Internet Engineering Task Force                                MMUSIC WG
Internet Draft                              Handley/Schulzrinne/Schooler
draft-ietf-mmusic-sip-04.txt
draft-ietf-mmusic-sip-05.txt                     ISI/Columbia U./Caltech
November 11, 1997
May 14, 1998
Expires: April 1, November 1998

                    SIP: Session Initiation Protocol

STATUS OF THIS MEMO

   This document is an Internet-Draft. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
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   Distribution of this document is unlimited.

                                 ABSTRACT

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

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

1 Introduction

1.1 Overview of SIP Functionality

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

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

   SIP transparently supports name mapping and redirection services,
   allowing the implementation of ISDN and Intelligent Network telephony
   subscriber services. Section 14 discusses these services in detail.

   SIP supports These facilities also enable personal mobility telecommunications intelligent network
   services, this is defined as: "Personal mobility is the ability of
   end users to originate and receive calls and access subscribed
   telecommunication services on any terminal in any location, and the
   ability of the network to identify end users as they move. Personal
   mobility is based on the use of a unique personal identity (i.e.,
   'personal number')." [3].  Personal
   mobility complements terminal mobility, i.e., the ability to maintain
   communications when moving a single end system from one network to
   another.

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

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

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

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

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

   SIP may also be used in conjunction with other call setup and
   signaling protocols. In that mode, an end system uses SIP protocol
   exchanges to determine the appropriate end system address and
   protocol from a given address that is protocol-independent. For
   example, SIP may be used to determine that the party may be reached
   via H.323, obtain the H.245 gateway and user address and then use
   H.225.0 to establish the call [4]. [8]. In another example, it may be used
   to determine that the callee is reachable via the public switched
   telephone network (PSTN) and indicate the phone number to be called,
   possibly suggesting an Internet-to-PSTN gateway to be used.

   SIP can also initiate multi-party calls using a multipoint control
   unit (MCU) or fully-meshed interconnection instead of multicast.

   Internet telephony gateways that connect PSTN parties may also use
   SIP to set up calls between them.

   SIP does not offer conference control services such as floor control
   or voting and does not prescribe how a conference is to be managed,
   but SIP can be used to introduce conference control protocols. SIP
   does not allocate multicast addresses, leaving this functionality
   to protocols such as SAP [2]. addresses.

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

   SIP is designed as part of

1.2 Terminology

   In this document, the overall IETF multimedia data key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
   and
   control architecture [5] currently incorporating protocols such "OPTIONAL" are to be interpreted as
   RSVP [6] for reserving network resources, the real-time transport
   protocol (RTP) [7] for transporting real-time data and providing QOS
   feedback, the real-time streaming protocol (RTSP) [8] for controlling
   delivery of streaming media, the session announcement protocol (SAP)
   [2] for advertising multimedia sessions via multicast and the session
   description protocol (SDP) [9] for describing multimedia sessions,
   but the functionality and operation of SIP does not depend on any of
   these protocols.

1.2 Terminology
   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
   and "OPTIONAL" are to be interpreted as described in RFC 2119 [10]
   and indicate requirement levels described in RFC 2119 [9] and
   indicate requirement levels for compliant SIP implementations.

1.3 Definitions

   This specification uses a number of terms to refer to the roles
   played by participants in SIP communications. The definitions of
   client, server and proxy are similar to those used by the Hypertext
   Transport Protocol (HTTP) [11]. (RFC 2068 [10]). The following terms have
   special significance for SIP.

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

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

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

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

   Final response: A response that terminates a SIP transaction, as
        opposed to a provisional response that does not. All 2xx, 3xx,
        4xx, 5xx and 6xx responses are final.

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

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

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

   Location server: See location service

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

   Provisional response: A response used by the server to indicate
        progress, but that does not terminate

   Parallel search: In a SIP transaction. All 1xx
        and 6xx responses are provisional. Other responses are
        considered final.

   Proxy, proxy server: An intermediary program that acts as both parallel search, a proxy issues several
        requests to possible user locations upon receiving an incoming
        request.  Rather than issuing one request and then waiting for
        the final response before issuing the next request as in a
        sequential search , a parallel search issues requests without
        waiting for the result of previous requests.

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

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

   Redirect server: A redirect server is a server that accepts a SIP
        request, maps the address into zero or more new addresses and
        returns these addresses to the client. Unlike a proxy server, server ,
        it does not initiate its own SIP request. Unlike a user agent server,
        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.

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

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

   Session: "A multimedia session is a set of multimedia senders and
        receivers and the data streams flowing from senders to
        receivers. A multimedia conference is an example of a multimedia
        session." [9] (Note: a (RFC 2327, [7]) (A session as defined here for SDP may
        comprise one or more RTP sessions.) Since As defined, a callee may be
        invited several times, by different calls, to the word same session.
        If SDP is used, a session is used
        differently defined by protocols relevant to SIP, this document avoids the term altogether. concatenation of the
        user name , session id , network type , address type and address
        elements in the origin field.

   (SIP) transaction: A SIP transaction occurs between a client and a
        server and comprises all messages from the first request sent
        from the client to the server up to a final (non-1xx) response
        sent from the server to the client. A transaction is for a
        single call (identified by a  Call-ID, Section 6.12). There 6.12) and may be
        identified by a  CSeq sequence number (Section 6.16).  If there
        is no  CSeq, there can only be one pending transaction between a
        server and client for each call id.

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

   URL-encoded: A character string encoded according to RFC 1738,
        Section 2.2 [11].

   User agent server, called client (UAC), calling user agent: The server A user agent client is a
        client application that
        contacts initiates the user when a SIP request is received and request.

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

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

1.4 Summary of SIP Operation

   This section explains the basic protocol functionality and operation.

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

   Table 1: Properties of the different SIP server types

1.4 Summary of SIP Operation

   This section explains the basic protocol functionality and operation.

   Callers and callees are identified by SIP addresses, described in
   Section 1.4.1. When making a SIP call, a caller first locates the
   appropriate server (Section 1.4.2) and then sends a SIP request
   (Section 1.4.3). The most common SIP operation is the invitation
   (Section 1.4.4). Instead of directly reaching the intended callee, a
   SIP request may be redirected or may trigger a chain of new SIP
   requests by proxies (Section 1.4.5). Users can register their
   location(s) with SIP servers (Section 4.2.5). 4.2.6).

1.4.1 SIP Addressing

   SIP addresses contain a user and host part. The user part is an
   operating-system a user name.
   name, a civil name or a telephone number. The host part is either a
   domain name having a DNS SRV (RFC 2052 [12]), MX (RFC 974 [13], CNAME
   or A (address) record, record (RFC 1035 [14]), or a numeric network address.
   Examples include:

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

   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.

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

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

   Examples of SIP names include

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

     mjh@metro.isi.edu
     watson@bell-telephone.com
     root@[193.175.132.42]
     root@193.175.132.42
   An address can designate an individual (possibly located at one of
   several end systems), the first available person from a group of
   individuals or a whole group. The form of the address, e.g.,
   sales@example.com , is not sufficient, in general, to determine the
   intent of the caller.

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

   When used within SIP, SIP addresses are written as SIP URLs (Section
   sec:url), e.g., sip://info@ietf.org as

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

     sip:info@ietf.org

1.4.2 Locating a SIP Server
_________________________

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

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

        1.   If client first attempts
   to contact the SIP address 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, contact a the client contacts
   the SIP server at that address.

        2.   If Otherwise, the SIP address does not contain a port number and if client follows the
   steps below.

        1.   If there is a SRV DNS resource record [13] (RFC 2052 [12]) of
             type  sip.udp, contact the listed SIP servers in the order
             of the preference values contained in those resource
             records, using UDP as a transport protocol at the port
             number given in the URL or, if none provided, the one
             listed in the DNS resource record. [TBD: What if the SIP URL
             contains a port number?]

        3.

        2.   If the SIP address does not contain a port number and if there is a SRV DNS resource record [13] (RFC 2052 [12]) of
             type  sip.tcp, contact the listed SIP servers in the order
             of the preference value contained in those resource
             records, using TCP as a transport protocol at the port
             number given in the URL or, if none provided, the one
             listed in the DNS resource record.

        4.

        3.   If there is a DNS MX record [12], (RFC 974 [13]), contact the
             hosts listed in their order of preference at the port
             number listed in the URL or the default SIP port number
             (TBD). if
             none. For each host listed, first try to contact the SIP
             server using UDP, then TCP.

        5.

        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.

        6.

   If all of the above methods fail, 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.

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

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

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

        This sequence is modeled after that described for SMTP,
        where MX records are to be checked before A records [15]. (RFC
        1123 [15]).

1.4.3 SIP Transaction

   Once the host part has been resolved to a SIP server, the client
   sends one or more SIP requests to that server and receives one or
   more responses from the server. A request (and its retransmissions)
   together with the responses triggered by that request make up a SIP
   transaction.  The  ACK request following an  INVITE is not part of
   the transaction since it may traverse a different set of hosts.

   If TCP is used, request and responses within a single SIP transaction
   are carried over the same TCP connection. Thus, the client SHOULD
   maintain the connection until a final response has been received. (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 sends via unicast UDP, the response is sent
   to the source address of the UDP request.
   (Implementation note: use recvfrom() to obtain contained in the source address and
   port next  Via header field (Section 6.43)
   of the request.) 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
   11).

        Need motivation why we ALWAYS want to have a multicast
        return.
   10).

   The SIP message format and operation is independent of the transport
   protocol.

1.4.4 SIP Invitation

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

   The  INVITE request typically contains a session description, for
   example written in SDP (RFC 2327, [7]) format, that provides the
   called party with enough information to join the session. For
   multicast sessions, the session description enumerates the media
   types and formats that may be distributed to that session. For
   unicast session, the session description enumerates the media types
   and formats that the caller is willing to receive and where it wishes
   the media data to be sent. In either case, if the callee wishes to
   accept the call, it responds to the invitation by returning a similar
   description listing the media it wishes to receive. For a multicast
   session, the callee should only return a session description if it is
   unable to receive the media indicated in the caller's description.
   The caller may ignore the session description returned or use it to
   change the global session description.

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

   The protocol exchanges for the  INVITE method are shown in Fig. 1 for
   a proxy server and in Fig. 2 for a redirect server. The In Fig. 1, the
   proxy server accepts the  INVITE request (step 1), contacts the
   location service with all or parts of the address (step 2) and
   obtains a more precise location (step 3). The proxy server then
   issues a SIP  INVITE request to the address(es) returned by the
   location service (step 4). The user agent server alerts the user
   (step 5) and returns a success indication to the proxy server (step
   6). The proxy server then returns the success result to the original
   caller (step 7). The receipt of this message is confirmed by the
   caller using an  ACK
   message, request, which is forwarded to the callee (steps
   8 and 9), with a
   response returned (steps 10 and 11). 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). 4), which is then acknowledged via an
   ACK request (step 5). The caller issues a new request, with the same
   call-ID but a new call-ID, higher  CSeq, to the address returned by the first
   server (step 6). In the example, the call succeeds (step 7). The
   caller and callee complete the
   handshanke handshake with an  ACK (steps 8 and 9). (step 8).

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

1.4.5 Locating a User

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

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 (Section 4.2.5) or a location server, typically for a single
   administrative domain, or a (Sections 1.4.7, 4.2.6). A location server may also use one or
   more other protocols, such as finger [16], rwho, (RFC 1288 [16]), rwhois (RFC
   2167 [17]), LDAP (RFC 1777 [18]), multicast-based protocols or operating-
   system
   operating-system dependent mechanism to actively determine the end
   system where a user might be reachable. The location services yield a list of a
   zero or more possible locations, possibly even sorted in order of
   likelihood of success.

   The A location server can be part of the SIP server or the SIP server
   may use a different protocol (e.g., finger [16] or LDAP [17]) to map
   addresses. A single user may be registered at different locations,
   either return
   several locations because she 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 simply returns the list to the
   client sending the request as  Location headers (Section 6.18). 6.25). A SIP
   proxy server can sequentially or in parallel try the addresses
                                            +....... cs.columbia.edu .......+
                                            :                               :
                                            : (~~~~~~~~~~)                  :
                                            : ( location )                  :
                                            : ( service  )                  :
                                            : (~~~~~~~~~~)                  :
                                            :   ^      |                    :
                                            :   |   hgs@play                :
                                            :  2|     3|                    :
                                            :   |      |                    :
                                            : henning  |                    :
   +.. cs.tu-berlin.de ..+ 1: INVITE        :   |      |                    :
   :                     :    henning@cs.col:   |      |                    :
   : cz@cs.tu-berlin.de =======================>  tune                      :
   :         ^ |        <.......................                            :
   :         . |         : 4: 302 Moved     :                               :
   +...........|.........+    hgs@play      :                               :
             . |                            :                               :
             . | 5: INVITE hgs@play.cs.columbia.edu                6: ring  :
             . ==================================================> play     :
             .....................................................          :
               7: 200 OK                    :                               :
                                            +...............................+

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

   Figure 2: Example of SIP redirect server until
   the call is successful (2xx response) or the callee has declined the
   call (60x (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 6.33) 6.43)
   headers. The  Via trace ensures that replies can take the same path
   back, thus ensuring correct operation through compliant firewalls and
   loop-free requests.
   avoiding request loops. On the reply response path, each host most MUST remove
   its  Via, so that routing internal information is hidden from the
   callee and outside networks. When a multicast request is made, first
   the host making the request, then the multicast address itself are
   added to the path. A proxy server MUST check that it does not
   generate a request to a host listed in the  Via list.  (Note: If a
   host has several names or network addresses, this may not always
   work. Thus, each host also checks if it is part of the Via list.)

   A SIP invitation may traverse more than one SIP proxy server. If one
   of these "forks" the request, i.e., issues more than one request in
   response to receiving the invitation request, it is possible that a
   client is reached, independently, by more than one copy of the
   invitation request. Each of these copies bears the same Call-ID. The
   user agent MUST return the appropriate status response, but
   SHOULD NOT alert the user.

   As discussed in Section 1.4.1, a SIP address may designate a group
   rather than response. Duplicate
   requests are not an individual. A client indicates using the  Reach
   request header whether it wants to reach the first available
   individual or treat the address as a group, to be invited as a whole.
   The default is to attempt to reach the first available callee.  If
   the address error, so there is designated as a group address, a proxy server MUST
   return the list of individuals instead of attempting to connect no need to
   these. (Otherwise, the proxy cannot report errors, redirections and
   call status individually. For example, some may be contacted
   successfully, while one of alert the group may be reachable under a
   different address.) user.

1.4.6 Changing an Existing Session

   In some circumstances, it may be necessary to change the parameters
   of an existing session. For example, two parties may have been
   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 and  UNREGISTER requests allow request allows a client to let a proxy or redirect
   server know which address address(es) it may be reached under. A client may
   also use it to install call handling features at the server.

1.5 Protocol Properties

1.5.1 Minimal State

   A single conference session or call may involve one or more SIP
   request-response transactions. Proxy server servers do not have to keep
   state for a particular call, however, they maintain state for a
   single SIP transaction, as discussed in Section 12. 11.

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

1.5.2 Transport-Protocol 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. protocols, among others. UDP allows the application to more
   carefully control the timing of messages and their retransmission, to
   perform parallel searches without requiring TCP connection state for
   each outstanding request, and to use multicast. Routers can more
   readily snoop SIP UDP packets. TCP allows easier passage through
   existing firewalls, and given the similar protocol design, allows
   common servers for SIP, HTTP and the Real Time Streaming Protocol
   (RTSP) [1].

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

   Clients

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

   For concreteness, this document will only refer to Internet
   protocols.  However, SIP may also be used directly with protocols
   such as ATM AAL5, IPX, frame relay or X.25. The necessary naming
   conventions are beyond the scope of this document.

1.5.3 Text-Based

   SIP is text based. 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 originator,
   current destination and final recipient of a SIP request, and to
   specify redirection addresses. A SIP URL may 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 SIP
   URL scheme is defined to allow setting SIP  request-header fields and
   the SIP  message-body. (This is similar to the  mailto: URL.)  URL [19].)

   A SIP URL follows the guidelines of RFC 1630 [18,19] 1630, as revised, [20,21] and
   takes the following form:

        SIP-URL                 =    short-sip-url |    full-sip-url
        full-sip-url            =    "sip://"    "sip:" ( user | phone ) [ ":" password ]
                                     "@" [ host | nhost ] : port
                                     url-parameters [ headers ]
        short-sip-url      =    ( user | phone) [ ":" password ]
                                "@" [ host | nhost ] : port
        user                    =    ;  defined in RFC 1738 [20] [11]
        phone                   =    "+" DIGIT    telephone-subscriber
        telephone-subscriber    =    ;
        defined in [22]
        password                =    *( DIGIT unreserved | "-" escaped | "."
                                     ";" | "                                     " | "=" | "+" | "$" | "," )
        host                    =    ;  defined in RFC 1738
        nhost                   =    "[" hostnumber "]" |    hostnumber
        hostnumber              =    digits "." digits "." digits "." digits
        port                    =    *digit    digits
        url-parameters          =    *( ";" url-parameter)
        url-parameter           =    transport-param |
                                     ttl-param | maddr-param | other-param
        transport-param         =    "transport=" ( "udp" | "tcp" )
        ttl-param               =    "ttl=" ttl
        ttl                     =    1*3DIGIT                                      ; 0 to 255
        maddr-param             =    "maddr=" maddr
        maddr                   =    ;  dotted decimal multicast address
        other-param             =    *uric
        headers                 =    "?" header *( "                             " header )
        header                  =    hname "=" hvalue
        hname                   =    *urlc    *uric
        hvalue                  =    *urlc
        urlc    *uric
        uric                    =    ;  defined in [19] [21]
        digits                  =    1*digit    1*DIGIT

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

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

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

   The  password parameter can be used for a basic authentication
   mechanism that takes the place of an unlisted telephone number. Also,
   for Internet telephony gateways, it may serve as a PIN. personal
   identification number (PIN). Including just the password in the URL
   is more convenient than including a whole authentication header. This
   approach may be reasonably secure if the URL is part of a secure secured web
   page. Unless the SIP transaction is carried over a secure network
   connection, this carries the same security risks as all URL-based
   passwords and should only be used when security requirements are low.
   In almost all circumstances, general, it is NOT RECOMMENDED and use of the Authorization
   (Section 6.10) 6.11) header is preferred.

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

   Examples of short and full-form SIP URLs are:

     j.doe@big.com
     sip://j.doe@big.com
     sip://j.doe:secret@big.com;transport=tcp
     sip://j.doe@big.com?subject=project
     sip://+1-212-555-1212:1234@gateway.com
     sip://alice@[10.1.2.3]
     sip://alice@10.1.2.3
     sip:j.doe@big.com
     sip:j.doe:secret@big.com;transport=tcp
     sip:j.doe@big.com?subject=project
     sip:+1-212-555-1212:1234@gateway.com
     sip:alice@[10.1.2.3]
     sip:alice@10.1.2.3

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

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

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

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

   SIP URLs can define specific parameters of the request, including the
   transport mechanism (UDP or TCP) and the use of multicast to make a
   request. These parameters are added after the  host and are separated
   by semi-colons. For example, to specify to call j.doe@big.com using
   multicast to 239.255.255.1 with a ttl of 15, the following URL would
   be used:

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

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

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

3 SIP Message Overview

   Since
   SIP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [23]). Lines are terminated by CRLF, but
   receivers should be prepared to also interpret CR and LF by
   themselves as line terminators.

   Except for the above difference in character sets, much of the
   message syntax is identical to HTTP/1.1, rather than repeating it
   here we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1
   specification [11]. (RFC 2068 [10]). In addition, we describe SIP in both
   prose and an augmented Backus-Naur form (BNF) [H2.1] described in
   detail in [21].

   All SIP messages are text-based and use HTTP/1.1 conventions [H4.1],
   except for the additional ability of RFC 2234 [24].

   Unlike HTTP, SIP to 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 1500 1400 bytes if the MTU is unknown.

        The 1400 bytes accommodates lower-layer packet headers
        within the "typical" MTU of around 1500 bytes. There are
        few MTU values around 1 kB; the next value is 1006 bytes
        for SLIP and 296 for low-delay PPP [22]. Recent
        studies
        [23] [25] indicate that an MTU of 1500 bytes is a
        reasonable assumption. The next lower common MTU values are
        1006 bytes for SLIP and 296 for low-delay PPP (RFC 1191
        [26]). Thus, another reasonable value would be a message
        size of 950 bytes, to accommodate packet headers within the
        SLIP MTU without fragmentation.

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

        SIP-message  ___   Request | Response  ; SIP messages

   Both  Request (section 4) and  Response (section 5) messages use the
   generic message
   generic-message format of RFC 822 [24] [27] for transferring entities (the
   payload
   body of the message). Both types of message consist of a  start-
   line,  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 line-feed (
   CRLF)) indicating the end of the header fields, and an optional
   message-body. To avoid confusion with similar-named headers in HTTP,
   we refer to the header describing the message body as entity headers.
   These components are described in detail in the upcoming sections.

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

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

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

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

4 Request

   The  Request message format is shown below:

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

4.1 Request-Line

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

        Request-Line  ___   Method SP Request-URI SP SIP-Version CRLF
        general-header     =     Call-ID                ; Section 6.12
                           |     CSeq                   ; Section 6.26 6.16
                           |     Date                   ; Section 6.15 6.17
                           |     Encryption             ; Section 6.18
                           |     Expires                ; Section 6.16 6.20
                           |     From                   ; Section 6.17 6.21
                           |     Via     Record-Route           ; Section 6.33
                           |     Timestamp              ; Section 6.39
                           |     Via                    ; Section 6.43
        entity-header      =     Content-Length     Content-Encoding       ; Section 6.13
                           |     Content-Type     Content-Length         ; Section 6.14
                           |     Content-Type           ; Section 6.15
                           |     ETag                   ; Section 6.19
        request-header     =     Accept                 ; Section 6.6 6.7
                           |     Accept-Encoding        ; Section 6.8
                           |     Accept-Language        ; Section 6.7 6.9
                           |     Authorization          ; Section 6.10 6.11
                           |     Call-Disposition     Hide                   ; Section 6.11 6.22
                           |     If-Match               ; Section 6.23
                           |     If-None-Match          ; Section 6.24
                           |     Location               ; Section 6.25
                           |     Max-Forwards           ; Section 6.26
                           |     Organization           ; Section 6.19 6.27
                           |     Priority               ; Section 6.20 6.28
                           |     Proxy-Authorization    ; Section 6.22 6.30
                           |     Proxy-Require          ; Section 6.31
                           |     Route                  ; Section 6.35
                           |     Require                ; Section 6.24 6.32
                           |     Response-Key           ; Section 6.34
                           |     Subject                ; Section 6.28 6.38
                           |     To                     ; Section 6.31 6.40
                           |     User-Agent             ; Section 6.32 6.42
        response-header    =     Location     Allow                  ; Section 6.18 6.10
                           |     Proxy-Authenticate     Location               ; Section 6.21 6.25
                           |     Public     Proxy-Authenticate     ; Section 6.23 6.29
                           |     Retry-After            ; Section 6.25 6.36
                           |     Server                 ; Section 6.27 6.37
                           |     Unsupported            ; Section 6.29 6.41
                           |     Warning                ; Section 6.34 6.44
                           |     WWW-Authenticate       ; Section 6.35 6.45

   Table 2: SIP headers

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

4.2 Methods

   The methods are defined below. Methods that are not supported by a
   proxy or redirect server SHOULD be 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 should cause a
   "501 Not Implemented" 501
   (Not Implemented) response to be returned (Section 7).

        method

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

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

   A

   If a user agent agents receives an  INVITE with a new  CSeq sequence
   number, it MUST check any version identifiers in the session
   description or, if there are no version identifiers, the content of
   the session description to see if it has changed. If the version number 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 accomodate
   accommodate the capabilities of new arrivals to a conference conference, add or
   delete media or change from a unicast to a multicast conference.)

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

4.2.2  ACK

   The  ACK request confirms that the client has received a final
   response to an  INVITE request. See Section 11 for details. This method MUST be
   supported 2xx responses are acknowledged by a SIP server and client.

4.2.3  OPTIONS

   The
   client user agents, all other final responses by the first proxy or
   client user agent to receive the response. The  Via is being queried as always
   initialized to its capabilities. A server the host that
   believes it can contact originates the user, such as a  ACK request, i.e., the
   client user agent after a 2xx response or the first proxy to receive
   a non-2xx final response. The  ACK request is forwarded as the
   corresponding  INVITE request, based on its  Request-URI. See Section
   10 for details. This method MUST be supported by a SIP server and
   client.

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

4.2.3  OPTIONS

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

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

4.2.4  BYE

   The user agent client indicates uses  BYE to indicate to the server that it
   wishes to abort the call
   attempt. The leaving party can use a  Location header field to
   indicate that the recipient of call. A  BYE request should contact is forwarded like an INVITE
   request. It terminates any on-going searches for the named
   address.  This implements the "call transfer" telephony
   functionality. call. A client
   caller SHOULD also use this method to indicate to
   the callee that it wishes to abort an on-going issue a  BYE request before aborting a call attempt.

        With UDP, ("hanging
   up"). Note that a  BYE request may also be issued by the caller has no other way to signal her intent
        to drop callee.

   If the call attempt and  INVITE request contained a  Location header, the callee side will keep
        "ringing".  When using TCP, a client MAY also close sends
   the
        connection  BYE request to abort a call attempt. Support of this that address rather than the From address.

   This method
        is OPTIONAL.

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

4.2.5  CANCEL

   The  CANCEL request cancels any pending searches, but does not
   terminate an accepted call at a particular user agent. (A call is
   considered accepted if the callee has returned a 200 (OK) status
   response.) A proxy SHOULD issue a  CANCEL request to all destinations
   that have not yet returned a final response after it has received a
   2xx or 6xx response for one or more of this method the parallel-search requests.
   A proxy that receives a  CANCEL request forwards the request to all
   destinations with pending requests triggered by an INVITE. The
   Call-ID,  To and  From in the CANCEL request are identical to those
   contained in the  CANCEL request, but the  Via header field is OPTIONAL.

4.2.5
   initialized to the proxy issuing the  CANCEL request.

   A redirect server or user agent server returns 200 (OK) if the Call-
   ID exists and 481 (Invalid Call-ID) if not, but takes no further
   action. In particular, any existing call is unaffected.

        The  BYE request cannot be used to cancel branches of a
        parallel search, since several branches may, through
        intermediate proxies, find the same user agent server and
        then terminate the call.

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

4.2.6  REGISTER

   A client uses the  REGISTER method to register the address listed in
   the request line  To header to a SIP server. The host part of the request-URI

   A user agent SHOULD correspond register with a local server on startup by
   sending a  REGISTER request to (one of the aliases of) name well-known "all SIP servers"
   multicast address, 224.0.1.75, with a time-to-live value of 1.

   SIP user agents on the server or same subnet MAY listen to
   the domain that address and use
   it represents, if location-independent. After
   registration, to become aware of the location of other local users [28];
   however, they do not respond to the request.

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

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

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

   Request-URI: The  Request-URI names the destination of the
        registration request, i.e., the domain of the registrar. The
        user name MUST be empty. Generally, the domains in the
        Request-URI and the  To header have the same value; however, it
        is possible to register as a "visitor", while maintaining one's
        name. For example, a traveller alice@acme.com may register under
        @atlanta.ayh.org

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

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

   Otherwise, future call control requests will be directed to the
   network source address of the  REGISTER request, using the  To
   address in the  REGISTER request as the  Request-URI. If the
   registration is changed while a user agent or proxy server processes
   an invitation, the new information should be used.

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

   After registration, the server MAY forward incoming SIP requests to
   the the network source address and port from that originated the registration
   request. A server SHOULD silently drop the registration after one
   hour, unless refreshed by the client. A client may request and a server may
   indicate or lower or
   higher refresh interval and indicate the
   interval through the  Expires header (Section 6.16). A single 6.20).
   Based on this request and its configuration, the server chooses the
   expiration interval and indicates it through the  Expires header in
   the response. A single address (if host-independent) may be
   registered from several different clients.

   If the

   A client cancels an existing registration by sending a  REGISTER
   request contains with an expiration time ( Expires) of zero seconds for a
   particular  Location header, requests for or the
   request-URI will be directed to wildcard  Location designated by a "*"
   for all registrations.

   The server SHOULD return the address(es) given.

   Support current list of this method is OPTIONAL. registrations in the 200
   response as  Location header fields.

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

4.2.6  UNREGISTER

   A client cancels an existing registration established for the
   Request-URI with  REGISTER with the  UNREGISTER method. If it
   unregisters Since a  Request-URI unknown client may not have easy access to the servers, host
        address or port number, using the server returns
   a 200 (OK) response. source address and port
        from the request itself seems simpler.

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

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

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

4.3.1 SIP Version

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

4.4 Option Tags

   Option tags are unique identifiers used to designate new options in
   SIP.  These tags are used in  Require (Section 6.24) 6.32) and Unsupported
   (Section 6.29) 6.41) fields.

   Syntax:

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

   The creator of a new SIP option should either prefix the option with
   a reverse domain name (e.g., 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"), or
   register the new option with the Internet Assigned Numbers Authority
   (IANA).

4.4.1 Registering New Option Tags "foo.com".  Options registered with IANA

   When registering a new SIP option, have the following information
   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-
   insensitive.

4.4.1 Registering New Option Tags with IANA

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

        oName

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

        oIndication

        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
          companies);

        oA

        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;

        oFor

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

        Borrowed from RTSP and the RTP AVP.

5 Response

   After receiving and interpreting a request message, the recipient
   responds with a SIP response message. The response message format is
   shown below:

        Response    =    Status-Line          ;  Section 5.1
                         *( general-header
                         | response-header
                         | entity-header )
                         CRLF
                         [ message-body ]     ;  Section 8

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

5.1 Status-Line

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

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

5.1.1 Status Codes and Reason Phrases

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

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

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

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

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

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

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

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

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

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

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

   Figure 3: Informational and success status codes

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

   Figure 4: Redirection status codes

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

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

   Figure 5: Client error status codes

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

   Figure 6: Server error status codes

   SIP header fields response codes are similar extensible. SIP applications are not required
   to HTTP header fields in both syntax
   and semantics [H4.2], [H14]. In general understand the ordering meaning of the header
   fields all registered response codes, though
   such understanding is not of importance (with obviously desirable. However, applications MUST
   understand the exception class of  Via fields, see
   below), but proxies MUST NOT reorder or otherwise modify header
   fields other than by adding 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
        Global-Failure   |    "600"    ;  Busy
                         |    "603"    ;  Decline
                         |    "604"    ;  Does not exist anywhere
                         |    "606"    ;  Not Acceptable

   Figure 7: Global failure status Codes

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

6 Header Field Definitions

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

   The header fields required, optional and not applicable for each
        Global-Failure   |    "600"    ;  Busy
                         |    "603"    ;  Decline
                         |    "604"    ;  Does not exist anywhere
                         |    "606"    ;  Not Acceptable

   Figure 7: Global failure status Codes
   method are listed in Table 3. The 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).

   Other headers may be added as required; a

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

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

   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 10 9 for use over UDP when the request
   has to fit into a single packet and size is an issue.

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

6.1 General Header Fields

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

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

6.2 Entity Header Fields

   Entity-header fields define meta-information about the message-body
   or, if no body is present, about the resource identified by the
   request. The term "entity header" is an HTTP 1.1 term where the reply
   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
                             type    ACK   BYE   INV   OPT   REG   UNR
     _________________________________________________________________
     Accept                   R       o     -     o     o     o     o
     Accept-Language          R       o     o     o     o     o     o
     Allow                   405      o     o     o     o     o     o
     Also                     R       -     -     o     -     -     -
     Authorization            R       o     o     o     o     o     o
     Call-Disposition         R       -     o     o     -     -     -
     Call-ID                  g       m     m     m     o     -     -
     Content-Length           g       -     -     *     *     -     -
     Content-Type             g       -     -     *     *     -     -
     CSeq                     g       o     o     o     o     o     o
     Date                     g       o     o     o     o     o     o
     Expires                  g       -     -     o     o     o     -
     From                     R       m     m     m     m     o     o
     Location                 R       -     o     -     -     o     -
     Location                 r       -     -     o     o     -     -
     Organization             R       -     -     o     o     -     -
     Proxy-Authenticate       R       o     o     o     o     o     o
     Proxy-Authorization      R       o     o     o     o     o     o
     Priority                 R       -     -     o     -     -     -
     Public                   r       -     -     -     o     -     -
     Require                  R       o     o     o     o     o     o
     Retry-After           600,603    -     -     o     -     -     -
     Server                   r       o     o     o     o     o     o
     Subject                  R       -     -     o     -     -     -
     Timestamp                g       o     o     o     o     o     o
     To                       g       m     m     m     m     m     m
     Unsupported              r       o     o     o     o     o     o
     User-Agent               R       o     o     o     o     o     o
     Via                      g       m     m     m     m     m     m
     Warning                  r       o     o     o     o     o     o
     WWW-Authenticate        401      o     o     o     o     o     o

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

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

6.4 Response Header Fields

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

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

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

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

6.6 Header Field Format

   Header fields ( general-header,  request-header, response-header, and
   entity-header) follow the same generic header format as that given in
   Section 3.1 of RFC 822 [24]. [27,29].

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

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

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

6.6 Accept

   See [H14.1]
                           type     enc.    e-e   ACK   BYE   CAN   INV   OPT   REG
   ________________________________________________________________________________
   Accept                   R                e     o     o     o     o     o     o
   Accept-Encoding          R                e     o     o     o     o     o     o
   Accept-Language          R        n       e     o     o     o     o     o     o
   Allow                   405               e     o     o     o     o     o     o
   Authorization            R                e     o     o     o     o     o     o
   Call-ID                  g        n       e     m     m     m     m     o     -
   Content-Encoding         e                e     *     -     -     *     *     *
   Content-Length           e                e     m     -     -     m     m     m
   Content-Type             e                e     *     -     -     *     *     *
   CSeq                     g        n       e     m     m     m     m     m     o
   Date                     g                e     o     o     o     o     o     o
   Encryption               g        n       e     o     o     o     o     o     o
   ETag                    200               e     -     -     -     o     -     -
   Expires                  g                e     -     -     -     o     o     o
   From                     R                e     m     m     m     m     m     m
   Hide                     R        n       h     o     o     o     o     o     o
   If-Match                 R                e     o     o     -     -     -     -
   If-None-Match            R                e     o     o     -     -     -     -
   Location                 R                e     -     -     -     -     -     o
   Location                3xx               e     -     -     o     o     o     o
   Location                2xx               e     -     -     o     o     o     -
   Max-Forwards             R        n       e     o     o     o     o     o     o
   Organization             R        c       e     -     -     -     o     o     o
   Proxy-Authenticate      407       n       h     o     o     o     o     o     o
   Proxy-Authorization      R        n       h     o     o     o     o     o     o
   Proxy-Require            R        n       h     o     o     o     o     o     o
   Priority                 R        c       e     -     -     -     o     -     -
   Require                  R        n       e     o     o     o     o     o     o
   Retry-After              R        c       e     -     -     -     -     -     o
   Retry-After           600,603     c       e     -     -     -     o     -     -
   Response-Key             R        c       e     -     o     o     o     o     o
   Record-Route             R                h     o     o     o     o     o     o
   Record-Route            2xx               h     o     o     o     o     o     o
   Route                    R                h     -     o     o     o     o     o
   Server                   r        c       e     o     o     o     o     o     o
   Subject                  R        c       e     -     -     -     o     -     -
   Timestamp                g                e     o     o     o     o     o     o
   To                       g        n       e     m     m     m     m     m     m
   Unsupported             420               e     o     o     o     o     o     o
   User-Agent               R        c       e     o     o     o     o     o     o
   Via                      g        n       e     m     m     m     m     m     m
   Warning                  r                e     o     o     o     o     o     o
   WWW-Authenticate        401       c       e     o     o     o     o     o     o

   Table 3: Summary of header fields
   separated list (i.e.,  #(values)). It MUST be possible to combine the
   multiple header fields into one "field-name: field-value" pair,
   without changing the semantics of the message, by appending each
   subsequent field-value to the first, each separated by a comma. The
   order in which header fields with the same field-name are received is
   therefore significant to the interpretation of the combined field
   value, and thus a proxy MUST NOT change the order of these field
   values when a message is forwarded.

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

6.7  Accept

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

   Example:

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

6.8  Accept-Encoding

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

6.9  Accept-Language

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

   Example:

     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] and [30]. A user agent that wishes to authenticate itself
   with a server -- usually, but not necessarily, after receiving a 401
   response -- MAY do so by including an  Authorization request-header
   field with the request. The Authorization field value consists of
   credentials containing the authentication information of the user
   agent for the realm of the resource being requested.

6.12  Call-ID

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

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

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

   The  Call-ID may be any string consisting of the unreserved URI
   characters that can be guaranteed to be globally unique for the
   duration of the request.  Call-IDs are case-sensitive and are not
   URL-encoded.

        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.

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

        Using cryptographically random identifiers provides some
        protection against session hijacking.

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

   Example:

     Call-ID: 3436538253725150855@foo.bar.com

6.13  Content-Encoding

   The  Content-Encoding entity-header field is used as a modifier to
   the media-type. When present, its value indicates what additional
   content codings have been applied to the entity-body, and thus what
   decoding mechanisms MUST be applied in order to obtain the media-type
   referenced by the  Content-Type header field.  Content-Encoding is
   primarily used to allow a document to be compressed without losing
   the identity of its underlying media type.  See [H14.11].

6.14  Content-Length

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

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

   An example is

     Content-Length: 3495

   Applications MUST use this field to indicate the size of the
   message-body to be transferred, regardless of the media type of the
   entity. Any  Content-Length greater than or equal to zero is a valid
   value. If no body is present in a message, then the Content-Length
   header MUST be set to zero. If a server receives a message without
   Content-Length, it MUST assume it to be zero.  Section 8 describes
   how to determine the length of the message body.

6.15  Content-Type

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

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

   Example of this header field are

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

6.16  CSeq

   Clients MUST add the  CSeq (command sequence) general-header field to
   every request. A  CSeq request header field contains a single decimal
   sequence number chosen by the requesting client and the request
   method. The sequence number MUST be expressible as a 64-bit unsigned
   integer. The initial value of the sequence number is arbitrary.
   Consecutive requests that differ in request method, headers or body,
   but have the same  Call-ID MUST contain strictly monotonically
   increasing and contiguous sequence numbers; sequence numbers do not
   wrap around. Retransmissions of the same request carry the same
   sequence number, but an  INVITE with a different message body (a
   "re-invitation") acquires a new, higher sequence number. A server
   responding to a request containing a  CSeq header MUST echo the value
   in the response. If the  Method value is missing, the server fills it
   it appropriately.

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

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

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

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

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

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

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

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

   Example:

     CSeq: 4711 INVITE

6.17  Date

   General header field. See [H14.19].

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

6.18  Encryption

   The  Encryption general-header field specifies that the content has
   been encrypted. Section 12 describes the overall SIP security
   architecture and algorithms. It is intended for end-to-end encryption
   of requests and responses. Requests are encrypted with a public key
   belonging to the entity named in the  To header field. Responses are
   encrypted with the public key conveyed in the Response-Key header
   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-params
        encryption-scheme    =    token
        encryption-params    =    token "=" ( token | quoted-string )

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

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

   INVITE sip:watson@boston.bell-telephone.com SIP/2.0
   Via: SIP/2.0/UDP 169.130.12.5
   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

   hQEMAxkp5GPd+j5xAQf/ZDIfGD/PDOM1wayvwdQAKgGgjmZWe+MTy9NEX8O25Red
   h0/pyrd/+DV5C2BYs7yzSOSXaj1C/tTK/4do6rtjhP8QA3vbDdVdaFciwEVAcuXs
   ODxlNAVqyDi1RqFC28BJIvQ5KfEkPuACKTK7WlRSBc7vNPEA3nyqZGBTwhxRSbIR
   RuFEsHSVojdCam4htcqxGnFwD9sksqs6LIyCFaiTAhWtwcCaN437G7mUYzy2KLcA
   zPVGq1VQg83b99zPzIxRdlZ+K7+bAnu8Rtu+ohOCMLV3TPXbyp+err1YiThCZHIu
   X9dOVj3CMjCP66RSHa/ea0wYTRRNYA/G+kdP8DSUcqYAAAE/hZPX6nFIqk7AVnf6
   IpWHUPTelNUJpzUp5Ou+q/5P7ZAsn+cSAuF2YWtVjCf+SQmBR13p2EYYWHoxlA2/
   GgKADYe4M3JSwOtqwU8zUJF3FIfk7vsxmSqtUQrRQaiIhqNyG7KxJt4YjWnEjF5E
   WUIPhvyGFMJaeQXIyGRYZAYvKKklyAJcm29zLACxU5alX4M25lHQd9FR9Zmq6Jed
   wbWvia6cAIfsvlZ9JGocmQYF7pcuz5pnczqP+/yvRqFJtDGD/v3s++G2R+ViVYJO
   z/lxGUZaM4IWBCf+4DUjNanZM0oxAE28NjaIZ0rrldDQmO8V9FtPKdHxkqA5iJP+
   6vGOFti1Ak4kmEz0vM/Nsv7kkubTFhRl05OiJIGr9S1UhenlZv9l6RuXsOY/EwH2
   z8X9N4MhMyXEVuC9rt8/AUhmVQ==
   =bOW+

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

6.19  ETag

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

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

   The entity tag consists of an opaque quoted string. An entity tag
   MUST be unique across all instances associated with a particular To
   URI. A given entity tag value may be used for different URIs without
   implying anything about the equivalence of those URIs. It is
   RECOMMENDED that the entity tag is a cryptographically random
   identifier with at least 32 bits of randomness.

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

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

6.20  Expires

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

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

   For  INVITE, it is a request and response-header field. In a request,
   the callee can limit the validity of an invitation. (For example, if
   a client wants to limit how long a search should take at most or when
   a conference invitation is time-limited. A user interface may take
   this is as a hint to leave the invitation window on the screen even
   if the user is not currently at the workstation.) This also limits
   the duration of a search. If the request expires before the search
   completes, the proxy returns a 408 (Request Timeout) status.  In a
   302 (Moved Temporarily) response, a server can advise the client of
   the maximal duration of the redirection.

   The value of this field can be either an  HTTP-date or an integer
   number of seconds (in decimal), measured from the receipt of the
   request.  The latter approach is preferable for short durations, as
   it does not depend on clients and servers sharing a synchronized
   clock.

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

   Two example of its use are
     Expires: Thu, 01 Dec 1994 16:00:00 GMT
     Expires: 5

6.21  From

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

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

   Examples:

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

6.22  Hide

   The  Hide request header field indicates that the path comprised of
   the  Via header fields (Section 6.43) should be hidden from
   subsequent proxies and user agents. It can take two forms:  Hide:
   route and  Hide:hop.  Hide header fields are typically added by the
   client user agent, but MAY be added by any proxy along the path.

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

   A server hides the previous hop by encrypting the  host and port
   parts of the top-most  Via header with an algorithm of its choice.
   Servers SHOULD add additional "salt" to the  host and  port
   information prior to encryption to prevent malicious downstream
   proxies from guessing earlier parts of the path based on seeing
   identical encrypted  Via headers. Hidden Via fields are marked with
   the  hidden  Via option, as described in Section 6.43.

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

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

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

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

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

   The encryption of  Via header fields is described in more detail in
   Section 12.

   The  Hide header field has the following syntax:

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

6.23  If-Match

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

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

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

   Example:

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

6.24  If-None-Match

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

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

        Example:

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

6.25  Location

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

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

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

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

   3xx responses: The  Location response-header field can be used with a
        3xx response codes to indicate one or more addresses to try.  It
        can appear in responses to  INVITE and  OPTIONS methods. The
        Location header field contains URIs giving the new locations or
        user names to try, or may simply specify additional transport
        parameters. A 301 (Moved Permanently) or 302 (Moved Temporarily)
        response SHOULD contain a  Location field containing URIs of new
        addressed to be tried. A 301 or 302 response may also give the
        same location and username that was being tried but specify
        additional transport parameters such as a multicast address to
        try or a change of SIP transport from UDP to TCP or vice versa.

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

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

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

   A  Location response header may contain any suitable URI indicating
   where the called party may be reached, not limited to SIP URLs. For
   example, it may contain a phone or fax URL [22], a  mailto: URL [19]
   or  irc: URL.

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

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

   action: The  action is only used when registering with the  REGISTER
        request. It indicates how the client wishes forwarding to occur,
        by proxying or by redirection. The action taken if this
        parameter is not specified depends on server configuration. In
        its response, the registrar SHOULD indicate the mode used. This
        parameter is ignored for other requests.

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

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

   Example:

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

6.26  Max-Forwards

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

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

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

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

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

   Example:

     Max-Forwards: 6

6.27  Organization

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

        Organization    =    "Organization" ":" *text

6.28  Priority

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

        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.

   Examples:

     Subject: A tornado is heading our way!
     Priority: emergency
     Subject: Weekend plans
     Priority: non-urgent

        These are the values of RFC 2076, with the addition of
        "emergency".

6.29  Proxy-Authenticate

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

   See [H14.33] for further details.

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

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

6.30  Proxy-Authorization

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

6.31  Proxy-Require

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

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

6.32  Require

   The  Require request header is used by clients to tell user agent
   servers about options that the client expects the server to support
   in order to properly process the request. If a server does not
   understand the option, it MUST respond by returning status code 420
   (Bad Extension) and list those options it does not understand in the
   Unsupported header.

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

   Example:

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

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

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

   Proxy and redirect servers MUST ignore features that are not
   understood. If a particular extension requires that intermediate
   devices support it, the extension should be tagged in the Proxy-
   Require field instead (see Section 6.31).

6.33  Record-Route

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

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

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

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

   The  Record-Route header field has the following syntax:

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

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

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

6.34  Response-Key

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

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

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

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

6.35  Route

   The  Route request header determines the route taken by a request.
   Each host removes the first entry and then proxies the request to the
   host listed in that entry, also using it as the Request-URI. The
   operation is further described in Section 6.33.

   The  Route header field has the following syntax:

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

6.36  Retry-After

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

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

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

        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 syntax. one hour starting
   at 21:00 GMT. In the last example, the delay is 2 minutes.

6.37  Server

   The  Server response-header field contains information about the
   software used by the user agent server to handle the request. See
   [H14.39].

6.38  Subject

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

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

   Example:

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

6.39  Timestamp

   The timestamp general header describes when the client sent the
   request to the server. The value of the timestamp is of significance
   only to the client and may use any timescale. The server MUST echo
   the exact same value and MAY, if it has accurate information about
   this, add a floating point number indicating the number of seconds
   that have elapsed since it has received the request. The timestamp is
   used by the client to compute the round-trip time to the server so
   that it can adjust the timeout value for retransmissions.

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

6.40  To

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

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

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

   Example:

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

6.41  Unsupported

   The  Unsupported response header lists the OPTIONS and  INVITE request methods to indicate what description
   formats are acceptable in features not supported by
   the response.

   Example:

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

6.7 Accept-Language server. See [H14.4] Section 6.32 for syntax. a usage example and motivation.

6.42  User-Agent

   The  Accept-Language request header can be
   used to allow  User-Agent request-header field contains information about the
   client to indicate to user agent originating the server in request. See [H14.42].

6.43  Via

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

6.43.1 Requests

   The client originating the request MUST insert into the request a Via
   field containing its host name or network address and, if not the
   default port number, the port number it would prefer wishes to receive reason phrases. This responses
   at. (Note that this port number may also be used as a
   hint by differ from the UDP source port
   number of the request.) A fully-qualified domain name is RECOMMENDED.
   Each subsequent proxy as to which destination to connect server that sends the call to
   (e.g., for selecting a human operator).

   Example:

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

6.8 Allow

   See [H14.7].

6.9 Also

   The  Also request header advises 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 callee to send invitations to original
   request.

   A proxy SHOULD check the addresses listed. This supports third-party call initiation
   (Section 13).

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

   Example:

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

6.10 Authorization

   See [H14.8].

6.11 Call-Disposition

   The  Call-Disposition request top-most  Via header field allows the client to
   indicate how ensure that it
   contains the server sender's correct network address, as seen from that
   proxy. If the sender's address is incorrect, the proxy should add an
   additional  received attribute, as described below.

        A host behind a network address translator (NAT) or
        firewall may not be able to handle insert a network address into
        the call. The following options  Via header that can be used singly reached by the next hop beyond
        the NAT. Hosts behind NATs or in combination:

   all: If NAPTs should insert the user part local
        port number of the SIP request address identifies a group outgoing socket, rather than an individual, the " all" feature indicates port
        number for incoming requests, as NAPTs assume that
        responses return with reversed source and destination
        ports.

   Additionally, if the message goes to a multicast address, an extra
   Via field is added by the sender before all
        members of the group should 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 alerted rather than guaranteed that a proxy server
        can always detect that the default
        of locating address returned by a location
        service refers to a host listed in the first available individual from  Via list, as a
        single host may have aliases or several network interfaces.

6.43.2 Receiver-tagged Via Fields

   Normally every host that sends or forwards a SIP message adds a Via
   field indicating the path traversed. However, it is possible that group.
        Section 1.4.1 describes
   Network Address Translators (NAT) may change the behavior source address of proxy servers when
        resolving group aliases.

   do-not-forward: The "do-not-forward" request prohibits proxies from
        forwarding the call to another individual (e.g.,
   the call is
        personal or request, in which case the caller does not want to Via field cannot be shunted relied on to route
   replies. To prevent this, a
        secretary if the line is busy.)

   queue: If proxy SHOULD check the called party is temporarily unreachable, e.g., because top-most  Via
   header to ensure that it is in another call, contains the caller can indicate sender's correct network
   address, as seen from that it wants to
        have its call queued rather than rejected immediately. proxy. If the
        call sender's address is queued,
   incorrect, the server returns "181 Queued" (see Section
        7.1.3). A pending call be terminated proxy should add a  received tag to the Via field
   inserted by the previous hop. Such a  BYE request (Section
        4.2.4).

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

   Example:

     Call-Disposition: all, do-not-forward, queue
        HS: This header is experimental. The name modified Via field is based on the
        SMTP Content-Disposition header.

6.12 Call-ID

   The  Call-ID general header uniquely identifies known as a particular
   invitation. Note that
   receiver-tagged  Via field.  An example is:

     Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
     Via: SIP/2.0/UDP 10.0.0.1:5060, received=199.172.136.3

   In this example, the message went from 10.0.0.1 and through a single multimedia conference may give rise NAT
   using external address border.ieee.org (199.172.136.3) to
   several calls
   erlang.bell-telephone.com tagged the previous hop's  Via field with different  Call-IDs, e.g., if
   the address that it actually came from.

6.43.3 Responses

   In the return path,  Via fields are processed by a user invites
   several different people. Since proxy or client
   according to the  Call-ID following rules:

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

        2.   If the second  Via field in a response is unique for each
   caller, a user may invited multicast
             address, remove that  Via field, and send the message to
             the same conference using several
   different  Call-IDs. multicast address indicated.

        3.   If desired, it must use identifiers within the
   session description to detect this duplication. Calls second  Via field is a receiver-tagged field
             (Section 6.43.2) send the message to different
   callee MUST always use different  Call-IDs unless they are the result
   of a proxy server "forking" a single request.

   The  Call-ID may be any URL-encoded string address in the
             received tag rather than that can be guaranteed to
   be globally unique for in the duration main part of the request. Using Via
             field.

        4.   If the
   initiator's IP-address, process id, and instance (if more than one
   request second  Via field exists, send the message to the
             address indicated. If there is being made simultaneously) satisfies no second  Via field, this requirement.

   The form  local-id@host is recommended, where  host
             response is either the
   fully qualified domain name or destined for this client.

   These rules ensure that a globally routable IP address, and
   local-id depends on the application and operating system of proxy server only has to check the host,
   but is an ID that can be guaranteed first
   Via field in a response to be unique during this session
   initiation request.

        Call-ID  ___ see if it needs processing.

6.43.4 Syntax

   The format for a  Via header is:

        Via                   =    ( "Call-ID" "Via" | "v") ":" 1#( sent-protocol sent-by
                                   *( ";" via-params ) [ comment ] )
        via-params            =    via-hidden | via-ttl | via-received | via-branch
        via-hidden            =    "hidden"
        via-ttl               =    "ttl" "=" ttl
        via-received          =    "received" "=" host
        via-branch            =    "branch" "=" token
        sent-protocol         =    [ protocol-name "/" ] protocol-version
        [ "/" transport ]
        protocol-name         =    "SIP" | token
        protocol-version      =    token
        transport             =    "UDP" | "TCP" | token
        sent-by               =    ( host [ ":" port ] ) | "i" ( concealed-host ) ":" atom "@" host

   Example:

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

6.13 Content-Length

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

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

   An example is
     Content-Length: 3495

   Applications SHOULD use this field to indicate the size of the
   message-body    token
        ttl                   =    1*3DIGIT                                            ; 0 to be transferred, regardless of the media type of 255

   The " ttl" parameter is included only if the
   entity. Any  Content-Length greater than or equal to zero address is a valid
   value. If no body multicast
   address. The " received" parameter is present in added only for receiver-added
   Via fields (Section 6.43.2).  For reasons of privacy, a message, then the Content-Length
   header MAY be omitted client or set
   proxy may wish to zero. hide its Via information by encrypting it (see
   Section 8 describes how to
   determine 6.22).  The " hidden" parameter is included if this header
   was hidden by the length upstream proxy (see 6.22).

   The " branch" parameter is included by every forking proxy.  The
   token uniquely identifies a branch of the message body.

6.14 Content-Type a particular search. The  Content-Type entity-header field indicates the media type
   identifier has to be unique only within a set of isomorphic requests.

   Note that privacy of the
   message-body sent to proxy relies on the recipient.

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

   An example cooperation of the field is

     Content-Type: application/sdp

6.15 Date

   General header field. See [H14.19].

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

6.16 Expires

   The  Expires entity-header field gives next
   hop, as the date next-hop proxy will, by necessity, know the IP address
   and time after which port number of the message content expires.

   This header source host.

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

6.44  Warning

   The  Warning response-header field is currently defined only for used to carry additional
   information about the  REGISTER and
   INVITE methods. For  REGISTER, it is status of a request and response-header
   field response.  Warning headers are sent
   with responses and allows have the client to indicate how long following format:

        Warning          =    "Warning" ":" 1#warning-value
        warning-value    =    warn-code SP warn-agent SP warn-text
        warn-code        =    3DIGIT "." 2DIGIT
        warn-agent       =    ( host [ ":" port ] ) | pseudonym
                              ;  the registration
   should be valid; name or pseudonym of the server uses it to indicate when adding
                              ;  the client has
   to re-register. Warning header, for use in debugging
        warn-text        =    quoted-string

   A response may carry more than one  Warning header.

   The server's choice overrides  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 client. The
   server MAY choose a shorter time interval than that requested by cache or user, the
   client, but SHOULD not choose a longer one.

   For  INVITE, it is a request and response-header field. In  Accept-Language field in a
   request, the callee can limit the validity of an invitation. (For example, if  Content-Language field in a client wants response, etc. The default
   language is English.

   Any server may add  Warning headers to limit how long a search should take at most or when a conference being invited to is time-limited. response. New Warning
   headers MUST be added after any existing  Warning headers. A user interface may
   take this is as proxy
   server MUST NOT delete any  Warning header that it received with a hint
   response.

   When multiple  Warning headers are attached to leave a response, the invitation window on user
   agent SHOULD display as many of them as possible, in the screen
   even if order that
   they appear in the user response. If it is not currently at the workstation.) In a 302
   response, a server can advise the client possible to display all of
   the maximal duration of warnings, the redirection.

   The value of 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.

   Example:

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

6.45  WWW-Authenticate

   The  WWW-Authenticate response-header field can MUST be either an  HTTP-date or an integer
   number of seconds (in decimal), measured from the receipt included in 401
   (Unauthorized) response messages. The field value consists of at
   least one challenge that indicates the
   request.

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

   Two example of its use are

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

6.17 From

   Requests MUST authentication scheme(s) and responses SHOULD contain a  From header field,
   indicating
   parameters applicable to the invitation initiator.  Request-URI.

   See [H14.46] and [30].

   The  WWW-Authenticate response-header field MUST be a SIP URL as
   defined included in Section 2. Only a single initiator and a single invited
   user are allowed to 401
   (Unauthorized) response messages.

   The content of the  realm parameter SHOULD be specified in displayed to the user.
   A user agent SHOULD cache the authorization credentials for a single SIP request.  The sense given
   value of the destination ( To header) and  From header fields is maintained from realm and attempt to re-use
   these values on the next request for that destination.

   In addition to
   response, i.e., if the  From header is sip://bob@example.edu in the
   request then it is MUST also be sip://bob@example.edu "basic" and "digest" authentication schemes
   defined in the response
   to that request.

   The  From field is a URL and not a simple specifications cited above, SIP address (Section 1.4.1
   address to allow a gateway to relay a call into defines a SIP request new scheme,
   PGP (RFC 2015, [32]), Section 13. Other schemes, such as S-MIME, are
   for further study.

7 Status Code Definitions

   The response codes are consistent with, and
   still produce an extend, HTTP/1.1 response
   codes. Not all HTTP/1.1 response codes are appropriate, and only
   those that are appropriate  From field.

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

   Examples:

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

6.18 Location

   The  Location are given here. Other HTTP/1.1 response header can
   codes should not be used with a 2xx or 3xx response used. Response codes not defined by HTTP/1.1 have
   codes x80 upwards to indicate a new location to try. It contains a URL giving the
   new location or username to try, or may simply specify additional
   transport parameters. A "301 Moved Permanently" or "302 Moved
   Temporarily" avoid clashes with future HTTP response SHOULD contain a  Location field containing the
   URL giving codes.
   Also, SIP defines a new address to try. A 301 or 302 class, 6xx. The default behavior for unknown
   response may also give codes is given for each category of codes.

7.1 Informational 1xx

   Informational responses indicate that the same location server or proxy contacted
   is performing some further action and username that was being tried but specify
   additional transport parameters such as does not yet have a multicast address to try or definitive
   response. The client SHOULD wait for a change of SIP transport further response from UDP to TCP or vice versa.

   A user agent or redirect the
   server, and the server sending SHOULD send such a definitive, positive response (2xx), SHOULD insert without further
   prompting. Typically a  Location server should send a 1xx response header indicating
   the SIP address under which if it
   expects to take more than 200 ms to obtain a final response.

7.1.1 100 Trying

   Some unspecified action is reachable most directly for future
   SIP requests. This may be the address being taken on behalf of this call (e.g.,
   a database is being consulted), but the server itself or that of user has not yet been
   located.

7.1.2 180 Ringing

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

7.1.3 181 Call Is Being Forwarded

   A proxy (e.g., if server MAY use this status code to indicate that the host call is behind
   being forwarded to a firewall).

   A different set of destinations. The new
   destinations are listed in  Location response header may contain any suitable URL indicating
   where the called party may headers. Proxies SHOULD be reached,
   configurable not limited to SIP URLs. For
   example, it may contain a phone or fax URL [25], reveal this information.

7.2 Successful 2xx

   The request was successful and MUST terminate a mailto: URL [26]
   or  irc. search.

7.2.1 200 OK

   The following parameters are defined:

   q: 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  qvalue indicates callee has agreed to participate; the relative preference among message body
        indicates the locations
        given.  qvalue values are decimal numbers from 0.0 to 1.0, with
        higher values indicating higher preference.

   class: callee's capabilities.

   OPTIONS: The class parameter whether this terminal is found callee has agreed to share its capabilities, included in a
        residential or business setting. (A caller may defer a personal
        call if only a business line is available, for example.)

   description: The description field further describes, as text,
        the
        terminal. It message body.

   REGISTER: The registration has succeeded. The message body is expected that empty.

7.3 Redirection 3xx

   3xx responses give information about the user interface will render
        this text.

   duplex: The duplex parameter lists whether user's new location, or
   about alternative services that may be able to satisfy the terminal can
        simultaneously send and receive ("full"), alternate between
        sending call. They
   SHOULD terminate an existing search, and receiving ("half"), can only receive ("receive-
        only") or only send ("send-only"). Typically, a caller will
        prefer a full-duplex terminal over MAY cause the initiator to
   begin a half-duplex terminal and
        these over receive-only or send-only terminals.

   features: The feature list enumerates additional features new search if appropriate.

   Any redirection (3xx) response MUST NOT suggest any of this
        terminal. Values for this field are for further study.

   language: The language parameter lists, the addresses
   in order the  Via (Section 6.43) path of preference, the
        languages spoken by request in the person answering. This feature may be
        used to have a caller automatically select Location header
   field. (Addresses match if their host and port number match.)

7.3.1 300 Multiple Choices

   The address in the appropriate
        attendant or customer service representative, without having request resolved to
        declare several choices, each with its
   own language skills.

   media: The media tag lists the media types supported by the terminal.
        Currently, specific location, and the names defined in SDP may be used [9]: "audio",
        "video", "whiteboard", "text" user (or user agent) can select a
   preferred communication end point and "data".

   mobility: redirect its request to that
   location.

   The mobility parameter indicates if response SHOULD include an entity containing a list of resource
   characteristics and location(s) from which the terminal is fixed
        or mobile. In some locales, this may affect voice quality user or
        charges.

   priority: The priority tag indicates the minimum priority level this
        terminal is to be used for. It user agent can be used for automatically
        restricting
   choose the choice of terminals available to one most appropriate, if allowed by the user.

   service:  Accept request
   header. The service tag describes what service entity format is being provided specified by the terminal.

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

        location-params       =    "q"                     "="    qvalue
                              |    "class"                 "="    ( "personal" | "business" )
                              |    "description"           "="    quoted-string
                              |    "duplex"                "="    ( "full" | "half" |
                                                                  "receive-only" | "send-only" )
                              |    "features"              "="    1# feature-list
                              |    "language"              "="    1# language-tag
                              |    "media"                 "="    1# media-tag
                              |    "mobility"              "="    ( "fixed" | "mobile" )
                              |    "priority"              "="    1# priority-tag
                              |    "service"               "="    1# service-tag
                              |    extension-attributes
        extension-attribute   =    extension-name          "="    extension-value

   Examples:

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

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

6.19 Organization
   Content-Type header field. The Organization request-header choices SHOULD also be listed as
   Location fields conveys (Section 6.25).  Unlike HTTP, the name SIP response may
   contain several  Location fields or a list of addresses in a
   Location field. User agents MAY use the
   organization  Location field value for
   automatic redirection or MAY ask the user to which confirm a choice.

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

        This header is appropriate if the callee belongs. It may can be inserted by
   proxies reached at the boundary of an organization
        several different locations and may the server cannot or
        prefers not to proxy the request.

7.3.2 301 Moved Permanently

   The user can no longer be used by found at the address in the Request-URI and
   the requesting client
   software should retry at the new address given by the
   Location header field (Section 6.25). The caller SHOULD update any
   local directories, address books and user location caches with this
   new value and redirect future requests to filter calls.

6.20 Priority the address(es) listed.

7.3.3 302 Moved Temporarily

   The priority requesting client should retry the request header signals at the urgency new address(es)
   given by the  Location header field (Section 6.25). The duration of
   the redirection can be indicated through an  Expires (Section 6.20)
   header.

7.3.4 380 Alternative Service

   The call to was not successful, but alternative services are possible.
   The alternative services are described in the
   callee.

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

   Example:

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

6.21 Proxy-Authenticate

   See [H14.33].

6.22 Proxy-Authorization

   See [H14.34].

6.23 Public

   See [H14.35].

6.24 Require message body of the
   response.

7.3.5 381 Ambiguous

   The  Require header is used by clients to query callee address provided in the server about
   options that it request was ambiguous. The
   response MAY contain a listing of possible unambiguous addresses in
   Location headers.

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

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

   381 Ambiguous SIP/2.0
   Location: carol.lee@example.com (Carol Lee)
   Location: p.lee@example.com (Ping Lee)
   Location: lee.foote@example.com (Lee M. Foote)
        Some email and voice mail systems provide this header by returning
        functionality. A status code "420 Bad Extension" and list
   those options it does not understand in separate from 300 is used
        since the  Unsupported header.

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

   Example:

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

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

   This semantics are different: for 300, it is to make sure assumed
        that the client-server interaction same person or service will proceed
   optimally when all options are understood be reached 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
        choices provided. While an automated choice or sequential
        search makes sense for a round-trip often 300 response, user intervention is
        required by negotiation
   mechanisms. In addition, it also removes ambiguity when the client
   requires features that the server does not understand.

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

   This field roughly corresponds to a 381 response.

7.4 Request Failure 4xx

   4xx responses are definite failure responses from a particular
   server.  The client SHOULD NOT retry the PEP field in same request without
   modification (e.g., adding appropriate authorization). However, the PEP draft.

6.25 Retry-After
   same request to a different server may be successful.

7.4.1 400 Bad Request

   The  Retry-After response header field can request could not be used with a "503
   Service Unavailable" response understood due to indicate how long 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 service request, but is
   expected to be unavailable refusing to the requesting client fulfill it.
   Authorization will not help, and with a "404
   Not Found", "600 Busy", "603 Decline" response to indicate when the
   called party may request should not be available again. repeated.

7.4.5 404 Not Found

   The value of this field can be
   either an HTTP-date or an integer number of seconds (in decimal)
   after server has definitive information that the time of user does not exist at
   the response. An optional comment can be used to
   indicate additional information about domain specified in the time of callback. An
   optional duration parameter indicates how long  Request-URI. This status is also
   returned if the called party will
   be reachable starting at domain in the initial time of availability.

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

   Examples  Request-URI does not match any of its use are

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

   In the third example,
   domains handled by the callee is reachable for one hour starting
   at 21:00 GMT. In recipient of the last example, request.

7.4.6 405 Method Not Allowed

   The method specified in the delay  Request-Line is 2 minutes.

6.26 CSeq not allowed for the
   address identified by the  Request-URI. The  CSeq (command sequence) response MUST include an
   Allow header field MAY be added by a SIP
   client making containing a request if it needs to distinguish responses to
   several consecutive requests sent with list of valid methods for the same  Call-ID. A  CSeq
   field contains a single decimal sequence number chosen by indicated
   address.

7.4.7 407 Proxy Authentication Required
   This code is similar to 401 (Unauthorized), but indicates that the
   requesting client. Consecutive different requests made
   client MUST first authenticate itself with the same
   Call-ID proxy. The proxy MUST contain strictly monotonically increasing sequence
   numbers; the sequence space MAY NOT be contiguous. Retransmissions of
   the same request carry the same sequence number. A server responding
   to
   return a request  Proxy-Authenticate header field (section 6.29) containing a sequence number MUST echo
   challenge applicable to the sequence
   number back in proxy for the response. requested resource. The  ACK request MUST contain the same
   CSeq value as
   client MAY repeat the  INVITE request that it refers to.

        CSeq = "CSeq" ":" 1*DIGIT

   CSeq with a suitable Proxy-Authorization
   header fields are NOT needed for field (section 6.30). SIP requests using the INVITE
   or  OPTIONS methods but may access authentication is explained
   in section 12.3 and [H11].

   This status code should be needed used for future methods.

   Example:

     CSeq: 4711

6.27 Server

   See [H14.39].

6.28 Subject

   This is intended to provide a summary, or indicate the nature, of the
   call, allowing call filtering without having applications where access to parse the session
   description. (Also, the session description may not necessarily use
   communication channel (e.g., a telephony gateway) rather than the same subject indication as
   callee herself requires authentication.

7.4.8 408 Request Timeout

   The server could not produce a response, e.g., a user location,
   within the invitation.)

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

   Example:

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

6.29 Unsupported

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

   See Section 6.24 for a usage example and motivation.

6.30 Timestamp  Expires header.
   The timestamp general header describes when the client sent MAY repeat the request to the server. without modifications at any later
   time.

7.4.9 412 Precondition Failed

   The value precondition given in one or more of the timestamp is of significance
   only request-header fields
   evaluated to false when it was tested on the client server. Preconditions
   include  If-Match (Section 6.23) and may use any timescale. If-None-Match (Section 6.24).

7.4.10 420 Bad Extension

   The server MUST echo did not understand the exact same value and MAY, if it has accurate information about
   this, add protocol extension specified in a floating point number indicating
   Require (Section 6.32) header field.

7.4.11 480 Temporarily Unavailable

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

7.4.12 481 Invalid Call-ID

   The server so
   that it can adjust the timeout received a  BYE or  CANCEL request with a Call-ID (Section
   6.12 value for retransmissions.

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

6.31 To it does not recognize.  (A server simply discards an  ACK
   with an invalid  Call-ID.)

7.4.13 482 Loop Detected
   The  To server received a request header field specifies the invited user, with a  Via path containing itself.

7.4.14 483 Too Many Hops

   The server received a request that contains more  Via entries (hops)
   than allowed by the
   same SIP URL syntax as the  From  Max-Forwards header field.

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

   If a SIP

7.4.15 484 Address Incomplete

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

        This status code allows overlapped dialing.

7.5 Server Failure 5xx

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

7.5.1 500 Server Internal Error

   The server encountered an unexpected condition that is unknown to it, prevented it from
   fulfilling the server returns a
   "400 bad request" response.

   Example:

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

6.32 User-Agent

   See [H14.42].

6.33 Via request.

7.5.2 501 Not Implemented

   The  Via field indicates server does not support the path taken by functionality required to fulfill the request so far.
   request. This
   prevents request looping and ensures replies take the same path as is the requests, which assists in firewall traversal and other unusual
   routing situations.

   The client originating appropriate response when the server does not
   recognize the request MUST insert method and is not capable of supporting it for
   any user.

7.5.3 502 Bad Gateway

   The server, while acting as a  Via field
   containing its address gateway or proxy, received an invalid
   response from the downstream server it accessed in attempting to
   fulfill the request. Each subsequent proxy

7.5.4 503 Service Unavailable

   The server
   that sends is currently unable to handle the request onwards MUST add its own additional  Via
   field, due to a
   temporary overloading or maintenance of the server. The implication
   is that this is a temporary condition which MUST will be added before any existing  Via fields.
   Additionally, if alleviated after
   some delay. If known, the message goes to length of the delay may be indicated in a multicast address, an extra
   Via field
   Retry-After header. If no  Retry-After is added before all given, the others giving client MUST
   handle the multicast address
   and TTL.

   If a proxy server receives a request which contains its own address, response as it MUST respond with would for a "482 Loop Detected" 500 response.

   Note: The existence of the 503 status code. (This
   prevents a malfunctioning proxy server from causing loops. Also, it
   cannot be guaranteed code does not imply that a proxy
   server can always detect that the
   address returned by a location service refers has to a host listed in use it when becoming overloaded. Some servers may wish
   to simply refuse the
   Via list, connection.

7.5.5 504 Gateway Timeout

   The server, while acting as a single host may have aliases or several network
   interfaces.)

   In the return path,  Via fields are processed by gateway, did not receive a timely
   response from the server (e.g., a proxy or client
   according location server) it accessed in
   attempting to complete the following rules:

        oIf request.

7.5.6 505 Version Not Supported

   The server does not support, or refuses to support, the first  Via field SIP protocol
   version that was used in the reply received request message. The server is the client's
   indicating that it is unable or server's local address, remove unwilling to complete the  Via field and process request
   using the reply.

        oIf same major version as the first  Via field in a reply client, other than with this
   error message. The response SHOULD contain an entity describing why
   that version is a multicast address,
         remove not supported and what other protocols are supported
   by that  Via field before sending to the multicast address.

   These rules ensure server.

7.6 Global Failures 6xx

   6xx responses indicate that a proxy server only has to check definitive information about
   a particular user, not just the first
   Via field particular instance indicated in a reply the
   Request-URI. All further searches for this user are doomed to see if it needs processing. failure
   and pending searches SHOULD be terminated.

7.6.1 600 Busy

   The format for 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  Via header is:

        Via                   =    ( "Via" | "v") ":" 1#( sent-protocol sent-by
                                   *( ";" via-params ) [ comment ] )
        via-params            =    "ttl" "=" ttl
        sent-protocol         =    [ protocol-name "/" ] protocol-version
        [ "/" transport ]
        protocol-name         =    "SIP" | token
        protocol-version      =    token
        transport             =    "UDP" | "TCP"
        sent-by               =    host [ ":" port ]
        ttl                   =    1*3DIGIT                                         ; 0 better time to call in the  Retry-After header. If the
   callee does not wish to 255 reveal the reason for declining the call, the
   callee should use status code 603 (Decline) instead.

7.6.2 603 Decline

   The "ttl" parameter is included only if callee's machine was successfully contacted but the address is user
   explicitly does not wish to or cannot participate. The response may
   indicate a multicast
   address.

   Example:

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

6.34 Warning better time to call in the  Retry-After header.

7.6.3 604 Does Not Exist Anywhere

   The  Warning response-header server has authoritative information that the user indicated in
   the To request field is used 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 carry additional
   information about
   communicate, but cannot adequately support the status session described. The
   606 (Not Acceptable) response MAY contain a list of reasons in a response.
   Warning headers are sent
   with responses and have header describing why the following format:

        Warning          =    "Warning" ":" 1#warning-value
        warning-value    =    warn-code SP warn-agent SP warn-text
        warn-code        =    2DIGIT
        warn-agent       =    ( host [ ":" port ] ) | pseudonym
                              ; session described cannot be
   supported. These reasons can be one or more of:

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

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

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

   606.4 Multicast Not Available: The site where the Warning header, for use in debugging
        warn-text        =    quoted-string

   A response may carry more than one  Warning header. user is located
        does not support multicast.

   606.5 Unicast Not Available: The  warn-text should be in a natural language and character set that site where the user is most located does
        not support unicast communication (usually due to the presence
        of a firewall).

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

8 SIP Message Body

8.1 Body Inclusion

   For a request message, the location presence of a body is signaled by the cache or user, the  Accept-Language field in
   inclusion of a
   request,  Content-Length header. Only  ACK,  INVITE, OPTIONS
   and  REGISTER requests may contain message bodies. For ACK,  INVITE
   and  OPTIONS, the  Content-Language field in message body is always a response, etc. session description. The default
   language
   use of message bodies for  REGISTER requests is English.

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

   When multiple  Warning headers are attached to for further study.

   For response messages, whether or not a response, the user
   agent SHOULD display as many of them as possible, in body is included is dependent
   on both the order that
   they appear in request method and the response. If response message's response code.
   All responses MAY include a body, although it is not possible to display all may be of zero length.
   Message bodies for 1xx responses contain advisory information about
   the warnings, the user agent should follow these heuristics:

        oWarnings that appear early in progress of the response take priority over
         those appearing later in request, 2xx responses contain session
   descriptions; for responses with status 300 or greater, the response.

        oWarnings in session
   body MAY contain additional, human-readable information about the user's preferred character set take priority
         over warnings in other character sets but with identical
         warn-codes and  warn-agents.

   Systems
   reasons for failure.  It is RECOMMENDED that generate multiple  Warning headers should order them
   with this user agent behavior information in mind.

   Example:

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

6.35 WWW-Authenticate

   See [H14.46].

7 Status Code Definitions

   The response codes are consistent with, 1xx and extend, HTTP/1.1 response
   codes. Not all HTTP/1.1 response codes are appropriate,
   300 and only
   those that are appropriate are given here. Response codes not defined
   by HTTP/1.1 have codes x80 upwards to avoid clashes with future HTTP
   response codes. Also, SIP defines a new class, 6xx. greater responses be of type text/plain or text/html

8.2 Message Body Type

   The default
   behavior for unknown response codes is Internet media type of the message body MUST be given for each category by the
   Content-Type header field, If the body has undergone any encoding
   (such as compression) then this MUST be indicated by the Content-
   Encoding header field, otherwise Content-Encoding MUST be omitted.

   If applicable, the character set of
   codes.

7.1 Informational 1xx

   Informational responses indicate that the server or proxy contacted message body is performing some further action and does not yet have a definitive
   response. The client SHOULD wait for a further response from indicated as
   part of the
   server, and  Content-Type header-field value.

8.3 Message Body Length

   The body length in bytes MUST be given by the server SHOULD send such a response without further
   prompting.  Content-Length header
   field. If UDP transport no body is being used, the client SHOULD
   periodically re-send the request present in case a message, then the final response is lost.
   Typically Content-Length
   header MUST set to zero. If a server should send receives a "1xx" response if message without
   Content-Length, it MUST assume it expects to take
   more than one second to obtain a final reply.

7.1.1 100 Trying

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

7.1.2 180 Ringing be zero.

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

7.1.3 181 Queued "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
   (Note: The called party was temporarily unavailable, but chunked encoding modifies the caller
   indicated via body of a "Call-Disposition: Queue" directive (Section 6.11) message in order
   to
   queue the call rather than reject it. When the callee becomes
   available, transfer it will return the appropriate final status response. The
   reason phrase MAY give further details about the status as a series of the call,
   e.g., "5 calls queued; expected waiting time chunks, each with its own size
   indicator.)

9 Compact Form

   When SIP is 15 minutes". The
   server carried over UDP with authentication and a complex
   session description, it may issue several 181 responses to update be possible that the caller about size of a request or
   response is larger than the
   status 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 queued call.

7.2 Successful 2xx

   The request was successful header in section 14.2 could also be written:

     INVITE schooler@vlsi.caltech.edu SIP/2.0
     v:SIP/2.0/UDP 239.128.16.254 16
     v:SIP/2.0/UDP 131.215.131.131
     v:SIP/2.0/UDP 128.16.64.19
     f:mjh@isi.edu
     t:schooler@cs.caltech.edu
     i:62729-27@128.16.64.19
     c:application/sdp
     l:187

     v=0
     o=user1 53655765 2353687637 IN IP4 128.3.4.5
     s=Mbone Audio
     i=Discussion of Mbone Engineering Issues
     e=mbone@somewhere.com
     c=IN IP4 224.2.0.1/127
     t=0 0
     m=audio 3456 RTP/AVP 0

   Mixing short field names and long field names is allowed, but not
   recommended. Servers MUST terminate accept both short and long field names for
   requests. Proxies MUST NOT translate a search.

7.2.1 200 OK

   The request was successful in contacting the user, between short and the user has
   agreed to participate.

7.3 Redirection 3xx

   3xx responses give information about the user's new location, 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
   about alternative services that may be able to satisfy multicast) or TCP
   as a transport protocol; it provides its own reliability mechanism.

10.1.1 Requests

   Stateful proxies mark outgoing requests with the call.
   They SHOULD terminate an existing search, and MAY cause  branch parameter in
   the initiator
   to begin a new search if appropriate.

7.3.1 300 Multiple Choices

   The requested resource corresponds  Via header.

   Servers ignore isomorphic requests, but retransmit the appropriate
   response. (SIP requests are said to any be idempotent , i.e., receiving
   more than one copy of a set of
   representations, each with its own specific location, and agent-
   driven negotiation (i.e., controlled by the SIP client) is being
   provided so that request does not change the user (or user agent) can select server state.)

   If a preferred
   communication end point and stateful proxy, user agent or redirect its request server cannot respond to that location.

   The
   a request with a final response SHOULD include an entity containing within 200 ms, it MUST issue a list of resource
   characteristics and location(s)
   provisional (1xx) response as soon as possible. Stateless proxies
   MUST NOT issue provisional responses on their own.

   After receiving a  CANCEL request from which the user or user agent can
   choose the one most appropriate. The entity format is specified an upstream client, a stateful
   proxy server SHOULD send a  CANCEL on all branches where it has not
   yet received a final response.

10.1.2 Responses

   Responses are mapped to requests by the media type given in the  Content-Type header field. Depending
   upon the format matching  To, From,  Call-ID,
   CSeq headers and the capabilities of the user agent, selection branch parameter of the most appropriate choice may be performed automatically. However,
   this specification does not define any standard for such automatic
   selection.

   The choices SHOULD also first  Via header.
   Responses terminate request retransmissions even if they have  Via
   headers that cause them to be listed as  Location fields (Section 6.18).
   Unlike HTTP, the SIP response delivered to an upstream client.

   A stateful proxy may contain several  Location fields.
   User agents MAY use receive a response that it does not have state
   for, that is, where it has no a record of an isomorphic request. If
   the  Location Via header field value for automatic
   redirection or MAY ask indicates that the user to confirm upstream server used TCP, the
   proxy actively opens a choice.

7.3.2 301 Moved Permanently

   The requesting client should retry TCP connection to that address. Thus, proxies
   have to be prepared to receive responses on the new address given 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
   Location field because proxy, a passive connection is one
   accepted by the user has permanently moved and proxy, but initiated by another entity.)

   100 responses are not forwarded, other 1xx responses MAY be
   forwarded, possibly after the address
   this response is in reply server eliminates responses with status
   codes that had already been sent earlier. 2xx responses are forwarded
   according to is no longer a current address for the
   user. A 301 response  Via header. Once a stateful proxy has received a
   2xx response, it MUST NOT suggest 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 of
   connections 20 seconds after receiving the hosts in first final response.

        The 20 second window is given by the  Via
   (Section 6.33) path maximum retransmission
        duration of the request as the user's new location.

7.3.3 302 Moved Temporarily

   The requesting client should retry 200 responses (10 times T4), in case the  ACK
        is lost somewhere on the new address(es) given by way to the Location header. A 302 response called user agent or
        the next stateful proxy.

10.2 Unicast UDP

   UDP packets MUST NOT suggest any of have a source address that is valid as a destination
   for requests and responses. Responses are returned to the hosts address
   listed in the  Via header field (Section 6.33) path of the request as 6.43), not the user's new
   location.  The duration source
   address of the redirection can request.

10.3 Multicast UDP

   Requests may be indicated through
   an  Expires (Section 6.16) header.

7.3.4 380 Alternative Service

   The call was not successful, but alternative services are possible.
   The alternative services are described in the message body multicast. Multicast requests SHOULD have a time-to-
   live value of no greater than one, i.e., be restricted to the
   response.

7.4 Request Failure 4xx

   4xx responses are definite failure responses from a particular
   server.  The client SHOULD NOT retry local
   network.

   If the same request without
   modification (e.g., adding appropriate authorization). However, was received via multicast, the
   same request to a different server may be successful.

7.4.1 400 Bad Request response is also
   returned via multicast. The request could server delays its response by a random
   interval between zero and one second. Servers do not be understood due return 404 (Not
   Found) responses and SHOULD suppress responses if they hear a lower-
   numbered or 6xx response from another group member prior to malformed syntax.

7.4.2 401 Unauthorized

   The sending.
   Servers do not respond to  CANCEL requests received via multicast.

10.4  BYE,  CANCEL,  OPTIONS

   A SIP client SHOULD retransmit a  BYE,  CANCEL, or  OPTIONS request requires user authentication.

7.4.3 402 Payment Required

   Reserved for future use.

7.4.4 403 Forbidden

   The server understood
   periodically with timer T1 until it receives a response, or until it
   has reached a set limit on the request, but number of retransmissions. If the
   response is refusing provisional, the client continues to fulfill it.
   Authorization will not help, and retransmit the request should not be repeated.

7.4.5 404 Not Found
   request, albeit less frequently, using timer T2. The server has definitive information that default values
   of timer T1 and T2 are 1 and 5 seconds, respectively. The default
   retransmit limit is 20 times. After the user does not exist at server sends a final
   response, it cannot be sure the domain specified in client has received the  Request-URI.

7.4.6 405 Method Not Allowed

   The method specified in response, and
   thus SHOULD cache the  Request-Line is not allowed results for at least 100 seconds to avoid
   having to, for example, contact the
   address identified by the  Request-URI. The response MUST include an
   Allow header containing user or user location server
   again upon receiving a list retransmission.

        The value of valid methods for the indicated
   address.

7.4.7 407 Proxy Authentication Required

   This code initial retransmission timer is similar to 401 (Unauthorized), but indicates smaller
        than that that for TCP since it is expected that network
        paths suitable for interactive communications have round-
        trip times smaller than 1.5 seconds. To avoid flooding the
   client MUST first authenticate itself
        network with packets every second even if the proxy. The proxy MUST
   return destination
        network is unreachable, the retransmission count has to be
        bounded. Given that most transactions should consist of one
        request and a  Proxy-Authenticate header field (section 6.21) containing few responses, round-trip time estimation
        seems less than helpful. If RTT estimation is desired to
        more quickly discover a
   challenge applicable missing final response, each
        request retransmission needs to be labeled with its own
        Timestamp (Section 6.39), returned in the proxy for the requested resource. response. The
        server caches the result until it can be sure that the
        client MAY repeat will not retransmit the same request with again.

10.5  REGISTER

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

        Retransmitting REGISTER indefinitely ensures that a suitable Proxy-Authorization
   header field (section 6.22). SIP access authentication is explained
   in section [H11].

   This status code should user
        will eventually be used for applications where access able to the
   communication channel (e.g., register after a telephony gateway) rather registrar
        recovers from a crash. The period is chosen so that even on
        a large LAN, there will not be more than the
   callee herself requires authentication.

7.4.8 408 Request Timeout about one
        REGISTER request per second.

10.6  ACK

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

10.7  INVITE

   Special considerations apply for the  INVITE method.

        1.   After receiving an invitation, considerable time that may elapse
             before the server
   was prepared to wait. The client MAY repeat can determine the request without
   modifications at any later time.

7.4.9 420 Bad Extension

   The server did not understand outcome. For example,
             the protocol extension specified with
   strength "must".

7.4.10 480 Temporarily Unavailable

   The callee's end system was contacted successfully but 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 is
   currently unavailable (e.g., are automated servers not logged in or logged in in such directly
             controlled by a
   manner as to preclude communication with the callee). The response human being, a call attempt may indicate be
             unbounded in time.

        2.   If a better time telephony user interface is modeled or if we need to
             interface to call in the  Retry-After header. The PSTN, the caller's user interface will
             provide "ringback", a signal that the callee is being
             alerted. (The status response 180 (Trying) may also be available elsewhere (unbeknownst to this host),
   thus, this response does not terminate any searches. The reason
   phrase SHOULD indicate the more precise cause as used to why
             initiate ringback.) Once the callee is
   unavailable.  This value SHOULD be setable by picks up, the user agent.

7.4.11 481 Invalid Call-ID
   The server received a  BYE or  ACK request with a Call-ID value it
   does not recognize.

7.4.12 482 Loop Detected

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

7.5 Server Failure 5xx

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

7.5.1 500 Server Internal Error

   The server encountered an unexpected condition caller
             needs to know so that prevented it from
   fulfilling can enable the request.

7.5.2 501 Not implemented voice path and stop
             ringback. The server does not support the functionality required callee's response to fulfill the
   request. This is invitation could get
             lost. Unless the appropriate response when is transmitted reliably, the server does not
   recognize
             caller will continue to hear ringback while the request method and is not capable of supporting it for
   any user.

7.5.3 502 Bad Gateway callee
             assumes that the call exists.

        3.   The server, while acting as a gateway or proxy, received client has to be able to terminate an invalid
   response from the upstream server on-going request,
             e.g., because it accessed in attempting is no longer willing to
   fulfill wait for the request.

7.5.4 503 Service Unavailable
             connection or search to succeed. The server is currently unable will have to handle the request due
             wait several round-trip times to a
   temporary overloading or maintenance interpret the lack of
             request retransmissions as the server. The implication
   is that this is end of a temporary condition which will be alleviated after
   some delay. call. If known, the length of call
             succeeds shortly after the delay may caller has given up, the callee
             will "pick up the phone" and not be indicated in "connected".

   A SIP client SHOULD retransmits a
   Retry-After header. SIP  INVITE request periodically
   with timer T1 until it receives a response, or until it has reached a
   set limit on the number of retransmissions. If no  Retry-After the response is given,
   provisional, the client SHOULD
   handle continues to retransmit the response as it would for a 500 response.

   Note: request, albeit
   less frequently, using timer T3. The existence default values of the 503 status code does not imply timer T1 and
   T3 are 1 and 30 seconds, respectively. The default retransmit limit
   is 20.

        The value of T3 was chosen so that for most normal phone
        calls, only one  INVITE request will be issued. Typically,
        a
   server must use it when becoming overloaded. Some servers may wish phone switches to
   simply refuse the connection.

7.5.5 504 Gateway Timeout

   The server, while acting as a gateway, did not receive an answering machine or voice mail
        after about 20--22 seconds.

   Upon receiving a timely
   response from final response, the upstream server (e.g., a location server) it
   accessed client sends an  ACK to the
   address listed in the  Location header field contained in the
   response. The  To header field in attempting to complete the request.

7.6 Global Failures 6xx

   6xx responses indicate that a server has definitive information about
   a particular user,  ACK request assumes the value
   of the  Location header field. If the response did not just contain a
   Location header, the particular instance indicated in client uses the
   Request-URI. All further searches same  To header field as 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
   INVITE request and does not wish to take sends the call at this time. The response
   may indicate a better time  ACK to call in the  Retry-After header. If same destination as the
   callee does not wish to reveal
   original  INVITE request.

   The server retransmits the reason final response at intervals of T4 (default
   value of T4 = 2 seconds) until it receives an  ACK request for declining the call,
   same  Call-ID and  CSeq from the
   callee should use status code 680 instead.

7.6.2 603 Decline

   The callee's machine was successfully contacted but same  From source or until it has
   retransmitted the user
   explicitly does not wish to participate. The final response may indicate a
   better time 10 times. The ACK request MUSTNOT be
   acknowledged to call in prevent a response- ACK feedback loop.

   Fig. 8 and 9 show the  Retry-After header.

7.6.3 604 Does not exist anywhere

   The client and server has authoritative information that state diagram for
   invitations.

        Using the user indicated mechanism in
   the To request field Sec. 10.4 does not exist anywhere. Searching work well for the user
   elsewhere will not yield any results.

7.6.4 606 Not Acceptable

   The user's agent was contacted successfully
        long delays between  INVITE and a final response.  If the
        200 response gets lost, the callee would believe the call
        to exist, but some aspects of the
   session profile (the requested media, bandwidth, or addressing style)
   were voice path would be dead since the caller
        does not acceptable.

   A "606 Not Acceptable" reply means know that the user wishes callee has picked up. Thus, the
        INVITE retransmission interval would have to
   communicate, but cannot adequately support be on the session described. The
   "606 Not Acceptable" reply MAY contain a list
        order of reasons in a Warning
   header describing why second or two to limit the session described cannot be supported.
   These reasons duration of this
        state confusion.

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

10.8 TCP Connections

   A single TCP connection can be serve one or more of:

   606.1 Insufficient Bandwidth: The bandwidth specified in the session
        description SIP transactions. A
   transaction contains zero or defined more provisional responses followed by the media exceeds that known to
   one or more final responses. (Typically, transactions contain exactly
   one final response, but there are exceptional circumstances, where,
                 +===========+
                 |  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    +-----------+
           +------|

    event
   -------
   message

   Figure 8: State transition diagram of client for  INVITE method

   for example, multiple 200 responses may be
        available.

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

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

   606.4 Multicast not available: The site where generated.)

   After sending an  INVITE request, the user is located
        does not support multicast.

   606.5 Unicast not available: The site where client keeps the user is located does
        not support unicast communication (usually due to connection
   open until the presence
        of first final response arrives. If that response has a firewall).

   Other reasons are likely to be added later. It
                 +===========+
                 |  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 |              |
           -     +-----------+              |
           +------|       |                 |
                          +-----------------+

    event
   -------
   message

   Figure 9: State transition diagram of server for  INVITE method

   status code of 300 or larger, the client sends an  ACK. If the
   response status code is hoped that
   negotiation will not frequently be needed, 2xx and when the client is a new user agent client, it
   sends an ACK. If the client is
   being invited to join a pre-existing lightweight session, negotiation
   may not be possible. It a user agent, the response is up to
   forwarded upstream.

   The client MAY close the invitation initiator to decide
   whether or not to act on a "606 Not Acceptable" reply.

8 SIP Message Body connection at any time. The session description body gives details of server SHOULD
   NOT close the session TCP connection until it has sent its final response, at
   which point it MAY close the user TCP connection if it wishes to. However,
   normally it is
   being invited the client's responsibility to join. Its Internet media type MUST be given by close the
   Content-type header field, connection.

   If the server leaves the connection open, and if the body length in bytes MUST be given
   by client so
   desires it may re-use the  Content-Length header field. If connection for further SIP requests or for
   requests from the body has undergone any
   encoding same family of protocols (such as compression) then this MUST be indicated by HTTP or stream
   control commands).

   If a server needs to return a response to a client and no longer has
   a connection open to that client, it MAY open a connection to the
   Content-encoding header field, otherwise Content-encoding
   address listed in the  Via header. Thus, a proxy MUST be
   omitted.

8.1 Body Inclusion

   For prepared to
   receive both requests and responses on a request message, the presence "passive" connection.

11 Behavior of a body is signaled by the
   inclusion SIP Servers

   This section describes behavior of a  Content-Length header. A body may be included SIP server in a
   request only when the request method allows one.

   For response messages, whether detail. Servers
   can operate in proxy or not redirect mode. Proxy servers can "fork"
   connections, i.e., a body is included is dependent
   on both the single incoming request method and the response message's response code.
   All 1xx informational responses 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 NOT include insert a body. All other
   responses MAY include " branch" parameter (Section 6.43). To
   prevent loops, a payload, although it may be of zero length.

8.2 Message Body Length

   If no body server MUST check if its own address is present already
   contained in a message, then the  Content-Length  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 omitted larger, smaller or set
   equal to zero. When y.

        This rule allows a body is included, its length in
   bytes is indicated proxy to serve as a go-between between
        two servers that have no version of the protocol in common.

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

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

11.1 Redirect Server
   A redirect server does not issue any SIP requests of the following:

        1.   Any response message which MUST NOT include its own. It can
   return a body (such as response that refuses or redirects the 1xx responses) is always terminated by request. After
   receiving an  INVITE request, once the first empty
             line after server has gathered the header fields, regardless if any  entity-
             header fields are present.

        2.   Otherwise, a  Content-Length header MUST be present (this
             requirement differs from HTTP/1.1). Its value in bytes
             represents list
   of alternative locations or has decided to refuse the length call, it
   returns the final response of class 3xx or 6xx. This ends the message body. SIP
   transaction. The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used redirect server maintains transaction state for SIP.
   (Note: The chunked encoding modifies the body of a message in order
   whole SIP transaction.

11.2 User Agent Server

   User agent servers behave similarly to transfer it as redirect servers, except that
   they may also accept a series of chunks, each call with its own size
   indicator.)

9 Examples

9.1 Invitation to Multimedia Conference

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

9.1.1 Request

   C->S: INVITE schooler@vlsi.cs.caltech.edu SIP/2.0
         Via: SIP/2.0/UDP 239.128.16.254 16
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19
         From: mjh@isi.edu (Mark Handley)
         Subject: 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 will be discussed, too
         To: schooler@cs.caltech.edu (Eve Schooler)
         Call-ID: 62729-27@oregon.isi.edu
         Content-type: application/sdp
         CSeq: 4711
         Content-Length: 187

         v=0
         o=user1 53655765 2353687637 IN IP4 128.3.4.5
         s=Mbone Audio
         i=Discussion request, that is, they do not "fork" requests.
   However, servers may choose to always operate in a mode that allows
   issuing of Mbone Engineering Issues
         e=mbone@somewhere.com
         c=IN IP4 224.2.0.1/127
         t=0 0
         m=audio 3456 RTP/AVP 0 several requests, as described in Section 11.4.

   The  Via fields list server can forward the hosts along request and any responses. It does not
   have to maintain any state for the path from invitation
   initiator (the first element of SIP transaction. Reliability is
   assured by the list) towards next redirect or stateful proxy server in the invitee. In server
   chain.

   A proxy server SHOULD cache the
   example above, result of any address translations
   and the message was last multicast response to speed forwarding of retransmissions. After the administratively
   scoped group 239.128.16.254 with
   cache entry has been expired, the server cannot tell whether an
   incoming request is actually a ttl retransmission of 16 from the host
   131.215.131.131 an older request.
   The server will treat it as a new request and commence another
   search.

11.4 Proxy Server Issuing Several  INVITE Requests

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

   All requests carry the same  Call-ID. For unicast, each of the
   requests has a different (host-dependent)  Request-URI. For
   multicast, a single request was initiated by
   mjh@isi.edu is issued, likely with a host-independent
   Request-URI. A client receiving a multicast query does not have to
   check whether the host 128.16.64.19 schooler@cs.caltech.edu is being
   invited; part of the message is currently being routed  Request-URI matches its own host
   or domain name. To avoid response implosion, servers MUST NOT answer
   multicast requests with a "404 Not Found" status code. Servers MAY
   decide not to
   schooler@vlsi.cs.caltech.edu

   In this case, answer multicast requests if their response would be
   5xx.  Responses to multicast requests are multicast with the session description same TTL
   as the request, where the TTL is using derived from the Session
   Description Protocol (SDP), as stated  ttl parameter in
   the  Content-type header.

   The Via header is terminated by (Section 6.43).

   Successful responses to an empty line and is followed by  INVITE request SHOULD contain a
   message body containing Location
   header so that the session description.

9.1.2 Reply

   The called user agent, directly following  ACK or indirectly through  BYE bypasses the proxy servers,
   indicates that it is alerting ("ringing") search
   mechanism. If the called party:

   S->C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP 239.128.16.254 16
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19 1
         From: mjh@isi.edu
         Call-ID: 62729-27@128.16.64.19
         Location: sip://es@jove.cs.caltech.edu
         CSeq: 4711

   A sample reply proxy requires future requests to the invitation is given below. The first line of
   the reply states the SIP version number, that be routed through
   it, it is adds a "200 OK" reply,
   which means  Record-Route header to the request was successful. (Section 6.33).

   The  Via headers are taken
   from the request, and entries are removed hop by hop as the reply
   retraces following pseudo-code describes the path behavior of the a proxy server
   issuing several requests in response to an incoming  INVITE request. A new authentication field MAY be
   added by the invited user's agent if required.
   The  Call-ID 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 taken
   directly from received and returns the original request, along response. close(a) closes the TCP
   connection to client with address a. response(s, l, L) sends a
   response to the remaining fields client with status s and list of locations L, with l
   entries. ismulticast() returns 1 if the request message. The original sense of  From field is
   preserved (i.e., it location is a multicast
   address and zero otherwise.  The variable timeleft indicates the session initiator).

   In addition, the  Location header gives details
   amount of time left until the host where maximum response time has expired. The
   variable recurse indicates whether the
   user was located, or alternatively 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 relevant proxy contact point
   which should be reachable server. Thus, an initial multicast
   request may trigger additional unicast requests.

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

     process_request(Method R, int N, address_t address[])
     {
       struct {
         address_t address;  /* address */
         int branch;         /* branch id */
         int done;           /* has responded */
       } outgoing;
       int done[];           /* address has responded */
       char *location[];     /* list of locations */
       int heard = 0;        /* number of sites heard from the caller's host.

   S->C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP 239.128.16.254 16
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19 1
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19
         Location: sip://es@jove.cs.caltech.edu
         CSeq: 4711 */
       int class;            /* class of status code */
       int best = 1000;      /* best response so far */
       int timeleft = 120;   /* sample timeout value */
       int loc = 0;          /* number of locations */
       struct {              /* response */
         int status;         /* response status; -2: BYE; -1: CANCEL */
         char *location;     /* redirect location */
         address_t a;        /* address of respondent */
         int branch;         /* branch identifier */
       } r;
       int i;

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

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

         if (class == 2) {
           best = r.status;
           break;
         }
         else if (class == 3) {
           /* A server may optionally recurse.  The caller confirms the invitation by sending a request to the server MUST check whether
            * it has tried this location named in before and whether the  Location header:

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

9.2 Two-party Call

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

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

   C->S: INVITE watson@boston.bell-telephone.com SIP/2.0
         Via: SIP/2.0/UDP 169.130.12.5
         From: a.g.bell@bell-telephone.com (A. Bell)
         To: watson@bell-telephone.com (T. A. Watson)
         Call-ID: 187602141351@worcester.bell-telephone.com
         Subject: Mr. Watson, come here.
         Content-type: application/sdp
         Content-Length: ...

         v=0
         o=bell 53655765 2353687637 IN IP4 128.3.4.5
         c=IN IP4 135.180.144.94
         m=audio 3456 RTP/AVP 0 3 4 5

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

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

   Watson can only receive PCMU and GSM. Note that Watson's list Via path of
   codecs may or may not the incoming request.  This check is
            * omitted here for brevity. Multicast locations MUST NOT be a subset of
            * returned to the one offered by Bell, as each
   party indicates client if the data types it server is willing to receive. Watson will
   send audio data to port 3456 at 135.180.144.94, Bell will send to
   port 5004 at 135.180.161.25.

9.3 Aborting a Call

   If the caller wants to abort a pending call, it sends a not recursing.
            */
           if (recurse) {
             multicast = 0;
             N++;
             request(R, r.location);
           } else if (!ismulticast(r.location)) {
             locations[loc++] = r.location;
             best = r.status;
           }
         }
         else if (class == 4) {
           request(ACK, r.a, r.branch);
           if (best >= 400) best = r.status;
         }
         else if (class == 5) {
           request(ACK, r.a, r.branch);
           if (best >= 500) best = r.status;
         }
         else if (class == 6) {
           request(ACK, r.a, r.branch);
           best = r.status;
           break;
         }
       }

       /* CANCEL or BYE request.

   C->S: */
       if (r.status < 0) {
         response(200, loc, 0);

         /* BYE schooler@jove.cs.caltech.edu
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19

9.3.1 Redirects

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

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

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

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

   Responses are processed as follows. The process completes when all
   requests have been answered by final status responses (for unicast)
   or 60 seconds have elapsed (for multicast). A proxy located at 131.215.131.131 is being
   advised to contact the multicast group 239.128.16.254 with MAY send a ttl of
   16
   CANCEL to all branches and UDP transport. In normal situations, a server would not
   suggest return a redirect 408 (Timeout) 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 after
   120 seconds or more.

   1xx: The proxy server SHOULD query the multicast address itself
   rather than sending MAY forward the redirect back response upstream towards the client as multicast
   may be scoped; this allows a client.

   2xx: The proxy within the appropriate scope
   regions to make MUST forward the query.

9.3.2 Alternative Services

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

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

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

   In this case, response upstream towards the answering server provides a session description
   that describes client,
        without sending an "answering machine". If the invitation initiator
   decides to take advantage of this service, it should  ACK downstream.

   3xx: The proxy MUST send an
   invitation request to  ACK and MAY recurse on the answering machine at 131.215.131.131 with listed
        Location addresses. Otherwise, the session description provided (modified as appropriate locations in the response are
        added to separate lists for a
   unicast session 300, 301 and 302 responses
        maintained by the proxy. The lowest-numbered 300 response is
        returned to contain the appropriate local address client on completion.

   4xx, 5xx: The proxy MUST send an  ACK and port for
   the invitation initiator). This request SHOULD contain a different
   Call-ID from remember the one in the original request. An example would be:

   C->S: INVITE mm-server@131.215.131.131 SIP/2.0
         Via: SIP/2.0/UDP 128.16.64.19
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-28@128.16.64.19
         Content-type: application/sdp
         Content-length: 146

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

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

9.3.3 Negotiation

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

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

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

   In this example, the original request specified 256 kb/s total
   bandwidth, lower status code than any previous 4xx and 5xx responses.
        On completion, the reply states that only 128 kb/s lowest-numbered response is available. returned if there
        were no 2xx or 3xx responses.

   6xx: The
   original request specified GSM audio, H.261 video, proxy MUST forward the response to the client and WB whiteboard. send an
        ACK. Other pending requests SHOULD be terminated with CANCEL.

   The audio coding and whiteboard are not available, but proxy server must maintain state until all responses have been
   received or for 60 seconds if the reply
   states that DVI, PCM request was multicast.

   After receiving a 2xx or LPC audio could be supported in order 6xx response, the server SHOULD terminate
   all other pending requests by sending a  CANCEL request and closing
   the TCP connection, if applicable. (Terminating pending requests is
   advisable as searches consume resources. Also,  INVITE requests may
   "ring" on a number of
   preference. The reply also states that multicast workstations if the callee is not available.
   In such currently logged
   in more than once.)

   When operating in this mode, a case, proxy server MUST ignore any responses
   received for  Call-IDs for which it might be appropriate does not have a pending
   transaction. (If server were to forward responses not belonging to set up a transcoding
   gateway and re-invite
   current transaction using the user.

9.4 OPTIONS Request

   A caller Alice can use an  OPTIONS  Via field, the requesting client would
   get confused if it has just issued another request to find out using the
   capabilities of same
   Call-ID.)

   If a potential callee Bob, without "ringing" proxy server receives a  BYE request for a pending search or the
   designated address.
   TCP connection initiating the search is closed by the upstream
   client, the proxy server SHOULD terminate all pending requests by
   sending a BYE request and closing the TCP connections, if applicable.

11.5 Proxy Server Issuing Several  ACK and  BYE Requests

   In this case, Bob indicates that he can be
   reached at three different addresses, ranging most cases,  ACK and  BYE requests will bypass proxies and reach
   the desired party directly, keeping proxies from voice-over-IP to a
   PSTN phone 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 pager.

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

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

   Alternatively, Bob could have returned a description period of

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

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

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

10 Compact Form

   When SIP is carried over UDP with authentication and its choosing. If a complex
   session description, it may be possible
   proxy still has list of destinations that it forwarded the size of a request 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
   reply is larger than have not yet responded to the MTU. To reduce this problem, a more compact
   form of SIP is also defined by using alternative names last
   INVITE.

   If the proxy has no call state for common
   header fields.  These short forms are NOT abbreviations, they are
   field names. No other abbreviations are allowed.

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

   Thus
   destination, it forwards the header in section 9.1 could also be written:

     INVITE schooler@vlsi.caltech.edu SIP/2.0
     v:SIP/2.0/UDP 239.128.16.254 16
     v:SIP/2.0/UDP 131.215.131.131
     v:SIP/2.0/UDP 128.16.64.19
     f:mjh@isi.edu
     t:schooler@cs.caltech.edu
     i:62729-27@128.16.64.19
     c:application/sdp
     l:187

     v=0
     o=user1 53655765 2353687637 IN IP4 128.3.4.5
     s=Mbone Audio
     i=Discussion of Mbone Engineering Issues
     e=mbone@somewhere.com
     c=IN IP4 224.2.0.1/127
     t=0 0
     m=audio 3456 RTP/AVP 0

   Mixing short field names request to all downstream servers.

12 Security Considerations

12.1 Confidentiality and Privacy: Encryption

12.1.1 SIP Transactions

   SIP transactions can contain sensitive information about the
   communication patterns and long field names is allowed, but not
   recommended. Servers MUST accept both short communication content of individuals and long field names
   thus should be protected against eavesdropping. The SIP message body
   may also contain encryption keys for
   requests. Proxies MUST NOT translate a request between short and long
   forms if authentication fields are present.

11 the session itself.

   SIP Transport supports three complementary forms of encryption to protect
   privacy:

        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 defined so it can use either UDP or TCP as calling whom;

        o hop-by-hop encryption of  Via fields to hide the route a transport
   protocol.

11.1 Achieving Reliability For UDP Transport

11.1.1 General Operation
          request has taken.

   Not all of the SIP assumes no additional reliability from IP. Requests or replies
   may request can be lost. A encrypted end-to-end because header
   fields such as  To and  Via need to be visible to proxies so that the
   SIP client SHOULD simply retransmit a request can be routed correctly. Hop-by-hop encryption encrypts
   the entire SIP request
   periodically with timer T1 (default value of T1: once a second) until
   it receives a response, or until it has reached a set limit response on the
   number of retransmissions. The default limit wire (the request may
   already have been end-to-end encrypted) so that packet sniffers or
   other eavesdroppers cannot see who is 20.

   SIP requests calling whom. Note that proxies
   can still see who is calling whom, and replies 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 matched up used to route a response back along the path
   taken by the client using request and to prevent infinite request loops. However,
   the
   Call-ID header field; thus, information given by them may also provide useful information to
   an attacker. Section 6.22 describes how a server sender can only have one outstanding request per call at any given time.

   If that Via
   fields be encrypted by cooperating proxies without compromising the reply is a provisional response,
   purpose of the initiating client SHOULD
   continue retransmitting Via field.

12.2 End-to-End Encryption

   End-to-end encryption relies on keys shared by the request, albeit less frequently, using
   timer T2. The default retransmission interval T2 two user agents
   involved in the transaction. Typically, the message is 5 seconds.

   After sent encrypted
   with the server sends a final response, it cannot be sure public key of the client
   has received recipient, so that only that recipient can
   read the response, and thus message. SIP does not limit the security mechanisms that may
   be used, but all implementations SHOULD cache support PGP-based encryption.

   A SIP request (or response) is end-to-end encrypted by splitting the results for at
   least 30 seconds
   message to avoid having to, for example, contact the user or
   user location server again upon receiving be sent into a retransmission.

11.1.2 INVITE

   Special considerations apply for part to be encrypted and a short header
   that will remain in the  INVITE method.

        1.   After receiving an invitation, considerable time may elapse
             before clear. Some parts of the server can determine SIP message, namely
   the outcome. For example, request line, the called party may response line and certain header fields marked
   with "n" in the "enc." column in Table 3 need to be "rung" or extensive searches may read and returned
   by proxies and thus MUST NOT be
             performed, so delays between the request encrypted end-to-end. Possibly
   sensitive information that needs to be made available as plaintext
   include destination address ( To) and a definitive
             response can reach several tens the forwarding path ( Via) of seconds.  If either
             caller
   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 callee are automated servers not directly
             controlled by "digest"
   authentication. The  From header field SHOULD normally remain in the
   clear, but MAY be encrypted if required, in which case some proxies
   MAY return a human being, 401 (Unauthorized) status if they require a call attempt may  From field.

   Other header fields MAY be
             unbounded encrypted or MAY travel in time.

        It is undesirable to retransmit the  INVITE request, clear as this
        introduces additional network traffic.
   desired by the sender. The retransmission
        interval would have to  Subject,  Allow, Call-ID, and  Content-
   Type header fields will typically be no more than about a second, since encrypted. The  Accept,
   Accept-Language, Date,  Expires,  Priority,  Require, Cseq, and
   Timestamp header fields will remain in the
        callee would encounter a "dead" voice path if clear.

   All fields that will remain in the "200 OK"
        response is lost.

        2.   It is possible clear MUST precede those that will
   be encrypted. The message is encrypted starting with the invitation request reaches first
   character of the
             callee first header field that will be encrypted and the callee is willing
   continuing through to take the call, but that the final response (200 OK, in this case) is lost on end of the
             way message body. If no header
   fields are to be encrypted, encrypting starts with the caller. If second CRLF
   pair after the session still exists but last header field, as shown below. Carriage return and
   line feed characters have been made visible as "$", and the encrypted
   part of the message is outlined.

     INVITE watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP 169.130.12.5$
     To: watson@bell-telephone.com (T. A. Watson)$
     From: a.g.bell@bell-telephone.com (A. Bell)$
     Encryption: PGP version=5.0$
     Content-Length: 224$
     CSeq: 488$
     $
   *******************************************************
   * Call-ID: 187602141351@worcester.bell-telephone.com$ *
   * Subject: Mr. Watson, come here.$                    *
   * Content-Type: application/sdp$                      *
   * $                                                   *
   * v=0$                                                *
   * o=bell 53655765 2353687637 IN IP4 128.3.4.5$        *
   * c=IN IP4 135.180.144.94$                            *
   * m=audio 3456 RTP/AVP 0 3 4 5$                       *
   *******************************************************

   An  Encryption header field MUST be added to indicate the
             initiator gives up on including encryption
   mechanism used. A  Content-Length field is added that indicates the user,
   length of the contacted
             user has sufficient information to be able to join encrypted body. The encrypted body is preceded by a
   blank line as a normal SIP message body would be.

   Upon receipt by the
             session. However, if called user agent possessing the session no longer exists because correct
   decryption key, the invitation initiator "hung up" before message body as indicated by the reply arrived  Content-Length
   field is decrypted, and the session was now-decrypted body is appended to be two-way, the conferencing system
             should be prepared to deal with this circumstance.

        3.   If a telephony user interface
   clear-text header fields. There is modeled or if we no need to
             interface to for an additional
   Content-Length header field within the PSTN, encrypted body because the caller will provide "ringback",
             a signal that
   length of the callee actual message body is being alerted. Once the callee
             picks up, unambiguous after decryption.

   Had no SIP header fields required encryption, the caller needs to know so message would have
   been as below. Note 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 encrypted body must then include a blank
   line (start with CRLF) to hear
             ringback while the callee assumes disambiguate between any possible SIP
   header fields that might have been present and the call exists.

        4.   The client has to SIP message body.

     INVITE watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP 169.130.12.5$
     To: watson@bell-telephone.com (T. A. Watson)$
     From: a.g.bell@bell-telephone.com (A. Bell)$
     Encryption: PGP version=5.0$
     Content-Type: application/sdp$
     Content-Length: 107$
     $
   *************************************************
   * $                                             *
   * v=0$                                          *
   * o=bell 53655765 2353687637 IN IP4 128.3.4.5$  *
   * c=IN IP4 135.180.144.94$                      *
   * m=audio 3456 RTP/AVP 0 3 4 5$                 *
   *************************************************

12.2.1 Privacy of SIP Responses

   SIP requests may be able sent securely using end-to-end encryption and
   authentication to terminate a called user agent that sends an on-going request,
             e.g., because it insecure
   response.  This is no longer willing to wait 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
             connection or search called user agent to succeed. One cannot rely on infer what a
   reasonable behaviour was. Thus when end-to-end encryption is used by
   the
             absence of request retransmission, since originator, the server would
             have encryption key to continue honoring the request for several request
             retransmission periods, that is, possible tens of seconds
             if only one or two packets can be lost.

   The first problem is solved by indicating progress to the caller: used for the
   server returns a provisional
   response indicating it is searching or
   ringing SHOULD be specified in the user.

   The second and third problems are solved by having request. If this were not done,
   it might be possible for the server
   retransmit called user agent to incorrectly infer
   an appropriate key to use in the final response at intervals of T3 (default value of T3
   = 2 seconds) until it receives response. Thus, to prevent key-
   guessing becoming an  ACK acceptable strategy, we specify that a called
   user agent receiving a request that does not specify a key to be used
   for the same Call-ID
   and  CSeq or until it has retransmitted the final response 10 times.
   The  ACK SHOULD send that response unencrypted.

   Any SIP header fields that were encrypted in a request is acknowledged only once. If should also be
   encrypted in an encrypted response.  Location response fields MAY be
   encrypted if the request information they contain is
   syntactically valid and the  Request-URI matches that sensitive, or MAY be
   left in the  INVITED clear to permit proxies more scope for localized
   searches.

12.2.2 Encryption by Proxies

   Normally, proxies are not allowed to alter end-to-end header fields
   and message bodies. Proxies MAY, however, encrypt an unsigned request
   or response with the same  Call-ID, key of the server answers with status code
   200, otherwise with status code 400.

   Fig. 8 and 9 show call recipient.

        Proxies may need to encrypt a SIP request if the client and server state diagram for
   invitations.

11.2 Connection Management for TCP

   A single TCP connection can serve one end system
        cannot perform encryption or more to enforce organizational
        security policies.

12.2.3 Hop-by-Hop Encryption

   It is RECOMMENDED that SIP transactions. A
   transaction contains zero or more provisional responses followed transactions are also protected by
   exactly one final response.

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

    event
   -------
   message

   Figure 8: State transition diagram of client for  INVITE method

   The client MAY close the connection
   security mechanisms at any time. Closing the
   connection before receiving a final response signals transport and network layer.

12.2.4 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 6.22) SHOULD encrypt the header field. As only the proxy that
   encrypts the client
   wishes to abort field will decrypt it, the request.

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

    event
   ------- 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

   Figure 9: State transition diagram looping, and so an
        additional ID must be added so that the proxy can detect loops.
        This should not normally be the address of server 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  INVITE method 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 SHOULD NOT close may use a secret key to encrypt the TCP connection until it has sent its
   final response, at which point it MAY close  Via field, a
          timestamp and an appropriate checksum in any such message with
          the TCP connection if it
   wishes to. However, normally it same secret key. The checksum is the client's responsibility needed to
   close detect whether
          successful decoding has occurred, and the connection.

   If timestamp is
          required to prevent possible response attacks and to ensure
          that no two requests from the server leaves same previous hop have the connection open, and if same
          encrypted  Via field.

   The latter is the client so
   desires it preferred solution, although proxy developers may re-use
   devise other methods that might also satisfy the connection for further requirements.

12.3 Message Integrity and Access Control: Authentication

   An active attacker may be able to modify and replay SIP requests or for
   requests from and
   responses unless protective measures are taken. In practice, the same family of protocols (such as HTTP or stream
   control commands).

12 Behavior
   cryptographic measures that are used to ensure the authenticity of
   the SIP Servers

   This section describes behavior message also serve to authenticate the originator of a SIP server in detail. Servers
   can operate in proxy the
   message.

   Transport-layer 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 network-layer authentication may be used for hop-
   by-hop authentication. SIP also extends the HTTP WWW-Authenticate
   (Section 6.45 and Authorization (Section 6.11) header field containing its own
   address into those requests that are caused by an incoming request.
   To prevent loops, and their
   Proxy- counterparts to include cryptographically strong signatures.
   SIP also supports the HTTP "basic" authentication scheme [33] that
   offers a server MUST check if its own address is already
   contained in very rudimentary mechanism of ascertaining the  Via header identity of
   the incoming request.

   We define an "A--B proxy" as a proxy that receives caller.

        Since SIP requests over
   transport protocol A and issues requests, acting as a are often sent to parties with which no
        prior communication relationship has existed, we do not
        specify authentication based on shared secrets.

   SIP client, requests may be authenticated using transport protocol B. If not stated explicitly, rules apply the  Authorization header
   field to
   any combination include a digital signature of transport protocols. For conciseness, we only
   describe behavior with UDP certain header fields, the
   request method and version number and TCP, but the same rules apply for any
   unreliable datagram or reliable protocol, respectively. payload, none of which are
   modified between client and called user agent. The detailed connection behavior for UDP Authorization
   header field may be used in requests to end-to-end authenticate the
   request originator to proxies and the called user agent, and TCP is described in
   Section 11.

12.1 Redirect Server

   A redirect server
   responses to authenticate the called user agent or proxies returning
   their own failure codes. It does not issue any provide hop-by-hop
   authentication, which may be provided if required using the IPSEC
   Authentication Header.

   SIP requests of its own. It can
   return 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 response 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 refuses or redirects all the request. After
   receiving an  INVITE header fields
   following the Authorization header field have been included in the
   signature.  To sign a request, a redirect server proceeds through client removes all of the
   following steps:

        1.   If SIP header
   from before where the  INVITE request cannot be answered immediately
             (e.g., because a location server needs to  Authorization field will be contacted), it
             returns one or more provisional responses.

        2.   Once added. It then
   prepends the server has gathered request method (in upper case) followed by the list of alternative
             locations or has decided SIP
   version number field (in upper case) directly to refuse the call, it returns start of the
             final response.
   message with no whitespace, CR or LF characters inserted. This ends the SIP transaction.

   The redirect server maintains transaction state for
   extended message is what is signed.

   For example, if the whole SIP
   transaction.

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

12.3 Proxies Issuing Single Unicast Requests

   Proxies in this category issue at most a single unicast request for
   each incoming SIP request, that is, they do not "fork" requests.
   Servers may choose is to always operate in the mode described in Section
   12.4.

   The server can forward be:

   INVITE watson@boston.bell-telephone.com SIP/2.0
   Via: SIP/2.0/UDP 169.130.12.5
   Authorization: PGP version=5.0, signature=...
   From: a.g.bell@bell-telephone.com (A. Bell)
   To: watson@bell-telephone.com (T. A. Watson)
   Call-ID: 187602141351@worcester.bell-telephone.com
   Subject: Mr. Watson, come here.
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=bell 53655765 2353687637 IN IP4 128.3.4.5
   c=IN IP4 135.180.144.94
   m=audio 3456 RTP/AVP 0 3 4 5

   Then the request data block that is signed is:

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

   v=0
   o=bell 53655765 2353687637 IN IP4 128.3.4.5
   c=IN IP4 135.180.144.94
   m=audio 3456 RTP/AVP 0 3 4 5

   Note that if a message is encrypted and any responses. It does not
   have to maintain any state for authenticated using a digital
   signature, when the SIP transaction. Reliability message is
   assured by generated encryption is performed
   before the next redirect server in digital signature is generated. On receipt, the server chain. digital
   signature is checked before decryption.

   A proxy client MAY require that a server SHOULD cache the result of any address translations
   and the sign its response to speed forwarding. After by including a
   Require: org.ietf.sip.signed-response request header field. The
   client indicates the cache entry has been
   expired, desired authentication method via the server cannot tell whether an incoming WWW-
   Authenticate header.

   The correct behaviour in handling unauthenticated responses to a
   request that requires authenticated responses is
   actually a retransmission of an older request, where the TCP side has
   terminated. described in section
   12.3.1.

12.3.1 Trusting responses

   It will treat it as a new request.

12.4 Proxy Server Issuing Several Requests

   All requests carry the same  Call-ID. For unicast, each of the may be possible for an eavesdropper to listen to requests has and to
   inject unauthenticated responses that would terminate, redirect or
   otherwise interfere with a different (host-dependent)  Request-URI. For
   multicast, call. (Even encrypted requests contain
   enough information to fake a single request is issued, likely response.)

   Client should be particularly careful with 3xx redirection responses.
   Thus a host-independent
   Request-URI. A client receiving receiving, for example, a multicast query does 301 (Moved Permanently) which
   was not have to
   check whether authenticated when the host part public key of the  Request-URI matches its own host
   or domain name. To avoid response implosion, servers called user agent is
   known to the client, and authentication was requested in the request
   SHOULD NOT
   answer multicast requests with be treated as suspicious. The correct behaviour in such a 404 (Not Found) status code.
   Servers MAY decide not to answer multicast requests if their response case
   would be 5xx.

   The server MAY respond to for the request immediately with called-user to form a "100 Trying"
   or "180 Ringing" response; otherwise it MAY wait until either dated response containing the
   first
   Location field to be used, to sign it, and give this signed stub
   response to its requests or the UDP retransmission interval.

   The following pseudo-code describes the behavior of a proxy server
   issuing several requests in that will provide the redirection. Thus the
   response to an incoming request. The
   function request(r, a) sends can be authenticated correctly. There may be circumstances
   where such unauthenticated responses are unavoidable, but a client
   SHOULD NOT automatically redirect such a SIP request r to address a.
   await_response() waits until the new location
   without alerting the user to the authentication failure before doing
   so.

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

   If TCP is received being used, a proxy SHOULD treat 4xx and returns the
   response. close(a) closes 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 TCP connection client, and failure to client with address
   a. response(s, l, L) sends do
   so SHOULD cause such a response client to ignore the client with status s 6xx response and
   list of locations L, with l entries. ismulticast() returns 1 if continue
   a search.

   With UDP, the
   location is 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 multicast address failure occurred when in fact it
   did not. Typically 4xx and zero otherwise. The variable
   timeleft indicates 5xx responses will not be signed by the amount
   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 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 SIP request, which removes any additional problems
   that UDP might have over TCP.

   These attacks are within the same domain as prevented by having the proxy server. Thus, an
   initial multicast request may trigger additional unicast requests.

     enum {INVITE,         /* request type */
       ACK, OPTIONS, BYE, REGISTER, UNREGISTER} R;
     int N = 0;            /* number of connection attempts */
     address_t address[];  /* list of addresses */
     int done[];           /* address has responded */
     location[];           /* list of locations */
     int heard = 0;        /* number of sites heard from */
     int class;            /* class of status code */
     int best = 1000;      /* best client require response so far */
     int timeleft = 120;   /* sample timeout value */
     int loc = 0;          /* number of locations */
     struct {              /*
   authentication and dropping unauthenticated responses. A server user
   agent that cannot perform response */
       int status;         /* authentication responds using the
   normal  Require response status */
       char *location;     /* redirect locations */
       address_t a;        /* address of respondent */
     } r;
     int i;

     if (multicast) {
       request(R, address[0]);
     } else {
       N = /* number 420 (Bad Extension).

12.4 Callee Privacy

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

12.5 Known Security Problems

   With either TCP or UDP, a denial of service attack exists by a rogue
   proxy sending 6xx responses. Although a client SHOULD choose to try */
       for (i = 0; i < N; i++) {
         request(R, address[i]);
         done[i] = 0;
       }
     }

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

       if (class >= 2) {
         heard++;
         for (i = 0; i < N; i++) {
   ignore such responses if (address[i] == r.a) {
             done[i] = 1;
             break;
           }
         }
       } 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 (class == 2) {
         best = r.status;
         break;
       }
       else 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 (class == 3) {
             /* 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
   follows:

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

   pgp-algorithm: A server may optionally recurse.  The server MUST string indicating the PGP message integrity check whether
              * it has tried
        (MIC) to be used to produce the signature. If this location before not present
        it is assumed to be "md5". The currently defined values are
        "md5" for the MD5 checksum, and whether "sha1" for the location is
              * part SHA.1 algorithm.

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

   Example:

   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 incoming request.  This check request, passing an Authorization
   header line, which is
              * omitted here for brevity. Multicast locations 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 NOT be
          * returned correspond to the client if  From header of the server is not recursing.
          */
         if (recurse) {
           multicast = 0;
           N++;
           request(R, r.location);
         } else if (!ismulticast(r.location)) {
           locations[loc++] = r.location;
           best = r.status;
         }
       }
       else if (class == 4) {
         if (best >= 400) best = r.status;
       }
       else if (class == 5) {
         if (best >= 500) best = r.status;
       }
       else if (class == 6) {
         best = r.status;
         break;
       }
     }
     /* We haven't heard anything useful from anybody. */
     if (best == 1000) {
       best = 404;
     }
     if (best/100 != 3) loc = 0;
     response(best, loc, locations);

     /*
      * Close request
   unless the other pending transactions by sending BYE.
      */
     for (i = 0; i < N; i++) {
       if (!done[i]) {
         request(BYE, address[i]);
         if (tcp) close(a);
       }
     }
   After receiving a 2xx or 6xx response,  signed-by parameter is provided.

   pgp-signature: The PGP ASCII-armored signature, as it appears between
        the server SHOULD terminate
   all other pending requests by sending a  BYE "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 closing version are prepended to the
   TCP connection, if applicable. (Terminating pending requests header fields without any
   white space. The signature is
   advisable computed across the headers as searches consume resources. Also,  INVITE requests may
   "ring" on a number of workstations if sent,
   including any folding and the callee terminating CRLF. The CRLF following
   the Authorization header is currently logged NOT included in more the signature.

        Using the ASCII-armored version is about 25% less space-
        efficient than once.)

   [TBD: How do we cancel multicast requests? Force receivers to listen including the binary signature, but it is
        significantly easier for a 200/6xx response and hope that they don't miss one?]

   When operating the receiver to piece together.
        Versions of the PGP program always include the full
        (compressed) signed text in this mode, a proxy server MUST ignore any responses
   received for  Call-IDs 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 it does proxy servers
        are not have a pending transaction
   for. (If server were allowed to forward responses 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 belonging to a current
   transaction using signed by the  Via field, entity
        listed in the requesting client would get
   confused if it has just issued another request using  From header, the same Call-
   ID.)

13 Third-Party Call Initiation

   In some circumstances, third-party call control is required, where  signed-by header indicates the calling party suggests to
        name of the called party to invite signing entity, expressed as a (small)
   number URI.

   Receivers of other parties. Third-party call control can be used to
   implement signed SIP messages SHOULD discard any end-to-end header
   fields above the following features:

   Multipoint-control unit (MCU): Some conferences use  Authorization header, as they may have been
   maliciously added en route by a multipoint
        control unit proxy.

   Example:

   Authorization: pgp version="5.0",
     realm="Your Startrek identity, please",
     signature="iQB1AwUBNNJiUaYBnHmiiQh1AQFYsgL/Wt3dk6TWK81/b0gcNDf
     VAUGU4rhEBW972IPxFSOZ94L1qhCLInTPaqhHFw1cb3lB01rA0RhpV4t5yCdUt
     SRYBSkOK29o5e1KlFeW23EzYPVUm2TlDAhbcjbMdfC+KLFX
     =aIrx"

13.2 PGP Encryption Scheme

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

   encoding: Describes the encoding or "armor" used by PGP. The value
        "ascii" refers to mix, convert the standard PGP ASCII armor, without the
        lines containing "BEGIN PGP MESSAGE" and "END PGP MESSAGE" and replicate media streams. While
        this solution has scaling problems, it
        without the version identifier. By default, the encrypted part
        is widely deployed in
        traditional telephony and ISDN conferencing settings, included as so-
        called conference bridges. In a MCU-based conference, binary.

   Example:

   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 ".

   Example:

   Response-Key: pgp version="2.6.2", encoding="ascii",
     key="mQBtAzNWHNYAAAEDAL7QvAdK2utY05wuUG+ItYK5tCF8HNJM60sU4rLaV+eUnkMk
     mOmJWtc2wXcZx1XaXb2lkydTQOesrUR75IwNXBuZXPEIMThEa5WLsT7VLme7njnx
     sE86SgWmAZx5ookIdQAFEbQxSGVubmluZyBTY2h1bHpyaW5uZSA8c2NodWx6cmlu
     bmVAY3MuY29sdW1iaWEuZWR1Pg==
     =+y19"

14 Examples

14.1 Registration

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

   C->S: REGISTER sip:@bell-tel.com SIP/2.0
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Location: sip:saturn.bell-tel.com:3890;transport=udp
         Expires: 7200

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

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

   C->S: REGISTER sip:@bell-tel.com SIP/2.0
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Expire: 0
         Location: *

   C->S: REGISTER sip:@bell-tel.com SIP/2.0
         From: sip:watson@bell-tel.com
         To: sip:watson@bell-tel.com
         Location: sip:tawatson@example.com
   Now, the
        conference initiator or server will forward any authorized member invites a new
        participant and indicate the address of the MCU in the  Also
        header. The invitee then contacts request for Watson to the MCU server at
   example.com , using the same session
        description and requests  Request-URI tawatson@example.com

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

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

   The request could be added send to either the registrar at bell-tel.com or
   the server at example.com example.com would proxy the call, just like a
        normal two-party call.

   Telephony call initiation ("click-to-call"): A SIP  INVITE request
        containing two addresses to the
   address indicated in the  Also  Request-URI. Then,  Max-Forwards header is sent
   could be used to restrict the registration to a PSTN
        service node that connects these two addresses by a telephone
        call.

   Fully-meshed small conference: For small conferences, such as adding
        a third party server.

14.2 Invitation to Multicast Conference

   The first example invites schooler@vlsi.cs.caltech.edu to a two-party call, multicast may not always
   session. All examples use the Session Description Protocol (SDP) (RFC
   2327 [7]) as the session description format.

14.2.1 Request

   C->S: INVITE sip:schooler@vlsi.cs.caltech.edu SIP/2.0
         Via: SIP/2.0/UDP 239.128.16.254 16
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19
         From: Mark Handley <sip:mjh@isi.edu>
         To: Eve Schooler <sip:schooler@caltech.edu>
         Subject: SIP will be
        appropriate or available. Instead, when inviting a new
        participant, discussed, too
         Call-ID: 19971205T234505.56.78@oregon.isi.edu
         Content-Type: application/sdp
         CSeq: 4711
         Content-Length: 187

         v=0
         o=user1 53655765 2353687637 IN IP4 128.3.4.5
         s=Mbone Audio
         i=Discussion of Mbone Engineering Issues
         e=mbone@somewhere.com
         c=IN IP4 224.2.0.1/127
         t=0 0
         m=audio 3456 RTP/AVP 0
   The  Via fields list the caller asks hosts along the new member to call path from invitation
   initiator (the last element of the
        remaining members. TBD: Should list) towards the call-id be invitee. In the same or
        different? Need to distinguish between new INVITE for same call
        and adding a party
   example above, the message was last multicast to a call. Include conference identifier?
   TBD: How about just transferring an SDP description the administratively
   scoped group 239.128.16.254 with multiple
   addresses?

   The  Also: header (Section 6.9) is used to indicate a list ttl of parties 16 from the host
   131.215.131.131

   The request header above states that the callee should invite.

14 ISDN and Intelligent Network Services

   SIP may be used to support a number of ISDN [27] and Intelligent
   Network [28] telephony services, described below. Due to request was initiated by
   mjh@isi.edu from the
   fundamental differences between Internet-based telephony and
   conferencing as compared to public switched telephone network
   (PSTN)-based services, service definitions cannot be precisely host 128.16.64.19 schooler@caltech.edu is being
   invited; the
   same.  Where large differences beyond addressing and location of
   implementation exist, this message is indicated below. The term address
   implies any SIP address. (Section 1.4.1).

   Call transfer (TRA) enables a user to transfer an established (i.e.,
        active) call currently being routed to a third party. SIP signals
   schooler@vlsi.cs.caltech.edu

   In this via the Location
        header in the  BYE (Section 4.2.4) method.

   Call forwarding (CF) permits case, the called user to forward particular
        pre-selected calls to another address. Unlike telephony, session description is using the
        choice of calls to be forwarded depends on program logic
        contained Session
   Description Protocol (SDP), as stated in any of the SIP servers  Content-Type header.

   The header is terminated by an empty line and can thus be made
        dependent on time-of-day, subject of call, media types, urgency
        or caller identity, rather than being restricted to matching
        list entries. This forwarding service encompasses:

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

   Selective call forwarding (SCF) permits
   message body containing the user to have her incoming
        calls addressed to another network destination, no matter what session description.

14.2.2 Response

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

   S->C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP 239.128.16.254 16 ;branch=17
         Via: SIP/2.0/UDP csvax.cs.caltech.edu ;branch=8348
         Via: SIP/2.0/UDP north.east.isi.edu
         To: Eve Schooler <sip:schooler@caltech.edu>
         From: Mark Handley <sip:mjh@isi.edu>
         Call-ID: 19971205T234505.56.78@north.east.isi.edu
         Location: sip:es@jove.cs.caltech.edu
         CSeq: 4711

   A sample response to the calling address invitation is included
        in, or excluded from, a screening list. given below. The user's originating
        service is unaffected.

   Completion first line of calls to busy subscriber (CCBS) allows a calling user
        encountering a busy destination to be informed when
   the busy
        destination becomes free, without having to make a new call
        attempt. response states the SIP supports services close to CCBS by allowing a
        callee to indicate version number, that it is a more opportune time to call back (Section
        6.25). Also, calling and called user agents can easily record 200 (OK)
   response, which means the URL of outcoming request was successful. The  Via headers
   are taken from the request, and incoming calls, so that a user can re-
        try or return calls with a single mouse click.

   Conferencing (CON) allows entries are removed hop by hop as the user to communicate simultaneously with
        multiple parties, which may also communicate among themselves.
        SIP can initiate IP multicast conferences with any number
   response retraces the path of
        participants, conferences where media are mixed the request. A new authentication field
   MAY be added by a conference
        bridge (multipoint control unit or MCU) and, for exceptional
        applications the invited user's agent if required. The  Call-ID is
   taken directly from the original request, along with a small number the remaining
   fields of participants, fully-meshed
        conferences, where each participant sends and receives data to
        all other participants.

   Conference calling add-on allows a user to add and drop participants
        once the conference request message. The original sense of  From field is active.

   Conference calling meet-me (MMC) allows
   preserved (i.e., it is the session initiator).

   In addition, the  Location header gives details of the host where the
   user to set up a
        conference was located, or multi-party call, indicating alternatively the date, time,
        conference duration, conference media and other parameters. relevant proxy contact point
   which should be reachable from the caller's host.

   S->C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP 239.128.16.254 16 ;branch=17
         Via: SIP/2.0/UDP csvax.cs.caltech.edu ;branch=8348
         Via: SIP/2.0/UDP north.east.isi.edu
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 19971205T234505.56.78@north.east.isi.edu
         Location: sip:es@jove.cs.caltech.edu
         CSeq: 4711

   The
        conference session description included in caller confirms the SIP invitation
        may indicate by sending a time in the future. For multicast conferences,
        participants do not have to connect using SIP at the actual time
        of the conference; instead, they simply subscribe request to the
        multicast addresses listed
   location named in the announcement.  Location header:

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

14.3 Two-party Call

   For MCU-based
        conferences, two-party Internet phone calls, the session description may response must contain the address a
   description of
        the MCU where to be called at the time of send the conference.

   Destination call routing (DCR) allows customers to specify data. In the
        routing of their incoming example below, Bell
   calls to destinations according to

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

        -area of call origination;

        -network address of caller;

        -service attributes;

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

        -charge rates applicable for the destination;

        -proportional routing of traffic.

   In SIP, destination call routing is implemented by proxy 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 169.130.12.5
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com>
         Call-ID: 1985853074@kensington.bell-tel.com
         Subject: Mr. Watson, come here.
         CSeq: 17
         Content-Type: application/sdp
         Content-Length: ...

         v=0
         o=bell 53655765 2353687637 IN IP4 128.3.4.5
         c=IN IP4 135.180.144.94
         m=audio 3456 RTP/AVP 0 3 4 5

   S->C: SIP/2.0 200 OK
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: sip:watson@bell-tel.com
         Call-ID: 1985853074@kensington.bell-tel.com
         CSeq: 17
         Location: sip:watson@boston.bell-tel.com
         Content-Length: ...

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

   Watson can only receive PCMU and redirect
   servers GSM. Note that implement custom call handling logic, with parameters
   including, but Watson's list of
   codecs may or may not limited to be a subset of the set listed above.

   Follow-me diversion (FMD) allows one offered by Bell, as each
   party indicates the service subscriber data types it is willing to remotely
        control the redirection (diversion) of calls from his primary
        network address receive. Watson will
   send audio data to other locations.

   In SIP, finding the current network-reachable location of a callee is
   left port 3456 at 135.180.144.94, Bell will send to
   port 5004 at 135.180.161.25.

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

   Since the scope of this
   specification. However, users may use two sides have agree on the  REGISTER method (Section
   4.2.5) to appraise their "home" SIP server set of their new location.

   Originating call screening (OCS) controlls media, Watson confirms
   the ability of call without enclosing another session description:

   C->S: ACK sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP 169.130.12.5
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. Watson <sip:watson@bell-tel.com>
         Call-ID: 1985853074@kensington.bell-tel.com
         CSeq: 17
         Content-Length: 0

14.4 Terminating a node to
        originate calls. In Call

   To terminate a fashion similar to closed user groups, call, caller or callee can send a
        firewall would have to be used to restrict  BYE request:

   C->S: BYE sip:watson@boston.bell-tel.com SIP/2.0
         From: A. Bell <sip:a.g.bell@bell-tel.com>
         To: T. A. Watson <sip:watson@bell-tel.com>
         Call-ID: 1985853074@kensington.bell-tel.com
         CSeq: 18
   If the ability callee wants to
        initiate SIP invitations outside a designated part of 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
        network. same proxies as the original INVITE.

14.5 Forking Proxy

   In many cases, gateways 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 PSTN will require
        appropriate authentication.

   Premium rate (PRM) allows to pay back part time of the call cost to 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 party, considered proxy.ieee.org
   registration for the home machine of Watson, at watson@h.bell-tel.com
   (H), as an added value provider. See
        discussion on billing services below.

   Split charging (SPL) allows well as a permanent registration at watson@acm.org (A). For
   brevity, the calling and called party being each
        charged examples omit the session description.

   Watson's user agent sends the invitation to the SIP server for one part of the call. See discussion on billing
        services below.

   Universal access number (UAN) allows a subscriber with several
        network
   ieee.org domain:

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

   The SIP server tries the four addresses in parallel. It sends the
   following message to be reached with the home machine:

   P->H: INVITE sip:watson@h.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP proxy.ieee.org ;branch=1
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 1985853074@c.bell-tel.com
         CSeq:    19

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

   The subscriber may specify which incoming calls are to be routed proxy  ACKs the response so that host H can stop retransmitting
   it:

   P->H: ACK sip:watson@h.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP proxy.ieee.org ;branch=1
         If-Match: "4711"
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     19

   Also, P attempts to which address. SIP offers this functionality reach Watson through proxies
        and redirection.

   Universal personal telecommunications (UPT) is a mobility service
        which enables subscribers to be reached the ACM server:

   P->A: INVITE sip:watson@acm.org SIP/2.0
         Via:      SIP/2.0/UDP proxy.ieee.org ;branch=2
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     19

   In parallel, the next attempt proceeds, with a unique personal
        telecommunication number (PTN) across multiple networks at any
        network access. The PTN will be translated to an appropriate
        destination address for routing based on the capabilities
        subscribed  INVITE to by each service subscriber. A person may have
        multiple PTNs, e.g., a business X and private PTN. In SIP, Y:

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

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

   As it happens, both Watson at X and a colleague in the
        host-independent address (Section 1.4.1) of other lab at
   host Y hear the form user@host
        serves as phones ringing and pick up. Both X and Y return 200s
   via the PTN, which is translated into one or more host-
        dependent addresses.

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

   Some telephony services can be provided by not strictly necessary here,
   since the end system, without
   involvement by SIP:

   Abbreviated dialing allows users  Location header is unambiguous.

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

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

   This response is forwarded to reach local subscribers without
        specifying the full address (domain or host name). For SIP, the
        user application completes Bell, using the address to be a fully qualified
        domain name.

   Call waiting (CW) allows  Via information.  At
   this point, the called party to receive ACM server is still searching its database. P can now
   cancel this attempt:

   P->A: CANCEL sip:watson@acm.org SIP/2.0
         Via:     SIP/2.0/UDP proxy.ieee.org ;branch=2
         From:    A. Bell <sip:a.g.bell@bell-tel.com>
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 1985853074@c.bell-tel.com
         CSeq:    19
   The ACM server gladly stops its neural-network database search and
   responds with a notification
        that another party is trying to reach her while she 200. The 200 will not travel any further, since P is busy
        talking to another calling party.

   For SIP-based telephony,
   the called party can maintain several call
   presences at last  Via stop.

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

   Bell gets the same time, limited by local resources. Thus, it 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
   up needed to the called party determine who he wants to decide whether
   talk to accept another call. The
   separation or he can automatically reject one of resource reservation the two calls. Here, he
   acknowledges both, separately and call control may lead directly to the
   situation that the called party accepts final destination:

   C->X: ACK sip:watson@x.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         If-Match: "1620"
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     19

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

   After a brief discussion between the incoming call, but three, it becomes clear that
   the network or system resource allocation fails. This cannot be
   completely prevented, but if the likely resource bottleneck
   Watson is at the
   local system, the user agent may be able to determine whether there
   are sufficient resources available or roughly track its own resource
   consumption.

   Consultation calling (COC) allows a subscriber to place X, thus Bell sends a call on
        hold, in order  BYE to initiate a new call for consultation. In
        systems using SIP, consultation calling can be implemented as
        two separate SIP calls, possibly Y, which is replied to:

   C->Y: BYE sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         If-Match: "2016"
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     20

   Y->C: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>
         To:       T. Watson <sip:t.watson@ieee.org>
         Call-ID:  1985853074@c.bell-tel.com
         CSeq:     20

14.6 Redirects

   Replies with status codes 301 (Moved Permanently) or 302 (Moved
   Temporarily) specify another location using the temporary release of
        reserved resources for  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 128.16.64.19
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 3779067998@oregon.isi.edu
         Location: sip:@239.128.16.254;ttl=16;transport=udp
         CSeq: 19
         Content-Length: 0

   In this example, the call proxy located at csvax.cs.caltech.edu is being put on hold.

   Customized ringing (CRG) allows the subscriber to allocate a
        distinctive ringing
   advised to contact the multicast group 239.128.16.254 with a list ttl of calling parties.
   16 and UDP transport. In normal situations, a SIP-based
        system, this feature 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 offered by within the user application, based
        on caller identification ( From, Section 6.17) provided by
   scope of the
        SIP INVITE request (Section 4.2.1).

   Malicious call identification (MCI) allows local group. If the service subscriber request is redirected to
        control the logging (making a record) of calls that received
        that are of multicast
   group, a malicious nature. In SIP, by default, all calls
        identify proxy server SHOULD query the calling party and multicast address itself
   rather than sending the SIP servers that have
        forwarded redirect back towards the call. In addition, calls client as multicast
   may be authenticated
        using standard HTTP methods or transport-layer security. A
        callee may decide only to accept calls that are authenticated.

   Multiway calling (MWC) scoped; this allows a proxy within the user appropriate scope
   regions to establish multiple,
        simultaneous calls with other parties. For a SIP-based end
        system, make the considerations for consultation calling apply.

   Terminating call screening (TCS) allows query.

14.7 Alternative Services

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

   S->C: SIP/2.0 350 Alternative Service
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19
         From: sip:mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 19971205T234505.56.78@oregon.isi.edu
         Location: recorder@131.215.131.131
         CSeq: 19
         Content-Type: application/sdp
         Content-Length: 146

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

   In this case, the subscriber to specify answering server provides a session description
   that incoming calls either be restricted or allowed, according describes an "answering machine". If the invitation initiator
   decides to a screening list and/or by time take advantage of day or other parameters.

   Billing features such this service, it should send an
   invitation request to the answering machine at 131.215.131.131 with
   the session description provided (modified as account card dialing , automatic alternative
   billing , credit card calling (CCC) , reverse charging , freephone
   (FPH) , premium rate (PRM) and split charging are supported through
   authentication. However, mechanisms appropriate for indicating billing
   preferences a
   unicast session to contain the appropriate local address and capabilities have not yet been specified port for SIP.

   Advice of charge allows
   the user paying for invitation initiator). This request SHOULD contain a call to be informed of
   usage-based charging information. Charges incurred by reserving
   resources different
   Call-ID from the one in the network are probably best indicated by original request. An example would be:

   C->S: INVITE sip:mm-server@131.215.131.131 SIP/2.0
         Via: SIP/2.0/UDP 128.16.64.19
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 19971205T234505.56.78@128.16.64.19
         CSeq: 20
         Content-Type: application/sdp
         Content-Length: 146

         v=0
         o=mm-server 2523535 0 IN IP4 131.215.131.131
         s=Answering Machine
         i=Leave an audio message
         c=IN IP4 128.16.64.19
         t=0 0
         m=audio 26472 RTP/AVP 0
   Invitation initiators MAY choose to treat a protocol
   closely affiliated with the reservation protocol. Advice of charge
   when using Internet-to-PSTN gateways through SIP appears feasible,
   but is for further study. Desirable facilities include indication 350 (Alternative Service)
   response as a failure if they wish to do so.

14.8 Negotiation

   An example of
   charges at call setup time, during a 606 (Not Acceptable) response is:

   S->C: SIP/2.0 606 Not Acceptable
         From: sip:mjh@isi.edu
         To: sip:schooler@cs.caltech.edu
         Call-ID: 19971205T234505.56.78@128.16.64.19
         Location: mjh@131.215.131.131
         Warning: 606.1 Insufficient bandwidth (only have ISDN),
           606.3 Incompatible format,
           606.4 Multicast not available
         Content-Type: application/sdp
         Content-Length: 50

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

   In this example, the call original request specified 256 kb/s total
   bandwidth, and at the end of the
   call

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

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

   Third-party signaling 128 kb/s is optionally supported within SIP (Section
   6.9). Third-party signaling can be used to indicate to callees who
   else to invite to a call for MCU available.
   The original request specified GSM audio, H.261 video, and fully-meshed conferences.
   Third-party signaling, combined with appropriate URLs, may be used to
   initiate PSTN phone calls from an Internet host.

15 Security Considerations

15.1 Confidentiality

   Unless SIP transactions are protected by lower-layer security
   mechanisms such as SSL , an attacker may be able to eavesdrop on call
   establishment WB
   whiteboard.  The audio coding and invitations and, through whiteboard are not available, but
   the  Subject header field response states that DVI, PCM or the session description, gain insights into the topic LPC audio could be supported in
   order of
   conversation.

15.2 Integrity

   Unless SIP transactions are protected by lower-layer security
   mechanisms preference. The response also states that multicast is not
   available.  In such as SSL , an active attacker may a case, it might be able appropriate to modify SIP
   requests.

15.3 Access Control

   SIP requests are not authenticated unless the SIP  Authorization and
   WWW-Authenticate headers are being used. The strengths set up a
   transcoding gateway and weaknesses
   of these authentication mechanisms are re-invite the same as for HTTP.

15.4 Privacy

   User location and SIP-initiated calls may violate user.

14.9  OPTIONS Request

   A caller Alice can use an  OPTIONS request to find out the
   capabilities of a callee's privacy.
   An implementation SHOULD potential callee Bob, without "ringing" the
   designated address. In this case, Bob indicates that he can be able
   reached at three different addresses, ranging from voice-over-IP to restrict, on a per-user basis,
   what kind of location and availability information is given out
   PSTN phone to
   certain classes a pager.

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

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

   Alternatively, Bob could have returned a description of callers.

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

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

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

A Minimal Implementation

A.1 Client

   All clients MUST be able to generate the  INVITE and  ACK requests
   and MUST be able to include the  Call-ID, Content-Length,  Content-
   Type,  CSeq, From and  To headers. A minimal implementation MUST
   understand SDP [9]. (RFC 2327, [7]). In responses, it must be able to
   parse the  Call-ID,  Content-Length, Content-Type,  Require headers.
   It must MUST be able to recognize the status code classes 1 through 6 and
   act accordingly.

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

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

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

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

   Authentication: If a client wishes to invite callees that require
        caller authentication, it must be able to recognize the 401
        "Unauthorized"
        (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 MUST be able to recognize the 407 "Proxy (Proxy Authentication Required" Required)
   status code, must 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  ACK,  OPTIONS and  BYE requests. It MUST parse the and generate,
   as appropriate, the  Call-ID,  Content-Length, Content-Type,
   Expires,  From,
   PEP, Max-Forwards,  Require,  To and  Via headers. It must MUST
   echo the  Sequence header  CSeq header in the response. It SHOULD include the  Server
   header in its responses.

A.3 Header Processing

   Table 4 lists the headers that different implementations support. UAC
   refers to a user-agent client (calling user agent), UAS to a user-
   agent server (called user-agent).

B Usage of SDP

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

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

   The caller should include all media types that it is willing to send
   so that the  Server header in its responses.

B receiver can provide matching media descriptions.

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

C Summary of Augmented BNF

   In this specification we use the Augmented Backus-Naur Form notation
   described in [21]. RFC 2234 [24]. For quick reference, the following is a
   brief summary of the main features of this ABNF.

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

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

   *term
        zero of more instances of  term;

   3*term
        three or more instances of  term;

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

   [ term ]
        term is optional;

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

   term1 | term2
        either  term1 or  term2 may appear but unimportant;

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

   #term
        a comma separated list of  term;
                                   type     UAC    proxy    UAS
           ____________________________________________________
           Accept                   R        -       o       o
           Accept-Language          R        -       b       b
           Allow                   405       o       -       -
           Authorization            R        a       o       a
           Call-ID                  g        m       m       m
           Content-Length           g        m       m       m
           Content-Type             g        m       -       m
           CSeq                     g        o       m       m
           Date                     g        o       o       o
           Encryption               g        e       -       e
           Expires                  g        -       o       o
           From                     R        m       o       m
           Hide                     R        o       o       -
           Location                 R        -       -       -
           Location                 r        r       r       -
           Max-Forwards             R        -       b       -
           Organization             R        -       o       o
           Proxy-Authenticate      407       a       -       -
           Proxy-Authorization      R        -       a       -
           Proxy-Require            R        -       m       -
           Priority                 R        -       o       o
           Require                  R        m       -       m
           Retry-After           600,603     o       o       -
           Response-Key             R        -       -       e
           Server                   r        o       o       -
           Subject                  R        o       o       o
           Timestamp                g        o       o       o
           To                       g        m       m       m
           Unsupported              r        b       b       -
           User-Agent               R        -       o       o
           Via                      g        -       m       m
           Warning                  r        o       o       -
           WWW-Authenticate        401       a       -       -

   Table 4: This table indicates which systems should be able  to  parse
   which  response  header fields. Type is as table 3. "-" indicates the
   field is  not both;

   #term  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 comma separated list  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  term; the field is purely optional.
   2#term
        a comma separated list of  term containing at least 2 items;

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

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

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

   Changes in Version -04

   Since version -03, a line
        SP       =    %d32            ;  space character
        TAB      =    %d09            ;  tab character
        LWS      =    *( SP | TAB)    ;  linear whitespace
        DIGIT    =    "0" .. "9"      ;  a single decimal digit

D IANA Considerations

   Section 4.4 describes a name space and mechanism for registering SIP
   options.

   Changes in Version -05

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

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

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

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

        o Updated phone URLs to [22].

        o Proxy-Require header added, as in RTSP.

        o "Emergency" priority value added.

        o 381 (Ambiguous) status code added.

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

        o The service definitions have been moved to a separate
          document.

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

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

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

        o Max-Forwards header and status 483 added.

        o The default port number 5060 has been made.

        oThe introduction assigned to SIP. The
          address 224.0.1.75 also has been reorganized assigned as the "all SIP
          servers" multicast address.

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

        o CSeq mandatory for requests and large parts
         rewritten.

        oCONNECTED changed minimal implementations.

        o Proxy search rules made more precise.

        o Added ability of (state-ful) proxies to  ACK, as it applies request to all responses, not
         just 200.

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

        oStatus code 481 (Invalid Call-ID) added.

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

        oAllow phone numbers in SIP URL ACK packets via  Record-Route and  Route.

        o Retransmission timer for easy connection  INVITE changed to Internet
         telephony gateways.

        oAdded  Also header T3 = 30 seconds to
          avoid multiple  INVITEs for third-party connectivity.

        oWhen doing parallel searches, pending searches should be
         aborted when one address was successful. The a typical phone call call.

        o ACKs may be
         ringing on a number of workstations where the user is logged in
         and would keep ringing.

        oAdded  duration contain session descriptions.

        o signed-by parameter added to  Authorization header.

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

        o OPTIONS allowed to return current status such as Busy or
          Decline.

        o Unicast example clarified.

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

        o Tentatively added  CANCEL request to terminate searches.

        o Status 181: Call Is Being Forwarded.

        o Content-Encoding for compressed payloads.

        o Retry-After may be used for  REGISTER.

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

        o Added  method to  CSeq response header.

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

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

        o Warning header syntax fixed to allow 606.3.

        o signed-by can be reachable at general URI. However, not clear yet whether
          URI schemes are the address given.

        oChanged  Sequence most appropriate mechanism to  CSeq for consistency with RTSP.

C denote a
          principal. In X.509 certificates, email addresses are used.

E Open Issues

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

   H.323:

E.1 H.323

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

   TRANSACTION: Should we have a transaction id in addition

   Solution: Leave to separate document.

   Status: Closed.

E.2 REGISTER

   Problem: How does the UA get a call
        ID? Call-IDs are "personalized" multicast address for
        multicast searches? It would be helpful if the end system, but a transaction ID server could
        return an indication of any local search multicast address the
        user agent is supposed to listen on. The server might also want
        to indicate whether all outgoing calls should be proxied through
        the server. Use of message bodies for
        a single SIP exchange. This is useful  REGISTER requests.

   Solution: Leave for Internet telephony,
        where a single call may trigger several transactions. Also,
        avoids BYE race condition: Proxy doing parallel search cancels
        pending search follow-on work. Possible to use Location with BYE after one of the addresses responds 200
        response, but already otherwise used to indicate registrations.

   Status: Closed.

E.3 Max-Forwards
   Problem: Extend  Max-Forwards with
        200. Through another proxy, a max fan out field.  [Not really
        related.] How do you limit fanout for multicast? What's the
        advantage of doing this?

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

   Status: Closed.

E.4 Cancellation and BYE reaches the same

   Problem: Determine how to cancel

        - searches

        - ringing

        - non-selected end system
        and cancels systems

        - calls

   Solution: In spec.

   Status: Closed.

E.5 IPv6 URLs

   Problem: Numeric IPv6 addresses in URLs.

   Solution: None; wait for others to take the successful call.

D lead. Note that this
        affects only numeric addresses, which should be rarely used.

   Status: Defered.

F Acknowledgments

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

   This work is based, inter alia, on [29,30]. Parameters of
   the terminal negotiation mechanism were influenced by Scott Petrack's
   CMA design.

E [34,35].

G Authors' Addresses

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

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

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

F

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   Lee, "Hypertext transfer protocol -- HTTP/1.1," Tech. Rep. RFC 2068, Internet
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   [17] S. Williamson, M. Kosters, D. Blacka, J. Singh, and K. Zeilstra,
   "Referral whois (rwhois) protocol V1.5," RFC 2167, Internet
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   [18] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access
   protocol," Tech. Rep. RFC 1777, Internet Engineering Task Force, Mar. 1995.

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   scheme," Internet Draft, Internet Engineering Task Force, Jan. 1998.
   Work in progress.

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   unifying syntax for the expression of names and addresses of objects
   on the network as used in the world-wide web," Tech. Rep. RFC 1630, Internet
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   locators (URL): Generic syntax and semantics,"
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   Engineering Task Force, Feb. 1998.  Work in progress.

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   2279, Internet Engineering Task Force, Dec. 1994.

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   specifications:  ABNF,"
   Internet Draft, Internet Engineering Task Force, Oct. 1996.  Work in
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   Nov. 1990.

   [23] 1997.

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   Reading, Massachusetts: Addison-Wesley, 1994.

   [24]

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   Internet Engineering Task Force, Nov. 1990.

   [27] D. Crocker, "Standard for the format of ARPA internet text
   messages," Tech.  Rep. Also STD0011, RFC STD 11, 822, Internet Engineering Task Force, Aug.
   1982.

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   synchronous rendezvous," Master's Thesis CS-TR-96-18, Department of
   Computer Science, California Institute of Technology, Pasadena,
   California, Aug. 1996.

   [29] P. Resnick, "Internet message format standard," Internet Draft,
   Internet Engineering Task Force, Aug. 1997. Mar. 1998.  Work in progress.

   [26]

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   Nielsen, and J. Zawinski, "The mailto URL
   scheme," Gettys, "Hypertext transfer protocol -- HTTP/1.1,"
   Internet Draft, Internet Engineering Task Force, Oct. 1997. Mar. 1998.  Work in
   progress.

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

   [28] International Telecommunication Union, "General recommendations
   on telephone switching

   [31] P. Leach and signaling -- intelligent network:
   Introduction to intelligent network capability set 1," Recommendation
   Q.1211, Telecommunication Standardization Sector of ITU, Geneva,
   Switzerland, R. Salz, "UUIDs and GUIDs," Internet Draft,
   Internet Engineering Task Force, Feb. 1998.  Work in progress.

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

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   Stewart, S. Lawrence, and A. Luotonen, "HTTP authentication: Basic
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   Engineering Task Force, Mar. 1993.

   [29] 1998.  Work in progress.

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   Multimedia Systems and Services , (Berlin, Germany), Mar. 1996.

   Full Copyright Statement

   Copyright (c) The Internet Society (1997). (1998). All Rights Reserved.

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                           Table of Contents

   1          Introduction ........................................    2
   1.1        Overview of SIP Functionality .......................    2
   1.2        Terminology .........................................    3
   1.3        Definitions .........................................    4
   1.4        Summary of SIP Operation ............................    6
   1.4.1      SIP Addressing ......................................    6    7
   1.4.2      Locating a SIP Server ...............................    7    8
   1.4.3      SIP Transaction .....................................    9
   1.4.4      SIP Invitation ......................................    9   10
   1.4.5      Locating a User .....................................   10   13
   1.4.6      Changing an Existing Session ........................   13
   1.4.7      Registration Services ...............................   13   14
   1.5        Protocol Properties .................................   13   14
   1.5.1      Minimal State .......................................   13   14
   1.5.2      Transport-Protocol      Lower-Layer-Protocol Neutral .......................... ........................   14
   1.5.3      Text-Based ..........................................   14   15
   2          SIP Uniform Resource Locators .......................   14   15
   3          SIP Message Overview ................................   17
   4          Request .............................................   18   19
   4.1        Request-Line ........................................   18   19
   4.2        Methods .............................................   19   21
   4.2.1       INVITE .............................................   20   21
   4.2.2       ACK ................................................   20   21
   4.2.3       OPTIONS ............................................   20   22
   4.2.4       BYE ................................................   20   22
   4.2.5       CANCEL .............................................   22
   4.2.6       REGISTER ...........................................   21
   4.2.6       UNREGISTER .........................................   21   23
   4.3        Request-URI .........................................   21   25
   4.3.1      SIP Version .........................................   22   25
   4.4        Option Tags .........................................   22   25
   4.4.1      Registering New Option Tags with IANA ...............   22   26
   5          Response ............................................   23   26
   5.1        Status-Line .........................................   23   27
   5.1.1      Status Codes and Reason Phrases .....................   23   27
   6          Header Field Definitions ............................   25   30
   6.1        General Header Fields ...............................   27   31
   6.2        Entity Header Fields ................................   27   31
   6.3        Request Header Fields ...............................   27   31
   6.4        Response Header Fields ..............................   29   32
   6.5        End-to-end and Hop-by-hop Headers ...................   32
   6.6        Header Field Format .................................   29
   6.6        Accept ..............................................   30   32
   6.7        Accept-Language .....................................   30         Accept .............................................   34
   6.8        Allow ...............................................   30         Accept-Encoding ....................................   34
   6.9        Also ................................................   30         Accept-Language ....................................   34
   6.10       Authorization .......................................   31        Allow ..............................................   34
   6.11       Call-Disposition ....................................   31        Authorization ......................................   35
   6.12        Call-ID .............................................   32 ............................................   35
   6.13       Content-Length ......................................   32        Content-Encoding ...................................   36
   6.14       Content-Type ........................................   33        Content-Length .....................................   36
   6.15       Date ................................................   33        Content-Type .......................................   37
   6.16       Expires .............................................   33        CSeq ...............................................   37
   6.17       From ................................................   34        Date ...............................................   38
   6.18       Location ............................................   35        Encryption .........................................   39
   6.19       Organization ........................................   37        ETag ...............................................   40
   6.20       Priority        Expires ............................................   37   41
   6.21       Proxy-Authenticate ..................................   38        From ...............................................   42
   6.22       Proxy-Authorization .................................   38        Hide ...............................................   42
   6.23       Public ..............................................   38        If-Match ...........................................   43
   6.24       Require .............................................   38        If-None-Match ......................................   44
   6.25       Retry-After .........................................   39        Location ...........................................   44
   6.26       CSeq ................................................   39        Max-Forwards .......................................   46
   6.27       Server ..............................................   40        Organization .......................................   47
   6.28       Subject .............................................   40        Priority ...........................................   47
   6.29       Unsupported .........................................   40        Proxy-Authenticate .................................   48
   6.30       Timestamp ...........................................   41        Proxy-Authorization ................................   48
   6.31       To ..................................................   41        Proxy-Require ......................................   48
   6.32       User-Agent ..........................................   41        Require ............................................   49
   6.33       Via .................................................   41        Record-Route .......................................   50
   6.34       Warning .............................................   43        Response-Key .......................................   50
   6.35        Route ..............................................   51
   6.36        Retry-After ........................................   51
   6.37        Server .............................................   52
   6.38        Subject ............................................   52
   6.39        Timestamp ..........................................   52
   6.40        To .................................................   53
   6.41        Unsupported ........................................   53
   6.42        User-Agent .........................................   53
   6.43        Via ................................................   53
   6.43.1     Requests ............................................   54
   6.43.2     Receiver-tagged Via Fields ..........................   54
   6.43.3     Responses ...........................................   55
   6.43.4     Syntax ..............................................   55
   6.44        Warning ............................................   56
   6.45        WWW-Authenticate ....................................   44 ...................................   57
   7          Status Code Definitions .............................   44   58
   7.1        Informational 1xx ...................................   44   58
   7.1.1      100 Trying ..........................................   44   58
   7.1.2      180 Ringing .........................................   44   58
   7.1.3      181 Queued ..........................................   45 Call Is Being Forwarded .........................   58
   7.2        Successful 2xx ......................................   45   59
   7.2.1      200 OK ..............................................   45   59
   7.3        Redirection 3xx .....................................   45   59
   7.3.1      300 Multiple Choices ................................   45   59
   7.3.2      301 Moved Permanently ...............................   46   60
   7.3.3      302 Moved Temporarily ...............................   46   60
   7.3.4      380 Alternative Service .............................   46   60
   7.3.5      381 Ambiguous .......................................   60
   7.4        Request Failure 4xx .................................   46   61
   7.4.1      400 Bad Request .....................................   46   61
   7.4.2      401 Unauthorized ....................................   46   61
   7.4.3      402 Payment Required ................................   46   61
   7.4.4      403 Forbidden .......................................   46   61
   7.4.5      404 Not Found .......................................   46   61
   7.4.6      405 Method Not Allowed ..............................   47   61
   7.4.7      407 Proxy Authentication Required ...................   47   61
   7.4.8      408 Request Timeout .................................   47   62
   7.4.9      412 Precondition Failed .............................   62
   7.4.10     420 Bad Extension ...................................   47
   7.4.10   62
   7.4.11     480 Temporarily Unavailable .........................   47
   7.4.11   62
   7.4.12     481 Invalid Call-ID .................................   47
   7.4.12   62
   7.4.13     482 Loop Detected ...................................   48   62
   7.4.14     483 Too Many Hops ...................................   63
   7.4.15     484 Address Incomplete ..............................   63
   7.5        Server Failure 5xx ..................................   48   63
   7.5.1      500 Server Internal Error ...........................   48   63
   7.5.2      501 Not implemented Implemented .................................   48   63
   7.5.3      502 Bad Gateway .....................................   48   63
   7.5.4      503 Service Unavailable .............................   48   63
   7.5.5      504 Gateway Timeout .................................   48   64
   7.5.6      505 Version Not Supported ...........................   64
   7.6        Global Failures 6xx .................................   49   64
   7.6.1      600 Busy ............................................   49   64
   7.6.2      603 Decline .........................................   49   64
   7.6.3      604 Does not exist anywhere Not Exist Anywhere .........................   49   64
   7.6.4      606 Not Acceptable ..................................   49   64
   8          SIP Message Body ....................................   50   65
   8.1        Body Inclusion ......................................   50   65
   8.2        Message Body Type ...................................   66
   8.3        Message Body Length .................................   50   66
   9          Examples ............................................   51
   9.1        Invitation to Multimedia Conference .................   51
   9.1.1      Request .............................................   51
   9.1.2      Reply ...............................................   52
   9.2        Two-party Call ......................................   53
   9.3        Aborting a Call .....................................   54
   9.3.1      Redirects ...........................................   54
   9.3.2      Alternative Services ................................   55
   9.3.3      Negotiation .........................................   56
   9.4        OPTIONS Request .....................................   57
   10          Compact Form ........................................   57
   11   66
   10         SIP Transport .......................................   58
   11.1       Achieving Reliability For UDP Transport .............   59
   11.1.1   67
   10.1       General Operation ...................................   59
   11.1.2 Remarks .....................................   67
   10.1.1     Requests ............................................   67
   10.1.2     Responses ...........................................   68
   10.2       Unicast UDP .........................................   68
   10.3       Multicast UDP .......................................   69
   10.4        BYE,  CANCEL,  OPTIONS .............................   69
   10.5        REGISTER ...........................................   69
   10.6        ACK ................................................   70
   10.7        INVITE ..............................................   59
   11.2       Connection Management for .............................................   70
   10.8       TCP .......................   60
   12 Connections .....................................   71
   11         Behavior of SIP Servers .............................   63
   12.1   74
   11.1       Redirect Server .....................................   63
   12.2   74
   11.2       User Agent Server ...................................   63
   12.3       Proxies   75
   11.3       Stateless Proxy: Proxy Servers Issuing Single
   Unicast Requests .............   64
   12.4 ...............................................   75
   11.4       Proxy Server Issuing Several  INVITE Requests ...............   64
   13         Third-Party Call Initiation .........................   67
   14         ISDN .......   75
   11.5       Proxy Server Issuing Several  ACK and Intelligent Network Services ...............   68
   15  BYE
   Requests .......................................................   79
   12         Security Considerations .............................   72
   15.1   80
   12.1       Confidentiality .....................................   72
   15.2 and Privacy: Encryption .............   80
   12.1.1     SIP Transactions ....................................   80
   12.2       End-to-End Encryption ...............................   81
   12.2.1     Privacy of SIP Responses ............................   83
   12.2.2     Encryption by Proxies ...............................   83
   12.2.3     Hop-by-Hop Encryption ...............................   83
   12.2.4     Via field encryption ................................   83
   12.3       Message Integrity ...........................................   72
   15.3 and Access Control ......................................   72
   15.4 Control:
   Authentication .................................................   84
   12.3.1     Trusting responses ..................................   86
   12.4       Callee Privacy ......................................   87
   12.5       Known Security Problems .............................   87
   13         SIP Security Using PGP ..............................   88
   13.1       PGP Authentication Scheme ...........................   88
   13.1.1     The  WWW-Authenticate Response Header ...............   88
   13.1.2     The  Authorization Request Header ...................   89
   13.2       PGP Encryption Scheme ...............................   90
   13.3        Response-Key Header Field for PGP ..................   90
   14         Examples ............................................   91
   14.1       Registration ........................................   91
   14.2       Invitation to Multicast Conference ..................   92
   14.2.1     Request .............................................   73   92
   14.2.2     Response ............................................   93
   14.3       Two-party Call ......................................   94
   14.4       Terminating a Call ..................................   95
   14.5       Forking Proxy .......................................   96
   14.6       Redirects ...........................................  100
   14.7       Alternative Services ................................  100
   14.8       Negotiation .........................................  102
   14.9        OPTIONS Request ....................................  102
   A          Minimal Implementation ..............................   73  103
   A.1        Client ..............................................   73  103
   A.2        Server ..............................................   74  104
   A.3        Header Processing ...................................  104
   B          Usage of SDP ........................................  104
   C          Summary of Augmented BNF ............................   74
   C  105
   D          IANA Considerations .................................  107
   E          Open Issues .........................................   76
   D  109
   E.1        H.323 ...............................................  109
   E.2        REGISTER ............................................  109
   E.3        Max-Forwards ........................................  109
   E.4        Cancellation and BYE ................................  110
   E.5        IPv6 URLs ...........................................  110
   F          Acknowledgments .....................................   76
   E  110
   G          Authors' Addresses ..................................   76
   F  110
   H          Bibliography ........................................   77  111