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Internet Engineering Task Force                                MMUSIC WG
Internet Draft                              Handley/Schulzrinne/Schooler
draft-ietf-mmusic-sip-05.txt                     ISI/Columbia U./Caltech
May 14, 1998
Expires: November 1998

                    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
   working documents as Internet-Drafts.

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   Distribution of this document is unlimited.


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

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

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1 Introduction

1.1 Overview of SIP Functionality

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

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

   SIP transparently supports name mapping and redirection services,
   allowing the implementation of ISDN and Intelligent Network telephony
   subscriber services. 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 network to

   SIP supports some or all of five facets of establishing and
   terminating multimedia communications:

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

   User capabilities: determination of the media and media parameters to
        be used;

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

   Call setup: "ringing", establishment of call parameters at both

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        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 [4] currently incorporating protocols such as
   RSVP (RFC 2205 [5]) for reserving network resources, the real-time
   transport protocol (RTP) (RFC 1889 [6]) for transporting real-time
   data and providing QOS feedback, the real-time streaming protocol
   (RTSP) [1] for controlling delivery of streaming media, the session
   announcement protocol (SAP) [2] for advertising multimedia sessions
   via multicast and the session description protocol (SDP) (RFC 2327
   [7]) for describing multimedia sessions, but the functionality and
   operation of SIP does not depend on any of these protocols.

   SIP may also be used in conjunction with other call setup and
   signaling protocols. In that mode, an end system uses SIP protocol
   exchanges to determine the appropriate end system address and
   protocol from a given address that is protocol-independent. For
   example, SIP may be used to determine that the party may be reached
   via H.323, obtain the H.245 gateway and user address and then use
   H.225.0 to establish the call [8]. 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-
   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 [9] and

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   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 [10]). 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.

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

   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.

   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,
        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.

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   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-

   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
        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

   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

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        order to service requests and sends back responses to those
        requests.  Servers are either proxy, redirect or user agent

   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, [7]) (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 for a
        single call (identified by a  Call-ID, Section 6.12) and may be
        identified by a  CSeq sequence number (Section 6.16).  If there
        is no  CSeq, there can only be one pending transaction between a
        server and client for each call id.

   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 [11].

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

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

   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 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.

1.4 Summary of SIP Operation

   This section explains the basic protocol functionality and operation.

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          property               redirect    proxy     user agent
                                  server     server      server
          also acts as client       no        yes          no
          return  1xx status       yes        yes         yes
          return  2xx status        no        yes         yes
          return  3xx status       yes        yes         yes
          return  4xx status       yes        yes         yes
          return  5xx status       yes        yes         yes
          return  6xx status        no        yes         yes
          insert  Via header        no        yes          no
          accept  ACK               no        yes         yes

   Table 1: Properties of the different SIP server types

   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

   SIP addresses contain a user and host part. The 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 [12]), MX (RFC 974 [13], CNAME
   or A record (RFC 1035 [14]), or a numeric network address.

   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, the SIP address can be the same as a user's electronic
   mail address, but this is not required. SIP can thus leverage off the
   domain name system (DNS) to provide a first-stage location

   Examples include:


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

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

   Since SIP requests and responses may also contain non-SIP addresses,
   e.g., telephone numbers, SIP addresses are written as SIP URLs
   (Section 2) when used within SIP headers. For example,


1.4.2 Locating a SIP Server

   A SIP client MUST follow the following steps to resolve the host part
   of a callee address. 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
   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 [12]) 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

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             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 [12]) 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 [13]), 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 call. 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 [15]).

1.4.3 SIP Transaction

   Once the host part has been resolved to a SIP server, the client
   sends one or more SIP requests to that server and receives one or
   more responses from the server. A request (and its retransmissions)
   together with the responses triggered by that request make up a SIP
   transaction.  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

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   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 6.43)
   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, [7]) format, that provides the
   called party with enough information to join the session. For
   multicast sessions, the session description enumerates the media
   types and formats that may be distributed to that session. For
   unicast session, the session description enumerates the media types
   and formats that the caller is willing to receive and where it wishes
   the media data to be sent. In either case, if the callee wishes to
   accept the call, it responds to the invitation by returning a similar
   description listing the media it wishes to receive. For a multicast
   session, the callee should only return a session description if it is
   unable to receive the media indicated in the caller's description.
   The caller may ignore the session description returned or use it to
   change the global session description.

   The protocol exchanges for the  INVITE method are shown in Fig. 1 for
   a proxy server and in Fig. 2 for a redirect server. 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

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   8 and 9). All requests and responses have the same Call-ID.

                                            +....... 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

   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.

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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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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 [16]), rwhois (RFC
   2167 [17]), LDAP (RFC 1777 [18]), multicast-based protocols or
   operating-system dependent mechanism 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 sending the request as  Location headers (Section 6.25). 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 6.43)
   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. When a multicast request is made, first
   the host making the request, then the multicast address itself are
   added to the path. A proxy server MUST check that it does not
   generate a request to a host listed in the  Via list.  (Note: If a
   host has several names or network addresses, this may not always
   work. Thus, each host also checks if it is part of the Via list.)

   A SIP invitation may traverse more than one SIP proxy server. If one
   of these "forks" the request, i.e., issues more than one request in
   response to receiving the invitation request, it is possible that a
   client is reached, independently, by more than one copy of the
   invitation request. Each of these copies bears the same Call-ID. The
   user agent MUST return the appropriate status response. Duplicate
   requests are not an error, so there is no need to alert the user.

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

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   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 which address(es) it may be reached under. A client may
   also use it to install call handling features at the server.

1.5 Protocol Properties

1.5.1 Minimal State

   A single conference session or call may involve one or more SIP
   request-response transactions. Proxy servers do not have to keep
   state for a particular call, however, they maintain state for a
   single SIP transaction, as discussed in Section 11.

   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
   existing firewalls, and given the similar protocol design, allows
   common servers for SIP, HTTP and the Real Time Streaming Protocol
   (RTSP) [1].

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

   User agents SHOULD implement both UDP and TCP transport, proxy and
   redirect servers MUST.

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   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.

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,
   current destination and final recipient of a SIP request, and to
   specify redirection addresses. A SIP URL can also be embedded in web
   pages or other hyperlinks to indicate that a user or service may be

   Because interaction with some resources may require message headers
   or message bodies to be specified as well as the SIP address, the SIP
   URL scheme is defined to allow setting SIP  request-header fields and
   the SIP  message-body. (This is similar to the  mailto:  URL [19].)

   A SIP URL follows the guidelines of RFC 1630, as revised, [20,21] and
   takes the following form:

        SIP-URL                 =    full-sip-url
        full-sip-url            =    "sip:" ( user | phone ) [ ":" password ]
                                     "@" [ host | nhost ] : port
                                     url-parameters [ headers ]
        user                    =    ;  defined in RFC 1738 [11]
        phone                   =    telephone-subscriber
        telephone-subscriber    =    ;
        defined in [22]
        password                =    *( unreserved | escaped |
                                     ";" | "                                     " | "=" | "+" | "$" | "," )
        host                    =    ;  defined in RFC 1738
        nhost                   =    hostnumber
        hostnumber              =    digits "." digits "." digits "." digits
        port                    =    digits
        url-parameters          =    *( ";" url-parameter)
        url-parameter           =    transport-param |

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                                     ttl-param | maddr-param | other-param
        transport-param         =    "transport=" ( "udp" | "tcp" )
        ttl-param               =    "ttl=" ttl
        ttl                     =    1*3DIGIT                                      ; 0 to 255
        maddr-param             =    "maddr=" maddr
        maddr                   =    ;  dotted decimal multicast address
        other-param             =    *uric
        headers                 =    "?" header *( "                             " header )
        header                  =    hname "=" hvalue
        hname                   =    *uric
        hvalue                  =    *uric
        uric                    =    ;  defined in [21]
        digits                  =    1*DIGIT

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

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

   The  mailto: URL and RFC 822 email addresses require that numeric
   host addresses ("host numbers") are enclosed in square brackets
   (presumably, since host names might be numeric), while host numbers
   without brackets are used for all other URLs. The SIP URL requires
   the latter form.

   The  password parameter can be used for a basic authentication
   mechanism that takes the place of an unlisted telephone number. Also,
   for Internet telephony gateways, it may serve as a personal
   identification number (PIN). Including just the password in the URL
   is more convenient than including a whole authentication header. This
   approach may be reasonably secure if the URL is part of a secured web
   page. Unless the SIP transaction is carried over a secure network
   connection, this carries the same security risks as all URL-based
   passwords and should only be used when security requirements are low.
   In general, it is NOT RECOMMENDED and use of the Authorization
   (Section 6.11) header is preferred.

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

   Examples of short and full-form SIP URLs are:

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   Within a SIP message, URLs are used to indicate the source and
   intended destination of a request, redirection addresses and the
   current destination of a request. Normally all these fields will
   contain SIP URLs. When additional parameters are not required, the
   short form SIP URL can be used unambiguously.

   SIP URLs are case-insensitive, so that sip:j.doe@example.com and
   SIP:J.Doe@Example.com are equivalent.

   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.

   SIP URLs can define specific parameters of the request, including the
   transport mechanism (UDP or TCP) and the use of multicast to make a
   request. These parameters are added after the  host and are separated
   by semi-colons. For example, to specify to call j.doe@big.com using
   multicast to with a ttl of 15, the following URL would
   be used:


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

3 SIP Message Overview

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   SIP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [23]). 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 [10]). In addition, we describe SIP in both
   prose and an augmented Backus-Naur form (BNF) [H2.1] described in
   detail in RFC 2234 [24].

   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 [25] 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
        [26]). 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 [27] for transferring entities (the
   body of the message). Both types of message consist of a  start-line,
   one or more header fields (also known as "headers"), an empty line
   (i.e., a line with nothing preceding the carriage-return line-feed (
   CRLF)) indicating the end of the header fields, and an optional
   message-body. To avoid confusion with similar-named headers in HTTP,
   we refer to the header describing the message body as entity headers.
   These components are described in detail in the upcoming sections.

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        generic-message    =    start-line
                                [ 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, the  CRLF should be ignored.

4 Request

   The  Request message format is shown below:

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

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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 6.20
                           |     From                   ; Section 6.21
                           |     Record-Route           ; Section 6.33
                           |     Timestamp              ; Section 6.39
                           |     Via                    ; Section 6.43
        entity-header      =     Content-Encoding       ; Section 6.13
                           |     Content-Length         ; Section 6.14
                           |     Content-Type           ; Section 6.15
                           |     ETag                   ; Section 6.19
        request-header     =     Accept                 ; Section 6.7
                           |     Accept-Encoding        ; Section 6.8
                           |     Accept-Language        ; Section 6.9
                           |     Authorization          ; Section 6.11
                           |     Hide                   ; Section 6.22
                           |     If-Match               ; Section 6.23
                           |     If-None-Match          ; Section 6.24
                           |     Location               ; Section 6.25
                           |     Max-Forwards           ; Section 6.26
                           |     Organization           ; Section 6.27
                           |     Priority               ; Section 6.28
                           |     Proxy-Authorization    ; Section 6.30
                           |     Proxy-Require          ; Section 6.31
                           |     Route                  ; Section 6.35
                           |     Require                ; Section 6.32
                           |     Response-Key           ; Section 6.34
                           |     Subject                ; Section 6.38
                           |     To                     ; Section 6.40
                           |     User-Agent             ; Section 6.42
        response-header    =     Allow                  ; Section 6.10
                           |     Location               ; Section 6.25
                           |     Proxy-Authenticate     ; Section 6.29
                           |     Retry-After            ; Section 6.36
                           |     Server                 ; Section 6.37
                           |     Unsupported            ; Section 6.41
                           |     Warning                ; Section 6.44
                           |     WWW-Authenticate       ; Section 6.45

   Table 2: SIP headers

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4.2 Methods

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

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

        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 agents receives an  INVITE with a new  CSeq sequence
   number, 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. 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 conference.)

   This method MUST be supported by a SIP server and client.

4.2.2  ACK

   The  ACK request confirms that the client has received a final
   response to an  INVITE request. 2xx responses are acknowledged by
   client user agents, all other final responses by the first proxy or

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   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. This method MUST be supported by a SIP server and

   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

4.2.3  OPTIONS

   The client is being queried as to its capabilities. A server that
   believes it can contact the user, such as a user agent where the user
   is logged in and has been recently active, MAY respond to this
   request with a capability set. Support of this method is REQUIRED.

   A called user agent MAY return a status reflecting how it would have
   responded to an invitation, e.g., 600 (Busy).

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. It terminates any on-going searches for the named call. A
   caller SHOULD issue a  BYE request before aborting a call ("hanging
   up"). Note that a  BYE request may also be issued by the callee.

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

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

4.2.5  CANCEL

   The  CANCEL request cancels any pending searches, but does not
   terminate an accepted call at a particular user agent. (A call is
   considered accepted if the callee has returned a 200 (OK) status
   response.) A proxy SHOULD issue a  CANCEL request to all 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 triggered by an INVITE. The
   Call-ID,  To and  From in the CANCEL request are identical to those

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   contained in the  CANCEL request, but the  Via header field is
   initialized to the proxy issuing the  CANCEL request.

   A redirect server or user agent server returns 200 (OK) if the Call-
   ID exists and 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.

   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 to a SIP server.

   A user agent SHOULD register with a local server on startup by
   sending a  REGISTER request to the well-known "all SIP servers"
   multicast address,, 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 [28];
   however, they do not respond to the request.

   The  REGISTER request interprets header fields as follows. We define
   "address-of-record" as the SIP address that the registry knows the
   registrand under, 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 alice@acme.com may register under

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   Location: If the request contains a  Location header field, requests
        for the  Request-URI will also be directed to the address(es)
        given. It is recommended that user agents include both SIP UDP
        and TCP addresses in their registration.  Registrations are

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

   Otherwise, future call control requests will be directed to the
   network source address of the  REGISTER request, using the  To
   address in the  REGISTER request as the  Request-URI. 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".

   After registration, the server MAY forward incoming SIP requests to
   the network source address and port that originated the registration
   request. 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 6.20).
   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.

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

        Beyond its use as a simple location service, this method is
        needed if there are several SIP servers on a single host,
        so that some cannot use the default port number. Each such
        server would register with a server for the administrative
        domain. 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. For example, a proxy may perform a lookup
   on the contents of the  To field to resolve a username from a mail
   alias, and then use this username as part of the  Request-URI field
   of requests it generates.

   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. The case where the Request-URI
   and server host name disagrees occurs, for example, for a firewall
   proxy that handles outgoing calls. It is similar to the operation 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
   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 6.32) and Unsupported
   (Section 6.41) fields.


        option-tag  ___   1*urlc

   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

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   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.

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

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   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.

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

   1xx: Informational -- request received, continuing process;

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

   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.

   Presented below are the individual values of the numeric response
   codes, and an example set of corresponding reason phrases for
   SIP/2.0. These reason phrases are only recommended; they may be

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   replaced by local equivalents without affecting the protocol. Note
   that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
   codes in the range starting at x80 to avoid conflicts with newly
   defined HTTP response codes, and extends these response codes in the
   6xx range.

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

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

   Figure 3: Informational and success status codes

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

   Figure 4: Redirection status codes

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        Client-Error    =    "400"    ;  Bad Request
                       |     "401"    ;  Unauthorized
                       |     "402"    ;  Payment Required
                       |     "403"    ;  Forbidden
                       |     "404"    ;  Not Found
                       |     "405"    ;  Method Not Allowed
                       |     "407"    ;  Proxy Authentication Required
                       |     "408"    ;  Request Timeout
                       |     "409"    ;  Conflict
                       |     "410"    ;  Gone
                       |     "411"    ;  Length Required
                       |     "412"    ;  Precondition Failed
                       |     "413"    ;  Request Message Body Too Large
                       |     "414"    ;  Request-URI Too Large
                       |     "415"    ;  Unsupported Media Type
                       |     "420"    ;  Bad Extension
                       |     "480"    ;  Temporarily not available
                       |     "481"    ;  Invalid Call-ID
                       |     "482"    ;  Loop Detected
                       |     "483"    ;  Too Many Hops

   Figure 5: Client error status codes

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

   Figure 6: Server error status codes

   SIP 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

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

   Figure 7: Global failure status Codes

   that there was something wrong with its request and treat the
   response as if it had received a 400 (Bad Request) response code. In
   such cases, user agents SHOULD present to the user the message body
   returned with the response, since that message body is likely to
   include human-readable information which will explain the unusual

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), but 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.

   The header fields required, optional and not applicable for each
   method are listed in Table 3. 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
   message (Section 8).

   The "type" column describes the request and response types the header
   field may be used for. 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.

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

6.1 General Header Fields

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

   General-header field names can be extended reliably only in
   combination with a change in the protocol version. However, new or
   experimental header fields may be given the semantics of general
   header fields if all parties in the communication recognize them to
   be general-header fields.

6.2 Entity Header Fields

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

   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.

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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 [27,29].

   Each header field consists of a name followed by a colon (":") and
   the field value. Field names are case-insensitive. The field value
   may be preceded by any amount of leading white space (LWS), though a
   single space (SP) is preferred. Header fields can be extended over
   multiple lines by preceding each extra line with at least one  SP or
   horizontal tab (HT). Applications SHOULD follow HTTP "common form"
   when generating these constructs, since there might exist some
   implementations that fail to accept anything beyond the common forms.

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

   The order in which header fields are received is not significant if
   the header fields have different field names. Multiple header fields
   with the same field-name may be present in a message if and only if
   the entire field-value for that header field is defined as a comma-

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                           type     enc.    e-e   ACK   BYE   CAN   INV   OPT   REG
   Accept                   R                e     o     o     o     o     o     o
   Accept-Encoding          R                e     o     o     o     o     o     o
   Accept-Language          R        n       e     o     o     o     o     o     o
   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     o     -
   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     o
   Date                     g                e     o     o     o     o     o     o
   Encryption               g        n       e     o     o     o     o     o     o
   ETag                    200               e     -     -     -     o     -     -
   Expires                  g                e     -     -     -     o     o     o
   From                     R                e     m     m     m     m     m     m
   Hide                     R        n       h     o     o     o     o     o     o
   If-Match                 R                e     o     o     -     -     -     -
   If-None-Match            R                e     o     o     -     -     -     -
   Location                 R                e     -     -     -     -     -     o
   Location                3xx               e     -     -     o     o     o     o
   Location                2xx               e     -     -     o     o     o     -
   Max-Forwards             R        n       e     o     o     o     o     o     o
   Organization             R        c       e     -     -     -     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        n       e     o     o     o     o     o     o
   Retry-After              R        c       e     -     -     -     -     -     o
   Retry-After           600,603     c       e     -     -     -     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               R        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 3: Summary of header fields

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   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 OPTIONS and  INVITE 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

   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. This may also be used as
   a hint by the proxy to which destination to connect the call to
   (e.g., for selecting a human operator).


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

6.10  Allow

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   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.

6.11  Authorization

   See [H14.8] and [30]. 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. 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)

   For an  INVITE request, a callee client application alerts the user
   only if the user has not 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 client application 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  Call-ID may be any string consisting of the unreserved URI
   characters that can be guaranteed to be globally unique for the
   duration of the request.  Call-IDs are case-sensitive and are not

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

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        case-ignoring string comparison.

   The form  local-id@host is recommended, where  host is either the
   fully qualified domain name or a globally routable IP address. The
   local-id is a version-4 (random) UUID [31].

        Using cryptographically random identifiers provides some
        protection against session hijacking.

        Call-ID    =    ( "Call-ID" | "i" ) ":" UUID "@" host
        UUID       =    ;  see [31]


     Call-ID: 3436538253725150855@foo.bar.com

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 [H14.11].

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 MUST use this field to indicate the size of the

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   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 message without
   Content-Length, it MUST assume it to be zero.  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.

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

   Example 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 and the request
   method. The sequence number MUST be expressible as a 64-bit unsigned
   integer. The initial value of the sequence number is arbitrary.
   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 (a
   "re-invitation") acquires a new, higher sequence number. A server
   responding to a request containing a  CSeq header MUST echo the value
   in the response. If the  Method value is missing, the server fills it
   it appropriately.

   The  ACK request MUST contain the same  CSeq value as the INVITE
   request that it refers to, while a  BYE or  CANCEL 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.

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

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

        Strictly speaking,  CSeq header fields are needed for any
        SIP request that can be cancelled by a  BYE or  CANCEL
        request or where a client can issue several requests for
        the same Call-ID in close succession. Without a sequence
        number, the response to an  INVITE could be mistaken for
        the response to the cancellation ( BYE or  CANCEL). Also,
        if the network duplicates packets or if an  ACK is delayed
        until the server has sent an additional response, the
        client could interpret an old response as the response to a
        re-invitation issued shortly thereafter. Using CSeq also
        makes it easy for the server to distinguish different
        versions of an invitation, without comparing the message

   The  Method value allows the client to distinguish the response to an
   INVITE request from that of a  CANCEL response.

   At 64 bits, a server could generate one request a second for about
   500 billion years before needing to wrap around.

   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 is useful for simple devices without
        their own clock.

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6.18  Encryption

   The  Encryption general-header field specifies that the content has
   been encrypted. Section 12 describes the overall SIP security
   architecture and algorithms. It is intended for end-to-end encryption
   of requests and responses. Requests are encrypted 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

        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 12.

   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

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   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


   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  ETag

   The  ETag response-header field labels an instance of the callee. A
   callee MUST include an  ETag in a 200 response to an  INVITE if the
   Location response-header field is not sufficient to uniquely identify
   the callee. Typically, this is the case if the  Location header
   points to a proxy rather than the callee itself. (If requests are
   forked, it is possible that two or more people "pick up the phone"
   for the same call.)

   If the caller receives a 2xx response containing an entity tag, it
   MUST include a  If-Match (Section 6.23) request-header field with
   that entity tag in the  ACK or  BYE. It would send a  BYE with the
   entity tag if it does not wish to talk to this particular instance of
   the callee.

   The entity tag consists of an opaque quoted string. An entity tag
   MUST be unique across all instances associated with a particular To
   URI. A given entity tag value may be used for different URIs without

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   implying anything about the equivalence of those URIs. It is
   RECOMMENDED that the entity tag is a cryptographically random
   identifier with at least 32 bits of randomness.

   Note that SIP does not use the concept of "weak" entity tags [H3.11].

        ETag          =    "ETag" ":" entity-tag
        entity-tag    =    quoted-string

6.20  Expires

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

   This header field is currently defined only for the  REGISTER and
   INVITE methods. For  REGISTER, it is a request and response-header
   field and allows the client to indicate how long the registration is
   to be valid; the server uses it to indicate when the client has to
   re-register. The server's choice overrides that of the client. The
   server MAY choose a shorter time interval than that requested by the
   client, but SHOULD not choose a longer one.

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

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

   Two example of its use are

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

6.21  From

   Requests and responses MUST contain a  From general-header field,
   indicating the invitation initiator. The server copies the To and
   From header fields from the request to the response.

        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

6.22  Hide

   The  Hide request header field indicates that the path comprised of
   the  Via header fields (Section 6.43) 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

   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 6.43.

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   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
   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 12.

   The  Hide header field has the following syntax:

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

6.23  If-Match

   The  If-Match request-header field is used with a method to make it
   conditional. The server MUST NOT perform the request unless it had
   returned one of the listed entity tags in a  ETag header field in a
   200 response for the same  Call-ID. The "*" tag matches all entity
   tags, as well as the case where the server that had not returned an
   entity tag. If there is no match, a user agent server MUST return a
   412 (Precondition Failed) response, while a proxy server forwards the
   request according to the  Request-URI. A proxy server that receives

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   an  ACK with a matching entity tag MUST NOT forward the request.

        If-Match    =    "If-Match" ":" ( "*" | 1#entity-tag )

   The  If-Match allows a caller to select among a set of callee that
   answer to the same  request-URI and  To.


     If-Match: "83ja", "148293289"

6.24  If-None-Match

   The  If-None-Match request-header field is used with a method to make
   the method conditional. The server compares the list of entity tags
   in the  If-None-Match header to the entity tag that it had returned
   in an  ETag header (Section 6.19) as part of a 200 response for the
   same  Call-ID. The value "*" matches any entity tag. If one of the
   entity tag in the  If-None-Match header matches the  ETag entity tag,
   a user agent server MUST NOT perform the method requested and respond
   with a status of 412 (Precondition Failed) instead while a proxy
   server forwards the request.  (The "*" value is included for symmetry
   with  If-Match only and currently has no practical application.)

        If-None-Match    =    "If-None-Match" ":" ( "*" | 1#entity-tag )


          If-None-Match: "83ja", "148293289"

6.25  Location

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

   REGISTER requests:  REGISTER requests MAY contain Location header
        fields. They indicate under which locations the user may be
        reachable. The  REGISTER request defines a wildcard Location
        field, "*". that is only used with  Expires:  0 to remove all

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        registrations for a particular user.

   INVITE and  ACK requests:  INVITE and  ACK requests MAY contain
        Location headers indicating the location the request is
        originating from.

        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 proxy.

   3xx responses: The  Location response-header field can be used with a
        3xx response codes to indicate one or more addresses to try.  It
        can appear in responses to  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 301 (Moved Permanently) or 302 (Moved Temporarily)
        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 multicast address to
        try or a change of SIP transport from UDP to TCP or vice versa.

   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. 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 and the  To header of subsequent
        ACK and  BYE requests for this call.

   REGISTER 2xx responses: Similarly, a  REGISTER response SHOULD return
        all locations that a user is currently reachable under.

   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 URL [22], a  mailto: URL [19]
   or  irc: URL.

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

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   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
        request. It indicates how the client wishes forwarding to occur,
        by proxying or by redirection. 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 requests.

        Location    =    ( "Location" | "m" ) ("*" | (1# (( SIP-URL | URI )
                         *( ";" location-params )))

        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

6.26  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.

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   Each proxy or gateway recipient of a request containing a Max-
   Forwards header field MUST check and update its value prior to
   forwarding the request. If the received value is zero (0), the
   recipient MUST NOT forward the request. Instead, for the  OPTIONS and
   REGISTER methods, it MUST respond as the final recipient. For all
   other methods, the server returns 483 (Too many hops).

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


     Max-Forwards: 6

6.27  Organization

   The  Organization request-header field conveys the name of the
   organization to which the callee 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

6.28  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, with the addition of

6.29  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.

6.30  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.

6.31  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

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   negatively acknowledged by the proxy to the client if not supported.
   Servers treat this field identically to the Require field.

   See Section 6.32 for more details on the mechanics of this message
   and a usage example.

6.32  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
   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 6.31).

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6.33  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 client 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 clientMUST 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# request-uri

   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

6.34  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

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        key-param       =    token "=" ( token | quoted-string )

   The  key-scheme gives the type of encryption to be used for response.
   Section 12 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 an 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.

6.35  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 6.33.

   The  Route header field has the following syntax:

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

6.36  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.

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        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

   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.

6.37  Server

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

6.38  Subject

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

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


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

6.39  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

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   this, add a floating point number indicating the number of seconds
   that have elapsed since it has received the request. The timestamp is
   used by the client to compute the round-trip time to the server so
   that it can adjust the timeout value for retransmissions.

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

6.40  To

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

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

   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

6.41  Unsupported

   The  Unsupported response header lists the features not supported by
   the server. See Section 6.32 for a usage example and motivation.

6.42  User-Agent

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

6.43  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.

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6.43.1 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 it wishes to receive responses
   at. (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

   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 below.

        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

   Additionally, if the message goes to a multicast address, an extra
   Via field is added by the sender before all the other Via fields
   giving the multicast address and TTL.

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

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

6.43.2 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

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   the request, 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 went from and through a NAT
   using external address border.ieee.org ( to
   erlang.bell-telephone.com tagged the previous hop's  Via field with
   the address that it actually came from.

6.43.3 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 message. If it does not, discard the
             message.  Otherwise, remove this  Via field.

        2.   If the second  Via field in a response is a multicast
             address, remove that  Via field, and send the message to
             the multicast address indicated.

        3.   If the second  Via field is a receiver-tagged field
             (Section 6.43.2) send the message to the address in the
             received tag rather than that in the main part of the Via

        4.   If the second  Via field exists, send the message to the
             address indicated. If there is no second  Via field, this
             response is destined for this client.

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

6.43.4 Syntax

   The format for a  Via header is:

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        Via                   =    ( "Via" | "v") ":" 1#( sent-protocol sent-by
                                   *( ";" via-params ) [ comment ] )
        via-params            =    via-hidden | via-ttl | via-received | via-branch
        via-hidden            =    "hidden"
        via-ttl               =    "ttl" "=" ttl
        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

   The " ttl" parameter is included only if the address is a multicast
   address. The " received" parameter is added only for receiver-added
   Via fields (Section 6.43.2).  For reasons of privacy, a client or
   proxy may wish to hide its Via information by encrypting it (see
   Section 6.22).  The " hidden" parameter is included if this header
   was hidden by the upstream proxy (see 6.22).

   The " branch" parameter is included by every forking proxy.  The
   token uniquely identifies a branch of a particular search. The
   identifier has to be unique only within a set of isomorphic requests.

   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.

     Via: SIP/2.0/UDP first.example.com:4000
     Via: SIP/2.0/UDP adk8

6.44  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

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        warn-code        =    3DIGIT "." 2DIGIT
        warn-agent       =    ( host [ ":" port ] ) | pseudonym
                              ;  the name or pseudonym of the server adding
                              ;  the Warning header, for use in debugging
        warn-text        =    quoted-string

   A response may carry more than one  Warning header.

   The  warn-text should be in a natural language that is most likely to
   be intelligible to the human user receiving the response.  This
   decision may be based on any available knowledge, such as the
   location of the cache or user, the  Accept-Language field in a
   request, the  Content-Language field in a response, etc. The default
   language is English.

   Any server may add  Warning headers to a response. New Warning
   headers MUST be added after any existing  Warning headers. A proxy
   server MUST NOT delete any  Warning header that it received with a

   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.


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

6.45  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 [30].

   The  WWW-Authenticate response-header field MUST be included in 401

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   (Unauthorized) response messages.

   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, [32]), Section 13. Other schemes, such as S-MIME, are
   for further study.

7 Status Code Definitions

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

7.1 Informational 1xx

   Informational responses indicate that the server or proxy contacted
   is performing some further action and does not yet have a definitive
   response. The client SHOULD wait for a further response from the
   server, and the server SHOULD send such a response without further
   prompting. Typically a server should send a 1xx response if it
   expects to take more than 200 ms to obtain a final response.

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 been recently and is trying to alert them.

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.

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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 message body is empty.

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 6.43) path of the request in the Location header
   field. (Addresses match if their host and port number match.)

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
   Content-Type header field. The choices SHOULD also be listed as
   Location fields (Section 6.25).  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.

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   However, this specification does not define any standard for such
   automatic selection.

        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 6.25). 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 6.25). The duration of
   the redirection can be indicated through an  Expires (Section 6.20)

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

7.3.5 381 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 :

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

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        Some email and voice mail systems provide this
        functionality. A status code separate from 300 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 300 response, user intervention is
        required for a 381 response.

7.4 Request Failure 4xx

   4xx responses are definite failure responses from a particular
   server.  The client SHOULD NOT retry the same request without
   modification (e.g., adding appropriate authorization). However, the
   same request to a different server may be successful.

7.4.1 400 Bad Request

   The request could not be understood due to malformed syntax.

7.4.2 401 Unauthorized

   The request requires user authentication.

7.4.3 402 Payment Required

   Reserved for future use.

7.4.4 403 Forbidden

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

7.4.5 404 Not Found

   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

7.4.7 407 Proxy Authentication Required

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   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 6.29) 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 6.30). SIP access authentication is explained
   in section 12.3 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.

7.4.8 408 Request Timeout

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

7.4.9 412 Precondition Failed

   The precondition given in one or more of the request-header fields
   evaluated to false when it was tested on the server. Preconditions
   include  If-Match (Section 6.23) and If-None-Match (Section 6.24).

7.4.10 420 Bad Extension

   The server did not understand the protocol extension specified in a
   Require (Section 6.32) header field.

7.4.11 480 Temporarily Unavailable

   The callee's end system was contacted successfully but the callee is
   currently unavailable (e.g., not logged in or logged in in such a
   manner as to preclude communication with the callee). The response
   may indicate a better time to call in the  Retry-After header. The
   user may also be available elsewhere (unbeknownst to this host),
   thus, this response does not terminate any searches. The reason
   phrase SHOULD indicate the more precise cause as to why the callee is
   unavailable. This value SHOULD be setable by the user agent.

7.4.12 481 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.)

7.4.13 482 Loop Detected

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

7.4.14 483 Too Many Hops

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

7.4.15 484 Address Incomplete

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

        This status code allows overlapped dialing.

7.5 Server Failure 5xx

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

7.5.1 500 Server Internal Error

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

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.

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   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
   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

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   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 describing why the session described cannot be
   supported. These reasons can be one or more of:

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

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

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

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

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

   Other reasons are likely to be added later. It is hoped that
   negotiation will not frequently be needed, and when a new user is
   being invited to join an already existing conference, negotiation may
   not be possible. It is up to the invitation initiator to decide
   whether or not to act on a 606 (Not Acceptable) response.

8 SIP Message Body

8.1 Body Inclusion

   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

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   descriptions; for responses with status 300 or greater, the session
   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 MUST be given by the  Content-Length header
   field. If no body is present in a message, then the Content-Length
   header MUST set to zero. If a server receives a message without
   Content-Length, it MUST assume it to be zero.

   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.

   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

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   Thus the header in section 14.2 could also be written:

     INVITE schooler@vlsi.caltech.edu SIP/2.0
     v:SIP/2.0/UDP 16

     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 SIP Transport

10.1 General Remarks

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

10.1.1 Requests

   Stateful proxies mark outgoing requests with the  branch parameter in
   the  Via header.

   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.)

   If a stateful proxy, user agent or redirect server 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

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   MUST NOT issue provisional responses on their own.

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

10.1.2 Responses

   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
   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 Unicast UDP

   UDP packets MUST have a source address that is valid as a destination
   for requests and responses. Responses are returned to the address
   listed in the  Via header field (Section 6.43), not the source
   address of the request.

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10.3 Multicast UDP

   Requests may be multicast. Multicast requests SHOULD have a time-to-
   live value of no greater than one, i.e., be restricted to the local

   If the request was received via multicast, the response is also
   returned via multicast. The server delays its response by a random
   interval between zero and one second. Servers do not return 404 (Not
   Found) responses and SHOULD suppress responses if they hear a lower-
   numbered or 6xx response from another group member prior to sending.
   Servers do not respond to  CANCEL requests received via multicast.


   A SIP client SHOULD retransmit a  BYE,  CANCEL, or  OPTIONS 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 user location server
   again upon receiving a retransmission.

        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 1.5 seconds. 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
        seems less than helpful. 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 6.39), returned in the response. The
        server caches the result until it can be sure that the
        client will not retransmit the same request again.


   A client MAY repeat its registration attempts at intervals of 2, 4,
   8, ..., 512, 512, ... seconds if it receives no response.

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        Retransmitting REGISTER indefinitely ensures that a user
        will eventually be able to register after a registrar
        recovers from a crash. The period is chosen so that even on
        a large LAN, there will not be more than about one
        REGISTER request per second.

10.6  ACK

   The  ACK request does not generate responses. It is only
   retransmitted when a response to an  INVITE request arrives.

10.7  INVITE

   Special considerations apply for the  INVITE method.

        1.   After receiving an invitation, considerable time may elapse
             before the server can determine the outcome. For example,
             the called party may be "rung" or extensive searches may be
             performed, so delays between the request and a definitive
             response can reach several tens of seconds.  If either
             caller or callee are automated servers not directly
             controlled by a human being, a call attempt may be
             unbounded in time.

        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 (Trying) 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".

   A SIP client SHOULD retransmits a SIP  INVITE 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 T3. The default values of timer T1 and

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   T3 are 1 and 30 seconds, respectively. The default retransmit limit
   is 20.

        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.

   Upon receiving a final response, the client sends an  ACK to the
   address listed in the  Location header field contained in the
   response. The  To header field in the  ACK request assumes the value
   of the  Location header field. If the response did not contain a
   Location header, the client uses the same  To header field as for the
   INVITE request and sends the  ACK to the same destination as the
   original  INVITE request.

   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 MUSTNOT be
   acknowledged to prevent a response- ACK feedback loop.

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

        Using the mechanism in Sec. 10.4 does not work well for the
        long delays between  INVITE and a final response.  If the
        200 response gets 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.

   Blindly retransmitting the response increases the probability of
   success, but at the cost of significantly higher processing and
   network load.

10.8 TCP Connections

   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,

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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 8: State transition diagram of client for  INVITE method

   for example, multiple 200 responses may be generated.)

   After sending an  INVITE request, the client keeps the connection
   open until the first final response arrives. If that response has a

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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 9: State transition diagram of server for  INVITE method

   status code of 300 or larger, the client sends an  ACK. If the
   response status code is 2xx and the client is a user agent client, it
   sends an ACK. If the client is not a user agent, the response is

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   forwarded upstream.

   The client MAY close the connection at any time. 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 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 MUST be prepared to
   receive both requests and responses on a "passive" connection.

11 Behavior of SIP Servers

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

   A proxy server always inserts a  Via header field containing its own
   address into those requests that are caused by an incoming request.
   Each proxy MUST insert a " branch" parameter (Section 6.43). To
   prevent loops, a server MUST check if its own address is already
   contained in the  Via header of the incoming request.

   A proxy server MAY convert a version-x SIP request or response to a
   version-y request or response, where x may be larger, smaller or
   equal to y.

        This rule allows a proxy to serve as a go-between between
        two servers that have no version of the protocol in common.

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

   The detailed connection behavior for UDP and TCP is described in
   Section 10.

11.1 Redirect Server

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   A redirect server does not issue any SIP requests of its own. It can
   return a response that refuses or redirects the request. After
   receiving an  INVITE request, once the server has gathered the list
   of alternative locations or has decided to refuse the call, it
   returns the final response of class 3xx or 6xx. This ends the SIP
   transaction. The redirect server maintains transaction state for the
   whole SIP transaction.

11.2 User Agent Server

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

11.3 Stateless Proxy: Proxy Servers Issuing Single Unicast Requests

   Proxies in this category issue at most a single unicast request for
   each incoming SIP request, that is, they do not "fork" requests.
   However, servers may choose to always operate in a mode that allows
   issuing of several requests, as described in Section 11.4.

   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

11.4 Proxy Server Issuing Several  INVITE Requests

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

   All requests carry the same  Call-ID. For unicast, each of the
   requests has a different (host-dependent)  Request-URI. For
   multicast, a single request is issued, likely with a host-independent
   Request-URI. A client receiving a multicast query does not have to
   check whether the host part of the  Request-URI matches its own host
   or domain name. To avoid response implosion, servers MUST NOT answer
   multicast requests with a "404 Not Found" status code. Servers MAY
   decide not to answer multicast requests if their response would be
   5xx.  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 6.43).

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   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 6.33).

   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
   connection to client with address a. response(s, l, L) sends a
   response to the client with status s and list of locations L, with l
   entries. ismulticast() returns 1 if the location is a multicast
   address and zero otherwise.  The variable timeleft indicates the
   amount of time left until the maximum response time has expired. The
   variable recurse indicates whether the server will recursively try
   addresses returned through a 3xx response. A server MAY decide to
   recursively try only certain addresses, e.g., 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 best = 1000;      /* best response so far */
       int timeleft = 120;   /* sample timeout value */
       int loc = 0;          /* number of locations */
       struct {              /* response */
         int status;         /* response status; -2: BYE; -1: CANCEL */
         char *location;     /* redirect location */
         address_t a;        /* address of respondent */
         int branch;         /* branch identifier */
       } r;
       int i;

       for (i = 0; i < N; i++) {

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         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 (r.status < 0) {

         if (class >= 2) {
           for (i = 0; i < N; i++) {
             if (r.branch == outgoing.branch[i]) {
               outgoing[i].done = 1;

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

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

       /* CANCEL or BYE */
       if (r.status < 0) {
         response(200, loc, 0);

         /* BYE */
         if (r.status == -2) {
           for (i = 0; i < N; i++) {
             request(BYE, address[i], i);
       else {
         /* We haven't heard anything useful from anybody. */
         if (best == 1000) {
           best = 404;
         if (best/100 != 3) loc = 0;
         response(best, loc, locations);

        * Close the other pending transactions by sending CANCEL.
       if (r.status > -1) {
         for (i = 0; i < N; i++) {
           if (!outgoing[i].done) {
             request(CANCEL, address[i], outgoing[i].branch);
            if (tcp) close(a);

   Responses are processed as follows. The process completes when all
   requests have been answered by final status responses (for unicast)
   or 60 seconds have elapsed (for multicast). A proxy MAY send a

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   CANCEL to all branches and return a 408 (Timeout) to the client after
   120 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.

   3xx: The proxy MUST send an  ACK and MAY recurse on the listed
        Location addresses. Otherwise, the locations in the response are
        added to separate lists for 300, 301 and 302 responses
        maintained by the proxy. The lowest-numbered 300 response is
        returned to the client on completion.

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

   6xx: The proxy MUST forward the response to the client and send an
        ACK. Other pending requests SHOULD be terminated with CANCEL.

   The proxy server must maintain state until all responses have been
   received or for 60 seconds if the request was multicast.

   After receiving a 2xx or 6xx response, the server SHOULD 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 once.)

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

   If a proxy server receives a  BYE request for a pending search or the
   TCP connection initiating the search is closed by the upstream
   client, the proxy server SHOULD terminate all pending requests by
   sending a BYE request and closing the TCP connections, if applicable.

11.5 Proxy Server Issuing Several  ACK and  BYE Requests

   In most cases,  ACK and  BYE requests will bypass proxies and reach
   the desired party directly, keeping proxies from having to make

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   forwarding decisions.

   User agent clients respond to  ACK and  BYE requests with unknown
   Call-ID with status code 481 (Invalid Call-ID).

   A proxy MAY maintain call state for a period of its choosing. If a
   proxy still has list of destinations that it forwarded the last
   INVITE to, it SHOULD direct  ACK requests only to those downstream
   servers. It SHOULD direct  BYE to only those servers that had
   previously responded with 2xx or have not yet responded to the last

   If the proxy has no call state for a particular  Call-ID and To
   destination, it forwards the request to all downstream servers.

12 Security Considerations

12.1 Confidentiality and Privacy: Encryption

12.1.1 SIP Transactions

   SIP transactions 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 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 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.

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   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 6.22 describes how a sender can request that Via
   fields be encrypted by cooperating proxies without compromising the
   purpose of the Via field.

12.2 End-to-End Encryption

   End-to-end encryption relies on keys shared by the two user agents
   involved in the transaction. 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 3 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.

     INVITE watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP$

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     To: watson@bell-telephone.com (T. A. Watson)$
     From: a.g.bell@bell-telephone.com (A. Bell)$
     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 watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP$
     To: watson@bell-telephone.com (T. A. Watson)$
     From: a.g.bell@bell-telephone.com (A. Bell)$
     Encryption: PGP version=5.0$
     Content-Type: application/sdp$
     Content-Length: 107$
   * $                                             *
   * v=0$                                          *

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

12.2.1 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

12.2.2 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.

12.2.3 Hop-by-Hop Encryption

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

12.2.4 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 6.22) SHOULD encrypt the header field. As only the proxy that
   encrypts the field will decrypt it, the algorithm chosen is entirely
   up to the proxy implementor. Two methods satisfy these requirements:

        o The server keeps a cache of  Via 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.

12.3 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. In practice, the same
   cryptographic measures that are used to ensure the authenticity of
   the SIP message also serve to authenticate the originator of the

   Transport-layer or network-layer authentication may be used for hop-
   by-hop authentication. SIP also extends the HTTP WWW-Authenticate
   (Section 6.45 and Authorization (Section 6.11) header and their
   Proxy- counterparts to include cryptographically strong signatures.
   SIP also supports the HTTP "basic" authentication scheme [33] that
   offers a very rudimentary mechanism of ascertaining the identity of

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   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 end-to-end authenticate the
   request originator 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 13.

   To sign a SIP request, the order of the SIP header fields is
   important.  Via header fields MUST precede all other SIP header
   fields as these are modified in transit. When an  Authorization
   header field is present, it indicates that all the header fields
   following the Authorization header field have been included in the
   signature.  To sign a request, a client removes all of the SIP header
   from before where the  Authorization field will be added. It then
   prepends the request method (in upper case) followed by the SIP
   version number field (in upper case) directly to the start of the
   message with no whitespace, CR or LF characters inserted. This
   extended message is what is signed.

   For example, if the SIP request is to be:

   INVITE watson@boston.bell-telephone.com SIP/2.0
   Via: SIP/2.0/UDP
   Authorization: PGP version=5.0, signature=...
   From: a.g.bell@bell-telephone.com (A. Bell)
   To: watson@bell-telephone.com (T. A. Watson)
   Call-ID: 187602141351@worcester.bell-telephone.com
   Subject: Mr. Watson, come here.
   Content-Type: application/sdp
   Content-Length: ...

   o=bell 53655765 2353687637 IN IP4

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   c=IN IP4
   m=audio 3456 RTP/AVP 0 3 4 5

   Then the data block that is signed is:

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

   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

12.3.1 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.)

   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

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   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).

12.4 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.

12.5 Known Security Problems

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

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   process will repeat.

13 SIP Security Using PGP

13.1 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.

13.1.1 The  WWW-Authenticate Response Header

        WWW-Authenticate  ___   "WWW-Authenticate" ":" "pgp" pgp-challenge
        pgp-challenge     ___   1# ( 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.


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

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13.1.2 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   ___   1# (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.

   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.

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   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",

13.2 PGP Encryption Scheme

        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"

13.3  Response-Key Header Field for PGP

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

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

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

14 Examples

14.1 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:@bell-tel.com SIP/2.0
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         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.

   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
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Expire: 0
         Location: *

   C->S: REGISTER sip:@bell-tel.com SIP/2.0
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Location: sip:tawatson@example.com

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   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
         From: sip:jon.diligent@bell-tel.com
         To: sip:watson@bell-tel.com
         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.

14.2 Invitation to Multicast Conference

   The first example invites schooler@vlsi.cs.caltech.edu to a multicast
   session. All examples use the Session Description Protocol (SDP) (RFC
   2327 [7]) as the session description format.

14.2.1 Request

   C->S: INVITE sip:schooler@vlsi.cs.caltech.edu SIP/2.0
         Via: SIP/2.0/UDP 16
         Via: SIP/2.0/UDP
         Via: SIP/2.0/UDP
         From: Mark Handley <sip:mjh@isi.edu>
         To: Eve Schooler <sip:schooler@caltech.edu>
         Subject: SIP will be discussed, too
         Call-ID: 19971205T234505.56.78@oregon.isi.edu
         Content-Type: application/sdp
         CSeq: 4711
         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

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   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.

14.2.2 Response

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

   S->C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP 16 ;branch=17
         Via: SIP/2.0/UDP csvax.cs.caltech.edu ;branch=8348
         Via: SIP/2.0/UDP north.east.isi.edu
         To: Eve Schooler <sip:schooler@caltech.edu>
         From: Mark Handley <sip:mjh@isi.edu>
         Call-ID: 19971205T234505.56.78@north.east.isi.edu
         Location: sip:es@jove.cs.caltech.edu
         CSeq: 4711

   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.

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   S->C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP 16 ;branch=17
         Via: SIP/2.0/UDP csvax.cs.caltech.edu ;branch=8348
         Via: SIP/2.0/UDP north.east.isi.edu
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 19971205T234505.56.78@north.east.isi.edu
         Location: sip:es@jove.cs.caltech.edu
         CSeq: 4711

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

   C->S: ACK sip:es@jove.cs.caltech.edu SIP/2.0
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 19971205T234505.56.78@oregon.isi.edu
         CSeq: 4711

14.3 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: 1985853074@kensington.bell-tel.com
         Subject: Mr. Watson, come here.
         CSeq: 17
         Content-Type: application/sdp
         Content-Length: ...

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

   S->C: SIP/2.0 200 OK

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         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: sip:watson@bell-tel.com
         Call-ID: 1985853074@kensington.bell-tel.com
         CSeq: 17
         Location: sip:watson@boston.bell-tel.com
         Content-Length: ...

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

   Watson can only receive PCMU and GSM. Note that Watson's list of
   codecs may or may not be a subset of the one offered by Bell, as each
   party indicates the data types it is willing to receive. Watson will
   send audio data to port 3456 at, 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 agree on the set of media, Watson confirms
   the call without enclosing another session description:

   C->S: ACK sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com>
         Call-ID: 1985853074@kensington.bell-tel.com
         CSeq: 17
         Content-Length: 0

14.4 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
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. A. Watson <sip:watson@bell-tel.com>
         Call-ID: 1985853074@kensington.bell-tel.com
         CSeq: 18

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   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.

14.5 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 proxy.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
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 1985853074@kensington.bell-tel.com
         CSeq:    19
         Via:     SIP/2.0/UDP c.bell-tel.com

   The SIP server 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 proxy.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: 1985853074@c.bell-tel.com
         CSeq:    19

   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 proxy.ieee.org ;branch=1
         Via:     SIP/2.0/UDP c.bell-tel.com

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         ETag:    "4711"
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 1985853074@c.bell-tel.com
         CSeq: 19

   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 proxy.ieee.org ;branch=1
         If-Match: "4711"
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     19

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

   P->A: INVITE sip:watson@acm.org SIP/2.0
         Via:      SIP/2.0/UDP proxy.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:  1985853074@c.bell-tel.com
         CSeq:     19

   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 proxy.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: 1985853074@c.bell-tel.com
         CSeq:    19

   P->Y: INVITE sip:watson@y.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP proxy.ieee.org ;branch=4
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>

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         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 1985853074@c.bell-tel.com
         CSeq:    19

   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. The  ETag is not strictly necessary here,
   since the  Location header is unambiguous.

   X->P: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP proxy.ieee.org ;branch=3
         Via:      SIP/2.0/UDP c.bell-tel.com
         ETag:     "1620"
         Location: sip:t.watson@x.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     19

   Y->P: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP proxy.ieee.org ;branch=4
         Via:      SIP/2.0/UDP c.bell-tel.com
         ETag:     "2016"
         Location: sip:t.watson@y.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     19

   This response is 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 proxy.ieee.org ;branch=2
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 1985853074@c.bell-tel.com
         CSeq:    19

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   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 proxy.ieee.org ;branch=3
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 1985853074@c.bell-tel.com
         CSeq:    19

   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>
         If-Match: "1620"
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     19

   C->Y: ACK sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         If-Match: "2016"
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     19

   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
         If-Match: "2016"
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>

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         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     20

   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>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     20

14.6 Redirects

   Replies with status codes 301 (Moved Permanently) or 302 (Moved
   Temporarily) specify another location using the  Location field:

   S->C: SIP/2.0 302 Moved temporarily
         Via: SIP/2.0/UDP csvax.cs.caltech.edu ;branch=8348
         Via: SIP/2.0/UDP
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 3779067998@oregon.isi.edu
         Location: sip:@;ttl=16;transport=udp
         CSeq: 19
         Content-Length: 0

   In this example, the proxy located at csvax.cs.caltech.edu is being
   advised to contact the multicast group with a ttl of
   16 and UDP transport. In normal situations, a server would not
   suggest a redirect to a local multicast group unless, as in the above
   situation, it knows that the previous proxy or client is within the
   scope of the local group. If the request is redirected to a multicast
   group, a proxy server SHOULD query the multicast address itself
   rather than sending the redirect back towards the client as multicast
   may be scoped; this allows a proxy within the appropriate scope
   regions to make the query.

14.7 Alternative Services

   An example of a 350 (Alternative Service) response is:

   S->C: SIP/2.0 350 Alternative Service
         Via: SIP/2.0/UDP

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         Via: SIP/2.0/UDP
         From: sip:mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 19971205T234505.56.78@oregon.isi.edu
         Location: recorder@
         CSeq: 19
         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:mm-server@ SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 19971205T234505.56.78@
         CSeq: 20
         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

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

14.8 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
         Call-ID: 19971205T234505.56.78@
         Location: 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
   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.

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

   C->S: OPTIONS sip:bob@example.com SIP/2.0
         From: Alice <sip:alice@anywhere.org>

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         To: Bob <sip:bob@example.com>
         Accept: application/sdp

   S->C: SIP/2.0 200 OK
         Location: sip:bob@host.example.com ;service=IP,voice-mail
                   ;media=audio ;duplex=full ;q=0.7
         Location: phone:+1-415-555-1212 ; service=ISDN;mobility=fixed;
                   language=en,es,iw ;q=0.5
         Location: phone:+1.800.555.1212 ; service=pager;mobility=mobile;
                   duplex=send-only;media=text; q=0.1

   Alternatively, Bob could have returned a description of

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

   S->C: SIP/2.0 200 OK
         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
   and MUST be able to include the  Call-ID, Content-Length,  Content-
   Type,  CSeq, From and  To headers. A minimal implementation MUST
   understand SDP (RFC 2327, [7]). In responses, it must be able to
   parse the  Call-ID,  Content-Length, Content-Type,  Require headers.
   It MUST be able to recognize the status code classes 1 through 6 and
   act accordingly.

   The following capability sets build on top of a minimal

   Basic: A basic implementation SHOULD add support for the BYE method
        to allow the interruption of a pending call attempt. It SHOULD

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        include a  User-Agent header in its requests and indicate its
        preferred language in the  Accept-Language header.

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

   Negotiation: A client MUST be able to request the  OPTIONS method and
        understand the 380 (Alternative Service) status and the Location
        parameters to participate in terminal and media negotiation. It
        SHOULD be able to parse the  Warning response header to provide
        useful feedback to the caller.

   Authentication: If a client wishes to invite callees that require
        caller authentication, it must be able to recognize the 401
        (Unauthorized) status code, must be able to generate the
        Authorization request header and 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. It MUST parse and generate,
   as appropriate, the  Call-ID,  Content-Length, Content-Type,
   Expires,  From, Max-Forwards,  Require,  To and  Via headers. It MUST
   echo the  CSeq header in the response. It SHOULD include the  Server
   header in its responses.

A.3 Header Processing

   Table 4 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
   descriptions appropriately.

   It is assumed that if caller or callee include a particular media

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   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
   as sendonly wants to neither receiver nor send a particular media
   type, it should set the port to zero. (RTCP ports are not needed in
   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 [24]. 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;

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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        o       m       m
           Date                     g        o       o       o
           Encryption               g        e       -       e
           Expires                  g        -       o       o
           From                     R        m       o       m
           Hide                     R        o       o       -
           Location                 R        -       -       -
           Location                 r        r       r       -
           Max-Forwards             R        -       b       -
           Organization             R        -       o       o
           Proxy-Authenticate      407       a       -       -
           Proxy-Authorization      R        -       a       -
           Proxy-Require            R        -       m       -
           Priority                 R        -       o       o
           Require                  R        m       -       m
           Retry-After           600,603     o       o       -
           Response-Key             R        -       -       e
           Server                   r        o       o       -
           Subject                  R        o       o       o
           Timestamp                g        o       o       o
           To                       g        m       m       m
           Unsupported              r        b       b       -
           User-Agent               R        -       o       o
           Via                      g        -       m       m
           Warning                  r        o       o       -
           WWW-Authenticate        401       a       -       -

   Table 4: This table indicates which systems should be able  to  parse
   which  response  header fields. Type is as table 3. "-" 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.
        a comma separated list of  term containing at least 2 items;

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   Common Tokens

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

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

D IANA Considerations

   Section 4.4 describes a name space and mechanism for registering SIP

   Changes in Version -05

   Since version -04, the following changes have been made.

        o Local part of  Call-ID needs to be cryptographically random.

        o Added  Response-Key,  Encryption,  Hide and "pgp"
          authentication method.

        o Removed  Public, following the revised HTTP/1.1 [30];  Allow
          can do the same thing.

        o Updated phone URLs to [22].

        o Proxy-Require header added, as in RTSP.

        o "Emergency" priority value added.

        o 381 (Ambiguous) status code added.

        o Sequence numbers should be contiguous to allow loss detection.
          Limited to 64 bits to allow numerical computations.

        o The service definitions have been moved to a separate

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        o The  Also header field and  Location parameters have moved to
          a separate document.

        o Call-ID defined consistently, with behavior for several calls
          with the same Call-ID and several calls with different Call-
          IDs for the same conference.

        o UNREGISTER was removed since  REGISTER with an expiration time
          of zero accomplishes the same thing.

        o Max-Forwards header and status 483 added.

        o The default port number 5060 has been assigned to SIP. The
          address also has been assigned as the "all SIP
          servers" multicast address.

        o Added table with list of headers used by each SIP
          server/client type.

        o CSeq mandatory for requests and minimal implementations.

        o Proxy search rules made more precise.

        o Added ability of (state-ful) proxies to request to see  BYE
          and ACK packets via  Record-Route and  Route.

        o Retransmission timer for  INVITE changed to T3 = 30 seconds to
          avoid multiple  INVITEs for a typical phone call.

        o ACKs may contain session descriptions.

        o signed-by parameter added to  Authorization header.

        o INVITE and  ACK requests may also contain  Location so that
          the callee can send the  BYE directly.

        o OPTIONS allowed to return current status such as Busy or

        o Unicast example clarified.

        o Added section on SDP usage for unicast. Currently, require
          "lining up" media types.

        o Tentatively added  CANCEL request to terminate searches.

        o Status 181: Call Is Being Forwarded.

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        o Content-Encoding for compressed payloads.

        o Retry-After may be used for  REGISTER.

        o ETag,  If-Match,  If-None-Match

        o Added  method to  CSeq response header.

        o Folded in Yaron Goland's "nit" comments: URL phone now
          conforms to "URLs for Telephony" I-D; production of password;
          port number; short SIP URLs removed [controversial?].

        o Cleaned up  From,  To,  Request-URI throughout spec to always
          include full SIP URL. (Otherwise, there is an ambiguity with
          users named "sip", since the user name can contain a colon.
          Search Four11 if you don't believe that the name exists...)

        o Warning header syntax fixed to allow 606.3.

        o signed-by can be general URI. However, not clear yet whether
          URI schemes are the most appropriate mechanism to denote a
          principal. In X.509 certificates, email addresses are used.

E Open Issues

E.1 H.323

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

   Solution: Leave to separate document.

   Status: Closed.


   Problem: How does the UA get a "personalized" multicast address for
        multicast searches? It would be helpful if the server could
        return an indication of any local search multicast address the
        user agent is supposed to listen on. The server might also want
        to indicate whether all outgoing calls should be proxied through
        the server. Use of message bodies for  REGISTER requests.

   Solution: Leave for follow-on work. Possible to use Location with 200
        response, but already otherwise used to indicate registrations.

   Status: Closed.

E.3 Max-Forwards

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   Problem: Extend  Max-Forwards with a max fan out field.  [Not really
        related.] How do you limit fanout for multicast? What's the
        advantage of doing this?

   Solution: Await operational experience as to whether users can
        actually set this value to anything meaningful.

   Status: Closed.

E.4 Cancellation and BYE

   Problem: Determine how to cancel

        - searches

        - ringing

        - non-selected end systems

        - calls

   Solution: In spec.

   Status: Closed.

E.5 IPv6 URLs

   Problem: Numeric IPv6 addresses in URLs.

   Solution: None; wait for others to take the lead. Note that this
        affects only numeric addresses, which should be rarely used.

   Status: Defered.

F Acknowledgments

   We wish to thank the members of the IETF MMUSIC WG for their comments
   and suggestions. Detailed comments were provided by Yaron Goland,
   Christian Huitema, Jonathan Lennox, Jonathan Rosenberg, and Moshe J.

   This work is based, inter alia, on [34,35].

G Authors' Addresses

   Mark Handley
   USC Information Sciences Institute
   c/o MIT Laboratory for Computer Science

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   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

H Bibliography

   [1] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
   Feb. 1998.  Work in progress.

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

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

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

   [5] B. Braden, L. Zhang, S. Berson, S. Herzog, and S. Jamin,
   "Resource ReSerVation protocol (RSVP) -- version 1 functional
   specification," RFC 2205, Internet Engineering Task Force, Oct. 1997.

   [6] 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.

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

   [8] P. Lantz, "Usage of H.323 on the Internet," Internet Draft,

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   Internet Engineering Task Force, Feb. 1997.  Work in progress.

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

   [10] 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.

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

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

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

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

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

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

   [17] 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.

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

   [19] L. Masinter, P. Hoffman, and J. Zawinski, "The mailto URL
   scheme," Internet Draft, Internet Engineering Task Force, Jan. 1998.
   Work in progress.

   [20] 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.

   [21] T. Berners-Lee, L. Masinter, and R. Fielding, "Uniform resource

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   identifiers (URI): generic syntax," Internet Draft, Internet
   Engineering Task Force, Mar. 1998.  Work in progress.

   [22] A. Vaha-Sipila, "URLs for telephony," Internet Draft, Internet
   Engineering Task Force, Feb. 1998.  Work in progress.

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

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

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

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

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

   [28] 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.

   [29] P. Resnick, "Internet message format standard," Internet Draft,
   Internet Engineering Task Force, Mar. 1998.  Work in progress.

   [30] 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

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

   [32] M. Elkins, "MIME security with pretty good privacy (PGP)," RFC
   2015, Internet Engineering Task Force, Oct. 1996.

   [33] J. Franks, E. Sink, P. Leach, J. Hostetler, P. Hallam-Baker, L.
   Stewart, S. Lawrence, and A. Luotonen, "HTTP authentication: Basic
   and digest access authentication," Internet Draft, Internet
   Engineering Task Force, Mar. 1998.  Work in progress.

   [34] E. M. Schooler, "Case study: multimedia conference control in a

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   packet-switched teleconferencing system," Journal of Internetworking:
   Research and Experience , vol. 4, pp. 99--120, June 1993.  ISI
   reprint series ISI/RS-93-359.

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

   Full Copyright Statement

   Copyright (c) The Internet Society (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

   1          Introduction ........................................    2
   1.1        Overview of SIP Functionality .......................    2
   1.2        Terminology .........................................    3
   1.3        Definitions .........................................    4

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   1.4        Summary of SIP Operation ............................    6
   1.4.1      SIP Addressing ......................................    7
   1.4.2      Locating a SIP Server ...............................    8
   1.4.3      SIP Transaction .....................................    9
   1.4.4      SIP Invitation ......................................   10
   1.4.5      Locating a User .....................................   13
   1.4.6      Changing an Existing Session ........................   13
   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 ................................   17
   4          Request .............................................   19
   4.1        Request-Line ........................................   19
   4.2        Methods .............................................   21
   4.2.1       INVITE .............................................   21
   4.2.2       ACK ................................................   21
   4.2.3       OPTIONS ............................................   22
   4.2.4       BYE ................................................   22
   4.2.5       CANCEL .............................................   22
   4.2.6       REGISTER ...........................................   23
   4.3        Request-URI .........................................   25
   4.3.1      SIP Version .........................................   25
   4.4        Option Tags .........................................   25
   4.4.1      Registering New Option Tags with IANA ...............   26
   5          Response ............................................   26
   5.1        Status-Line .........................................   27
   5.1.1      Status Codes and Reason Phrases .....................   27
   6          Header Field Definitions ............................   30
   6.1        General Header Fields ...............................   31
   6.2        Entity Header Fields ................................   31
   6.3        Request Header Fields ...............................   31
   6.4        Response Header Fields ..............................   32
   6.5        End-to-end and Hop-by-hop Headers ...................   32
   6.6        Header Field Format .................................   32
   6.7         Accept .............................................   34
   6.8         Accept-Encoding ....................................   34
   6.9         Accept-Language ....................................   34
   6.10        Allow ..............................................   34
   6.11        Authorization ......................................   35
   6.12        Call-ID ............................................   35
   6.13        Content-Encoding ...................................   36
   6.14        Content-Length .....................................   36
   6.15        Content-Type .......................................   37
   6.16        CSeq ...............................................   37
   6.17        Date ...............................................   38

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   6.18        Encryption .........................................   39
   6.19        ETag ...............................................   40
   6.20        Expires ............................................   41
   6.21        From ...............................................   42
   6.22        Hide ...............................................   42
   6.23        If-Match ...........................................   43
   6.24        If-None-Match ......................................   44
   6.25        Location ...........................................   44
   6.26        Max-Forwards .......................................   46
   6.27        Organization .......................................   47
   6.28        Priority ...........................................   47
   6.29        Proxy-Authenticate .................................   48
   6.30        Proxy-Authorization ................................   48
   6.31        Proxy-Require ......................................   48
   6.32        Require ............................................   49
   6.33        Record-Route .......................................   50
   6.34        Response-Key .......................................   50
   6.35        Route ..............................................   51
   6.36        Retry-After ........................................   51
   6.37        Server .............................................   52
   6.38        Subject ............................................   52
   6.39        Timestamp ..........................................   52
   6.40        To .................................................   53
   6.41        Unsupported ........................................   53
   6.42        User-Agent .........................................   53
   6.43        Via ................................................   53
   6.43.1     Requests ............................................   54
   6.43.2     Receiver-tagged Via Fields ..........................   54
   6.43.3     Responses ...........................................   55
   6.43.4     Syntax ..............................................   55
   6.44        Warning ............................................   56
   6.45        WWW-Authenticate ...................................   57
   7          Status Code Definitions .............................   58
   7.1        Informational 1xx ...................................   58
   7.1.1      100 Trying ..........................................   58
   7.1.2      180 Ringing .........................................   58
   7.1.3      181 Call Is Being Forwarded .........................   58
   7.2        Successful 2xx ......................................   59
   7.2.1      200 OK ..............................................   59
   7.3        Redirection 3xx .....................................   59
   7.3.1      300 Multiple Choices ................................   59
   7.3.2      301 Moved Permanently ...............................   60
   7.3.3      302 Moved Temporarily ...............................   60
   7.3.4      380 Alternative Service .............................   60
   7.3.5      381 Ambiguous .......................................   60
   7.4        Request Failure 4xx .................................   61
   7.4.1      400 Bad Request .....................................   61
   7.4.2      401 Unauthorized ....................................   61

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   7.4.3      402 Payment Required ................................   61
   7.4.4      403 Forbidden .......................................   61
   7.4.5      404 Not Found .......................................   61
   7.4.6      405 Method Not Allowed ..............................   61
   7.4.7      407 Proxy Authentication Required ...................   61
   7.4.8      408 Request Timeout .................................   62
   7.4.9      412 Precondition Failed .............................   62
   7.4.10     420 Bad Extension ...................................   62
   7.4.11     480 Temporarily Unavailable .........................   62
   7.4.12     481 Invalid Call-ID .................................   62
   7.4.13     482 Loop Detected ...................................   62
   7.4.14     483 Too Many Hops ...................................   63
   7.4.15     484 Address Incomplete ..............................   63
   7.5        Server Failure 5xx ..................................   63
   7.5.1      500 Server Internal Error ...........................   63
   7.5.2      501 Not Implemented .................................   63
   7.5.3      502 Bad Gateway .....................................   63
   7.5.4      503 Service Unavailable .............................   63
   7.5.5      504 Gateway Timeout .................................   64
   7.5.6      505 Version Not Supported ...........................   64
   7.6        Global Failures 6xx .................................   64
   7.6.1      600 Busy ............................................   64
   7.6.2      603 Decline .........................................   64
   7.6.3      604 Does Not Exist Anywhere .........................   64
   7.6.4      606 Not Acceptable ..................................   64
   8          SIP Message Body ....................................   65
   8.1        Body Inclusion ......................................   65
   8.2        Message Body Type ...................................   66
   8.3        Message Body Length .................................   66
   9          Compact Form ........................................   66
   10         SIP Transport .......................................   67
   10.1       General Remarks .....................................   67
   10.1.1     Requests ............................................   67
   10.1.2     Responses ...........................................   68
   10.2       Unicast UDP .........................................   68
   10.3       Multicast UDP .......................................   69
   10.4        BYE,  CANCEL,  OPTIONS .............................   69
   10.5        REGISTER ...........................................   69
   10.6        ACK ................................................   70
   10.7        INVITE .............................................   70
   10.8       TCP Connections .....................................   71
   11         Behavior of SIP Servers .............................   74
   11.1       Redirect Server .....................................   74
   11.2       User Agent Server ...................................   75
   11.3       Stateless Proxy: Proxy Servers Issuing Single
   Unicast Requests ...............................................   75
   11.4       Proxy Server Issuing Several  INVITE Requests .......   75
   11.5       Proxy Server Issuing Several  ACK and  BYE

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   Requests .......................................................   79
   12         Security Considerations .............................   80
   12.1       Confidentiality and Privacy: Encryption .............   80
   12.1.1     SIP Transactions ....................................   80
   12.2       End-to-End Encryption ...............................   81
   12.2.1     Privacy of SIP Responses ............................   83
   12.2.2     Encryption by Proxies ...............................   83
   12.2.3     Hop-by-Hop Encryption ...............................   83
   12.2.4     Via field encryption ................................   83
   12.3       Message Integrity and Access Control:
   Authentication .................................................   84
   12.3.1     Trusting responses ..................................   86
   12.4       Callee Privacy ......................................   87
   12.5       Known Security Problems .............................   87
   13         SIP Security Using PGP ..............................   88
   13.1       PGP Authentication Scheme ...........................   88
   13.1.1     The  WWW-Authenticate Response Header ...............   88
   13.1.2     The  Authorization Request Header ...................   89
   13.2       PGP Encryption Scheme ...............................   90
   13.3        Response-Key Header Field for PGP ..................   90
   14         Examples ............................................   91
   14.1       Registration ........................................   91
   14.2       Invitation to Multicast Conference ..................   92
   14.2.1     Request .............................................   92
   14.2.2     Response ............................................   93
   14.3       Two-party Call ......................................   94
   14.4       Terminating a Call ..................................   95
   14.5       Forking Proxy .......................................   96
   14.6       Redirects ...........................................  100
   14.7       Alternative Services ................................  100
   14.8       Negotiation .........................................  102
   14.9        OPTIONS Request ....................................  102
   A          Minimal Implementation ..............................  103
   A.1        Client ..............................................  103
   A.2        Server ..............................................  104
   A.3        Header Processing ...................................  104
   B          Usage of SDP ........................................  104
   C          Summary of Augmented BNF ............................  105
   D          IANA Considerations .................................  107
   E          Open Issues .........................................  109
   E.1        H.323 ...............................................  109
   E.2        REGISTER ............................................  109
   E.3        Max-Forwards ........................................  109
   E.4        Cancellation and BYE ................................  110
   E.5        IPv6 URLs ...........................................  110
   F          Acknowledgments .....................................  110
   G          Authors' Addresses ..................................  110
   H          Bibliography ........................................  111

Handley/Schulzrinne/Schooler                                [Page 118]

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Handley/Schulzrinne/Schooler                                [Page 119]

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