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Internet Engineering Task Force                                MMUSIC WG
Internet Draft                              Handley/Schulzrinne/Schooler
draft-ietf-mmusic-sip-03.txt                           ISI/Columbia U./Caltech
July 31, 1997
Expires: January 20, 1998


                    SIP: Session Initiation Protocol

STATUS OF THIS MEMO

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

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

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

   Distribution of this document is unlimited.

                                 ABSTRACT


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

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




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1 Introduction

1.1 Overview of SIP Functionality

   The Session Initiation Protocol (SIP) is an application-layer
   protocol that can establish and control multimedia sessions or calls.
   These multimedia sessions include multimedia conferences, distance
   learning, Internet telephony and similar applications. SIP can
   initiate both unicast and multicast sessions; the initiator does not
   necessarily have to be a member of the session. 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 [4].)

   SIP transparently supports name mapping and redirection services,
   allowing the implementation of telephony services such as selective
   call forwarding, selective call rejection, conditional and
   unconditional call forwarding, call forwarding busy, call forwarding
   no response. SIP may use multicast to try several possible callee
   locations at the same time.

   SIP supports personal mobility telecommunications intelligent network
   services, this is defined as:  "Personal mobility is the ability of
   end users to originate and receive calls and access subscribed
   telecommunication services on any terminal in any location, and the
   ability of the network to identify end users as they move. Personal
   mobility is based on the use of a unique personal identity (i.e.,
   'personal number')." [1].  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 four facets of establishing multimedia
   communications:

        1.   user location: determination of the end system to be used
             for communication;

        2.   user capabilities: determination of the media and media
             parameters to be used;

        3.   user availability: determination of the willingness of the
             called party to engage in communications;

        4.   call setup ("ringing", establishment of call parameters at
             both called and calling party)
        In particular, SIP can be used to locate a user and determine



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        the appropriate end system, leaving the actual call
        establishment to other protocols such as H.323.

   SIP may also be used to terminate and transfer a call. SIP can also
   initiate multi-party calls using a multipoint control unit (MCU) or
   fully-meshed interconnection instead of multicast.

        These features are for further study.

   SIP is not a conference control protocol, but can be used to
   introduce conference control protocols to a session.

   SIP is designed as part of the overall IETF multimedia data and
   control architecture currently incorporating protocols such as RSVP
   [2] for reserving network resources, RTP [3] for transporting real-
   time data and providing QOS feedback, RTSP [4] for controlling
   delivery of streaming media, SAP [5] for advertising multimedia
   sessions via multicast and SDP [6] for describing multimedia
   sessions, but SIP does not depend for its operation on any of these
   protocols.

1.2 Finding Multimedia Sessions

   There are two basic ways to locate and to participate in a multimedia
   session:

   Advertisement: The session is advertised, potential participants see
        the advertisement, then join the session address to participate.

   Invitation: Users are invited by others to participate in a session,
        which may or may not be advertised.

   Sessions may be advertised using multicast protocols such as SAP [5],
   electronic mail, news groups, web pages or directories (LDAP), among
   others. SIP serves the role of the invitation protocol.

   SIP does not prescribe how a conference is to be managed, e.g.,
   whether it uses a central server to manage conference and participant
   state or distributes state via multicast.

   SIP does not allocate multicast addresses, leaving this functionality
   to protocols such as SAP [5].

   SIP can invite users to conferences 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



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   to the agent performing session invitation.

   SIP offers some of the same functionality as H.323, but may also be
   used in conjunction with it. In this mode, H.323 is used to locate
   the appropriate terminal identified by a H.245 address [TBD: what
   does this look like?]. An H.323-capable terminal then proceeds with a
   normal H.323/H.245 invitation [7].

1.3 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 [8] and
   indicate requirement levels for compliant SIP implementations.

1.4 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) [9].

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

   Final response: A response that terminates a  ->  SIP transaction, as
        opposed to a  ->  provisional response 3xx, 4xx, and 5xx
        responses are final.

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

   Invitation: A request sent to a user (or service) requesting
        participation in a session.

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

   Location server: A program that is contacted by a  ->  client and
        that returns one or more possible locations of the called party
        or service. Location servers may be invoked by SIP redirect and
        proxy servers and may be Co-located with a SIP server.

   Location service: A service used by a  ->  redirect or  ->  proxy
        server to obtain information about a callee's possible location.




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   Provisional response: A response used by the server to indicate
        progress, but that does not terminate a  ->  SIP transaction.
        All 1xx and 6xx responses are provisional. Other responses are
        considered  ->  final.

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

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

   Server: An application program that accepts requests in order to
        service requests and sends back responses to those requests.
        Servers are either proxy, redirect or user agent servers. An
        application program may act as both server and client.

   Session: "A multimedia session is a set of multimedia senders and
        receivers and the data streams flowing from senders to
        receivers. A multimedia conference is an example of a multimedia
        session." [6] For SIP, a session is equivalent to a "call".
        (Note: a session as defined here may comprise one or more RTP
        sessions.)

   (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 6.11).
        There can only be one pending transaction between a server and
        client for each call id.

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

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



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1.5 Protocol Properties

1.5.1 Minimal State

   There is no concept of an ongoing SIP session that lasts for the
   duration of the conference or call. Rather, a single conference
   session or call may involve one or more SIP request-response
   transactions. For example, a conference control protocol may use SIP
   to add or remove a media stream, but again, once the information has
   been successfully conveyed to the participants, SIP is then no longer
   involved.

   At most, a server has to maintain state for a single SIP transaction.
   In some cases, it can process each message without regard to previous
   messages ( stateless server ), as described in Section 12.

1.5.2 Transport-Protocol Neutral

   SIP is able to utilize both UDP and TCP as transport protocols. UDP
   allows the application to more carefully control the timing of
   messages and their retransmission, to perform parallel searches
   without requiring connection state for each outstanding request, and
   to use multicast.  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)
   [4].

   When TCP is used, SIP can use one or more connections to attempt to
   contact a user or to modify parameters of an existing session. The
   concept of a session is not implicitly bound to a TCP connection, so
   the initial SIP request and a subsequent SIP request may use
   different TCP connections or a single persistent connection as
   appropriate.

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

1.5.3 Text-Based

   SIP is text based. This allows easy implementation in languages such
   as Tcl and Perl, allows easy debugging, and most importantly, makes
   SIP flexible and extensible. As SIP is primarily used for session
   initiation, it is believed that the additional overhead of using a
   text-based protocol is not significant.

1.6 SIP Addressing

   SIP uses two kinds of address identifiers, host-specific addresses
   and host-independent addresses form user@host , where user is any



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   alphanumeric identifier and the form of host depends on the address
   type. Note that SIP does not distinguish between the two and can,
   while inviting a user, map repeatedly between the two address types.

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

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



   A user's host-specific 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.

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

   Host-independent 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 [10], SRV [11]
   or A [12] 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 host-independent 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
   host-independent names include

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



   An address can designate an individual (possibly located at one of
   several end systems), the first available person from a group of
   individuals or a whole group. The form of the address, e.g.,
_________________________
  [1] We avoid the term  location-independent  ,  since
the  address  may  indeed refer to a specific location,
e.g., a company department.




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   sales@example.com , is not sufficient, in general, to determine the
   intent of the caller.

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

1.7 Locating a SIP Server

   Call setup may proceed in several phases. A SIP client MUST follow
   the following steps to resolve the user part of a callee address. If
   a client only supports TCP or UDP, but not both, the respective
   address type is omitted.

        1.   If there is a SRV DNS resource record [11] of type sip.udp
             , contact the listed SIP servers in order of preference
             value using UDP as a transport protocol at the port number
             listed in the DNS resource record.

        2.   If there is a SRV DNS resource record [11] of type sip.tcp
             , contact the listed SIP servers in order of preference
             value using TCP as a transport protocol at the port number
             listed in the DNS resource record.

        3.   If there is a DNS MX record [10], contact the hosts listed
             in their order of preference at the default port number
             (TBD).  For each host listed, first try to contact the
             server using UDP, then TCP.

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

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

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



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   beginning of the sequence when the cached address fails.

   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 [13].

1.8 SIP Transactions

   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. If the invitation is SIP request is
   an invitation, it contains a session description, for example written
   in SDP format, that provides the called party with enough information
   to join the session.

   If TCP is used, request and responses within a single SIP transaction
   are carried over the same TCP connection. Thus, the client SHOULD
   maintain the connection until a final response has been received.
   Several SIP requests from the same client to the same server may use
   the same TCP connection or may open a new connection for each
   request. If the client sent the request sends via unicast UDP, the
   response is sent to the source address of the UDP request. If the
   request is sent via multicast UDP, the response is directed to the
   same multicast address and destination port. For UDP, reliability is
   achieved using retransmission (Section 11).


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

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

   The basic message flow is shown in Fig. 1 and Fig. 2, for proxy and
   redirect modes, respectively.



1.9 Locating a User

   A callee may move between a number of different end systems over
   time.  These locations can be dynamically registered with a location
   server, typically for a single administrative domain, or a location



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

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


   Figure 1: Example of SIP proxy server


   server may use other protocols, such as finger [14], rwho,
   multicast-based protocols or operating-system dependent mechanism to
   actively determine the end system where a user is reachable. The
   location services yield a list of a zero or more possible locations,
   possibly even sorted in order of likelihood of success.

   The location server can be part of the SIP server or the SIP server
   may use a different protocol (e.g., finger [14] or LDAP [15]) to map
   addresses. A single user may be registered at different locations,
   either because she is logged in at several hosts simultaneously or
   because the location server has (temporarily) inaccurate information.

   The action taken on receiving a list of locations varies with the
   type of SIP server. A SIP redirect server simply returns the list to
   the client sending the request as  Location headers (Section 6.17). A
   SIP proxy server can sequentially try the addresses until the call is
   successful (2xx response) or the callee has declined the call (40x
   response). Alternatively, the server may issue several requests in
   parallel. A proxy server can only issue more than one sequential or
   parallel connection request if it is the first in the chain of hosts



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

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


   Figure 2: Example of SIP redirect server


   noted in the  Via header to do so. If it is not the first, it must
   issue a redirect response.

   If a proxy server forwards a SIP request, it SHOULD add itself to the
   end of the list of forwarders noted in the  Via (Section 6.31)
   headers. A proxy server also notes whether it is attempting to reach
   several possible locations at once ("connection forking"). The  Via
   trace ensures that replies can take the same path back, thus ensuring
   correct operation through compliant firewalls and loop-free requests.
   On the reply path, each host most remove its Via, so that routing
   internal information is hidden from the callee and outside networks.
   When a multicast request is made, first the host making the request,
   then the multicast address itself are added to the path.




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   As discussed in Section 1.6, a SIP address may designate a group
   rather than an individual. A client indicates using the  Reach
   request header whether it wants to reach the first available
   individual or treat the address as a group, to be invited as a whole.
   The default is to attempt to reach the first available callee.  If
   the address is designated as a group address, a proxy server MUST
   return the list of individuals instead of attempting to connect to
   these.

        Otherwise, the proxy cannot report errors and call status
        appropriately.

2 SIP Uniform Resource Locators

   SIP URLs are used within SIP messages to indicate the originator and
   recipient of a SIP request, and to specify redirection addresses. A
   SIP URL may be embedded in web pages or other hyperlinks to indicate
   that a user or service may be called. Within SIP messages, an email
   address could have been used, but this would have made it more
   difficult to gateway between SIP and other protocols with other
   addressing schemes.

   For greater functionality, because interaction with some resources
   may require message headers or message bodies to be specified as well
   as the SIP address, the sip URL scheme is extended to allow setting
   SIP request-header fields and the SIP  message-body.

   A SIP URL follows the guidelines of RFC 1630 [16,17] and takes the
   following form:



        SIP-URL            =    short-sip-url | full-sip-url
        full-sip-url       =    "sip://" user [ ":" password ] "@" host
                                url-parameters [ headers ]
        short-sip-url      =    user [ ":" password ] "@" host : port
        user               =    ;  defined in RFC 1738 [18]
        host               =    ;  defined in RFC 1738
        port               =    *digit
        url-parameters     =    *( ";" url-parameter)
        url-parameter      =    transport-param |
                                ttl-param | maddr-param
        transport-param    =    "transport=" ( "udp" | "tcp" )
        ttl-param          =    "ttl=" ttl
        ttl                =    1*3DIGIT                                     ; 0 to 255
        maddr-param        =    "maddr=" maddr
        maddr              =    ;  dotted decimal multicast address
        headers            =    "?" header *( "                            " header )



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        header             =    hname "=" hvalue
        hname              =    *urlc
        hvalue             =    *urlc
        urlc               =    ;  defined in [17]


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

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

   Examples of short and full form SIP URLs with identical address 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



   The  password parameter allows to easily specify a call-back address
   on a secure web page, but carries the same security risks as all
   URL-based passwords and should only be used under special
   circumstances where security requirements are low or all transport
   paths are secured.

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

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

   Clearly not all URLs are appropriate to be used in a SIP message as a
   user address. It is unlikely, for example, that HTTP or FTP URLs are
   useful in this context. The correct behavior when an unknown scheme



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   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



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

3 SIP Message Overview

   Since much of the message syntax is identical to HTTP/1.1, rather
   than repeating it here we use [HX.Y] to refer to Section X.Y of the
   current HTTP/1.1 specification [9]. In addition, we describe SIP in
   both prose and an augmented Backus-Naur form (BNF) [H2.1] described
   in detail in [19].

   All SIP messages are text-based and use HTTP/1.1 conventions [H4.1],
   except for the additional ability of SIP to use UDP. When sent over
   TCP or UDP, multiple SIP transactions can be carried in a single TCP
   connection or UDP datagram. UDP datagrams, including all headers,
   should not normally be larger than the path maximum transmission unit
   (MTU) if the MTU is known, or 1500 bytes if the MTU is unknown.


        The 1400 bytes accommodates lower-layer packet headers
        within the "typical" MTU of around 1500 bytes. There are
        few MTU values around 1 kB; the next value is 1006 bytes
        for SLIP and 296 for low-delay PPP [20]. Recent studies
        [21] indicate that an MTU of 1500 bytes is a reasonable
        assumption. Thus, another reasonable value would be a
        message size of 950 bytes, to accommodate packet headers
        within the SLIP MTU without fragmentation.

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


        SIP-message = Request | Response  ; SIP messages



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   Both  Request (section 4) and  Response (section 5) messages use the
   generic message format of RFC 822 [22] for transferring entities (the
   payload of the message). Both types of message consist of a  start-
   line, one or more header fields (also known as "headers"), an empty
   line (i.e., a line with nothing preceding the carriage-return line-
   feed ( CRLF)) indicating the end of the header fields, and an
   optional message-body. To avoid confusion with similar-named headers
   in HTTP, we refer to the header describing the message body as entity
   headers.  These components are described in detail in the upcoming
   sections.



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

        start-line         =    Request-Line | Status-Line




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

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



   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:





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   general-header     =     Call-ID                ; Section 6.11
                      |     Date                   ; Section 6.14
                      |     Expires                ; Section 6.15
                      |     From                   ; Section 6.16
                      |     Sequence               ; Section 6.26
                      |     Via                    ; Section 6.31
   entity-header      =     Content-Length         ; Section 6.12
                      |     Content-Type           ; Section 6.13
   request-header     =     Accept                 ; Section 6.6
                      |     Accept-Language        ; Section 6.7
                      |     Authorization          ; Section 6.9
                      |     Organization           ; Section 6.18
                      |     Priority               ; Section 6.20
                      |     Proxy-Authorization    ; Section 6.22
                      |     Reach                  ; Section 6.24
                      |     Subject                ; Section 6.28
                      |     To                     ; Section 6.29
                      |     User-Agent             ; Section 6.30
   response-header    =     Location               ; Section 6.17
                      |     Proxy-Authenticate     ; Section 6.21
                      |     Public                 ; Section 6.23
                      |     Retry-After            ; Section 6.25
                      |     Server                 ; Section 6.27
                      |     Warning                ; Section 6.32
                      |     WWW-Authenticate       ; Section 6.33


   Table 1: SIP headers

        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



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4.1.1 Methods

   The following methods are defined:



        method    =    "INVITE" | "CONNECTED" | "OPTIONS" | "BYE"
                 |     "REGISTER" | "UNREGISTER"


   INVITE: The user or service is being invited to participate in the
        session. This method MUST be supported by a SIP server.

   CONNECTED: A  CONNECTED request confirms that the client has received
        a successful response to an  INVITE request. See Section 11 for
        details. This method MUST be supported by a SIP server.

   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.

   BYE: The client indicates to the server that it wishes to abort the
        call attempt. The leaving party can use a  Location header field
        to indicate that the recipient of request should contact the
        named address. This implements the "call transfer" telephony
        functionality.  A client SHOULD also use this method to indicate
        to the callee that it wishes to abort an on-going call attempt.

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

   REGISTER: A client uses the  REGISTER method to register the address
        listed in the request line to a SIP server. In the future, the
        server MAY use the source address and port to forward SIP
        requests to.  A server SHOULD silently drop the registration
        after one hour, unless refreshed by the client. A server may set
        or lower or higher refresh interval and indicate the interval
        through the  Expires header (Section 6.15). A single address (if
        host-independent) may be registered from several different
        clients. Support of this method is OPTIONAL.


        Beyond its use as a simple location service, this method is



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        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.

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


        BYE and REGISTER are experimental and need to be discussed.

   Methods that are not supported by a proxy server SHOULD be treated by
   that proxy as if they were an INVITE method, and relayed through
   unchanged or cause a redirection as appropriate.

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

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

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

4.1.3 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".

5 Response

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




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


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

5.1 Status-Line

   The first line of a  Response message is the  Status-Line, consisting
   of the protocol version ((Section 4.1.3) followed by a numeric
   Status-Code and its associated textual phrase, with each element
   separated by SP characters. No  CR or LF is allowed except in the
   final  CRLF sequence.


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

5.1.1 Status Codes and Reason Phrases

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

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

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

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



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   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.


   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

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

   To,  From, and  Call-ID header MUST be present in each request with
   method  INVITE. 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).

   A server MUST understand the  PEP-Require header.

   Other headers may be added as required; a server MAY ignore headers
   that it does not understand. A compact form of these header fields is



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   Status-Code       =    "100"                         ;  Trying
                    |     "180"                         ;  Ringing
                    |     "200"                         ;  OK
                    |     "300"                         ;  Multiple Choices
                    |     "301"                         ;  Moved Permanently
                    |     "302"                         ;  Moved Temporarily
                    |     "303"                         ;  See Other
                    |     "305"                         ;  Use Proxy
                    |     "380"                         ;  Alternative Service
                    |     "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
                    |     "500"                         ;  Internal Server Error
                    |     "501"                         ;  Not Implemented
                    |     "502"                         ;  Bad Gateway
                    |     "503"                         ;  Service Unavailable
                    |     "504"                         ;  Gateway Timeout
                    |     "505"                         ;  SIP Version not supported
                    |     "600"                         ;  Busy
                    |     "603"                         ;  Decline
                    |     "604"                         ;  Does not exist anywhere
                    |     "606"                         ;  Not Acceptable
                    |     extension-code
   extension-code    =    3DIGIT
   Reason-Phrase     =    *<TEXT,  excluding CR, LF>


   Figure 3: Status Codes


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




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6.1 General Header Fields

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

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

6.2 Entity Header Fields

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

6.3 Request Header Fields

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



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   be  response-header fields. Unrecognized header fields are treated as
   entity-header fields.

6.5 Header Field Format

   Header fields ( general-header,  request-header, response-header, and
   entity-header) follow the same generic header format as that given in
   Section 3.1 of RFC 822 [22].

   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]. This request header field is used only with the OPTIONS
   request to indicate what description formats are acceptable.




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   Example:


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



6.7 Accept-Language

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

   Example:


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



6.8 Allow

   See [H14.7].

6.9 Authorization

   See [H14.8].

6.10 Authentication

   Authentication fields provide a digital signature of the remaining
   fields for authentication purposes. They are not yet defined The use
   of authentication headers is optional. If used, authentication
   headers MUST be added to the header after the  Via fields and before
   the rest of the fields.


        HS: Should probably re-use S/MIME here rather than invent
        our own. Maybe better to fold into Authorization field.

6.11 Call-ID

   The  Call-ID uniquely identifies a particular invitation. Note that a
   single multimedia conference may give rise to several calls, e.g., if
   a user invites several different people. Calls to different callee
   MUST always use different  Call-IDs unless they are the result of a



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   proxy server "forking" a single request.

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

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



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


   Example:

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



6.12 Content-Length

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


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

   An example is

     Content-Length: 3495



   Applications SHOULD use this field to indicate the size of the
   message-body to be transferred, regardless of the media type of the
   entity. Any  Content-Length greater than or equal to zero is a valid
   value. If no body is present in a message, then the Content-Length
   header MAY be omitted or set to zero.  Section 8 describes how to
   determine the length of the message body.

6.13 Content-Type

   The  Content-Type entity-header field indicates the media type of the



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   message-body sent to the recipient.


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

   An example of the field is

     Content-Type: application/sdp



6.14 Date

   See [H14.19].


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

6.15 Expires

   The  Expires header field gives the date/time after which the
   registration expires.

   This header field is currently defined only for the  REGISTER and
   INVITE methods.  For  REGISTER, it is a response-header field and
   allows the server to indicate when the client has to re-register. For
   INVITE, it is a request-header with which the callee can limit the
   validity of an invitation. (For example, if a client wants to limit
   how long a search should take at most or when a conference being
   invited to is time-limited. A user interface may take this is as a
   hint to leave the invitation window on the screen even if the user is
   not currently at the workstation.)

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


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

   Two example of its use are

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






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6.16 From

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

   The  From field is a URL and not a simple SIP address (Section 1.6
   address to allow a gateway to relay a call into a SIP request and
   still produce an appropriate  From field.  An example might be a
   telephone call relayed into a SIP request where the from field might
   contain a  phone:// URL. Normally however this field will contain a
   sip:// URL in either the long or short form.

   If a SIP agent or proxy receives a request sourced  From a URL
   indicating a scheme other that SIP that is unknown to it, this MUST
   NOT be treated as an error.


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

   Example:


     From: mjh@isi.edu (Mark Handley)



6.17 Location

   The  Location response header can be used with a 2xx or 3xx response
   codes to indicate a new location to try. It contains a SIP URL giving
   the new location or username to try, or may simply specify addition
   transport parameters. For example, a "301 Moved Permanently" response
   SHOULD contain a  Location field containing the SIP URL giving the
   new location and username to try. However, a "302 Moved Temporarily"
   MAY give simply 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 transport from UDP to TCP or vice
   versa.

   A user agent or redirect server sending a definitive, positive
   response (2xx), SHOULD insert a  Location response header indicating



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   the SIP address under which it is reachable most directly for future
   SIP requests. This may be the address of the server itself or that of
   a proxy (e.g., if the host is behind a firewall).



        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] >
        service-tag          =     "fax" | "IP" | "PSTN" | "ISDN" | "pager" | "voice-mail
                                   | "attendant"
        media-tag            =      < see SDP: "audio" | "video" | ...
        feature-list         =      to be determined



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


   Examples:


     Location: sip://hgs@erlang.cs.columbia.edu ;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



   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



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   is meant for indicating general communication capability, sufficient
   for the caller to choose an appropriate address.

6.18 Organization

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

6.19 PEP

   This corresponds to the  PEP header in the "Protocol Extension
   Protocol" defined in RFC XXXX. The Protocol Extension Protocol (PEP)
   is an extension mechanism designed to accommodate dynamic extension
   of applications such as SIP clients and servers by software
   components.  The  PEP general header declares new headers and whether
   an application must or may understand them. Servers MUST parse this
   field and MUST return "420 Bad Extension" when there is a PEP
   extension of strength "must" (see RFC XXXX) that they do not
   understand.

6.20 Priority

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



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


   Example:


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



6.21 Proxy-Authenticate

   See [H14.33].

6.22 Proxy-Authorization

   See [H14.34].



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6.23 Public

   See [H14.35].

6.24 Reach

   The  Reach request header field allows the client to indicate whether
   it wants to reach the group identified by the user part of the
   address (value "all") or the first available individual (value
   "first"). If not present, a value of "first" is implied. The "do-
   not-forward" request prohibits proxies from forwarding the call to
   another individual (e.g., the call is personal or the caller does not
   want to be shunted to a secretary if the line is busy.)  Section 1.6
   describes the behavior of proxy servers when resolving group aliases.


        Reach = "Reach" ":" 1#( "first" | "all" ) ( "do-not-
        forward" )

   Example:

     Reach: first, do-not-forward




        HS: This header is experimental.

6.25 Retry-After

   The  Retry-After response header field can be used with a "503
   Service Unavailable" response to indicate how long the service is
   expected to be unavailable to the requesting client and with a "404
   Not Found" or "451 Busy" 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.


        Retry-After = "Retry-After" ":" ( HTTP-date | delta-seconds
        )

   Two examples of its use are

     Retry-After: Mon, 21 Jul 1997 18:48:34 GMT
     Retry-After: 120





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   In the latter example, the delay is 2 minutes.

6.26 Sequence

   The  Sequence header field MAY be added by a SIP client making a
   request if it needs to distinguish responses to several consecutive
   requests sent with the same  Call-ID. A  Sequence field contains a
   single decimal sequence number chosen by the requesting client.
   Consecutive different requests made with the same  Call-ID MUST
   contain strictly monotonically increasing sequence numbers although
   the sequence space MAY NOT be contiguous. A server responding to a
   request containing a sequence number MUST echo the sequence number
   back in the response.


        Sequence = "Sequence" ":" 1*DIGIT

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

   Example:


     Sequence: 4711



6.27 Server

   See [H14.39].

6.28 Subject

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


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

   Example:


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






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6.29 To

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


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

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

   Example:

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



6.30 User-Agent

   See [H14.42].

6.31 Via

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

   In the request path, an initiator MUST add its own  Via field to each
   request. This  Via field MUST be the first field in the request. Each
   subsequent client or proxy that sends the message onwards MUST add
   its own additional  Via field, which MUST be added before any
   existing  Via fields. Additionally, if the message goes to a
   multicast address, an extra  Via field is added before all the others
   giving the multicast address and TTL.

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

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

        o If the first  Via field in a reply you are going to send is a
         multicast address, remove that  Via field before sending to the
         multicast address.




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

   When a reply passes through a proxy on the reverse path, that proxies
   Via field MUST be removed from the reply.

   The format for a  Via header is:


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


   The "ttl" parameter is included only if the address is a multicast
   address. The "fanout" parameter indicates that this proxy has
   initiated several connection attempts and that subsequent proxies
   should not do the same.

   Example:


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



6.32 Warning

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



        Warning          =    "Warning" ":" 1#warning-value
        warning-value    =    warn-code SP warn-agent SP warn-text
        warn-code        =    2DIGIT
        warn-agent       =    ( host [ ":" port ] ) | pseudonym
                              ;  the name or pseudonym of the server adding
                              ;  the Warning header, for use in debugging



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        warn-text        =    quoted-string


   A response may carry more than one  Warning header.

   The  warn-text should be in a natural language and character set that
   is most likely to be intelligible to the human user receiving the
   response. This decision may be based on any available knowledge, such
   as the location of the cache or user, the  Accept-Language field in a
   request, the  Content-Language field in a response, etc. The default
   language is English and the default character set is ISO- 8859-1.

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

   When multiple  Warning headers are attached to a response, the user
   agent SHOULD display as many of them as possible, in the order that
   they appear in the response. If it is not possible to display all of
   the warnings, the user agent should follow these heuristics:

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

        o Warnings in the user's preferred character set take priority
         over warnings in other character sets but with identical
         warn-codes and  warn-agents.

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

   Example:


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



6.33 WWW-Authenticate

   See [H14.46].

7 Status Code Definitions

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



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

7.1 Informational 1xx

   Informational responses indicate that the server or proxy contacted
   is performing some further action and does not yet have a definitive
   response. The client SHOULD wait for a further response from the
   server, and the server SHOULD send such a response without further
   prompting. If UDP transport is being used, the client SHOULD
   periodically re-send the request in case the final response is lost.
   Typically a server should send a "1xx" response if it expects to take
   more than one second to obtain a final reply.

7.1.1 100 Trying

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

7.1.2 180 Ringing

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

7.2 Successful 2xx

   The request was successful and MUST terminate a search.

7.2.1 200 OK

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

7.3 Redirection 3xx

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

7.3.1 300 Multiple Choices

   The requested resource corresponds to any one of a set of
   representations, each with its own specific location, and agent-
   driven negotiation (i.e., controlled by the SIP client) is being



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   provided so that the user (or user agent) can select a preferred
   communication end point and redirect its request to that location.

   The response SHOULD include an entity containing a list of resource
   characteristics and location(s) from which the user or user agent can
   choose the one most appropriate. The entity format is specified by
   the media type given in the  Content-Type header field. Depending
   upon the format and the capabilities of the user agent, selection of
   the most appropriate choice may be performed automatically. However,
   this specification does not define any standard for such automatic
   selection.

   The choices SHOULD also be listed as  Location fields (Section 6.17).
   Unlike HTTP, the SIP response may contain several  Location fields.
   User agents MAY use the  Location field value for automatic
   redirection or MAY ask the user to confirm a choice.

7.3.2 301 Moved Permanently

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

7.3.3 302 Moved Temporarily

   The requesting client should retry on the new address(es) given by
   the Location header. A 302 response MUST NOT suggest any of the hosts
   in the  Via path of the request as the user's new location.

7.3.4 380 Alternative Service

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

7.4 Request Failure 4xx

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

7.4.1 400 Bad Request

   The request could not be understood due to malformed syntax.




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7.4.2 401 Unauthorized

   The request requires user authentication.

7.4.3 402 Payment Required

   Reserved for future use.

7.4.4 403 Forbidden

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

7.4.5 404 Not Found

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

7.4.6 405 Method Not Allowed

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

7.4.7 407 Proxy Authentication Required

   This code is similar to 401 (Unauthorized), but indicates that the
   client MUST first authenticate itself with the proxy. The proxy MUST
   return a  Proxy-Authenticate header field (section 6.21) containing a
   challenge applicable to the proxy for the requested resource. The
   client MAY repeat the request with a suitable Proxy-Authorization
   header field (section 6.22). SIP access authentication is explained
   in section [H11].

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

7.4.8 408 Request Timeout

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

7.4.9 420 Bad Extension

   The server did not understand the protocol extension specified with



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   strength "must".

7.4.10 480 Temporarily Unavailable

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

7.5 Server Failure 5xx

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

7.5.1 500 Server Internal Error

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

7.5.2 501 Not implemented

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

7.5.3 502 Bad Gateway

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

7.5.4 503 Service Unavailable

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

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




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7.5.5 504 Gateway Timeout

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

7.6 Global Failures

   6xx responses indicate that a server has definitive information about
   a particular user, not just the particular instance indicated in the
   Request-URI. All further searches for this user are doomed to failure
   and pending searches SHOULD be terminated.

7.6.1 600 Busy

   The callee's end system was contacted successfully but the callee is
   busy and does not wish to take the call at this time. The response
   may indicate a better time to call in the  Retry-After header. If the
   callee does not wish to reveal the reason for declining the call, the
   callee should use status code 680 instead.

7.6.2 603 Decline

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

7.6.3 604 Does not exist anywhere

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

7.6.4 606 Not Acceptable

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

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

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



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

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

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

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

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

8 SIP Message Body

   The session description body gives details of the session the user is
   being invited to join. Its Internet media type MUST be given by the
   Content-type header field, and the body length in bytes MUST be given
   by the  Content-Length header field. If the body has undergone any
   encoding (such as compression) then this MUST be indicated by the
   Content-encoding header field, otherwise Content-encoding MUST be
   omitted.

   If required, the session description can be encrypted using public
   key cryptography, and then can also carry private session keys for
   the session. If this is the case, four random bytes are added to the
   beginning of the session description before encryption and are
   removed after decryption but before parsing.

8.1 Body Inclusion

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

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

8.2 Message Body Length




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

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

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

   The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.

9 Examples

9.1 Invitation

9.1.1 Request

   The example below is a request message en route from initiator to
   invitee:


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






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   The first line above states that this is a SIP version 2.0 request.

   The  Via fields give the hosts along the path from invitation
   initiator (the first element of the list) towards the invitee. In the
   example above, the message was last multicast to the administratively
   scoped group 239.128.16.254 with a ttl of 16 from the host
   131.215.131.131

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

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

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

9.1.2 Reply

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


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



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

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



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   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



   For two-party Internet phone calls, the response must contain a
   description of where to send data to, for example the reply from
   schooler to mjh :


   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
         Content-Length: 102

         v=0
         o=schooler 4858949 4858949 IN IP4 192.1.2.3
         t=0 0
         m=audio 5004 RTP/AVP 0
         c=IN IP4 131.215.131.147



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


   C->S: CONNECTED 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



9.1.3 Aborting a Call

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



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



9.1.4 Redirects

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


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



   In this example, the proxy located at 131.215.131.131 is being
   advised to contact the multicast group 239.128.16.254 with a ttl of
   16 and UDP transport. In normal situations, a server would not
   suggest a redirect to a local multicast group unless, as in the above
   situation, it knows that the previous proxy or client is within the
   scope of the local group. If the request is redirected to a multicast
   group, a proxy server SHOULD query the multicast address itself
   rather than sending the redirect back towards the client as multicast
   may be scoped; this allows a proxy within the appropriate scope
   regions to make the query.

9.1.5 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



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

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



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


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

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



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

9.1.6 Negotiation

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




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

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



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

9.2 OPTIONS Request

   A caller Alice can use an  OPTIONS request to find out the
   capabilities of a potential callee Bob, without "ringing" the
   designated address. In this case, Bob indicates that he can be
   reached at three different addresses, ranging from voice-over-IP to a
   PSTN phone to a pager.


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

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



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         Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
                   duplex=send-only;media=text; q=0.1



   Alternatively, Bob could have returned a description of

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

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

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



10 Compact Form

   When SIP is carried over UDP with authentication and a complex
   session description, it may be possible that the size of a request or
   reply is larger than the MTU. To reduce this problem, a more compact
   form of SIP is also defined by using alternative names for common
   header fields.  These short forms are NOT abbreviations, they are
   field names. No other abbreviations are allowed.


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


   Thus the header in section 9.1 could also be written:





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

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



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

11 SIP Transport

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

11.1 Achieving Reliability For UDP Transport

11.1.1 General Operation

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

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


        HS: A transaction or request ID would remove this
        limitation.



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

   After the server sends a final response, it cannot be sure the client
   has received the response, and thus SHOULD cache the results for at
   least 30 seconds to avoid having to, for example, contact the user or
   user location server again upon receiving a retransmission.

11.1.2 INVITE

   Special considerations apply for the  INVITE method.

        1.   After receiving an invitation, considerable time may elapse
             before the server can determine the outcome. For example,
             the called party may be "rung" or extensive searches may be
             performed, so delays can reach several tens of seconds.

        2.   It is possible that the invitation request reaches the
             callee and the callee is willing to take the call, but that
             the final response (200 OK, in this case) is lost on the
             way to the caller. If the session still exists but the
             initiator gives up on including the user, the contacted
             user has sufficient information to be able to join the
             session. However, if the session no longer exists because
             the invitation initiator "hung up" before the reply arrived
             and the session was to be two-way, the conferencing system
             should be prepared to deal with this circumstance.

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

        4.   The client has to be able to terminate an on-going request,
             e.g., because it is no longer willing to wait for the
             connection or search to succeed. One cannot rely on the
             absence of request retransmission, since the server would
             have to continue honoring the request for several request
             retransmission periods, that is, possible tens of seconds
             if only one or two packets can be lost.

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



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   ringing the user.

   The server retransmits the final response at intervals of T3 (default
   value of T3 = 2 seconds) until it receives a  CONNECTED request for
   the same  Call-ID or until it has retransmitted the final response 10
   times. The  CONNECTED request is acknowledged only once. If the
   request is syntactically valid and the  Request-URI matches that in
   the  INVITED request with the same  Call-ID, the server answers with
   status code 200, otherwise with status code 400.

   Fig. 4 and 5 show the client and server state diagram for
   invitations.



11.2 Connection Management for TCP

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

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

   The server SHOULD NOT close the TCP connection until it has sent its
   final response, at which point it MAY close the TCP connection if it
   wishes to. However, normally it is the client's responsibility to
   close the connection.

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

12 Behavior of SIP Servers

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

   A proxy server always inserts a  Via header field containing their
   own address into requests it issues that are caused by an incoming
   request.

   We define an "A--B proxy" as a proxy that receives SIP requests over
   transport protocol A and issues requests, acting as a SIP client,



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                           +===========+
                           |  Initial  |
                           +===========+
                                 |
                                 |
                                 |    -
                                 |  ------
                                 |  INVITE
                     +------v    v
                    T1     +-----------+
                  ------   |  Calling  |-------------------+
                  INVITE   +-----------+                   |
                     +------| |  |                         |
             +----------------+  |                         |
             |                   |                         |
             |                   |                         |
             |                   |                         |
             |                   |                         |
             |       +------v    v    v-----|              |
             |      T2     +-----------+   1xx             |
             |    ------   |  Ringing  |   ---             |
             |    INVITE   +-----------+    -              |
             |       +------|    |  | |-----+              |
             |                   |  +--------------+       |
             |     2xx           |                 | >=300 |
             |  ---------        |    2xx          | ----- |
             |  CONNECTED        | ---------       |   -   |
             |                   | CONNECTED       |       |
             +----------------+  |                 |       |
                     +------v |  v                 v       v
                    2xx    +-----------+         +-----------+
                 --------- | Connected |         |  Failure  |
                 CONNECTED +-----------+         +-----------+
                     +------|

              event
             -------
             message


   Figure 4: State transition diagram of client for  INVITE method


   using transport protocol B. If not stated explicitly, rules apply to
   any combination of transport protocols. For conciseness, we only
   describe behavior with UDP and TCP, but the same rules apply for any
   unreliable datagram or reliable protocol, respectively.



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                            +===========+
              +------------>|  Initial  |<-------------+
              |             +===========+              |
              |                   |                    |
              |   failure         |                    |
              | -----------       |  INVITE            |
              | 3xx,4xx,5xx       |  ------            |
              |                   |   1xx              |
              |       +------v    v                    |
              |    INVITE   +-----------+              |
              |    ------   | Searching |              |
              |      1xx    +-----------+              |
              |       +------| |  |  +---------------->+
              |                |  |                    |
              |                |  |  callee picks up   |
              +----------------+  |  ---------------   |
                                  |       200          |
                                  |                    | BYE
                      +------v    v    v-----|         | ---
                   INVITE   +-----------+   T3         | 200
                   ------   | Answered  |   ---        |
                     1xx    +-----------+   200        |
                      +------|    |  | |-----+         |
                                  |  +---------------->+
                                  |                    |
                                  |  CONNECTED         |
                                  |  ---------         |
                                  |     200            |
                                  |                    |
                      +------v    v                    |
                  CONNECTED +-----------+              |
                  --------- | Connected |              |
                     200    +-----------+              |
                      +------|       |                 |
                                     +-----------------+

               event
              -------
              message


   Figure 5: State transition diagram of server for  INVITE method


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




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12.1 Redirect Server

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

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

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

   The redirect server maintains transaction state for the whole SIP
   transaction. Servers in user agents are redirect servers.

12.2 Proxies Issuing Single Unicast Requests

   Proxies in this category issue at most a single unicast request for
   each incoming SIP request, that is, they do not "fork" requests.
   Servers may choose to always operate in the mode described in Section
   12.3.

12.2.1 UDP--UDP Proxy Server

   The UDP--UDP server can forward the request and any responses. It
   does not have to maintain any state for the SIP transaction. UDP
   reliability is assured by the next redirect server in the server
   chain.

12.2.2 UDP--TCP Proxy Server

   A proxy server issuing a single request over TCP maintains state for
   the whole SIP transaction indexed by the  Call-ID.

   If it receives a UDP retransmission of the same request for an
   existing session, it retransmits the last response received from the
   TCP side.  Any changes in the message body compared to the last
   request for the Call-ID are silently ignored. (Otherwise, the proxy
   would have to remember and compare the message body; this also
   violates the notion of a SIP transaction. TBD) The server SHOULD
   cache the final response for a particular  Call-ID after the SIP
   transaction on the TCP side has completed.

   After the cache entry has been expired, the server cannot tell
   whether an incoming request is actually a retransmission of an older



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   request, where the TCP side has terminated. It will treat it as a new
   request.

12.3 Proxy Server Issuing Several Requests

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

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

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


     int N = 0;            /* number of connection attempts */
     address_t address[];  /* list of addresses */
     location[];           /* list of locations */
     int heard = 0;        /* number of sites heard from */
     int class;            /* class of status code */
     int best = 1000;      /* best response so far */
     int timeleft = 120;   /* sample timeout value */
     int loc = 0;          /* number of locations */
     struct {              /* response */
       int status;         /* response status */
       char *location;     /* redirect locations */
       address_t a;        /* address of respondent */



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     } r;
     int i;

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

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

       if (class >= 2) {
         heard++;
         if (tcp) request_close(a);
       }

       if (class == 2) {
         best = r.status;
         break;
       }
       else if (class == 3) {
             /* A server may optionally recurse.  The server MUST check whether
              * it has tried this location before and whether the location is
              * part of the Via path of the incoming request.  This check is
              * omitted here for brevity. Multicast locations MUST NOT be
          * returned to the client if the server is not recursing.
          */
         if (recurse) {
           multicast = 0;
           N++;
           request(r.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) {



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         best = r.status;
         break;
       }
     }
     /* We haven't heard anything useful from anybody. */
     if (best == 1000) {
       best = 404;
     }
     if (best/100 != 3) locs = 0;
     response(best, locs, locations);



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

13 Security Considerations

13.1 Confidentiality

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

13.2 Integrity

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

13.3 Access Control

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

13.4 Privacy

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



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A Summary of Augmented BNF

   In this specification we use the Augmented Backus-Naur Form notation
   described in [19]. For quick reference, the following is a brief
   summary of the main features of this ABNF.

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

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

   *term
        zero of more instances of  term;

   3*term
        three or more instances of  term;

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

   [ term ]
        term is optional;

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

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

   #term
        a comma separated list of  term;

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

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


   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.




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        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

   Since version -01, the following things have changed:

        o Added personal note to "Searching" section indicating that 6xx
         codes may not be necessary. Added figures.

        o Initial author's note removed; dated.

        o Introduction rewritten to give quick, concise overview as to
         what SIP does.

        o Conference control (tight vs. loose) seems less and less
         appropriate. All share some state such as notions of
         membership; some (ITU versions) tend to keep it in a central
         server, others distribute it. Some state is synchronized at
         larger timescales than other. (After all, even a server won't
         know if a participant disconnects from the network until TCP
         keep-alive, if any, kicks in.)

        o Added list of related protocols to emphasize that this is part
         of a whole architecture.

        o Terminology: user always reminds me of controlled substances;
         thus, this term is avoided where better terminology exists.
         Since this protocol sits at the boundary between traditional
         Internet and telephony services, some of the terminology
         familiar in that realm is introduced.

        o Terminology: user location server replaced by redirect server,
         since a proxy server may also invoke user location. Also, the
         actual user location server (e.g., an LDAP, ULS or similar
         directory) may be invoked using protocols other than SIP.

        o Rearranged ordering of address resolution to correspond to
         host requirements for MX and suggestions in DNS SRV RFC. Adding
         note about caching and socket implementation. Added note about
         using SMTP EXPN to get an alternate address.



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        o Defined SIP transaction, provisional and final responses.

        o Assigned values to timeout parameters; otherwise, there will
         be unnecessary retransmissions between different
         implementations.

        o Retransmission was greatly simplified; there does not seem to
         be a need for all the rules governing transitions between TCP
         and UDP domains. A proxy should look just like a server to one
         side and like a client to the other. Proxies need to maintain
         transaction state in any event since they need to remember
         where they forwarded the last SIP request to ( Confirm wouldn't
         work otherwise, for example.).  Invoking a location service may
         yield inconsistent results, introduces additional failure modes
         (what if the location server is temporarily unavailable?),
         increases delay and processing overhead. UDP--UDP proxies can
         still be built without state; they just forward packets and
         responses. Proxies with TCP on one and UDP on the other side
         will have to act like a normal UDP server and issue 100
         responses.

        o Removed redundancies and contradictions from request and
         response definitions (space vs. SP, duplicate CRLF definition,
         recursive request-header, ...).

        o Added the experimental methods  CONNECTED,  REGISTER,
         UNREGISTER and  BYE.

        o Re-engineered the invitation reliability mechanism to use a
         separate confirmation message.

        o Tentative increase of MTU to 1500 bytes, as per discussion in
         Stevens.

        o Added  Reach,  Organization,  Subject, Priority,
         Authorization,  WWW-Authentication headers for improved call
         handling. WWW "basic" authentication isn't great, but it is
         widely deployed and probably sufficient for giving out
         "private" telephone numbers, particularly those where the
         callee incurs a charge.  (I want to be able to give somebody a
         password to call my 800 number via an Internet gateway;
         authenticating who that person is requires that I modify a
         script on my server to add another distinguished name to the
         list of allowable callees.)

        o Renamed  Reason to  Warning (to align with HTTP) header since
         the response line already offers a failure reason.
         Unfortunately, listing several failures is not all that helpful



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         since the calling party cannot determine which of the media
         within the description causes the difficulty or whether it was
         the set of media as a whole, but it may give the user agent
         some indication as to what's going on.

        o SEP and CRLF in headers removed, since this is always implied
         between items. Missing ":" added. CRLF was already in the
         message definition. Also, unlike RFC 822 and HTTP, the
         definition did not allow spaces between the field name and the
         colon.

        o Added (reluctantly) password to URL. It's no worse than ftp
         and needed to easily call from a secure web page, without
         having to type in a password manually.

        o Added port to SIP URL to specify non-standard port.

        o CAPABILITIES to OPTIONS for closer alignment with HTTP and
         RTSP;

        o Path to Via for closer alignment with HTTP and RTSP;

        o Content type meta changed to application, since "meta" doesn't
         exist as a top-level Internet media type.

        o Formatting closer to HTTP and RTSP.

        o Explain relationship to H.323.

B Open Issues

   RELIABLE: How to provide reliability?

   BYE: Use of BYE method?

   REGISTER: Use of REGISTER method?

   H.323: Interaction with H.323 and H.245.

   TRANSACTION: Should we have a transaction id in addition to a call
        ID? Call-IDs are for the end system, but a transaction ID is for
        a single SIP exchange. This is useful for Internet telephony,
        where a single call may trigger several transactions.

C Acknowledgments

   We wish to thank the members of the IETF MMUSIC WG for their comments
   and suggestions. This work is based, inter alia, on [23,24].



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   Parameters of the terminal negotiation mechanism were influenced by
   Scott Petrack's CMA design.

D 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

E Bibliography

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

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

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

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

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



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

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

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

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

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

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

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

   [13] R. Braden, "Requirements for internet hosts - application and
   support," STD 3, RFC 1123, Internet Engineering Task Force, Oct.
   1989.

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

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

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

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

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




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

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

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

   [22] D. Crocker, "Standard for the format of ARPA internet text
   messages," STD 11, RFC 822, Internet Engineering Task Force, Aug.
   1982.

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

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




                           Table of Contents



   1          Introduction ........................................    2
   1.1        Overview of SIP Functionality .......................    2
   1.2        Finding Multimedia Sessions .........................    3
   1.3        Terminology .........................................    4
   1.4        Definitions .........................................    4
   1.5        Protocol Properties .................................    6
   1.5.1      Minimal State .......................................    6
   1.5.2      Transport-Protocol Neutral ..........................    6
   1.5.3      Text-Based ..........................................    6
   1.6        SIP Addressing ......................................    6
   1.7        Locating a SIP Server ...............................    8
   1.8        SIP Transactions ....................................    9
   1.9        Locating a User .....................................    9
   2          SIP Uniform Resource Locators .......................   12
   3          SIP Message Overview ................................   14
   4          Request .............................................   15
   4.1        Request-Line ........................................   16
   4.1.1      Methods .............................................   17



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   4.1.2      Request-URI .........................................   18
   4.1.3      SIP Version .........................................   18
   5          Response ............................................   18
   5.1        Status-Line .........................................   19
   5.1.1      Status Codes and Reason Phrases .....................   19
   6          Header Field Definitions ............................   20
   6.1        General Header Fields ...............................   22
   6.2        Entity Header Fields ................................   22
   6.3        Request Header Fields ...............................   22
   6.4        Response Header Fields ..............................   22
   6.5        Header Field Format .................................   23
   6.6        Accept ..............................................   23
   6.7        Accept-Language .....................................   24
   6.8        Allow ...............................................   24
   6.9        Authorization .......................................   24
   6.10       Authentication ......................................   24
   6.11       Call-ID .............................................   24
   6.12       Content-Length ......................................   25
   6.13       Content-Type ........................................   25
   6.14       Date ................................................   26
   6.15       Expires .............................................   26
   6.16       From ................................................   27
   6.17       Location ............................................   27
   6.18       Organization ........................................   29
   6.19       PEP .................................................   29
   6.20       Priority ............................................   29
   6.21       Proxy-Authenticate ..................................   29
   6.22       Proxy-Authorization .................................   29
   6.23       Public ..............................................   30
   6.24       Reach ...............................................   30
   6.25       Retry-After .........................................   30
   6.26       Sequence ............................................   31
   6.27       Server ..............................................   31
   6.28       Subject .............................................   31
   6.29       To ..................................................   32
   6.30       User-Agent ..........................................   32
   6.31       Via .................................................   32
   6.32       Warning .............................................   33
   6.33       WWW-Authenticate ....................................   34
   7          Status Code Definitions .............................   34
   7.1        Informational 1xx ...................................   35
   7.1.1      100 Trying ..........................................   35
   7.1.2      180 Ringing .........................................   35
   7.2        Successful 2xx ......................................   35
   7.2.1      200 OK ..............................................   35
   7.3        Redirection 3xx .....................................   35
   7.3.1      300 Multiple Choices ................................   35
   7.3.2      301 Moved Permanently ...............................   36



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   7.3.3      302 Moved Temporarily ...............................   36
   7.3.4      380 Alternative Service .............................   36
   7.4        Request Failure 4xx .................................   36
   7.4.1      400 Bad Request .....................................   36
   7.4.2      401 Unauthorized ....................................   37
   7.4.3      402 Payment Required ................................   37
   7.4.4      403 Forbidden .......................................   37
   7.4.5      404 Not Found .......................................   37
   7.4.6      405 Method Not Allowed ..............................   37
   7.4.7      407 Proxy Authentication Required ...................   37
   7.4.8      408 Request Timeout .................................   37
   7.4.9      420 Bad Extension ...................................   37
   7.4.10     480 Temporarily Unavailable .........................   38
   7.5        Server Failure 5xx ..................................   38
   7.5.1      500 Server Internal Error ...........................   38
   7.5.2      501 Not implemented .................................   38
   7.5.3      502 Bad Gateway .....................................   38
   7.5.4      503 Service Unavailable .............................   38
   7.5.5      504 Gateway Timeout .................................   39
   7.6        Global Failures .....................................   39
   7.6.1      600 Busy ............................................   39
   7.6.2      603 Decline .........................................   39
   7.6.3      604 Does not exist anywhere .........................   39
   7.6.4      606 Not Acceptable ..................................   39
   8          SIP Message Body ....................................   40
   8.1        Body Inclusion ......................................   40
   8.2        Message Body Length .................................   40
   9          Examples ............................................   41
   9.1        Invitation ..........................................   41
   9.1.1      Request .............................................   41
   9.1.2      Reply ...............................................   42
   9.1.3      Aborting a Call .....................................   43
   9.1.4      Redirects ...........................................   44
   9.1.5      Alternative Services ................................   44
   9.1.6      Negotiation .........................................   45
   9.2        OPTIONS Request .....................................   46
   10         Compact Form ........................................   47
   11         SIP Transport .......................................   48
   11.1       Achieving Reliability For UDP Transport .............   48
   11.1.1     General Operation ...................................   48
   11.1.2     INVITE ..............................................   49
   11.2       Connection Management for TCP .......................   50
   12         Behavior of SIP Servers .............................   50
   12.1       Redirect Server .....................................   53
   12.2       Proxies Issuing Single Unicast Requests .............   53
   12.2.1     UDP--UDP Proxy Server ...............................   53
   12.2.2     UDP--TCP Proxy Server ...............................   53
   12.3       Proxy Server Issuing Several Requests ...............   54



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   13         Security Considerations .............................   56
   13.1       Confidentiality .....................................   56
   13.2       Integrity ...........................................   56
   13.3       Access Control ......................................   56
   13.4       Privacy .............................................   56
   A          Summary of Augmented BNF ............................   57
   B          Open Issues .........................................   60
   C          Acknowledgments .....................................   60
   D          Authors' Addresses ..................................   61
   E          Bibliography ........................................   61









































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