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Versions: 01 02 03 04 05 06 07 08 09 10 11 12 RFC 2543

Internet Engineering Task Force                                MMUSIC WG
Internet Draft                    Handley/Schulzrinne/Schooler/Rosenberg
ietf-mmusic-sip-08.txt                ISI/Columbia U./Caltech/Bell Labs.
August 7, 1998
Expires: February 1999

                    SIP: Session Initiation Protocol


   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
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   Distribution of this document is unlimited.


         The Session Initiation Protocol (SIP) is an application-
         layer control (signaling) protocol for creating,
         modifying and terminating sessions with one or more
         participants. These sessions include Internet multimedia
         conferences, Internet telephone calls and multimedia
         distribution. Members in a session can communicate via
         multicast or via a mesh of unicast relations, or a
         combination of these.

         SIP invitations used to create sessions carry session
         descriptions which allow participants to agree on a set
         of compatible media types. It supports user mobility by
         proxying and redirecting requests to the user's current
         location. Users can register their current location.  SIP
         is not tied to any particular conference control
         protocol. SIP is designed to be independent of the

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         lower-layer transport protocol and can be extended with
         additional capabilities.

         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.

1 Introduction

1.1 Overview of SIP Functionality

   The Session Initiation Protocol (SIP) is an application-layer control
   protocol that can establish, modify and terminate multimedia sessions
   or calls. These multimedia sessions include multimedia conferences,
   distance learning, Internet telephony and similar applications. SIP
   can invite both persons and "robots", such as a media storage
   service.  SIP can invite parties to both unicast and multicast
   sessions; the initiator does not necessarily have to be a member of
   the session to which it is inviting. Media and participants can be
   added to an existing session.

   SIP can be used to initiate sessions as well as invite members to
   sessions that have been advertised and established by other means.
   Sessions may be advertised using multicast protocols such as SAP,
   electronic mail, news groups, web pages or directories (LDAP), among

   SIP transparently supports name mapping and redirection services,
   allowing the implementation of ISDN and Intelligent Network telephony
   subscriber services. These facilities also enable personal mobility
   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.,
   mobility complements terminal mobility, i.e., the ability to maintain
   communications when moving a single end system from one subnet to

   SIP supports five facets of establishing and terminating multimedia

   User location: determination of the end system to be used for

   User capabilities: determination of the media and media parameters to

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        be used;

   User availability: determination of the willingness of the called
        party to engage in communications;

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

   Call handling: including transfer and termination of calls.

   SIP can also initiate multi-party calls using a multipoint control
   unit (MCU) or fully-meshed interconnection instead of multicast.
   Internet telephony gateways that connect PSTN parties may also use
   SIP to set up calls between them.

   SIP is designed as part of the overall IETF multimedia data and
   control architecture currently incorporating protocols such as RSVP
   (RFC 2205 [2]) for reserving network resources, the real-time
   transport protocol (RTP) (RFC 1889 [3]) for transporting real-time
   data and providing QOS feedback, the real-time streaming protocol
   (RTSP) (RFC 2326 [4]) for controlling delivery of streaming media,
   the session announcement protocol (SAP) for advertising multimedia
   sessions via multicast and the session description protocol (SDP)
   (RFC 2327 [5]) for describing multimedia sessions. However, the
   functionality and operation of SIP does not depend on any of these

   SIP may also be used in conjunction with other call setup and
   signaling protocols. In that mode, an end system uses SIP exchanges
   to determine the appropriate end system address and protocol from a
   given address that is protocol-independent. For example, SIP could be
   used to determine that the party may be reached via H.323, obtain the
   H.245 gateway and user address and then use H.225.0 to establish the

   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 does not offer conference control services such as floor control
   or voting and does not prescribe how a conference is to be managed,
   but SIP can be used to introduce conference control protocols. SIP
   does not allocate multicast addresses.

   SIP can invite users to sessions with and without resource
   reservation.  SIP does not reserve resources, but may convey to the
   invited system the information necessary to do this. Quality-of-

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   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.

1.2 Terminology

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   and "OPTIONAL" are to be interpreted as described in RFC 2119 [6] and
   indicate requirement levels for compliant SIP implementations.

1.3 Definitions

   This specification uses a number of terms to refer to the roles
   played by participants in SIP communications. The definitions of
   client, server and proxy are similar to those used by the Hypertext
   Transport Protocol (HTTP) (RFC 2068 [7]). The following terms have
   special significance for SIP.

   Call: A call consists of all participants in a conference invited by
        a common source. A SIP call is identified by a globally unique
        call-id (Section 6.12). Thus, if a user is, for example, invited
        to the same multicast session by several people, each of these
        invitations will be a unique call. A point-to-point Internet
        telephony conversation maps into a single SIP call. In a MCU-
        based call-in conference, each participant uses a separate call
        to invite himself to the MCU.

   Call leg: A call leg is identified by the combination of Call-ID, To
        and From.

   Client: An application program that establishes connections for the
        purpose of sending requests. Clients may or may not interact
        directly with a human user. User agents and proxies contain
        clients (and servers).

   Conference: A multimedia session (see below), identified by a common
        session description. A conference may have zero or more members
        and includes the cases of a multicast conference, a full-mesh
        conference and a two-party "telephone call", as well as
        combinations of these.  Any number of calls may be used to
        create a conference.

   Downstream: Requests sent in the direction from the caller to the

   Final response: A response that terminates a SIP transaction, as
        opposed to a provisional response that does not. All 2xx, 3xx,

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        4xx, 5xx and 6xx responses are final.

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

   Invitation: A request sent to a user (or service) requesting
        participation in a session. A successful SIP invitation consists
        of two transactions: an INVITE request followed by an ACK

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

   Isomorphic request or response: Two requests or responses are defined
        to be isomorphic for the purposes of this document if they have
        the same values for the Call-ID, To, From and CSeq header
        fields. In addition, requests have to have the same Request-URI.

   Location server: See location service

   Location service: A location service is used by a SIP redirect or
        proxy server to obtain information about a callee's possible
        location(s). Location services are offered by location servers.
        Location servers may be co-located with a SIP server, but the
        manner in which a SIP server requests location services is
        beyond the scope of this document.

   Parallel search: In a parallel search, a proxy issues several
        requests to possible user locations upon receiving an incoming
        request.  Rather than issuing one request and then waiting for
        the final response before issuing the next request as in a
        sequential search , a parallel search issues requests without
        waiting for the result of previous requests.

   Provisional response: A response used by the server to indicate
        progress, but that does not terminate a SIP transaction. 1xx
        responses are provisional, other responses are considered final

   Proxy, proxy server: An intermediary program that acts as both a
        server and a client for the purpose of making requests on behalf
        of other clients. Requests are serviced internally or by passing
        them on, possibly after translation, to other servers. A proxy
        must interpret, and, if necessary, rewrite a request message
        before forwarding it.

   Redirect server: A redirect server is a server that accepts a SIP
        request, maps the address into zero or more new addresses and

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        returns these addresses to the client. Unlike a proxy server ,
        it does not initiate its own SIP request. Unlike a user agent
        server , it does not accept calls.

   Registrar: A registrar is server that accepts REGISTER requests. A
        registrar is typically co-located with a proxy or redirect
        server and may offer location services.

   Ringback: Ringback is the signaling tone produced by the calling
        client's application indicating that a called party is being
        alerted (ringing).

   Server: A server is an application program that accepts requests in
        order to service requests and sends back responses to those
        requests.  Servers are either proxy, redirect or user agent
        servers or registrars.

   Session: "A multimedia session is a set of multimedia senders and
        receivers and the data streams flowing from senders to
        receivers. A multimedia conference is an example of a multimedia
        session." (RFC 2327 [5]) (A session as defined for SDP may
        comprise one or more RTP sessions.) As defined, a callee may be
        invited several times, by different calls, to the same session.
        If SDP is used, a session is defined by the concatenation of the
        user name , session id , network type , address type and address
        elements in the origin field.

   (SIP) transaction: A SIP transaction occurs between a client and a
        server and comprises all messages from the first request sent
        from the client to the server up to a final (non-1xx) response
        sent from the server to the client. A transaction is identified
        by the CSeq sequence number (Section 6.16) within a single call
        leg The ACK request has the same CSeq number as the
        corresponding INVITE request, but comprises a transaction of its

   Upstream: Responses sent in the direction from the called client to
        the caller.

   URL-encoded: A character string encoded according to RFC 1738,
        Section 2.2 [8].

   User agent client (UAC), calling user agent: A user agent client is a
        client application that initiates the SIP request.

   User agent server (UAS), called user agent: A user agent server is a
        server application that contacts the user when a SIP request is
        received and that returns a response on behalf of the user. The

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        response may accept, reject or redirect the request.

   An application program may be capable of acting both as a client and
   a server. For example, a typical multimedia conference control
   application would act as a user agent client to initiate calls or to
   invite others to conferences and as a user agent server to accept
   invitations. The properties of the different SIP server types are
   summarized in Table 1.

   property                     redirect    proxy     user agent    registrar
                                 server     server      server
   also acts as a SIP client       no        yes          no           no
   returns 1xx status             yes        yes         yes           yes
   returns 2xx status              no        yes         yes           yes
   returns 3xx status             yes        yes         yes           yes
   returns 4xx status             yes        yes         yes           yes
   returns 5xx status             yes        yes         yes           yes
   returns 6xx status              no        yes         yes           no
   inserts Via header              no        yes          no           no
   accepts ACK                    yes        yes         yes           no

   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 may trigger a chain of new SIP
   requests by proxies (Section 1.4.5). Users can register their
   location(s) with SIP servers (Section 4.2.6).

1.4.1 SIP Addressing

   The "objects" addressed by SIP are users at hosts, identified by a
   SIP URL. The SIP URL takes the form similar to a mailto or telnet
   URL, i.e., user@host user part is a user name, a civil name or a
   telephone number. The host part is either a domain name having a DNS
   SRV (RFC 2052 [9]), MX (RFC 974 [10], CNAME or A record (RFC 1035
   [11]), or a numeric network address.

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   A user's SIP address can be obtained out-of-band, can be learned via
   existing media agents, can be included in some mailers' message
   headers, or can be recorded during previous invitation interactions.
   In many cases, a user's SIP URL can be guessed from his email

   Examples of SIP URLs include:


   A SIP URL address can designate an individual (possibly located at
   one of several end systems), the first available person from a group
   of individuals or a whole group. The form of the address, e.g.,
   sip:sales@example.com , is not sufficient, in general, to determine
   the intent of the caller.

   If a user or service chooses to be reachable at an address that is
   guessable from the person's name and organizational affiliation, the
   traditional method of ensuring privacy by having an unlisted "phone"
   number is compromised. However, unlike traditional telephony, SIP
   offers authentication and access control mechanisms and can avail
   itself of lower-layer security mechanisms, so that client software
   can reject unauthorized or undesired call attempts.

1.4.2 Locating a SIP Server

   When a client wishes to send a request, the client either sends it to
   locally configured SIP proxy server (as in HTTP), independent of the
   Request-URI, or sends it to the IP address and port corresponding to
   the Request-URI. For the latter case, the client performs the
   following steps to obtain the server's IP address.

   A SIP client MUST follow the following steps to resolve the host part
   of the Request-URI. If a client supports only TCP or UDP, but not
   both, the client omits the respective address type. If the SIP
   address contains a port number, that number is to be used, otherwise,
   the default port number 5060 is to be used. The default port number
   is the same for UDP and TCP. In all cases, the client first attempts
   to contact the server using UDP, then TCP.

   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. (For socket-based programs: For TCP, connect() returns

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   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 so that a second write()
   fails with ECONNREFUSED.  )

   If the SIP address contains a numeric IP address, the client contacts
   the SIP server at that address. Otherwise, the client follows the
   steps below.

        1.   If there is a SRV DNS resource record (RFC 2052 [9]) 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 given in the URL or, if none provided, the one
             listed in the DNS resource record.

        2.   If there is a SRV DNS resource record (RFC 2052 [9]) 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 given in the URL or, if none provided, the one
             listed in the DNS resource record.

        3.   If there is a DNS MX record (RFC 974 [10]), contact the
             hosts listed in their order of preference at the port
             number listed in the URL or the default SIP port number if
             none. For each host listed, first try to contact the SIP
             server using UDP, then TCP.

        4.   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.

   If all of the above methods fail to locate a server, 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.

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

        This sequence is modeled after that described for SMTP,
        where MX records are to be checked before A records (RFC
        1123 [12]).

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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.  The ACK request following an INVITE is not part of the
   transaction since it may traverse a different set of hosts.

   If TCP is used, request and responses within a single SIP transaction
   are carried over the same TCP connection (see Section 10). Several
   SIP requests from the same client to the same server may use the same
   TCP connection or may open a new connection for each request.

   If the client sent the request via unicast UDP, the response is sent
   to the address contained in the next Via header field (Section 6.40)
   of the response. If the request is sent via multicast UDP, the
   response is directed to the same multicast address and destination
   port. For UDP, reliability is achieved using retransmission (Section

   The SIP message format and operation is independent of the transport

1.4.4 SIP Invitation

   A successful SIP invitation consists of two requests, INVITE followed
   by ACK. The INVITE (Section 4.2.1) request asks the callee to join a
   particular conference or establish a two-party conversation. After
   the callee has agreed to participate in the call, the caller confirms
   that it has received that response by sending an ACK (Section 4.2.2)
   request. If the caller no longer wants to participate in the call, it
   sends a BYE request instead of an ACK.

   The INVITE request typically contains a session description, for
   example written in SDP (RFC 2327 [5]) 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 a
   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 or
   wants to receive data via unicast.

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   The protocol exchanges for the INVITE method are shown in Fig. 1 for
   a proxy server and in Fig. 2 for a redirect server. (Note that the
   messages shown in the figures have been abbreviated slightly.) In
   Fig. 1, the proxy server accepts the INVITE request (step 1),
   contacts the location service with all or parts of the address (step
   2) and obtains a more precise location (step 3). The proxy server
   then issues a SIP INVITE request to the address(es) returned by the
   location service (step 4). The user agent server alerts the user
   (step 5) and returns a success indication to the proxy server (step
   6). The proxy server then returns the success result to the original
   caller (step 7). The receipt of this message is confirmed by the
   caller using an ACK request, which is forwarded to the callee (steps
   8 and 9). Note that an ACK can also be sent directly to the callee,
   bypassing the proxy. All requests and responses have the same Call-

                                            +....... 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 ========================>(~~~~~~)=========>(~~~~~~) :
   :                    <........................(      )<.........(      ) :
   :                     : 7: 200 OK        :    (      )6: 200 OK (      ) :
   :                     :                  :    ( tune )          ( play ) :
   :                     : 8: ACK           :    (      )9: ACK    (      ) :
   :                    ========================>(~~~~~~)=========>(~~~~~~) :
   +.....................+                  +...............................+

     ====> SIP request
     ....> SIP response
     ----> non-SIP protocols

   Figure 1: Example of SIP proxy server

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   The redirect server shown in Fig. 2 accepts the INVITE request (step
   1), contacts the location service as before (steps 2 and 3) and,
   instead of contacting the newly found address itself, returns the
   address to the caller (step 4), which is then acknowledged via an ACK
   request (step 5). The caller issues a new request, with the same
   call-ID but a higher CSeq, to the address returned by the first
   server (step 6). In the example, the call succeeds (step 7). The
   caller and callee complete the handshake with an ACK (step 8).

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

1.4.5 Locating a User

   A callee may move between a number of different end systems over
   time.  These locations can be dynamically registered with the SIP
   server (Sections 1.4.7, 4.2.6). A location server may also use one or
   more other protocols, such as finger (RFC 1288 [13]), rwhois (RFC
   2167 [14]), LDAP (RFC 1777 [15]), multicast-based protocols [16] or
   operating-system dependent mechanisms to actively determine the end
   system where a user might be reachable. A location server may return
   several locations because the user is logged in at several hosts
   simultaneously or because the location server has (temporarily)
   inaccurate information. The SIP server combines the results to yield
   a list of a zero or more locations. It is recommended that each
   location server sorts results according to the likelihood of success.

   The action taken on receiving a list of locations varies with the
   type of SIP server. A SIP redirect server returns the list to the
   client as Location headers (Section 6.22). A SIP proxy server can
   sequentially or in parallel try the addresses until the call is
   successful (2xx response) or the callee has declined the call (6xx
   response). With sequential attempts, a proxy server can implement an
   "anycast" service.

   If a proxy server forwards a SIP request, it MUST add itself to the
   end of the list of forwarders noted in the Via (Section 6.40)
   headers. The Via trace ensures that replies can take the same path
   back, ensuring correct operation through compliant firewalls and
   avoiding request loops. On the response path, each host MUST remove
   its Via, so that routing internal information is hidden from the
   callee and outside networks. A proxy server MUST check that it does
   not generate a request to a host listed in the Via sent-by, via-
   received or via-maddr parameters (Section 6.40).  (Note: If a host
   has several names or network addresses, this may not always work.
   Thus, each host also checks if it is part of the Via list.)

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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 =======================>(~~~~~~)                    :
   :       | ^ |        <.......................(      )                    :
   :       | . |         : 4: 302 Moved     :   (      )                    :
   :       | . |         :    hgs@play      :   ( tune )                    :
   :       | . |         :                  :   (      )                    :
   :       | . |         : 5: ACK           :   (      )                    :
   :       | . |        =======================>(~~~~~~)                    :
   :       | . |         :                  :                               :
   +.......|...|.........+                  :                               :
           | . |                            :                               :
           | . |                            :                               :
           | . |                            :                               :
           | . |                            :                               :
           | . | 6: INVITE hgs@play.cs.columbia.edu                (~~~~~~) :
           | . ==================================================> (      ) :
           | ..................................................... (      ) :
           |     7: 200 OK                  :                      ( play ) :
           |                                :                      (      ) :
           |     8: ACK                     :                      (      ) :
           ======================================================> (~~~~~~) :

     ====> SIP request
     ....> SIP response
     ----> non-SIP protocols

   Figure 2: Example of SIP redirect server

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   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. Duplicate
   requests are not an error.

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 with
   the old call identifier.

1.4.7 Registration Services

   The REGISTER request allows a client to let a proxy or redirect
   server know at which address(es) it may be reached. A client may also
   use it to install call handling features at the server.

1.5 Protocol Properties

1.5.1 Minimal State

   A single conference session or call may involve one or more SIP
   request-response transactions. Proxy servers do not have to keep
   state for a particular call, however, they MAY maintain state for a
   single SIP transaction, as discussed in Section 12.

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

1.5.2 Lower-Layer-Protocol Neutral

   SIP makes minimal assumptions about the underlying transport and
   network-layer protocols. The lower-layer may provide either a packet
   or a byte stream service, with reliable or unreliable service.

   In an Internet context, SIP is able to utilize both UDP and TCP as
   transport protocols, among others. UDP allows the application to more
   carefully control the timing of messages and their retransmission, to
   perform parallel searches without requiring TCP connection state for
   each outstanding request, and to use multicast. Routers can more
   readily snoop SIP UDP packets. TCP allows easier passage through

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   existing firewalls, and given the similar protocol design, allows
   common servers for SIP, HTTP and the Real Time Streaming Protocol
   (RTSP) (RFC 2326 [4]).

   When TCP is used, SIP can use one or more connections to attempt to
   contact a user or to modify parameters of an existing conference.
   Different SIP requests for the same SIP call may use different TCP
   connections or a single persistent connection, as appropriate.

   For concreteness, this document will only refer to Internet
   protocols.  However, SIP may also be used directly with protocols
   such as ATM AAL5, IPX, frame relay or X.25. The necessary naming
   conventions are beyond the scope of this document. User agents SHOULD
   implement both UDP and TCP transport, proxy and redirect servers

1.5.3 Text-Based

   SIP is text-based, using ISO 10646 in UTF-8 encoding throughout. This
   allows easy implementation in languages such as Java, Tcl and Perl,
   allows easy debugging, and most importantly, makes SIP flexible and
   extensible. As SIP is used for initiating multimedia conferences
   rather than delivering media data, it is believed that the additional
   overhead of using a text-based protocol is not significant.

2 SIP Uniform Resource Locators

   SIP URLs are used within SIP messages to indicate the originator
   (From), current destination (Request-URI) and final recipient (To) of
   a SIP request, and to specify redirection addresses (Location). A SIP
   URL can 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.

   A SIP URL follows the guidelines of RFC 1630 [17], as revised [18],
   and has the syntax shown in Fig. 3. Note that reserved characters
   have to be escaped.

   The URI character classes referenced above are described in Section
   C. The URI specification is currently being revised. It is
   anticipated that future versions of this specification will reference
   the revised edition. Note that all URL reserved characters MUST be

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     SIP-URL         = "sip:" [ userinfo "@" ] hostport
                       url-parameters [ headers ]
     userinfo        = user [ ":" password ]
     user            = *( unreserved | escaped
                     | "&" | "=" | "+" | "$" | "," )
     password        = *( unreserved | escaped
                     | "&" | "=" | "+" | "$" | "," )
     hostport        = host [ ":" port ]
     host            = hostname | IPv4address
     hostname        = *( domainlabel "." ) toplabel [ "." ]
     domainlabel     = alphanum | alphanum *( alphanum | "-" ) alphanum
     toplabel        = alpha | alpha *( alphanum | "-" ) alphanum
     IPv4address     = 1*digit "." 1*digit "." 1*digit "." 1*digit
     port            = *digit
     url-parameters  = *( ";" url-parameter )
     url-parameter   = transport-param | user-param
                     | ttl-param | maddr-param | tag-param | other-param
     transport-param = "transport=" ( "udp" | "tcp" )
     ttl-param       = "ttl=" ttl
     ttl             = 1*3DIGIT       ; 0 to 255
     maddr-param     = "maddr=" host
     user-param      = "user=" ( "phone" $|$ "ip" )
     tag-param       = "tag=" UUID
     UUID            = 1*( hex | "-" )
     other-param     = *uric
     headers         = "?" header *( "&" header )
     header          = hname "=" hvalue
     hname           = *uric
     hvalue          = *uric
     uric            = reserved | unreserved | escaped
     reserved        = ";" | "/" | "?" | ":" | "@" | "&" | "=" | "+" |
                       "$" | ","
     digits          = 1*DIGIT

   Figure 3: SIP URL syntax

   Figure 4: SIP URL syntax; telephone subscriber


   host: The mailto: URL and RFC 822 email addresses require that
        numeric host addresses ("host numbers") are enclosed in square
        brackets (presumably, since host names might be numeric), while

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        host numbers without brackets are used for all other URLs. The
        SIP URL requires the latter form, without brackets.

   userinfo: The SIP scheme MAY use the format "user:password" in the
        userinfo field. The use of passwords in the userinfo is NOT
        RECOMMENDED, because the passing of authentication information
        in clear text (such as URIs) has proven to be a security risk in
        almost every case where it has been used.

   If the host is an Internet telephony gateway, the userinfo field can
   also encode a telephone number using the notation of telephone-
   subscriber (Fig. 4).  The telephone number is a special case of a
   user name and cannot be distinguished by a BNF. Thus, a URL
   parameter, user, is added to distinguish telephone numbers from user
   names. The phone identifier is to be used when connecting to a
   telephony gateway. Even without this parameter, recipients of SIP
   URLs MAY interpret the pre-@ part as a phone number if local
   restrictions on the name space for user name allow it.

   If a server handles SIP addresses for another domain, it MUST URL-
   encode the "@" character (%40). The ";" character MUST be URL-
   encoded, as otherwise it is not possible to distinguish, in one
   parsing pass, the case host;parameter and user;moreuser@host

   URL parameters: SIP URLs can define specific parameters of the
        request: The transport parameter determines the the transport
        mechanism (UDP or TCP). UDP is to be assumed when no explicit
        transport parameter is included. The maddr parameter provides
        the server address to be contacted for this user, overriding the
        address supplied in the host field. This address is typically a
        multicast address, but could also be the address of a backup
        server. The ttl parameter determines the time-to-live value of
        the UDP multicast packet and MUST only be used if maddr is a
        multicast address and the transport protocol is UDP. The user
        parameter was described above, the tag parameter is described in
        its own section below.  URL parameters are added after the host
        component and are separated by semi-colons. For example, to
        specify to call j.doe@big.com using multicast to
        with a ttl of 15, the following URL would be used:


   The transport, maddr, and ttl parameters MUST NOT be used in the From
   and To header fields and the Request-URI; they are ignored if

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   Headers: Headers of the SIP request can be defined with the "?"
        mechanism within a SIP URL. The special hname "body" indicates
        that the associated hvalue is the message-body of the SIP INVITE
        request. Headers MUST NOT be used in the From and To header
        fields and the Request-URI; they are ignored if present.

   Tag: The tag parameter serves as a general mechanism to distinguish
        multiple instances of a user identified by a single SIP URL.
        Such distinction is needed in two cases. First, as proxies can
        fork requests, the same request can reach multiple instances of
        a user (mobile and home phones, for example). As each can
        respond, there needs to be a means to distinguish the responses
        from each at the caller. The situation also arises with
        multicast requests. The tag serves to distinguish responses at
        the UAC. It MUST be placed in the To field of the response by
        each instance when there is a possibility that the request was
        forked at an intermediate proxy. This, in general, means that
        the Tag MUST be inserted when the URL in the To does not refer
        to a fully qualified hostname. The tag MUST be added by UAS,
        registrars and redirect servers, but MUST NOT be inserted into
        responses forwarded upstream by proxies. The Tag is added for
        all responses for all methods. All subsequent transactions
        between two entities MUST include the Tag parameter, as
        described in Section 11.

   Secondly, the tag MAY appear in the From field of a call invitation.
   This is needed when it is anticipated that two instances of a user
   sharing a SIP address may make call invitations with the same Call-
   ID. In this case, it MUST be present.

   The use of version-1 (time based) or version-4 (random) UUID [19] is
   OPTIONAL. The tag value is designed to be globally unique and
   cryptographically random with at least 32 bits of randomness. It
   SHOULD NOT be included in long-lived SIP URLs, e.g., those found on
   web pages or user databases.  A single user maintains the same tag
   throughout the call identified by the Call-ID. The tag parameter in
   To headers is ignored when matching responses to requests that did
   not contain a tag in their To header. (See Section 6.37.)

   Table 2 summarizes where the components of the SIP URL can be used.

   Examples of SIP URLs are:


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                      Request-URI    To    From    Location    external
   user               x              x     x       x           x
   password                          x             x           x
   host               x              x     x       x           x
   user-param         x              x     x       x           x
   tag-param          x              x     x       x
   maddr-param                                     x           x
   ttl-param                                       x           x
   transport-param                                 x           x
   headers                                         x           x

   Table 2: Use of URL  components  for  SIP  headers,  Request-URI  and


   Within a SIP message, URLs are used to indicate the source and
   intended destination of a request, redirection addresses and the
   current destination of a request. Normally all these fields will
   contain SIP URLs.

   SIP URLs are case-insensitive, so that for example the two URLs
   sip:j.doe@example.com and SIP:J.Doe@Example.com are equivalent.  All
   URL parameters are included when comparing SIP URLs for equality.

   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.
   To allow for this flexibility, SIP headers that specify user
   addresses allow these addresses to be SIP and non-SIP 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.

3 SIP Message Overview

   SIP is a text-based protocol and uses the ISO 10646 character set in

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   UTF-8 encoding (RFC 2279 [20]). Lines are terminated by CRLF, but
   receivers should be prepared to also interpret CR and LF by
   themselves as line terminators.

   Except for the above difference in character sets, 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 (RFC 2068 [7]). In addition, we describe SIP in both
   prose and an augmented Backus-Naur form (BNF) [H2.1] described in
   detail in RFC 2234 [21].

   Unlike HTTP, SIP MAY use UDP. When sent over TCP or UDP, multiple SIP
   transactions can be carried in a single TCP connection or UDP
   datagram. UDP datagrams, including all headers, should not normally
   be larger than the path maximum transmission unit (MTU) if the MTU is
   known, or 1400 bytes if the MTU is unknown.

        The 1400 bytes accommodates lower-layer packet headers
        within the "typical" MTU of around 1500 bytes. Recent
        studies [22] indicate that an MTU of 1500 bytes is a
        reasonable assumption. The next lower common MTU values are
        1006 bytes for SLIP and 296 for low-delay PPP (RFC 1191
        [23]). Thus, another reasonable value would be a message
        size of 950 bytes, to accommodate packet headers within the
        SLIP MTU without fragmentation.

   A SIP message is either a request from a client to a server, or a
   response from a server to a client.

        SIP-message  ___   Request | Response

   Both Request (section 4) and Response (section 5) messages use the
   generic-message format of RFC 822 [24] for transferring entities (the
   body of the message). Both types of messages consist of a start-line,
   one or more header fields (also known as "headers"), an empty line
   (i.e., a line with nothing preceding the carriage-return line-feed
   (CRLF)) indicating the end of the header fields, and an optional
   message-body. To avoid confusion with similar-named headers in HTTP,
   we refer to the header describing the message body as entity headers.
   These components are described in detail in the upcoming sections.

        generic-message    =    start-line

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                                [ message-body ]

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

        message-header    =    ( general-header
                               | request-header
                               | response-header
                               | entity-header )

   In the interest of robustness, any leading empty line(s) MUST be
   ignored. In other words, if the Request or Response message begins
   with a CRLF, CR, or LF, these characters should be ignored.

4 Request

   The Request message format is shown below:

        Request    =    Request-Line         ;  Section 4.1
                        *( general-header
                        | request-header
                        | entity-header )
                        [ message-body ]     ;  Section 8

4.1 Request-Line

   The Request-Line begins with a method token, followed by the
   Request-URI and the protocol version, and ending with CRLF. The
   elements are separated by SP characters.  No CR or LF are allowed
   except in the final CRLF sequence.

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

4.2 Methods

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        general-header     =    Call-ID                ; Section 6.12
                           |    CSeq                   ; Section 6.16
                           |    Date                   ; Section 6.17
                           |    Encryption             ; Section 6.18
                           |    Expires                ; Section 6.19
                           |    From                   ; Section 6.20
                           |    Record-Route           ; Section 6.29
                           |    Timestamp              ; Section 6.36
                           |    To                     ; Section 6.37
                           |    Via                    ; Section 6.40
        entity-header      =    Content-Encoding       ; Section 6.13
                           |    Content-Length         ; Section 6.14
                           |    Content-Type           ; Section 6.15
        request-header     =    Accept                 ; Section 6.7
                           |    Accept-Encoding        ; Section 6.8
                           |    Accept-Language        ; Section 6.9
                           |    Authorization          ; Section 6.11
                           |    Hide                   ; Section 6.21
                           |    Location               ; Section 6.22
                           |    Max-Forwards           ; Section 6.23
                           |    Organization           ; Section 6.24
                           |    Priority               ; Section 6.25
                           |    Proxy-Authorization    ; Section 6.27
                           |    Proxy-Require          ; Section 6.28
                           |    Route                  ; Section 6.33
                           |    Require                ; Section 6.30
                           |    Response-Key           ; Section 6.31
                           |    Subject                ; Section 6.35
                           |    User-Agent             ; Section 6.39
        response-header    =    Allow                  ; Section 6.10
                           |    Location               ; Section 6.22
                           |    Proxy-Authenticate     ; Section 6.26
                           |    Retry-After            ; Section 6.32
                           |    Server                 ; Section 6.34
                           |    Unsupported            ; Section 6.38
                           |    Warning                ; Section 6.41
                           |    WWW-Authenticate       ; Section 6.42

   Table 3: SIP headers

   The methods are defined below. Methods that are not supported by a
   proxy or redirect server are treated by that server as if they were
   an OPTIONS method and forwarded accordingly.  Methods that are not
   supported by a user agent server or registrar cause a 501 (Not
   Implemented) response to be returned (Section 7).

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        Method    =    "ACK" | "BYE" | "CANCEL" | "INVITE"
                 |     "OPTIONS" | "REGISTER"

4.2.1 INVITE

   The INVITE method indicates that the user or service is being invited
   to participate in a session. The message body contains a description
   of the session to which the callee is being invited. For two-party
   calls, the caller indicates the type of media it is able to receive
   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.

   If a user agent receives an INVITE request for an existing Call-ID
   with a higher CSeq sequence number than any previous INVITE for the
   same Call-ID, it MUST check any version identifiers in the session
   description or, if there are no version identifiers, the content of
   the session description to see if it has changed. It MUST also
   inspect any other header fields for changes and act accordingly. If
   the session description has changed, the user agent server MUST
   adjust the session parameters accordingly, possibly after asking the
   user for confirmation. (Versioning of the session description may be
   used to accommodate the capabilities of new arrivals to a conference,
   add or delete media or change from a unicast to a multicast

   This method MUST be supported by SIP proxy, redirect and user agent
   servers as well as clients.

4.2.2 ACK

   The ACK request confirms that the client has received a final
   response to an INVITE request. (ACK is used only with INVITE
   requests.) 2xx responses are acknowledged by client user agents, all
   other final responses by the first proxy or client user agent to
   receive the response. The Via is always initialized to the host that
   originates the ACK request, i.e., the client user agent after a 2xx
   response or the first proxy to receive a non-2xx final response. The
   ACK request is forwarded as the corresponding INVITE request, based
   on its Request-URI. See Section 10 for details.

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   The ACK request MAY contain a message body with the final session
   description to be used by the callee. If the ACK message body is
   empty, the callee uses the session description in the INVITE request.

   This method MUST be supported by SIP proxy, redirect and user agent
   servers as well as clients.


   The server is being queried as to its capabilities. A server that
   believes it can contact the user, such as a user agent where the user
   is logged in and has been recently active, MAY respond to this
   request with a capability set. A called user agent MAY return a
   status reflecting how it would have responded to an invitation, e.g.,
   600 (Busy). Such a server SHOULD return an Allow header field
   indicating the methods that it supports. Proxy and redirect servers
   simply forward the request without indicating their capabilities.

   This method MUST be supported by SIP proxy, redirect and user agent
   servers, registrars and clients.

4.2.4 BYE

   The user agent client uses BYE to indicate to the server that it
   wishes to abort the call. A BYE request is forwarded like an INVITE
   request. A caller SHOULD issue a BYE request before aborting a call
   ("hanging up"). Note that a BYE request may also be issued by the

   If the INVITE request contained a Location header, the callee MAY
   send a BYE request to that address rather than the From address.

   This method MUST be supported by proxy servers and SHOULD be
   supported by redirect and user agent SIP servers.

4.2.5 CANCEL

   The CANCEL request cancels a pending request with the same Call-ID,
   To, From and CSeq (sequence number only) header values, but does not
   affect a completed request. (A request is considered completed if the
   server has returned a final status response.)

   A user agent client or proxy client MAY issue a CANCEL request at any
   time. A proxy, in particular, MAY choose to send a CANCEL to
   destinations that have not yet returned a final response after it has
   received a 2xx or 6xx response for one or more of the parallel-search
   requests. A proxy that receives a CANCEL request forwards the request
   to all destinations with pending requests.

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   The Call-ID, To, the numeric part of CSeq and From headers in the
   CANCEL request are identical to those in the original request. This
   allows a CANCEL request to be matched with the request it cancels.
   However, to allow the client to distinguish responses to the CANCEL
   from those to the original request, the CSeq Method component is set
   to CANCEL. The Via header field is initialized to the proxy issuing
   the CANCEL request. (Thus, responses to this CANCEL request only
   reach the issuing proxy.)

   Once a user agent server has received a CANCEL, it MUST NOT issue a
   2xx response for the cancelled original request.

   A redirect or user agent server receiving a CANCEL request responds
   with a status of 200 (OK) if the Call-ID exists and a status of 481
   (Invalid Call-ID) if not, but takes no further action. In particular,
   any existing call is unaffected.

        The BYE request cannot be used to cancel branches of a
        parallel search, since several branches may, through
        intermediate proxies, find the same user agent server and
        then terminate the call.  To terminate a call instead of
        just pending searches, the UAC must use BYE instead of or
        in addition to CANCEL. While CANCEL can terminate any
        pending request other than ACK or CANCEL, it is typically
        useful only for INVITE. 200 responses to INVITE and 200
        responses to CANCEL are distinguished by the method in the
        Cseq header field, so there is no ambiguity.

   This method MUST be supported by proxy servers and SHOULD be
   supported by all other SIP server types.


   A client uses the REGISTER method to register the address listed in
   the To header with a SIP server.

   A user agent MAY register with a local server on startup by sending a
   REGISTER request to the well-known "all SIP servers" multicast
   address "sip.mcast.net" (, with a time-to-live value of 1.
   SIP user agents on the same subnet MAY listen to that address and use
   it to become aware of the location of other local users [16];
   however, they do not respond to the request.  A user agent MAY also
   be configured with the address of a registrar server to which it
   sends a REGISTER request upon startup.

   The meaning of the REGISTER request-header fields is defined as
   follows. We define "address-of-record" as the SIP address that the

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   registry knows the registrand, typically of the form "user@domain"
   rather than "user@host". In third-party registration, the entity
   issuing the request is different from the entity being registered.

   To: The To header field contains the address-of-record whose
        registration is to be created or updated.

   From: The From header field contains the address-of-record of the
        person responsible for the registration. For first-party
        registration, it is identical to the To header field value.

   Request-URI: The Request-URI names the destination of the
        registration request, i.e., the domain of the registrar. The
        user name MUST be empty. Generally, the domains in the Request-
        URI and the To header have the same value; however, it is
        possible to register as a "visitor", while maintaining one's
        name. For example, a traveller sip:alice@acme.com (To) may
        register under the Request-URI sip:@atlanta.ayh.org , with the
        former as the To field and the latter as the Request-URI. The
        request is no longer forwarded once it reached the server whose
        authoritative domain is the one listed in the Request-URI.

   Location: The request MAY contain a Location header field; future
        non-REGISTER requests for the URI given in the To field will be
        directed to the address(es) given in the Location header. It is
        RECOMMENDED that user agents include SIP URLs with both UDP and
        TCP transport parameters in their registration. If the
        registration contains a Location field whose URL includes a
        transport parameter, future requests will use that protocol.
        Otherwise, requests use the same transport protocol as used by
        the registration. However, a multicast REGISTER request still
        causes future requests to be unicast unless the maddr URL
        parameter explicitly requests otherwise. If the Location header
        does not contain a port number, the default SIP port number is
        used for future requests.

        We cannot require that registration and subsequent INVITE
        requests use the same transport protocol, as multicast
        registrations may be quite useful.

   If the request does not contain a Location header, the registration
   remains unchanged. Registrations that differ in one or more of host,
   port, transport or maddr from an existing registration are added to
   the list of registrations. If these parameters are the same as an
   existing registration, the new registration replaces the old one if
   it has a higher q value or, for the same value of q, if the ttl value
   is higher. All current locations MUST share the same action value.

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   Registrations that have a different action than current registrations
   for the same user are rejected with status of 409 (Conflict).

   A proxy server ignores the q parameter in processing non-REGISTER
   requests, while a redirect server simply returns the parameter in its
   Location header.

        Having the proxy server interpret the q parameter is not
        sufficient to guide proxy behavior, as it is not clear, for
        example, how long it should wait between trying addresses.

   If the registration is changed while a user agent or proxy server
   processes an invitation, the new information should be used.

        This allows a service known as "directed pick-up".

   A server SHOULD silently drop the registration after one hour, unless
   refreshed by the client. A client may request a lower or higher
   refresh interval through the Expires header (Section 6.19). Based on
   this request and its configuration, the server chooses the expiration
   interval and indicates it through the Expires header in the response.
   A single address (if host-independent) may be registered from several
   different clients.

   A client cancels an existing registration by sending a REGISTER
   request with an expiration time (Expires) of zero seconds for a
   particular Location or the wildcard Location designated by a "*" for
   all registrations. Registrations are matched based on the user, host,
   port and maddr parameters.

   The server SHOULD return the current list of registrations in the 200
   response as Location header fields.

   It is particularly important that REGISTER requests are authenticated
   since they allow to redirect future requests.

        Beyond its use as a simple location service, this method is
        needed if there are several SIP servers on a single host.
        In that case, only one of the servers can use the default
        port number. Each server that cannot would register with a
        server for the administrative domain.  Since a client may
        not have easy access to the host address or port number,
        using the source address and port from the request itself
        seems simpler.

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   Support of this method is RECOMMENDED.

4.3 Request-URI

   The Request-URI field is a SIP URL as described in Section 2 or a
   general URI. It indicates the user or service to which this request
   is being addressed. Unlike the To field, the Request-URI field may be
   re-written by proxies.

   When used as a Request-URI, a SIP-URL MUST NOT contain the
   transport-param, maddr-param, ttl-param, or headers elements. A
   server that receives a SIP-URL with these elements removes them
   before further processing.

        Typically, the UAC sets the Request-URI and To to the same
        SIP URL, presumed to remain unchanged over long time
        periods. However, if the UAC has cached a more direct path
        to the callee, e.g., from the Location header of a response
        to a previous request, the To would still contain the
        long-term, "public" address, while the Request-URI would be
        set to the cached address.

   Proxy and redirect servers may use the information in the Request-URI
   and request header fields to handle the request and possibly rewrite
   the Request-URI. For example, a request addressed to the generic
   address sip:sales@acme.com might be proxied to the particular person,
   e.g., sip:bob@ny.acme.com , with the To remaining as sales@acme.com
   ny.acme.com , Bob may have designated Alice as the temporary

   The host part of the Request-URI typically agrees with one of the
   host names of the server. If it does not, the server SHOULD proxy the
   request to the address indicated or return a 404 (Not Found) response
   if it is unwilling or unable to do so. For example, the Request-URI
   and server host name may disagree in the case of a firewall proxy
   that handles outgoing calls. This mode of operation similar to that
   of HTTP proxies.

   If a SIP server receives a request with a URI indicating a scheme
   other than SIP which that server does not understand, the server MUST
   return a 400 (Bad Request) response. It MUST do this even if the To
   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

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   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 6.30) and Unsupported
   (Section 6.38) fields.


        option-tag  ___   1*uric

   The creator of a new SIP option should either prefix the option with
   a reverse domain name or register the new option with the Internet
   Assigned Numbers Authority (IANA). For example,
   "com.foo.mynewfeature" is an apt name for a feature whose inventor
   can be reached at "foo.com". Options registered with IANA have the
   prefix "org.ietf.sip.", options described in RFCs have the prefix
   "org.ietf.rfc.N", where N is the RFC number. Option tags are case-

4.4.1 Registering New Option Tags with IANA

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

        o Name and description of option. The name may be of any length,
          but SHOULD be no more than twenty characters long. The name
          MUST NOT contain any spaces, control characters or periods.

        o Indication of who has change control over the option (for
          example, IETF, ISO, ITU-T, other international standardization
          bodies, a consortium or a particular company or group of

        o A reference to a further description, if available, for
          example (in order of preference) an RFC, a published paper, a
          patent filing, a technical report, documented source code or a
          computer manual;

        o For proprietary options, contact information (postal and email

        Borrowed from RTSP and the RTP AVP.

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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 )
                         [ 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

   The Status-Code is a 3-digit integer result code that indicates the
   outcome of the attempt to understand and satisfy the request. The
   Reason-Phrase is intended to give a short textual description of the
   Status-Code. The Status-Code is intended for use by automata, whereas
   the Reason-Phrase is intended for the human user. The client is not
   required to examine or display the Reason-Phrase.

        Status-Code       =    Informational                  Fig. 5
                         |     Success                        Fig. 5
                         |     Redirection                    Fig. 6
                         |     Client-Error                   Fig. 7

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                         |     Server-Error                   Fig. 8
                         |     Global-Failure                 Fig. 9
                         |     extension-code
        extension-code    =    3DIGIT
        Reason-Phrase     =    *<TEXT,  excluding CR, LF>

   We provide an overview of the Status-Code below, and provide full
   definitions in Section 7. The first digit of the Status-Code defines
   the class of response. The last two digits do not have any
   categorization role. SIP/2.0 allows 6 values for the first digit:

   1xx: Informational -- request received, continuing to process the

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

   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

   6xx: Global Failure -- the request is invalid at any server.

   Figures 5 through 9 present the individual values of the numeric
   response codes, and an example set of corresponding reason phrases
   for SIP/2.0. These reason phrases are only recommended; they may be
   replaced by local equivalents without affecting the protocol. Note
   that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
   codes in the range starting at x80 to avoid conflicts with newly
   defined HTTP response codes, and adds a new class, 6xx, of response

   SIP response codes are extensible. SIP applications are not required
   to understand the meaning of all registered response codes, though
   such understanding is obviously desirable. However, applications MUST
   understand the class of any response code, as indicated by the first
   digit, and treat any unrecognized response as being equivalent to the
   x00 response code of that class, with the exception that an
   unrecognized response MUST NOT be cached. For example, if a client
   receives an unrecognized response code of 431, it can safely assume
   that there was something wrong with its request and treat the
   response as if it had received a 400 (Bad Request) response code. In

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   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

        Informational    =    "100"    ;  Trying
                        |     "180"    ;  Ringing
                        |     "181"    ;  Call Is Being Forwarded
                        |     "182"    ;  Queued
        Success          =    "200"    ;  OK

   Figure 5: Informational and success status codes

        Redirection    =    "300"    ;  Multiple Choices
                      |     "301"    ;  Moved Permanently
                      |     "302"    ;  Moved Temporarily
                      |     "303"    ;  See Other
                      |     "305"    ;  Use Proxy
                      |     "380"    ;  Alternative Service

   Figure 6: Redirection status codes

6 Header Field Definitions

   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). The only requirement is that header fields which are hop-by-
   hop MUST appear before any header fields which are end-to-end.
   Proxies MUST NOT reorder or otherwise modify header fields other than
   by adding a new Via or other hop-by-hop field. Proxies MUST NOT, for
   example, change how header fields are broken across lines. This
   allows an authentication field to be added after the Via fields that
   will not be invalidated by proxies.

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        Client-Error    =    "400"    ;  Bad Request
                       |     "401"    ;  Unauthorized
                       |     "402"    ;  Payment Required
                       |     "403"    ;  Forbidden
                       |     "404"    ;  Not Found
                       |     "405"    ;  Method Not Allowed
                       |     "406"    ;  Not Acceptable
                       |     "407"    ;  Proxy Authentication Required
                       |     "408"    ;  Request Timeout
                       |     "409"    ;  Conflict
                       |     "410"    ;  Gone
                       |     "411"    ;  Length Required
                       |     "413"    ;  Request 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
                       |     "483"    ;  Too Many Hops
                       |     "484"    ;  Address Incomplete
                       |     "485"    ;  Ambiguous

   Figure 7: 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 8: Server error status codes

   The header fields required, optional and not applicable for each
   method are listed in Table 4. The table uses "o" to indicate
   optional, "m" mandatory and "-" for not applicable. A "*" indicates
   that the header fields are needed only if message body is not empty:
   The Content-Type and Content-Length headers are required when there
   is a valid message body (of non-zero length) associated with the

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

   Figure 9: Global failure status codes

   message (Section 8).

   The "type" column describes the request and response types for which
   the header field may be used. A numeric value indicates the status
   code for a response, while "R" refers to any request header, "r" to
   any response header. "g" and "e" designate general (Section 6.1) and
   entity header (Section 6.2) fields, respectively.

   The "enc." column describes whether this message header may be
   encrypted end-to-end. A "n" designates fields that MUST NOT be
   encrypted, while "c" designates fields that SHOULD be encrypted if
   encryption is used.

   The "e-e" column has a value of "e" for end-to-end and a value of "h"
   for hop-by-hop headers.

   Other headers may be added as required; a server MAY ignore optional
   headers that it does not understand. A compact form of these header
   fields is also defined in Section 9 for use over UDP when the request
   has to fit into a single packet and size is an issue.

   Table 5 in Appendix A indicates which system components should be
   capable of parsing which header fields.

6.1 General Header Fields

   General header fields apply to both request and response messages.
   The general-header field names can be extended reliably only in
   combination with a change in the protocol version. However, new or
   experimental header fields may be given the semantics of general
   header fields if all parties in the communication recognize them to
   be general-header fields. Unrecognized header fields are treated as
   entity-header fields.

6.2 Entity Header Fields

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   The entity-header fields define meta-information about the message-
   body or, if no body is present, about the resource identified by the
   request. The term "entity header" is an HTTP 1.1 term where the
   response body 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
   equivalent to the parameters of a programming language method

   The request-header field names can be extended reliably only in
   combination with a change in the protocol version. However, new or
   experimental header fields MAY be given the semantics of request-
   header fields if all parties in the communication recognize them to
   be request-header fields. Unrecognized header fields are treated as
   entity-header fields.

6.4 Response Header Fields

   The response-header fields allow the server to pass additional
   information about the response which cannot be placed in the Status-
   Line. These header fields give information about the server and about
   further access to the resource identified by the Request-URI.

   Response-header field names can be extended reliably only in
   combination with a change in the protocol version. However, new or
   experimental header fields MAY be given the semantics of response-
   header fields if all parties in the communication recognize them to
   be response-header fields. Unrecognized header fields are treated as
   entity-header fields.

6.5 End-to-end and Hop-by-hop Headers

   End-to-end headers must be transmitted unmodified across all proxies,
   while hop-by-hop headers may be modified or added by proxies.

6.6 Header Field Format

   Header fields (general-header, request-header, response-header, and
   entity-header) follow the same generic header format as that given in
   Section 3.1 of RFC 822 [24]. Each header field consists of a name
   followed by a colon (":") and the field value. Field names are case-

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                                 type            enc.    e-e   ACK   BYE   CAN   INV   OPT   REG
   Accept                          R                      e     -     -     -     o     o     o
   Accept-Encoding                 R                      e     -     -     -     o     o     o
   Accept-Language                 R              n       e     -     o     o     o     o     o
   Allow                          200                     e     -     -     -     -     m     -
   Allow                          405                     e     o     o     o     o     o     o
   Authorization                   R                      e     o     o     o     o     o     o
   Call-ID                         g              n       e     m     m     m     m     m     m
   Content-Encoding                e                      e     *     -     -     *     *     *
   Content-Length                  e                      e     m     -     -     m     m     m
   Content-Type                    e                      e     *     -     -     *     *     *
   CSeq                            g              n       e     m     m     m     m     m     m
   Date                            g                      e     o     o     o     o     o     o
   Encryption                      g              n       e     o     o     o     o     o     o
   Expires                         g                      e     -     -     -     o     -     o
   From                            g              n       e     m     m     m     m     m     m
   Hide                            R              n       h     o     o     o     o     o     o
   Location                        R                      e     o     -     -     o     o     o
   Location                       1xx                     e     -     -     -     o     o     -
   Location                       2xx                     e     -     -     -     o     o     o
   Location                       3xx                     e     -     o     -     o     o     o
   Location                       485                     e     -     o     -     o     o     o
   Max-Forwards                    R              n       e     o     o     o     o     o     o
   Organization                    g              c       h     -     -     -     o     o     o
   Proxy-Authenticate             407             n       h     o     o     o     o     o     o
   Proxy-Authorization             R              n       h     o     o     o     o     o     o
   Proxy-Require                   R              n       h     o     o     o     o     o     o
   Priority                        R              c       e     -     -     -     o     -     -
   Require                         R                      e     o     o     o     o     o     o
   Retry-After                     R              c       e     -     -     -     -     -     o
   Retry-After            404,480,503,600,603     c       e     o     o     o     o     o     o
   Response-Key                    R              c       e     -     o     o     o     o     o
   Record-Route                    R                      h     o     o     o     o     o     o
   Record-Route                   2xx                     h     o     o     o     o     o     o
   Route                           R                      h     -     o     o     o     o     o
   Server                          r              c       e     o     o     o     o     o     o
   Subject                         R              c       e     -     -     -     o     -     -
   Timestamp                       g                      e     o     o     o     o     o     o
   To                              g              n       e     m     m     m     m     m     m
   Unsupported                    420                     e     o     o     o     o     o     o
   User-Agent                      g              c       e     o     o     o     o     o     o
   Via                             g              n       e     m     m     m     m     m     m
   Warning                         r                      e     o     o     o     o     o     o
   WWW-Authenticate               401             c       e     o     o     o     o     o     o

   Table 4: Summary of header fields

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   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,
   without changing the semantics of the message, by appending each
   subsequent field-value to the first, each separated by a comma. The
   order in which header fields with the same field-name are received is
   therefore significant to the interpretation of the combined field
   value, and thus a proxy MUST NOT change the order of these field
   values when a message is forwarded.

   Field names are not case-sensitive, although their values may be.

6.7 Accept

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


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

6.8 Accept-Encoding

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   The Accept-Encoding request-header field is similar to Accept, but
   restricts the content-codings [H3.4.1] that are acceptable in the
   response. See [H14.3].

6.9 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, session descriptions or
   status responses carried as message bodies. A proxy may use this
   field to help select the destination for the call, for example, a
   human operator conversant in a language spoken by the caller.


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

6.10 Allow

   See [H14.7]. The Allow entity-header field lists the set of methods
   supported by the resource identified by the Request-URI. The purpose
   of this field is strictly to inform the recipient of valid methods
   associated with the resource. An Allow header field MUST be present
   in a 405 (Method Not Allowed) response and SHOULD be present in an
   OPTIONS response.

6.11 Authorization

   See [H14.8].

   A user agent that wishes to authenticate itself with a server --
   usually, but not necessarily, after receiving a 401 response -- MAY
   do so by including an Authorization request-header field with the
   request. The Authorization field value consists of credentials
   containing the authentication information of the user agent for the
   realm of the resource being requested.

6.12 Call-ID

   The Call-ID general header uniquely identifies a particular
   invitation or all registrations of a particular client. Note that a
   single multimedia conference may give rise to several calls with
   different Call-IDs, e.g., if a user invites a single individual
   several times to the same (long-running) conference.

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   For an INVITE request, a callee user agent server SHOULD NOT alert
   the user if the user has responded previously to the Call-ID in the
   INVITE request. If the user is already a member of the conference and
   the conference parameters contained in the session description have
   not changed, a callee user agent server MAY silently accept the call,
   regardless of the Call-ID. An invitation for an existing Call-ID or
   session may change the parameters of the conference. A client
   application MAY decide to simply indicate to the user that the
   conference parameters have been changed and accept the invitation
   automatically or it MAY require user confirmation.

   A user may be invited to the same conference or call using several
   different Call-IDs. If desired, the client may use identifiers within
   the session description to detect this duplication. For example, SDP
   contains a session id and version number in the origin (o) field.

   The REGISTER and OPTIONS methods use the Call-ID value to
   unambiguously match requests and responses. All REGISTER requests
   issued by a single client MUST use the same Call-ID.

        Since the Call-ID is generated by and for SIP, there is no
        reason to deal with the complexity of URL-encoding and
        case-ignoring string comparison.

        Call-ID     =    ( "Call-ID" | "i" ) ":" local-id "@" host
        local-id    =    *uric

   host MUST be either a fully qualified domain name or a globally
   routable IP address, while the local-id is a random identifier
   consisting of URI characters that is unique within host. It MUST NOT
   be reused for a different call.  Call-IDs are case-sensitive.  The
   use of a UUID as local-id is OPTIONAL. The UUID is a version-4
   (random) UUID [19].

        Using cryptographically random identifiers provides some
        protection against session hijacking.  Call-ID, To and From
        are needed to identify a call leg calls with third-party


     Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com

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6.13 Content-Encoding

   The Content-Encoding entity-header field is used as a modifier to the
   media-type. When present, its value indicates what additional content
   codings have been applied to the entity-body, and thus what decoding
   mechanisms MUST be applied in order to obtain the media-type
   referenced by the Content-Type header field.  Content-Encoding is
   primarily used to allow a document to be compressed without losing
   the identity of its underlying media type.  See [H14.12].

6.14 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

     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 MUST be set to zero. If a server receives a UDP request
   without Content-Length, it MUST assume that the request encompasses
   the remainder of the packet. If a response does not contain a
   Content-Length, the client assumes that it encompasses the remainder
   of the UDP packet or the data until the TCP connection is closed, as
   applicable.  Section 8 describes how to determine the length of the
   message body.

6.15 Content-Type

   The Content-Type entity-header field indicates the media type of the
   message-body sent to the recipient. The media-type element is defined
   in [H3.7].

        Content-Type    =    ( "Content-Type" ":" media-type

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   Examples of this header field are

     Content-Type: application/sdp
     Content-Type: text/html; charset=ISO-8859-4

6.16 CSeq

   Clients MUST add the CSeq (command sequence) general-header field to
   every request. A CSeq request header field contains a single decimal
   sequence number chosen by the requesting client, unique within a
   single value of Call-ID. The sequence number MUST be expressible as a
   32-bit unsigned integer. The initial value of the sequence number is
   arbitrary, but MUST be less than 2**31.  Consecutive requests that
   differ in request method, headers or body, but have the same Call-ID
   MUST contain strictly monotonically increasing and contiguous
   sequence numbers; sequence numbers do not wrap around.
   Retransmissions of the same request carry the same sequence number,
   but an INVITE with a different message body or different header
   fields (a "re-invitation") acquires a new, higher sequence number. A
   server MUST echo the CSeq value from the request in its response. If
   the Method value is missing, the server fills it in appropriately.

   The ACK and CANCEL requests MUST contain the same CSeq value as the
   INVITE request that it refers to, while a BYE request cancelling an
   invitation MUST have a higher sequence number.

   A user agent server MUST remember the highest sequence number for any
   INVITE request with the same Call-ID value. The server MUST respond
   to, but ignore any INVITE request with a lower sequence number.

   All requests spawned in a parallel search have the same CSeq value as
   the request triggering the parallel search.

        CSeq    =    "CSeq" ":" 1*DIGIT Method

        Strictly speaking, CSeq header fields are needed for any
        SIP request that can be cancelled by a BYE or CANCEL
        request or where a client can issue several requests for
        the same Call-ID in close succession. Without a sequence
        number, the response to an INVITE could be mistaken for the
        response to the cancellation (BYE or CANCEL). Also, if the
        network duplicates packets or if an ACK is delayed until

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        the server has sent an additional response, the client
        could interpret an old response as the response to a re-
        invitation issued shortly thereafter. Using CSeq also makes
        it easy for the server to distinguish different versions of
        an invitation, without comparing the message body.

   The Method value allows the client to distinguish the response to an
   INVITE request from that of a CANCEL response. CANCEL requests can be
   generated by proxies; if they were to increase the sequence number,
   it might conflict with a later request issued by the user agent for
   the same call.

   With a length of 32 bits, a server could generate, within a single
   call, one request a second for about 136 years before needing to wrap
   around.  The initial value of the sequence number is chosen so that
   subsequent requests within the same call will not wrap around. A
   non-zero initial value allows to use a time-based initial sequence
   number, which protects against ambiguities when clients are re-
   invited to the same call after rebooting. A client could, for
   example, choose the 31 most significant bits of a 32-bit second clock
   as an initial sequence number.

   Forked requests must have the same CSeq as there would be ambiguity
   otherwise between these forked requests and later BYE issued by the
   client user agent.


     CSeq: 4711 INVITE

6.17 Date

   General header field. See [H14.19].

        The Date header field can be used by simple end systems
        without a battery-backed clock to acquire a notion of
        current time.

6.18 Encryption

   The Encryption general-header field specifies that the content has
   been encrypted. Section 13 describes the overall SIP security
   architecture and algorithms. This header field is intended for end-
   to-end encryption of requests and responses. Requests are encrypted

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   with a public key belonging to the entity named in the To header
   field. Responses are encrypted with the public key conveyed in the
   Response-Key header field.

        SIP chose not to adopt HTTP's Content-Transfer-Encoding
        header because the encrypted body may contain additional
        SIP header fields as well as the body of the message.

   For any encrypted message, at least the message body and possibly
   other message header fields are encrypted. An application receiving a
   request or response containing an Encryption header field decrypts
   the body and then concatenates the plaintext to the request line and
   headers of the original message. Message headers in the decrypted
   part completely replace those with the same field name in the
   plaintext part.  (Note: If only the body of the message is to be
   encrypted, the body has to be prefixed with CRLF to allow proper
   concatenation.) Note that the request method and Request-URI cannot
   be encrypted.

        Encryption only provides privacy; the recipient has no
        guarantee that the request or response came from the party
        listed in the From message header, only that the sender
        used the recipients public key. However, proxies will not
        be able to modify the request or response.

        Encryption           =    "Encryption" ":" encryption-scheme 1*SP
        encryption-scheme    =    token
        encryption-params    =    token "=" ( token | quoted-string )

        The token indicates the form of encryption used; it is
        described in section 13.

   The following example for a message encrypted with ASCII-armored PGP
   was generated by applying "pgp -ea" to the payload to be encrypted.

   INVITE sip:watson@boston.bell-telephone.com SIP/2.0
   Via: SIP/2.0/UDP
   From: <sip:a.g.bell@bell-telephone.com>
   To: T. A. Watson <sip:watson@bell-telephone.com>
   Call-ID: 187602141351@worcester.bell-telephone.com
   Content-Length: 885
   Encryption: PGP version=2.6.2,encoding=ascii

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   Since proxies may base their forwarding decision on any combination
   of SIP header fields, there is no guarantee that an encrypted request
   "hiding" header fields will reach the same destination as an
   otherwise identical un-encrypted request.

6.19 Expires

   The Expires entity-header field gives the date and time after which
   the message content expires.

   This header field is currently defined only for the REGISTER and
   INVITE methods. For REGISTER, it is a request and response-header
   field. In a REGISTER request, the client indicates how long it wishes
   the registration to be valid. In the response, the server indicates
   the earliest expiration time of all registrations. The server MAY
   choose a shorter time interval than that requested by the client, but
   SHOULD NOT choose a longer one.

   For INVITE, 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 invitation is time-limited. A user interface may take this
   as a hint to leave the invitation window on the screen even if the
   user is not currently at the workstation. This also limits the
   duration of a search. If the request expires before the search
   completes, the proxy returns a 408 (Request Timeout) status. In a 302
   (Moved Temporarily) response, a server can advise the client of the
   maximal duration of the redirection.

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

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   request. The latter approach is preferable for short durations, as it
   does not depend on clients and servers sharing a synchronized clock.

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

   Two examples of its use are

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

6.20 From

   Requests and responses MUST contain a From general-header field,
   indicating the initiator of the request. The From field MAY contain
   the Tag parameter. The server copies the To and From header fields
   from the request to the response  and MUST add the tag parameter to
   the To field in the response if the URL in the To field is not a
   fully qualified hostname . The optional display-name is meant to be
   rendered by a human-user interface.

   The SIP-URL MUST NOT contain the transport-param, maddr-param, ttl-
   param, or headers elements.  A server that receives a SIP-URL with
   these elements removes them before further processing.

        From            =    ( "From" | "f" ) ":" ( name-addr | addr-spec )
        name-addr       =    [ display-name ] "<" addr-spec ">"
        addr-spec       =    SIP-URL | URI
        display-name    =    *token | quoted-string


     From: A. G. Bell <sip:agb@bell-telephone.com>
     From: sip:+12125551212@server.phone2net.com
     From: Anonymous <sip:c8oqz84zk7z@privacy.org>

        Call-ID, To and From are needed to identify a call leg

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        matters in calls with third-party control. The format is
        similar to the equivalent RFC 822 [24] header, but with a
        URI instead of just an email address.

6.21 Hide

   The Hide request header field indicates that the path comprised of
   the Via header fields (Section 6.40) should be hidden from subsequent
   proxies and user agents. It can take two forms:  Hide: route and
   Hide: hop. Hide header fields are typically added by the client user
   agent, but MAY be added by any proxy along the path.

   If a request contains the "Hide: route" header field, all following
   proxies SHOULD hide their previous hop. If a request contains the
   "Hide: hop" header field, only the next proxy SHOULD hide the
   previous hop and then remove the Hide option unless it also wants to
   remain anonymous.

   A server hides the previous hop by encrypting the host and port parts
   of the top-most Via header with an algorithm of its choice. Servers
   SHOULD add additional "salt" to the host and port information prior
   to encryption to prevent malicious downstream proxies from guessing
   earlier parts of the path based on seeing identical encrypted Via
   headers. Hidden Via fields are marked with the hidden Via option, as
   described in Section 6.40.

   A server that is capable of hiding Via headers MUST attempt to
   decrypt all Via headers marked as "hidden" to perform loop detection.
   Servers that are not capable of hiding can ignore hidden Via fields
   in their loop detection algorithm.

        If hidden headers were not marked, a proxy would have to
        decrypt all headers to detect loops, just in case one was
        encrypted, as the Hide: Hop option may have been removed
        along the way.

   A host MUST NOT add such a "Hide: hop" header field unless it can
   guarantee it will only send a request for this destination to the
   same next hop. The reason for this is that it is possible that the
   request will loop back through this same hop from a downstream proxy.
   The loop will be detected by the next hop if the choice of next hop
   is fixed, but could loop an arbitrary number of times otherwise.

   A client requesting "Hide: route" can only rely on keeping the
   request path private if it sends the request to a trusted proxy.
   Hiding the route of a SIP request may be of limited value if the
   request results in data packets being exchanged directly between the

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   calling and called user agent.

   The use of Hide header fields is discouraged unless path privacy is
   truly needed; Hide fields impose extra processing costs and
   restrictions for proxies and can cause requests to generate 482 (Loop
   Detected) responses that could otherwise be avoided.

   The encryption of Via header fields is described in more detail in
   Section 13.

   The Hide header field has the following syntax:

        Hide    =    "Hide" ":" ( "route" | "hop" )

6.22 Location

   The Location general-header field can appear in requests, 1xx, 2xx
   responses and 3xx responses.

   REGISTER requests: REGISTER requests MAY contain a Location header
        field indicating at which locations the user may be reachable.
        The REGISTER request defines a wildcard Location field, "*",
        which MUST only be used with Expires: 0 to remove all
        registrations for a particular user.

   INVITE and ACK requests: INVITE and ACK requests SHOULD contain
        Location headers indicating from which location the request is

        This allows the callee to send a BYE directly to the caller
        instead of through a series of proxies. The Via header is
        not sufficient since the desired address may be that of a

   INVITE 2xx responses: A user agent server sending a definitive,
        positive response (2xx) MAY insert a Location response header
        indicating the SIP address under which it is reachable most
        directly for future SIP requests, such as ACK, within the same
        Call-ID. This may be the address of the server itself or that of
        a proxy, e.g., if the host is behind a firewall. The value of
        this Location header is copied into the Request-URI of
        subsequent requests for this call.

        The Location value MUST NOT be cached across calls, as it
        may not represent the most desirable location for a

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        particular destination address.

   INVITE 1xx responses: A UAS sending a provisional response (1xx) MAY
        insert a Location response header. It has the same semantics in
        a 1xx response as a 2xx INVITE response.

   REGISTER 2xx responses: Similarly, a REGISTER response MAY return all
        locations at which the user is currently reachable.

   3xx and 485 responses: The Location response-header field can be used
        with a 3xx or 485 response codes to indicate one or more
        alternate addresses to try. It can appear in responses to BYE,
        INVITE and OPTIONS methods. The Location header field contains
        URIs giving the new locations or user names to try, or may
        simply specify additional transport parameters. A 300 (Multiple
        Choices), 301 (Moved Permanently), 302 (Moved Temporarily) or
        485 (Ambiguous) response SHOULD contain a Location field
        containing URIs of new addressed to be tried. A 301 or 302
        response may also give the same location and username that was
        being tried but specify additional transport parameters such as
        a different server or multicast address to try or a change of
        SIP transport from UDP to TCP or vice versa. The client copies
        the host, user and tag elements of the Location URI into the
        Request-URI of the redirected request and directs the request to
        the address specified in the maddr and port parameter, using the
        transport protocol given in the transport parameter. If maddr is
        a multicast address, the value of ttl is used as the time-to-
        live value.

   Note that the Location header may also refer to a different entity
   than the one originally called. For example, a SIP call connected to
   GSTN gateway may need to deliver a special information announcement
   such as "The number you have dialed has been changed."

   A Location response header may contain any suitable URI indicating
   where the called party may be reached, not limited to SIP URLs. For
   example, it may contain a phone or fax

   a mailto: (RFC 2368, [25]) or irc: URL.

   The following parameters are defined. Additional parameters may be
   defined in other specifications.

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

   action: The action is only used when registering with the REGISTER

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        request. It indicates whether the client wishes that the server
        proxies or redirects future requests intended for the client.
        The action taken if this parameter is not specified depends on
        server configuration. In its response, the registrar SHOULD
        indicate the mode used. This parameter is ignored for other

        Location    =    ( "Location" | "m" ) ":"
                         ("*" | (1# (( SIP-URL | URI )
                         [ LWS *( ";" location-params ) ] [ comment ] ))

        location-params       =    "q"                     "="     qvalue
                              |    "action"                "="     "proxy" | "redirect"
                              |    extension-attribute
        extension-attribute   =    extension-name         [ "="    extension-value ]


     Location: sip:watson@worcester.bell-telephone.com ;q=0.7,
        mailto:watson@bell-telephone.com ;q=0.1

6.23 Max-Forwards

   The Max-Forwards request-header field may be used with any SIP method
   to limit the number of proxies or gateways that can forward the
   request to the next inbound server. This can also be useful when the
   client is attempting to trace a request chain which appears to be
   failing or looping in mid-chain. [H14.31]

        Max-Forwards    =    "Max-Forwards" ":" 1*DIGIT

   The Max-Forwards value is a decimal integer indicating the remaining
   number of times this request message may be forwarded.

   Each proxy or gateway recipient of a request containing a Max-
   Forwards header field MUST check and update its value prior to

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   forwarding the request. If the received value is zero (0), the
   recipient MUST NOT forward the request. Instead, for the OPTIONS and
   REGISTER methods, it MUST respond as the final recipient. For all
   other methods, the server returns 483 (Too many hops).

   If the received Max-Forwards value is greater than zero, then the
   forwarded message MUST contain an updated Max-Forwards field with a
   value decremented by one (1).


     Max-Forwards: 6

6.24 Organization

   The Organization general-header field conveys the name of the
   organization to which the entity issuing the request or response
   belongs.  It may also be inserted by proxies at the boundary of an
   organization and may be used by client software to filter calls.

        Organization    =    "Organization" ":" *text

6.25 Priority

   The Priority request header signals the urgency of the call to the

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

   The value of "emergency" should only be used when life, limb or
   property are in imminent danger.


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

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     Subject: Weekend plans
     Priority: non-urgent

        These are the values of RFC 2076 [26], with the addition of

6.26 Proxy-Authenticate

   The Proxy-Authenticate response-header field MUST be included as part
   of a 407 (Proxy Authentication Required) response. The field value
   consists of a challenge that indicates the authentication scheme and
   parameters applicable to the proxy for this Request-URI. See [H14.33]
   for further details.

   A client SHOULD cache the credentials used for a particular proxy
   server and realm for the next request to that server. Credentials
   are, in general, valid for a specific value of the Request-URI at a
   particular proxy server. If a client contacts a proxy server that has
   required authentication in the past, but the client does not have
   credentials for the particular Request-URI, it MAY attempt to use the
   most-recently used credential. The server responds with 401
   (Unauthorized) if the client guessed wrong.

        This suggested caching behavior is motivated by proxies
        restricting phone calls to authenticated users. It seems
        likely that in most cases, all destinations require the
        same password. Note that end-to-end authentication is
        likely to be destination-specific.

6.27 Proxy-Authorization

   The Proxy-Authorization request-header field allows the client to
   identify itself (or its user) to a proxy which requires
   authentication. The Proxy-Authorization field value consists of
   credentials containing the authentication information of the user
   agent for the proxy and/or realm of the resource being requested. See
   [H14.34] for further details.

6.28 Proxy-Require

   The Proxy-Require header is used to indicate proxy-sensitive features
   that MUST be supported by the proxy. Any Proxy-Require header
   features that are not supported by the proxy MUST be negatively
   acknowledged by the proxy to the client if not supported. Servers

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   treat this field identically to the Require field.

   See Section 6.30 for more details on the mechanics of this message
   and a usage example.

6.29 Record-Route

   The Record-Route request and response header field is added to an
   INVITE request by any proxy that insists on being in the path of
   subsequent ACK and BYE requests for the same call. It contains a
   globally reachable Request-URI that identifies the proxy server. Each
   proxy server adds its Request-URI to the beginning of the list.

   The server copies the Record-Route header unchanged into the
   response. (Record-Route is only relevant for 2xx responses.)

   The calling user agent client copies the Record-Route header into a
   Route header of subsequent requests, reversing the order of requests,
   so that the first entry is closest to the caller. If the response
   contained a Location header field, the calling user agent adds its
   content as the last Route header. Unless this would cause a loop, any
   client MUST send any subsequent requests for this Call-ID to the
   first Request-URI in the Route request header and remove that entry.

        Some proxies, such as those controlling firewalls or in an
        automatic call distribution (ACD) system, need to maintain
        call state and thus need to receive any BYE and ACK packets
        for the call.

   The Record-Route header field has the following syntax:

        Record-Route    =    "Record-Route" ":" 1# SIP-URL

   Example for a request that has traversed the hosts ieee.org and
   bell-telephone.com , in that order:

     Record-Route: sip:a.g.bell@bell-telephone.com, sip:a.bell@ieee.org

6.30 Require

   The Require request header is used by clients to tell user agent
   servers about options that the client expects the server to support
   in order to properly process the request. If a server does not

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   understand the option, it MUST respond by returning status code 420
   (Bad Extension) and list those options it does not understand in the
   Unsupported header.

        Require    =    "Require" ":" 1#option-tag


   C->S:   INVITE sip:watson@bell-telephone.com SIP/2.0
           Require: com.example.billing
           Payment: sheep_skins, conch_shells

   S->C:   SIP/2.0 420 Bad Extension
           Unsupported: com.example.billing

        This is to make sure that the client-server interaction
        will proceed without delay when all options are understood
        by both sides, and only slow down if options are not
        understood (as in the example above).  For a well-matched
        client-server pair, the interaction proceeds quickly,
        saving a round-trip often required by negotiation
        mechanisms. In addition, it also removes ambiguity when the
        client requires features that the server does not
        understand. Some features, such as call handling fields,
        are only of interest to end systems.

   Proxy and redirect servers MUST ignore features that are not
   understood. If a particular extension requires that intermediate
   devices support it, the extension should be tagged in the Proxy-
   Require field instead (see Section 6.28).

6.31 Response-Key

   The Response-Key request header field can be used by a client to
   request the key that the called user agent SHOULD use to encrypt the
   response with. The syntax is:

        Response-Key    =    "Response-Key" ":" key-scheme 1*SP #key-param
        key-scheme      =    token
        key-param       =    token "=" ( token | quoted-string )

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   The key-scheme gives the type of encryption to be used for the
   response. Section 13 describes security schemes.

   If the client insists that the server return an encrypted response,
   it includes a
                  Require: org.ietf.sip.encrypt-response
   header field in its request. If the client cannot encrypt for
   whatever reason, it MUST follow normal Require header field
   procedures and return a 420 (Bad Extension) response. If this Require
   header is not present, a client SHOULD still encrypt, but MAY return
   an unencrypted response if unable to.

6.32 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), or 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.

   A REGISTER request may include this header field when deleting
   registrations with Location: *; Expires: 0. The Retry-After value
   then indicates when the user might again be reachable. The registrar
   MAY then include this information in responses to future calls.

   An optional comment can be used to indicate additional information
   about the time of callback. An optional duration parameter indicates
   how long the called party will be reachable starting at the initial
   time of availability. If no duration parameter is given, the service
   is assumed to be available indefinitely.

        Retry-After    =    "Retry-After" ":" ( 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 GMT;duration=3600
     Retry-After: 120

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   In the third example, the callee is reachable for one hour starting
   at 21:00 GMT. In the last example, the delay is 2 minutes.

6.33 Route

   The Route request header determines the route taken by a request.
   Each host removes the first entry and then proxies the request to the
   host listed in that entry, also using it as the Request-URI. The
   operation is further described in Section 6.29.

   The Route header field has the following syntax:

        Route    =    "Route" ":" 1# request-uri

6.34 Server

   The Server response-header field contains information about the
   software used by the user agent server to handle the request. See

6.35 Subject

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

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


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

6.36 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

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   that have elapsed since it has received the request. The timestamp is
   used by the client to compute the round-trip time to the server so
   that it can adjust the timeout value for retransmissions.

        Timestamp    =    "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
        delay        =    *(DIGIT) [ "." *(DIGIT) ]

6.37 To

   The To general-header field specifies recipient of the request, with
   the same SIP URL syntax as the From field.

        To    =    ( "To" | "t" ) ":" ( name-addr | addr-spec )

   The UAS copies the To header into its response, and MUST add a tag
   parameter if the URL is not a full qualified hostname.

   A SIP server returns a 400 (Bad Request) response if it receives a
   request with a To header field containing a URI with a scheme it does
   not recognize.


     To: The Operator <sip:operator@cs.columbia.edu>
     To: sip:+12125551212@server.phone2net.com

        Call-ID, To and From are needed to identify a call leg
        matters in calls with third-party control. The tag is added
        to the To header in the response to allow forking of future
        requests for the same call by proxies, while addressing
        only one of the possibly several responding user agent
        servers. It also allows several instances of the callee to
        send requests that can be distinguished.

6.38 Unsupported

   The Unsupported response header lists the features not supported by
   the server. See Section 6.30 for a usage example and motivation.

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6.39 User-Agent

   The User-Agent general-header field contains information about the
   client user agent originating the request. See [H14.42].

6.40 Via

   The Via field indicates the path taken by the request so far.  This
   prevents request looping and ensures replies take the same path as
   the requests, which assists in firewall traversal and other unusual
   routing situations.

6.40.1 Requests

   The client originating the request MUST insert into the request a Via
   field containing its host name or network address and, if not the
   default port number, the port number at which it wishes to receive
   responses. (Note that this port number may differ from the UDP source
   port number of the request.) A fully-qualified domain name is
   RECOMMENDED. Each subsequent proxy server that sends the request
   onwards MUST add its own additional Via field before any existing Via
   fields. A proxy that receives a redirection (3xx) response and then
   searches recursively, MUST use the same Via headers as on the
   original request.

   A proxy SHOULD check the top-most Via header to ensure that it
   contains the sender's correct network address, as seen from that
   proxy. If the sender's address is incorrect, the proxy should add an
   additional received attribute, as described 6.40.2.

        A host behind a network address translator (NAT) or
        firewall may not be able to insert a network address into
        the Via header that can be reached by the next hop beyond
        the NAT. Hosts behind NATs or NAPTs should insert the local
        port number of the outgoing socket, rather than the port
        number for incoming requests, as NAPTs assume that
        responses return with reversed source and destination

   A proxy sending a request to a multicast address MUST add the maddr
   parameter to its Via header field, and SHOULD add the ttl parameter.
   If a server receives a request which contained an maddr parameter in
   the topmost Via field, it should SHOULD send the response to the
   multicast address listed in the maddr parameter.

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

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        This prevents a malfunctioning proxy server from causing
        loops. Also, it cannot be guaranteed that a proxy server
        can always detect that the address returned by a location
        service refers to a host listed in the Via list, as a
        single host may have aliases or several network interfaces.

6.40.2 Receiver-tagged Via Fields

   Normally, every host that sends or forwards a SIP message adds a Via
   field indicating the path traversed. However, it is possible that
   Network Address Translators (NAT) may change the source address of
   the request (e.g., from net-10 to a globally routable address), in
   which case the Via field cannot be relied on to route replies. To
   prevent this, a proxy SHOULD check the top-most Via header to ensure
   that it contains the sender's correct network address, as seen from
   that proxy. If the sender's address is incorrect, the proxy should
   add a received tag to the Via field inserted by the previous hop.
   Such a modified Via field is known as a receiver-tagged Via field. An
   example is:

     Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
     Via: SIP/2.0/UDP ;received=

   In this example, the message originated from and traversed a
   NAT with the external address border.ieee.org ( to
   reach erlang.bell-telephone.com and tagged the previous hop's Via
   field with the address that it actually came from.

6.40.3 Responses

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

        1.   The first Via field should indicate the proxy or client
             processing this response. If it does not, discard the
             message.  Otherwise, remove this Via field.

        2.   If the second Via field contains a maddr parameter, the
             response is sent to the address listed there, using the
             port indicated in sent-by, or 5060 if none is present. The
             response SHOULD be sent using the TTL indicated in the ttl
             parameter, or with a TTL of 1 if none is present.

        3.   Otherwise, if the second Via field is a receiver-tagged
             field (Section 6.40.2), send the message to the address in

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             the received tag, using the port present in sent-by, or
             port 5060 if none is present.

        4.   Otherwise, send the message to the address indicated by
             sent-by in the second Via field.

        5.   If there is no second Via field, this response is destined
             for this client.

   A user agent server or redirect server should send a response by
   pretending to insert the received tag into the topmost Via header in
   the request, and treating this header as the second Via in the above

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

6.40.4 Syntax

   The format for a Via header is shown in Fig. 10.

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

   Figure 10: Syntax of Via header field

   The defaults for "protocol-name" and "transport" are "SIP" and "UDP",
   respectively. The "maddr" parameter, designating the multicast
   address, and the "ttl" parameter, designating the time-to-live (TTL)

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   value, are included only if the request was sent via multicast. The
   "received" parameter is added only for receiver-added Via fields
   (Section 6.40.2). For reasons of privacy, a client or proxy may wish
   to hide its Via information by encrypting it (see Section 6.21). The
   "hidden" parameter is included if this header was hidden by the
   upstream proxy (see 6.21).  Note that privacy of the proxy relies on
   the cooperation of the next hop, as the next-hop proxy will, by
   necessity, know the IP address and port number of the source host.

   The "branch" parameter is included by every forking proxy.  The token
   MUST be unique for each distinct request generated when a proxy
   forks. When a response arrives at the proxy it can use the branch
   value to figure out which branch the response corresponds to. A proxy
   which generates a single request (non-forking) MAY also insert the
   "branch" parameter. The identifier has to be unique only within a set
   of isomorphic requests.

     Via: SIP/2.0/UDP first.example.com:4000;ttl=16
       ;maddr= (Example)
     Via: SIP/2.0/UDP adk8

6.41 Warning

   The Warning response-header field is used to carry additional
   information about the status of a response. Warning headers are sent
   with responses and have the following format:

        Warning          =    "Warning" ":" 1#warning-value
        warning-value    =    warn-code SP warn-agent SP warn-text
        warn-code        =    3DIGIT
        warn-agent       =    ( host [ ":" port ] ) | pseudonym
                              ;  the name or pseudonym of the server adding
                              ;  the Warning header, for use in debugging
        warn-text        =    quoted-string

   A response may carry more than one Warning header.

   The warn-text should be in a natural language that is most likely to
   be intelligible to the human user receiving the response.  This
   decision 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

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   language is English.

   Any server may add Warning headers to a response. Proxy servers MUST
   place additional Warning headers before any Authorization headers.
   Within that constraint, Warning headers MUST be added after any
   existing Warning headers not covered by a signature. A proxy server
   MUST NOT delete any Warning header that it received with a response.

   When multiple Warning headers are attached to a response, the user
   agent SHOULD display as many of them as possible, in the order that
   they appear in the response. If it is not possible to display all of
   the warnings, the user agent first displays warnings that appear
   early in the response. Systems that generate multiple Warning headers
   should order them with this user agent behavior in mind.

   The warn-code consists of three digits. The first digit indicates the
   significance of the warning, with 3xx indicating a warning that did
   not cause the request to fail and 4xx indicating a fatal error
   condition that contributed to the failure of the request.

   This is a list of the currently-defined warn-codes, each with a
   recommended warn-text in English, and a description of its meaning.
   Additional warn-codes may be defined through IANA. Note that these
   warnings describe failures induced by the session description.

   x01 Insufficient bandwidth: The bandwidth specified in the session
        description or defined by the media exceeds that known to be

   x02 Incompatible transport protocol: One or more transport protocols
        described in the request are not available.

   x03 Incompatible network protocol: One or more network protocols
        described in the request are not available.

   x04 Incompatible network address formats: One or more network address
        formats described in the request are not available.

   x05 Incompatible media format: One or more media formats described in
        the request are not available.

   x06 Incompatible bandwidth description: One or more bandwidth
        descriptions contained in the request were not understood.

   x07 Multicast not available: The site where the user is located does
        not support multicast.

   x08 Unicast not available: The site where the user is located does

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        not support unicast communication (usually due to the presence
        of a firewall).

   x09 Media type not available: One or more media types contained in
        the request are not available.

   x10 Attribute not understood: One or more of the media attributes in
        the request are not supported.

   x09 Session description parameter not understood: A parameter other
        than those listed above was not understood.

   x99 Miscellaneous warning: The warning text may include arbitrary
        information to be presented to a human user, or logged. A system
        receiving this warning MUST NOT take any automated action.

        1xx and 2xx have been taken by HTTP/1.1.


     Warning: 309 isi.edu "Session parameter 'foo' not understood"
     Warning: 404 isi.edu "Incompatible network address type 'E.164'"

6.42 WWW-Authenticate

   The WWW-Authenticate response-header field MUST be included in 401
   (Unauthorized) response messages. The field value consists of at
   least one challenge that indicates the authentication scheme(s) and
   parameters applicable to the Request-URI. See [H14.46] and [27].

   The content of the realm parameter SHOULD be displayed to the user. A
   user agent SHOULD cache the authorization credentials for a given
   value of the destination (To header) and realm and attempt to re-use
   these values on the next request for that destination.

   In addition to the "basic" and "digest" authentication schemes
   defined in the specifications cited above, SIP defines a new scheme,
   PGP (RFC 2015, [28]), Section 14. Other schemes, such as S-MIME, are
   for further study.

7 Status Code Definitions

   The response codes are consistent with, and extend, HTTP/1.1 response
   codes. Not all HTTP/1.1 response codes are appropriate, and only

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   those that are appropriate are given here. Other HTTP/1.1 response
   codes should not be used. Response codes not defined by HTTP/1.1 have
   codes x80 upwards to avoid clashes with future HTTP response codes.
   Also, SIP defines a new class, 6xx. The default behavior for unknown
   response codes is given for each category of codes.

7.1 Informational 1xx

   Informational responses indicate that the server or proxy contacted
   is performing some further action and does not yet have a definitive
   response. The client SHOULD wait for a further response from the
   server, and the server SHOULD send such a response without further
   prompting. Typically a server should send a 1xx response if it
   expects to take more than 200 ms to obtain a final response.  A
   server can issue zero or more 1xx responses, with no restriction on
   their ordering or uniqueness. Note that 1xx responses are not
   transmitted reliably, that is, they do not cause the client to send
   an ACK. Servers are free to retransmit informational responses and
   clients can inquire about the current state of call processing by
   re-sending the request.

7.1.1 100 Trying

   Some unspecified action is being taken on behalf of this call (e.g.,
   a database is being consulted), but the user has not yet been

7.1.2 180 Ringing

   The called user agent has located a possible location where the user
   has registered recently and is trying to alert the user.

7.1.3 181 Call Is Being Forwarded

   A proxy server MAY use this status code to indicate that the call is
   being forwarded to a different set of destinations. The new
   destinations are listed in Location headers. Proxies SHOULD be
   configurable not to reveal this information.

7.1.4 182 Queued

   The called party is temporarily unavailable, but the callee has
   decided to queue the call rather than reject it. When the callee
   becomes available, it will return the appropriate final status
   response. The reason phrase MAY give further details about the status
   of the call, e.g., "5 calls queued; expected waiting time is 15
   minutes". The server MAY issue several 182 responses to update the
   caller about the status of the queued call.

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7.2 Successful 2xx

   The request was successful and MUST terminate a search.

7.2.1 200 OK

   The request has succeeded. The information returned with the response
   depends on the method used in the request, for example:

   BYE: The call has been terminated. The message body is empty.

   CANCEL: The search has been cancelled. The message body is empty.

   INVITE: The callee has agreed to participate; the message body
        indicates the callee's capabilities.

   OPTIONS: The callee has agreed to share its capabilities, included in
        the message body.

   REGISTER: The registration has succeeded. The client treats the
        message body according to its Content-Type.

7.3 Redirection 3xx

   3xx responses give information about the user's new location, or
   about alternative services that 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.

   Any redirection (3xx) response MUST NOT suggest any of the addresses
   in the Via (Section 6.40) path of the request in the Location header
   field. (Addresses match if their host and port number match.)

   To avoid forwarding loops, a user agent client or proxy MUST check
   whether the address returned by a redirect server equals an address
   tried earlier.

7.3.1 300 Multiple Choices

   The address in the request resolved to several choices, each with its
   own specific location, and the user (or user agent) can select a
   preferred communication end point and redirect its request to that

   The response SHOULD include an entity containing a list of resource
   characteristics and location(s) from which the user or user agent can
   choose the one most appropriate, if allowed by the Accept request
   header. The entity format is specified by the media type given in the

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   Content-Type header field. The choices SHOULD also be listed as
   Location fields (Section 6.22).  Unlike HTTP, the SIP response may
   contain several Location fields or a list of addresses in a Location
   field. User agents MAY use the Location field value for automatic
   redirection or MAY ask the user to confirm a choice. However, this
   specification does not define any standard for such automatic

        This header is appropriate if the callee can be reached at
        several different locations and the server cannot or
        prefers not to proxy the request.

7.3.2 301 Moved Permanently

   The user can no longer be found at the address in the Request-URI and
   the requesting client should retry at the new address given by the
   Location header field (Section 6.22). The caller SHOULD update any
   local directories, address books and user location caches with this
   new value and redirect future requests to the address(es) listed.

7.3.3 302 Moved Temporarily

   The requesting client should retry the request at the new address(es)
   given by the Location header field (Section 6.22). The duration of
   the redirection can be indicated through an Expires (Section 6.19)

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

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.

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7.4.3 402 Payment Required

   Reserved for future use.

7.4.4 403 Forbidden

   The server understood the request, but is refusing to fulfill it.
   Authorization will not help, and the request should not be repeated.

7.4.5 404 Not Found

   The server has definitive information that the user does not exist at
   the domain specified in the Request-URI. This status is also returned
   if the domain in the Request-URI does not match any of the domains
   handled by the recipient of the request.

7.4.6 405 Method Not Allowed

   The method specified in the Request-Line is not allowed for the
   address identified by the Request-URI. The response MUST include an
   Allow header containing a list of valid methods for the indicated

7.4.7 406 Not Acceptable

   The resource identified by the request is only capable of generating
   response entities which have content characteristics not acceptable
   according to the accept headers sent in the request.

7.4.8 407 Proxy Authentication Required

   This code is similar to 401 (Unauthorized), but indicates that the
   client MUST first authenticate itself with the proxy. The proxy MUST
   return a Proxy-Authenticate header field (section 6.26) containing a
   challenge applicable to the proxy for the requested resource. The
   client MAY repeat the request with a suitable Proxy-Authorization
   header field (section 6.27). SIP access authentication is explained
   in section 13.2 and [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.

7.4.9 408 Request Timeout

   The server could not produce a response, e.g., a user location,
   within the time indicated in the Expires request-header field. The
   client MAY repeat the request without modifications at any later

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7.4.10 409 Conflict

   The request could not be completed due to a conflict with the current
   state of the resource. This response is returned is the action
   parameter in a REGISTER request conflicts with existing

7.4.11 414 Request-URI Too Long

   The server is refusing to service the request because the Request-URI
   is longer than the server is willing to interpret.

7.4.12 415 Unsupported Media Type

   The server is refusing to service the request because the message
   body of the request is in a format not supported by the requested
   resource for the requested method.

7.4.13 420 Bad Extension

   The server did not understand the protocol extension specified in a
   Require (Section 6.30) header field.

7.4.14 480 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 a more precise cause as to why the callee is
   unavailable. This value SHOULD be setable by the user agent.

7.4.15 481 Invalid Call-ID

   The server received a BYE or CANCEL request with a Call-ID (Section
   6.12) value it does not recognize. (A server simply discards an ACK
   with an invalid Call-ID.)

7.4.16 482 Loop Detected

   The server received a request with a Via (Section 6.40) path
   containing itself.

7.4.17 483 Too Many Hops

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   The server received a request that contains more Via entries (hops)
   (Section 6.40) than allowed by the Max-Forwards (Section 6.23) header

7.4.18 484 Address Incomplete

   The server received a request with a To (Section 6.37) address or
   Request-URI that was incomplete. Additional information should be

        This status code allows overlapped dialing. With overlapped
        dialing, the client does not know the length of the dialing
        string. It sends strings of increasing lengths, prompting
        the user for more input, until it no longer receives a 484
        status response.

7.4.19 485 Ambiguous

   The callee address provided in the request was ambiguous. The
   response MAY contain a listing of possible unambiguous addresses in
   Location headers.

   Revealing alternatives may infringe on privacy concerns of the user
   or the organization. It MUST be possible to configure a server to
   respond with status 404 (Not Found) or to suppress the listing of
   possible choices if the request address was ambiguous.

   Example response to a request with the URL lee@example.com :

   485 Ambiguous SIP/2.0
   Location: sip:carol.lee@example.com (Carol Lee)
   Location: sip:p.lee@example.com (Ping Lee)
   Location: sip:lee.foote@example.com (Lee M. Foote)

        Some email and voice mail systems provide this
        functionality. A status code separate from 3xx is used
        since the semantics are different: for 300, it is assumed
        that the same person or service will be reached by the
        choices provided. While an automated choice or sequential
        search makes sense for a 3xx response, user intervention is
        required for a 485 response.

7.5 Server Failure 5xx

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   5xx responses are failure responses given when a server itself has
   erred. They are not definitive failures, and MUST NOT terminate a
   search if other possible locations remain untried.

7.5.1 500 Server Internal Error

   The server encountered an unexpected condition that prevented it from
   fulfilling the request.

7.5.2 501 Not Implemented

   The server does not support the functionality required to fulfill the
   request. This is the appropriate response when the server does not
   recognize the request method and is not capable of supporting it for
   any user.

7.5.3 502 Bad Gateway

   The server, while acting as a gateway or proxy, received an invalid
   response from the downstream server it accessed in attempting to
   fulfill the request.

7.5.4 503 Service Unavailable

   The server is currently unable to handle the request due to a
   temporary overloading or maintenance of the server. The implication
   is that this is a temporary condition which will be alleviated after
   some delay. If known, the length of the delay may be indicated in a
   Retry-After header. If no Retry-After is given, the client MUST
   handle the response as it would for a 500 response.

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

7.5.5 504 Gateway Timeout

   The server, while acting as a gateway, did not receive a timely
   response from the server (e.g., a location server) it accessed in
   attempting to complete the request.

7.5.6 505 Version Not Supported

   The server does not support, or refuses to support, the SIP protocol
   version that was used in the request message. The server is
   indicating that it is unable or unwilling to complete the request
   using the same major version as the client, other than with this
   error message. The response SHOULD contain an entity describing why

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   that version is not supported and what other protocols are supported
   by that server.

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 603 (Decline) instead.

7.6.2 603 Decline

   The callee's machine was successfully contacted but the user
   explicitly does not wish to or cannot participate. The response may
   indicate a better time to call in the Retry-After header.

7.6.3 604 Does Not Exist Anywhere

   The server has authoritative information that the user indicated in
   the To request field does not exist anywhere. Searching for the user
   elsewhere will not yield any results.

7.6.4 606 Not Acceptable

   The user's agent was contacted successfully but some aspects of the
   session description such as the requested media, bandwidth, or
   addressing style were not acceptable.

   A 606 (Not Acceptable) response means that the user wishes to
   communicate, but cannot adequately support the session described. The
   606 (Not Acceptable) response MAY contain a list of reasons in a
   Warning header or headers describing why the session described cannot
   be supported. Reasons are listed in Section 6.41.  It is hoped that
   negotiation will not frequently be needed, and when a new user is
   being invited to join an already existing conference, negotiation may
   not be possible. It is up to the invitation initiator to decide
   whether or not to act on a 606 (Not Acceptable) response.

8 SIP Message Body

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8.1 Body Inclusion

   For a request message, the presence of a body is signaled by the
   inclusion of a Content-Length header. Only ACK, INVITE, OPTIONS and
   REGISTER requests may contain message bodies. For ACK, INVITE and
   OPTIONS, the message body is always a session description. The use of
   message bodies for REGISTER requests is for further study.

   For response messages, whether or not a body is included is dependent
   on both the request method and the response message's response code.
   All responses MAY include a body, although it may be of zero length.
   Message bodies for 1xx responses contain advisory information about
   the progress of the request. 2xx responses contain session
   descriptions.  In 3xx respones, the message body MAY contain the
   description of alternative destinations or services, as described in
   Section 7.3.  For responses with status 400 or greater, the message
   body MAY contain additional, human-readable information about the
   reasons for failure. It is RECOMMENDED that information in 1xx and
   300 and greater responses be of type text/plain or text/html

8.2 Message Body Type

   The Internet media type of the message body MUST be given by the
   Content-Type header field, If the body has undergone any encoding
   (such as compression) then this MUST be indicated by the Content-
   Encoding header field, otherwise Content-Encoding MUST be omitted. If
   applicable, the character set of the message body is indicated as
   part of the Content-Type header-field value.

8.3 Message Body Length

   The body length in bytes SHOULD be given by the Content-Length header
   field. Section 6.22 describes the behavior in detail.

   The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
   (Note: The chunked encoding modifies the body of a message in order
   to transfer it as a series of chunks, each with its own size

9 Compact Form

   When SIP is carried over UDP with authentication and a complex
   session description, it may be possible that the size of a request or
   response is larger than the MTU. To 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 header field abbreviations are allowed.

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   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 15.2 could also be written:

     INVITE sip:schooler@vlsi.caltech.edu SIP/2.0
     CSeq: 4711 INVITE

     o=user1 53655765 2353687637 IN IP4
     s=Mbone Audio
     i=Discussion of Mbone Engineering Issues
     c=IN IP4
     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.

10 Behavior of SIP Clients and Servers

10.1 General Remarks

   SIP is defined so it can use either UDP (unicast or multicast) or TCP
   as a transport protocol; it provides its own reliability mechanism.

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10.1.1 Requests

   Servers ignore isomorphic requests, but retransmit the appropriate
   response. (SIP requests are said to be idempotent , i.e., receiving
   more than one copy of a request does not change the server state.)

   After receiving a CANCEL request from an upstream client, a stateful
   proxy server MAY send a CANCEL on all branches where it has not yet
   received a final response.

   When a user agent receives a request, it checks the Call-ID against
   those of in-progress calls. If the Call-ID was found, it compares the
   tag value of To with the user's tag and rejects the request if the
   two do not match. If the From header, including any tag value,
   matches the value for an existing call leg, the server compares the
   CSeq header value. If less than or equal to the current sequence
   number, the request is a retransmission. Otherwise, it is a new
   request. If the From header does not match an existing call leg, a
   new call leg is created.

   If the Call-ID was not found, a new call leg is created, with entries
   for the To, From and Call-ID headers.  In this case, the To header
   should not have contained a tag.  The server returns a response
   containing the same To value, but with a unique tag added. The tag
   MAY be omitted if the To refers to a fully qualified host name.

10.1.2 Responses

   A server MAY issue one or more provisional responses at any time
   before sending a final response. If a stateful proxy, user agent
   server, redirect server or registrar cannot respond to a request with
   a final response within 200 ms, it MUST issue a provisional (1xx)
   response as soon as possible. Stateless proxies MUST NOT issue
   provisional responses on their own.

   Responses are mapped to requests by the matching To, From, Call-ID,
   CSeq headers and the branch parameter of the first Via header.
   Responses terminate request retransmissions even if they have Via
   headers that cause them to be delivered to an upstream client.

   A stateful proxy may receive a response that it does not have state
   for, that is, where it has no a record of an isomorphic request. If
   the Via header field indicates that the upstream server used TCP, the
   proxy actively opens a TCP connection to that address. Thus, proxies
   have to be prepared to receive responses on the incoming side of
   passive TCP connections, even though most responses will arrive on
   the incoming side of an active connection. (An active connection is a
   TCP connection initiated by the proxy, a passive connection is one

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   accepted by the proxy, but initiated by another entity.)

   100 responses are not forwarded, other 1xx responses MAY be
   forwarded, possibly after the server eliminates responses with status
   codes that had already been sent earlier. 2xx responses are forwarded
   according to the Via header. Once a stateful proxy has received a 2xx
   response, it MUST NOT forward non-2xx final responses.  Responses
   with status 300 and higher are retransmitted by each stateful proxy
   until the next upstream proxy sends an ACK (see below for timing
   details) or CANCEL.

   A stateful proxy can remove state for a call attempt and close any
   connections 20 seconds after receiving the first final response.

        The 20 second window is given by the maximum retransmission
        duration of 200 responses (10 times T4), in case the ACK is
        lost somewhere on the way to the called user agent or the
        next stateful proxy.

10.2 Source Addresses, Destination Addresses and Connections

10.2.1 Unicast UDP

   Responses are returned to the address listed in the Via header field
   (Section 6.40), not the source address of the request.

10.2.2 Multicast UDP

   Requests may be multicast; multicast requests likely feature a host-
   independent Request-URI. Multicast requests SHOULD have a time-to-
   live value of no greater than one, i.e., be restricted to the local

   A client receiving a multicast query does not have to check whether
   the host part of the Request-URI matches its own host or domain name.
   If the request was received via multicast, the response is also
   returned via multicast. Responses to multicast requests are multicast
   with the same TTL as the request, where the TTL is derived from the
   ttl parameter in the Via header (Section 6.40).

   To avoid response implosion, servers MUST NOT answer multicast
   requests with a status code other than 2xx or 6xx. Servers only
   return 6xx responses if the To represents a single individual rather
   than a group of people. The server delays its response by a random
   interval between zero and one second. Servers MAY suppress responses
   if they hear a lower-numbered or 6xx response from another group
   member prior to sending. Servers do not respond to CANCEL requests

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   received via multicast to avoid request implosion. A proxy or UAC
   SHOULD send a CANCEL on receiving the first 2xx or 6xx response to a
   multicast request.

        Server response suppression is a MAY since it requires a
        server to violate some basic message processing rules. Lets
        say A sends a multicast request, and it is received by B,C,
        and D. B sends a 200 response. The topmost Via field in the
        response will contain the address of A. C will also receive
        this response, and could use it to suppress its own
        response. However, C would normally not examine this
        response, as the topmost Via is not its own. Normally, a
        response received with an incorrect topmost Via MUST be
        dropped, but not in this case. To distinguish this packet
        from a misrouted or multicast looped packet is fairly
        complex, and for this reason the procedure is a MAY. The
        CANCEL, instead, provides a simpler and more standard way
        to perform response suppression. It is for this reason that
        the use of CANCEL here is a SHOULD

10.3 TCP

   A single TCP connection can serve one or more SIP transactions. A
   transaction contains zero or more provisional responses followed by
   one or more final responses. (Typically, transactions contain exactly
   one final response, but there are exceptional circumstances, where,
   for example, multiple 200 responses may be generated.)

   The client MAY close the connection at any time, but SHOULD keep the
   connection open at least until the first final response arrives.  The
   server SHOULD NOT close the TCP connection until it has sent its
   final response, at which point it MAY close the TCP connection if it
   wishes to. However, normally it is the client's responsibility to
   close the connection.

   If the server leaves the connection open, and if the client so
   desires it may re-use the connection for further SIP requests or for
   requests from the same family of protocols (such as HTTP or stream
   control commands).

   If a client closes a connection or the connection is reset (e.g.,
   because the client has crashed and rebooted), the server treats this
   as equivalent to having received a CANCEL request.

   If a server needs to return a response to a client and no longer has
   a connection open to that client, it MAY open a connection to the
   address listed in the Via header. Thus, a proxy or user agent MUST be
   prepared to receive both requests and responses on a "passive"

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10.4 Reliability for BYE, CANCEL, OPTIONS, REGISTER Requests

10.4.1 UDP

   A SIP client using UDP SHOULD retransmit a BYE, CANCEL, OPTIONS, or
   REGISTER request periodically with timer T1 until it receives a
   response, or until it has reached a set limit on the number of
   retransmissions. If the response is provisional, the client continues
   to retransmit the request, albeit less frequently, using timer T2.
   The default values of timer T1 and T2 are 1 and 5 seconds,
   respectively. The default retransmit limit is 20 times. 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
   100 seconds to avoid having to, for example, contact the user or
   location server again upon receiving a retransmission.

   Each server in a proxy chain generates its own final response to a
   CANCEL request.  The server responds immediately upon receipt of the
   CANCEL request rather than not waiting until it has received final
   responses from the CANCEL requests it generates.

   BYE and OPTIONS final responses are generated by redirect and user
   agent servers; REGISTER final responses are generated by registrars.
   Note that responses to these commands are not acknowledged via ACK.

        The value of the initial retransmission timer is smaller
        than that that for TCP since it is expected that network
        paths suitable for interactive communications have round-
        trip times smaller than 1 second. To avoid flooding the
        network with packets every second even if the destination
        network is unreachable, the retransmission count has to be
        bounded. Given that most transactions should consist of one
        request and a few responses, round-trip time estimation is
        not likely to be very useful. If RTT estimation is desired
        to more quickly discover a missing final response, each
        request retransmission needs to be labeled with its own
        Timestamp (Section 6.36), returned in the response. The
        server caches the result until it can be sure that the
        client will not retransmit the same request again.

10.4.2 TCP

   Clients using TCP do not need to retransmit requests.

10.5 Reliability for ACK Requests

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   The ACK request does not generate responses. It is only retransmitted
   when a response to an INVITE request arrives. This behavior is
   independent of the transport protocol. Note that the ACK request MAY
   take a different path than the original INVITE request, and even may
   cause a new TCP connection to be opened in order to send it.

10.6 Reliability for INVITE Requests

   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.

        2.   If a telephony user interface is modeled or if we need to
             interface to the PSTN, the caller's user interface will
             provide "ringback", a signal that the callee is being
             alerted. (The status response 180 (Ringing) may be used to
             initiate ringback.) Once the callee picks up, the caller
             needs to know so that it can enable the voice path and stop
             ringback. The callee's response to the invitation could get
             lost. Unless the response is transmitted reliably, the
             caller will continue to hear ringback while the callee
             assumes that the call exists.

        3.   The client has to be able to terminate an on-going request,
             e.g., because it is no longer willing to wait for the
             connection or search to succeed. The server will have to
             wait several round-trip times to interpret the lack of
             request retransmissions as the end of a call. If the call
             succeeds shortly after the caller has given up, the callee
             will "pick up the phone" and not be "connected".

10.6.1 UDP

   For UDP, A SIP client SHOULD retransmit a SIP INVITE request
   periodically with timer T1 until it receives a response. If the
   client receives no response, it ceases retransmission after 20
   attempts.  If the response is provisional, the client continues to
   retransmit the request, albeit less frequently, using timer T3. The
   default values of timer T1 and T3 are 1 and 30 seconds, respectively.

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        The value of T3 was chosen so that for most normal phone
        calls, only one INVITE request will be issued. Typically, a
        phone switches to an answering machine or voice mail after
        about 20--22 seconds. The number of retransmissions after
        receiving a provisional response is unlimited to allow call
        queueing. Clients may limit the number of invitations sent
        for each call attempt.

   For 2xx final responses, only the user agent client generates an ACK.
   If the response contained a Location header, the ACK MAY be sent to
   the address listed in that Location header field.  If the response
   did not contain a Location header, the client uses the same To header
   field and Request-URI as for the INVITE request and sends the ACK to
   the same destination as the original INVITE request. ACKs for final
   responses other than 2xx are sent to the same server that the
   original request was sent to, using the same Request-URI as the
   original request. Note, however, that the To field in the ACK is
   copied from the response being acknowledged, not the request, and
   thus may additionally contain the tag parameter. Also note than
   unlike 2xx final responses, a proxy generates an ACK for non-2xx
   final responses.

   The server retransmits the final response at intervals of T4 (default
   value of T4 = 2 seconds) until it receives an ACK request for the
   same Call-ID and CSeq from the same From source or until it has
   retransmitted the final response 10 times. The ACK request MUST NOT
   be acknowledged to prevent a response-ACK feedback loop.

   Fig. 11 and 12 show the client and server state diagram for

        The mechanism in Sec. 10.4 would not work well for INVITE
        because of the long delays between INVITE and a final
        response. If the 200 response were to get lost, the callee
        would believe the call to exist, but the voice path would
        be dead since the caller does not know that the callee has
        picked up. Thus, the INVITE retransmission interval would
        have to be on the order of a second or two to limit the
        duration of this state confusion. Retransmitting the
        response a fixed number of times increases the probability
        of success, but at the cost of significantly higher
        processing and network load.

10.6.2 TCP

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                 |  Initial  |
                       |    -
                       |  ------
                       |  INVITE
           +------v    v
          T1     +-----------+
        ------   |  Calling  |--------+
        INVITE   +-----------+        |
           +------| |  |              |
   +----------------+  |              |
   |                   | 1xx          |  >= 200
   |                   | ---          |  ------
   |                   |  -           |   ACK
   |                   |              |
   |       +------v    v    v-----|   |
   |      T3     +-----------+   1xx  |
   |    ------   |  Ringing  |   ---  |
   |    INVITE   +-----------+    -   |
   |       +------|    |    |-----+   |
   |                   |              |
   |     2xx           |              |
   |     ---           | 2xx          |
   |     ACK           | ---          |
   |                   | ACK          |
   +----------------+  |              |
           +------v |  v              |
          xxx    +-----------+        |
          ---    | Completed |<-------+
          ACK    +-----------+


   Figure 11: State transition diagram of client for INVITE method

   A client using TCP MUST NOT retransmit requests, but uses the same
   algorithm as for UDP (Section 10.6.1) to retransmit responses until
   it receives an ACK. (An implementation can simply set T1 and T3 to
   infinity and otherwise maintain the same state diagram.)

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                 |  Initial  |<-------------+
                 +===========+              |
                       |                    |
                       |                    |
                       |  INVITE            |
                       |  ------            |
                       |   1xx              |
           +------v    v                    |
        INVITE   +-----------+              |
        ------   | Searching |              |
          1xx    +-----------+              |
           +------| |  |  +---------------->+
                    |  |                    |
          failure   |  |  callee picks up   |
          -------   |  |  ---------------   |
          >= 300    |  |       200          |
                    |  |                    | BYE
           +------v v  v    v-----|         | ---
        INVITE   +-----------+    T4        | 200
        ------   | Answered  |  ------      |
        status   +-----------+  status      |
           +------|    |  | |-----+         |
                       |  +---------------->+
                       |                    |
                       | ACK                |
                       | ---                |
                       |  -                 |
                       |                    |
           +------v    v                    |
          ACK    +-----------+              |
          ---    | Connected |              |
           -     +-----------+              |
           +------|       |                 |


   Figure 12: State transition diagram of server for INVITE method

        It is necessary to retransmit 2xx responses as their
        reliability is assured end-to-end only. If the chain of
        proxies has a UDP link in the middle, it could lose the

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        response, with no possibility of recovery. For simplicity,
        we also retransmit non-2xx responses, although that is not
        strictly necessary.

11 Behavior of SIP User Agents

   This section describes the rules for user agent client and servers
   for generating and processing requests and responses.

11.1 Caller Issues Initial INVITE Request

   When a user agent client desires to initiate a call, it formulates an
   INVITE request. The To field in the request contains the address of
   the callee. The Request-URI contains the same address. The From field
   contains the address of the caller.  If the From address can appear
   in requests generated by other user agent clients for the same call,
   the caller should insert the tag parameter in the From field. A UAC
   MAY optionally add a Location header containing an address where it
   would like to be contacted for transactions from the callee back to
   the caller.

11.2 Callee Issues Response

   When the initial INVITE request is received at the callee, the callee
   may accept, redirect, or reject the call. In all of these cases, it
   formulates a response. The response MUST copy the To, From, Call-ID,
   CSeq and Via fields from the request. Additionally, the responding
   UAS MUST add the tag parameter to the To field in the response if the
   To field in the request was not the fully-qualified hostname of the
   UAS. Since a request from a UAC may fork and arrive at multiple
   hosts, the tag parameter serves to distinguish, at the UAC, multiple
   responses from different UAS's. The UAS MAY add a Location header in
   the response. It contains an address where the callee would like to
   be contacted for subsequent transactions, including the ACK for the
   current INVITE. The UAS stores the values of the To and From field,
   including any tags. These become the local and remote addresses of
   the call leg, respectively.

11.3 Caller Receives Response to Initial Request

   Multiple responses may arrive at the UAC for a single INVITE request,
   due to a forking proxy. Each response is distinguished by the "tag"
   parameter in the To header field, and each represents a distinct call
   leg. The caller may choose to acknowledge or terminate the call with
   each responding UAS. To acknowledge, it sends an ACK request, and to
   terminate it sends a BYE request. The To field in the ACK or BYE MUST
   be the same as the To field in the 200 response, including any tag.
   The From header field MUST be the same as the From field in the 200

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   response, including any tag. The Request-URI of the ACK or BYE
   request MAY be set to whatever address was found in the Location
   field in the 200 response, if present. Alternately, a UAC may copy
   the address from the To field into the Request-URI. The UAC also
   notes the value of the To and From field in each response. For each
   call leg, the To header field becomes the remote address, and the
   From header field becomes the local address.

11.4 Caller or Callee Generate Subsequent Requests

   Once the call has been established, either the caller or callee may
   generate either INVITE or BYE requests to change or terminate the
   call. Regardless of whether the caller or callee is generating the
   new request, the fields in the request are set as follows. For the
   desired call leg, the To header field is set to the remote address,
   and the From header field is set to the local address (both including
   any tags). The Location header field MAY be different than the
   Location header field sent in a previous response or request. The
   Request-URI MAY be set to the value of the Location header field
   received in a previous request or response from the remote party, or
   to the value of the remote address.

11.5 Receiving Subsequent Requests

   When a request is received subsequently, the following checks are

        1.   If the Call-ID is new, the request is for a new call,
             regardless of the values of the To and From header fields.

        2.   If the Call-ID exists, the request is for an existing call.
             If the To, From, Call-ID, and CSeq values exactly match
             (including tags) those of any requests received previously,
             the request is a retransmission.

        3.   If there was no match to the previous step, the To and From
             fields are compared against existing call leg local and
             remote addresses. If there is a match, and the CSeq in the
             request is higher than the last CSeq received on that leg,
             the request is a new transaction for an existing call leg.

12 Behavior of SIP Proxy and Redirect Servers

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

12.1 Redirect Server

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   A redirect server does not issue any SIP requests of its own. After
   receiving a request, the server gathers the list of alternative
   locations and returns a final response of class 3xx or it refuses the
   request. For CANCEL requests, it SHOULD also return a 2xx response.
   This response ends the SIP transaction. The redirect server maintains
   transaction state for the whole SIP transaction. It is up to the
   client to detect forwarding loops between redirect servers.

12.2 User Agent Server

   User agent servers behave similarly to redirect servers, except that
   they may also accept requests and return a response of class 2xx.

12.3 Proxy Server

   This section outlines processing rules for proxy servers. A proxy
   server can either be stateful or stateless. When stateful, a proxy
   remembers the incoming request which generated outgoing requests, and
   the outgoing requests. A stateless proxy forgets all information once
   an outgoing request is generated. A forking proxy SHOULD be stateful.
   A stateful proxy MAY become stateless at any time, but SHOULD remain
   stateful until it sends a definitive response upstream.

   A stateful proxy acts as a virtual UAS/UAC. It implements the server
   state machine when receiving requests, and the client state machine
   for generating outgoing requests, with the exception of receiving a
   2xx response to an INVITE. Instead of generating an ACK, the 2xx
   response is always forwarded upstream towards the caller.
   Furthermore, ACK's for 200 responses to INVITE's are always proxied
   downstream towards the UAS, as they would be for a stateless proxy.

   A stateless proxy does not act as a virtual UAS/UAC (as this would
   require state). Rather, a stateless proxy forwards every request it
   receives downstream, and every response it receives upstream.

12.3.1 Proxying Requests

   To prevent loops, a server MUST check if its own address is already
   contained in the Via header of the incoming request.

   The To, From, Call-ID, and Location tags are copied exactly from the
   original request. The proxy SHOULD change the Request-URI to indicate
   the server where it intends to send the request.

   A proxy server always inserts a Via header field containing its own
   address into those requests that are caused by an incoming request.
   Each proxy MUST insert a "branch" parameter (Section 6.40).

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12.3.2 Proxying Responses

   A proxy only processes a response if the topmost Via field matches
   one of its addresses. A response with a non-matching top Via field
   MUST be dropped.

12.3.3 Stateless Proxy: Proxying Responses

   A stateless proxy removes its own Via field, and checks the address
   in the next Via field. If the field indicates TCP as the transport
   protocol, the proxy checks to see if it has a connection currently
   open to that address. If so, the response is sent on that connection.
   Otherwise, a new TCP connection is opened to the address and port in
   the Via field, and the response is sent there. In the case of UDP,
   the response is sent to the address listed in the "maddr" tag if
   present, otherwise to the "received" tag if present, and finally to
   the address in the "sent-by" field. Note that this implies that a UAC
   or proxy must be prepared to receive responses on the incoming side
   of a TCP connection.

   A stateless proxy MUST NOT generate its own provisional responses.

12.3.4 Stateful Proxy: Receiving Requests

   When a stateful proxy receives a request, it checks the To, From
   (including tags), Call-ID and CSeq against existing request records.
   If the tuple exists, the request is a retransmission. The provisional
   or final response sent previously is retransmitted, as per the server
   state machine. If the tuple does not exist, the request corresponds
   to a new transaction, and the request should be proxied.

   A stateful proxy server MAY generate its own provisional (1xx)

12.3.5 Stateful Proxy: Receiving ACKs

   When an ACK request is received, it must either be processed locally
   or proxied. To make this determination, the To, From, CSeq and Call-
   ID fields are compared against those in previous requests. If there
   is no match, the ACK request is proxied as if it were an INVITE
   request. If there is a match, and if the server had ever sent a 200
   response upstream, the ACK is proxied.  If the server had never sent
   any responses upstream, the ACK is also proxied. If the server had
   sent a 3xx, 4xx, 5xx or 6xx response, but no 2xx response, the ACK is
   processed locally, as it is acknowledging the response generated by
   the proxy.

12.3.6 Stateful Proxy: Receiving Responses

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   When a proxy server receives a response that has passed the Via
   checks, the proxy server checks the To (without the tag), From
   (including the tag), Call-ID and CSeq against values seen in previous
   requests. If there is no match, the response is forwarded upstream to
   the address listed in the Via field. If there is a match, the
   "branch" tag in the Via field is examined. If it matches a known
   branch identifier, the response is for the given branch, and
   processed by the virtual client for the given branch. Otherwise, the
   response is dropped.

   A stateful proxy should obey the rules in Section 12.4 to determine
   if the response should be proxied upstream. If it is to be proxied,
   the same rules for stateless proxies above are followed.

12.3.7 Stateless, Non-Forking Proxy

   Proxies in this category issue at most a single unicast request for
   each incoming SIP request, that is, they do not "fork" requests.
   However, servers may choose to always operate in a mode that allows
   issuing of several requests, as described in Section 12.4.

   The server can forward the request and any responses. It does not
   have to maintain any state for the SIP transaction. Reliability is
   assured by the next redirect or stateful proxy server in the server

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

12.4 Forking Proxy

   The server MUST respond to the request immediately with a 100
   (Trying) response.

   Successful responses to an INVITE request SHOULD contain a Location
   header so that the following ACK or BYE bypasses the proxy search
   mechanism. If the proxy requires future requests to be routed through
   it, it adds a Record-Route header to the request (Section 6.29).

   The following pseudo-code describes the behavior of a proxy server
   issuing several requests in response to an incoming INVITE request.
   The function request(r, a, b) sends a SIP request of type r to
   address a, with branch id b. await_response() waits until a response
   is received and returns the response. close(a) closes the TCP

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   connection to client with address a. response(r) sends a response to
   the client. ismulticast() returns 1 if the location is a multicast
   address and zero otherwise.  The variable timeleft indicates the
   amount of time left until the maximum response time has expired. The
   variable recurse indicates whether the server will recursively try
   addresses returned through a 3xx response. A server MAY decide to
   recursively try only certain addresses, e.g., those which are within
   the same domain as the proxy server. Thus, an initial multicast
   request may trigger additional unicast requests.

     /* request type */
     typedef enum {INVITE, ACK, BYE, OPTIONS, CANCEL, REGISTER} Method;

     process_request(Method R, int N, address_t address[])
       struct {
         address_t address;  /* address */
         int branch;         /* branch id */
         int done;           /* has responded */
       } outgoing[];
       int done[];           /* address has responded */
       char *location[];     /* list of locations */
       int heard = 0;        /* number of sites heard from */
       int class;            /* class of status code */
       int timeleft = 120;   /* sample timeout value */
       int loc = 0;          /* number of locations */
       struct {              /* response */
         int status;         /* response: CANCEL=-1 */
         int locations;      /* number of redirect locations */
         char *location[];   /* redirect locations */
         address_t a;        /* address of respondent */
         int branch;         /* branch identifier */
       } r, best;            /* response, best response */
       int i;

       best.status = 1000;
       for (i = 0; i < N; i++) {
         request(R, address[i], i);
         outgoing[i].done = 0;
         outgoing[i].branch = i;

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

         /* If final response, mark branch as done. */

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         if (class >= 2) {
           for (i = 0; i < N; i++) {
             if (r.branch == outgoing[i].branch) {
               outgoing[i].done = 1;
         /* CANCEL: respond, fork and wait for responses */
         else if (class < 0) {
           best.status = 200;
           for (i = 0; i < N; i++) {
             request(CANCEL, address[i], outgoing[i].branch);
           best.status = -1;

         if (class == 2) {
           if (r.status < best) best = r;
         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 += r.locations;
             for (i = 0; i < r.locations; i++) {
               request(R, r.location[i]);
           } else if (!ismulticast(r.location)) {
             best = r;
           if (R == INVITE) request(ACK, r.a, r.branch);
         else if (class == 4) {
           if (R == INVITE) request(ACK, r.a, r.branch);
           if (best.status >= 400) best = r;
         else if (class == 5) {
           if (R == INVITE) request(ACK, r.a, r.branch);

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           if (best.status >= 500) best = r;
         else if (class == 6) {
           if (R == INVITE) request(ACK, r.a, r.branch);
           best = r;

       /* We haven't heard anything useful from anybody. */
       if (best.status == 1000) {
         best.status = 404;
       if (best.status/100 != 3) loc = 0;

   Responses are processed as follows. The process completes (and state
   can be freed) when all requests have been answered by final status
   responses (for unicast) or 60 seconds have elapsed (for multicast). A
   proxy MAY send a CANCEL to all branches and return a 408 (Timeout) to
   the client after 60 seconds or more.

   1xx: The proxy MAY forward the response upstream towards the client.

   2xx: The proxy MUST forward the response upstream towards the client,
        without sending an ACK downstream. After receiving a 2xx, the
        server MAY terminate all other pending requests by sending a
        CANCEL request and closing the TCP connection, if applicable.
        (Terminating pending requests is advisable as searches consume
        resources. Also, INVITE requests may "ring" on a number of
        workstations if the callee is currently logged in more than

   3xx: The proxy MUST send an ACK and MAY recurse on the listed
        Location addresses. Otherwise, the lowest-numbered response is
        returned if there were no 2xx responses.

        Location lists are not merged as that would prevent
        forwarding of authenticated responses. Also, some responses
        may have message bodies, so that merging is not feasible.

   4xx, 5xx: The proxy MUST send an ACK and remember the response if it
        has a lower status code than any previous 4xx and 5xx responses.
        On completion, the lowest-numbered response is returned if there
        were no 2xx or 3xx responses.

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   6xx: The proxy MUST forward the response to the client and send an
        ACK. Other pending requests MAY be terminated with CANCEL as
        described for 2xx responses.

   A proxy server forwards any response for Call-IDs for which it does
   not have a pending transaction according to the response's Via
   header. User agent servers respond to BYE requests with an unknown
   Call-ID with status code 481 (Invalid Call-ID).  ACK requests are
   silently dropped.

   Special considerations apply for choosing forwarding destinations for
   ACK and BYE requests. In most cases, these requests will bypass
   proxies and reach the desired party directly, keeping proxies from
   having to make forwarding decisions.

   A proxy MAY maintain call state for a period of its choosing. If a
   proxy still has list of destinations that it forwarded the last
   INVITE to, it SHOULD direct ACK requests only to those downstream

13 Security Considerations

13.1 Confidentiality and Privacy: Encryption

13.1.1 End-to-End Encryption

   SIP requests and responses can contain sensitive information about
   the communication patterns and communication content of individuals
   and thus should be protected against eavesdropping. The SIP message
   body may also contain encryption keys for the session itself.

   SIP supports three complementary forms of encryption to protect

        o End-to-end encryption of the SIP message body and certain
          sensitive header fields;

        o hop-by-hop encryption to prevent eavesdropping that tracks who
          is calling whom;

        o hop-by-hop encryption of Via fields to hide the route a
          request has taken.

   Not all of the SIP request or response can be encrypted end-to-end
   because header fields such as To and Via need to be visible to
   proxies so that the SIP request can be routed correctly.  Hop-by-hop
   encryption encrypts the entire SIP request or response on the wire
   (the request may already have been end-to-end encrypted) so that

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   packet sniffers or other eavesdroppers cannot see who is calling
   whom.  Note that proxies can still see who is calling whom, and this
   information may also be deducible by performing a network traffic
   analysis, so this provides a very limited but still worthwhile degree
   of protection.

   SIP Via fields are used to route a response back along the path taken
   by the request and to prevent infinite request loops. However, the
   information given by them may also provide useful information to an
   attacker. Section 6.21 describes how a sender can request that Via
   fields be encrypted by cooperating proxies without compromising the
   purpose of the Via field.

   End-to-end encryption relies on keys shared by the two user agents
   involved in the request. Typically, the message is sent encrypted
   with the public key of the recipient, so that only that recipient can
   read the message. SIP does not limit the security mechanisms that may
   be used, but all implementations SHOULD support PGP-based encryption.

   A SIP request (or response) is end-to-end encrypted by splitting the
   message to be sent into a part to be encrypted and a short header
   that will remain in the clear. Some parts of the SIP message, namely
   the request line, the response line and certain header fields marked
   with "n" in the "enc." column in Table 4 need to be read and returned
   by proxies and thus MUST NOT be encrypted end-to-end. Possibly
   sensitive information that needs to be made available as plaintext
   include destination address (To) and the forwarding path (Via) of the
   call. The Authorization header MUST remain in the clear if it
   contains a digital signature as the signature is generated after
   encryption, but MAY be encrypted if it contains "basic" or "digest"
   authentication. The From header field SHOULD normally remain in the
   clear, but MAY be encrypted if required, in which case some proxies
   MAY return a 401 (Unauthorized) status if they require a From field.

   Other header fields MAY be encrypted or MAY travel in the clear as
   desired by the sender. The Subject, Allow, Call-ID, and Content-Type
   header fields will typically be encrypted. The Accept, Accept-
   Language, Date, Expires, Priority, Require, Cseq, and Timestamp
   header fields will remain in the clear.

   All fields that will remain in the clear MUST precede those that will
   be encrypted. The message is encrypted starting with the first
   character of the first header field that will be encrypted and
   continuing through to the end of the message body. If no header
   fields are to be encrypted, encrypting starts with the second CRLF
   pair after the last header field, as shown below. Carriage return and
   line feed characters have been made visible as "$", and the encrypted
   part of the message is outlined.

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     INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP$
     To: T. A. Watson <sip:watson@bell-telephone.com>$
     From: A. Bell <sip:a.g.bell@bell-telephone.com>$
     Encryption: PGP version=5.0$
     Content-Length: 224$
     CSeq: 488$
   * Call-ID: 187602141351@worcester.bell-telephone.com$ *
   * Subject: Mr. Watson, come here.$                    *
   * Content-Type: application/sdp$                      *
   * $                                                   *
   * v=0$                                                *
   * o=bell 53655765 2353687637 IN IP4$        *
   * c=IN IP4$                            *
   * m=audio 3456 RTP/AVP 0 3 4 5$                       *

   An Encryption header field MUST be added to indicate the encryption
   mechanism used. A Content-Length field is added that indicates the
   length of the encrypted body. The encrypted body is preceded by a
   blank line as a normal SIP message body would be.

   Upon receipt by the called user agent possessing the correct
   decryption key, the message body as indicated by the Content-Length
   field is decrypted, and the now-decrypted body is appended to the
   clear-text header fields. There is no need for an additional
   Content-Length header field within the encrypted body because the
   length of the actual message body is unambiguous after decryption.

   Had no SIP header fields required encryption, the message would have
   been as below. Note that the encrypted body must then include a blank
   line (start with CRLF) to disambiguate between any possible SIP
   header fields that might have been present and the SIP message body.

     INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP$
     To: T. A. Watson <sip:watson@bell-telephone.com>$
     From: A. Bell <a.g.bell@bell-telephone.com>$
     Encryption: PGP version=5.0$
     Content-Type: application/sdp$
     Content-Length: 107$

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   * $                                             *
   * v=0$                                          *
   * o=bell 53655765 2353687637 IN IP4$  *
   * c=IN IP4$                      *
   * m=audio 3456 RTP/AVP 0 3 4 5$                 *

13.1.2 Privacy of SIP Responses

   SIP requests may be sent securely using end-to-end encryption and
   authentication to a called user agent that sends an insecure
   response.  This is allowed by the SIP security model, but is not a
   good idea.  However, unless the correct behaviour is explicit, it
   would not always be possible for the called user agent to infer what
   a reasonable behaviour was. Thus when end-to-end encryption is used
   by the request originator, the encryption key to be used for the
   response SHOULD be specified in the request. If this were not done,
   it might be possible for the called user agent to incorrectly infer
   an appropriate key to use in the response. Thus, to prevent key-
   guessing becoming an acceptable strategy, we specify that a called
   user agent receiving a request that does not specify a key to be used
   for the response SHOULD send that response unencrypted.

   Any SIP header fields that were encrypted in a request should also be
   encrypted in an encrypted response. Location response fields MAY be
   encrypted if the information they contain is sensitive, or MAY be
   left in the clear to permit proxies more scope for localized

13.1.3 Encryption by Proxies

   Normally, proxies are not allowed to alter end-to-end header fields
   and message bodies. Proxies MAY, however, encrypt an unsigned request
   or response with the key of the call recipient.

        Proxies may need to encrypt a SIP request if the end system
        cannot perform encryption or to enforce organizational
        security policies.

13.1.4 Hop-by-Hop Encryption

   It is RECOMMENDED that SIP requests and responses are also protected
   by security mechanisms at the transport and network layer.

13.1.5 Via field encryption

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   When Via fields are to be hidden, a proxy that receives a request
   containing an appropriate "Hide: hop" header field (as specified in
   section 6.21) SHOULD encrypt the header field. As only the proxy that
   encrypts the field will decrypt it, the algorithm chosen is entirely
   up to the proxy implementor. Two methods satisfy these requirements:

        o The server keeps a cache of Via fields and the associated To
          field, and replaces the Via field with an index into the
          cache. On the reverse path, take the Via field from the cache
          rather than the message.

        This is insufficient to prevent message looping, and so an
        additional ID must be added so that the proxy can detect loops.
        This should not normally be the address of the proxy as the goal
        is to hide the route, so instead a sufficiently large random
        number should be used by the proxy and maintained in the cache.
        Obtaining sufficiently much randomness to give sufficient
        protection against clashes may be hard.

        It may also be possible for replies to get directed to the wrong
        originator if the cache entry gets reused, so great care must be
        taken to ensure this does not happen.

        o The server may use a secret key to encrypt the Via field, a
          timestamp and an appropriate checksum in any such message with
          the same secret key. The checksum is needed to detect whether
          successful decoding has occurred, and the timestamp is
          required to prevent possible response attacks and to ensure
          that no two requests from the same previous hop have the same
          encrypted Via field.

   The latter is the preferred solution, although proxy developers may
   devise other methods that might also satisfy the requirements.

13.2 Message Integrity and Access Control: Authentication

   An active attacker may be able to modify and replay SIP requests and
   responses unless protective measures are taken. The same
   cryptographic measures that are used to ensure the authenticity of
   the SIP message also serve to authenticate the originator of the
   message. However, the "basic" and "digest" authentication mechanism
   offer authentication only, without message integrity.

   Transport-layer or network-layer authentication may be used for hop-
   by-hop authentication. SIP also extends the HTTP WWW-Authenticate
   (Section 6.42) and Authorization (Section 6.11) header and their
   Proxy counterparts to include cryptographically strong signatures.
   SIP also supports the HTTP "basic" and "digest" schemes and other

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   HTTP authentication schemes to be defined that offer a rudimentary
   mechanism of ascertaining the identity of the caller.

        Since SIP requests are often sent to parties with which no
        prior communication relationship has existed, we do not
        specify authentication based on shared secrets.

   SIP requests may be authenticated using the Authorization header
   field to include a digital signature of certain header fields, the
   request method and version number and the payload, none of which are
   modified between client and called user agent. The Authorization
   header field may be used in requests to authenticate the request
   originator end-to-end to proxies and the called user agent, and in
   responses to authenticate the called user agent or proxies returning
   their own failure codes. It does not provide hop-by-hop
   authentication, which may be provided if required using the IPSEC
   Authentication Header.

   SIP does not dictate which digital signature scheme is used for
   authentication, but does define how to provide authentication using
   PGP in Section 14. As indicated above, SIP may also use "basic" and
   "digest" authentication and other authentication mechanisms defined
   for HTTP. Note that "basic" authentication has severe security
   limitations. The following does not apply to these schemes.

   To cryptographically sign a SIP request, the order of the SIP header
   fields is important.  When an Authorization header field is present,
   it indicates that all header fields following the Authorization
   header field have been included in the signature. Therefore, hop-by-
   hop header fields which MUST or SHOULD be modified by proxies MUST
   precede the Authorization header as they will generally be modified
   or added-to by proxy servers. Hop-by-hop header fields which MAY be
   modified by a proxy MAY appear before or after the Authorization
   header. When the appear before, the MAY be modified by a proxy. When
   they appear after, they MUST NOT be modified by a proxy. To sign a
   request, a client constructs a message from the request method (in
   upper case) followed, without LWS, by the SIP version number,
   followed, again without LWS, by the request headers to be signed and
   the message body. The message thus constructed is then signed.

   For example, if the SIP request is to be:

   INVITE sip:watson@boston.bell-telephone.com SIP/2.0
   Via: SIP/2.0/UDP
   Authorization: PGP version=5.0, signature=...
   From: A. Bell <sip:a.g.bell@bell-telephone.com>
   To: T. A. Watson <sip:watson@bell-telephone.com>

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   Call-ID: 187602141351@worcester.bell-telephone.com
   Subject: Mr. Watson, come here.
   Content-Type: application/sdp
   Content-Length: ...

   o=bell 53655765 2353687637 IN IP4
   c=IN IP4
   m=audio 3456 RTP/AVP 0 3 4 5

   Then the data block that is signed is:

   INVITESIP/2.0From: A. Bell <sip:a.g.bell@bell-telephone.com>
   To: T. A. Watson <sip:watson@bell-telephone.com>
   Call-ID: 187602141351@worcester.bell-telephone.com
   Subject: Mr. Watson, come here.
   Content-Type: application/sdp
   Content-Length: ...

   o=bell 53655765 2353687637 IN IP4
   c=IN IP4
   m=audio 3456 RTP/AVP 0 3 4 5

   Note that if a message is encrypted and authenticated using a digital
   signature, when the message is generated encryption is performed
   before the digital signature is generated. On receipt, the digital
   signature is checked before decryption.

   A client MAY require that a server sign its response by including a
   Require: org.ietf.sip.signed-response request header field. The
   client indicates the desired authentication method via the WWW-
   Authenticate header.

   The correct behaviour in handling unauthenticated responses to a
   request that requires authenticated responses is described in section

13.2.1 Trusting responses

   It may be possible for an eavesdropper to listen to requests and to
   inject unauthenticated responses that would terminate, redirect or
   otherwise interfere with a call. (Even encrypted requests contain
   enough information to fake a response.)

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   Client should be particularly careful with 3xx redirection responses.
   Thus a client receiving, for example, a 301 (Moved Permanently) which
   was not authenticated when the public key of the called user agent is
   known to the client, and authentication was requested in the request
   SHOULD be treated as suspicious. The correct behaviour in such a case
   would be for the called-user to form a dated response containing the
   Location field to be used, to sign it, and give this signed stub
   response to the proxy that will provide the redirection. Thus the
   response can be authenticated correctly. There may be circumstances
   where such unauthenticated responses are unavoidable, but a client
   SHOULD NOT automatically redirect such a request to the new location
   without alerting the user to the authentication failure before doing

   Another problem might be responses such as 6xx failure responses
   which would simply terminate a search, or "4xx" and "5xx" response

   If TCP is being used, a proxy SHOULD treat 4xx and 5xx responses as
   valid, as they will not terminate a search. However, 6xx responses
   from a rogue proxy may terminate a search incorrectly. 6xx responses
   SHOULD be authenticated if requested by the client, and failure to do
   so SHOULD cause such a client to ignore the 6xx response and continue
   a search.

   With UDP, the same problem with 6xx responses exists, but also an
   active eavesdropper can generate 4xx and 5xx responses that might
   cause a proxy or client to believe a failure occurred when in fact it
   did not. Typically 4xx and 5xx responses will not be signed by the
   called user agent, and so there is no simple way to detect these
   rogue responses. This problem is best prevented by using hop-by-hop
   encryption of the SIP request, which removes any additional problems
   that UDP might have over TCP.

   These attacks are prevented by having the client require response
   authentication and dropping unauthenticated responses. A server user
   agent that cannot perform response authentication responds using the
   normal Require response of 420 (Bad Extension).

13.3 Callee 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.

13.4 Known Security Problems

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   With either TCP or UDP, a denial of service attack exists by a rogue
   proxy sending 6xx responses. Although a client SHOULD choose to
   ignore such responses if it requested authentication, a proxy cannot
   do so. It is obliged to forward the 6xx response back to the client.
   The client can then ignore the response, but if it repeats the
   request it will probably reach the same rogue proxy again, and the
   process will repeat.

14 SIP Security Using PGP

14.1 PGP Authentication Scheme

   The "pgp" authentication scheme is based on the model that the client
   must authenticate itself with a request signed with the client's
   private key. The server can then ascertain the origin of the request
   if it has access to the public key, preferably signed by a trusted
   third party.

14.1.1 The WWW-Authenticate Response Header

        WWW-Authenticate    =    "WWW-Authenticate" ":" "pgp" pgp-challenge
        pgp-challenge       =    * (";" pgp-params )
        pgp-params          =    realm | pgp-version | pgp-algorithm
        realm               =    "realm" "=" realm-value
        realm-value         =    quoted-string
        pgp-version         =    "version" "=" digit *( "." digit ) *letter
        pgp-algorithm       =    "algorithm" "=" ( "md5" | "sha1" | token )

   The meanings of the values of the parameters used above are as

   realm: A string to be displayed to users so they know which identity
        to use. This string should contain at least the name of the host
        performing the authentication and might additionally indicate
        the collection of users who might have access. An example might
        be " Users with call-out privileges ".

   pgp-algorithm: A string indicating the PGP message integrity check
        (MIC) to be used to produce the signature. If this not present
        it is assumed to be "md5". The currently defined values are
        "md5" for the MD5 checksum, and "sha1" for the SHA.1 algorithm.

   pgp-version: The version of PGP that the client MUST use. Common
        values are "2.6.2" and "5.0". The default is 5.0.

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   WWW-Authenticate: pgp ;version="5.0"
     ;realm="Your Startrek identity, please" ;algorithm="md5"

14.1.2 The Authorization Request Header

   The client is expected to retry the request, passing an Authorization
   header line, which is defined as follows.

        Authorization  ___   "Authorization" ":" "pgp" *( ";" pgp-response )
        pgp-response   ___   realm | pgp-version | pgp-signature | signed-by
        pgp-signature  ___   "signature" "=" quoted-string
        signed-by      ___   "signed-by" "=" URI

   The signature MUST correspond to the From header of the request
   unless the signed-by parameter is provided.

   pgp-signature: The PGP ASCII-armored signature, as it appears between
        the "BEGIN PGP MESSAGE" and "END PGP MESSAGE" delimiters,
        without the version indication. The signature is included
        without any linebreaks.

   The signature is computed across the request method, request version
   and header fields following the Authorization header and the message
   body, in the same order as they appear in the message. The request
   method and version are prepended to the header fields without any
   white space. The signature is computed across the headers as sent,
   including any folding and the terminating CRLF. The CRLF following
   the Authorization header is NOT included in the signature.

        Using the ASCII-armored version is about 25% less space-
        efficient than including the binary signature, but it is
        significantly easier for the receiver to piece together.
        Versions of the PGP program always include the full
        (compressed) signed text in their output unless ASCII-
        armored mode ( -sta ) is specified.  Typical signatures are
        about 200 bytes long. -- The PGP signature mechanism allows
        the client to simply pass the request to an external PGP
        program. This relies on the requirement that proxy servers
        are not allowed to reorder or change header fields.

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   realm: The realm is copied from the corresponding WWW-Authenticate
        header field parameter.

   signed-by: If and only if the request was not signed by the entity
        listed in the From header, the signed-by header indicates the
        name of the signing entity, expressed as a URI.

   Receivers of signed SIP messages SHOULD discard any end-to-end header
   fields above the Authorization header, as they may have been
   maliciously added en route by a proxy.


   Authorization: pgp version="5.0"
     ;realm="Your Startrek identity, please"

14.2 PGP Encryption Scheme

   The PGP encryption scheme uses the following syntax:

        Encryption    ___   "Encryption" ":" "pgp" pgp-eparams
        pgp-eparams   ___   1# ( pgp-version | pgp-encoding )
        pgp-encoding  ___   "encoding" "=" "ascii" | token

   encoding: Describes the encoding or "armor" used by PGP. The value
        "ascii" refers to the standard PGP ASCII armor, without the
        lines containing "BEGIN PGP MESSAGE" and "END PGP MESSAGE" and
        without the version identifier. By default, the encrypted part
        is included as binary.


   Encryption: pgp version="2.6.2", encoding="ascii"

14.3 Response-Key Header Field for PGP

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        Response-Key  ___   "Response-Key" ":" "pgp" pgp-eparams
        pgp-eparams   ___   1# ( pgp-version | pgp-encoding | pgp-key)
        pgp-key       ___   "key" "=" quoted-string

   If ASCII encoding has been requested via the encoding parameter, the
   key parameter contains the user's public key as extracted with the
   "pgp -kxa user ".


   Response-Key: pgp version="2.6.2", encoding="ascii",

15 Examples

   In the following examples, we often omit the message body and the
   corresponding Content-Length and Content-Type headers for brevity.

15.1 Registration

   A user at host saturn.bell-tel.com registers on start-up, via
   multicast, with the local SIP server named sip.bell-tel.com the
   example, the user agent on saturn expects to receive SIP requests on
   UDP port 3890.

   C->S: REGISTER sip:sip.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 4236500900@saturn.bell-tel.com
         CSeq: 1 REGISTER
         Location: sip:saturn.bell-tel.com:3890;transport=udp
         Expires: 7200

   The registration expires after two hours. Any future invitations for
   watson@bell-tel.com arriving at sip.bell-tel.com will now be
   redirected to watson@saturn.bell-tel.com , UDP port 3890.

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   If Watson wants to be reached elsewhere, say, an on-line service he
   uses while traveling, he updates his reservation after first
   cancelling any existing locations:

   C->S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 1345441868@saturn.bell-tel.com
         CSeq: 1 REGISTER
         Location: *
         Expires: 0

   C->S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 81791800@saturn.bell-tel.com
         CSeq: 1 REGISTER
         Location: sip:tawatson@example.com

   Now, the server will forward any request for Watson to the server at
   example.com , using the Request-URI tawatson@example.com

   It is possible to use third-party registration. Here, the secretary
   jon.diligent registers his boss:

   C->S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:jon.diligent@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 1212759220@saturn.bell-tel.com
         CSeq: 1 REGISTER
         Location: sip:tawatson@example.com

   The request could be send to either the registrar at bell-tel.com or
   the server at example.com example.com would proxy the request to the
   address indicated in the Request-URI. Then, Max-Forwards header could
   be used to restrict the registration to that server.

15.2 Invitation to a Multicast Conference

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

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   session. All examples use the Session Description Protocol (SDP) (RFC
   2327 [5]) as the session description format.

15.2.1 Request

   C->S: INVITE sip:schooler@vlsi.cs.caltech.edu SIP/2.0
         Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Mark Handley <sip:mjh@isi.edu>
         To: Eve Schooler <sip:schooler@caltech.edu>
         Call-ID: 2963313058@oregon.isi.edu
         CSeq: 1 INVITE
         Subject: SIP will be discussed, too
         Content-Type: application/sdp
         Content-Length: 187

         o=user1 53655765 2353687637 IN IP4
         s=Mbone Audio
         i=Discussion of Mbone Engineering Issues
         c=IN IP4
         t=0 0
         m=audio 3456 RTP/AVP 0

   The Via fields list the hosts along the path from invitation
   initiator (the last element of the list) towards the invitee. In the
   example above, the message was last multicast to the administratively
   scoped group with a ttl of 16 from the host

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

   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.

15.2.2 Response

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   The called user agent, directly or indirectly through proxy servers,
   indicates that it is alerting ("ringing") the called party:

   S->C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Mark Handley <sip:mjh@isi.edu>
         To: Eve Schooler <sip:schooler@caltech.edu;tag=9883472>
         Call-ID: 2963313058@oregon.isi.edu
         CSeq: 1 INVITE

   A sample response to the invitation is given below. The first line of
   the response states the SIP version number, that it is a 200 (OK)
   response, which means the request was successful. The Via headers are
   taken from the request, and entries are removed hop by hop as the
   response retraces the path of the request. A new authentication field
   MAY be added by the invited user's agent if required. The Call-ID is
   taken directly from the original request, along with the remaining
   fields of the request message. The original sense of From field is
   preserved (i.e., it is the session initiator).

   In addition, the 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 csvax.cs.caltech.edu;branch=8348
           ;maddr= 16;ttl=16
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Mark Handley <sip:mjh@isi.edu>
         To: Eve Schooler <sip:schooler@caltech.edu;tag=9883472>
         Call-ID: 2963313058@oregon.isi.edu
         CSeq: 1 INVITE
         Location: sip:es@jove.cs.caltech.edu

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

   C->S: ACK sip:es@jove.cs.caltech.edu SIP/2.0
         Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348

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         From: Mark Handley <sip:mjh@isi.edu>
         To: Eve Schooler <sip:schooler@caltech.edu;tag=9883472>
         Call-ID: 2963313058@oregon.isi.edu
         CSeq: 1 ACK

15.3 Two-party Call

   For two-party Internet phone calls, the response must contain a
   description of where to send the data. In the example below, Bell
   calls Watson. Bell indicates that he can receive RTP audio codings 0
   (PCMU), 3 (GSM), 4 (G.723) and 5 (DVI4).

   C->S: INVITE sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com>
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Subject: Mr. Watson, come here.
         Content-Type: application/sdp
         Content-Length: ...

         o=bell 53655765 2353687637 IN IP4
         s=Mr. Watson, come here.
         c=IN IP4
         m=audio 3456 RTP/AVP 0 3 4 5

   S->C: SIP/2.0 100 Trying
         Via: SIP/2.0/UDP
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com;tag=37462311>
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S->C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com;tag=37462311>
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

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   S->C: SIP/2.0 182 Queued, 2 callers ahead
         Via: SIP/2.0/UDP
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com;tag=37462311>
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S->C: SIP/2.0 182 Queued, 1 caller ahead
         Via: SIP/2.0/UDP
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com;tag=37462311>
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S->C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: sip:watson@bell-tel.com;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 INVITE
         Location: sip:watson@boston.bell-tel.com
         Content-Length: ...

         o=watson 4858949 4858949 IN IP4
         s=I'm on my way
         c=IN IP4
         m=audio 5004 RTP/AVP 0 3

   The example illustrates the use of informational status responses.
   Here, the reception of the call is confirmed immediately (100), then,
   possibly after some database mapping delay, the call rings (180) and
   is then queued, with periodic status updates.

   Watson can only receive PCMU and GSM. Note that Watson's list of
   codecs may or may not be a subset of the one offered by Bell, as each
   party indicates the data types it is willing to receive. Watson will
   send audio data to port 3456 at, Bell will send to
   port 5004 at

   By default, the media session is one RTP session. Watson will receive
   RTCP packets on port 5005, while Bell will receive them on port 3457.

   Since the two sides have agreed on the set of media, Watson confirms

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   the call without enclosing another session description:

   C->S: ACK sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com;tag=37462311>
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 ACK

15.4 Terminating a Call

   To terminate a call, caller or callee can send a BYE request:

   C->S: BYE sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. A. Watson <sip:watson@bell-tel.com;tag=37462311>
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 2 BYE

   If the callee wants to abort the call, it simply reverses the To and
   From fields. Note that it is unlikely that an BYE from the callee
   will traverse the same proxies as the original INVITE.

15.5 Forking Proxy

   In this example, Bell ( a.g.bell@bell-tel.com ) (C), currently seated
   at host c.bell-tel.com wants to call Watson ( t.watson@ieee.org ). At
   the time of the call, Watson is logged in at two workstations,
   watson@x.bell-tel.com (X) and watson@y.bell-tel.com (Y), and has
   registered with the IEEE proxy server (P) called sip.ieee.org
   registration for the home machine of Watson, at watson@h.bell-tel.com
   (H), as well as a permanent registration at watson@acm.org (A). For
   brevity, the examples omit the session description.

   Watson's user agent sends the invitation to the SIP server for the
   ieee.org domain:

   C->P: INVITE sip:watson@ieee.org SIP/2.0
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>

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         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 INVITE

   The SIP server at ieee.org tries the four addresses in parallel.  It
   sends the following message to the home machine:

   P->H: INVITE sip:watson@h.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=1
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 INVITE

   This request immediately yields a 404 (Not Found) response, since
   Watson is not currently logged in at home:

   H->P: SIP/2.0 404 Not Found
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=1
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org;tag=87454273>
         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 INVITE

   The proxy ACKs the response so that host H can stop retransmitting

   P->H: ACK sip:watson@h.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=1
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org;tag=37462311>
         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 ACK

   Also, P attempts to reach Watson through the ACM server:

   P->A: INVITE sip:watson@acm.org SIP/2.0

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         Via:     SIP/2.0/UDP sip.ieee.org ;branch=2
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 INVITE

   In parallel, the next attempt proceeds, with an INVITE to X and Y:

   P->X: INVITE sip:watson@x.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 INVITE

   P->Y: INVITE sip:watson@y.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=4
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 INVITE

   As it happens, both Watson at X and a colleague in the other lab at
   host Y hear the phones ringing and pick up. Both X and Y return 200s
   via the proxy to Bell.

   X->P: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP sip.ieee.org ;branch=3
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org;tag=192137601>
         Call-ID:  31415@kton.bell-tel.com
         CSeq:     1 INVITE
         Location: sip:t.watson@x.bell-tel.com

   Y->P: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP sip.ieee.org ;branch=4
         Via:      SIP/2.0/UDP c.bell-tel.com
         Location: sip:t.watson@y.bell-tel.com

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         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org;tag=35253448>
         Call-ID:  31415@kton.bell-tel.com
         CSeq:     1 INVITE

   Both responses are forwarded to Bell, using the Via information.  At
   this point, the ACM server is still searching its database. P can now
   cancel this attempt:

   P->A: CANCEL sip:watson@acm.org SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=2
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 CANCEL

   The ACM server gladly stops its neural-network database search and
   responds with a 200. The 200 will not travel any further, since P is
   the last Via stop.

   A->P: SIP/2.0 200 OK
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=2
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 CANCEL

   Bell gets the two 200 responses from X and Y in short order. Bell's
   reaction now depends on his software. He can either send an ACK to
   both if human intelligence is needed to determine who he wants to
   talk to or he can automatically reject one of the two calls. Here, he
   acknowledges both, separately and directly to the final destination:

   C->X: ACK sip:watson@x.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org;tag=192137601>
         Call-ID:  31415@c.bell-tel.com
         CSeq:     1 ACK

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   C->Y: ACK sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org;tag=35253448>
         Call-ID:  31415@c.bell-tel.com
         CSeq:     1 ACK

   After a brief discussion between the three, it becomes clear that
   Watson is at X, thus Bell sends a BYE to Y, which is replied to:

   C->Y: BYE sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org;tag=35253448>
         Call-ID:  31415@c.bell-tel.com
         CSeq:     2 BYE

   Y->C: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org;tag=35253448>
         Call-ID:  31415@c.bell-tel.com
         CSeq:     2 BYE

15.6 Redirects

   Replies with status codes 301 (Moved Permanently) or 302 (Moved
   Temporarily) specify another location using the Location field.
   Continuing our earlier example, the server at ieee.org decides to
   redirect rather than proxy the request:

   S->C: SIP/2.0 302 Moved temporarily
         Via:     SIP/2.0/UDP sip.ieee.org
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org;tag=72538263>
         Call-ID: 31415@kton.bell-tel.com
         CSeq:    1 INVITE
         Location: sip:watson@h.bell-tel.com,
                   sip:watson@acm.org, sip:watson@x.bell-tel.com,
         CSeq: 1 INVITE

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   As another example, assume Alice wants to delegate her calls to Bob
   while she is on vacation until July 29th. Any calls meant for her
   will reach Bob with Alice's To field, indicating to him what role he
   is to play. In the example below, Charlie calls Alice.

   S->C: SIP/2.0 302 Moved temporarily
         From: sip:charlie@caller.com
         To: sip:alice@anywhere.com;tag=2332462
         Call-ID: 27182@caller.com
         Location: sip:bob@anywhere.com
         Expires: Wed, 29 Jul 1998 9:00:00 GMT
         CSeq: 1 INVITE

   Charlie then sends the following request to the SIP server of the
   anywhere.com domain.

   C->S: INVITE sip:bob@anywhere.com SIP/2.0
         From: sip:charlie@caller.com
         To: sip:alice@anywhere.com
         Call-ID: 27182@caller.com
         CSeq: 2 INVITE

   In the third redirection example, we assume that all requests are
   directed through a local firewall at caller.com firewall happens to
   be overloaded and thus redirects the call from Charlie to a secondary

   S->C: SIP/2.0 302 Moved temporarily
         From: sip:charlie@caller.com
         To: sip:alice@anywhere.com;tag=273462236
         Call-ID: 27182@caller.com
         CSeq: 2 INVITE
         Location: sip:alice@anywhere.com:5080 ;maddr=secondary.caller.com

   Charlie directs the invitation to the secondary server at port 5080,
   but maintains the same Request-URI as before.

   C->S: INVITE sip:alice@anywhere.com SIP/2.0

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         From: sip:charlie@caller.com
         To: sip:alice@anywhere.com
         Call-ID: 27182@caller.com
         CSeq: 3 INVITE

15.7 Alternative Services

   An example of a 380 (Alternative Service) response is:

   S->C: SIP/2.0 380 Alternative Service
         Via: SIP/2.0/UDP
         Via: SIP/2.0/UDP
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu;tag=11223647
         Call-ID: 14142@oregon.isi.edu
         CSeq: 1 INVITE
         Location: sip:recorder@
         Content-Type: application/sdp
         Content-Length: 146

         o=mm-server 2523535 0 IN IP4
         s=Answering Machine
         i=Leave an audio message
         c=IN IP4
         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 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 sip:recorder@ SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 1732@oregon.isi.edu

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         CSeq: 1 INVITE
         Content-Type: application/sdp
         Content-Length: 146

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

   Invitation initiators MAY choose to treat a 350 (Alternative Service)
   response as a failure if they wish to do so.

15.8 Negotiation

   An example of a 606 (Not Acceptable) response is:

   S->C: SIP/2.0 606 Not Acceptable
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu;tag=7434264
         Call-ID: 14142@oregon.isi.edu
         CSeq: 1 INVITE
         Location: sip:mjh@
         Warning: 606.1 Insufficient bandwidth (only have ISDN),
           606.3 Incompatible format,
           606.4 Multicast not available
         Content-Type: application/sdp
         Content-Length: 50

         s=Lets talk
         c=IN IP4
         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 response 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 response states that DVI, PCM or LPC audio could be supported in

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   order of preference. The response also states that multicast is not
   available.  In such a case, it might be appropriate to set up a
   transcoding gateway and re-invite the user.

15.9 OPTIONS Request

   A caller Alice can use an OPTIONS request to find out the
   capabilities of a potential callee Bob, without "ringing" the
   designated address. Bob returns a description indicating that he is
   capable of receiving audio and video, with a list of supported

   C->S: OPTIONS sip:bob@example.com SIP/2.0
         From: Alice <sip:alice@anywhere.org>
         To: Bob <sip:bob@example.com>
         Call-ID: 6378@host.anywhere.org
         CSeq: 1 OPTIONS
         Accept: application/sdp

   S->C: SIP/2.0 200 OK
         From: Alice <sip:alice@anywhere.org>
         To: Bob <sip:bob@example.com;tag=376364382>
         Call-ID: 6378@host.anywhere.org
         Content-Length: 81
         Content-Type: application/sdp

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

A Minimal Implementation

A.1 Client

   All clients MUST be able to generate the INVITE and ACK requests.
   Clients MUST generate and parse the Call-ID, Content-Length,
   Content-Type, CSeq, From and To headers. Clients MUST also parse the
   Require header. A minimal implementation MUST understand SDP (RFC
   2327, [5]). It MUST be able to recognize the status code classes 1
   through 6 and act accordingly.

   The following capability sets build on top of the minimal
   implementation described in the previous paragraph:

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   Basic: A basic implementation adds support for the BYE method to
        allow the interruption of a pending call attempt. It includes a
        User-Agent header in its requests and 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 MUST understand the WWW-
        Authenticate response header.

   If a client wishes to use proxies that require caller authentication,
   it MUST be able to recognize the 407 (Proxy Authentication Required)
   status code, MUST be able to generate the Proxy-Authorization request
   header and understand the Proxy-Authenticate response header.

A.2 Server

   A minimally compliant server implementation MUST understand the
   INVITE, ACK, OPTIONS and BYE requests. A proxy server MUST also
   understand CANCEL. It MUST parse and generate, as appropriate, the
   Call-ID, Content-Length, Content-Type, CSeq, Expires, From, Max-
   Forwards, Require, To and Via headers. It MUST echo the CSeq and
   Timestamp headers in the response. It SHOULD include the Server
   header in its responses.

A.3 Header Processing

   Table 5 lists the headers that different implementations support. UAC
   refers to a user-agent client (calling user agent), UAS to a user-
   agent server (called user-agent).

B Usage of SDP

   By default, the nth media session in a unicast INVITE request will
   become a single RTP session with the nth media session in the
   response. Thus, the callee should be careful to order media

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                                   type    UAC    proxy    UAS
            Accept                  R       -       o       o
            Accept-Language         R       -       b       b
            Allow                  405      o       -       -
            Authorization           R       a       o       a
            Call-ID                 g       m       m       m
            Content-Length          g       m       m       m
            Content-Type            g       m       -       m
            CSeq                    g       m       m       m
            Encryption              g       e       -       e
            Expires                 g       -       o       o
            From                    g       m       o       m
            Location                R       -       -       -
            Location                r       r       r       -
            Max-Forwards            R       -       b       -
            Proxy-Authenticate     407      a       -       -
            Proxy-Authorization     R       -       a       -
            Proxy-Require           R       -       m       -
            Require                 R       m       -       m
            Response-Key            R       -       -       e
            Timestamp               g       o       o       m
            To                      g       m       m       m
            Unsupported             r       b       b       -
            Via                     g       m       m       m
            WWW-Authenticate       401      a       -       -

   Table 5: This table indicates which systems should be able  to  parse
   which  header  fields. Type is as in Table 4. "-" indicates the field
   is not meaningful to this system (although it might be  generated  by
   it).  "m"  indicates  the field MUST be understood. "b" indicates the
   field SHOULD be understood by a Basic implementation.  "r"  indicates
   the  field  SHOULD  be  understood if the system claims to understand
   redirection.  "a" indicates the field SHOULD  be  understood  if  the
   system  claims  to  support  authentication.  "e" indicates the field
   SHOULD be understood if the system claims to support encryption.  "o"
   indicates  support  of  the  field  is purely optional. Headers whose
   support is optional for all implementations are not shown.

   descriptions appropriately.

   It is assumed that if caller or callee include a particular media
   type, they want to both send and receive media data. If the callee
   does not want to send a particular media type, it should mark the
   media entry as recvonly receive a particular media type, it may mark
   it as sendonly wants to neither receive nor send a particular media
   type, it should set the port to zero. (RTCP ports are not needed in

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   this case.)

   The caller should include all media types that it is willing to send
   so that the receiver can provide matching media descriptions.

   The callee should set the port to zero if callee and caller only want
   to receive a media type.

C Summary of Augmented BNF

   In this specification we use the Augmented Backus-Naur Form notation
   described in RFC 2234 [21]. For quick reference, the following is a
   brief summary of the main features of this ABNF.

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

        The character with ASCII code decimal 32 (space);

        zero of more instances of  term;

        three or more instances of  term;

        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;

        a comma separated list of  term;

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

        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 of this document, and
   not defined elsewhere. Their meaning is well understood but we
   include it here for completeness.

        CR       =    %d13                                           ;  US-ASCII CR, carriage return character
        LF       =    %d10                                           ;  US-ASCII LF, line feed character
        CRLF     =    CR LF                                          ;  typically the end of a line
        SP       =    %d32                                           ;  US-ASCII SP, space character
        HT       =    %d09                                           ;  US-ASCII HT, horizontal tab character
        LWS      =    [CRLF] 1*( SP | HT )                           ;  linear whitespace
        DIGIT    =    "0" .. "9"                                     ;  a single decimal digit
        CHAR     =    <any US-ASCII character (octets 0 - 127)>
        CTL      =    <any US-ASCII control character
                      (octets 0 - 31) and DEL (127)>
        OCTET    =    <any 8-bit sequence of data>
        TEXT     =    <any OCTET except CTLs, but including LWS>

        unreserved    =    alphanum | mark
        mark          =    "-" | "_" | "." | "!" | "~" | "*" | "'"
                     |     "(" | ")"
        separators    =    "(" | ")" | "<" | ">" | "@" |
                           "," | ";" | ":" | "/" | <"> |
                           "/" | "[" | "]" | "?" | "=" |
                           "" | "" | SP | HT
        escaped       =    "%" hex hex
        hex           =    digit | "A" | "B" | "C" | "D" | "E" | "F" |
                           "a" | "b" | "c" | "d" | "e" | "f"
        alphanum      =    alpha | digit
        alpha         =    lowalpha | upalpha
        lowalpha      =    "a" | "b" | "c" | "d" | "e" | "f" | "g" | "h" | "i" |
                           "j" | "k" | "l" | "m" | "n" | "o" | "p" | "q" | "r" |
                           "s" | "t" | "u" | "v" | "w" | "x" | "y" | "z"
        upalpha       =    "A" | "B" | "C" | "D" | "E" | "F" | "G" | "H" | "I" |
                           "J" | "K" | "L" | "M" | "N" | "O" | "P" | "Q" | "R" |
                           "S" | "T" | "U" | "V" | "W" | "X" | "Y" | "Z"
        digit         =    "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" |
                           "8" | "9"
        token         =    1*<any CHAR except CTL's or separators>
        comment       =    "(" *(ctext | quoted-pair | comment) ")"
        ctext         =    <any TEXT excluding "(" and ")">

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D IANA Considerations

   Section 4.4 describes a name space and mechanism for registering SIP

   Section 6.41 describes the name space for registering SIP warn-codes.

E Changes in Version -08

   Since version -07, the following changes have been made.

        o Allow maddr to contain host names, not just IP multicast
          addresses. This is useful to allow multicast addresses of the
          form mcast.net as well as for redirecting requests to a
          different server without changing the Request-URI. Added note
          to that effect to description of 302.

        o Added references to symbolic name sip.mcast.net

        o REGISTER requests without Location are quite useful to obtain
          the current lists of registrations.

        o Added "comment" to Location header to allow responses like
          Location: alice@example.com (Alice) solution conforming to the
          format of From and To is, unfortunately, not backward
          compatible with HTTP.

        o Allow Content-Length to be optional, in line with HTTP.  This
          does not seem to complicate matters, but makes the processing
          of cgi-style output much easier, since the server doesn't have
          to store the output to compute the Content-Length.

        o Fixed table 4: CSeq is now mandatory for all requests.

        o Since To, From, and Call-ID are used as a callleg identifier
          even in proxies, encryption of From needs to be disallowed.

        o Organization is now a general header; no reason not to allow
          the callee to insert its organization.

        o Record-Route BNF changed to SIP-URL.

        o Via operation clarified.

        o Additional BNF added to appendix.

        o Tag usage descriptions made consistent.

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F Acknowledgments

   We wish to thank the members of the IETF MMUSIC WG for their comments
   and suggestions. Detailed comments were provided by Jim Buller, Dave
   Devanathan, Yaron Goland, Christian Huitema, Gadi Karmi, Jonathan
   Lennox, Moshe J. Sambol, and Eric Tremblay.

   This work is based, inter alia, on [29,30].

G Authors' Addresses

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

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

   Eve Schooler
   Computer Science Department 256-80
   California Institute of Technology
   Pasadena, CA 91125
   electronic mail:  schooler@cs.caltech.edu

   Jonathan Rosenberg
   Lucent Technologies, Bell Laboratories
   Rm. 4C-526
   101 Crawfords Corner Road
   Holmdel, NJ 07733
   electronic mail:  jdrosen@bell-labs.com

H Bibliography

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

   [2] B. Braden, L. Zhang, S. Berson, S. Herzog, and S. Jamin,

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   "Resource ReSerVation protocol (RSVP) -- version 1 functional
   specification," RFC 2205, Internet Engineering Task Force, Oct. 1997.

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

   [4] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming
   protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.

   [5] M. Handley and V. Jacobson, "SDP: session description protocol,"
   RFC 2327, Internet Engineering Task Force, Apr. 1998.

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

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

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

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

   [10] C. Partridge, "Mail routing and the domain system," RFC STD 14,
   974, Internet Engineering Task Force, Jan. 1986.

   [11] P. Mockapetris, "Domain names - implementation and
   specification," RFC STD 13, 1035, Internet Engineering Task Force,
   Nov. 1987.

   [12] B. Braden, "Requirements for internet hosts - application and
   support," RFC STD 3, 1123, Internet Engineering Task Force, Oct.

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

   [14] S. Williamson, M. Kosters, D. Blacka, J. Singh, and K. Zeilstra,
   "Referral whois (rwhois) protocol V1.5," RFC 2167, Internet
   Engineering Task Force, June 1997.

   [15] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access

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   protocol," RFC 1777, Internet Engineering Task Force, Mar. 1995.

   [16] E. M. Schooler, "A multicast user directory service for
   synchronous rendezvous," Master's Thesis CS-TR-96-18, Department of
   Computer Science, California Institute of Technology, Pasadena,
   California, Aug. 1996.

   [17] 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," RFC 1630, Internet
   Engineering Task Force, June 1994.

   [18] T. Berners-Lee, L. Masinter, and R. Fielding, "Uniform resource
   identifiers (URI): generic syntax," Internet Draft, Internet
   Engineering Task Force, Mar. 1998.  Work in progress.

   [19] P. Leach and R. Salz, "UUIDs and GUIDs," Internet Draft,
   Internet Engineering Task Force, Feb. 1998.  Work in progress.

   [20] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
   2279, Internet Engineering Task Force, Jan. 1998.

   [21] D. Crocker and P. Overell, "Augmented BNF for syntax
   specifications:  ABNF," RFC 2234, Internet Engineering Task Force,
   Nov. 1997.

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

   [23] J. Mogul and S. Deering, "Path MTU discovery," RFC 1191,
   Internet Engineering Task Force, Nov. 1990.

   [24] D. Crocker, "Standard for the format of ARPA internet text
   messages," RFC STD 11, 822, Internet Engineering Task Force, Aug.

   [25] P. Hoffman, L. Masinter, and J. Zawinski, "The mailto URL
   scheme," RFC 2368, Internet Engineering Task Force, July 1998.

   [26] J. Palme, "Common internet message headers," RFC 2076, Internet
   Engineering Task Force, Feb. 1997.

   [27] J. Mogul, T. Berners-Lee, L. Masinter, P. Leach, R. Fielding, H.
   Nielsen, and J. Gettys, "Hypertext transfer protocol -- HTTP/1.1,"
   Internet Draft, Internet Engineering Task Force, Mar. 1998.  Work in

   [28] M. Elkins, "MIME security with pretty good privacy (PGP)," RFC

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   2015, Internet Engineering Task Force, Oct. 1996.

   [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 (IDMS) , (Berlin, Germany), Mar.

   Full Copyright Statement

   Copyright (c) The Internet Society (1998). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   This document and the information contained herein is provided on an

                           Table of Contents

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   1          Introduction ........................................    2
   1.1        Overview of SIP Functionality .......................    2
   1.2        Terminology .........................................    4
   1.3        Definitions .........................................    4
   1.4        Summary of SIP Operation ............................    7
   1.4.1      SIP Addressing ......................................    7
   1.4.2      Locating a SIP Server ...............................    8
   1.4.3      SIP Transaction .....................................   10
   1.4.4      SIP Invitation ......................................   10
   1.4.5      Locating a User .....................................   12
   1.4.6      Changing an Existing Session ........................   14
   1.4.7      Registration Services ...............................   14
   1.5        Protocol Properties .................................   14
   1.5.1      Minimal State .......................................   14
   1.5.2      Lower-Layer-Protocol Neutral ........................   14
   1.5.3      Text-Based ..........................................   15
   2          SIP Uniform Resource Locators .......................   15
   3          SIP Message Overview ................................   19
   4          Request .............................................   21
   4.1        Request-Line ........................................   21
   4.2        Methods .............................................   21
   4.2.1      INVITE ..............................................   23
   4.2.2      ACK .................................................   23
   4.2.3      OPTIONS .............................................   24
   4.2.4      BYE .................................................   24
   4.2.5      CANCEL ..............................................   24
   4.2.6      REGISTER ............................................   25
   4.3        Request-URI .........................................   28
   4.3.1      SIP Version .........................................   28
   4.4        Option Tags .........................................   29
   4.4.1      Registering New Option Tags with IANA ...............   29
   5          Response ............................................   30
   5.1        Status-Line .........................................   30
   5.1.1      Status Codes and Reason Phrases .....................   30
   6          Header Field Definitions ............................   32
   6.1        General Header Fields ...............................   34
   6.2        Entity Header Fields ................................   34
   6.3        Request Header Fields ...............................   35
   6.4        Response Header Fields ..............................   35
   6.5        End-to-end and Hop-by-hop Headers ...................   35
   6.6        Header Field Format .................................   35
   6.7        Accept ..............................................   37
   6.8        Accept-Encoding .....................................   37
   6.9        Accept-Language .....................................   38
   6.10       Allow ...............................................   38
   6.11       Authorization .......................................   38
   6.12       Call-ID .............................................   38
   6.13       Content-Encoding ....................................   40

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   6.14       Content-Length ......................................   40
   6.15       Content-Type ........................................   40
   6.16       CSeq ................................................   41
   6.17       Date ................................................   42
   6.18       Encryption ..........................................   42
   6.19       Expires .............................................   44
   6.20       From ................................................   45
   6.21       Hide ................................................   46
   6.22       Location ............................................   47
   6.23       Max-Forwards ........................................   49
   6.24       Organization ........................................   50
   6.25       Priority ............................................   50
   6.26       Proxy-Authenticate ..................................   51
   6.27       Proxy-Authorization .................................   51
   6.28       Proxy-Require .......................................   51
   6.29       Record-Route ........................................   52
   6.30       Require .............................................   52
   6.31       Response-Key ........................................   53
   6.32       Retry-After .........................................   54
   6.33       Route ...............................................   55
   6.34       Server ..............................................   55
   6.35       Subject .............................................   55
   6.36       Timestamp ...........................................   55
   6.37       To ..................................................   56
   6.38       Unsupported .........................................   56
   6.39       User-Agent ..........................................   57
   6.40       Via .................................................   57
   6.40.1     Requests ............................................   57
   6.40.2     Receiver-tagged Via Fields ..........................   58
   6.40.3     Responses ...........................................   58
   6.40.4     Syntax ..............................................   59
   6.41       Warning .............................................   60
   6.42       WWW-Authenticate ....................................   62
   7          Status Code Definitions .............................   62
   7.1        Informational 1xx ...................................   63
   7.1.1      100 Trying ..........................................   63
   7.1.2      180 Ringing .........................................   63
   7.1.3      181 Call Is Being Forwarded .........................   63
   7.1.4      182 Queued ..........................................   63
   7.2        Successful 2xx ......................................   64
   7.2.1      200 OK ..............................................   64
   7.3        Redirection 3xx .....................................   64
   7.3.1      300 Multiple Choices ................................   64
   7.3.2      301 Moved Permanently ...............................   65
   7.3.3      302 Moved Temporarily ...............................   65
   7.3.4      380 Alternative Service .............................   65
   7.4        Request Failure 4xx .................................   65
   7.4.1      400 Bad Request .....................................   65

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   7.4.2      401 Unauthorized ....................................   65
   7.4.3      402 Payment Required ................................   66
   7.4.4      403 Forbidden .......................................   66
   7.4.5      404 Not Found .......................................   66
   7.4.6      405 Method Not Allowed ..............................   66
   7.4.7      406 Not Acceptable ..................................   66
   7.4.8      407 Proxy Authentication Required ...................   66
   7.4.9      408 Request Timeout .................................   66
   7.4.10     409 Conflict ........................................   67
   7.4.11     414 Request-URI Too Long ............................   67
   7.4.12     415 Unsupported Media Type ..........................   67
   7.4.13     420 Bad Extension ...................................   67
   7.4.14     480 Temporarily Unavailable .........................   67
   7.4.15     481 Invalid Call-ID .................................   67
   7.4.16     482 Loop Detected ...................................   67
   7.4.17     483 Too Many Hops ...................................   67
   7.4.18     484 Address Incomplete ..............................   68
   7.4.19     485 Ambiguous .......................................   68
   7.5        Server Failure 5xx ..................................   68
   7.5.1      500 Server Internal Error ...........................   69
   7.5.2      501 Not Implemented .................................   69
   7.5.3      502 Bad Gateway .....................................   69
   7.5.4      503 Service Unavailable .............................   69
   7.5.5      504 Gateway Timeout .................................   69
   7.5.6      505 Version Not Supported ...........................   69
   7.6        Global Failures 6xx .................................   70
   7.6.1      600 Busy ............................................   70
   7.6.2      603 Decline .........................................   70
   7.6.3      604 Does Not Exist Anywhere .........................   70
   7.6.4      606 Not Acceptable ..................................   70
   8          SIP Message Body ....................................   70
   8.1        Body Inclusion ......................................   71
   8.2        Message Body Type ...................................   71
   8.3        Message Body Length .................................   71
   9          Compact Form ........................................   71
   10         Behavior of SIP Clients and Servers .................   72
   10.1       General Remarks .....................................   72
   10.1.1     Requests ............................................   73
   10.1.2     Responses ...........................................   73
   10.2       Source Addresses, Destination Addresses and
   Connections ....................................................   74
   10.2.1     Unicast UDP .........................................   74
   10.2.2     Multicast UDP .......................................   74
   10.3       TCP .................................................   75
   10.4       Reliability for BYE, CANCEL, OPTIONS, REGISTER
   Requests .......................................................   76
   10.4.1     UDP .................................................   76
   10.4.2     TCP .................................................   76

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   10.5       Reliability for ACK Requests ........................   76
   10.6       Reliability for INVITE Requests .....................   77
   10.6.1     UDP .................................................   77
   10.6.2     TCP .................................................   78
   11         Behavior of SIP User Agents .........................   81
   11.1       Caller Issues Initial INVITE Request ................   81
   11.2       Callee Issues Response ..............................   81
   11.3       Caller Receives Response to Initial Request .........   81
   11.4       Caller or Callee Generate Subsequent Requests .......   82
   11.5       Receiving Subsequent Requests .......................   82
   12         Behavior of SIP Proxy and Redirect Servers ..........   82
   12.1       Redirect Server .....................................   82
   12.2       User Agent Server ...................................   83
   12.3       Proxy Server ........................................   83
   12.3.1     Proxying Requests ...................................   83
   12.3.2     Proxying Responses ..................................   84
   12.3.3     Stateless Proxy: Proxying Responses .................   84
   12.3.4     Stateful Proxy: Receiving Requests ..................   84
   12.3.5     Stateful Proxy: Receiving ACKs ......................   84
   12.3.6     Stateful Proxy: Receiving Responses .................   84
   12.3.7     Stateless, Non-Forking Proxy ........................   85
   12.4       Forking Proxy .......................................   85
   13         Security Considerations .............................   89
   13.1       Confidentiality and Privacy: Encryption .............   89
   13.1.1     End-to-End Encryption ...............................   89
   13.1.2     Privacy of SIP Responses ............................   92
   13.1.3     Encryption by Proxies ...............................   92
   13.1.4     Hop-by-Hop Encryption ...............................   92
   13.1.5     Via field encryption ................................   92
   13.2       Message Integrity and Access Control:
   Authentication .................................................   93
   13.2.1     Trusting responses ..................................   95
   13.3       Callee Privacy ......................................   96
   13.4       Known Security Problems .............................   96
   14         SIP Security Using PGP ..............................   97
   14.1       PGP Authentication Scheme ...........................   97
   14.1.1     The WWW-Authenticate Response Header ................   97
   14.1.2     The Authorization Request Header ....................   98
   14.2       PGP Encryption Scheme ...............................   99
   14.3       Response-Key Header Field for PGP ...................   99
   15         Examples ............................................  100
   15.1       Registration ........................................  100
   15.2       Invitation to a Multicast Conference ................  101
   15.2.1     Request .............................................  102
   15.2.2     Response ............................................  102
   15.3       Two-party Call ......................................  104
   15.4       Terminating a Call ..................................  106
   15.5       Forking Proxy .......................................  106

Handley/Schulzrinne/Schooler/Rosenberg                      [Page 127]

Internet Draft                    SIP                     August 7, 1998

   15.6       Redirects ...........................................  110
   15.7       Alternative Services ................................  112
   15.8       Negotiation .........................................  113
   15.9       OPTIONS Request .....................................  114
   A          Minimal Implementation ..............................  114
   A.1        Client ..............................................  114
   A.2        Server ..............................................  115
   A.3        Header Processing ...................................  115
   B          Usage of SDP ........................................  115
   C          Summary of Augmented BNF ............................  117
   D          IANA Considerations .................................  119
   E          Changes in Version -08 ..............................  119
   F          Acknowledgments .....................................  120
   G          Authors' Addresses ..................................  120
   H          Bibliography ........................................  120

Handley/Schulzrinne/Schooler/Rosenberg                      [Page 128]

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