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

Internet Engineering Task Force                                MMUSIC WG
Internet Draft                    Handley/Schulzrinne/Schooler/Rosenberg
ietf-mmusic-sip-07.txt                ISI/Columbia U./Caltech/Bell Labs.
July 16, 1998
Expires: December 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.


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

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

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

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

1 Introduction

1.1 Overview of SIP Functionality

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

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

   SIP transparently supports name mapping and redirection services,
   allowing the implementation of ISDN and Intelligent Network telephony
   subscriber services. These facilities also enable personal mobility
   services, this is defined as: "Personal mobility is the ability of
   end users to originate and receive calls and access subscribed
   telecommunication services on any terminal in any location, and the
   ability of the network to identify end users as they move. Personal
   mobility is based on the use of a unique personal identity (i.e.,
   mobility complements terminal mobility, i.e., the ability to maintain
   communications when moving a single end system from one subnet to

   SIP supports five facets of establishing and terminating multimedia

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

   User capabilities: determination of the media and media parameters to

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        be used;

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

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

   Call handling: including transfer and termination of calls.

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

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

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

   In another example, it may be used to determine that the callee is
   reachable via the public switched telephone network (PSTN) and
   indicate the phone number to be called, possibly suggesting an
   Internet-to-PSTN gateway to be used.

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

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

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   service guarantees, if required, may depend on knowing the full
   membership of the session; this information may or may not be known
   to the agent performing session invitation.

1.2 Terminology

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

1.3 Definitions

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

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

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

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

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

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

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

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

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

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

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

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

   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

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        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
        server and may offer location services.

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

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

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

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

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

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

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

   User agent server (UAS), called user agent: A user agent server is a
        server application that contacts the user when a SIP request is

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        received and that returns a response on behalf of the user. The
        response may accept, reject or redirect the request.

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

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

   Table 1: Properties of the different SIP server types

1.4 Summary of SIP Operation

   This section explains the basic protocol functionality and operation.
   Callers and callees are identified by SIP addresses, described in
   Section 1.4.1. When making a SIP call, a caller first locates the
   appropriate server (Section 1.4.2) and then sends a SIP request
   (Section 1.4.3). The most common SIP operation is the invitation
   (Section 1.4.4). Instead of directly reaching the intended callee, a
   SIP request may be redirected or may trigger a chain of new SIP
   requests by proxies (Section 1.4.5). Users can register their
   location(s) with SIP servers (Section 4.2.6).

1.4.1 SIP Addressing

   The "objects" addressed by SIP are users at hosts, identified by a
   SIP URL. The SIP URL takes the form similar to a mailto or telnet
   URL, i.e., user@host 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 [9]), MX (RFC 974 [10], CNAME or A record (RFC 1035
   [11]), or a numeric network address.

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

   Examples of SIP URLs include:


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

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

1.4.2 Locating a SIP Server

   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

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   the SIP server at that address. Otherwise, the client follows the
   steps below.

        1.   If there is a SRV DNS resource record (RFC 2052 [9]) of
             type  sip.udp, contact the listed SIP servers in the order
             of the preference values contained in those resource
             records, using UDP as a transport protocol at the port
             number given in the URL or, if none provided, the one
             listed in the DNS resource record.

        2.   If there is a SRV DNS resource record (RFC 2052 [9]) of
             type  sip.tcp, contact the listed SIP servers in the order
             of the preference value contained in those resource
             records, using TCP as a transport protocol at the port
             number given in the URL or, if none provided, the one
             listed in the DNS resource record.

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

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

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

   A client MAY cache the result of the reachability steps for a
   particular address and retry that host address for the next 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 [12]).

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)

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   together with the responses triggered by that request make up a SIP
   transaction.  The  ACK request following an  INVITE is not part of
   the transaction since it may traverse a different set of hosts.

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

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

   The SIP message format and operation is independent of the transport

1.4.4 SIP Invitation

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

   The  INVITE request typically contains a session description, for
   example written in SDP (RFC 2327 [5]) format, that provides the
   called party with enough information to join the session. For
   multicast sessions, the session description enumerates the media
   types and formats that may be distributed to that session. For a
   unicast session, the session description enumerates the media types
   and formats that the caller is willing to receive and where it wishes
   the media data to be sent. In either case, if the callee wishes to
   accept the call, it responds to the invitation by returning a similar
   description listing the media it wishes to receive. For a multicast
   session, the callee should only return a session description if it is
   unable to receive the media indicated in the caller's description or
   wants to receive data via unicast.

   The protocol exchanges for the  INVITE method are shown in Fig. 1 for
   a proxy server and in Fig. 2 for a redirect server. (Note that the
   messages shown in the figures have been abbreviated slightly.) In
   Fig. 1, the proxy server accepts the  INVITE request (step 1),
   contacts the location service with all or parts of the address (step

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   2) and obtains a more precise location (step 3). The proxy server
   then issues a SIP  INVITE request to the address(es) returned by the
   location service (step 4).  The user agent server alerts the user
   (step 5) and returns a success indication to the proxy server (step
   6). The proxy server then returns the success result to the original
   caller (step 7). The receipt of this message is confirmed by the
   caller using an  ACK request, which is forwarded to the callee (steps
   8 and 9). 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

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   server (step 6). In the example, the call succeeds (step 7). The
   caller and callee complete the handshake with an  ACK (step 8).

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

1.4.5 Locating a User

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

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

   If a proxy server forwards a SIP request, it MUST add itself to the
   end of the list of forwarders noted in the  Via (Section 6.40)
   headers. The  Via trace ensures that replies can take the same path
   back, ensuring correct operation through compliant firewalls and
   avoiding request loops. On the response path, each host MUST remove
   its  Via, so that routing internal information is hidden from the
   callee and outside networks. 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

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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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   user agent MUST return the appropriate status response. Duplicate
   requests are not an error.

1.4.6 Changing an Existing Session

   In some circumstances, it may be necessary to change the parameters
   of an existing session. For example, two parties may have been
   conversing and then want to add a third party, switching to multicast
   for efficiency. One of the participants invites the third party with
   the new multicast address and simultaneously sends an  INVITE to the
   second party, with the new multicast session description, but with
   the old call identifier.

1.4.7 Registration Services

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

1.5 Protocol Properties

1.5.1 Minimal State

   A single conference session or call may involve one or more SIP
   request-response transactions. Proxy servers do not have to keep
   state for a particular call, however, they MAY maintain state for a
   single SIP transaction, as discussed in Section 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) (RFC 2326 [4]).

   When TCP is used, SIP can use one or more connections to attempt to

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

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

1.5.3 Text-Based

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

2 SIP Uniform Resource Locators

   SIP URLs are used within SIP messages to indicate the originator (
   From), current destination ( Request-URI) and final recipient ( To)
   of a SIP request, and to specify redirection addresses ( Location). A
   SIP URL can also be embedded in web pages or other hyperlinks to
   indicate that a user or service may be called.

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

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

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

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

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

   Figure 3: SIP URL syntax

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

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

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     telephone-subscriber  = global-phone-number | local-phone-number
     global-phone-number   = "+" 1*phonedigit [isdn-subaddress]
     local-phone-number    = 1*(phonedigit | dtmf-digit |
                               pause-character) [isdn-subaddress]
     isdn-subaddress       = ";isub=" 1*phonedigit
     post-dial             = ";postd=" 1*(phonedigit | dtmf-digit
                           |  pause-character)
     phonedigit            = DIGIT | visual-separator
     visual-separator      = "-" | "."
     pause-character       = one-second-pause | wait-for-dial-tone
     one-second-pause      = "p"
     wait-for-dial-tone    = "w"
     dtmf-digit            = "*" | "#" | "A" | "B" | "C" | "D"

   Figure 4: SIP URL syntax; telephone subscriber

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

   If a server handles SIP addresses for another domain, it MUST URL-
   encode the "@" character (%40).

   URL parameters: 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.

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   Transport parameters MUST NOT be used in the  From and To header
   fields and the  Request-URI; they are ignored if present.

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

   Tag: The  tag parameter allows several instances of a user that share
        the same  host and  port values to be distinguished from each
        other, for example, where the  host designates a firewall or
        proxy. The  tag value is a random string consisting of hex
        digits. The use of version-1 (time-based) or version-4 (random)
        UUID [19] is OPTIONAL.  The  tag value is designed to be
        globally unique and cryptographically random with at least 32
        bits of randomness. It SHOULD NOT be included in long-lived SIP
        URLs, e.g., those found on web pages or user databases. A single
        user maintains the same tag throughout the call identified by
        the  Call-ID. The tag parameter in  To headers is ignored when
        matching responses to requests that did not contain a  tag in
        their To header. (See Section 6.37.)

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

                       Request-URI     To     From     Location    external
   user               x               x      x        x            x
   password                           x               x            x
   host               x               x      x        x            x
   tag                x               x      x        x
   headers                                            x            x
   transport para.                                    x            x

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

   Examples of 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.

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

   In some circumstances a non-SIP URL may be used in a SIP message. An
   example might be making a call from a telephone which is relayed by a
   gateway onto the internet as a SIP request. In such a case, the
   source of the call is really the telephone number of the caller, and
   so a SIP URL is inappropriate and a phone URL might be used instead.
   To allow for this flexibility, SIP headers that specify user
   addresses allow these addresses to be SIP and non-SIP URLs.

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

3 SIP Message Overview

   SIP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [20]). Lines are terminated by CRLF, but
   receivers should be prepared to also interpret CR and LF by
   themselves as line terminators.

   Except for the above difference in character sets, much of the
   message syntax is identical to HTTP/1.1; rather than repeating it
   here we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1
   specification (RFC 2068 [7]). In addition, we describe SIP in both
   prose and an augmented Backus-Naur form (BNF) [H2.1] described in
   detail in RFC 2234 [21].

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

        The 1400 bytes accommodates lower-layer packet headers

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

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

        SIP-message  ___   Request | Response

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

        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

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

   Table 3: SIP headers

   ignored. In other words, if the  Request or  Response message begins
   with a  CRLF,  CR, or  LF, these characters should be ignored.

4 Request

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   The  Request message format is shown below:

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

4.1 Request-Line

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

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

4.2 Methods

   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.

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   A server MAY automatically respond to an invitation for a conference
   the user is already participating in, identified either by the SIP
   Call-ID or a globally unique identifier within the session
   description, with a 200 (OK) response.

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

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

4.2.2  ACK

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

   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

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

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

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   600 (Busy).

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

4.2.4  BYE

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

   If the  INVITE request contained a  Location header, the callee 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 redirect and user agent SIP servers.

4.2.5  CANCEL

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

   A user agent client or proxy client MAY issue a  CANCEL request at
   any time. A proxy, in particular, MAY choose to send a  CANCEL to
   destinations that have not yet returned a final response after it has
   received a 2xx or 6xx response for one or more of the parallel-search
   requests. A proxy that receives a  CANCEL request forwards the
   request to all destinations with pending requests. The Call-ID,  To
   and  From in the  CANCEL request are identical to those contained in
   the request being canceled, but the Via header field is initialized
   to the proxy issuing the CANCEL request. (Thus, responses to this
   CANCEL request only reach the issuing proxy.)

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

   A redirect 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

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        then terminate the call.  To terminate a call instead of
        just pending searches, the UAC must use BYE instead of or
        in addition to  CANCEL. While  CANCEL can terminate any
        pending request other than  ACK or  CANCEL, it is typically
        useful only for  INVITE. 200 responses to  INVITE and 200
        responses to  CANCEL are distinguished by the method in the
        Cseq header field, so there is no ambiguity.

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


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

   A user agent 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 [16]; however,
   they do not respond to the request.

   The  REGISTER request-header fields are defined as follows. We define
   "address-of-record" as the SIP address that the registry knows the
   registrand, typically of the form "user@domain" rather than
   "user@host". In third-party registration, the entity issuing the
   request is different from the entity being registered.

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

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

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

   Location: The request MUST contain a Location header field; requests

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

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

   Registrations are additive, but all current locations must share the
   same  action value. A proxy server ignores the  q parameter, while a
   redirect server simply returns the parameter in its Location header.

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

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

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

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

   A client cancels an existing registration by sending a  REGISTER
   request with an expiration time ( Expires) of zero seconds for a
   particular  Location or the wildcard  Location designated by a "*"
   for all registrations.

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

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

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

   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.

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

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

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

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

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   that handles outgoing calls. This mode of operation similar to that
   of HTTP proxies.

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

4.3.1 SIP Version

   Both request and response messages include the version of SIP in use,
   and basically follow [H3.1], with HTTP replaced by SIP. To be
   conditionally compliant with this specification, applications sending
   SIP messages MUST include a  SIP-Version of "SIP/2.0".

4.4 Option Tags

   Option tags are unique identifiers used to designate new options in
   SIP.  These tags are used in  Require (Section 6.30) and Unsupported
   (Section 6.38) fields.


        option-tag  ___   1*uric

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

4.4.1 Registering New Option Tags with IANA

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

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

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

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        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
   HTTP codes.

5.1 Status-Line

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

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

5.1.1 Status Codes and Reason Phrases

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

        Status-Code       =    Informational                  Fig. 5
                         |     Success                        Fig. 5
                         |     Redirection                    Fig. 6
                         |     Client-Error                   Fig. 7
                         |     Server-Error                   Fig. 8
                         |     Global-Failure                 Fig. 9
                         |     extension-code
        extension-code    =    3DIGIT
        Reason-Phrase     =    *<TEXT,  excluding CR, LF>

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

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

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

   3xx: Redirection -- further action must be taken in order to complete
        the request;

   4xx: Client Error -- the request contains bad syntax or cannot be
        fulfilled at this server;

   5xx: Server Error -- the server failed to fulfill an apparently valid

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

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

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   defined HTTP response codes, and adds a new class, 6xx, of response

   SIP response codes are extensible. SIP applications are not required
   to understand the meaning of all registered response codes, though
   such understanding is obviously desirable. However, applications MUST
   understand the class of any response code, as indicated by the first
   digit, and treat any unrecognized response as being equivalent to the
   x00 response code of that class, with the exception that an
   unrecognized response MUST NOT be cached. For example, if a client
   receives an unrecognized response code of 431, it can safely assume
   that there was something wrong with its request and treat the
   response as if it had received a 400 (Bad Request) response code. In
   such cases, user agents SHOULD present to the user the message body
   returned with the response, since that message body is likely to
   include human-readable information which will explain the unusual

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

   Figure 5: Informational and success status codes

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

   Figure 6: Redirection status codes

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        Client-Error    =    "400"    ;  Bad Request
                       |     "401"    ;  Unauthorized
                       |     "402"    ;  Payment Required
                       |     "403"    ;  Forbidden
                       |     "404"    ;  Not Found
                       |     "405"    ;  Method Not Allowed
                       |     "406"    ;  Not Acceptable
                       |     "407"    ;  Proxy Authentication Required
                       |     "408"    ;  Request Timeout
                       |     "409"    ;  Conflict
                       |     "410"    ;  Gone
                       |     "411"    ;  Length Required
                       |     "413"    ;  Request Message Body Too Large
                       |     "414"    ;  Request-URI Too Large
                       |     "415"    ;  Unsupported Media Type
                       |     "420"    ;  Bad Extension
                       |     "480"    ;  Temporarily not available
                       |     "481"    ;  Invalid Call-ID
                       |     "482"    ;  Loop Detected
                       |     "483"    ;  Too Many Hops
                       |     "484"    ;  Address Incomplete
                       |     "485"    ;  Ambiguous

   Figure 7: Client error status codes

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

   Figure 8: Server error status codes

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

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

   Figure 9: Global failure status codes

   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 4. The table uses "o" to indicate
   optional, "m" mandatory and "-" for not applicable. A "*" indicates
   that the header fields are needed only if message body is not empty:
   The  Content-Type and  Content-Length headers are required when there
   is a valid message body (of non-zero length) associated with the
   message (Section 8).

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

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

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

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

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

6.1 General Header Fields

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                           type     enc.    e-e   ACK   BYE   CAN   INV   OPT   REG
   Accept                   R                e     -     -     -     o     o     o
   Accept-Encoding          R                e     -     -     -     o     o     o
   Accept-Language          R        n       e     -     o     o     o     o     o
   Allow                   405               e     o     o     o     o     o     o
   Authorization            R                e     o     o     o     o     o     o
   Call-ID                  g        n       e     m     m     m     m     m     m
   Content-Encoding         e                e     *     -     -     *     *     *
   Content-Length           e                e     m     -     -     m     m     m
   Content-Type             e                e     *     -     -     *     *     *
   CSeq                     g        n       e     m     m     m     m     m     o
   Date                     g                e     o     o     o     o     o     o
   Encryption               g        n       e     o     o     o     o     o     o
   Expires                  g                e     -     -     -     o     o     o
   From                     g                e     m     m     m     m     m     m
   Hide                     R        n       h     o     o     o     o     o     o
   Location                 R                e     o     -     -     o     -     m
   Location                2xx               e     -     -     -     o     o     -
   Location                3xx               e     -     o     -     o     o     o
   Location                485               e     -     o     -     o     o     o
   Max-Forwards             R        n       e     o     o     o     o     o     o
   Organization             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 4: Summary of header fields

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

6.2 Entity Header Fields

   The  entity-header fields define meta-information about the message-
   body or, if no body is present, about the resource identified by the
   request. The term "entity header" is an HTTP 1.1 term where the
   response body may contain a transformed version of the message body.
   The original message body is referred to as the "entity". We retain
   the same terminology for header fields but usually refer to the
   "message body" rather then the entity as the two are the same in SIP.

6.3 Request Header Fields

   The  request-header fields allow the client to pass additional
   information about the request, and about the client itself, to the
   server. These fields act as request modifiers, with semantics
   equivalent to the parameters of a programming language method

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

6.4 Response Header Fields

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

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

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

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

6.6 Header Field Format

   Header fields ( general-header,  request-header, response-header, and
   entity-header) follow the same generic header format as that given in
   Section 3.1 of RFC 822 [24]. Each header field consists of a name
   followed by a colon (":") and the field value. Field names are case-
   insensitive. The field value may be preceded by any amount of leading
   white space (LWS), though a single space (SP) is preferred. Header
   fields can be extended over multiple lines by preceding each extra
   line with at least one SP or horizontal tab (HT). Applications SHOULD
   follow HTTP "common form" when generating these constructs, since
   there might exist some implementations that fail to accept anything
   beyond the common forms.

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

   The order in which header fields are received is not significant if
   the header fields have different field names. Multiple header fields
   with the same field-name may be present in a message if and only if
   the entire field-value for that header field is defined as a comma-
   separated list (i.e.,  #(values)). It MUST be possible to combine the
   multiple header fields into one "field-name: field-value" pair,
   without changing the semantics of the message, by appending each
   subsequent field-value to the first, each separated by a comma. The
   order in which header fields with the same field-name are received is
   therefore significant to the interpretation of the combined field
   value, and thus a proxy MUST NOT change the order of these field
   values when a message is forwarded.

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

6.7  Accept

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

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     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. A proxy may use this
   field to help select the destination for the call, for example, a
   human operator conversant in a language spoken by the caller.


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

6.10  Allow

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

6.11  Authorization

   See [H14.8].

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

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6.12  Call-ID

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

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

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

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

   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
        case-ignoring string comparison.

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

   host MUST be either a fully qualified domain name or a globally
   routable IP address, while the  local-id is a random identifier
   unique within  host. The use of a  UUID as local-id is OPTIONAL. The

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   UUID is a version-4 (random) UUID [19].

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


     Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@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 [H14.12].

6.14  Content-Length

   The  Content-Length entity-header field indicates the size of the
   message-body, in decimal number of octets, sent to the recipient.

        Content-Length    =    "Content-Length" ":" 1*DIGIT

   An example is

     Content-Length: 3495

   Applications MUST use this field to indicate the size of the
   message-body to be transferred, regardless of the media type of the
   entity. Any  Content-Length greater than or equal to zero is a valid
   value. If no body is present in a message, then the Content-Length
   header MUST be set to zero. If a server receives a 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

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

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

   Examples of this header field are

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

6.16  CSeq

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

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

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

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

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

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        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.  CANCEL requests can
   be generated by proxies; if they were to increase the sequence
   number, it might conflict with a later request issued by the user
   agent for the same call.

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

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


     CSeq: 4711 INVITE

6.17  Date

   General header field. See [H14.19].

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

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        current time.

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. This header field 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 field.

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

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

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

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

        The token indicates the form of encryption used; it is
        described in section 12.

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

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


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

6.19  Expires

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

   This header field is currently defined only for the  REGISTER and
   INVITE methods. For  REGISTER, it is a request and response-header
   field 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 addresses contained in the request. 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

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   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 clock.

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

   Two examples of its use are

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

6.20  From

   Requests and responses MUST contain a  From general-header field,
   indicating the initiator of the request. The server copies the To and
   From header fields from the request to the response. The optional
   display-name is meant to be rendered by a human-user interface.

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

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


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

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     From: Anonymous <sip:c8oqz84zk7z@privacy.org>

        Call-ID,  To and  From are needed to identify a call leg
        matters in calls with third-party control. The format is
        similar to the equivalent RFC 822 [24] header, but with a
        URI instead of just an email address.

6.21  Hide

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

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

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

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

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

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

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   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" )

6.22  Location

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

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

   INVITE and  ACK requests:  INVITE and  ACK requests SHOULD contain
        Location headers indicating from which location the request is
        originating. If the SIP address does not refer to the user agent
        server, the SIP URL MUST contain a  tag parameter uniquely
        identifying the user agent. (The same person may be logged on at
        several locations within the same domain served by the proxy.)

        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.

   INVITE 2xx responses: A user agent server sending a definitive,
        positive response (2xx) MAY insert a  Location response header

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        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. If the SIP address does not refer to
        the user agent server, the SIP URL MUST contain a  tag parameter
        uniquely identifying the user agent. (The same person may be
        logged on at several locations within the same domain served by
        the proxy.) The value of this  Location header is copied into
        the  Request-URI of subsequent requests for this call.

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

   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  BYE,  INVITE and  OPTIONS methods.
        The  Location header field contains URIs giving the new
        locations or user names to try, or may simply specify additional
        transport parameters. A 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.

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

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

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

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

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

   action: The  action is only used when registering with the  REGISTER
        request. It indicates whether the client wishes that the server
        proxies or redirects future requests intended for the client.
        The action taken if this parameter is not specified depends on

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        server configuration. In its response, the registrar SHOULD
        indicate the mode used. This parameter is ignored for other

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

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


     Location: sip:watson@worcester.bell-telephone.com;tag=123
        mailto:watson@bell-telephone.com ;q=0.1

6.23  Max-Forwards

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

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

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

   Each proxy or gateway recipient of a request containing a Max-
   Forwards header field MUST check and update its value prior to
   forwarding the request. If the received value is zero (0), the
   recipient MUST NOT forward the request. Instead, for the  OPTIONS and

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   REGISTER methods, it MUST respond as the final recipient. For all
   other methods, the server returns 483 (Too many hops).

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


     Max-Forwards: 6

6.24  Organization

   The  Organization 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

6.25  Priority

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

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

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


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

     Subject: Weekend plans
     Priority: non-urgent

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

6.26  Proxy-Authenticate

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

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

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

6.27  Proxy-Authorization

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

6.28  Proxy-Require

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

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

6.29  Record-Route

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

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

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

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

   The  Record-Route header field has the following syntax:

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

6.30  Require

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

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

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

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

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

6.31  Response-Key

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

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

   The  key-scheme gives the type of encryption to be used for the
   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

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   whatever reason, it MUST follow normal  Require header field
   procedures and return a 420 (Bad Extension) response. If this Require
   header is not present, a client SHOULD still encrypt, but MAY return
   an unencrypted response if unable to.

6.32  Retry-After

   The  Retry-After response header field can be used with a 503
   (Service Unavailable) response to indicate how long the service is
   expected to be unavailable to the requesting client and with a 404
   (Not Found), 600 (Busy), or 603 (Decline) response to indicate when
   the called party may be available again. The value of this field can
   be either an HTTP-date or an integer number of seconds (in decimal)
   after the time of the response.

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

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

        Retry-After    =    "Retry-After" ":" ( HTTP-date | delta-seconds )
                            [ comment ] [ ";duration" "=" delta-seconds ]

   Examples of its use are

     Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting)
     Retry-After: Mon,  1 Jan 9999 00:00:00 GMT
       (Dear John: Don't call me back, ever)
     Retry-After: Fri, 26 Sep 1997 21:00:00 GMT;duration=3600
     Retry-After: 120

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

6.33  Route

   The  Route request header determines the route taken by a request.

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

   The  Route header field has the following syntax:

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

6.34  Server

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

6.35  Subject

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

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


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

6.36  Timestamp

   The timestamp general header describes when the client sent the
   request to the server. The value of the timestamp is of significance
   only to the client and may use any timescale. The server MUST echo
   the exact same value and MAY, if it has accurate information about
   this, add a floating point number indicating the number of seconds
   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.

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

6.37  To

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

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

   The UAS copies the  To header into its response, but SHOULD add a
   tag parameter if not already present. It MAY forego adding the  tag
   parameter if there is no chance that another UAS responds to the same

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


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

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

6.38  Unsupported

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

6.39  User-Agent

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

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6.40  Via

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

6.40.1 Requests

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

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

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

   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.

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6.40.2 Receiver-tagged  Via Fields

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

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

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

6.40.3 Responses

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

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

        2.   If the second  Via field is a receiver-tagged field
             (Section 6.40.2), send the message to the address in the
             received tag. Otherwise, if the  Via header contains a
             maddr multicast address, send the response to that
             multicast address, using the value of the  ttl parameter if
             given.  Otherwise, send the message to the address
             indicated in the sent-by parameter.

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

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

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   A user agent server or redirect server returns the response to the
   network address where the request came from. (Since these servers do
   not forward the request, they do not add a  Via header field or
   received tag.)

6.40.4 Syntax

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

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

   Figure 10: Syntax of  Via header field

   The defaults for " protocol-name" and " transport" are "SIP" and
   "UDP", respectively. The " maddr" parameter, designating the
   multicast address, and the " ttl" parameter, designating the time-
   to-live (TTL) value, are included only if the request was sent via
   multicast. The " received" parameter is added only for receiver-added
   Via fields (Section 6.40.2). For reasons of privacy, a client or
   proxy may wish to hide its  Via information by encrypting it (see
   Section 6.21). The " hidden" parameter is included if this header was
   hidden by the upstream proxy (see 6.21).

   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

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   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;ttl=16
       ;maddr= (Example)
     Via: SIP/2.0/UDP adk8

6.41  Warning

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

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

   A response may carry more than one  Warning header.

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

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

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

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

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

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

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

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

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

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

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

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

   x08 Unicast not available: The site where the user is located does
        not support unicast communication (usually due to the presence
        of a firewall).

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

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

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

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

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


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

6.42  WWW-Authenticate

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

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

   In addition to the "basic" and "digest" authentication schemes
   defined in the specifications cited above, SIP defines a new scheme,
   PGP (RFC 2015, [28]), Section 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.  A
   server can issue zero or more 1xx responses, with no restriction on
   their ordering or uniqueness. Note that 1xx responses are not
   transmitted reliably, that is, they do not cause the client to send

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   an ACK. Servers are free to retransmit informational responses and
   clients can inquire about the current state of call processing by
   re-sending the request.

7.1.1 100 Trying

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

7.1.2 180 Ringing

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

7.1.3 181 Call Is Being Forwarded

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

7.1.4 182 Queued

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

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.

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

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

7.3 Redirection 3xx

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

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

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

7.3.1 300 Multiple Choices

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

   The response SHOULD include an entity containing a list of resource
   characteristics and location(s) from which the user or user agent can
   choose the one most appropriate, if allowed by the  Accept request
   header. The entity format is specified by the media type given in the
   Content-Type header field. The choices SHOULD also be listed as
   Location fields (Section 6.22).  Unlike HTTP, the SIP response may
   contain several  Location fields or a list of addresses in a
   Location field. User agents MAY use the  Location field value for
   automatic redirection or MAY ask the user to confirm a choice.
   However, this specification does not define any standard for such
   automatic 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

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   the requesting client should retry at the new address given by the
   Location header field (Section 6.22). The caller SHOULD update any
   local directories, address books and user location caches with this
   new value and redirect future requests to the address(es) listed.

7.3.3 302 Moved Temporarily

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

7.3.4 380 Alternative Service

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

7.4 Request Failure 4xx

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

7.4.1 400 Bad Request

   The request could not be understood due to malformed syntax.

7.4.2 401 Unauthorized

   The request requires user authentication.

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.

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7.4.6 405 Method Not Allowed

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

7.4.7 406 Not Acceptable

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

7.4.8 407 Proxy Authentication Required

   This code is similar to 401 (Unauthorized), but indicates that the
   client MUST first authenticate itself with the proxy. The proxy MUST
   return a  Proxy-Authenticate header field (section 6.26) containing a
   challenge applicable to the proxy for the requested resource. The
   client MAY repeat the request with a suitable Proxy-Authorization
   header field (section 6.27). SIP access authentication is explained
   in section 12.2 and [H11].

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

7.4.9 408 Request Timeout

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

7.4.10 414 Request-URI Too Long

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

7.4.11 415 Unsupported Media Type

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

7.4.12 420 Bad Extension

   The server did not understand the protocol extension specified in a

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   Require (Section 6.30) header field.

7.4.13 480 Temporarily Unavailable

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

7.4.14 481 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.)

7.4.15 482 Loop Detected

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

7.4.16 483 Too Many Hops

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

7.4.17 484 Address Incomplete

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

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

7.4.18 485 Ambiguous

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

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

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

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

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

7.5 Server Failure 5xx

   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

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

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

7.5.5 504 Gateway Timeout

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

7.5.6 505 Version Not Supported

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

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7.6.3 604 Does Not Exist Anywhere

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

7.6.4 606 Not Acceptable

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

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

8 SIP Message Body

8.1 Body Inclusion

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

   For response messages, whether or not a body is included is dependent
   on both the request method and the response message's response code.
   All responses MAY include a body, although it may be of zero length.
   Message bodies for 1xx responses contain advisory information about
   the progress of the request. 2xx responses contain session
   descriptions.  For responses with status 300 or greater, the messaage
   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

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   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

   Thus, the header in section 14.2 could also be written:

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

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     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

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

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

   If the  To header user and host information does not match an address
   supported by the server, the server returns a 404 (Not Found) error
   response. Otherwise, it searches for the  Call-ID value.

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

   If the  Call-ID was not found, a new call leg is created, with

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

10.1.2 Responses

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

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

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

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10.2 Source Addresses, Destination Addresses and Connections

10.2.1 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 6.40), not the source
   address of the request.

10.2.2 Multicast UDP

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

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

   To avoid response implosion, servers MUST NOT answer multicast
   requests with a status code other than 2xx or 6xx. Servers only
   return 6xx responses if the  To represents a single individual rather
   than a group of people. The server delays its response by a random
   interval between zero and one second. Servers 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 to avoid request implosion.

10.3 TCP

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

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

   If the server leaves the connection open, and if the client so

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   desires it may re-use the connection for further SIP requests or for
   requests from the same family of protocols (such as HTTP or stream
   control commands).

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

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

10.4 Reliability for  BYE,  CANCEL,  OPTIONS,  REGISTER Requests

10.4.1 UDP

   A SIP client using UDP SHOULD retransmit a  BYE,  CANCEL, OPTIONS, or
   REGISTER request periodically with timer T1 until it receives a
   response, or until it has reached a set limit on the number of
   retransmissions. If the response is provisional, the client continues
   to retransmit the request, albeit less frequently, using timer T2.
   The default values of timer T1 and T2 are 1 and 5 seconds,
   respectively. The default retransmit limit is 20 times. After the
   server sends a final response, it cannot be sure the client has
   received the response, and thus SHOULD cache the results for at least
   100 seconds to avoid having to, for example, contact the user or
   location server again upon receiving a retransmission.

   Each server in a proxy chain generates its own final response to a
   CANCEL request;  BYE and  OPTIONS final responses are generated by
   redirect and user agent servers;  REGISTER final responses are
   generated by registrars. Note that responses to these commands are
   not acknowledged via  ACK.

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

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        Timestamp (Section 6.36), returned in the response. The
        server caches the result until it can be sure that the
        client will not retransmit the same request again.

10.4.2 TCP

   Clients using TCP do not need to retransmit requests.

10.5 Reliability for  ACK Requests

   The  ACK request does not generate responses. It is only
   retransmitted when a response to an  INVITE request arrives. This
   behavior is independent of the transport protocol.

10.6 Reliability for  INVITE Requests

   Special considerations apply for the  INVITE method.

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

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

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

10.6.1 UDP

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

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

   For 2xx final responses, only the user agent client generates an
   ACK.  If the response contained a  Location header, the  ACK is sent
   to the address listed in that  Location header field. If the response
   did not contain a  Location header, the client uses the same  To
   header field and  Request-URI as for the INVITE request and sends the
   ACK to the same destination as the original  INVITE request.  ACKs
   for final responses other than 2xx are sent to the source of the
   response by each client.

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

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

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

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


   Figure 11: State transition diagram of client for  INVITE method

10.6.2 TCP

   A client using TCP MUST NOT retransmit requests, but uses the same
   algorithm as for UDP (Section 10.6.1) to retransmit responses until

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


   Figure 12: State transition diagram of server for  INVITE method

   it receives an  ACK. (An implementation can simply set T1 and T3 to
   infinity and otherwise maintain the same state diagram.)

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        It is necessary to retransmit 2xx responses as their
        reliability is assured end-to-end only. If the chain of
        proxies has a UDP link in the middle, it could lose the
        response, with no possibility of recovery. For simplicity,
        we also retransmit non-2xx responses, although that is not
        strictly necessary.

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 6.40). 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.

11.1 Redirect Server

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

11.2 User Agent Server

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

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.

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   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 Requests

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

   All outgoing requests carry the same  Call-ID,  To, From and  CSeq
   headers as the request received. Each of the requests has a different
   (host-dependent)  Request-URI.

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

   The following pseudo-code describes the behavior of a proxy server
   issuing several requests in response to an incoming  INVITE request.
   The function request(r, a, b) sends a SIP request of type r to
   address a, with branch id b. await_response() waits until a response
   is received and returns the response. close(a) closes the TCP
   connection to client with address a. response(r) sends a response to
   the client. ismulticast() returns 1 if the location is a multicast
   address and zero otherwise.  The variable timeleft indicates the
   amount of time left until the maximum response time has expired. The
   variable recurse indicates whether the server will recursively try
   addresses returned through a 3xx response. A server MAY decide to
   recursively try only certain addresses, e.g., those which are within
   the same domain as the proxy server. Thus, an initial multicast
   request may trigger additional unicast requests.

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

     process_request(Method R, int N, address_t address[])
       struct {

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         address_t address;  /* address */
         int branch;         /* branch id */
         int done;           /* has responded */
       } outgoing[];
       int done[];           /* address has responded */
       char *location[];     /* list of locations */
       int heard = 0;        /* number of sites heard from */
       int class;            /* class of status code */
       int timeleft = 120;   /* sample timeout value */
       int loc = 0;          /* number of locations */
       struct {              /* response */
         int status;         /* response: BYE=-2; CANCEL=-1 */
         int locations;      /* number of redirect locations */
         char *location[];   /* redirect locations */
         address_t a;        /* address of respondent */
         int branch;         /* branch identifier */
       } r, best;            /* response, best response */
       int i;

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

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

         if (r.status < 0) {

         /* If final response, mark branch as done. */
         if (class >= 2) {
           for (i = 0; i < N; i++) {
             if (r.branch == outgoing[i].branch) {
               outgoing[i].done = 1;

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

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         else if (class == 3) {
                   /* A server may optionally recurse.  The server MUST check
                    * whether it has tried this location before and whether the
                    * location is part of the Via path of the incoming request.
                    * This check is omitted here for brevity.  Multicast locations
                    * MUST NOT be returned to the client if the server is not
                    * recursing.
           if (recurse) {
             multicast = 0;
             N += r.locations;
             for (i = 0; i < r.locations; i++) {
               request(R, r.location[i]);
           } else if (!ismulticast(r.location)) {
             best = r;
           if (R == INVITE} request(ACK, r.a, r.branch);
         else if (class == 4) {
           if (R == INVITE} request(ACK, r.a, r.branch);
           if (best.status >= 400) best = r;
         else if (class == 5) {
           if (R == INVITE} request(ACK, r.a, r.branch);
           if (best.status >= 500) best = r;
         else if (class == 6) {
           if (R == INVITE} request(ACK, r.a, r.branch);
           best = r;

       /* CANCEL */
       if (r.status == -1) {
         best.status = 200;
       /* BYE */
       else if (r.status == -2) {
         for (i = 0; i < N; i++) {
           request(BYE, address[i], i);
       /* INVITE */
       else {

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

            * If complete or CANCELed, close the other pending transactions by
            * sending CANCEL.
       if (r.status > 0 || 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 (and state
   can be freed) when all requests have been answered by final status
   responses (for unicast) or 60 seconds have elapsed (for multicast). A
   proxy MAY send a  CANCEL to all branches and return a 408 (Timeout)
   to the client after 60 seconds or more.

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

   2xx: The proxy MUST forward the response upstream towards the client,
        without sending an  ACK downstream. After receiving a 2xx, the
        server 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

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

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

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   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 as
        described for 2xx responses.

   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, the
   proxy MUST terminate all pending requests by sending a  BYE request.

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

   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
   INVITE. If the proxy has no call state for a particular Call-ID and
   To destination, it forks the request as it would for an  INVITE

12 Security Considerations

12.1 Confidentiality and Privacy: Encryption

12.1.1 End-to-End Encryption

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

   SIP supports three complementary forms of encryption to protect

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

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

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

   Not all of the SIP request or response can be encrypted end-to-end
   because header fields such as  To and  Via need to be visible to
   proxies so that the SIP request can be routed correctly.  Hop-by-hop
   encryption encrypts the entire SIP request or response on the wire
   (the request may already have been end-to-end encrypted) so that
   packet sniffers or other eavesdroppers cannot see who is calling
   whom.  Note that proxies can still see who is calling whom, and this
   information may also be deducible by performing a network traffic
   analysis, so this provides a very limited but still worthwhile degree
   of protection.

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

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

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

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

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

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

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   Content-Length header field within the encrypted body because the
   length of the actual message body is unambiguous after decryption.

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

     INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP$
     To: T. A. Watson <sip:watson@bell-telephone.com>$
     From: A. Bell <a.g.bell@bell-telephone.com>$
     Encryption: PGP version=5.0$
     Content-Type: application/sdp$
     Content-Length: 107$
   * $                                             *
   * v=0$                                          *
   * o=bell 53655765 2353687637 IN IP4$  *
   * c=IN IP4$                      *
   * m=audio 3456 RTP/AVP 0 3 4 5$                 *

12.1.2 Privacy of SIP Responses

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

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

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12.1.3 Encryption by Proxies

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

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

12.1.4 Hop-by-Hop Encryption

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

12.1.5 Via field encryption

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

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

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

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

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

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

12.2 Message Integrity and Access Control: Authentication

   An active attacker may be able to modify and replay SIP requests and
   responses unless protective measures are taken. 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 6.42) and Authorization (Section 6.11) header and their
   Proxy- counterparts to include cryptographically strong signatures.
   SIP also supports the HTTP "basic" authentication scheme

   that offers a very rudimentary mechanism of ascertaining the identity
   of the caller.

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

   SIP requests may be authenticated using the  Authorization header
   field to include a digital signature of certain header fields, the
   request method and version number and the payload, none of which are
   modified between client and called user agent. The Authorization
   header field may be used in requests to 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

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

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

   Then the data block that is signed is:

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

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

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

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

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

12.2.1 Trusting responses

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

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

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

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

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

   These attacks are prevented by having the client require response
   authentication and dropping unauthenticated responses. A server user
   agent that cannot perform response authentication responds using the

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   normal  Require response of 420 (Bad Extension).

12.3 Callee Privacy

   User location and SIP-initiated calls may violate a callee's privacy.
   An implementation SHOULD be able to restrict, on a per-user basis,
   what kind of location and availability information is given out to
   certain classes of callers.

12.4 Known Security Problems

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

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

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        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"

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-

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

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

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

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


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

13.2 PGP Encryption Scheme

   The PGP encryption scheme uses the following syntax:

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

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

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   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 ".


   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:@sip.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Location: sip:saturn.bell-tel.com:3890;transport=udp
         Call-ID: 123@saturn.bell-tel.com
         Expires: 7200
         CSeq: 1 REGISTER

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

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

   C->S: REGISTER sip:@bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 1234@saturn.bell-tel.com
         Expire: 0
         Location: *

   C->S: REGISTER sip:@bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Call-ID: 1235@saturn.bell-tel.com
         Location: sip:tawatson@example.com

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

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

   C->S: REGISTER sip:@bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:jon.diligent@bell-tel.com
         To: sip:watson@bell-tel.com
         Location: sip:tawatson@example.com
         Call-ID: 1236@saturn.bell-tel.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

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   The first example invites schooler@vlsi.cs.caltech.edu to a multicast
   session. All examples use the Session Description Protocol (SDP) (RFC
   2327 [5]) 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;maddr=;ttl=16
         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: 42100bb8-1dd2-11b2-8d70-c91e31477491@oregon.isi.edu
         Content-Type: application/sdp
         CSeq: 4711 INVITE
         Content-Length: 187

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

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

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

   In this case, the session description is using the Session
   Description Protocol (SDP), as stated in the  Content-Type header.

   The header is terminated by an empty line and is followed by a
   message body containing the session description.

14.2.2 Response

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

   S->C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348;
         Via: SIP/2.0/UDP north.east.isi.edu
         To: Eve Schooler <sip:schooler@caltech.edu>
         From: Mark Handley <sip:mjh@isi.edu>
         Call-ID: 42100bb8-1dd2-11b2-8d70-c91e31477491@north.east.isi.edu
         Location: sip:es@jove.cs.caltech.edu
         CSeq: 4711 INVITE

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

   In addition, the  Location header gives details of the host where the
   user was located, or alternatively the relevant proxy contact point
   which should be reachable from the caller's host.

   S->C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=8348
           maddr= 16;ttl=16
         Via: SIP/2.0/UDP north.east.isi.edu
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 42100bb8-1dd2-11b2-8d70-c91e31477491@oregon.isi.edu
         Location: sip:es@jove.cs.caltech.edu
         CSeq: 4711 INVITE

   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

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         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 42100bb8-1dd2-11b2-8d70-c91e31477491@oregon.isi.edu
         CSeq: 4711 ACK

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: 2d978243-b270-33dc-a261-d1fe3e2aa05a@kton.bell-tel.com
         Subject: Mr. Watson, come here.
         CSeq: 17 INVITE
         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/2.0 100 Trying
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com>
         Call-ID: 2d978243-b270-33dc-a261-d1fe3e2aa05a@kton.bell-tel.com
         Content-Length: 0

   S->C: SIP/2.0 180 Ringing
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com>
         Call-ID: 2d978243-b270-33dc-a261-d1fe3e2aa05a@kton.bell-tel.com
         Content-Length: 0

   S->C: SIP/2.0 182 Queued, 2 callers ahead
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com>
         Call-ID: 2d978243-b270-33dc-a261-d1fe3e2aa05a@kton.bell-tel.com
         Content-Length: 0

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   S->C: SIP/2.0 182 Queued, 1 caller ahead
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com>
         Call-ID: 2d978243-b270-33dc-a261-d1fe3e2aa05a@kton.bell-tel.com
         Content-Length: 0

   S->C: SIP/2.0 200 OK
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: sip:watson@bell-tel.com
         Call-ID: 2d978243-b270-33dc-a261-d1fe3e2aa05a@kton.bell-tel.com
         CSeq: 17 INVITE
         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

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

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

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

   Since the two sides have 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: 2d978243-b270-33dc-a261-d1fe3e2aa05a@kton.bell-tel.com
         CSeq: 17 ACK
         Content-Length: 0

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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@kton.bell-tel.com
         CSeq: 18 BYE

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

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: 88323b2a-0a09-3888-88b4-2f93ee7808ea@kton.bell-tel.com
         CSeq:    19 INVITE
         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>

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         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 88323b2a-0a09-3888-88b4-2f93ee7808ea@kton.bell-tel.com
         CSeq:    19 INVITE

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

   H->P: SIP/2.0 404 Not Found
         Via:     SIP/2.0/UDP 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: 88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:    19 INVITE

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

   P->H: ACK sip:watson@h.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP proxy.ieee.org ;branch=1
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:     19 ACK

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

   P->A: INVITE sip:watson@acm.org SIP/2.0
         Via:      SIP/2.0/UDP 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:  88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:     19 INVITE

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

   P->X: INVITE sip:watson@x.bell-tel.com SIP/2.0

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         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: 88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:    19 INVITE

   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>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:    19 INVITE

   As it happens, both Watson at X and a colleague in the other lab at
   host Y hear the phones ringing and pick up. Both X and Y return 200s
   via the proxy to Bell. The  tag URI parameter 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
         Location: sip:t.watson@x.bell-tel.com;tag=1620
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:     19 INVITE

   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
         Location: sip:t.watson@y.bell-tel.com;tag=2016
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:     19 INVITE

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

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   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: 88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:    19 CANCEL

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

   A->P: SIP/2.0 200 OK
         Via:     SIP/2.0/UDP 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: 88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:    19 CANCEL

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

   C->X: ACK sip:watson@x.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:     19 ACK

   C->Y: ACK sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:     19 ACK

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

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   C->Y: BYE sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:     20 BYE

   Y->C: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  88323b2a-0a09-3888-88b4-2f93ee7808ea@c.bell-tel.com
         CSeq:     20 BYE

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: 46842902-e7b0-3583-ae6a-bee550833c34@oregon.isi.edu
         Location: sip:@;ttl=16;transport=udp
         CSeq: 19 INVITE
         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:

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   S->C: SIP/2.0 350 Alternative Service
         Via: SIP/2.0/UDP
         Via: SIP/2.0/UDP
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 46842902-e7b0-3583-ae6a-bee550833c34oregon.isi.edu
         Location: sip:recorder@
         CSeq: 19 INVITE
         Content-Type: application/sdp
         Content-Length: 146

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

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

   C->S: INVITE sip:recorder@ SIP/2.0
         Via: SIP/2.0/UDP
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 9469f230-70e0-3216-8482-fe1a2a150386@
         CSeq: 20 INVITE
         Content-Type: application/sdp
         Content-Length: 146

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

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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: 9469f230-70e0-3216-8482-fe1a2a150386@
         Location: sip:mjh@
         Warning: 606.1 Insufficient bandwidth (only have ISDN),
           606.3 Incompatible format,
           606.4 Multicast not available
         Content-Type: application/sdp
         Content-Length: 50

         s=Lets talk
         c=IN IP4
         m=audio 3456 RTP/AVP 7 0 13
         m=video 2232 RTP/AVP 31

   In this example, the original request specified 256 kb/s total
   bandwidth, and the response states that only 128 kb/s is available.
   The original request specified GSM audio, H.261 video, and WB
   whiteboard.  The audio coding and whiteboard are not available, but
   the response states that DVI, PCM or LPC audio could be supported in
   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. Bob returns a description indicating that he is
   capable of receiving audio and video, with a list of supported

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

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         To: Bob <sip:bob@example.com>
         Call-ID: 45869@host.anywhere.org
         Accept: application/sdp

   S->C: SIP/2.0 200 OK
         From: Alice <sip:alice@anywhere.org>
         To: Bob <sip:bob@example.com>
         Call-ID: 45869@host.anywhere.org
         Content-Length: 81
         Content-Type: application/sdp

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

A Minimal Implementation

A.1 Client

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

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

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

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

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

   Authentication: If a client wishes to invite callees that require

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        caller authentication, it must be able to recognize the 401
        (Unauthorized) status code, must be able to generate the
        Authorization request header and MUST understand the WWW-
        Authenticate response header.

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

A.2 Server

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

A.3 Header Processing

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

B Usage of SDP

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

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

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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
            Encryption              g       e       -       e
            Expires                 g       -       o       o
            From                    R       m       o       m
            Location                R       -       -       -
            Location                r       r       r       -
            Max-Forwards            R       -       b       -
            Proxy-Authenticate     407      a       -       -
            Proxy-Authorization     R       -       a       -
            Proxy-Require           R       -       m       -
            Require                 R       m       -       m
            Response-Key            R       -       -       e
            Timestamp               g       o       o       m
            To                      g       m       m       m
            Unsupported             r       b       b       -
            Via                     g       -       m       m
            WWW-Authenticate       401      a       -       -

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

C Summary of Augmented BNF

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

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

        The character with ASCII code decimal 32 (space);

        zero of more instances of  term;

        three or more instances of  term;

        two, three or four instances of  term;

   [ term ]
        term is optional;

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

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

        a comma separated list of  term;

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

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

   Common Tokens

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

        CR       =    %d13            ;  carriage return character
        LF       =    %d10            ;  line feed character
        CRLF     =    CR LF           ;  typically the end of a line
        SP       =    %d32            ;  space character

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        TAB      =    %d09            ;  tab character
        LWS      =    *( SP | TAB)    ;  linear whitespace
        DIGIT    =    "0" .. "9"      ;  a single decimal digit

        unreserved    =    alphanum | mark
        mark          =    "-" | "_" | "." | "!" | "~" | "*" | "'"
                     |     "(" | ")"
        escaped       =    "%" hex hex
        hex           =    digit | "A" | "B" | "C" | "D" | "E" | "F" |
                           "a" | "b" | "c" | "d" | "e" | "f"
        alphanum      =    alpha | digit
        alpha         =    lowalpha | upalpha
        lowalpha      =    "a" | "b" | "c" | "d" | "e" | "f" | "g" | "h" | "i" |
                           "j" | "k" | "l" | "m" | "n" | "o" | "p" | "q" | "r" |
                           "s" | "t" | "u" | "v" | "w" | "x" | "y" | "z"
        upalpha       =    "A" | "B" | "C" | "D" | "E" | "F" | "G" | "H" | "I" |
                           "J" | "K" | "L" | "M" | "N" | "O" | "P" | "Q" | "R" |
                           "S" | "T" | "U" | "V" | "W" | "X" | "Y" | "Z"
        digit         =    "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" |
                           "8" | "9"

D IANA Considerations

   Section 4.4 describes a name space and mechanism for registering SIP

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

E Changes in Version -07

   Since version -06, the following changes have been made.

        o Removed references to Internet Drafts.

        o Expanded URI definition to be independent of I-Ds.

        o Clarified redirect behavior for BYE.

        o Call-ID mandatory for all requests to allow to match requests
          with responses.

        o Clarified that  INVITE retransmit limit only applies if there
          has been no provisional response. Otherwise, call queueing is
          not possible.

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        o Removed open issues list.

        o Removed  REGISTER as special case of reliability mechanism.

        o Split out large syntax diagrams into figures to avoid empty

        o Abstract rewritten to reflect current protocol functionality.

        o Reorganized "SIP Transport" chapter to more clearly reflect
          behavior for UDP and TCP.

        o Modified syntax for  Via to include multicast address as a

        o "Ambiguous" status code moved from 381 to 485, to give
          precedence to other, more definitive 3xx responses. Also, 381
          was the only 3xx response that a proxy could not automatically
          recurse on.

        o Response merging made stricter, to avoid difficulties with
          merging bodies and non-standard headers of 3xx responses.

        o REGISTER MUST have  Location header.

        o Responses SHOULD add a tag to the  To header to allow requests
          (e.g.,  BYE) from several instances to be distinguished.

        o REGISTER examples were missing  Via headers.

F Acknowledgments

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

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

G Authors' Addresses

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

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

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

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

H Bibliography

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

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

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

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

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

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

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

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   Engineering Task Force, Jan. 1997.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

   Full Copyright Statement

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

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it

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   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 .........................................    4
   1.3        Definitions .........................................    4
   1.4        Summary of SIP Operation ............................    7
   1.4.1      SIP Addressing ......................................    7
   1.4.2      Locating a SIP Server ...............................    8
   1.4.3      SIP Transaction .....................................    9
   1.4.4      SIP Invitation ......................................   10
   1.4.5      Locating a User .....................................   12
   1.4.6      Changing an Existing Session ........................   14
   1.4.7      Registration Services ...............................   14
   1.5        Protocol Properties .................................   14
   1.5.1      Minimal State .......................................   14
   1.5.2      Lower-Layer-Protocol Neutral ........................   14
   1.5.3      Text-Based ..........................................   15
   2          SIP Uniform Resource Locators .......................   15
   3          SIP Message Overview ................................   19
   4          Request .............................................   21

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   4.1        Request-Line ........................................   22
   4.2        Methods .............................................   22
   4.2.1       INVITE .............................................   22
   4.2.2       ACK ................................................   23
   4.2.3       OPTIONS ............................................   23
   4.2.4       BYE ................................................   24
   4.2.5       CANCEL .............................................   24
   4.2.6       REGISTER ...........................................   25
   4.3        Request-URI .........................................   27
   4.3.1      SIP Version .........................................   28
   4.4        Option Tags .........................................   28
   4.4.1      Registering New Option Tags with IANA ...............   28
   5          Response ............................................   29
   5.1        Status-Line .........................................   29
   5.1.1      Status Codes and Reason Phrases .....................   29
   6          Header Field Definitions ............................   32
   6.1        General Header Fields ...............................   33
   6.2        Entity Header Fields ................................   35
   6.3        Request Header Fields ...............................   35
   6.4        Response Header Fields ..............................   35
   6.5        End-to-end and Hop-by-hop Headers ...................   35
   6.6        Header Field Format .................................   36
   6.7         Accept .............................................   36
   6.8         Accept-Encoding ....................................   37
   6.9         Accept-Language ....................................   37
   6.10        Allow ..............................................   37
   6.11        Authorization ......................................   37
   6.12        Call-ID ............................................   38
   6.13        Content-Encoding ...................................   39
   6.14        Content-Length .....................................   39
   6.15        Content-Type .......................................   39
   6.16        CSeq ...............................................   40
   6.17        Date ...............................................   41
   6.18        Encryption .........................................   42
   6.19        Expires ............................................   43
   6.20        From ...............................................   44
   6.21        Hide ...............................................   45
   6.22        Location ...........................................   46
   6.23        Max-Forwards .......................................   48
   6.24        Organization .......................................   49
   6.25        Priority ...........................................   49
   6.26        Proxy-Authenticate .................................   50
   6.27        Proxy-Authorization ................................   50
   6.28        Proxy-Require ......................................   50
   6.29        Record-Route .......................................   50
   6.30        Require ............................................   51
   6.31        Response-Key .......................................   52
   6.32        Retry-After ........................................   53

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   6.33        Route ..............................................   53
   6.34        Server .............................................   54
   6.35        Subject ............................................   54
   6.36        Timestamp ..........................................   54
   6.37        To .................................................   55
   6.38        Unsupported ........................................   55
   6.39        User-Agent .........................................   55
   6.40        Via ................................................   56
   6.40.1     Requests ............................................   56
   6.40.2     Receiver-tagged  Via Fields .........................   57
   6.40.3     Responses ...........................................   57
   6.40.4     Syntax ..............................................   58
   6.41        Warning ............................................   59
   6.42        WWW-Authenticate ...................................   61
   7          Status Code Definitions .............................   61
   7.1        Informational 1xx ...................................   61
   7.1.1      100 Trying ..........................................   62
   7.1.2      180 Ringing .........................................   62
   7.1.3      181 Call Is Being Forwarded .........................   62
   7.1.4      182 Queued ..........................................   62
   7.2        Successful 2xx ......................................   62
   7.2.1      200 OK ..............................................   62
   7.3        Redirection 3xx .....................................   63
   7.3.1      300 Multiple Choices ................................   63
   7.3.2      301 Moved Permanently ...............................   63
   7.3.3      302 Moved Temporarily ...............................   64
   7.3.4      380 Alternative Service .............................   64
   7.4        Request Failure 4xx .................................   64
   7.4.1      400 Bad Request .....................................   64
   7.4.2      401 Unauthorized ....................................   64
   7.4.3      402 Payment Required ................................   64
   7.4.4      403 Forbidden .......................................   64
   7.4.5      404 Not Found .......................................   64
   7.4.6      405 Method Not Allowed ..............................   65
   7.4.7      406 Not Acceptable ..................................   65
   7.4.8      407 Proxy Authentication Required ...................   65
   7.4.9      408 Request Timeout .................................   65
   7.4.10     414 Request-URI Too Long ............................   65
   7.4.11     415 Unsupported Media Type ..........................   65
   7.4.12     420 Bad Extension ...................................   65
   7.4.13     480 Temporarily Unavailable .........................   66
   7.4.14     481 Invalid Call-ID .................................   66
   7.4.15     482 Loop Detected ...................................   66
   7.4.16     483 Too Many Hops ...................................   66
   7.4.17     484 Address Incomplete ..............................   66
   7.4.18     485 Ambiguous .......................................   66
   7.5        Server Failure 5xx ..................................   67
   7.5.1      500 Server Internal Error ...........................   67

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   7.5.2      501 Not Implemented .................................   67
   7.5.3      502 Bad Gateway .....................................   67
   7.5.4      503 Service Unavailable .............................   67
   7.5.5      504 Gateway Timeout .................................   68
   7.5.6      505 Version Not Supported ...........................   68
   7.6        Global Failures 6xx .................................   68
   7.6.1      600 Busy ............................................   68
   7.6.2      603 Decline .........................................   68
   7.6.3      604 Does Not Exist Anywhere .........................   69
   7.6.4      606 Not Acceptable ..................................   69
   8          SIP Message Body ....................................   69
   8.1        Body Inclusion ......................................   69
   8.2        Message Body Type ...................................   69
   8.3        Message Body Length .................................   70
   9          Compact Form ........................................   70
   10         SIP Transport .......................................   71
   10.1       General Remarks .....................................   71
   10.1.1     Requests ............................................   71
   10.1.2     Responses ...........................................   72
   10.2       Source Addresses, Destination Addresses and
   Connections ....................................................   73
   10.2.1     Unicast UDP .........................................   73
   10.2.2     Multicast UDP .......................................   73
   10.3       TCP .................................................   73
   10.4       Reliability for  BYE,  CANCEL,  OPTIONS,  REGISTER
   Requests .......................................................   74
   10.4.1     UDP .................................................   74
   10.4.2     TCP .................................................   75
   10.5       Reliability for  ACK Requests .......................   75
   10.6       Reliability for  INVITE Requests ....................   75
   10.6.1     UDP .................................................   75
   10.6.2     TCP .................................................   77
   11         Behavior of SIP Servers .............................   79
   11.1       Redirect Server .....................................   79
   11.2       User Agent Server ...................................   79
   11.3       Stateless Proxy: Proxy Servers Issuing Single
   Unicast Requests ...............................................   79
   11.4       Proxy Server Issuing Several Requests ...............   80
   12         Security Considerations .............................   84
   12.1       Confidentiality and Privacy: Encryption .............   84
   12.1.1     End-to-End Encryption ...............................   84
   12.1.2     Privacy of SIP Responses ............................   87
   12.1.3     Encryption by Proxies ...............................   88
   12.1.4     Hop-by-Hop Encryption ...............................   88
   12.1.5     Via field encryption ................................   88
   12.2       Message Integrity and Access Control:
   Authentication .................................................   89
   12.2.1     Trusting responses ..................................   91

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   12.3       Callee Privacy ......................................   92
   12.4       Known Security Problems .............................   92
   13         SIP Security Using PGP ..............................   92
   13.1       PGP Authentication Scheme ...........................   92
   13.1.1     The  WWW-Authenticate Response Header ...............   92
   13.1.2     The  Authorization Request Header ...................   93
   13.2       PGP Encryption Scheme ...............................   94
   13.3        Response-Key Header Field for PGP ..................   95
   14         Examples ............................................   95
   14.1       Registration ........................................   95
   14.2       Invitation to Multicast Conference ..................   96
   14.2.1     Request .............................................   97
   14.2.2     Response ............................................   97
   14.3       Two-party Call ......................................   99
   14.4       Terminating a Call ..................................  101
   14.5       Forking Proxy .......................................  101
   14.6       Redirects ...........................................  105
   14.7       Alternative Services ................................  105
   14.8       Negotiation .........................................  107
   14.9        OPTIONS Request ....................................  107
   A          Minimal Implementation ..............................  108
   A.1        Client ..............................................  108
   A.2        Server ..............................................  109
   A.3        Header Processing ...................................  109
   B          Usage of SDP ........................................  109
   C          Summary of Augmented BNF ............................  110
   D          IANA Considerations .................................  112
   E          Changes in Version -07 ..............................  112
   F          Acknowledgments .....................................  113
   G          Authors' Addresses ..................................  113
   H          Bibliography ........................................  114

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