTP/AVP 0



   In this case, the answering server provides a session description
   that describes an "answering machine". If the invitation initiator
   decides to take advantage of this service, it should send an
   invitation request to the answering machine at 131.215.131.131 with
   the session description provided (modified as appropriate for a
   unicast session to contain the appropriate local address and port for
   the invitation initiator). This request SHOULD contain a different
   Call-ID from the one in the original request. An example would be:


   C->S: INVITE mm-server@131.215.131.131 SIP/2.0
         Via: SIP/2.0/UDP 128.16.64.19
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-28@128.16.64.19



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         Content-type: application/sdp
         Content-length: 146

         v=0
         o=mm-server 2523535 0 IN IP4 131.215.131.131
         s=Answering Machine
         i=Leave an audio message
         c=IN IP4 128.16.64.19
         t=0 0
         m=audio 26472 RTP/AVP 0



   Invitation initiators MAY choose to treat a "350 Alternative Service"
   reply as a failure if they wish to do so.

9.3.3 Negotiation

   An example of a "606 Not Acceptable" reply is:


   S->C: SIP/2.0 606 Not Acceptable
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID:62729-27@128.16.64.19
         Location: mjh@131.215.131.131
         Warning: 606.1 Insufficient bandwidth (only have ISDN),
           606.3 Incompatible format,
           606.4 Multicast not available
         Content-Type: application/sdp
         Content-Length: 50

         v=0
         s=Lets talk
         b=CT:128
         c=IN IP4 131.215.131.131
         m=audio 3456 RTP/AVP 7 0 13
         m=video 2232 RTP/AVP 31



   In this example, the original request specified 256 kb/s total
   bandwidth, and the reply states that only 128 kb/s is available. The
   original request specified GSM audio, H.261 video, and WB whiteboard.
   The audio coding and whiteboard are not available, but the reply
   states that DVI, PCM or LPC audio could be supported in order of
   preference. The reply also states that multicast is not available.
   In such a case, it might be appropriate to set up a transcoding



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   gateway and re-invite the user.

9.4 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. In this case, Bob indicates that he can be
   reached at three different addresses, ranging from voice-over-IP to a
   PSTN phone to a pager.


   C->S: OPTIONS bob@example.com SIP/2.0
         From: alice@anywhere.org (Alice)
         To: bob@example.com (Bob)
         Accept: application/sdp

   S->C: SIP/2.0 200 OK
         Location: sip://bob@host.example.com ;service=IP,voice-mail
                   ;media=audio ;duplex=full ;q=0.7
         Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed;
                   language=en,es,iw ;q=0.5
         Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
                   duplex=send-only;media=text; q=0.1



   Alternatively, Bob could have returned a description of

   C->S: OPTIONS bob@example.com SIP/2.0
         From: alice@anywhere.org (Alice)
         To: bob@example.com (Bob)
         Accept: application/sdp

   S->C: SIP/2.0 200 OK
         Content-Length: 81
         Content-Type: application/sdp

         v=0
         m=audio 0 RTP/AVP 0 1 3 99
         m=video 0 RTP/AVP 29 30
         a:rtpmap:98 SX7300/8000



10 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



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   reply is larger than the MTU. To reduce this problem, a more compact
   form of SIP is also defined by using alternative names for common
   header fields.  These short forms are NOT abbreviations, they are
   field names. No other abbreviations are allowed.


   short field name    long field name      note
   c                    Content-Type
   e                    Content-Encoding
   f                    From
   i                    Call-ID
   l                    Content-Length
   m                    Location            from "moved"
   s                    Subject
   t                    To
   v                    Via


   Thus the header in section 9.1 could also be written:


     INVITE schooler@vlsi.caltech.edu SIP/2.0
     v:SIP/2.0/UDP 239.128.16.254 16
     v:SIP/2.0/UDP 131.215.131.131
     v:SIP/2.0/UDP 128.16.64.19
     f:mjh@isi.edu
     t:schooler@cs.caltech.edu
     i:62729-27@128.16.64.19
     c:application/sdp
     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



   Mixing short field names and long field names is allowed, but not
   recommended. Servers MUST accept both short and long field names for
   requests. Proxies MUST NOT translate a request between short and long
   forms if authentication fields are present.

11 SIP Transport



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   SIP is defined so it can use either UDP or TCP as a transport
   protocol.

11.1 Achieving Reliability For UDP Transport

11.1.1 General Operation

   SIP assumes no additional reliability from IP. Requests or replies
   may be lost. A SIP client SHOULD simply retransmit a SIP request
   periodically with timer T1 (default value of T1: once a second) until
   it receives a response, or until it has reached a set limit on the
   number of retransmissions. The default limit is 20.

   SIP requests and replies are matched up by the client using the
   Call-ID header field; thus, a server can only have one outstanding
   request per call at any given time.

   If the reply is a provisional response, the initiating client SHOULD
   continue retransmitting the request, albeit less frequently, using
   timer T2. The default retransmission interval T2 is 5 seconds.

   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 30 seconds to avoid having to, for example, contact the user or
   user location server again upon receiving a retransmission.

11.1.2 INVITE

   Special considerations apply for the  INVITE method.

        1.   After receiving an invitation, considerable time may elapse
             before the server can determine the outcome. For example,
             the called party may be "rung" or extensive searches may be
             performed, so 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 may be
             unbounded in time.

        It is undesirable to retransmit the  INVITE request, as this
        introduces additional network traffic. The retransmission
        interval would have to be no more than about a second, since the
        callee would encounter a "dead" voice path if the "200 OK"
        response is lost.

        2.   It is possible that the invitation request reaches the
             callee and the callee is willing to take the call, but that
             the final response (200 OK, in this case) is lost on the



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             way to the caller. If the session still exists but the
             initiator gives up on including the user, the contacted
             user has sufficient information to be able to join the
             session. However, if the session no longer exists because
             the invitation initiator "hung up" before the reply arrived
             and the session was to be two-way, the conferencing system
             should be prepared to deal with this circumstance.

        3.   If a telephony user interface is modeled or if we need to
             interface to the PSTN, the caller will provide "ringback",
             a signal that the callee is being alerted. 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.

        4.   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. One cannot rely on the
             absence of request retransmission, since the server would
             have to continue honoring the request for several request
             retransmission periods, that is, possible tens of seconds
             if only one or two packets can be lost.

   The first problem is solved by indicating progress to the caller: the
   server returns a provisional response indicating it is searching or
   ringing the user.

   The second and third problems are solved by having the server
   retransmit the final response at intervals of T3 (default value of T3
   = 2 seconds) until it receives an  ACK request for the same Call-ID
   and  CSeq or until it has retransmitted the final response 10 times.
   The  ACK request is acknowledged only once. If the request is
   syntactically valid and the  Request-URI matches that in the  INVITED
   request with the same  Call-ID, the server answers with status code
   200, otherwise with status code 400.

   Fig. 8 and 9 show the client and server state diagram for
   invitations.



11.2 Connection Management for TCP

   A single TCP connection can serve one or more SIP transactions. A
   transaction contains zero or more provisional responses followed by
   exactly one final response.



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                 +===========+
                 |  Initial  |
                 +===========+
                       |
                       |
                       |    -
                       |  ------
                       |  INVITE
           +------v    v
          T1     +-----------+
        ------   |  Calling  |--------+
        INVITE   +-----------+        |
           +------| |  |              |
   +----------------+  |              |
   |                   | 1xx          |  >= 200
   |                   | ---          |  ------
   |                   |  -           |   ACK
   |                   |              |
   |       +------v    v    v-----|   |
   |      T2     +-----------+   1xx  |
   |    ------   |  Ringing  |   ---  |
   |    INVITE   +-----------+    -   |
   |       +------|    |    |-----+   |
   |                   |              |
   |     2xx           |              |
   |     ---           | 2xx          |
   |     ACK           | ---          |
   |                   | ACK          |
   +----------------+  |              |
           +------v |  v              |
          xxx    +-----------+        |
          ---    | Completed |<-------+
          ACK    +-----------+
           +------|

    event
   -------
   message

   Figure 8: State transition diagram of client for  INVITE method


   The client MAY close the connection at any time. Closing the
   connection before receiving a final response signals that the client
   wishes to abort the request.




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                 +===========+
                 |  Initial  |<-------------+
                 +===========+              |
                       |                    |
                       |                    |
                       |  INVITE            |
                       |  ------            |
                       |   1xx              |
           +------v    v                    |
        INVITE   +-----------+              |
        ------   | Searching |              |
          1xx    +-----------+              |
           +------| |  |  +---------------->+
                    |  |                    |
          failure   |  |  callee picks up   |
          -------   |  |  ---------------   |
          >= 300    |  |       200          |
                    |  |                    | BYE
           +------v v  v    v-----|         | ---
        INVITE   +-----------+    T3        | 200
        ------   | Answered  |  ------      |
        status   +-----------+  status      |
           +------|    |  | |-----+         |
                       |  +---------------->+
                       |                    |
                       | ACK                |
                       | ---                |
                       | 200                |
                       |                    |
           +------v    v                    |
          ACK    +-----------+              |
          ---    | Connected |              |
          200    +-----------+              |
           +------|       |                 |
                          +-----------------+

    event
   -------
   message

   Figure 9: State transition diagram of server for  INVITE method


   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



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   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).

12 Behavior of SIP Servers

   This section describes behavior of a SIP server in detail. Servers
   can operate in proxy or redirect mode. Proxy servers can "fork"
   connections, i.e., a single incoming request spawns several outgoing
   (client) requests.

   A proxy server always inserts a  Via header field containing its own
   address into those requests that are caused by an incoming request.
   To prevent loops, a server MUST check if its own address is already
   contained in the  Via header of the incoming request.

   We define an "A--B proxy" as a proxy that receives SIP requests over
   transport protocol A and issues requests, acting as a SIP client,
   using transport protocol B. If not stated explicitly, rules apply to
   any combination of transport protocols. For conciseness, we only
   describe behavior with UDP and TCP, but the same rules apply for any
   unreliable datagram or reliable protocol, respectively.

   The detailed connection behavior for UDP and TCP is described in
   Section 11.

12.1 Redirect Server

   A redirect server does not issue any SIP requests of its own. It can
   return a response that refuses or redirects the request. After
   receiving an  INVITE request, a redirect server proceeds through the
   following steps:

        1.   If the  INVITE request cannot be answered immediately
             (e.g., because a location server needs to be contacted), it
             returns one or more provisional responses.

        2.   Once the server has gathered the list of alternative
             locations or has decided to refuse the call, it returns the
             final response. This ends the SIP transaction.

   The redirect server maintains transaction state for the whole SIP
   transaction.

12.2 User Agent Server



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   Servers in user agents behave similarly to redirect servers, except
   that they may also accept a call.

12.3 Proxies Issuing Single Unicast Requests

   Proxies in this category issue at most a single unicast request for
   each incoming SIP request, that is, they do not "fork" requests.
   Servers may choose to always operate in the mode 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 server in the server chain.

   A proxy server SHOULD cache the result of any address translations
   and the response to speed forwarding. After the cache entry has been
   expired, the server cannot tell whether an incoming request is
   actually a retransmission of an older request, where the TCP side has
   terminated.  It will treat it as a new request.

12.4 Proxy Server Issuing Several Requests

   All requests carry the same  Call-ID. For unicast, each of the
   requests has a different (host-dependent)  Request-URI. For
   multicast, a single request is issued, likely with a host-independent
   Request-URI. 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. To avoid response implosion, servers SHOULD NOT
   answer multicast requests with a 404 (Not Found) status code.
   Servers MAY decide not to answer multicast requests if their response
   would be 5xx.

   The server MAY respond to the request immediately with a "100 Trying"
   or "180 Ringing" response; otherwise it MAY wait until either the
   first response to its requests or the UDP retransmission interval.

   The following pseudo-code describes the behavior of a proxy server
   issuing several requests in response to an incoming request. The
   function request(r, a) sends a SIP request r to address a.
   await_response() waits until a response is received and returns the
   response. close(a) closes the TCP connection to client with address
   a. response(s, l, L) sends a response to the client with status s and
   list of locations L, with l entries. 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.,



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   those which are within the same domain as the proxy server. Thus, an
   initial multicast request may trigger additional unicast requests.


     enum {INVITE,         /* request type */
       ACK, OPTIONS, BYE, REGISTER, UNREGISTER} R;
     int N = 0;            /* number of connection attempts */
     address_t address[];  /* list of addresses */
     int done[];           /* address has responded */
     location[];           /* list of locations */
     int heard = 0;        /* number of sites heard from */
     int class;            /* class of status code */
     int best = 1000;      /* best response so far */
     int timeleft = 120;   /* sample timeout value */
     int loc = 0;          /* number of locations */
     struct {              /* response */
       int status;         /* response status */
       char *location;     /* redirect locations */
       address_t a;        /* address of respondent */
     } r;
     int i;

     if (multicast) {
       request(R, address[0]);
     } else {
       N = /* number of addresses to try */
       for (i = 0; i < N; i++) {
         request(R, address[i]);
         done[i] = 0;
       }
     }

     while (timeleft > 0 && (heard < N || multicast)) {
       r = await_response();
       class = r.status / 100;

       if (class >= 2) {
         heard++;
         for (i = 0; i < N; i++) {
           if (address[i] == r.a) {
             done[i] = 1;
             break;
           }
         }
       }

       if (class == 2) {
         best = r.status;



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         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.
          */
         if (recurse) {
           multicast = 0;
           N++;
           request(R, r.location);
         } else if (!ismulticast(r.location)) {
           locations[loc++] = r.location;
           best = r.status;
         }
       }
       else if (class == 4) {
         if (best >= 400) best = r.status;
       }
       else if (class == 5) {
         if (best >= 500) best = r.status;
       }
       else if (class == 6) {
         best = r.status;
         break;
       }
     }
     /* We haven't heard anything useful from anybody. */
     if (best == 1000) {
       best = 404;
     }
     if (best/100 != 3) loc = 0;
     response(best, loc, locations);

     /*
      * Close the other pending transactions by sending BYE.
      */
     for (i = 0; i < N; i++) {
       if (!done[i]) {
         request(BYE, address[i]);
         if (tcp) close(a);
       }
     }






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   After receiving a 2xx or 6xx response, the server SHOULD terminate
   all other pending requests by sending a  BYE request and closing the
   TCP connection, if applicable. (Terminating pending requests is
   advisable as searches consume resources. Also,  INVITE requests may
   "ring" on a number of workstations if the callee is currently logged
   in more than once.)

   [TBD: How do we cancel multicast requests? Force receivers to listen
   for a 200/6xx response and hope that they don't miss one?]

   When operating in this mode, a proxy server MUST ignore any responses
   received for  Call-IDs that it does not have a pending transaction
   for. (If server were to forward responses not belonging to a current
   transaction using the  Via field, the requesting client would get
   confused if it has just issued another request using the same Call-
   ID.)

13 Third-Party Call Initiation

   In some circumstances, third-party call control is required, where
   the calling party suggests to the called party to invite a (small)
   number of other parties. Third-party call control can be used to
   implement the following features:

   Multipoint-control unit (MCU): Some conferences use a multipoint
        control unit to mix, convert and replicate media streams. While
        this solution has scaling problems, it is widely deployed in
        traditional telephony and ISDN conferencing settings, as so-
        called conference bridges. In a MCU-based conference, the
        conference initiator or any authorized member invites a new
        participant and indicate the address of the MCU in the  Also
        header. The invitee then contacts the MCU using the same session
        description and requests to be added to the call, just like a
        normal two-party call.

   Telephony call initiation ("click-to-call"): A SIP  INVITE request
        containing two addresses in the  Also header is sent to a PSTN
        service node that connects these two addresses by a telephone
        call.

   Fully-meshed small conference: For small conferences, such as adding
        a third party to a two-party call, multicast may not always be
        appropriate or available. Instead, when inviting a new
        participant, the caller asks the new member to call the
        remaining members. TBD: Should the call-id be the same or
        different? Need to distinguish between new INVITE for same call
        and adding a party to a call. Include conference identifier?




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   TBD: How about just transferring an SDP description with multiple
   addresses?

   The  Also: header (Section 6.9) is used to indicate a list of parties
   that the callee should invite.

14 ISDN and Intelligent Network Services

   SIP may be used to support a number of ISDN [27] and Intelligent
   Network [28] telephony services, described below. Due to the
   fundamental differences between Internet-based telephony and
   conferencing as compared to public switched telephone network
   (PSTN)-based services, service definitions cannot be precisely the
   same.  Where large differences beyond addressing and location of
   implementation exist, this is indicated below. The term address
   implies any SIP address. (Section 1.4.1).

   Call transfer (TRA) enables a user to transfer an established (i.e.,
        active) call to a third party. SIP signals this via the Location
        header in the  BYE (Section 4.2.4) method.

   Call forwarding (CF) permits the called user to forward particular
        pre-selected calls to another address. Unlike telephony, the
        choice of calls to be forwarded depends on program logic
        contained in any of the SIP servers and can thus be made
        dependent on time-of-day, subject of call, media types, urgency
        or caller identity, rather than being restricted to matching
        list entries. This forwarding service encompasses:

   Call forwarding busy/don't answer (CFB/CFNR, SCF-BY/DA) allows the
        called user to forward particular pre-selected calls if the
        called user is busy or does not answer within a set time.

   Selective call forwarding (SCF) permits the user to have her incoming
        calls addressed to another network destination, no matter what
        the called party status is, if the calling address is included
        in, or excluded from, a screening list. The user's originating
        service is unaffected.

   Completion of calls to busy subscriber (CCBS) allows a calling user
        encountering a busy destination to be informed when the busy
        destination becomes free, without having to make a new call
        attempt. SIP supports services close to CCBS by allowing a
        callee to indicate a more opportune time to call back (Section
        6.25). Also, calling and called user agents can easily record
        the URL of outcoming and incoming calls, so that a user can re-
        try or return calls with a single mouse click.




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   Conferencing (CON) allows the user to communicate simultaneously with
        multiple parties, which may also communicate among themselves.
        SIP can initiate IP multicast conferences with any number of
        participants, conferences where media are mixed by a conference
        bridge (multipoint control unit or MCU) and, for exceptional
        applications with a small number of participants, fully-meshed
        conferences, where each participant sends and receives data to
        all other participants.

   Conference calling add-on allows a user to add and drop participants
        once the conference is active.

   Conference calling meet-me (MMC) allows the user to set up a
        conference or multi-party call, indicating the date, time,
        conference duration, conference media and other parameters. The
        conference session description included in the SIP invitation
        may indicate a time in the future. For multicast conferences,
        participants do not have to connect using SIP at the actual time
        of the conference; instead, they simply subscribe to the
        multicast addresses listed in the announcement. For MCU-based
        conferences, the session description may contain the address of
        the MCU to be called at the time of the conference.

   Destination call routing (DCR) allows customers to specify the
        routing of their incoming calls to destinations according to

        -time of day, day of week, etc.;

        -area of call origination;

        -network address of caller;

        -service attributes;

        -priority (e.g., from input of a PIN or password);

        -charge rates applicable for the destination;

        -proportional routing of traffic.

   In SIP, destination call routing is implemented by proxy and redirect
   servers that implement custom call handling logic, with parameters
   including, but not limited to the set listed above.

   Follow-me diversion (FMD) allows the service subscriber to remotely
        control the redirection (diversion) of calls from his primary
        network address to other locations.




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   In SIP, finding the current network-reachable location of a callee is
   left to the location service and is outside the scope of this
   specification. However, users may use the  REGISTER method (Section
   4.2.5) to appraise their "home" SIP server of their new location.

   Originating call screening (OCS) controlls the ability of a node to
        originate calls. In a fashion similar to closed user groups, a
        firewall would have to be used to restrict the ability to
        initiate SIP invitations outside a designated part of the
        network. In many cases, gateways to the PSTN will require
        appropriate authentication.

   Premium rate (PRM) allows to pay back part of the call cost to the
        called party, considered as an added value provider. See
        discussion on billing services below.

   Split charging (SPL) allows the calling and called party being each
        charged for one part of the call. See discussion on billing
        services below.

   Universal access number (UAN) allows a subscriber with several
        network addresses to be reached with a single, unique address.
        The subscriber may specify which incoming calls are to be routed
        to which address. SIP offers this functionality through proxies
        and redirection.

   Universal personal telecommunications (UPT) is a mobility service
        which enables subscribers to be reached with a unique personal
        telecommunication number (PTN) across multiple networks at any
        network access. The PTN will be translated to an appropriate
        destination address for routing based on the capabilities
        subscribed to by each service subscriber. A person may have
        multiple PTNs, e.g., a business and private PTN. In SIP, the
        host-independent address (Section 1.4.1) of the form user@host
        serves as the PTN, which is translated into one or more host-
        dependent addresses.

   User-defined routing (UDR) allows a subscriber to specify how
        outgoing calls, from the subscriber's location, shall be routed.
        SIP cannot specify routing preferences; this is presumed to be
        handled by a policy-based routing protocol, source routing or
        similar mechanisms.

   Some telephony services can be provided by the end system, without
   involvement by SIP:

   Abbreviated dialing allows users to reach local subscribers without
        specifying the full address (domain or host name). For SIP, the



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        user application completes the address to be a fully qualified
        domain name.

   Call waiting (CW) allows the called party to receive a notification
        that another party is trying to reach her while she is busy
        talking to another calling party.

   For SIP-based telephony, the called party can maintain several call
   presences at the same time, limited by local resources. Thus, it is
   up to the called party to decide whether to accept another call. The
   separation of resource reservation and call control may lead to the
   situation that the called party accepts the incoming call, but that
   the network or system resource allocation fails. This cannot be
   completely prevented, but if the likely resource bottleneck is at the
   local system, the user agent may be able to determine whether there
   are sufficient resources available or roughly track its own resource
   consumption.

   Consultation calling (COC) allows a subscriber to place a call on
        hold, in order to initiate a new call for consultation. In
        systems using SIP, consultation calling can be implemented as
        two separate SIP calls, possibly with the temporary release of
        reserved resources for the call being put on hold.

   Customized ringing (CRG) allows the subscriber to allocate a
        distinctive ringing to a list of calling parties. In a SIP-based
        system, this feature is offered by the user application, based
        on caller identification ( From, Section 6.17) provided by the
        SIP INVITE request (Section 4.2.1).

   Malicious call identification (MCI) allows the service subscriber to
        control the logging (making a record) of calls that received
        that are of a malicious nature. In SIP, by default, all calls
        identify the calling party and the SIP servers that have
        forwarded the call. In addition, calls may be authenticated
        using standard HTTP methods or transport-layer security. A
        callee may decide only to accept calls that are authenticated.

   Multiway calling (MWC) allows the user to establish multiple,
        simultaneous calls with other parties. For a SIP-based end
        system, the considerations for consultation calling apply.

   Terminating call screening (TCS) allows the subscriber to specify
        that incoming calls either be restricted or allowed, according
        to a screening list and/or by time of day or other parameters.

   Billing features such as account card dialing , automatic alternative
   billing , credit card calling (CCC) , reverse charging , freephone



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   (FPH) , premium rate (PRM) and split charging are supported through
   authentication. However, mechanisms for indicating billing
   preferences and capabilities have not yet been specified for SIP.

   Advice of charge allows the user paying for a call to be informed of
   usage-based charging information. Charges incurred by reserving
   resources in the network are probably best indicated by a protocol
   closely affiliated with the reservation protocol. Advice of charge
   when using Internet-to-PSTN gateways through SIP appears feasible,
   but is for further study. Desirable facilities include indication of
   charges at call setup time, during the call and at the end of the
   call

   Closed user groups (CUGs) that restrict members to communicate only
   within the group can be implemented using firewalls and SIP proxies.

   User-to-user signaling is supported within SIP through the addition
   of headers, with predefined header fields such as  Subject or
   Organization.

   Third-party signaling is optionally supported within SIP (Section
   6.9). Third-party signaling can be used to indicate to callees who
   else to invite to a call for MCU and fully-meshed conferences.
   Third-party signaling, combined with appropriate URLs, may be used to
   initiate PSTN phone calls from an Internet host.

15 Security Considerations

15.1 Confidentiality

   Unless SIP transactions are protected by lower-layer security
   mechanisms such as SSL , an attacker may be able to eavesdrop on call
   establishment and invitations and, through the  Subject header field
   or the session description, gain insights into the topic of
   conversation.

15.2 Integrity

   Unless SIP transactions are protected by lower-layer security
   mechanisms such as SSL , an active attacker may be able to modify SIP
   requests.

15.3 Access Control

   SIP requests are not authenticated unless the SIP  Authorization and
   WWW-Authenticate headers are being used. The strengths and weaknesses
   of these authentication mechanisms are the same as for HTTP.




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15.4 Privacy

   User location and SIP-initiated calls may 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.

A Minimal Implementation

A.1 Client

   All clients MUST be able to generate the  INVITE and  ACK requests
   and MUST be able to include the  Call-ID, Content-Length,  Content-
   Type,  From and  To headers. A minimal implementation MUST understand
   SDP [9]. In responses, it must be able to parse the  Call-ID,
   Content-Length,  Content-Type,  Require headers. It must be able to
   recognize the status code classes 1 through 6 and act accordingly.

   The following capability sets build on top of a minimal
   implementation:

   Basic: A basic implementation SHOULD add support for the BYE method
        to allow the interruption of a pending call attempt. It SHOULD
        include a  User-Agent header in its requests and indicate its
        preferred language in the  Accept-Language header.

   Redirection: To support call forwarding, a client needs to be able to
        understand the  Location header, but only the  SIP-URL part, not
        the parameters.

   Negotiation: A client MUST be able to request the  OPTIONS method and
        understand the 380 "Alternative Service" status and the Location
        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 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.




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A.2 Server

   A minimally compliant server implementation MUST understand the
   INVITE,  ACK and  BYE requests. It MUST parse the generate, as
   appropriate, the  Call-ID,  Content-Length, Content-Type,  From,
   PEP,  To and  Via headers.  It must echo the  Sequence header in the
   response. It SHOULD include the  Server header in its responses.

B Summary of Augmented BNF

   In this specification we use the Augmented Backus-Naur Form notation
   described in [21]. For quick reference, the following is a brief
   summary of the main features of this ABNF.

   "abc"
        The case-insensitive string of characters "abc" (or "Abc",
        "aBC", etc.);

   %d32
        The character with ASCII code decimal 32 (space);

   *term
        zero of more instances of  term;

   3*term
        three or more instances of  term;

   2*4term
        two, three or four instances of  term;

   [ term ]
        term is optional;

   term1 term2 term3
        set notation:  term1,  term2 and  term3 must all appear but
        their order is unimportant;

   term1 | term2
        either  term1 or  term2 may appear but not both;

   #term
        a comma separated list of  term;

   2#term
        a comma separated list of  term containing at least 2 items;

   2#4term
        a comma separated list of  term containing 2 to 4 items.



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   Common Tokens

   Certain tokens are used frequently in the BNF this document, and not
   defined elsewhere. Their meaning is well understood but we include it
   here for completeness.



        CR       =    %d13            ;  carriage return character
        LF       =    %d10            ;  line feed character
        CRLF     =    CR LF           ;  typically the end of a line
        SP       =    %d32            ;  space character
        TAB      =    %d09            ;  tab character
        LWS      =    *( SP | TAB)    ;  linear whitespace
        DIGIT    =    "0" .. "9"      ;  a single decimal digit



   Changes in Version -04

   Since version -03, the following changes have been made.

        oThe introduction has been reorganized and large parts
         rewritten.

        oCONNECTED changed to  ACK, as it applies to all responses, not
         just 200.

        oStatus code 181 (Queued) and  Call-Disposition: Queue added.

        oStatus code 481 (Invalid Call-ID) added.

        oStatus code 482 (Loop Detected) added. Via description contains
         motivation.

        oAllow phone numbers in SIP URL for easy connection to Internet
         telephony gateways.

        oAdded  Also header for third-party connectivity.

        oWhen doing parallel searches, pending searches should be
         aborted when one address was successful. The phone call may be
         ringing on a number of workstations where the user is logged in
         and would keep ringing.

        oAdded  duration parameter to  Retry-After to indicate how long
         the callee is likely to be reachable at the address given.




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        oChanged  Sequence to  CSeq for consistency with RTSP.

C Open Issues

   Full meshes: How about just transferring an SDP description with
        multiple addresses?

   H.323: Detailed interaction with H.323 and H.245.

   TRANSACTION: Should we have a transaction id in addition to a call
        ID? Call-IDs are for the end system, but a transaction ID is for
        a single SIP exchange. This is useful for Internet telephony,
        where a single call may trigger several transactions. Also,
        avoids BYE race condition: Proxy doing parallel search cancels
        pending search with BYE after one of the addresses responds with
        200. Through another proxy, this BYE reaches the same end system
        and cancels the successful call.

D Acknowledgments

   We wish to thank the members of the IETF MMUSIC WG for their comments
   and suggestions. Detailed comments were provided by Jonathan
   Rosenberg.  This work is based, inter alia, on [29,30]. Parameters of
   the terminal negotiation mechanism were influenced by Scott Petrack's
   CMA design.

E Authors' Addresses

   Mark Handley
   USC Information Sciences Institute
   c/o MIT Laboratory for Computer Science
   545 Technology Square
   Cambridge, MA 02139
   USA
   electronic mail:  mjh@isi.edu

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   USA
   electronic mail:  schulzrinne@cs.columbia.edu

   Eve Schooler
   Computer Science Department 256-80
   California Institute of Technology
   Pasadena, CA 91125



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   USA
   electronic mail:  schooler@cs.caltech.edu

F Bibliography

   [1] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
   Mar. 1997.  Work in progress.

   [2] M. Handley, "SAP: Session announcement protocol," Internet Draft,
   Internet Engineering Task Force, Nov. 1996.  Work in progress.

   [3] R. Pandya, "Emerging mobile and personal communication systems,"
   IEEE Communications Magazine , vol. 33, pp. 44--52, June 1995.

   [4] P. Lantz, "Usage of H.323 on the Internet," Internet Draft,
   Internet Engineering Task Force, Feb. 1997.  Work in progress.

   [5] M. Handley, J. Crowcroft, C. Bormann, and J. Ott, "The internet
   multimedia conferencing architecture," Internet Draft, Internet
   Engineering Task Force, July 1997.  Work in progress.

   [6] R. Braden, L. Zhang, S. Berson, S. Herzog, and S. Jamin,
   "Resource reservation protocol (RSVP) -- version 1 functional
   specification," Internet Draft, Internet Engineering Task Force, June
   1997.  Work in progress.

   [7] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
   transport protocol for real-time applications," Tech. Rep. RFC 1889,
   Internet Engineering Task Force, Jan. 1996.

   [8] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
   July 1997.  Work in progress.

   [9] M. Handley and V. Jacobson, "SDP: Session description protocol,"
   Internet Draft, Internet Engineering Task Force, Mar. 1997.  Work in
   progress.

   [10] S. Bradner, "Key words for use in RFCs to indicate requirement
   level," Tech. Rep. RFC 2119, Internet Engineering Task Force, Mar.
   1997.

   [11] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-
   Lee, "Hypertext transfer protocol -- HTTP/1.1," Tech. Rep. RFC 2068,
   Internet Engineering Task Force, Jan. 1997.

   [12] C. Partridge, "Mail routing and the domain system," Tech. Rep.



Handley/Schulzrinne/Schooler                                 [Page 77]


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   RFC 974, Internet Engineering Task Force, Jan. 1986.

   [13] A. Gulbrandsen and P. Vixie, "A DNS RR for specifying the
   location of services (DNS SRV)," Tech. Rep. RFC 2052, Internet
   Engineering Task Force, Oct. 1996.

   [14] P. V. Mockapetris, "Domain names - implementation and
   specification," Tech.  Rep. RFC 1035, Internet Engineering Task
   Force, Nov. 1987.

   [15] R. T. Braden, "Requirements for internet hosts - application and
   support," Tech. Rep. RFC 1123, Internet Engineering Task Force, Oct.
   1989.

   [16] D. Zimmerman, "The finger user information protocol," Tech. Rep.
   RFC 1288, Internet Engineering Task Force, Dec. 1991.

   [17] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access
   protocol," Tech. Rep. RFC 1777, Internet Engineering Task Force, Mar.
   1995.

   [18] T. Berners-Lee, "Universal resource identifiers in WWW: a
   unifying syntax for the expression of names and addresses of objects
   on the network as used in the world-wide web," Tech. Rep. RFC 1630,
   Internet Engineering Task Force, June 1994.

   [19] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
   locators (URL): Generic syntax and semantics," Internet Draft,
   Internet Engineering Task Force, May 1997.  Work in progress.

   [20] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
   locators (URL)," Tech. Rep. RFC 1738, Internet Engineering Task
   Force, Dec. 1994.

   [21] D. Crocker, "Augmented BNF for syntax specifications: ABNF,"
   Internet Draft, Internet Engineering Task Force, Oct. 1996.  Work in
   progress.

   [22] J. C. Mogul and S. E. Deering, "Path MTU discovery," Tech. Rep.
   RFC 1191, Internet Engineering Task Force, Nov. 1990.

   [23] W. R. Stevens, TCP/IP illustrated: the protocols , vol. 1.
   Reading, Massachusetts: Addison-Wesley, 1994.

   [24] D. Crocker, "Standard for the format of ARPA internet text
   messages," Tech.  Rep. Also STD0011, RFC 822, Internet Engineering
   Task Force, Aug. 1982.




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   [25] A. Vaha-Sipila, "URLs for telephony," Internet Draft, Internet
   Engineering Task Force, Aug. 1997.  Work in progress.

   [26] L. Masinter, P. Hoffman, and J. Zawinski, "The mailto URL
   scheme," Internet Draft, Internet Engineering Task Force, Oct. 1997.
   Work in progress.

   [27] International Telecommunication Union, "Integrated services
   digital network (ISDN) service capabilities -- definition of
   supplementary services," Recommendation I.250, Telecommunication
   Standardization Sector of ITU, Geneva, Switzerland, 1993.

   [28] International Telecommunication Union, "General recommendations
   on telephone switching and signaling -- intelligent network:
   Introduction to intelligent network capability set 1," Recommendation
   Q.1211, Telecommunication Standardization Sector of ITU, Geneva,
   Switzerland, Mar. 1993.

   [29] E. M. Schooler, "Case study: multimedia conference control in a
   packet-switched teleconferencing system," Journal of Internetworking:
   Research and Experience , vol. 4, pp. 99--120, June 1993.  ISI
   reprint series ISI/RS-93-359.

   [30] H. Schulzrinne, "Personal mobility for multimedia services in
   the Internet," in European Workshop on Interactive Distributed
   Multimedia Systems and Services , (Berlin, Germany), Mar. 1996.


   Full Copyright Statement

   Copyright (c) The Internet Society (1997). 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 implmentation 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.



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   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.




                           Table of Contents



   1          Introduction ........................................    2
   1.1        Overview of SIP Functionality .......................    2
   1.2        Terminology .........................................    3
   1.3        Definitions .........................................    4
   1.4        Summary of SIP Operation ............................    6
   1.4.1      SIP Addressing ......................................    6
   1.4.2      Locating a SIP Server ...............................    7
   1.4.3      SIP Transaction .....................................    9
   1.4.4      SIP Invitation ......................................    9
   1.4.5      Locating a User .....................................   10
   1.4.6      Changing an Existing Session ........................   13
   1.4.7      Registration Services ...............................   13
   1.5        Protocol Properties .................................   13
   1.5.1      Minimal State .......................................   13
   1.5.2      Transport-Protocol Neutral ..........................   14
   1.5.3      Text-Based ..........................................   14
   2          SIP Uniform Resource Locators .......................   14
   3          SIP Message Overview ................................   17
   4          Request .............................................   18
   4.1        Request-Line ........................................   18
   4.2        Methods .............................................   19
   4.2.1       INVITE .............................................   20
   4.2.2       ACK ................................................   20
   4.2.3       OPTIONS ............................................   20
   4.2.4       BYE ................................................   20
   4.2.5       REGISTER ...........................................   21
   4.2.6       UNREGISTER .........................................   21
   4.3        Request-URI .........................................   21
   4.3.1      SIP Version .........................................   22
   4.4        Option Tags .........................................   22
   4.4.1      Registering New Option Tags with IANA ...............   22
   5          Response ............................................   23
   5.1        Status-Line .........................................   23
   5.1.1      Status Codes and Reason Phrases .....................   23



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   6          Header Field Definitions ............................   25
   6.1        General Header Fields ...............................   27
   6.2        Entity Header Fields ................................   27
   6.3        Request Header Fields ...............................   27
   6.4        Response Header Fields ..............................   29
   6.5        Header Field Format .................................   29
   6.6        Accept ..............................................   30
   6.7        Accept-Language .....................................   30
   6.8        Allow ...............................................   30
   6.9        Also ................................................   30
   6.10       Authorization .......................................   31
   6.11       Call-Disposition ....................................   31
   6.12       Call-ID .............................................   32
   6.13       Content-Length ......................................   32
   6.14       Content-Type ........................................   33
   6.15       Date ................................................   33
   6.16       Expires .............................................   33
   6.17       From ................................................   34
   6.18       Location ............................................   35
   6.19       Organization ........................................   37
   6.20       Priority ............................................   37
   6.21       Proxy-Authenticate ..................................   38
   6.22       Proxy-Authorization .................................   38
   6.23       Public ..............................................   38
   6.24       Require .............................................   38
   6.25       Retry-After .........................................   39
   6.26       CSeq ................................................   39
   6.27       Server ..............................................   40
   6.28       Subject .............................................   40
   6.29       Unsupported .........................................   40
   6.30       Timestamp ...........................................   41
   6.31       To ..................................................   41
   6.32       User-Agent ..........................................   41
   6.33       Via .................................................   41
   6.34       Warning .............................................   43
   6.35       WWW-Authenticate ....................................   44
   7          Status Code Definitions .............................   44
   7.1        Informational 1xx ...................................   44
   7.1.1      100 Trying ..........................................   44
   7.1.2      180 Ringing .........................................   44
   7.1.3      181 Queued ..........................................   45
   7.2        Successful 2xx ......................................   45
   7.2.1      200 OK ..............................................   45
   7.3        Redirection 3xx .....................................   45
   7.3.1      300 Multiple Choices ................................   45
   7.3.2      301 Moved Permanently ...............................   46
   7.3.3      302 Moved Temporarily ...............................   46
   7.3.4      380 Alternative Service .............................   46



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   7.4        Request Failure 4xx .................................   46
   7.4.1      400 Bad Request .....................................   46
   7.4.2      401 Unauthorized ....................................   46
   7.4.3      402 Payment Required ................................   46
   7.4.4      403 Forbidden .......................................   46
   7.4.5      404 Not Found .......................................   46
   7.4.6      405 Method Not Allowed ..............................   47
   7.4.7      407 Proxy Authentication Required ...................   47
   7.4.8      408 Request Timeout .................................   47
   7.4.9      420 Bad Extension ...................................   47
   7.4.10     480 Temporarily Unavailable .........................   47
   7.4.11     481 Invalid Call-ID .................................   47
   7.4.12     482 Loop Detected ...................................   48
   7.5        Server Failure 5xx ..................................   48
   7.5.1      500 Server Internal Error ...........................   48
   7.5.2      501 Not implemented .................................   48
   7.5.3      502 Bad Gateway .....................................   48
   7.5.4      503 Service Unavailable .............................   48
   7.5.5      504 Gateway Timeout .................................   48
   7.6        Global Failures 6xx .................................   49
   7.6.1      600 Busy ............................................   49
   7.6.2      603 Decline .........................................   49
   7.6.3      604 Does not exist anywhere .........................   49
   7.6.4      606 Not Acceptable ..................................   49
   8          SIP Message Body ....................................   50
   8.1        Body Inclusion ......................................   50
   8.2        Message Body Length .................................   50
   9          Examples ............................................   51
   9.1        Invitation to Multimedia Conference .................   51
   9.1.1      Request .............................................   51
   9.1.2      Reply ...............................................   52
   9.2        Two-party Call ......................................   53
   9.3        Aborting a Call .....................................   54
   9.3.1      Redirects ...........................................   54
   9.3.2      Alternative Services ................................   55
   9.3.3      Negotiation .........................................   56
   9.4        OPTIONS Request .....................................   57
   10         Compact Form ........................................   57
   11         SIP Transport .......................................   58
   11.1       Achieving Reliability For UDP Transport .............   59
   11.1.1     General Operation ...................................   59
   11.1.2     INVITE ..............................................   59
   11.2       Connection Management for TCP .......................   60
   12         Behavior of SIP Servers .............................   63
   12.1       Redirect Server .....................................   63
   12.2       User Agent Server ...................................   63
   12.3       Proxies Issuing Single Unicast Requests .............   64
   12.4       Proxy Server Issuing Several Requests ...............   64



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   13         Third-Party Call Initiation .........................   67
   14         ISDN and Intelligent Network Services ...............   68
   15         Security Considerations .............................   72
   15.1       Confidentiality .....................................   72
   15.2       Integrity ...........................................   72
   15.3       Access Control ......................................   72
   15.4       Privacy .............................................   73
   A          Minimal Implementation ..............................   73
   A.1        Client ..............................................   73
   A.2        Server ..............................................   74
   B          Summary of Augmented BNF ............................   74
   C          Open Issues .........................................   76
   D          Acknowledgments .....................................   76
   E          Authors' Addresses ..................................   76
   F          Bibliography ........................................   77




































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