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Versions: (RFC 2543) 00 01 02 03 04 05 06 07 08 RFC 3261

Internet Engineering Task Force                                   SIP WG
Internet Draft                    Handley/Schulzrinne/Schooler/Rosenberg
draft-ietf-sip-rfc2543bis-02.txt   ACIRI/Columbia U./Caltech/dynamicsoft
November 24, 2000
Expires: April 2001


                    SIP: Session Initiation Protocol

STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   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".

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.

Abstract

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

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





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1 Introduction

1.1 Overview of SIP Functionality

   The Session Initiation Protocol (SIP) is an application-layer control
   protocol that can establish, modify and terminate multimedia sessions
   (conferences) or Internet telephony calls. SIP can invite
   participants to unicast and multicast sessions; the initiator does
   not necessarily have to be a member of the session to which it is
   inviting. Media and participants can be added to an existing session.

   SIP transparently supports name mapping and redirection services,
   allowing the implementation of ISDN and Intelligent Network telephony
   subscriber services. These facilities also enable personal mobility.
   In the parlance of 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 subnet to another.

   SIP supports five facets of establishing and terminating multimedia
   communications:

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

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

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

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

        Call handling: including transfer and termination of calls.

   SIP is designed as part of the overall IETF multimedia data and
   control architecture currently incorporating protocols such as RSVP
   (RFC 2205 [2]) for reserving network resources, the real-time
   transport protocol (RTP) (RFC 1889 [3]) for transporting real-time
   data and providing QOS feedback, the real-time streaming protocol
   (RTSP) (RFC 2326 [4]) for controlling delivery of streaming media,
   the session announcement protocol (SAP) [5] for advertising



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   multimedia sessions via multicast and the session description
   protocol (SDP) (RFC 2327 [6]) for describing multimedia sessions.
   However, the functionality and operation of SIP does not depend on
   any of these protocols.

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

1.2 Terminology

   In this document, the key words "MUST", "MUSTNOT", "REQUIRED",
   "SHALL", "SHALLNOT", "SHOULD", "SHOULDNOT", "RECOMMENDED", "MAY", and
   "OPTIONAL" are to be interpreted as described in RFC 2119 [7] and
   indicate requirement levels for compliant SIP implementations.

1.3 Overview of SIP Operation

   This section explains the basic protocol functionality and operation.
   Terms are defined more precisely in Section 1.4. In SIP, protocol
   participants are identified by SIP URLs, described in Section 1.4.1.
   SIP is a request-response protocol, with requests sent by clients and
   received by servers. A single implementation typically combines both
   client and server functionality.  SIP requests can be sent using any
   reliable or unreliable protocol, including UDP, SCTP and TCP.
   Protocol operation is largely independent of the lower-layer
   transport protocol.

   This specification defines six SIP request methods: INVITE (Section
   4.2.1) initiates sessions, ACK (Section 4.2.2) confirms session
   establishment, OPTIONS (Section 4.2.3) requests information about
   capabilities, BYE (Section 4.2.4) terminates a sessions, CANCEL
   (Section 4.2.5) cancels a pending session and REGISTER (Section
   4.2.6) allows a client to bind a permanent SIP URL to a temporary SIP
   URL reflecting the current network location.

   SIP requests and responses consists of a request (or status) line, a
   number of header lines and a message body (Section 3).

   SIP requests can be sent directly from a user agent client to a user
   agent server, or they can traverse one or more proxy servers along
   the way. Often, proxy servers are associated with DNS domains,
   similar to SMTP MTAs.

   User agents send requests either directly to the address indicated in
   the SIP URI or to a designated proxy ("outbound proxy"), independent



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   of the destination address. The current destination address is
   carried in the Request-URI. Each proxy can forward the request based
   on local policy and information contained in the SIP request. The
   proxy MAY rewrite the request URI. A proxy MAY also forward the
   request to another designated proxy regardless of the request URI.
   For example, a departmental proxy could forward all authorized
   requests to a corporate-wide proxy which then forwards it to the
   proxy operated by the Internet service provider, which finally routes
   the request based on the request URI.

   For example, if the user agent wants to contact the user
   sip:alice@example.com, it sends the request to the server handling
   the example.com domain (Section 1.4.2). If that host acts as a proxy
   server, it looks up whether it has a mapping from alice@example.com
   to another address. If so, it substitutes that address, say
   alice@sales.example.com, into the Request-URI and then sends the
   request to the server for the sales.example.com domain. Any server
   can also return a response indicating a different destination to be
   tried by the upstream client or indicating that the request cannot be
   forwarded.

   Typically, only the first request within a call traverses all
   proxies, while subsequent requests are exchanged directly between
   user agents.  However, a proxy can indicate that it wants to remain
   in the request path via a Record-Route (Section 6.35) header field.

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) (RFC 2616 [8]). The terms and generic
   syntax of URI and URL are defined in RFC 2396 [9]. The following
   terms have special significance for SIP.

        Call: A call consists of all participants in a conference
             invited by a common source. A SIP call is identified by a
             globally unique call-id (Section 6.13). Thus, if a user is,
             for example, invited to the same multicast session by
             several people, each of these invitations will be a unique
             call. A point-to-point Internet telephony conversation maps
             into a single SIP call. In a multiparty conference unit
             (MCU) based call-in conference, each participant uses a
             separate call to invite himself to the MCU.

        Call leg: A call leg is identified by the combination of the
             Call-ID header field and the addr-spec and tag of the To
             and From header fields. SIP URIs are compared according to



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             Section 2.1, non-SIP URIs according to Section 2.2. Within
             the same Call-ID, requests with From A and To value B
             belong to the same call leg as the requests in the opposite
             direction, i.e., From B and To A.

        Client: An application program that sends SIP requests. Clients
             may or may not interact directly with a human user. User
             agents and proxies contain clients (and servers).

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

        Downstream: Requests sent in the direction from the caller to
             the callee (i.e., user agent client to user agent server).

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

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

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

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

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

        Location server: See location service.

        Location service: A location service is used by a SIP redirect
             or proxy server to obtain information about a callee's
             possible location(s). Examples of sources of location



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             information include SIP registrars, databases or mobility
             registration protocols. Location services are offered by
             location servers. Location servers MAY be part of a SIP
             server, but the manner in which a SIP server requests
             location services is beyond the scope of this document.

        Outbound proxy: A proxy that is located near the originator of
             requests. It receives all outgoing requests from a
             particular UAC, including those requests whose Request-URLs
             identify a host other than the outbound proxy. The outbound
             proxy sends these requests, after any local processing, to
             the address indicated in the request-URI. (All other proxy
             servers are simply referred as proxies, not inbound
             proxies.)

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

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

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


             Proxy servers are, for example, used to route
             requests, enforce policies, control firewalls.

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

        Registrar: A registrar is a server that accepts REGISTER
             requests. A registrar is typically co-located with a proxy
             or redirect server and MAY make its information available



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             through the location server.

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

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

        Session: From the SDP specification: "A multimedia session is a
             set of multimedia senders and receivers and the data
             streams flowing from senders to receivers. A multimedia
             conference is an example of a multimedia session." (RFC
             2327 [6]) (A session as defined for SDP can comprise one or
             more RTP sessions.) As defined, a callee can be invited
             several times, by different calls, to the same session. If
             SDP is used, a session is defined by the concatenation of
             the user name , session id , network type , address type
             and address elements in the origin field.

        (SIP) transaction: A SIP transaction occurs between a client and
             a server and comprises all messages from the first request
             sent from the client to the server up to a final (non-1xx)
             response sent from the server to the client. A transaction
             is identified by the CSeq sequence number (Section 6.21)
             within a single call leg.  The ACK request has the same
             CSeq number as the corresponding INVITE request, but
             comprises a transaction of its own.

        Stateless Proxy: A logical entity that does not maintain state
             for a SIP transaction. A stateless proxy forwards every
             request it receives downstream and every response it
             receives upstream.

        Stateful Proxy: A logical entity that maintains state
             information at least for the duration of a SIP transaction.
             The behavior of a stateful proxy is further defined in
             Section 12.3.

        Upstream: Responses sent in the direction from the user agent
             server to the user agent client.

        URL-encoded: A character string encoded according to RFC 1738,
             Section 2.2 [10].

        User agent client (UAC): A user agent client is a client



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             application that initiates a SIP request.

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

        User agent (UA): An application which can act both as a user
             agent client and user agent server.

   An application program MAY be capable of acting both as a client and
   a server. For example, a typical multimedia conference control
   application would act as a user agent client to initiate calls or to
   invite others to conferences and as a user agent server to accept
   invitations. The role of UAC and UAS as well as proxy and redirect
   servers are defined on a request-by-request basis. For example, the
   user agent initiating a call acts as a UAC when sending the initial
   INVITE request and as a UAS when receiving a BYE request from the
   callee. Similarly, the same software can act as a proxy server for
   one request and as a redirect server for the next request.

   Proxy, redirect, location and registrar servers defined above are
   logical entities; implementations MAY combine them into a single
   application program. The properties of the different SIP server types
   are summarized in Table 1.


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


   Table 1: Properties of the different SIP server types


1.4.1 SIP Addressing

   The "objects" addressed by SIP are users at hosts, identified by a
   SIP URL. The SIP URL takes a form similar to a mailto or telnet URL,
   i.e., user@host.  The user part is a user name or a telephone number.


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   The host part is either a domain name or a numeric network address.
   See section 2 for a detailed discussion of SIP URL's.

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

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

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

1.4.2 Locating a SIP Server

   When a client wishes to send a request, the client either sends it to
   a locally configured SIP proxy server, the so-called outbound proxy ,
   independent of the Request-URI, or sends it to the IP address and
   port corresponding to the Request-URI. The outbound proxy can be
   configured by any mechanism, including DHCP [11].

   If the Request-URI is used, the client needs to determine the
   protocol, port and IP address of a server to which to send the
   request. A client SHOULD follow the steps below to obtain this
   information.

   At each step, unless stated otherwise, the client SHOULD try to
   contact a server at the port number listed in the Request-URI.  If no
   port number is present in the Request-URI, the client uses port 5060.
   If the Request-URI specifies a protocol, the client contacts the
   server using that protocol. If no protocol is specified, the client
   tries UDP (if UDP is supported). If the attempt fails, or if the
   client doesn't support UDP but supports other protocols, it tries
   those protocols in some unspecified order.

   A client SHOULD be able to interpret explicit network notifications
   (such as ICMP messages) which indicate that a server is not
   reachable, rather than relying solely on timeouts. (For socket-based



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   programs:  For TCP, connect() returns ECONNREFUSED if the client
   could not connect to a server at that address. For UDP, the socket
   needs to be bound to the destination address using connect() rather
   than sendto() or similar so that a second write() or send() fails
   with ECONNREFUSED if there is no server listening) If the client
   finds the server is not reachable at a particular address, it SHOULD
   behave as if it had received a 400-class error response to that
   request.

   The client tries to find one or more addresses for the SIP server by
   querying DNS. If a step elicits no addresses, the client continues to
   the next step. However if a step elicits one or more addresses, but
   no SIP server at any of those addresses responds, then the client
   concludes the server is down and does not continue on to the next
   step.

   The service identifier for DNS SRV records [12] is "_sip".

        1.   If the maddr parameter exists, it becomes the destination
             address used below; if not, the host element in the
             Request-URI is the destination address.

        2.   If the destination address is an IP address, the client
             contacts the server at the given address and the port
             number specified in the Request-URI or the default port and
             ignores the remaining steps.

        3.   The Request-URI is examined. If it contains no port number
             or port 5060, the transport parameter is inspected:

             1.   If the Request-URI does not specify a transport
                  protocol, DNS SRV records are retrieved according to
                  RFC 2782 [12]. The results of the query or queries are
                  merged and ordered based on priority, keeping only
                  records with transport protocols that the client
                  supports. Then, the searching technique outlined in
                  RFC 2782 [12] is used to select servers in order.
                  Server selection across requests is independent of
                  previous choices, except as noted below for stateless
                  proxies. The client attempts to contact each server in
                  the order listed, at the port number specified in the
                  SRV record. If none of the servers can be contacted,
                  the client gives up. If there are no SRV records (with
                  any transport protocol), DNS address records are used,
                  as described below.

                  If the Request-URI specifies a transport protocol and
                  the transport protocol is supported by the client, the



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                  procedure in the paragraph above is used, limited to
                  DNS resource records with the transport protocol
                  specified in the Request-URI.

                  If the Request-URI specifies a transport protocol that
                  is not supported by the client, the client gives up.

             If the Request-URI contains a port number other than 5060
             or if there are no SRV records, the client queries the DNS
             server for address records for the destination address.
             Address records include A RR's, AAAA RR's, or other similar
             records, chosen according to the client's network protocol
             capabilities. If the DNS server returns no address records,
             the client gives up.

   Within a transaction, a stateless proxy MUST always select the same
   destination within the set of hosts with the same priority. This can
   be accomplished, for example, by using the modulo N of a hash of the
   Call-ID value or some other combination of transaction-identifying
   headers as the uniform random number described in the weighting
   algorithm of RFC 2782. Here, N is the sum of weights within the
   priority class.

   There are no mandatory rules on how to select a host name for a SIP
   server. Users are encouraged to name their SIP servers using the
   sip.domainname (e.g., sip.example.com ) convention, as specified in
   RFC 2219 [13]. Users may only know an email address instead of a full
   SIP URL for a callee, however. In that case, implementations may be
   able to increase the likelihood of reaching a SIP server for that
   domain by constructing a SIP URL from that email address by prefixing
   the domain name with "sip.". In the future, this mechanism is likely
   to become unnecessary as SRV records, described above, become widely
   available.

   A client MAY cache the list of DNS query results if one of the
   addresses was contacted successfully. Request for the same
   transaction SHOULD be sent to the same network address. Other
   requests from the same client select a server from the list of
   addresses cached, using the SRV load-balancing mechanism if
   applicable. The client must invalidate this list and retry the DNS
   query according to the rules in RFC1035 [14].

   A client MAY omit attempting to reach a server which it had failed to
   reach for a previous request.

   The results of the DNS lookup operation do not, in general, lead to a
   modification of the Request-URI.




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        A proxy is free to modify the Request-URI to any value
        desired, but the DNS lookups are usually based on the
        Request-URI obtained from a location server.


        If the DNS time-to-live value exceeds a few minutes,
        servers generating a large number of requests are probably
        well advised to retry failed servers every few minutes.

1.4.3 SIP Transaction

   Once the host part has been resolved to a SIP server, the client
   sends one or more SIP requests to that server and receives one or
   more responses from the server. A request (and its retransmissions)
   together with the responses triggered by that request make up a SIP
   transaction.  All responses to a request contain the same values in
   the Call-ID, CSeq, To, and From fields (with the possible addition of
   a tag in the To field (section 6.44)). This allows responses to be
   matched with requests. The ACK request confirming the receipt of an
   INVITE response is not part of the transaction since it may traverse
   a different set of hosts.

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

   If a client sends the request via a unicast datagram protocol such as
   UDP, the receiving user agent directs the response according to the
   information contained in the Via header fields (Section 6.47). Each
   proxy server in the forward path of the request forwards the response
   using these Via header fields, as described in detail in Sections
   6.47.3 and 6.47.4. For datagram protocols, reliability is achieved
   using retransmission (Section 10).

1.4.4 Initiating a Session

   A session is initiated with the INVITE request. A successful SIP
   invitation consists of two requests, INVITE followed by ACK. The
   INVITE (Section 4.2.1) request asks the callee to join a particular
   conference or establish a two-party conversation. After the callee
   has agreed to participate in the call, the caller confirms that it
   has received that response by sending an ACK (Section 4.2.2) request.

   The INVITE request typically contains a session description, for
   example written in SDP (RFC 2327 [6]) format, that provides the
   called party with enough information to join the session. For



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   multicast sessions, the session description enumerates the media
   types and formats that are allowed to be distributed to that session.
   For a unicast session, the session description enumerates the media
   types and formats that the caller is willing to use and where it
   wishes the media data to be sent. In either case, if the callee
   wishes to accept the call, it responds to the invitation by returning
   a similar description listing the media it wishes to use. For a
   multicast session, the callee SHOULD only return a session
   description if it is unable to receive the media indicated in the
   caller's description or wants to receive data via unicast.

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


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


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

1.4.5 Locating a User

   A callee may move between a number of different end systems over
   time.  These locations can be dynamically registered with the SIP
   server (Sections 1.4.7, 4.2.6). A location server MAY also use one or
   more other protocols, such as finger (RFC 1288 [15]), rwhois (RFC
   2167 [16]), LDAP (RFC 1777 [17]), multicast-based protocols [18] or



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                                         +....... cs.columbia.edu .......+
                                         :                               :
                                         : (~~~~~~~~~~)                  :
                                         : ( location )                  :
                                         : ( service  )                  :
                                         : (~~~~~~~~~~)                  :
                                         :     ^    |                    :
                                         :     | hgs@lab                 :
                                         :    2|   3|                    :
                                         :     |    |                    :
                                         : henning  |                    :
+.. cs.tu-berlin.de ..+ 1: INVITE        :     |    |                    :
:                     :    henning@cs.col:     |   \/ 4: INVITE  5: ring :
: cz@cs.tu-berlin.de ========================>(~~~~~~)=========>(~~~~~~) :
:                    <........................(      )<.........(      ) :
:                     : 7: 200 OK        :    (      )6: 200 OK (      ) :
:                     :                  :    ( work )          ( lab  ) :
:                     : 8: ACK           :    (      )9: ACK    (      ) :
:                    ========================>(~~~~~~)=========>(~~~~~~) :
+.....................+                  +...............................+

  ====> SIP request
  ....> SIP response

   ^
   |    non-SIP protocols
   |


   Figure 1: Example of SIP proxy server


   operating-system dependent mechanisms to actively determine the end
   system where a user might be reachable. A location server MAY return
   several locations because the user is logged in at several hosts
   simultaneously or because the location server has (temporarily)
   inaccurate information. The SIP server combines the results to yield
   a list of a zero or more locations.

   The action taken on receiving a list of locations varies with the
   type of SIP server. A SIP redirect server returns the list to the
   client as Contact headers (Section 6.15). A SIP proxy server can
   sequentially or in parallel try the addresses until the call is
   successful (2xx response) or the callee has declined the call (6xx



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                                         +....... cs.columbia.edu .......+
                                         :                               :
                                         : (~~~~~~~~~~)                  :
                                         : ( location )                  :
                                         : ( service  )                  :
                                         : (~~~~~~~~~~)                  :
                                         :    ^   |                      :
                                         :    | hgs@lab                  :
                                         :   2|  3|                      :
                                         :    |   |                      :
                                         : henning|                      :
+.. cs.tu-berlin.de ..+ 1: INVITE        :    |   |                      :
:                     :    henning@cs.col:    |   \/                     :
: cz@cs.tu-berlin.de =======================>(~~~~~~)                    :
:       | ^ |        <.......................(      )                    :
:       | . |         : 4: 302 Moved     :   (      )                    :
:       | . |         :    hgs@lab       :   ( work )                    :
:       | . |         :                  :   (      )                    :
:       | . |         : 5: ACK           :   (      )                    :
:       | . |        =======================>(~~~~~~)                    :
:       | . |         :                  :                               :
+.......|...|.........+                  :                               :
        | . |                            :                               :
        | . |                            :                               :
        | . |                            :                               :
        | . |                            :                               :
        | . | 6: INVITE hgs@lab.cs.columbia.edu                 (~~~~~~) :
        | . ==================================================> (      ) :
        | ..................................................... (      ) :
        |     7: 200 OK                  :                      ( lab  ) :
        |                                :                      (      ) :
        |     8: ACK                     :                      (      ) :
        ======================================================> (~~~~~~) :
                                         +...............................+

  ====> SIP request
  ....> SIP response

    ^
    |   non-SIP protocols
    |




   Figure 2: Example of SIP redirect server

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   response). With sequential attempts, a proxy server can implement an
   "anycast" service.

   If a proxy server forwards a SIP request, it MUST add itself to the
   beginning of the list of forwarders noted in the Via (Section 6.47)
   headers. The Via trace ensures that replies can take the same path
   back, ensuring correct operation through compliant firewalls and
   avoiding request loops. On the response path, each host MUST remove
   its Via, so that routing internal information is hidden from the
   callee and outside networks.

   A SIP invitation may traverse more than one SIP proxy server. If one
   of these "forks" the request, i.e., issues more than one request in
   response to receiving the invitation request, it is possible that a
   client is reached, independently, by more than one copy of the
   invitation request. Each of these copies bears the same Call-ID, but
   a different topmost Via header branch parameter (see Section 6.47).
   The user agent MAY choose which final response to return for each
   such request, typically returning a 200 (OK) for only one of them.

1.4.6 Changing an Existing Session

   In some circumstances, it is desirable to change the parameters of an
   existing session. This is done by re-issuing the INVITE within the
   same call leg, but within a new or different body or header fields to
   convey the new information. This re INVITE MUST have a higher CSeq
   than any previous request from the client to the server.

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

1.4.7 Registration Services

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

1.5 Protocol Properties

1.5.1 Minimal State

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



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   server MAY cache the results of location service requests.

1.5.2 Lower-Layer-Protocol Neutral

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

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

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

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

1.5.3 Text-Based

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

2 SIP Uniform Resource Locators

   SIP URLs are used within SIP messages to indicate the originator
   (From), current destination (Request-URI) and final recipient (To) of
   a SIP request, and to specify redirection addresses (Contact). A SIP
   URL can also be embedded in web pages or other hyperlinks to indicate
   that a particular user or service can be called via SIP. When used as
   a hyperlink, the SIP URL indicates the use of the INVITE method.

   The SIP URL scheme is defined to allow setting SIP request-header
   fields and the SIP message-body.



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        This corresponds to the use of mailto: URLs. It makes it
        possible, for example, to specify the subject, urgency or
        media types of calls initiated through a web page or as
        part of an email message.

   A SIP URL follows the guidelines of RFC 2396 [9] and has the syntax
   shown in Fig. 3. The syntax is described using Augmented Backus-Naur
   Form (see Section C). Note that reserved characters have to be
   escaped and that the "set of characters reserved within any given URI
   component is defined by that component. In general, a character is
   reserved if the semantics of the URI changes if the character is
   replaced with its escaped US-ASCII encoding" [9]. Excluded US-ASCII
   characters [9], such as space and control characters and characters
   used as URL delimiters, also MUST be escaped.  URLs MUSTNOT contain
   unescaped space and control characters.


   The URI character classes referenced above are described in Appendix
   C.

   The components of the SIP URI have the following meanings.

        user: The name of the user addressed. Note that this field MAY
             be empty where the destination host does not have a notion
             of users, e.g., for embedded devices.

        telephone-subscriber: If the host is an Internet telephony
             gateway, a telephone-subscriber field MAY be used instead
             of a user field. The telephone-subscriber field uses the
             notation of RFC 2806 [19]. Any characters of the un-escaped
             "telephone-subscriber" that are not either in the set
             "unreserved" or "user-unreserved" MUST be escaped. The set
             of characters not reserved in the RFC 2806 description of
             telephone-subscriber contains a number of characters in
             various syntax elements that need to be escaped when used
             in SIP URLs, for example quotation marks (%22), hash (%23),
             colon (%3a), at-sign (%40) and the "unwise" characters,
             i.e., punctuation of %5b and above.

             The telephone number is a special case of a user name and
             cannot be distinguished by a BNF. Thus, a URL parameter,
             user, is added to distinguish telephone numbers from user
             names.

             The user parameter value "phone" indicates that the user
             part contains a telephone number. Even without this
             parameter, recipients of SIP URLs MAY interpret the pre-@
             part as a telephone number if local restrictions on the



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  SIP-URL         = "sip:" [ userinfo "@" ] hostport
                    url-parameters [ headers ]
  userinfo        = [ user | telephone-subscriber ] [ ":" password ]
  user            = *( unreserved | escaped
                  | "&" | "=" | "+" | "$" | "," | ";" | "?" | "/" )
  password        = *( unreserved | escaped
                  | "&" | "=" | "+" | "$" | "," )
  hostport        = host [ ":" port ]
  host            = hostname | IPv4address | IPv6reference
  hostname        = *( domainlabel "." ) toplabel [ "." ]
  domainlabel     = alphanum | alphanum *( alphanum | "-" ) alphanum
  toplabel        = alpha | alpha *( alphanum | "-" ) alphanum
  IPv4address     = 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
  IPv6reference   = "[" IPv6address "]"
  IPv6address     = hexpart [ ":" IPv4address ]
  hexpart         = hexseq | hexseq "::" [ hexseq ] | "::" [ hexseq ]
  hexseq          = hex4 *( ":" hex4)
  hex4            = 1*4HEX
  port            = 1*DIGIT
  url-parameters  = *( ";" url-parameter )
  url-parameter   = transport-param | user-param | method-param
                  | ttl-param | maddr-param | other-param
  transport-param = "transport="
                    ( "udp" | "tcp" | "sctp" | "tls" | other-transport )
  other-transport = token
  ttl-param       = "ttl=" ttl
  ttl             = 1*3DIGIT       ; 0 to 255
  maddr-param     = "maddr=" host
  user-param      = "user=" ( "phone" | "ip" | other-user )
  other-user      = token
  method-param    = "method=" Method
  other-param     = pname [ "=" pvalue ]
  pname           = 1*paramchar
  pvalue          = 1*paramchar
  paramchar       = param-reserved | unreserved | escaped
  param-reserved  = "[" | "]" | "/" | "?" | ":" | "&" | "+" | "$"
  headers         = "?" header *( "&" header )
  header          = hname "=" hvalue
  hname           = 1*( hnv-unreserved | unreserved | escaped )
  hvalue          = *( hnv-unreserved | unreserved | escaped )
  hnv-unreserved  = "[" | "]" | "/" | "?" | ":" | "+" | "$"


   Figure 3: SIP URL syntax


             name space for user name allow it.

        password: The SIP scheme MAY use the format "user:password" in
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             clear text (such as URIs) has proven to be a security risk
             in almost every case where it has been used.

        host:  The host part SHOULD be a fully-qualified domain name or
             numeric IP address.

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

        port: The port number to send a request to. If not present, the
             procedures outlined in Section 1.4.2 are used to determine
             the port number to send a request to.

        URL parameters: SIP URLs can define specific parameters of the
             request. URL parameters are added after the host component
             and are separated by semi-colons. The transport parameter
             determines the the transport mechanism to be used for
             sending SIP requests and responses. SIP can use any network
             transport protocol; parameter names are defined for UDP
             [20], TCP [21], TLS [22], and SCTP. UDP is to be assumed
             when no explicit transport parameter is included. The maddr
             parameter indicates the server address to be contacted for
             this user, overriding the address supplied in the host
             field. This address is typically, but not necessarily, a
             multicast address.


             The maddr field can be used to force requests from
             traveling users to visit a home proxy even if an
             outbound proxy is used.

             The ttl parameter determines the time-to-live value of the
             UDP multicast packet and MUST only be used if maddr is a
             multicast address and the transport protocol is UDP. The
             user parameter was described above. 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, maddr, and ttl parameters MUSTNOT be used in



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             the From and To header fields; they are ignored if present.


             For Request-URIs, these parameters are useful
             primarily for outbound proxies.

             Receivers MUST silently ignore any URI parameters that they
             do not understand.

        Headers: Headers of the SIP request can be defined with the "?"
             mechanism within a SIP URL. The special hname "body"
             indicates that the associated hvalue is the message-body of
             the SIP INVITE request. Headers MUSTNOT be used in the From
             and To header fields and the Request-URI; they are ignored
             if present.  hname and hvalue are encodings of a SIP header
             name and value, respectively. All URL reserved characters
             in the header names and values MUST be escaped.

        Method: The method of the SIP request can be specified with the
             method parameter. This parameter MUSTNOT be used in the
             From and To header fields and the Request-URI; they are
             ignored if present.

   Table 2 summarizes where the components of the SIP URL can be used.
   Entries marked "m" are mandatory, those marked "o" are optional, and
   those marked "-" are not allowed. For optional elements, the second
   column indicates the default value if the element is not present.


                  default    Req.-URI  To  From  Contact  Rec.-Route  external
   user           --            o      o    o       o         o          o
   password       --            o      o    -       o         o          o
   host           mandatory     m      m    m       m         m          m
   port           5060          o      o    o       o         o          o
   user-param     ip            o      o    o       o         o          o
   method         INVITE        -      -    -       o         -          o
   maddr-param    --            o      -    -       o         m          o
   ttl-param      1             o      -    -       o         -          o
   transp.-param  udp           o      -    -       o         -          o
   other-param    o             o      o    o       o         o          o
   headers        --            -      -    -       o         -          o


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


   Examples of SIP URLs are:



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     sip:j.doe@big.com
     sip:j.doe:secret@big.com;transport=tcp
     sip:j.doe@big.com?subject=project
     sip:+1-212-555-1212:1234@gateway.com;user=phone
     sip:1212@gateway.com
     sip:alice@10.1.2.3
     sip:alice@example.com
     sip:alice
     sip:alice@registrar.com;method=REGISTER



2.1 SIP URL Comparison

   SIP URLs are compared for equality according to the following rules:

        o Comparisons of scheme name ("sip"), domain names, parameter
          names and header names are case-insensitive, all other
          comparisons are case-sensitive.

        o The ordering of parameters and headers is not significant in
          comparing SIP URLs.

        o user or telephone-subscriber, password, host, port and any
          url-parameter parameters of the URI must match. If a component
          is omitted, it matches based on its default value. (For
          example, otherwise equivalent URLs without a port
          specification and with port 5060 match.) Components not found
          in both URLs being compared are ignored.

        o Characters other than those in the "reserved" and "unsafe"
          sets (see RFC 2396 [9]) are equivalent to their ""%" HEX HEX"
          encoding.

        o An IP address that is the result of a DNS lookup of a host
          name does not match that host name.

   Thus, the following URLs are equivalent:

   sip:juser@
   sip:juser@ExAmPlE.CoM;Transport=udp


   while

   SIP:JUSER@ExAmPlE.CoM;Transport=udp
   sip:juser@ExAmPlE.CoM;Transport=UDP




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   are not.

   Header fields such as Contact, From and To are equal if and only if
   their URIs match under the rules above and if their header parameters
   (such as contact-param, from-param and to-param) match in name and
   parameter value, where parameter names and token parameter values are
   compared ignoring case and quoted-string parameter values are case-
   sensitive.

2.2 Non-SIP URLs

   SIP header fields and the Request-URI MAY contain non-SIP URLs, with
   the exceptions noted below. As an example, if a call from a telephone
   is relayed to the Internet via SIP, the SIP From header field might
   contain a tel: URL [19].

   In the following locations, only SIP URLs are allowed:

        o Request-URI in a REGISTER request;

        o Contact header field in INVITE, OPTIONS and BYE and 2xx
          responses.

   Implementations MAY compare non-SIP URLs by treating them as generic
   URIs [9] or, alternatively, compare them byte-by-byte.

3 SIP Message Overview

   SIP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [23]). Senders MUST terminate lines with a
   CRLF, but receivers MUST also interpret CR and LF by themselves as
   line terminators.  Only the combinations CR CR, LF LF and CRLF CRLF
   terminate the message header. Implementations MUST only send CRLF
   CRLF.

        CR and LF instead of CRLF is for backwards-compatibility;
        their use is deprecated.

   Except for the above difference in character sets and line
   termination, much of the message syntax is and header fields are
   identical to HTTP/1.1; rather than repeating the syntax and semantics
   here we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1
   specification (RFC 2616 [8]). In addition, we describe SIP in both
   prose and an augmented Backus-Naur form (ABNF). See section C for an
   overview of ABNF.

   Note, however, that SIP is not an extension of HTTP.




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   Unlike HTTP, SIP MAY use UDP or other unreliable datagram protocols.
   Each such datagram carries one request or response. Datagrams,
   including all headers, SHOULDNOT be larger than the path maximum
   transmission unit (MTU) if the MTU is known, or 1500 bytes if the MTU
   is unknown. However, implementations MUST be able to handle messages
   up to the maximum datagram packet size. For UDP, this size is 65,535
   bytes, including headers.


        The MTU of 1500 bytes accommodates encapsulation within the
        "typical" ethernet MTU without IP fragmentation. Recent
        studies [24] indicate that an MTU of 1500 bytes is a
        reasonable assumption. The next lower common MTU values are
        1006 bytes for SLIP and 296 for low-delay PPP (RFC 1191
        [25]). Thus, another reasonable value would be a message
        size of 950 bytes, to accommodate packet headers within the
        SLIP MTU without fragmentation.

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



        SIP-message  =  Request | Response


   Both Request (section 4) and Response (section 5) messages use the
   generic-message format of RFC 822 [26] for transferring entities (the
   body of the message). Both types of messages consist of a start-line,
   one or more header fields (also known as "headers"), an empty line
   (i.e., a line with nothing preceding the carriage-return line-feed
   (CRLF)) indicating the end of the header fields, and an optional
   message-body. To avoid confusion with similar-named headers in HTTP,
   we refer to the headers describing the message body as entity
   headers. These components are described in detail in the upcoming
   sections.



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

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





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



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

4 Request

   The Request message format is shown below:



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


4.1 Request-Line

   The Request-Line begins with a method token, followed by the
   Request-URI and the protocol version, and ending with CRLF. The
   elements are separated by SP characters.  No CR or LF are allowed
   except in the final CRLF sequence.  No LWS is allowed in any of the
   elements. The Request-URI MUSTNOT be enclosed in "<>". absoluteURI is
   defined in [H3.2.1].



        Request-Line  =  Method SP Request-URI SP SIP-Version CRLF
        Request-URI   =  SIP-URL | absoluteURI
        SIP-Version   =  "SIP/2.0"


4.2 Methods

   The methods are defined below. Proxy and redirect servers treat all
   methods other than INVITE and CANCEL in the same way, by forwarding
   them accordingly. Thus, no method-specific support is required in
   these servers for methods other than INVITE and CANCEL.  Methods that
   are not supported by a user agent server or registrar cause a 501



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   (Not Implemented) response to be returned (Section 7). As in HTTP,
   the Method token is case-sensitive.



        Method            =  "INVITE" | "ACK" | "OPTIONS" | "BYE"
                             | "CANCEL" | "REGISTER" | extension-method
        extension-method  =  token


4.2.1 INVITE

   The INVITE method indicates that the user or service is being invited
   to participate in a session. The message body MAY contain a
   description of the session to which the callee is being invited. For
   two-party calls, the caller indicates the type of media it is able to
   receive and possibly the media it is willing to send as well as their
   parameters such as network destination. A success response MUST
   indicate in its message body which media the callee wishes to receive
   and MAY indicate the media the callee is going to send.


        Not all session description formats have the ability to
        indicate sending media.

   The caller MAY choose to omit the request body (i.e., not send a
   session description) or send a session description that does not list
   any media types. This indicates that the caller does not know its
   desired media characteristics until the call has been accepted. In
   this case, the UAS SHOULD still return a session description in its
   informational (1xx) or success (2xx) response, containing those media
   streams and codecs it supports.

   If the INVITE request did not contain a complete session description,
   the caller MUST include one in the ACK request. A UAC MUSTNOT send an
   updated session description in an ACK request if it had already sent
   a complete session description in the INVITE request.  If the UAC
   wishes to modify the session after the call setup has begun, it MUST
   use another INVITE request instead.


        Delaying the session description until the ACK request is
        useful for gateways from H.323v1 to SIP, where the H.323
        media characteristics are not known until the call is
        established.

   A server MAY automatically respond to an invitation for a conference
   the user is already participating in, identified either by the SIP



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        general-header   =  Accept               ; Section 6.6
                         |  Accept-Encoding      ; Section 6.7
                         |  Accept-Language      ; Section 6.8
                         |  Call-ID              ; Section 6.13
                         |  Call-Info            ; Section 6.14
                         |  Contact              ; Section 6.15
                         |  CSeq                 ; Section 6.21
                         |  Date                 ; Section 6.22
                         |  Encryption           ; Section 6.23
                         |  From                 ; Section 6.26
                         |  MIME-Version         ; Section 6.29
                         |  Organization         ; Section 6.30
                         |  Record-Route         ; Section 6.35
                         |  Require              ; Section 6.36
                         |  Supported            ; Section 6.42
                         |  Timestamp            ; Section 6.43
                         |  To                   ; Section 6.44
                         |  User-Agent           ; Section 6.46
                         |  Via                  ; Section 6.47
        entity-header    =  Allow                ; Section 6.10
                         |  Content-Disposition  ; Section 6.16
                         |  Content-Encoding     ; Section 6.17
                         |  Content-Language     ; Section 6.18
                         |  Content-Length       ; Section 6.19
                         |  Content-Type         ; Section 6.20
                         |  Expires              ; Section 6.25
        request-header   =  Alert-Info           ; Section 6.9
                         |  Authorization        ; Section 6.12
                         |  In-Reply-To          ; Section 6.27
                         |  Max-Forwards         ; Section 6.28
                         |  Priority             ; Section 6.31
                         |  Proxy-Authorization  ; Section 6.33
                         |  Proxy-Require        ; Section 6.34
                         |  Route                ; Section 6.39
                         |  Response-Key         ; Section 6.37
                         |  Subject              ; Section 6.41
        response-header  =  Error-Info           ; Section 6.24
                         |  Proxy-Authenticate   ; Section 6.32
                         |  Retry-After          ; Section 6.38
                         |  Server               ; Section 6.40
                         |  Unsupported          ; Section 6.45
                         |  Warning              ; Section 6.48
                         |  WWW-Authenticate     ; Section 6.49


   Table 3: SIP headers



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   The behavior of UAS depend on whether they are Internet telephony
   gateways to the PSTN. A UAS not acting as a gateway which receives an
   INVITE with a Request-URI that does not correspond to one of its
   configured addresses, MUST respond with 404 (Not Found).

   A UAS acting as a gateway translates the INVITE request into a
   telephony signaling message. If the INVITE has a Call-ID value that
   matches a recent call, the UAS compares the Request-URI with the
   Request-URI of the previous INVITE request for the same Call-ID. If
   the Request-URI contains additional digits in the "user" part, the
   UAS treats the INVITE as adding additional digits to the original
   dialed string. This is known as overlap dialing.

   If the gateway knows that the telephone number is incomplete, it
   returns a 484 (Address Incomplete) status response.

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

   If an INVITE request for an existing session fails, the session
   description agreed upon in the last successful INVITE transaction
   remains in force.

   A UAC MUSTNOT issue another INVITE request for the same call leg
   before the previous transaction has completed. A UAS that receives an
   INVITE before it sent the final response to an INVITE with a lower
   CSeq number MUST return a 400 (Bad Request) response and MUST include
   a Retry-After header field with a randomly chosen value of between 0
   and 10 seconds.

   If a UA A sends an INVITE request to B and receives an INVITE request
   from B before it has received the response to its request from B, A
   MAY return a 500 (Internal Server Error), which SHOULD include a
   Retry-After header field specifying when the request should be
   resubmitted.





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        In most cases, a UA can assume that the order of messages
        received corresponds to the order they were sent. In rare
        circumstances, the response from B and the request from B
        may be reordered on the wire.

   In addition, if A or B change multicast addresses, strict transaction
   ordering is necessary so that both sides agree on the final result.

   A UAC MUST be prepared to receive media data according to the session
   description as soon as it sends an INVITE (or re-INVITE) and can
   start sending media data when it receives a provisional or final
   response containing a session description.

   The initial INVITE from the UAC SHOULD contain the Allow and
   Supported header fields, and MAY contain the Accept header field. A
   200 (OK) response to the initial INVITE for a call SHOULD contain the
   Allow and Supported header fields, and MAY contain the Accept header
   field.


        Including these header fields allows the UAC to determine
        the features and extensions supported by the UAS for the
        duration of the call, without probing.

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

4.2.2 ACK

   The ACK request confirms that the client has received a final
   response to an INVITE request. (ACK is used only with INVITE
   requests.) 2xx responses are acknowledged by client user agents, all
   other final responses by the first proxy or client user agent to
   receive the response. The Via is always initialized to the host that
   originates the ACK request, i.e., the client user agent after a 2xx
   response or the first proxy to receive a non-2xx final response. The
   ACK request is forwarded as the corresponding INVITE request, based
   on its Request-URI and thus MAY take a different path than the
   original INVITE request, and MAY even cause a new transport
   connection to be opened in order to send it.

   The ACK request does not generate responses for any transport
   protocol.

   The ACK request MAY contain a message body with the final session
   description to be used by the callee. If the ACK message body is
   empty, the callee uses the session description in the INVITE request.
   See Section 4.2.1 for further details on the relationship between



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   session descriptions in INVITE and ACK requests.

   A proxy server receiving an ACK request after having sent a 3xx, 4xx,
   5xx, or 6xx response must make a determination about whether the ACK
   is for it, or for some user agent or proxy server further downstream.
   This determination is made by examining the tag in the To field. If
   the tag in the ACK To header field matches the tag in the To header
   field of the response, and the From, CSeq and Call-ID header fields
   in the response match those in the ACK, the ACK is meant for the
   proxy server. Otherwise, the ACK SHOULD be proxied downstream as any
   other request.


        It is possible for a user agent client or proxy server to
        receive multiple 3xx, 4xx, 5xx, and 6xx responses to a
        request along a single branch. This can happen under
        various error conditions, typically when a forking proxy
        transitions from stateful to stateless before receiving all
        responses. The various responses will all be identical,
        except for the tag in the To field, which is different for
        each one. It can therefore be used as a means to
        disambiguate them.

   This method MUST be supported by SIP user agents.

4.2.3 OPTIONS

   The server is being queried as to its capabilities. A server that
   believes it can contact the user, such as a user agent where the user
   is logged in and has been recently active, MAY respond to this
   request with a capability set. A called user agent MAY return a
   status reflecting how it would have responded to an invitation, e.g.,
   600 (Busy).  A server SHOULD return Allow, Accept, Accept-Encoding,
   Accept-Language and Supported header fields. The response MAY contain
   a message body indicating the capabilities of the end system (rather
   than properties of any existing call).

   The use of the Call-ID header field is discussed in Section 6.13. An
   OPTIONS requests for an existing call-id has no impact on that call.

   This method MUST be supported by SIP user agents and registrars.

4.2.4 BYE

   The user agent client uses BYE to indicate to the server that it
   wishes to release the call. A BYE request is forwarded like an INVITE
   request and MAY be issued by either caller or callee. A party to a
   call SHOULD issue a BYE request before releasing a call ("hanging



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   up"). A party receiving a BYE request MUST cease transmitting media
   streams specifically directed at the party issuing the BYE request.

   A BYE request from either called or calling party terminates any
   pending INVITE at a UA, but the INVITE request transaction MUST be
   completed with a final response and ACK.

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

   This method SHOULD be supported by user agent servers.

4.2.5 CANCEL

   The CANCEL request cancels a pending request with the same Call-ID,
   To, From and CSeq (sequence number only) header field values, but
   does not affect a completed request or existing calls. (A request is
   considered completed if the server has returned a final status
   response.) The UAC can use a BYE request to terminate a call if the
   CANCEL arrived too late.

   A user agent client or proxy client MAY issue a CANCEL request at any
   time. A proxy client generates a CANCEL request for branches without
   a final response after it has forked a request and receives a 2xx or
   6xx response from one of the branches. A UAC or proxy client also
   sends a CANCEL if the time noted in the Expires header of the request
   has elapsed or no provisional or final response was received after a
   client-determined timeout interval. Finally, internal logic such as
   scripts, can trigger CANCEL requests.

   A proxy that receives a CANCEL request forwards the request to all
   destinations with pending requests.

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

   The behavior of the user agent or redirect server on receiving a
   CANCEL request depends on whether the server has already sent a final
   response for the original request. If it has, the CANCEL request has
   no effect on the original request, any call state and on the
   responses generated for the original request. If the server has not
   issued a final response for the original request, it immediately



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   sends a 487 (Request Terminated) for the original request. For INVITE
   requests, the UAC as usual sends an ACK request to confirm receipt of
   any final response. The CANCEL request itself is answered with a 200
   (OK) response in either case.

   A proxy client or UAC cannot rely on receiving a 487 (Request
   Terminated) response, as a RFC 2543-compliant UAS will not generate
   such a response. If there has been no final response after 32
   seconds, the client MAY consider the original transaction to have
   been cancelled.


        The BYE request cannot be used to cancel branches of a
        parallel search, since several branches may, through
        intermediate proxies, find the same user agent server and
        then terminate the call.  To terminate a call instead of
        just pending searches, the UAC must use BYE instead of or
        in addition to CANCEL. While CANCEL can terminate any
        pending request other than ACK or CANCEL, it is typically
        useful only for INVITE. 200 responses to INVITE and 200
        responses to CANCEL can be distinguished by the method in
        the Cseq header field.

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

4.2.6 REGISTER

   A client uses the REGISTER method to bind the address listed in the
   To header field with a SIP server to one or more URIs where the
   client can be reached, contained in the Contact header fields.  These
   URIs may use any URI scheme, not limited to SIP.

   It is particularly important that REGISTER requests are authenticated
   since they allow to redirect future requests (see Section 13.2).

4.2.6.1 Where to Register

   A user agent SHOULD attempt to register periodically according to the
   rules below.  A UA is said to be "visiting" if its From address
   domain differs from the current network domain and is said to be "at
   home" if the two are the same.

        Local server: If an outbound proxy is configured, the UA SHOULD
             send a REGISTER request to it. If the UA is visiting, it
             uses the From address consisting of the URL-escaped user
             identity at the visited domain, e.g., the user identified
             as alice@wonderland.com would register as



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             alice%40wonderland.com@example.com if she is visiting the
             example.com domain.

        Multicast: If no local outbound proxy is configured, multicast
             registrations are addressed to the well-known "all SIP
             servers" multicast address "sip.mcast.net" (224.0.1.75).
             This request SHOULD be scoped to ensure it is not forwarded
             beyond the boundaries of the administrative system. This
             MAY be done with either TTL or administrative scopes [27],
             depending on what is implemented in the network. SIP user
             agents MAY listen to that address and use it to become
             aware of the location of other local users [18]; however,
             they do not respond to the request.


             Multicast registration may be inappropriate in some
             environments, for example, if multiple businesses
             share the same local area network.

        Home server: If the UA is visiting, it SHOULD also send a
             registration to its home SIP server, identified by its home
             address. For example, alice@wonderland.com would send a
             registration to the SIP server for the domain
             wonderland.com when she is visiting another network. TBD:
             What Contact should be used?

   A user agent SHOULD register with a local server on startup and
   periodically thereafter by sending a REGISTER request. The period is
   given by the expiration time indicated in the registration response.
   It is RECOMMENDED that the UA registers via multicast and send a
   registration to its "home" address, i.e., the server for the domain
   that it uses as its From address in outgoing requests.

4.2.6.2 REGISTER Header Fields

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




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        Call-ID: All registrations from a client SHOULD use the same
             Call-ID header value, at least within the same reboot
             cycle.

        Cseq: Registrations with the same Call-ID MUST have increasing
             CSeq header values. However, the server does not reject
             out-of-order requests.

4.2.6.3 Registering Contact Locations

   REGISTER requests are processed in the order received. Clients SHOULD
   avoid sending a new registration (as opposed to a retransmission)
   until they have received the response from the server for the
   previous one.


        Clients may register from different locations, by necessity
        using different Call-ID values. Thus, the CSeq value cannot
        be used to enforce ordering. Since registrations are
        additive, ordering is less of a problem than if each
        REGISTER request completely replaced all earlier ones.

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

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

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

        Contact: The request MAY contain a Contact header field. Future
             non-REGISTER requests for the URI given in the To header
             field SHOULD be directed to the address(es) given in the
             Contact header.

             If the request does not contain a Contact header, the
             registration remains unchanged.

             This is useful to obtain the current list of
             registrations in the response, as described below.

             If a SIP URI in a registration Contact header field differs
             from existing registrations according to the rules in



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             Section 2.1, it is added to the list of registration. If it
             is equivalent, according to these rules, to an existing
             registration, all Contact header field parameters for this
             entry are updated accordingly. URIs other than SIP URIs are
             compared according to the standard URI equivalency rules
             for the URI schema.

             All current registrations MUST share the same action value.
             Registrations that have a different action than current
             registrations for the same user MUST be rejected with
             status of 409 (Conflict).

             A proxy server ignores the q parameter when processing
             non-REGISTER requests, while a redirect server simply
             returns that parameter in its Contact response header
             field.


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

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


        This allows a service known as "directed pick-up". In the
        telephone network, directed pickup permits a user at a
        remote station who hears his own phone ringing to pick up
        at that station, dial an access code, and be connected to
        the calling user as if he had answered his own phone.

4.2.6.4 Registration Expiration

   An optional "expires" Contact parameter indicates the desired
   expiration time of the registration. If a Contact entry does not have
   an "expires" parameter, the Expires header field is used as the
   default value. If neither of these mechanisms is used, SIP URIs are
   assumed to expire after one hour. Other URI schemes have no
   expiration times. Registrations not refreshed after this amount of
   time SHOULD be silently discarded. A registrar SHOULDNOT use a longer
   lifetime than the one requested, but MAY use a shorter one.

   Registration refreshes SHOULD be sent to the same address as the
   original registration, unless redirected.

4.2.6.5 List of Current Registrations



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   2xx REGISTER responses SHOULD list all current registration in the
   Contact header field.  An "expires" parameter MUST indicate the
   expiration time of the registration.

4.2.6.6 Removing Registrations

   Registrations expire as described above or may be removed explicitly
   by setting the expires parameter for an existing registration to zero
   or including an Expires: 0 header field. Registrations are matched
   based on the user, host, port and maddr parameters. A client can
   remove all registrations by including a single Contact header field
   with the wildcard address "*".

   Support of this method is RECOMMENDED; registrars MUST support it.

4.3 Request-URI

   The Request-URI is a SIP URL as described in Section 2 or a general
   URI (RFC 2396 [9]).  In particular, it MUSTNOT contain unescaped
   spaces or control characters. It indicates the user or service to
   which this request is being addressed. Unlike the To field, the
   Request-URI MAY be re-written by proxies.

   As shown in Table 2, the Request-URI MAY contain the user-param
   parameter as well as transport-related parameters. A server that
   receives a SIP-URL with illegal elements removes them before further
   processing.


        Transport-related parameters are needed when a UAC proxies
        all requests to a default proxy, which would then need this
        information to generate the appropriate request.


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

   Proxy and redirect servers MAY use the information in the Request-URI
   and request header fields to handle the request and possibly rewrite
   the Request-URI. For example, a request addressed to the generic
   address sip:sales@acme.com is proxied to the particular person, e.g.,
   sip:bob@ny.acme.com , with the To field remaining as
   sip:sales@acme.com.  At ny.acme.com , Bob then designates Alice as



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   the temporary substitute.

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

   SIP servers MAY support Request-URIs with schemes other than "sip",
   for example the "tel" URI scheme [19]. It MAY translate non-SIP URIs
   using any mechanism at its disposal, resulting in either a SIP URI or
   some other scheme.

   If a SIP server receives a request with a URI indicating a scheme the
   server does not understand, the server MUST return a 400 (Bad
   Request) response. It MUST do this even if the To header field
   contains a scheme it does understand, since proxies are responsible
   for processing the Request-URI. (The To field is only of interest to
   the UAS.)

4.3.1 SIP Version

   Both request and response messages include the version of SIP in use,
   and follow [H3.1] (with HTTP replaced by SIP, and HTTP/1.1 replaced
   by SIP/2.0) regarding version ordering, compliance requirements, and
   upgrading of version numbers. To be compliant with this
   specification, applications sending SIP messages MUST include a SIP-
   Version of "SIP/2.0".  The string is case-insensitive, but
   implementations MUST use upper-case.

        Unlike HTTP/1.1, SIP treats the version number as a literal
        string. In practice, this should make no difference.

4.4 Option Tags

   Option tags are unique identifiers used to designate new options in
   SIP.  These tags are used in Require (Section 6.36), Supported
   (Section 6.42) and Unsupported (Section 6.45) header fields.

   Syntax:


        option-tag  =  token


   See Section C for the definition of token. The creator of a new SIP



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   option MUST either prefix the option with their reverse domain name
   or register the new option with the Internet Assigned Numbers
   Authority (IANA).

   An example of a reverse-domain-name option is "com.foo.mynewfeature",
   whose inventor can be reached at "foo.com". For these features,
   individual organizations are responsible for ensuring that option
   names do not collide within the same domain. The host name part of
   the option MUST use lower-case; the option name is case-sensitive.

   Options registered with IANA do not contain periods and are globally
   unique. IANA option tags are case-sensitive.

4.4.1 Registering New Option Tags with IANA

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

        o Name and description of option. The name MAY be of any length,
          but SHOULD be no more than twenty characters long. The name
          MUST consist of alphanum (See Figure 3) characters only;

        o A listing of any new SIP header fields, header parameter
          fields or parameter values defined by this option. A SIP
          option MUSTNOT redefine header fields or parameters defined in
          either RFC 2543, any standards-track extensions to RFC 2543,
          or other extensions registered through IANA.

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

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

        o Contact information (postal and email address).

   Registrations should be sent to iana@iana.org


        This procedure has been borrowed from RTSP [4] and the RTP
        AVP [28].

5 Response




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



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


   SIP's structure of responses is similar to [H6], but is defined
   explicitly here.

5.1 Status-Line

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



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


5.1.1 Status Codes and Reason Phrases

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



        Status-Code     =  Informational                     ;Fig. 4
                       |   Success                           ;Fig. 4
                       |   Redirection                       ;Fig. 5
                       |   Client-Error                      ;Fig. 6
                       |   Server-Error                      ;Fig. 7
                       |   Global-Failure                    ;Fig. 8
                       |   extension-code



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        extension-code  =  3DIGIT
        Reason-Phrase   =  *<TEXT-UTF8,  excluding CR, LF>


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

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

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

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

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

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

        6xx: Global Failure -- the request cannot be fulfilled at any
             server.

   Figures 4 through 8 present the individual values of the numeric
   response codes, and an example set of corresponding reason phrases
   for SIP/2.0. These reason phrases are only recommended; they may be
   replaced by local equivalents without affecting the protocol. Note
   that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
   codes in the range starting at x80 to avoid conflicts with newly
   defined HTTP response codes, and adds a new class, 6xx, of response
   codes.

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



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   include human-readable information which will explain the unusual
   status.



        Informational  =  "100"  ;  Trying
                      |   "180"  ;  Ringing
                      |   "181"  ;  Call Is Being Forwarded
                      |   "182"  ;  Queued
                      |   "183"  ;  Session Progress
        Success        =  "200"  ;  OK


   Figure 4: Informational and success status codes





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


   Figure 5: Redirection status codes






6 Header Field Definitions

   SIP header fields are similar to HTTP header fields in both syntax
   and semantics. In particular, SIP header fields follow the syntax for
   message-header as described in [H4.2]. The rules for extending header
   fields over multiple lines, and use of multiple message-header fields
   with the same field-name, described in [H4.2] also apply to SIP. The
   rules in [H4.2] regarding ordering of header fields apply to SIP,
   with the exception of Via fields, see below, whose order matters.

   The header fields required, optional and not applicable for each
   method are listed in Table 4 and Table 5.  The table uses "o" to
   indicate optional, "m" mandatory and "-" for not applicable.
   "Optional" means that a UA MAY include the header field in a request
   or response, and a UA MAY ignore the header field if present in the



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        Client-Error  =  "400"  ;  Bad Request
                     |   "401"  ;  Unauthorized
                     |   "402"  ;  Payment Required
                     |   "403"  ;  Forbidden
                     |   "404"  ;  Not Found
                     |   "405"  ;  Method Not Allowed
                     |   "406"  ;  Not Acceptable
                     |   "407"  ;  Proxy Authentication Required
                     |   "408"  ;  Request Timeout
                     |   "409"  ;  Conflict
                     |   "410"  ;  Gone
                     |   "411"  ;  Length Required
                     |   "413"  ;  Request Entity Too Large
                     |   "414"  ;  Request-URI Too Large
                     |   "415"  ;  Unsupported Media Type
                     |   "420"  ;  Bad Extension
                     |   "480"  ;  Temporarily not available
                     |   "481"  ;  Call Leg/Transaction Does Not Exist
                     |   "482"  ;  Loop Detected
                     |   "483"  ;  Too Many Hops
                     |   "484"  ;  Address Incomplete
                     |   "485"  ;  Ambiguous
                     |   "486"  ;  Busy Here
                     |   "487"  ;  Request Cancelled
                     |   "488"  ;  Not Acceptable Here


   Figure 6: Client error status codes




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


   Figure 7: Server error status codes


   request or response. A "mandatory" request header field MUST be
   present in a request, and MUST be understood by the UAS receiving the
   request. A mandatory response header field MUST be present in the



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


   Figure 8: Global failure status codes


   response, and the header field MUST be understood by the UAC
   processing the response. "Not applicable" means for request header
   fields that the header field MUSTNOT be present in a request.  If one
   is placed in a request by mistake, it MUST be ignored by the UAS
   receiving the request. Similarly, a header field labeled "not
   applicable" for a response means that the UAS MUSTNOT place the
   header in the response, and the UAC MUST ignore the header in the
   response. "m*" indicates a header that SHOULD be sent, but servers
   need to be prepared to receive requests without that header field. A
   "*" indicates that the header fields are needed only if message body
   is not empty. See sections 6.19, 6.20 and 8 for details.

   The "where" column describes the request and response types with
   which the header field can be used. "R" refers to header fields that
   can be used in requests (that is, request and general header fields).
   "r" designates a response or general-header field as applicable to
   all responses, while a list of numeric values indicates the status
   codes with which the header field can be used. "g" and "e" designate
   general (Section 6.1) and entity header (Section 6.2) fields,
   respectively. If a header field is marked "c", it is copied from the
   request to the response.

   The "proxy" column describes whether proxies can add comma-separated
   elements to headers ("c", for concatenate or comma), can modify the
   header ("m"), can add the header if not present ("a") or need to read
   the header ("r"). Headers that need to be read cannot be encrypted.
   Proxies MUSTNOT alter any fields that are authenticated (see Section
   13.2), but MAY add copies of fields that were authenticated by the UA
   if indicated in the table. Depending on local policy, proxies MAY
   inspect any non-encrypted header fields and MAY modify any non-
   authenticated header field, but proxies cannot rely on fields other
   than the ones indicated in the table to be readable or modifiable.

   If authentication is used, the rules in Section 13.2 apply. Proxies
   SHOULDNOT re-order header fields.





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        Header field         where  proxy ACK BYE CAN INV OPT REG
        __________________________________________________________
        Accept                 R           -   o   o   o   o   o
        Accept                415          -   o   o   o   o   o
        Accept                 r           -   -   -   o   o   o
        Accept-Encoding        R           -   o   o   o   o   o
        Accept-Encoding       415          -   o   o   o   o   o
        Accept-Language        R           -   o   o   o   o   o
        Accept-Language       415          -   o   o   o   o   o
        Alert-Info             R     am    -   -   -   o   -   -
        Allow                  R           o   o   o   o   o   o
        Allow                 200          -   -   -   o   o   o
        Allow                 405          m   m   m   m   m   m
        Also                   R           -   o   -   -   -   -
        Authorization          R           o   o   o   o   o   o
        Authorization          r           o   o   o   o   o   o
        Call-ID               gc      r    m   m   m   m   m   m
        Call-Info              g     am    -   -   -   o   o   o
        Contact                R           o   -   -   m   o   o
        Contact               1xx          -   -   -   o   o   -
        Contact               2xx          -   -   -   m   o   o
        Contact               3xx          -   o   -   o   o   o
        Contact               485          -   o   -   o   o   o
        Content-Disposition    e           o   o   -   o   o   o
        Content-Encoding       e           o   o   -   o   o   o
        Content-Language       e           o   o   o   o   o   o
        Content-Length         e      r    m*  m*  m*  m*  m*  m*
        Content-Type           e           *   *   -   *   *   *
        CSeq                  gc      r    m   m   m   m   m   m
        Date                   g      a    o   o   o   o   o   o
        Encryption             g      r    o   o   o   o   o   o
        Error-Info             R           o   o   o   o   o   o
        Expires                g           -   -   -   o   -   o
        From                  gc      r    m   m   m   m   m   m
        In-Reply-To            R           -   -   -   o   -   -
        Max-Forwards           R     rm    o   o   o   o   o   o
        MIME-Version           g           o   o   o   o   o   o
        Organization           g     am    -   -   -   o   o   o


   Table 4: Summary of header fields, A--O


   Other header fields can be added as required; a server MUST ignore
   header fields not defined in this specification that it does not
   understand. A proxy MUSTNOT remove or modify header fields not
   defined in this specification that it does not understand. A compact
   form of these header fields is also defined in Section 9 for use over


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    Header field              where       proxy ACK BYE CAN INV OPT REG
    ___________________________________________________________________
    Priority                    R           a    -   -   -   o   -   -
    Proxy-Authenticate       401,407             o   o   o   o   o   o
    Proxy-Authorization         R           r    o   o   o   o   o   o
    Proxy-Require               R           r    o   o   o   o   o   o
    Record-Route                R          amr   o   o   o   o   o   o
    Record-Route           2xx,401,484           o   o   o   o   o   o
    Require                     g          acr   o   o   o   o   o   o
    Response-Key                R                -   o   o   o   o   o
    Retry-After                 R                -   -   -   -   -   o
    Retry-After          404,413,480,486         o   o   o   o   o   o
                             500,503             o   o   o   o   o   o
                             600,603             o   o   o   o   o   o
    Route                       R           r    o   o   o   o   o   o
    Server                      r                o   o   o   o   o   o
    Subject                     R                -   -   -   o   -   -
    Supported                   g                -   o   o   o   o   o
    Timestamp                   g                o   o   o   o   o   o
    To                        gc(1)         r    m   m   m   m   m   m
    Unsupported                 R                o   o   o   o   o   o
    Unsupported                420               o   o   o   o   o   o
    User-Agent                  g                o   o   o   o   o   o
    Via                        gc         acmr   m   m   m   m   m   m
    Warning                     r                o   o   o   o   o   o
    WWW-Authenticate            R                o   o   o   o   o   o
    WWW-Authenticate           401               o   o   o   o   o   o


   Table 5: Summary of header fields, P--Z; (1):  copied  with  possible
   addition of tag

   UDP when the request has to fit into a single packet and size is an
   issue.

   Table 6 in Appendix A lists those header fields that different client
   and server types MUST be able to parse.

6.1 General Header Fields

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



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6.2 Entity Header Fields

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

6.3 Request Header Fields

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

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

6.4 Response Header Fields

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

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

6.5 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 [26]. 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.



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   Header fields can be extended over multiple lines by preceding each
   extra line with at least one SP or horizontal tab (HT). Applications
   MUST 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-UTF8
                            or combinations of token,
                            separators, and quoted-string>


   The relative order of header fields with different field names is not
   significant. 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 MUSTNOT
   change the order of these field values when a message is forwarded.

   Unless otherwise stated, parameter names, parameter values and tokens
   are case-insensitive. Values expressed as quoted strings are case-
   sensitive.

   The Contact, From and To header fields contain a URL. If the URL
   contains a comma, question mark or semicolon, the URL MUST be
   enclosed in angle brackets (< and >). Any URL parameters are
   contained within these brackets. If the URL is not enclosed in angle
   brackets, any semicolon-delimited parameters are header-parameters,
   not URL parameters.

6.6 Accept

   The Accept header follows the syntax defined in [H14.1]. The
   semantics are also identical, with the exception that if no Accept
   header is present, the server SHOULD assume a default value of
   application/sdp

   As a request-header field, it is used only with those methods that



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   take message bodies. In a 415 (Unsupported Media Type) response, it
   indicates which content types are acceptable in requests. In 200 (OK)
   responses for INVITE, it lists the content types acceptable for
   future requests in this call.

   Example:


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



6.7 Accept-Encoding

   The Accept-Encoding general-header field is similar to Accept, but
   restricts the content-codings [H3.4.1] that are acceptable in the
   response. See [H14.3]. The syntax of this header is defined in
   [H14.3]. The semantics in SIP are identical to those defined in
   [H14.3].


        Note: An empty Accept-Encoding header field is permissible,
        even though the syntax in [H14.3] does not provide for it.
        It is equivalent to Accept-Encoding: identity, i.e., only
        the identity encoding, meaning no encoding, is permissible.

   If no Accept-Encoding header field is present in a request, the
   server MUST use the "identity" encoding.


        HTTP/1.1 [H14.3] states that the server SHOULD use the
        "identity" encoding unless it has additional information
        about the capabilities of the client. This is needed for
        backwards-compatibility with old HTTP clients and does not
        affect SIP.

6.8 Accept-Language

   The Accept-Language general-header follows the syntax defined in
   [H14.4]. The rules for ordering the languages based on the q
   parameter apply to SIP as well. When used in SIP, the Accept-Language
   general-header field can be used to allow the client to indicate to
   the server in which language it would prefer to receive reason
   phrases, session descriptions or status responses carried as message
   bodies. A proxy MAY use this field to help select the destination for
   the call, for example, a human operator conversant in a language
   spoken by the caller.




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   Example:


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



6.9 Alert-Info

   The Alert-Info header field indicates that the content indicated in
   the URLs should be rendered instead of ring tone. A user SHOULD be
   able to disable this feature selectively to prevent unauthorized
   disruptions.



        Alert-Info     =  "Alert-Info" ":" # ( "<" URI ">" *( ";" generic-param ))
        generic-param  =  token [ "=" ( token | host | quoted-string ) ]


   Example:

   Alert-Info: <http://wwww.example.com/sounds/moo.wav>



6.10 Allow

   The Allow header field lists the set of methods supported by the
   resource identified by the Request-URI. The purpose of this field is
   strictly to inform the recipient of valid methods associated with the
   resource. An Allow header field MUST be present in a 405 (Method Not
   Allowed) response, SHOULD be present in an OPTIONS response SHOULD be
   present in the 200 (OK) response to the initial INVITE for a call and
   MAY be present in final responses for other methods.  All methods,
   including ACK and CANCEL, understood by the UAS are included.

   The Allow header field MAY also be included in requests, to indicate
   the requestor's capabilities for this Call-ID.


        Supplying an Allow header in responses to methods other
        than OPTIONS cuts down on the number of messages needed.



        Allow  =  "Allow" ":" 1#Method




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6.11 Also

   The Also header is used only in BYE requests. It indicates to the
   receiving UA that it should initiate an INVITE request to the
   addresses indicated. The BYE request takes effect regardless of
   whether the INVITE succeeds and is responded to immediately.


        This mechanism allows unsupervised call transfer.



        Also  =  "Also" ":" 1# (( name-addr | addr-spec )


6.12 Authorization

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

   Section 13.2 overviews the use of the Authorization header field, and
   Section 15 describes the syntax and semantics when used with PGP-
   based authentication.

6.13 Call-ID

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

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




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

   The REGISTER and OPTIONS methods use the Call-ID value (in addition
   to the CSeq value) to unambiguously match requests and responses. All
   REGISTER requests issued by a single client SHOULD use the same
   Call-ID, at least within the same boot cycle. For these requests, it
   makes no difference whether the Call-ID value matches an existing
   call or not.


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



        callid   =  token [ "@" token ]
        Call-ID  =  ( "Call-ID" | "i" ) ":" callid


   The callid MUST be a globally unique identifier and MUSTNOT be reused
   for later calls. Use of cryptographically random identifiers [29] is
   RECOMMENDED. Implementations MAY use the form "localid@host". Call-
   IDs are case-sensitive and are simply compared byte-by-byte.


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

   For systems which have tight bandwidth constraints, many of the
   mandatory SIP headers have a compact form, as discussed in Section 9.
   These are alternate names for the headers which occupy less space in
   the message. In the case of Call-ID, the compact form is i.

   For example, both of the following are valid:

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


   or

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



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6.14 Call-Info

   The Call-Info general header field provides additional information
   about the caller or callee, depending on whether it is found in a
   request or response. The purpose of the URI is described by the
   "purpose" parameter. "icon" designates an image suitable as an iconic
   representation of the caller or callee; "info" describes the caller
   or callee in general, e.g., through a web page; "card" provides a
   business card (e.g., in vCard [30] or LDIF [31] formats).



        Call-Info   =  "Call-Info" ":" # ( "<" URI ">" *( ";" info-param) )
        info-param  =  "purpose" "=" ( "icon" | "info" | "card" | token )
                   |   generic-param


   Example:

   Call-Info: <http://wwww.example.com/alice/photo.jpg> ;purpose=icon,
     <http://www.example.com/alice/> ;purpose=info



6.15 Contact

   The Contact general-header field can appear in INVITE, OPTIONS, ACK,
   and REGISTER requests, and in 1xx, 2xx, 3xx, and 485 responses. In
   general, it provides a URL where the user can be reached for further
   communications.

   In some of the cases below, the client uses information from the
   Contact header field in Request-URI of future requests. In these
   cases, the client copies all but the "method-param" and "header"
   elements of the addr-spec part of the Contact header field into the
   Request-URI of the request. It uses the "header" parameters to create
   headers for the request, replacing any default headers normally used.
   Unless the client is configured to use a default proxy for all
   outgoing requests, it then directs the request to the address and
   port specified by the "maddr" and "port" parameters, using the
   transport protocol given in the "transport" parameter. If "maddr" is
   a multicast address, the value of "ttl" is used as the time-to-live
   value.

        INVITE, OPTIONS and ACK requests: INVITE requests MUST and ACK
             requests MAY contain Contact headers indicating from which
             location the request is originating. The URL in the Contact
             header field is then used by subsequent requests from the



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             callee. For OPTIONS, Contact provides a hint where future
             SIP requests can be sent or the user can be contacted via
             non-SIP means.


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

        INVITE 1xx responses: A UAS sending a provisional response (1xx)
             MAY insert a Contact response header. It has the same
             semantics in a 1xx response as a 2xx INVITE response. Note
             that CANCEL requests MUSTNOT be sent to that address, but
             rather follow the same path as the original request.

        INVITE and OPTIONS 2xx responses: A user agent server sending a
             definitive, positive response (2xx) MUST insert a Contact
             response header field indicating the SIP address under
             which it is reachable most directly for future SIP
             requests, such as ACK, within the same Call-ID. The Contact
             header field contains the address of the server itself or
             that of a proxy, e.g., if the host is behind a firewall.
             The value of this Contact header is copied into the
             Request-URI of subsequent requests for this call if the
             response did not also contain a Record-Route header. If the
             response also contains a Record-Route header field, the
             address in the Contact header field is added as the last
             item in the Route header field. See Section 6.35 for
             details.

             If a UA supports both UDP and TCP, it SHOULDNOT indicate a
             transport parameter in the URI.


             The Contact value SHOULDNOT be cached across calls, as
             it may not represent the most desirable location for a
             particular destination address.

        REGISTER requests and responses: See Section 4.2.6.

        3xx and 485 responses: The Contact response-header field can be
             used with a 3xx or 485 (Ambiguous) response codes to
             indicate one or more alternate addresses to try. It can
             appear in responses to BYE, INVITE and OPTIONS methods. The
             Contact header field contains URIs giving the new locations
             or user names to try, or may simply specify additional
             transport parameters. A 300 (Multiple Choices), 301 (Moved



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             Permanently), 302 (Moved Temporarily) or 485 (Ambiguous)
             response SHOULD contain a Contact field containing URIs of
             new addresses to be tried. A 301 or 302 response may also
             give the same location and username that was being tried
             but specify additional transport parameters such as a
             different server or multicast address to try or a change of
             SIP transport from UDP to TCP or vice versa.  The client
             copies information from the Contact header field into the
             Request-URI as described above.

        4xx, 5xx and 6xx responses: The Contact response-header field
             can be used with a 4xx, 5xx or 6xx response to indicate the
             location where additional information about the error can
             be found.

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

   A Contact response header field can contain any suitable URI
   indicating where the called party can be reached, not limited to SIP
   URLs. For example, it could contain URL's for phones, fax, or irc (if
   they were defined) or a mailto: (RFC 2368, [32]) URL.

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

        q: The "qvalue" indicates the relative preference among the
             locations given. "qvalue" values are decimal numbers from 0
             to 1, with higher values indicating higher preference. The
             default value is 0.5.

        action: The "action" parameter is used only when registering
             with the REGISTER request. It indicates whether the client
             wishes that the server proxy or redirect future requests
             intended for the client. If this parameter is not specified
             the action taken depends on server configuration. In its
             response, the registrar SHOULD indicate the mode used. This
             parameter is ignored for other requests.

        expires: The "expires" parameter indicates how long the URI is
             valid. The parameter is either a number indicating seconds
             or a quoted string containing a SIP-date. If this parameter
             is not provided, the value of the Expires header field
             determines how long the URI is valid. Implementations MAY
             treat values larger than 2**32-1 (4294967295 seconds or 136
             years) as equivalent to 2**32-1.



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   Contact = ( "Contact" | "m" ) ":"
             ("*" | (1# (( name-addr | addr-spec )
             *( ";" contact-params ) )))

   name-addr      = [ display-name ] "<" addr-spec ">"
   addr-spec      = SIP-URL | URI
   display-name   = *token | quoted-string

   contact-params = "q"       "=" qvalue
                  | "action"  "=" "proxy" | "redirect"
                  | "expires" "=" delta-seconds | <"> SIP-date <">
                  | contact-extension

   contact-extension = generic-param
   qvalue            = ( "0" [ "." 0*3DIGIT ] )
                     | ( "1" [ "." 0*3("0") ] )


   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, semicolon or question mark.
   Note that there may or may not be LWS between the display-name and
   the "<".


        The Contact header field fulfills functionality similar to
        the Location header field in HTTP. However, the HTTP header
        only allows one address, unquoted. Since URIs can contain
        commas and semicolons as reserved characters, they can be
        mistaken for header or parameter delimiters, respectively.
        The current syntax corresponds to that for the To and From
        header, which also allows the use of display names.

   Example:


     Contact: "Mr. Watson" <sip:watson@worcester.bell-telephone.com>
        ;q=0.7; expires=3600,
        "Mr. Watson" <mailto:watson@bell-telephone.com> ;q=0.1



6.16 Content-Disposition



        Content-Disposition   =  "Content-Disposition" ":"
                                 disposition-type *( ";" disposition-param )
        disposition-type      =  "render" | "session" | "icon" | "alert"



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                             |   disp-extension-token
        disposition-param     =  "handling" "="
                                 ( "optional" | "required" | other-handling )
                             |   generic-param
        other-handling        =  token
        disp-extension-token  =  token


   The Content-Disposition header field describes how the message body
   or, in the case of multipart messages, a message body part is to be
   interpreted by the UAC or UAS. The SIP header extends the MIME
   Content-Type (RFC 1806 [33]).

   The value "session" indicates that the body part describes a session,
   for either calls or early (pre-call) media. The value "render"
   indicates that the body part should be displayed or otherwise
   rendered to the user. For backward-compatibility, if the Content-
   Disposition header is not missing, bodies of Content-Type
   application/sdp imply the disposition "session", while other content
   types imply "render".

   The disposition type "icon" indicates that the body part contains an
   image suitable as an iconic representation of the caller or callee.
   The value "alert" indicates that the body part contains information,
   such as an audio clip, that should be rendered instead of ring tone.

   The handling parameter, handling-parm, describes how the UAS should
   react if it receives a message body whose content type or disposition
   type it does not understand. If the parameter has the value
   "optional", the UAS MUST ignore the message body; if it has the value
   "required", the UAS MUST return 415 (Unsupported Media Type).  If the
   handling parameter is missing, the value "required" is to be assumed.

   If this header field is missing, the MIME type determines the default
   content disposition. If there is none, "render" is assumed.

6.17 Content-Encoding



        Content-Encoding  =  ( "Content-Encoding" | "e" ) ":"
                             1#content-coding


   The Content-Encoding entity-header field is used as a modifier to the
   "media-type". When present, its value indicates what additional
   content codings have been applied to the entity-body, and thus what
   decoding mechanisms MUST be applied in order to obtain the media-type



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   referenced by the Content-Type header field.  Content-Encoding is
   primarily used to allow a body to be compressed without losing the
   identity of its underlying media type.

   If multiple encodings have been applied to an entity, the content
   codings MUST be listed in the order in which they were applied.

   All content-coding values are case-insensitive. The Internet Assigned
   Numbers Authority (IANA) acts as a registry for content-coding value
   tokens. See [H3.5] for a definition of the syntax for content-coding.

   Clients MAY apply content encodings to the body in requests. If the
   server is not capable of decoding the body, or does not recognize any
   of the content-coding values, it MUST send a 415 "Unsupported Media
   Type" response, listing acceptable encodings in the Accept-Encoding
   header. A server MAY apply content encodings to the bodies in
   responses. The server MUST only use encodings listed in the Accept-
   Encoding header in the request.

6.18 Content-Language

   See [H14.12].

6.19 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" | "l" ) ":" 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. (The size of the message-body does not include the CRLF
   separating headers and body.) 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 field MUST be set to zero.  If a
   server receives a datagram request without Content-Length, it MUST
   assume that the request encompasses the remainder of the packet. If a
   server receives a datagram request with a Content-Length, but the



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   value differs from the size of the body sent in the request, the
   server SHOULD return a 400 (Bad Request) response.

   If a response does not contain a Content-Length, the client assumes
   that it encompasses the remainder of the datagram packet or the data
   until the stream connection is closed, as applicable.  Section 8
   describes how to determine the length of the message body.


        The ability to omit Content-Length simplifies the creation
        of cgi-like scripts that dynamically generate responses.

6.20 Content-Type

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



        Content-Type  =  ( "Content-Type" | "c" ) ":" media-type


   Examples of this header field are

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



6.21 CSeq

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



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   The ACK and CANCEL requests MUST contain the same CSeq value as the
   INVITE request that it refers to, while a BYE request cancelling an
   invitation MUST have a higher sequence number. A BYE request with a
   CSeq that is not higher should cause a 400 response to be generated.

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

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



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



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

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

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

   Forked requests MUST have the same CSeq as there would be ambiguity



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   otherwise between these forked requests and later BYE issued by the
   client user agent.

   Example:


     CSeq: 4711 INVITE



6.22 Date

   Date is a general-header field. Its syntax is:



        Date      =  "Date" ":" SIP-date
        SIP-date  =  rfc1123-date


   See [H14.18] for a definition of rfc1123-date. Note that unlike
   HTTP/1.1, SIP only supports the most recent RFC 1123 [34] formatting
   for dates.  As in [H3.3], SIP restricts the timezone in SIP-date to
   "GMT", while RFC 1123 allows any timezone.

        The consistent use of GMT between Date, Expires and Retry-
        After headers allows implementation of simple clients that
        do not have a notion of absolute time.  Note that rfc1123-
        date is case-sensitive.

   The Date header field reflects the time when the request or response
   is first sent. Thus, retransmissions have the same Date header field
   value as the original.

   Registrars MUST include this header in REGISTER responses if they use
   absolute expiration times and SHOULD include it for all responses.


        The Date header field can be used by simple end systems
        without a battery-backed clock to acquire a notion of
        current time. However, in its GMT-form, it requires clients
        to know their offset from GMT.

6.23 Encryption

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



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   to-end encryption of requests and responses. Requests are encrypted
   based on the public key belonging to the entity named in the To
   header field. Responses are encrypted based on the public key
   conveyed in the Response-Key header field. Note that the public keys
   themselves may not be used for the encryption. This depends on the
   particular algorithms used.

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


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



        Encryption         =  "Encryption" ":" encryption-scheme 1*SP
                              #encryption-params
        encryption-scheme  =  token
        encryption-params  =  generic-param

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

   The example in Figure 9 shows a message encrypted with ASCII-armored
   PGP that was generated by applying "pgp -ea" to the payload to be
   encrypted.



   Figure 9: PGP Encryption Example



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



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   INVITE sip:watson@boston.bell-telephone.com SIP/2.0
   Via: SIP/2.0/UDP 169.130.12.5
   From: <sip:a.g.bell@bell-telephone.com>
   To: T. A. Watson <sip:watson@bell-telephone.com>
   Call-ID: 187602141351@worcester.bell-telephone.com
   Cseq: 1 INVITE
   Content-Length: 829
   Encryption: PGP version=2.6.2,encoding=ascii

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

   otherwise identical un-encrypted request.

6.24 Error-Info

   The Error-Info response header provides a pointer to additional
   information about the error status response. This header field is
   only contained in 3xx, 4xx, 5xx and 6xx responses.



        Error-Info  =  "Error-Info" ":" # ( "<" URI ">" *( ";" generic-param ))


6.25 Expires

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

   This header field is currently defined only for the REGISTER and
   INVITE methods. For REGISTER, it is a request and response-header
   field. In a REGISTER request, the client indicates how long it wishes
   the registration to be valid. In the response, the server indicates
   the earliest expiration time of all registrations. The server MAY



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   choose a shorter time interval than that requested by the client, but
   SHOULDNOT choose a longer one.  If a registration updates an existing
   registration, the Expires value of the most recent registration is
   used, even if it is shorter than the earlier registration.

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

   Note that the expiration time does not affect the duration of the
   actual session that may result from the invitation. Session
   description protocols may offer the ability to express time limits on
   the session duration, however.

   The value of this field can be either a SIP-date or an integer number
   of seconds (in decimal), measured from the receipt of the request.
   The latter approach is preferable for short durations, as it does not
   depend on clients and servers sharing a synchronized clock.
   Implementations MAY treat values larger than 2**32-1 (4294967295 or
   136 years) as equivalent to 2**32-1.



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


   Two examples of its use are

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



6.26 From

   Requests and responses MUST contain a From general-header field,
   indicating the initiator of the request.  (Note that this may be
   different from the initiator of the call leg. Requests sent by the
   callee to the caller use the callee's address in the From header
   field.)  The From field MAY contain the "tag" parameter.  The server
   copies the From header field from the request to the response. The



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   optional "display-name" is meant to be rendered by a human-user
   interface. A system SHOULD use the display name "Anonymous" if the
   identity of the client is to remain hidden.

   The SIP-URL MUSTNOT contain the "transport-param", "maddr-param",
   "ttl-param", or "headers" elements. A server that receives a SIP-URL
   with these elements ignores them.

   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, question mark, or
   semicolon.  Syntax issues are discussed in Section 6.5.



        From        =  ( "From" | "f" ) ":" ( name-addr | addr-spec )
                       *( ";" from-param )
        from-param  =  tag-param | generic-param
        tag-param   =  "tag" "=" token


   Examples:


     From: "A. G. Bell" <sip:agb@bell-telephone.com> ;tag=a48s
     From: sip:+12125551212@server.phone2net.com
     From: Anonymous <sip:c8oqz84zk7z@privacy.org>



   The "tag" MAY appear in the From field of a request. It MUST be
   present when it is possible that two instances of a user sharing a
   SIP address can make call invitations with the same Call-ID.

   The "tag" value MUST be globally unique and cryptographically random
   with at least 32 bits of randomness.  The UA SHOULD use different
   tags for From and To header fields, but use the same pair of tags at
   least within the same Call-ID. It is RECOMMENDED to maintain the same
   tag pair across calls and instances of the UA application.


        Maintaining the same tag pair allow restarting of a user
        agent within interrupting existing calls. Using different
        tag values for From and To header fields simplifies users
        calling themselves.

   For the purpose of identifying call legs, two From or To header
   fields are equal if and only if:




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        o The addr-spec component is equal, according to the rules in
          Section 2.1.

        o Any "tag" and "generic-param" parameters are equal, compared
          according to the case-sensitivity rules in Section 6. Only
          parameters that appear in both header fields are compared.


        Call-ID, To and From are needed to identify a call leg.
        The distinction between call and call leg matters in calls
        with multiple responses to a forked request. The format is
        similar to the equivalent RFC 822 [26] header, but with a
        URI instead of just an email address.

6.27 In-Reply-To

   The In-Reply-To request header field enumerates the call-IDs that
   this call references or returns.


        This allows automatic call distribution systems to route
        return calls to the originator of the first call and allows
        callees to filter calls, so that only calls that return
        calls they have originated will be accepted. This field is
        not a substitute for request authentication.



        In-Reply-To  =  "In-Reply-To" ":" 1# callid


   Example:

   In-Reply-To: 70710@saturn.bell-tel.com, 17320@saturn.bell-tel.com



6.28 Max-Forwards

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



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



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

   Each proxy or gateway recipient of a request containing a Max-
   Forwards header field MUST check and update its value prior to
   forwarding the request. If the received value is zero (0), the
   recipient MUSTNOT forward the request and returns 483 (Too many
   hops). Instead, a server MAY act as a final recipient for OPTIONS
   requests. It is RECOMMENDED that the server include Supported, Server
   and Allow header fields in the response.

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

   Example:

     Max-Forwards: 6



6.29 MIME-Version

   See [H19.4.1].

6.30 Organization

   The Organization general-header field conveys the name of the
   organization to which the entity issuing the request or response
   belongs. It MAY also be inserted by proxies at the boundary of an
   organization.


        The field MAY be used by client software to filter calls.



        Organization  =  "Organization" ":" TEXT-UTF8-TRIM


6.31 Priority

   The Priority request-header field indicates the urgency of the
   request as perceived by the client.



        Priority        =  "Priority" ":" priority-value



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        priority-value  =  "emergency" | "urgent" | "normal"
                        |  "non-urgent" | other-priority
        other-priority  =  token


   It is RECOMMENDED that the value of "emergency" only be used when
   life, limb or property are in imminent danger.

   Examples:


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

     Subject: Weekend plans
     Priority: non-urgent




        These are the values of RFC 2076 [35], with the addition of
        "emergency".

6.32 Proxy-Authenticate

   The Proxy-Authenticate response-header field MUST be included as part
   of a 407 (Proxy Authentication Required) response.  It may also occur
   in a 401 (Unauthorized) response if the request was forked.  The
   field value consists of a challenge that indicates the authentication
   scheme and parameters applicable to the proxy for this Request-URI.

   Unlike its usage within HTTP, the Proxy-Authenticate header MUST be
   passed upstream in the response to the UAC. In SIP, only UAC's can
   authenticate themselves to proxies.

   The syntax for this header is defined in [H14.33]. See 14 for further
   details on its usage.

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





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

6.33 Proxy-Authorization

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

   Unlike Authorization, the Proxy-Authorization header field applies
   only to the next outbound proxy that demanded authentication using
   the Proxy- Authenticate field. When multiple proxies are used in a
   chain, the Proxy-Authorization header field is consumed by the first
   outbound proxy that was expecting to receive credentials. A proxy MAY
   relay the credentials from the client request to the next proxy if
   that is the mechanism by which the proxies cooperatively authenticate
   a given request.

   See [H14.34] for a definition of the syntax, and section 14 for a
   discussion of its usage.

6.34 Proxy-Require

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

   See Section 6.36 for more details on the mechanics of this message
   and a usage example.



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


6.35 Record-Route

6.35.1 Operation

   The Record-Route request and response header field is added to a
   request by any proxy that insists on being in the path of subsequent



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   requests for the same call leg.  A proxy SHOULD add it to any request
   for robustness, but a request route, once established, persists until
   the end of the call leg, regardless of whether the Record-Route
   header is present in subsequent requests.

   The Record-Route header field contains a globally reachable SIP-URI
   that identifies the proxy server, including an address parameter
   (maddr) that identifies its location. Each such proxy server adds the
   Request-URI of the incoming request to the beginning of the list.
   Requests between both user agents involved in the call leg, in either
   direction, traverse this route.


        Some proxies, such as those controlling firewalls or in an
        automatic call distribution (ACD) system, need to maintain
        call state and thus need to receive any BYE, re-INVITE and
        ACK packets for the call. Note that proxy servers have to
        add Record-Route headers to each request as long as they
        want to be "visited" by the next request for the call leg.

   Proxies MUST include an maddr parameter in the URI in the Record-
   Route header, but MUSTNOT include a transport parameter. If the
   Request-URI did not contain a port number and the server port number
   differs from the default, the proxy MUST add a port number to the
   hostport portion of the Request-URI.


        If it is important that all requests go to the same host,
        server administrators are advised to be careful in
        selecting the appropriate name or address to ensure that
        name resolution does indeed resolve to the same host. For
        example, a domain name having an SRV record may resolve to
        a different network addresses on each attempt.  Inclusion
        of, say, a TCP transport parameter may prevent a UA that
        supports only UDP but reached the proxy inserting the
        Record-Route via another proxy from reaching this proxy.

   The UAS copies the Record-Route request header field unchanged into
   the response. (Record-Route is only relevant for 2xx responses and
   responses where the server can expect the client to retry for the
   same Call-Id, as in 401 (Unauthorized) or 484 (Address Incomplete).)

6.35.2 Construction of Route Header

   Once a proxy P inserts a Record-Route header in a request from UA A
   to UA B, all subsequent requests from A to B and from B to A visit P.

   A UA builds the Route header field for subsequent requests from the



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   Record-Route header fields received in either a response or a
   request.

   If a UAC finds a Record-Route header in a final response, it copies
   it, including all parameters, into Route header fields of all
   subsequent requests within the same call leg, reversing the order of
   fields, so that the first entry is the server closest to the UAC. If
   the response contained a Contact header field, the user agent adds
   its content as the last Route header.

   If a UAS finds a Record-Route header in a request, it copies the
   Record-Route maddr parameters as well as other Record-Route
   parameters and any port value, maintaining their ordering, to the
   Route header field of future requests issued as a UAC. Since the URIs
   contained in the Record-Route header fields are not useful for the
   reverse request path, the UA fills all other components of the Route
   name-addr value with the name-addr value found in the Contact or the
   From header field. The latter is used only if there is no Contact
   header field. All URI parameters in the Contact or From header field
   are copied.

   If the request featured a Contact header field, the Contact header
   value is appended to the Route header list.

6.35.3 Request Destination

   Unless this would cause a loop, any client, including the UAC, SHOULD
   send the next request for this call leg to the first Request-URI in
   the Route request header field. A client MAY forward the request to a
   designated proxy instead, for example, if it lacks DNS resolution
   capability. If a client uses the first Route entry to route the
   request, it removes it.

6.35.4 Syntax

   The Record-Route header field has the following syntax:


        Record-Route  =  "Record-Route" ":" 1# ( name-addr *( ";" rr-param ))
        rr-param      =  generic-param


   Proxy servers MUST include their address in a "maddr" URL parameter
   to ensure that subsequent requests are guaranteed to reach exactly
   the same server.

6.35.5 Example




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   Example for a request where the proxy servers ieee.org and bell-
   telephone.com , in that order, insist on being part of subsequent
   request paths:

     Record-Route: <sip:a.g.bell@bell-telephone.com;maddr=s.bell-telephone.com>,
       <sip:a.bell@ieee.org;maddr=199.172.136.40>



6.36 Require

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



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


   Example:

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

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




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

   Proxy and redirect servers MUST ignore features that are not
   understood. If a particular extension requires that intermediate



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   devices support it, the extension MUST be tagged in the Proxy-Require
   field as well (see Section 6.34).

6.37 Response-Key

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



        Response-Key  =  "Response-Key" ":" key-scheme 1*SP #key-param
        key-scheme    =  token
        key-param     =  generic-param


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

   If the client insists that the server return an encrypted response,
   it includes a

                  Require: org.ietf.sip.encrypt-response

   header field in its request. If the server cannot encrypt for
   whatever reason, it MUST follow normal Require header field
   procedures and return a 420 (Bad Extension) response. If this Require
   header field is not present, a server SHOULD still encrypt if it can.

6.38 Retry-After

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

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

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



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   service is assumed to be available indefinitely.



        Retry-After  =  "Retry-After" ":" ( SIP-date | delta-seconds )
                        [ comment ] *( ";" retry-param )
        retry-param  =  "duration" "=" delta-seconds
                    |   generic-param


   Examples of its use are

     Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting)
     Retry-After: Mon, 01 Jan 9999 00:00:00 GMT
       (Dear John: Don't call me back, ever)
     Retry-After: Fri, 26 Sep 1997 21:00:00 GMT;duration=3600
     Retry-After: 120



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

6.39 Route

   The Route request-header field determines the route taken by a
   request. Each host removes the first entry and then proxies the
   request to the host listed in that entry, also using it as the
   Request-URI. The operation is described in more detail in Section
   6.35.

   The Route header field has the following syntax:


        Route  =  "Route" ":" 1# ( name-addr *( ";" rr-param ))


6.40 Server

   The Server response-header field contains information about the
   software used by the user agent server to handle the request. The
   syntax for this field is defined in [H14.38].

6.41 Subject

   This header field provides a summary or indicates the nature of the
   call, allowing call filtering without having to parse the session
   description. (Note that the session description does not have to use



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   the same subject indication as the invitation.)



        Subject  =  ( "Subject" | "s" ) ":" TEXT-UTF8-TRIM


   Example:


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



6.42 Supported

   The Supported general-header field enumerates all the capabilities of
   the client or server. This header field SHOULD be included in all
   requests (except ACK) and in all responses.


        Including the header field in all responses greatly
        simplifies the use of extensions for call control in
        subsequent transactions with the same server.

   Syntax:


        Supported  =  ( "Supported" | "k" ) ":" 1#option-tag


6.43 Timestamp

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



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





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   Note that there MUSTNOT be any LWS between a DIGIT and the decimal
   point.

6.44 To

   The To general-header field specifies the "logical" recipient of the
   request.



        To        =  ( "To" | "t" ) ":" ( name-addr | addr-spec )
                     *( ";" to-param )
        to-param  =  tag-param | generic-param


   Requests and responses MUST contain a To general-header field,
   indicating the desired recipient of the request. The optional
   "display-name" is meant to be rendered by a human-user interface. The
   UAS or redirect server copies the To header field into its response,
   and MUST add a "tag" parameter.


        If there was more than one Via header field, the request
        was handled by at least one proxy server. Since the
        receiver cannot know whether any of the proxy servers
        forked the request, it is safest to assume that they might
        have.

   The SIP-URL MUSTNOT contain the "transport-param", "maddr-param",
   "ttl-param", or "headers" elements. A server that receives a SIP-URL
   with these elements removes them before further processing.

   The "tag" parameter serves as a general mechanism to distinguish
   multiple instances of a user identified by a single SIP URL. As
   proxies can fork requests, the same request can reach multiple
   instances of a user (mobile and home phones, for example). As each
   can respond, there needs to be a means to distinguish the responses
   from each at the caller. The situation also arises with multicast
   requests. The tag in the To header field serves to distinguish
   responses at the UAC. It MUST be placed in the To field of the
   response by user agent, registrar and redirect servers, but MUSTNOT
   be inserted into responses forwarded upstream by proxies. However,
   responses generated locally by a proxy, and then sent upstream, MUST
   contain a tag.

   A UAS or redirect server MUST add a "tag" parameter for all final
   responses for all transactions within a call leg. All such parameters
   have the same value within the same call leg. These servers SHOULD



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   add the tag for informational responses during the initial INVITE
   transaction, but MUST add a tag to informational responses for all
   subsequent transactions.

   See Section 6.26 for details of the "tag" parameter. The "tag"
   parameter in To headers is ignored when matching responses to
   requests that did not contain a "tag" in their To header.

   Section 11 describes when the "tag" parameter MUST appear in
   subsequent requests. Note that if a request already contained a tag,
   this tag MUST be mirrored in the response; a new tag MUSTNOT be
   inserted.

   Section 6.26 describes how To and From header fields are compared for
   the purpose of matching requests to call legs.

   UAS SHOULD accept requests even if they do not recognize the URI
   scheme (e.g., a tel: URI) or if the To header does not address the
   user. Only Request-URI that do not match the recipient should cause
   requests to be rejected.

   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, question mark, or
   semicolon.  Note that LWS is common, but not mandatory between the
   display-name and the "<".

   The following are examples of valid To headers:

     To: The Operator <sip:operator@cs.columbia.edu>;tag=287447
     To: sip:+12125551212@server.phone2net.com




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

6.45 Unsupported

   The Unsupported response-header field lists the features not
   supported by the server. See Section 6.36 for a usage example and



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   motivation.

   Syntax:


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


6.46 User-Agent

   The User-Agent general-header field contains information about the
   client user agent originating the request. The syntax and semantics
   are defined in [H14.42].

6.47 Via

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

6.47.1 Requests

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

   A proxy SHOULD check the top-most Via header field to ensure that it
   contains the sender's correct network address, as seen from that
   proxy. If the sender's address is incorrect, the proxy MUST add an
   additional "received" attribute, as described in Section 6.47.2.


        A multi-homed host may not be able to insert a network
        address into the Via header field that can be reached by
        the next hop, for example because if one of the networks is
        private. The address placed into the Via header may differ
        from the interface actually used, as that interface is
        selected only at packet sending time by the IP layer.




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   A client that sends a request to a multicast address MUST add the
   "maddr" parameter to its Via header field, and SHOULD add the "ttl"
   parameter. (In that case, the maddr parameter SHOULD contain the
   destination multicast address, although under exceptional
   circumstances it MAY contain a unicast address.) If a server receives
   a request which contained an "maddr" parameter in the topmost Via
   field, it SHOULD send the response to the address listed in the
   "maddr" parameter.

   Loop detection is described in Section 12.3.1.

6.47.2 Receiver-tagged Via Header Fields

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

   An example is:


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



   In this example, the message originated from 10.0.0.1 and traversed a
   NAT with the external address border.ieee.org (199.172.136.3) to
   reach erlang.bell-telephone.com.  The latter noticed the mismatch,
   and added a parameter to the previous hop's Via header field,
   containing the address that the packet actually came from. (Note that
   the NAT border.ieee.org is not a SIP server.)

6.47.3 Responses

   Via header fields in responses are processed by a proxy or UAC
   according to the following rules:

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



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        2.   If there is no second Via header field, this response is
             destined for this client. Otherwise, use this Via field as
             the destination, as described in Section 6.47.5.

6.47.4 User Agent and Redirect Servers

   A UAS or redirect server copies the Via header fields into the
   response, without changing their order, and uses the top (first) Via
   element as the destination, as described in the next section.

6.47.5 Forwarding Responses

   Given a destination described by a Via header field, the response is
   sent according to the following rules:

        o If the "sent-protocol" is a reliable transport protocol such
          as TCP, TLS or SCTP, send the response using the existing TCP
          connection to the source of the original request.

        o Otherwise, if the Via header field contains a "maddr"
          parameter, forward the response to the address listed there,
          using the port indicated in "sent-by", or port 5060 if none is
          present. If the address is a multicast address, the response
          SHOULD be sent using the TTL indicated in the "ttl" parameter,
          or with a TTL of 1 if that parameter is not present.

        o Otherwise, if it is a receiver-tagged field (Section 6.47.2),
          send the response to the address in the "received" parameter,
          using the port indicated in the "sent-by" value, or using port
          5060 if none is specified explicitly.

        o Otherwise, if it is not receiver-tagged, send the response to
          the address indicated by the "sent-by" value in the second Via
          header field.

   Note that the response to an unreliable datagram request is not
   returned to the port from which the request came.

6.47.6 Syntax

   The format for a Via header field is shown in Fig. 10. The "maddr"
   parameter, designating the multicast address, and the "ttl"
   parameter, designating the time-to-live (TTL) value, are included
   only if the request was sent via multicast. The "received" parameter
   is added only for receiver-added Via fields (Section 6.47.2).


   The "branch" parameter is included by every proxy. The token MUST be



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  Via              = ( "Via" | "v") ":" 1#( sent-protocol sent-by
                     *( ";" via-params ) [ comment ] )
  via-params       = via-hidden | via-ttl | via-maddr
                   | via-received | via-branch | via-extension
  via-hidden       = "hidden"
  via-ttl          = "ttl" "=" ttl
  via-maddr        = "maddr" "=" host
  via-received     = "received" "=" host
  via-branch       = "branch" "=" token
  via-extension    = generic-param
  sent-protocol    = protocol-name "/" protocol-version "/" transport
  protocol-name    = "SIP" | token
  protocol-version = token
  transport        = "UDP" | "TCP" | token
  sent-by          = host [ ":" port ]


   Figure 10: Syntax of Via header field


   unique for each distinct request. The precise format of the token is
   implementation-defined. In order to be able to both detect loops and
   associate responses with the corresponding request, the parameter
   SHOULD consist of two parts separable by the implementation. One
   part, used for loop detection (Section 12.3.1), MAY be computed as a
   cryptographic hash of the To, From, Call-ID header fields, the
   Request-URI of the request received (before translation) and the
   sequence number from the CSeq header field. The algorithm used to
   compute the hash is implementation-dependent, but MD5 [36], expressed
   in hexadecimal, is a reasonable choice. (Note that base64 is not
   permissible for a token.) The other part, used for matching responses
   to requests, is a globally unique function of the branch taken, for
   example, a hash of a sequence number, local IP address and request-
   URI of the request sent on the branch.

   For example: 7a83e5750418bce23d5106b4c06cc632.1


        The "branch" parameter MUST depend on the incoming
        request-URI to distinguish looped requests from requests
        whose request-URI is changed and which then reach a server
        visited earlier.

   CANCEL requests MUST have the same branch value as the corresponding
   forked request. When a response arrives at the proxy it can use the
   branch value to figure out which branch the response corresponds to.



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     Via: SIP/2.0/UDP first.example.com:4000;ttl=16
       ;maddr=224.2.0.1 ;branch=a7c6a8dlze.1 (Acme server)
     Via: SIP/2.0/UDP adk8



6.48 Warning

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



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


   A response MAY carry more than one Warning header.

   The "warn-text" should be in a natural language that is most likely
   to be intelligible to the human user receiving the response.  This
   decision can be based on any available knowledge, such as the
   location of the cache or user, the Accept-Language field in a
   request, or the Content-Language field in a response. The default
   language is i-default [37].

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

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

   The warn-code consists of three digits. A first digit of "3"
   indicates warnings specific to SIP.



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   This is a list of the currently-defined "warn-code"s, each with a
   recommended warn-text in English, and a description of its meaning.
   Note that these warnings describe failures induced by the session
   description.

   Warnings 300 through 329 are reserved for indicating problems with
   keywords in the session description, 330 through 339 are warnings
   related to basic network services requested in the session
   description, 370 through 379 are warnings related to quantitative QoS
   parameters requested in the session description, and 390 through 399
   are miscellaneous warnings that do not fall into one of the above
   categories.

        300 Incompatible network protocol: One or more network protocols
             contained in the session description are not available.

        301 Incompatible network address formats: One or more network
             address formats contained in the session description are
             not available.

        302 Incompatible transport protocol: One or more transport
             protocols described in the session description are not
             available.

        303 Incompatible bandwidth units: One or more bandwidth
             measurement units contained in the session description were
             not understood.

        304 Media type not available: One or more media types contained
             in the session description are not available.

        305 Incompatible media format: One or more media formats
             contained in the session description are not available.

        306 Attribute not understood: One or more of the media
             attributes in the session description are not supported.

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

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

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

        370 Insufficient bandwidth: The bandwidth specified in the



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             session description or defined by the media exceeds that
             known to be available.

        399 Miscellaneous warning: The warning text can include
             arbitrary information to be presented to a human user, or
             logged. A system receiving this warning MUSTNOT take any
             automated action.


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

   Additional "warn-code"s, as in the example below, can be defined
   through IANA.

   Examples:


     Warning: 307 isi.edu "Session parameter 'foo' not understood"
     Warning: 301 isi.edu "Incompatible network address type 'E.164'"



6.49 WWW-Authenticate

   The WWW-Authenticate response-header field MUST be included in 401
   (Unauthorized) response messages. The field value consists of at
   least one challenge that indicates the authentication scheme(s) and
   parameters applicable to the Request-URI. See [H14.46] for a
   definition of the syntax, and Section 14 for an overview of usage.

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

7 Status Code Definitions

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

7.1 Informational 1xx

   Informational responses indicate that the server or proxy contacted



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   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. A server SHOULD send a 1xx response if it expects to take
   more than 200 ms to obtain a final response. A server MAY issue zero
   or more 1xx responses, with no restriction on their ordering or
   uniqueness. Note that 1xx responses are not transmitted reliably,
   that is, they do not cause the client to send an ACK. Servers are
   free to retransmit informational responses and clients can inquire
   about the current state of call processing by re-sending the request.

   Informational (1xx) responses other than 100 (Trying) MAY contain
   message bodies, including session descriptions. If a 1xx response
   contains a session description, a UAC SHOULD cease generating local
   ringback tone. Session descriptions in 1xx responses are interpreted
   in the same manner as those in 2xx responses. In particular, the
   session description MUST be formatted in such a way that it would be
   valid in a 2xx response. Thus, the UAS can only include a session
   description in its provisional response if the UAC has included one
   in an earlier INVITE. (SIP extensions may specify additional
   circumstances where session descriptions may be included.) If a later
   provisional response or 2xx contains a different session description,
   this new description is treated as if it were the original response
   to the session description in the INVITE.

   The UAS can remove the media stream by setting the port number to
   zero in a subsequent session description contained in a provisional
   response and thus restore normal ringback behavior. The UAS cannot
   add media streams beyond those offered by the UAC in the INVITE. A
   provisional response without a session description has no effect on
   any early media that have already been set up.

   The media streams are assumed to be bidirectional unless marked as
   send-only or receive-only. For SDP, this is described in Section B.
   Client behavior when receiving several different session descriptions
   from different branches is undefined.

7.1.1 100 Trying

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

7.1.2 180 Ringing

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




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7.1.3 181 Call Is Being Forwarded

   A proxy server MAY use this status code to indicate that the call is
   being forwarded to a different set of destinations.

7.1.4 182 Queued

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

7.1.5 183 Session Progress

   The 183 (Session Progress) response is used to convey information
   about the progress of the call which is not otherwise classified. The
   Reason-Phrase MAY be used to convey more details about the call
   progress.

7.2 Successful 2xx

   The request was successful and MUST terminate a search.

7.2.1 200 OK

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

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

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

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

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

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

7.3 Redirection 3xx

   3xx responses give information about the user's new location, or



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   about alternative services that might be able to satisfy the call.
   They SHOULD terminate an existing search, and MAY cause the initiator
   to begin a new search if appropriate.

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

7.3.1 300 Multiple Choices

   The address in the request resolved to several choices, each with its
   own specific location, and the user (or user agent) can select a
   preferred communication end point and redirect its request to that
   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, if allowed by the Accept request
   header. The entity format is specified by the media type given in the
   Content-Type header field. The choices SHOULD also be listed as
   Contact fields (Section 6.15).  Unlike HTTP, the SIP response MAY
   contain several Contact fields or a list of addresses in a Contact
   field. User agents MAY use the Contact header field value for
   automatic redirection or MAY ask the user to confirm a choice.
   However, this specification does not define any standard for such
   automatic selection.


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

7.3.2 301 Moved Permanently

   The user can no longer be found at the address in the Request-URI and
   the requesting client SHOULD retry at the new address given by the
   Contact header field (Section 6.15). The caller SHOULD update any
   local directories, address books and user location caches with this
   new value and redirect future requests to the address(es) listed.

7.3.3 302 Moved Temporarily

   The requesting client SHOULD retry the request at the new address(es)
   given by the Contact header field (Section 6.15).  The Request-URI of
   the new request uses the value of the Contact header in the response.
   The new request can take two different forms. In the first approach,
   the To, From, Call-ID, and CSeq header fields in the new request are
   the same as in the original request, with a new branch identifier in



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   the Via header field. Proxies MUST follow this behavior and UACs MAY.
   UAs MAY also use the Contact information for the To header field, as
   well as a new Call-ID value.


        Reusing the CSeq value allows proxies to avoid forwarding
        the request to the same destination twice, as a proxy will
        consider it a retransmission.

   The duration of the redirection can be indicated through an Expires
   (Section 6.25) header. If there is no explicit expiration time, the
   address is only valid for this call and MUSTNOT be cached for future
   calls.

7.3.4 305 Use Proxy

   The requested resource MUST be accessed through the proxy given by
   the Contact field. The Contact field gives the URI of the proxy. The
   recipient is expected to repeat this single request via the proxy.
   305 responses MUST only be generated by user agent servers.

7.3.5 380 Alternative Service

   The call was not successful, but alternative services are possible.
   The alternative services are described in the message body of the
   response.  Formats for such bodies are not defined here, and may be
   the subject of future standardization.

7.4 Request Failure 4xx

   4xx responses are definite failure responses from a particular
   server.  The client SHOULDNOT retry the same request without
   modification (e.g., adding appropriate authorization). However, the
   same request to a different server might be successful.

7.4.1 400 Bad Request

   The request could not be understood due to malformed syntax.  The
   Reason-Phrase SHOULD identify the syntax problem in more detail,
   e.g., "Missing Call-ID header".

7.4.2 401 Unauthorized

   The request requires user authentication.  This response is issued by
   user agent servers and registrars, while 407 (Proxy Authentication
   Required) is used by proxy servers.

7.4.3 402 Payment Required



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   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 SHOULDNOT be repeated.

7.4.5 404 Not Found

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

7.4.6 405 Method Not Allowed

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

7.4.7 406 Not Acceptable

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

7.4.8 407 Proxy Authentication Required

   This code is similar to 401 (Unauthorized), but indicates that the
   client MUST first authenticate itself with the proxy. The proxy MUST
   return a Proxy-Authenticate header field (section 6.32) 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 6.33). SIP access authentication is explained
   in section 13.2 and 14.

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

7.4.9 408 Request Timeout

   The server could not produce a response within a suitable amount of
   time, for example, since it could not determine the location of the
   user in time. The amount of time may have been indicated in the
   Expires request-header field or may be set by the server. The client
   MAY repeat the request without modifications at any later time.



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7.4.10 409 Conflict

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

7.4.11 410 Gone

   The requested resource is no longer available at the server and no
   forwarding address is known. This condition is expected to be
   considered permanent. If the server does not know, or has no facility
   to determine, whether or not the condition is permanent, the status
   code 404 (Not Found) SHOULD be used instead.

7.4.12 411 Length Required

   The server refuses to accept the request without a defined Content-
   Length. The client MAY repeat the request if it adds a valid
   Content-Length header field containing the length of the message-body
   in the request message.

7.4.13 413 Request Entity Too Large

   The server is refusing to process a request because the request
   entity is larger than the server is willing or able to process. The
   server MAY close the connection to prevent the client from continuing
   the request.

   If the condition is temporary, the server SHOULD include a Retry-
   After header field to indicate that it is temporary and after what
   time the client MAY try again.

7.4.14 414 Request-URI Too Long

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

7.4.15 415 Unsupported Media Type

   The server is refusing to service the request because the message
   body of the request is in a format not supported by the server for
   the requested method. The server SHOULD return a list of acceptable
   formats using the Accept, Accept-Encoding and Accept-Language header
   fields.

7.4.16 420 Bad Extension




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   The server did not understand the protocol extension specified in a
   Proxy-Require (Section 6.34) or Require (Section 6.36) header field.

7.4.17 480 Temporarily Unavailable

   The callee's end system was contacted successfully but the callee is
   currently unavailable (e.g., not logged in, logged in in such a
   manner as to preclude communication with the callee or activated the
   "do not disturb" feature). The response MAY indicate a better time to
   call in the Retry-After header. The user could also be available
   elsewhere (unbeknownst to this host), thus, this response does not
   terminate any searches. The reason phrase SHOULD indicate a more
   precise cause as to why the callee is unavailable. This value SHOULD
   be setable by the user agent. Status 486 (Busy Here) MAY be used to
   more precisely indicate a particular reason for the call failure.

   This status is also returned by a redirect server that recognizes the
   user identified by the Request-URI, but does not currently have a
   valid forwarding location for that user.

7.4.18 481 Call Leg/Transaction Does Not Exist

   This status is returned under three conditions: The server received a
   BYE request that does not match any existing call leg, the server
   received a CANCEL request that does not match any existing
   transaction or the server received an INVITE with a To tag that does
   not match the local tag value. (A server simply discards an ACK
   referring to an unknown transaction.)

7.4.19 482 Loop Detected

   The server received a request with a Via (Section 6.47) path
   containing itself.

7.4.20 483 Too Many Hops

   The server received a request that contains a Max-Forwards (Section
   6.28) header with the value zero.

7.4.21 484 Address Incomplete

   The server received a request with a To (Section 6.44) address or
   Request-URI that was incomplete. Additional information SHOULD be
   provided.


        This status code allows overlapped dialing. With overlapped
        dialing, the client does not know the length of the dialing



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        string. It sends strings of increasing lengths, prompting
        the user for more input, until it no longer receives a 484
        status response.

7.4.22 485 Ambiguous

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

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

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

   485 Ambiguous SIP/2.0
   Contact: Carol Lee <sip:carol.lee@example.com>
   Contact: Ping Lee <sip:p.lee@example.com>
   Contact: Lee M. Foote <sip:lee.foote@example.com>




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

7.4.23 486 Busy Here

   The callee's end system was contacted successfully but the callee is
   currently not willing or able to take additional calls at this end
   system. The response MAY indicate a better time to call in the
   Retry-After header. The user could also be available elsewhere, such
   as through a voice mail service, thus, this response does not
   terminate any searches. Status 600 (Busy Everywhere) SHOULD be used
   if the client knows that no other end system will be able to accept
   this call.

7.4.24 487 Request Terminated

   The request was terminated by a BYE or CANCEL request. This response
   is never returned for a CANCEL request itself.



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7.4.25 488 Not Acceptable Here

   The response has the same meaning as 606 (Not Acceptable), but only
   applies to the specific entity addressed by the Request-URI and the
   request may succeed elsewhere.

7.5 Server Failure 5xx

   5xx responses are failure responses given when a server itself has
   erred. They are not definitive failures, and MUSTNOT 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. The client MAY display the specific error
   condition, and MAY retry the request after several seconds.

   If the condition is temporary, the server MAY indicate when the
   client may retry the request using the Retry-After header.

7.5.2 501 Not Implemented

   The server does not support the functionality required to fulfill the
   request. This is the appropriate response when a UAS does not
   recognize the request method and is not capable of supporting it for
   any user. (Proxies forward all requests regardless of method.)

7.5.3 502 Bad Gateway

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

7.5.4 503 Service Unavailable

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

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

7.5.5 504 Server Time-out



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   The server did not receive a timely response from the server (e.g., a
   location server) it accessed in attempting to process the request.
   Note that 408 (Request Timeout) should be used if there was no
   response within the period specified in the Expires header field from
   the upstream server.

7.5.6 505 Version Not Supported

   The server does not support, or refuses to support, the SIP protocol
   version that was used in the request message. The server is
   indicating that it is unable or unwilling to complete the request
   using the same major version as the client, other than with this
   error message. The response MAY contain an entity describing why that
   version is not supported and what other protocols are supported by
   that server. The format for such an entity is not defined here and
   may be the subject of future standardization.

7.6 Global Failures 6xx

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

7.6.1 600 Busy Everywhere

   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 uses status code 603 (Decline) instead. This status response
   is returned only if the client knows that no other end point (such as
   a voice mail system) will answer the request. Otherwise, 486 (Busy
   Here) should be returned.

7.6.2 603 Decline

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

7.6.3 604 Does Not Exist Anywhere

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

7.6.4 606 Not Acceptable



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

   A 606 (Not Acceptable) response means that the user wishes to
   communicate, but cannot adequately support the session described. The
   606 (Not Acceptable) response MAY contain a list of reasons in a
   Warning header field describing why the session described cannot be
   supported. Reasons are listed in Section 6.48.  It is hoped that
   negotiation will not frequently be needed, and when a new user is
   being invited to join an already existing conference, negotiation may
   not be possible. It is up to the invitation initiator to decide
   whether or not to act on a 606 (Not Acceptable) response.

8 SIP Message Body

8.1 Body Inclusion

   Requests MAY contain message bodies unless otherwise noted. In this
   specification, the CANCEL request MUSTNOT contain a message body.

   The use of message bodies for REGISTER requests is for further study.

   For response messages, the request method and the response status
   code determine the type and interpretation of any message body. All
   responses MAY include a body. Message bodies for 1xx responses
   contain advisory information about the progress of the request.  1xx
   responses to INVITE requests MAY contain session descriptions.  Their
   interpretation depends on the response status code, but generally
   informs the caller what kind of session the callee is likely to
   establish, subject to later modification in the 2xx response.
   Request methods not defined in this specification MAY also contain
   session descriptions.  2xx responses to INVITE requests contain
   session descriptions. In 3xx responses, the message body MAY contain
   the description of alternative destinations or services, as described
   in Section 7.3. For responses with status 400 or greater, the message
   body MAY contain additional, human-readable information about the
   reasons for failure. It is RECOMMENDED that information in 1xx and
   300 and greater responses be of type text/plain or text/html

8.2 Message Body Type

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



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   The "multipart" MIME type [38] MAY be used within the body of the
   message. Clients that send requests containing multipart message
   bodies MUST be able to send a session description as a non-multipart
   message body if the server requests this through an Accept header
   field.

8.3 Message Body Length

   The body length in bytes SHOULD be given by the Content-Length header
   field. Section 6.19 describes the behavior in detail.

   The "chunked" transfer encoding of HTTP/1.1 MUSTNOT be used for SIP.
   (Note: The chunked encoding modifies the body of a message in order
   to transfer it as a series of chunks, each with its own size
   indicator.)

9 Compact Form

   When SIP is carried over UDP with authentication and a complex
   session description, it may be possible that the size of a request or
   response is larger than the MTU. To address this problem, a more
   compact form of SIP is also defined by using abbreviations for the
   common header fields listed below:


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


   Thus, the message in section 16.2 could also be written:


     INVITE sip:bob@example.com SIP/2.0
     v:SIP/2.0/UDP 131.215.131.131;maddr=239.128.16.254;ttl=16
     v:SIP/2.0/UDP 216.112.6.38
     f:sip:alice@wonderland.com
     t:sip:bob@example.com
     m:sip:alice@mouse.wonderland.com
     i:62729-27@216.112.6.38



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     c:application/sdp
     CSeq: 4711 INVITE
     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
     a=rtpmap:0 PCMU/8000



   Clients MAY mix short field names and long field names within the
   same request. Servers MUST accept both short and long field names for
   requests. Proxies MAY change header fields between their long and
   short forms, but this MUSTNOT be done to fields following an
   Authorization header.

10 Behavior of SIP Clients and Servers

10.1 Multicast Unreliable Transport Protocols

   Requests MAY be multicast; multicast requests likely feature a host-
   independent Request-URI. This request SHOULD be scoped to ensure it
   is not forwarded beyond the boundaries of the administrative scope.
   This MAY be done with either TTL or administrative scopes [27],
   depending on what is implemented in the network.

   A client receiving a multicast query does not have to check whether
   the host part of the Request-URI matches its own host or domain name.
   If the request was received via multicast, the response MUST be
   returned to the address listed in the maddr parameter of the Via
   header field. (This parameter is REQUIRED.) Generally, this will be a
   multicast address. Such multicast responses are multicast with the
   same TTL as the request, where the TTL is derived from the ttl
   parameter in the Via header (Section 6.47).

   To avoid response implosion, servers MUSTNOT answer multicast
   requests with a status code other than 2xx, 401, 407, 484 or 6xx. The
   server delays its response by a random interval uniformly distributed
   between zero and one second. Servers MAY suppress responses if they
   hear a lower-numbered or 6xx response from another group member prior
   to sending. Servers do not respond to CANCEL requests received via
   multicast to avoid request implosion. A proxy or UAC SHOULD send a



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   CANCEL on receiving the first 2xx, 401, 407 or 6xx response to a
   multicast request.


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

10.1 Reliable Transport Protocols

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

   The client SHOULD keep the connection open at least until the first
   final response arrives. If the client closes or resets the connection
   prior to receiving the first final response, the server treats this
   action as equivalent to a CANCEL request for all pending requests
   using that connection.


        This behavior makes it less likely that malfunctioning
        clients cause a proxy server to keep connection state
        indefinitely.

   The server SHOULDNOT close the 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 closes the connection prematurely, the
   client SHOULD interpret this as being equivalent to a 500 (Server
   Internal Error) response.

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



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

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

10.2 Reliability for Requests Other Than INVITE

10.2.1 Unreliable Transport Protocols

   A SIP client using an unreliable transport protocol such as UDP
   SHOULD retransmit requests other than INVITE or ACK with an
   exponential backoff, starting at a T1 second interval, doubling the
   interval for each packet, and capping off at a T2 second interval.
   This means that after the first packet is sent, the second is sent T1
   seconds later, the next 2*T1 seconds after that, the next 4*T1
   seconds after that, and so on, until the interval reaches T2.
   Subsequent retransmissions are spaced by T2 seconds. If the client
   receives a provisional response, it continues to retransmit the
   request, but with an interval of T2 seconds. Retransmissions cease
   when the client has sent a total of eleven packets, or receives a
   definitive response. Default values for T1 and T2 are 500 ms and 4 s,
   respectively. Clients MAY use larger values, but SHOULDNOT use
   smaller ones. Servers retransmit the response upon receipt of a
   request retransmission. 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 10*T2 seconds to avoid having to, for
   example, contact the user or location server again upon receiving a
   request retransmission.


        Use of the exponential backoff is for congestion control
        purposes. However, the back-off must cap off, since request
        retransmissions are used to trigger response
        retransmissions at the server. Without a cap, the loss of a
        single response could significantly increase transaction
        latencies.

   The value of the initial retransmission timer is smaller than that
   that for TCP since it is expected that network paths suitable for
   interactive communications have round-trip times smaller than 500 ms.
   For congestion control purposes, the retransmission count has to be
   bounded.  Given that most transactions are expected to consist of one
   request and a few responses, round-trip time estimation is not likely



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   to be very useful. If RTT estimation is desired to more quickly
   discover a missing final response, each request retransmission needs
   to be labeled with its own Timestamp (Section 6.43), returned in the
   response. The server caches the result until it can be sure that the
   client will not retransmit the same request again.

   Each server in a proxy chain generates its own final response to a
   CANCEL request. The server responds immediately upon receipt of the
   CANCEL request rather than waiting until it has received final
   responses from the CANCEL requests it generates.

   BYE and OPTIONS final responses are generated by redirect and user
   agent servers; REGISTER final responses are generated by registrars.
   Note that in contrast to the reliability mechanism described in
   Section 10.3, responses to these requests are not retransmitted
   periodically and not acknowledged via ACK.

10.2.2 Reliable Transport Protocol

   Clients using a reliable transport protocol such as TCP, SCTP or TLS
   do not need to retransmit requests, but MAY give up after receiving
   no response for an extended period of time.

10.3 Reliability for INVITE Requests

   Special considerations apply for the INVITE method.

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

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

        3.   The client has to be able to terminate an on-going request,



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             e.g., because it is no longer willing to wait for the
             connection or search to succeed. The server will have to
             wait several retransmission intervals to interpret the lack
             of request retransmissions as the end of a call. If the
             call succeeds shortly after the caller has given up, the
             callee will "pick up the phone" and not be "connected".

10.3.1 Unreliable Transport Protocols

   A SIP client using an unreliable transport protocol SHOULD retransmit
   a SIP INVITE request with an interval that starts at T1 seconds, and
   doubles after each packet transmission. The client ceases
   retransmissions if it receives a provisional or definitive response,
   or once it has sent a total of seven request packets. A UAC MAY send
   a BYE or CANCEL request after the seventh retransmission. It is
   RECOMMENDED to send both. (This avoids call establishment in case the
   network path loses packets asymmetrically.)

   A server which transmits a provisional response should retransmit it
   upon reception of a duplicate request. A server which transmits a
   final response should retransmit it with an interval that starts at
   T1 seconds, and doubles for each subsequent packet until it reaches
   T2 seconds.  Response retransmissions cease when an ACK request is
   received or the response has been transmitted seven times. The value
   of a final response is not changed by the arrival of a BYE or CANCEL
   request.

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

   Fig. 11 and 12 show the client and server state diagram for INVITE
   transactions. The "terminated" event occurs if the server receives
   either a CANCEL or BYE request. Note that the state diagram only
   shows the behavior for the INVITE transaction; the responses for BYE
   and CANCEL are not shown and follow the rules laid in Section 10.2.





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              +===========+
              *           *
              *  Initial  *
              *           *
              +===========+
                    |
                    |    -
                    |  INVITE
                    |
                    v
              *************
    T1*2^n <--*           *
    INVITE -->*  Calling  *--------+
              *           *        |
              *************        |
                :   |              |
  ..............:   | 1xx      xxx |
  : 7 INVITE sent   |  -       ACK |
  :                 |              |
  :                 v              |
  :           *************        |
  :           *   Call    *        |
  :           * proceeding*<->1xx  |
  :           *           *        |
  :           *************        |
  :                 |              |
  :                 |<-------------+
  :..............   |
                .   v
              *************
      xxx  <--*           *
      ACK  -->* Completed *
              *           *
              *************

 event (xxx=status)
     message


   Figure 11: State transition diagram of client for INVITE method




        The mechanism in Sec. 10.2 would not work well for INVITE
        because of the long delays between INVITE and a final



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                      +===============+
                      *               *
                      *   Initial     *
                      *               *
                      +===============+
                              !
                              ! INVITE/1xx
                              !
                              !
                              v
                      *****************
             |------->*               *<----------|
        INVITE/1xx    *  Proceeding   *    status change/1xx
             ^--------*               *-----------^
;;;;;;;;;;;;;;;;;;;;;;*****************
; terminated/487            !   !
:                           !   !
;                           !   !
;            failure/>=300  !   !  picks up/2xx
;             +-------------+   +-----------+
;             v                             v
;        ***********                   ***********
;INVITE/>*         *<-min(T1*2^n,T2)/->*         *<-----|
;status <* failure *->    status     <-* success *  INVITE/2xx
;        *         *                   *         *------^
;;;;;;;;;***********                   ***********
              !                             !
              !                             !
              !                             !
              !                             !
              +--------------+--------------+
event/message sent           ! ACK/-
                             v
                     *****************
                |--->*               *
              ACK/-  *   Confirmed   *
                ^--->*               *
                     *****************


   Figure 12: State transition diagram of server for INVITE method


        response. If the 200 response were to get lost, the callee
        would believe the call to exist, but the voice path would
        be dead since the caller does not know that the callee has



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        picked up. Thus, the INVITE retransmission interval would
        have to be on the order of a second or two to limit the
        duration of this state confusion. Retransmitting the
        response with an exponential back-off helps ensure that the
        response is received, without placing an undue burden on
        the network.

10.3.2 Reliable Transport Protocol

   but uses the same algorithm as for unreliable transport protocols
   (Section 10.3.1) to retransmit responses until it receives an ACK. A
   client MAY give up on the request if there is no response within a
   client-defined timeout interval.


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

10.4 ICMP Handling

   Handling of ICMP messages in the case of unreliable transport
   protocol messages is straightforward. For requests, a host, network,
   port, or protocol unreachable error SHOULD be treated as if a 400-
   class response was received. For responses, these errors SHOULD cause
   the server to cease retransmitting the response.

   Source quench ICMP messages SHOULD be ignored. TTL exceeded errors
   SHOULD be ignored. Parameter problem errors SHOULD be treated as if a
   400-class response was received.

11 Behavior of SIP User Agents

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

11.1 Caller Issues Initial INVITE Request

   When a user agent client desires to initiate a call, it formulates an
   INVITE request. The To field in the request contains the address of
   the callee, and remains unaltered as the request traverses proxies.
   The Request-URI contains the same address, but may be rewritten by
   proxies. The From field contains the address of the caller. If the
   From address can appear in requests generated by other user agent
   clients for the same call, the caller MUST insert the tag parameter



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   in the From field. A UAC MUST add a Contact header containing an
   address where it would like to be contacted for transactions from the
   callee back to the caller.

11.2 Callee Issues Response

   When the initial INVITE request is received at the callee, the callee
   can accept, redirect, or reject the call. In all of these cases, it
   formulates a response. The response MUST copy the To, From, Call-ID,
   CSeq and Via fields from the request. Additionally, the responding
   UAS MUST add the tag parameter to the To field in the response. Since
   a request from a UAC may fork and arrive at multiple hosts, the tag
   parameter serves to distinguish, at the UAC, multiple responses from
   different UAS's.

   The UAS MUST add a Contact header field in the response. It contains
   an address where the callee would like to be contacted for subsequent
   transactions, including the ACK for the current INVITE.  The UAS
   stores the values of the To and From field, including any tags. These
   become the local and remote addresses of the call leg, respectively.

11.3 Caller Receives Response to Initial Request

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

   If the 200 (OK) response contained a Contact header field, it is
   copied into the Request-URI of the ACK or BYE requests. Otherwise,
   the To header field address is copied into the Request-URI.

   The UAC also notes the value of the To and From header fields in each
   response. For each call leg, the To header field becomes the remote
   address, and the From header field becomes the local address.

11.4 Caller or Callee Generate Subsequent Requests

   Once the call has been established, either the caller or callee MAY
   generate INVITE or BYE requests to change or terminate the call.
   Regardless of whether the caller or callee is generating the new
   request, the header fields in the request are set as follows. For the
   desired call leg, the To header field is set to the remote address,



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   and the From header field is set to the local address (both including
   any tags). A UAC copies the tag from the final response into the ACK,
   but it MUSTNOT copy the tag into any subsequent requests unless the
   response was a 200-class response to an INVITE request. The To field
   of CANCEL requests always contain exactly the same value as the
   request it is cancelling.

   The Contact header field MAY be different than the Contact header
   field sent in a previous response or request.  The Request-URI MAY be
   set to the value of the Contact header field received in a previous
   request or response from the remote party.

   The callee's requests use the caller's To header field value as the
   From header value and the From header field value as the To header
   field value.

   The network destination and Request-URI of requests is determined
   according to the following rules:

        o If the response from the previous request contained a Record-
          Route header field, the UAC sends the request to the last
          entry in the list and removes that entry. As described in
          Section 6.35, the Request-URI is set to that value.

        o Otherwise, if the response for the previous request contained
          a Contact header field, the request is directed to the host
          and port identified there. The Request-URI is set to the value
          of the Contact header. The request does not contain a Route
          header field in this case.

        o Otherwise, the Request-URI contains the same URL as the To
          header.

   If the UAC is configured with the address of an outbound proxy
   server, the UAC sends the request there, independent of the Request-
   URI. The outbound proxy is NOT named in the Request-URI. If there is
   no outbound proxy server, the Request-URI determines the network
   destination.

   If a UAC does not support DNS resolution or the full Record-
   Route/Route mechanism, it MAY send all requests to a locally
   configured outbound proxy. In that case, that proxy behaves as
   described above. The UAC MUST, however, perform the mapping of
   Record-Route to Route header fields and MUST include all Route header
   fields, i.e., the UAC does not remove the first Route header field.

11.5 Receiving Subsequent Requests




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   When a request is received during a call, the following checks are
   made:

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

             It is possible that the To header in an INVITE request has
             a tag, but the UAS believes this to be a new call. This
             will occur if the UAS crashed and rebooted in the middle of
             a call, and the UAC has sent what it believes to be a re-
             INVITE. The UAS MAY either accept or reject the request.
             Accepting the request provides robustness, so that calls
             can persist even through crashes. UAs wishing to support
             this capability must choose monotonically increasing CSeq
             numbers even across reboots. This is because subsequent
             requests from the crashed-and-rebooted UA towards the other
             UA need to have a CSeq number higher than previous requests
             in that direction.

             Note also that the crashed-and-rebooted UA will have lost
             any Route headers which would need to be inserted into a
             subsequent request. Therefore, it is possible that the
             requests may not be properly forwarded by proxies.


             RTP media agents allowing restarts need to be robust
             by accepting out-of-range timestamps and sequence
             numbers.

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

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

12 Behavior of SIP Proxy and Redirect Servers

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

12.1 Redirect Server




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

12.2 User Agent Server

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

12.3 Proxy Server

   This section outlines processing rules for proxy servers. A proxy
   server can either be stateful or stateless. When stateful, a proxy
   remembers the incoming request which generated outgoing requests, and
   the outgoing requests. A stateless proxy forgets all information once
   an outgoing request is generated. A forking proxy SHOULD be stateful.
   Proxies that accept TCP connections MUST be stateful when handling
   the TCP connection.


        Otherwise, if the proxy were to lose a request, the TCP
        client would never retransmit it.

   A stateful proxy SHOULDNOT become stateless until after it sends a
   definitive response upstream, and at least 32 seconds after it
   received a definitive response.

   A stateful proxy acts similar to a virtual UAS/UAC, but cannot be
   viewed as just a UAS and UAC glued together at the back. (In
   particular, it does not originate requests except ACK and CANCEL.)
   It implements the server state machine when receiving requests, and
   the client state machine for generating outgoing requests, with the
   exception of receiving a 2xx response to an INVITE. Instead of
   generating an ACK, the 2xx response is always forwarded upstream
   towards the caller. Furthermore, ACK's for 200 responses to INVITE's
   are always proxied downstream towards the UAS, as they would be for a
   stateless proxy.

   A stateless proxy forwards every request it receives downstream, and
   every response it receives upstream.

12.3.1 Proxying Requests




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   A proxy server MUST check for forwarding loops before proxying a
   request. A request has been looped if the server finds its own
   address in the Via header field and the hash computation over the
   fields enumerated in Section 6.47.6 yields the same value as the hash
   part of the "branch" parameter in the Via entry containing the proxy
   server's address.

   A proxy server MUSTNOT forward a request to a multicast group which
   already appears in any of the Via headers.

   The proxy server MUST copy all request header fields to the outgoing
   request. It MAY add other header fields.

   A proxy server always inserts a Via header field containing its own
   address into those requests that are caused by an incoming request.
   Each proxy MUST insert a "branch" parameter (Section 6.47).

   Proxies other than outbound proxies SHOULD change the Request-URI to
   indicate the server where it intends to send the request.

12.3.2 Proxying Responses

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

12.3.3 Stateless Proxy: Proxying Responses

   A stateless proxy removes its own Via field, and checks the address
   in the next Via field. In the case of UDP, the response is sent to
   the address listed in the "maddr" parameter if present, otherwise to
   the "received" tag if present, and finally to the address in the
   "sent-by" field.  A proxy MUST remain stateful when handling requests
   received via TCP.

   A stateless proxy MUSTNOT generate its own provisional responses.

12.3.4 Stateful Proxy: Receiving Requests

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

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



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12.3.5 Stateful Proxy: Receiving ACKs

   When an ACK request is received, it is proxied unless the request's
   To (including the tag), From, CSeq and Call-ID header fields match
   those of a (non-2xx) response sent by the proxy. In that case, the
   request is processed locally and stops retransmissions of responses.

12.3.6 Stateful Proxy: Receiving Responses

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

   A stateful proxy should obey the rules in Section 12.4 to determine
   if the response should be proxied upstream. If it is to be proxied,
   the same rules for stateless proxies above are followed, with the
   following addition for TCP. If a request was received via TCP
   (indicated by the protocol in the top Via header), the proxy checks
   to see if it has a connection currently open to that address. If so,
   the response is sent on that connection.  Otherwise, a new TCP
   connection is opened to the address and port in the Via field, and
   the response is sent there. Note that this implies that a UAC or
   proxy MUST be prepared to receive responses on the incoming side of a
   TCP connection. Definitive non 200-class responses MUST be
   retransmitted by the proxy, even over a TCP connection.

12.3.7 Stateless, Non-Forking Proxy

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

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

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



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   The server will treat it as a new request and commence another
   search.

12.4 Forking Proxy

   The server must respond to the request (other than ACK) immediately
   with a 100 (Trying) response if it expects to take more than 200 ms
   to obtain a final response.

   Successful responses to an INVITE request MAY contain a Contact
   header field so that the following ACK or BYE bypasses the proxy
   search mechanism. If the proxy requires future requests to be routed
   through it, it adds a Record-Route header to the request (Section
   6.35).

   The following C-code describes the behavior of a proxy server issuing
   several requests in response to an incoming INVITE request with
   method R which is to be proxied to a list of N destination enumerated
   in ' address .I expires

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


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

     process_request(Method R, int N, address_t address[], int expires)
     {
       struct {
         char *branch;         /* branch token */
         int branch_seq;       /* branch sequence number part */
         int done;             /* has responded */
       } outgoing[];
       char *location[];       /* list of locations */
       int heard = 0;          /* number of sites heard from */
       int class;              /* class of status code */
       int timeleft = expires; /* expiration value */



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       int loc = 0;            /* number of locations */
       struct {                /* response */
         int status;           /* response: CANCEL=-1 */
         int locations;        /* number of redirect locations */
         char *location[];     /* redirect locations */
         address_t a;          /* address of respondent */
         char *branch;         /* branch token */
         int branch_seq;       /* branch sequence number */
       } r, best;              /* response, best response */
       int i;

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

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

         /* If final response, mark branch as done. */
         if (class >= 2) {
           heard++;
           for (i = 0; i < N; i++) {
             if (r.branch_seq == outgoing[i].branch_seq) {
               outgoing[i].done = 1;
               break;
             }
           }
         }
         /* CANCEL: respond, fork and wait for responses */
         /* terminate INVITE with 40
         else if (class < 0) {
           best.status = 200;
           response(best);
           for (i = 0; i < N; i++) {
             if (!outgoing[i].done)
               request(CANCEL, address[i], outgoing[i].branch);
           }
           best.status = -1;
         }

         /* Send an ACK */
         if (class != 2) {
           if (R == INVITE) request(ACK, r.a, r.branch);



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         }

         if (class == 2) {
           if (r.status < best.status) best = r;
           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 += r.locations;
             for (i = 0; i < r.locations; i++) {
               request(R, r.location[i]);
             }
           } else if (!ismulticast(r.location)) {
             best = r;
           }
         }
         else if (class == 4) {
           if (best.status >= 400) best = r;
         }
         else if (class == 5) {
           if (best.status >= 500) best = r;
         }
         else if (class == 6) {
           best = r;
           break;
         }
       }

       /* We haven't heard anything useful from anybody. */
       if (best.status == 1000) {
         best.status = 408; /* request expired */
       }
       if (best.status/100 != 3) loc = 0;
       response(best);
     }



   Responses are processed as follows. The process completes (and state
   can be freed) when all requests have been answered by final status



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   responses (for unicast) or 60 seconds have elapsed (for multicast). A
   proxy MAY send a CANCEL to all incomplete branches and return the
   best available final status to the client if not all responses have
   been received after 60 seconds or the expiration period specified in
   the Expires header field of the request. If no responses have been
   received, the proxy returns a 408 (Timeout) response to the client.

   When forwarding responses, a proxy MUST forward the whole response,
   including all header fields of the selected response as well as the
   body.

        1xx: The proxy SHOULD forward provisional responses greater than
             100 upstream towards the client and SHOULDNOT forward 100
             (Trying) responses.

        2xx: If the request was an INVITE, the proxy MUST forward the
             response upstream towards the client, without sending an
             ACK downstream. For other requests, it should only forward
             the response upstream if it has not forwarded any other
             responses upstream.

             After receiving a 2xx, the server MAY terminate all other
             pending requests by sending a CANCEL request and closing
             the TCP connection, if applicable. (Terminating pending
             requests is advisable as searches consume resources. Also,
             INVITE requests could "ring" on a number of workstations if
             the callee is currently logged in more than once.)

             If the request was not an INVITE, the proxy SHOULD drop 2xx
             responses if it had already forwarded a final response
             upstream.

        3xx: For INVITE requests, the proxy MUST send an ACK.  It MAY
             recurse on the listed Contact addresses. Otherwise, the
             lowest-numbered response is returned if there were no 2xx
             or 6xx responses.

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

        4xx, 5xx: For INVITE requests, the proxy MUST send an ACK. It
             remembers the response if it has a lower status code class
             than any previous 4xx and 5xx response. On completion, a
             response with the lowest response class is returned if
             there were no 2xx, 3xx or 6xx responses. Within the set of
             responses from the lowest-numbered class, the proxy server



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             may choose any response.

             The proxy SHOULD collect all WWW-Authenticate and Proxy-
             Authenticate headers from all 401 and 407 responses and
             return all of them in the response if either 401 or 407 is
             the lowest-numbered response.

        6xx: For INVITE requests, the proxy sends an ACK. It forwards
             the 6xx response unless a 2xx response has been received.
             Other pending requests MAY be terminated with CANCEL as
             described for 2xx responses. Unlike for 2xx responses, only
             one 6xx response is forwarded, since ACKs are generated
             locally.

   A proxy server forwards any response for Call-IDs for which it does
   not have a pending transaction according to the response's Via
   header. User agent servers respond to BYE requests for unknown call
   legs with status code 481 (Transaction Does Not Exist); they drop ACK
   requests with unknown call legs silently.

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

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

13 SIP Authentication using HTTP Basic and Digest Schemes

   SIP implementations MAY use HTTP's basic and digest authentication
   mechanisms (RFC 2617 [41]) to provide a rudimentary form of security.
   This section overviews usage of these mechanisms in SIP. The basic
   operation is almost completely identical to that for HTTP [41]. This
   section outlines this operation, pointing to RFC 2617 [41] for
   details, and noting the differences when used in SIP.

13.1 Framework

   The framework for SIP authentication parallels that for HTTP (RFC
   2617 [41]). In particular, the BNF for auth-scheme, auth-param,
   challenge, realm, realm-value, and credentials is identical. The 401
   response is used by user agent servers in SIP to challenge the
   authorization of a user agent client. Additionally, registrars and
   redirect servers MAY make use of 401 responses for authorization, but
   proxies MUSTNOT, and instead MAY use the 407 response. The



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   requirements for inclusion of the Proxy-Authenticate, Proxy-
   Authorization, WWW-Authenticate, and Authorization in the various
   messages is identical to RFC 2617 [41].

   Since SIP does not have the concept of a canonical root URL, the
   notion of protections spaces are interpreted differently for SIP. The
   realm is a protection domain for all SIP URIs with the same value for
   the userinfo, host and port part of the SIP Request-URI. For example:


      INVITE sip:alice.wonderland@example.com SIP/2.0
      WWW-Authenticate:  Basic realm="business"



   and


      INVITE sip:aw@example.com SIP/2.0
      WWW-Authenticate: Basic realm="business"



   define different protection realms according to this rule.

   When a UAC resubmits a request with its credentials after receiving a
   401 or 407 response, it MUST increment the CSeq header field as it
   would normally do when sending an updated request.

13.2 Basic Authentication

   The rules for basic authentication follow those defined in [41] but
   with the words "origin server" replaced with "user agent server,
   redirect server , or registrar".

   Since SIP URIs are not hierarchical, the paragraph in [41] that
   states that "all paths at or deeper than the depth of the last
   symbolic element in the path field of the Request-URI also are within
   the protection space specified by the Basic realm value of the
   current challenge" does not apply for SIP. SIP clients MAY
   preemptively send the corresponding Authorization header with
   requests for SIP URIs within the same protection realm (as defined
   above) without receipt of another challenge from the server.

13.3 Digest Authentication

   The rules for digest authentication follow those defined in [41],
   with "HTTP 1.1" replaced by "SIP/2.0" in addition to the following



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   differences:

        1.   The URI included in the challenge has the following BNF:


             URI  =  SIP-URL


        2.   The BNF in RFC 2617 has an error in that the URI is not
             enclosed in quotation marks. (The example in Section 3.5 is
             correct.) For SIP, the URI MUST be enclosed in quotation
             marks.

        3.   The BNF for digest-uri-value is:


             digest-uri-value  =  Request-URI ; as defined in Section
             4.3


        4.   The example procedure for choosing a nonce based on Etag
             does not work for SIP.

        5.   The Authentication-Info and Proxy-Authentication-Info
             fields are not used in SIP.

        6.   The text in RFC 2617 [41] regarding cache operation does
             not apply to SIP.

        7.   RFC 2617 [41] requires that a server check that the URI in
             the request line, and the URI included in the Authorization
             header, point to the same resource. In a SIP context, these
             two URI's may actually refer to different users, due to
             forwarding at some proxy.  Therefore, in SIP, a server MAY
             check that the request-uri in the Authorization header
             corresponds to a user that the server is willing to accept
             forwarded or direct calls for.

13.4 Proxy-Authentication

   The use of the Proxy-Authentication and Proxy-Authorization parallel
   that as described in [41], with one difference. Proxies MUSTNOT add
   the Proxy-Authorization header. 407 (Proxy Authentication Required)
   responses MUST be forwarded upstream towards the client following the
   procedures for any other response. It is the client's responsibility
   to add the Proxy-Authorization header containing credentials for the
   proxy which has asked for authentication.




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        If a proxy were to resubmit a request with a Proxy-
        Authorization header field, it would need to increment the
        CSeq in the new request. However, this would mean that the
        UAC which submitted the original request would discard a
        response from the UAS, as the CSeq value would be
        different.

   See sections 6.32 and 6.33 for additional information on usage of
   these fields as they apply to SIP.

   It is also possible that a 401 (Unauthorized) response contains
   several challenges, from a mixture of proxies and user agent servers,
   if the request was forked.

14 SIP Security Using PGP

14.1 PGP Authentication Scheme

   The "pgp" authentication scheme is based on the model that the client
   authenticates itself with a request signed with the client's private
   key. The server can then ascertain the origin of the request if it
   has access to the public key, preferably signed by a trusted third
   party.  Implementations supporting this scheme MUST implement the
   definitions and default algorithms of RFC 2440 [42] and MAY implement
   the older version, based upon PGP 2.6, described in RFC 1991 [39].

14.1.1 The WWW-Authenticate Response Header



        WWW-Authenticate  =  "WWW-Authenticate" ":" "pgp" pgp-challenge
        pgp-challenge     =  1# pgp-params
        pgp-params        =  realm | pgp-version | pgp-micalgorithm
                         |   pgp-pubalgorithm | nonce
        realm             =  "realm" "=" realm-value
        realm-value       =  quoted-string
        pgp-version       =  "version" "="
                             <"> digit *( "." digit ) *alpha <">
        pgp-micalgorithm  =  "algorithm" "=" ( "md5" | "sha1" | token
                             | "ripemd160" | "MD2" | "TIGER192" | "HAVAL-5-160")
        pgp-pubalgorithm  =  "pubkey" "=" ( "rsa" | "rsa-encrypt"
                             | "rsa-sign" | "elgamal" | "dsa" | token )
        nonce             =  "nonce" "=" nonce-value
        nonce-value       =  quoted-string


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



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

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

        pgp-pubalgorithm: The value of this parameter indicates the PGP
             public-key algorithm to be used for signing and encrypting
             messages. The currently defined values are "rsa" for the
             RSA algorithm and "dsa" for the DSA algorithm. If this
             parameter is not present, it is assumed to be "dsa".

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

        nonce: A server-specified data string which should be uniquely
             generated each time a 401 response is returned. It is
             RECOMMENDED that this string be base64 [43] or hexadecimal
             data. Specifically, since the string is passed in the
             header lines as a quoted string, the double-quote character
             is not allowed. The contents of the nonce are
             implementation dependent. The quality of the implementation
             depends on a good choice. Since the nonce is used only to
             prevent replay attacks and is signed, a time stamp in units
             convenient to the server is sufficient.


             Replay attacks within the duration of the call setup
             are of limited interest, so that timestamps with a
             resolution of a few seconds are often sufficient. In
             that case, the server does not have to keep a record
             of the nonces.

   Example:

   WWW-Authenticate: pgp version="5.0"
     realm="Your Startrek identity, please", algorithm=md5,
     nonce="913082051"






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14.1.2 The Authorization Request Header

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



        Authorization  =  "Authorization" ":" "pgp" # pgp-response
        pgp-response   =  realm | pgp-version | pgp-signature
                          | signed-by | nonce
        pgp-signature  =  "signature" "=" quoted-string
        signed-by      =  "signed-by" "=" <"> URI <">


   The client MUST increment the CSeq header before resubmitting the
   request. The signature MUST correspond to the From header of the
   request unless the signed-by parameter is provided.

        pgp-signature: The PGP ASCII-armored signature [39] and [42], as
             it appears between the "BEGIN PGP MESSAGE" and "END PGP
             MESSAGE" delimiters, without the version indication. The
             signature is included without any linebreaks.

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

             UACs MAY attempt to authenticate themselves without a nonce
             on the first INVITE request rather than waiting for a 401
             response if the UAC knows, e.g., from past requests or
             local configuration, that the UAS supports PGP
             authentication. A server MAY be configured not to generate
             nonces only if replay attacks are not a concern.


             Not generating nonces avoids the additional set of
             request, 401 response and ACK messages and reduces
             delay by one round-trip time.


             Using the ASCII-armored version is about 25% less
             space-efficient than including the binary signature,



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

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

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

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

   Example:

   Authorization: pgp version="5.0",
     realm="Your Startrek identity, please",
     nonce="913082051",
     signature="iQB1AwUBNNJiUaYBnHmiiQh1AQFYsgL/Wt3dk6TWK81/b0gcNDf
     VAUGU4rhEBW972IPxFSOZ94L1qhCLInTPaqhHFw1cb3lB01rA0RhpV4t5yCdUt
     SRYBSkOK29o5e1KlFeW23EzYPVUm2TlDAhbcjbMdfC+KLFX
     =aIrx"



14.2 PGP Encryption Scheme

   The PGP encryption scheme uses the following syntax:



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


        encoding: Describes the encoding or "armor" used by PGP. The
             value "ascii" refers to the standard PGP ASCII armor,
             without the lines containing "BEGIN PGP MESSAGE" and "END



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             PGP MESSAGE" and without the version identifier. By
             default, the encrypted part is included as binary.

   Example:

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



14.3 Response-Key Header Field for PGP



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


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

   Example:

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



15 Examples

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

15.1 Registration

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


   C->S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP saturn.bell-tel.com



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         From: <sip:watson@bell-tel.com>;tag=19
         To: sip:watson@bell-tel.com
         Call-ID: 70710@saturn.bell-tel.com
         CSeq: 1 REGISTER
         Contact: <sip:watson@saturn.bell-tel.com:3890;transport=udp>
         Expires: 7200



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

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


   C->S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP saturn.bell-tel.com
         From: <sip:watson@bell-tel.com>;tag=19
         To: sip:watson@bell-tel.com
         Call-ID: 70710@saturn.bell-tel.com
         CSeq: 2 REGISTER
         Contact: *
         Expires: 0

   C->S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP saturn.bell-tel.com
         From: <sip:watson@bell-tel.com>;tag=19
         To: sip:watson@bell-tel.com
         Call-ID: 70710@saturn.bell-tel.com
         CSeq: 3 REGISTER
         Contact: sip:tawatson@example.com



   Now, the server will forward any request for Watson to the server at
   example.com, using the Request-URI tawatson@example.com. For the
   server at example.com to reach Watson, he will need to send a
   REGISTER there, or inform the server of his current location through
   some other means.

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

   C->S: REGISTER sip:bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP pluto.bell-tel.com



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         From: <sip:jon.diligent@bell-tel.com>;tag=7
         To: sip:watson@bell-tel.com
         Call-ID: 17320@pluto.bell-tel.com
         CSeq: 1 REGISTER
         Contact: sip:tawatson@example.com



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

15.2 Invitation to a Multicast Conference

   The first example invites bob@example.com to a multicast session.
   All examples use the Session Description Protocol (SDP) (RFC 2327
   [6]) as the session description format.

15.2.1 Request


   C->S: INVITE sip:bob@one.example.com SIP/2.0
         Via: SIP/2.0/UDP sip.example.com;branch=7c337f30d7ce.1
           ;maddr=239.128.16.254;ttl=16
         Via: SIP/2.0/UDP mouse.wonderland.com
         From: Alice <sip:alice@wonderland.com>;tag=1
         To: Bob <sip:bob@example.com>
         Call-ID: 602214199@mouse.wonderland.com
         CSeq: 1 INVITE
         Contact: Alice <sip:alice@mouse.wonderland.com>
         Subject: SIP will be discussed, too
         Content-Type: application/sdp
         Content-Length: 187

         v=0
         o=user1 53655765 2353687637 IN IP4 128.3.4.5
         s=Mbone Audio
         t=3149328700 0
         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
         a=rtpmap:0 PCMU/8000





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   The From request header above states that the request was initiated
   by alice@wonderland.com and addressed to bob@example.com (From header
   fields). The Via fields list the hosts along the path from invitation
   initiator (the last element of the list) towards the callee. 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
   sip.example.com. The second Via header field indicates that it was
   originally sent from the outbound proxy mouse.wonderland.com. The
   Request-URI indicates that the request is currently being being
   addressed to bob@one.example.com, the local address that the SIP
   server for the example.com domain looked up for the callee.

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

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

15.2.2 Response

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


   S->C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP csvax.cs.caltech.edu;branch=7c337f30d7ce.1
           ;maddr=239.128.16.254;ttl=16
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Alice <sip:alice@wonderland.com>;tag=1
         To: Bob <sip:bob@example.com> ;tag=3141593
         Call-ID: 602214199@mouse.wonderland.com
         CSeq: 1 INVITE



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

   In addition, the Contact 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 csvax.cs.caltech.edu;branch=7c337f30d7ce.1
           ;maddr=239.128.16.254;ttl=16
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Alice <sip:alice@wonderland.com>;tag=1
         To: Bob <sip:bob@example.com> ;tag=3141593
         Call-ID: 602214199@mouse.wonderland.com
         CSeq: 1 INVITE
         Contact: <sip:bob@one.example.com>



   The caller confirms the invitation by sending an ACK request to the
   location named in the Contact header:


   C->S: ACK sip:bob@one.example.com SIP/2.0
         Via: SIP/2.0/UDP north.east.isi.edu
         From: Alice <sip:alice@wonderland.com>;tag=1
         To: Bob <sip:bob@example.com> ;tag=3141593
         Call-ID: 602214199@mouse.wonderland.com
         CSeq: 1 ACK



15.3 Two-party Call

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


   C->S: INVITE sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To: T. Watson <sip:watson@bell-tel.com>
         Call-ID: 662606876@kton.bell-tel.com
         CSeq: 1 INVITE
         Contact: <sip:a.g.bell@kton.bell-tel.com>
         Subject: Mr. Watson, come here.
         Content-Type: application/sdp
         Content-Length: ...

         v=0
         o=bell 53655765 2353687637 IN IP4 128.3.4.5
         s=Mr. Watson, come here.
         t=3149328600 0



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         c=IN IP4 kton.bell-tel.com
         m=audio 3456 RTP/AVP 0 3 4 5
         a=rtpmap:0 PCMU/8000
         a=rtpmap:3 GSM/8000
         a=rtpmap:4 G723/8000
         a=rtpmap:5 DVI4/8000

   S->C: SIP/2.0 100 Trying
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
         Call-ID: 662606876@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S->C: SIP/2.0 180 Ringing
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
         Call-ID: 662606876@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S->C: SIP/2.0 182 Queued, 2 callers ahead
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
         Call-ID: 662606876@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S->C: SIP/2.0 182 Queued, 1 caller ahead
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
         Call-ID: 662606876@kton.bell-tel.com
         CSeq: 1 INVITE
         Content-Length: 0

   S->C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To: <sip:watson@bell-tel.com> ;tag=37462311
         Call-ID: 662606876@kton.bell-tel.com
         CSeq: 1 INVITE
         Contact: sip:watson@boston.bell-tel.com
         Content-Type: application/sdp
         Content-Length: ...



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         v=0
         o=watson 4858949 4858949 IN IP4 192.1.2.3
         s=I'm on my way
         t=3149329600 0
         c=IN IP4 boston.bell-tel.com
         m=audio 5004 RTP/AVP 0 3
         a=rtpmap:0 PCMU/8000
         a=rtpmap:3 GSM/8000



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

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

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

   Since the two sides have agreed on the set of media, Bell confirms
   the call without enclosing another session description:


   C->S: ACK sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 1 ACK



15.4 Terminating a Call

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


   C->S: BYE sip:watson@boston.bell-tel.com SIP/2.0
         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To: T. A. Watson <sip:watson@bell-tel.com> ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com



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         CSeq: 2 BYE



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

15.5 Forking Proxy

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

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


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



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


   P->H: INVITE sip:watson@h.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.1
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE
         Contact: a.g.bell@c.bell-tel.com





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


   H->P: SIP/2.0 404 Not Found
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.1
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>;tag=87454273
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE



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

   P->H: ACK sip:watson@h.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.1
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>;tag=87454273
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 ACK



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

   P->A: INVITE sip:watson@acm.org SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.2
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE
         Contact: a.g.bell@c.bell-tel.com



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


   P->X: INVITE sip:t.watson@x.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.3
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@c.bell-tel.com



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         CSeq:    1 INVITE
         Contact: a.g.bell@c.bell-tel.com

   P->Y: INVITE sip:watson@y.bell-tel.com SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.4
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE
         Contact: a.g.bell@c.bell-tel.com



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


   X->P: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.3
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:       T. Watson <sip:t.watson@ieee.org> ;tag=192137601
         Call-ID:  31415@c.bell-tel.com
         CSeq:     1 INVITE
         Contact:  sip:t.watson@x.bell-tel.com

   Y->P: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.4
         Via:      SIP/2.0/UDP c.bell-tel.com
         Contact:  sip:t.watson@y.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:       T. Watson <sip:t.watson@ieee.org> ;tag=35253448
         Call-ID:  31415@c.bell-tel.com
         CSeq:     1 INVITE



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


   P->A: CANCEL sip:watson@acm.org SIP/2.0
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.2
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>



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         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 CANCEL



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


   A->P: SIP/2.0 200 OK
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.2
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 CANCEL



   In addition, P responds to the original INVITE request with a 487
   (Request Terminated):

   A->P: SIP/2.0 487 Request Terminated
         Via:     SIP/2.0/UDP sip.ieee.org ;branch=3d8a50dbf5a28d.2
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE


   This response terminates at P.

   Bell gets the two 200 responses from X and Y in short order and sends
   and ACK to both directly. Bell can now keep both call legs or
   terminate one with a BYE request. Here, he temporarily keeps both to
   determine where the real Watson is located.


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

   C->Y: ACK sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com



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         From:     A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:       T. Watson <sip:t.watson@ieee.org>;tag=35253448
         Call-ID:  31415@c.bell-tel.com
         CSeq:     1 ACK



   After a brief discussion between Bell with X and Y, it becomes clear
   that Watson is at X. (Note that this is not a three-way call; only
   Bell can talk to X and Y, but X and Y cannot talk to each other.)
   Thus, Bell sends a BYE to Y, which is replied to:


   C->Y: BYE sip:watson@y.bell-tel.com SIP/2.0
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:       T. Watson <sip:t.watson@ieee.org>;tag=35253448
         Call-ID:  31415@c.bell-tel.com
         CSeq:     2 BYE

   Y->C: SIP/2.0 200 OK
         Via:      SIP/2.0/UDP c.bell-tel.com
         From:     A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:       T. Watson <sip:t.watson@ieee.org>;tag=35253448
         Call-ID:  31415@c.bell-tel.com
         CSeq:     2 BYE



15.6 Redirects

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


   P->C: SIP/2.0 302 Moved temporarily
         Via:     SIP/2.0/UDP c.bell-tel.com
         From:    A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To:      T. Watson <sip:t.watson@ieee.org>;tag=72538263
         Call-ID: 31415@c.bell-tel.com
         CSeq:    1 INVITE
         Contact: sip:watson@h.bell-tel.com,
                   sip:watson@acm.org, sip:t.watson@x.bell-tel.com,
                   sip:watson@y.bell-tel.com





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   As another example, assume Alice (A) wants to delegate her calls to
   Bob (B) while she is on vacation until July 29th, 1998. Any calls
   meant for her will reach Bob with Alice's To field, indicating to him
   what role he is to play. Charlie (C) calls Alice (A), whose server
   returns:


   A->C: SIP/2.0 302 Moved temporarily
         From: Charlie <sip:charlie@caller.com>;tag=5
         To: Alice <sip:alice@wonderland.com> ;tag=2332462
         Call-ID: 27182@caller.com
         Contact: sip:bob@example.com
         Expires: Wed, 29 Jul 1998 9:00:00 GMT
         CSeq: 1 INVITE



   Charlie then sends the following request to the SIP server of the
   example.com domain. Note that the server at example.com forwards the
   request to Bob based on the Request-URI.


   C->B: INVITE sip:bob@example.com SIP/2.0
         Via: SIP/2.0/UDP h.caller.com
         From: <sip:charlie@caller.com>;tag=5
         To: sip:alice@wonderland.com
         Call-ID: 27182@caller.com
         CSeq: 2 INVITE
         Contact: sip:charlie@h.caller.com



   In the third redirection example, we assume that all outgoing
   requests are directed through a local firewall F ("outbound proxy")
   at caller.com, with Charlie again inviting Alice:


   C->F: INVITE sip:alice@wonderland.com SIP/2.0
         Via: SIP/2.0/UDP h.caller.com
         From: <sip:charlie@caller.com>;tag=5
         To: Alice <sip:alice@wonderland.com>
         Call-ID: 27182@caller.com
         CSeq: 1 INVITE
         Contact: sip:charlie@h.caller.com



   The local firewall at caller.com happens to be overloaded and thus



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   redirects the call from Charlie to a secondary server S:


   F->C: SIP/2.0 302 Moved temporarily
         Via: SIP/2.0/UDP h.caller.com
         From: <sip:charlie@caller.com>;tag=5
         To: Alice <sip:alice@wonderland.com>
         Call-ID: 27182@caller.com
         CSeq: 1 INVITE
         Contact: <sip:alice@wonderland.com:5080;maddr=spare.caller.com>



   Based on this response, Charlie directs the same invitation to the
   secondary server spare.caller.com at port 5080, but maintains the
   same Request-URI as before:


   C->S: INVITE sip:alice@wonderland.com SIP/2.0
         Via: SIP/2.0/UDP h.caller.com
         From: <sip:charlie@caller.com>;tag=5
         To: Alice <sip:alice@wonderland.com>
         Call-ID: 27182@caller.com
         CSeq: 2 INVITE
         Contact: sip:charlie@h.caller.com



15.7 Negotiation

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


   S->C: SIP/2.0 606 Not Acceptable
         Via: SIP/2.0/UDP c.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3
         To: T. Watson <sip:t.watson@ieee.org> ;tag=7434264
         Call-ID: 14142@c.bell-tel.com
         CSeq: 1 INVITE
         Warning: 370 "Insufficient bandwidth (only have ISDN)",
           305 "Incompatible media format",
           330 "Multicast not available"
         Content-Type: application/sdp
         Content-Length: ...

         v=0
         o=c 3149329138 3149329165 IN IP4 38.245.76.2
         s=Let's talk



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         t=3149328630 0
         b=CT:128
         c=IN IP4 x.bell-tel.com
         m=audio 3456 RTP/AVP 5 0 7
         a=rtpmap:5 DVI4/8000
         a=rtpmap:0 PCMU/8000
         a=rtpmap:7 LPC/8000
         m=video 2232 RTP/AVP 31
         a=rtpmap:31 H261/90000



   In this example, the original request specified a bandwidth that was
   higher than the access link could support, requested multicast, and
   requested a set of media encodings. The response states that only 128
   kb/s is available and that (only) DVI, PCM or LPC audio could be
   supported in order of preference.

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

15.8 OPTIONS Request

   A caller Alice can use an OPTIONS request to find out the
   capabilities of a potential callee Bob, without "ringing" the
   designated address. Bob returns a description indicating that he is
   capable of receiving audio encodings PCM mu-law (RTP payload type 0),
   1016 (payload type 1), GSM (payload type 3), and SX7300/8000 (dynamic
   payload type 99), and video encodings H.261 (payload type 31) and
   H.263 (payload type 34).


   C->S: OPTIONS sip:bob@example.com SIP/2.0
         Via: SIP/2.0/UDP cat.wonderland.com
         From: Alice <sip:alice@wonderland.com>;tag=1
         To: Bob <sip:bob@example.com>
         Call-ID: 6378@cat.wonderland.com
         CSeq: 1 OPTIONS
         Accept: application/sdp

   S->C: SIP/2.0 200 OK
         From: Alice <sip:alice@wonderland.com>;tag=1
         To: Bob <sip:bob@example.com> ;tag=376364382
         Call-ID: 6378@cat.wonderland.com
         Content-Length: 81
         Content-Type: application/sdp




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         v=0
         o=alice 3149329138 3149329165 IN IP4 24.124.37.3
         s=Security problems
         t=3149328650 0
         c=IN IP4 24.124.37.3
         m=audio 0 RTP/AVP 0 1 3 99
         a=rtpmap:0 PCMU/8000
         a=rtpmap:1 1016/8000
         a=rtpmap:3 GSM/8000
         a=rtpmap:99 SX7300/8000
         m=video 0 RTP/AVP 31 34
         a=rtpmap:31 H261/90000
         a=rtpmap:34 H263/90000



A Minimal Implementation

A.1 Transport Protocol Support

   User agents and stateless proxies MUST support UDP and MAY support
   TCP or other transport protocols, stateful proxies MUST support both
   UDP and TCP.

A.2 Client

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

   The following capability sets build on top of the minimal
   implementation described in the previous paragraph. In general, each
   capability listed below builds on the ones above it:

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

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

        Firewall-friendly: A firewall-friendly client understands the
             Route and Record-Route header fields and can be configured



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             to use a local proxy for all outgoing requests.

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

        Authentication: If a client wishes to invite callees that
             require caller authentication, it MUST be able to recognize
             the 401 (Unauthorized) status code, MUST be able to
             generate the Authorization request header and MUST
             understand the WWW-Authenticate response header.

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

A.3 Server

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

A.4 Header Processing

   Table 6 lists the headers that different implementations support. UAC
   refers to a user-agent client (calling user agent), UAS to a user-
   agent server (called user-agent).

   The fields in the table have the following meaning. Type is as in
   Table 4 and 5. "-" indicates the field is not meaningful to this
   system (although it might be generated by it). "m" indicates the
   field MUST be understood. "b" indicates the field SHOULD be
   understood by a basic implementation.  "r" indicates the field SHOULD
   be understood if the system claims to understand redirection. "a"
   indicates the field SHOULD be understood if the system claims to
   support authentication. "e" indicates the field SHOULD be understood
   if the system claims to support encryption. "o" indicates support of
   the field is purely optional. Headers whose support is optional for
   all implementations are not shown.




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


   Table 6: Header Field Processing Requirements


B Usage of the Session Description Protocol (SDP)

   This section describes the use of the Session Description Protocol
   (SDP) (RFC 2327 [6]). SDP is identified as Content-Type
   "application/sdp".  Each SIP message body may only contain one SDP
   message unless the SIP message body is of type "multipart". SDP
   messages in later SIP request supersede earlier ones for the same
   call leg.

B.1 Configuring Media Streams




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   The caller and callee align their media descriptions so that the nth
   media stream ("m=" line) in the caller's session description
   corresponds to the nth media stream in the callee's description.

   All media descriptions SHOULD contain "a=rtpmap" mappings from RTP
   payload types to encodings.

        This allows easier migration away from static payload
        types.

   User agents MUSTNOT reuse dynamic payload types for different
   encodings when issuing a re-INVITE.

   If the callee wants to neither send nor receive a stream offered by
   the caller, the callee sets the port number of that stream to zero in
   its media description.


        There currently is no other way than port zero for the
        callee to refuse a bidirectional stream offered by the
        caller. Both caller and callee need to be aware what media
        tools are to be started.

   For example, assume that the caller Alice has included the following
   description in her INVITE request. It includes a bidirectional audio
   stream and two bidirectional video streams, using H.261 (payload type
   31) and MPEG (payload type 32).


   v=0
   o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
   s=New board design
   t=0 0
   c=IN IP4 host.anywhere.com
   m=audio 49170 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   m=video 51372 RTP/AVP 31
   a=rtpmap:31 H261/90000
   m=video 53000 RTP/AVP 32
   a=rtpmap:32 MPV/90000



   The callee, Bob, does not want to receive or send the first video
   stream, so it returns the media description below:

   v=0
   o=bob 2890844730 2890844730 IN IP4 host.example.com



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   s=New board design
   t=0 0
   c=IN IP4 host.example.com
   m=audio 47920 RTP/AVP 0 1
   a=rtpmap:0 PCMU/8000
   m=video 0 RTP/AVP 31
   m=video 53000 RTP/AVP 32
   a=rtpmap:32 MPV/90000



B.2 Setting SDP Values for Unicast

   If a session description from a caller contains a media stream which
   is listed as send (receive) only, it means that the caller is only
   willing to send (receive) this stream, not receive (send). The same
   is true for the callee.  Media streams are marked as send-only or
   receive-only media streams using the SDP "a=sendonly" and
   "a=recvonly" attributes, respectively.

   For receive-only and send-or-receive streams, the port number and
   address in the session description indicate where the media stream
   should be sent to by the recipient of the session description, either
   caller or callee.  For send-only RTP streams, the address and port
   number indicate where RTCP reports are to be sent. (RTCP reports are
   sent to the port number one higher than the number indicated.)

   The list of payload types for each media stream conveys two pieces of
   information, namely the set of codecs that the caller or callee is
   capable of sending or receiving, and the RTP payload type numbers
   used to identify those codecs. For receive-only or send-and-receive
   media streams, a caller SHOULD list all of the codecs it is capable
   of supporting in the session description in an INVITE or ACK. For
   send-only streams, the caller SHOULD indicate only those it wishes to
   send for this session. For receive-only streams, the payload type
   numbers indicate the value of the payload type field in RTP packets
   the caller is expecting to receive for that codec type. For send-only
   streams, the payload type numbers indicate the value of the payload
   type field in RTP packets the caller is planning to send for that
   codec type.  For send-and-receive streams, the payload type numbers
   indicate the value of the payload type field the caller expects to
   both send and receive.

   If a media stream is listed as receive-only by the caller, the callee
   lists, in the response, those codecs it intends to use from among the
   ones listed in the request. If a media stream is listed as send-only
   by the caller, the callee lists, in the response, those codecs it is
   willing to receive among the ones listed in the the request. If the



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   media stream is listed as both send and receive, the callee lists
   those codecs it is capable of sending or receiving among the ones
   listed by the caller in the INVITE. The actual payload type numbers
   in the callee's session description corresponding to a particular
   codec MUST be the same as the caller's session description.

   If caller and callee have no media formats in common for a particular
   stream, the callee MUST return a session description containing the
   particular "m=" line, but with the port number set to zero. Any media
   formats listed are ignored. (SDP requires at least one "fmt" entry.)

   If there are no media formats in common for all streams, the callee
   SHOULD return a 488 or 606 response, with a 304 Warning header field.

B.3 Multicast Operation

   The interpretation of send-only and receive-only for multicast media
   sessions differs from that for unicast sessions. For multicast,
   send-only means that the recipient of the session description (caller
   or callee) SHOULD only send media streams to the address and port
   indicated. Receive-only means that the recipient of the session
   description SHOULD only receive media on the address and port
   indicated.

   For multicast, receive and send multicast addresses are the same and
   all parties use the same port numbers to receive media data. If the
   session description provided by the caller is acceptable to the
   callee, the callee can choose not to include a session description or
   MAY echo the description in the response.

   A callee MAY, in the response, return a session description with some
   of the payload types removed, or port numbers set to zero (but no
   other value). This indicates to the caller that the callee does not
   support the given stream or media types which were removed. A callee
   MUSTNOT change whether a given stream is send-only, receive-only, or
   send-and-receive.

   If a callee does not support multicast at all, it SHOULD return a 400
   status response and include a 330 Warning.

B.4 Delayed Media Streams

   In some cases, a caller may not know the set of media formats which
   it can support at the time it would like to issue an invitation. This
   is the case when the caller is actually a gateway to another protocol
   which performs media format negotiation after call setup. When this
   occurs, a caller MAY issue an INVITE with a session description that
   contains no media lines. The callee SHOULD interpret this to mean



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   that the caller wishes to participate in a multimedia session
   described by the session description, but that the media streams are
   not yet known. The callee SHOULD return a session description
   indicating the streams and media formats it is willing to support,
   however. The caller MAY update the session description either in the
   ACK request or in a re-INVITE at a later time, once the streams are
   known.

B.5 Adding and Deleting Media Streams

   To add a stream to an existing call leg, either party appends an
   additional "m" line to the previous session description when sending
   a re-INVITE.

   To remove a stream from a call leg, either party sets its port to
   zero in the session description when sending a re-INVITE.

   UAs receiving SDP SHOULD accept descriptions that are not aligned, in
   terms of "m=" lines, with earlier descriptions. If a UA receives such
   a description, it SHOULD line up "m=" lines by media type ("audio",
   "video", ...). If it receives a new description in an INVITE request,
   it MAY line up the "m=" lines by media type, address and port and
   position in the description and start or delete media streams
   accordingly.


        Receiving a new description in an INVITE request that omits
        media with zero ports or is otherwise different may occur
        if the UA crashed and restarted.

B.6 Putting Media Streams on Hold

   If a party in a call wants to put the other party "on hold", i.e.,
   request that it temporarily stops sending one or more media streams,
   a party re-invites the other by sending an INVITE request with a
   modified session description. The session description is the same as
   in the original invitation (or response), but the "c" destination
   addresses for the media streams to be put on hold are set to zero
   (0.0.0.0).

B.7 Subject and SDP "s=" Line

   The SDP "s=" line and the SIP Subject header field have different
   meanings when inviting to a multicast session. The session
   description line describes the subject of the multicast session,
   while the SIP Subject header field describes the reason for the
   invitation. The example in Section 16.2 illustrates this point. For
   invitations to two-party sessions, the SDP "s=" line MAY consist of a



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   single space character (0x20).


        Unfortunately, SDP does not allow to leave the "s=" line
        empty.

B.8 The SDP "o=" Line

   The "o=" line MUST be present for all sessions, including two-party
   sessions, to allow re-use of SDP-based tools.

C Summary of Augmented BNF

   All of the mechanisms specified in this document are described in
   both prose and an augmented Backus-Naur Form (BNF) similar to that
   used by RFC 822 [26] and RFC 2234 [44]. Implementors will need to be
   familiar with the notation in order to understand this specification.
   The augmented BNF includes the following constructs:



        name  =  definition


   The name of a rule is simply the name itself (without any enclosing
   "<" and ">") and is separated from its definition by the equal "="
   character. White space is only significant in that indentation of
   continuation lines is used to indicate a rule definition that spans
   more than one line. Certain basic rules are in uppercase, such as SP,
   LWS, HT, CRLF, DIGIT, ALPHA, etc. Angle brackets are used within
   definitions whenever their presence will facilitate discerning the
   use of rule names.


   "literal"


   Quotation marks surround literal text. Unless stated otherwise, the
   text is case-insensitive.


   rule1 | rule2


   Elements separated by a bar ("|") are alternatives, e.g., "yes | no"
   will accept yes or no.





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   (rule1 rule2)


   Elements enclosed in parentheses are treated as a single element.
   Thus, "(elem (foo | bar) elem)" allows the token sequences "elem foo
   elem" and "elem bar elem".


   *rule


   The character "*" preceding an element indicates repetition. The full
   form is "<n>*<m>element" indicating at least <n> and at most <m>
   occurrences of element. Default values are 0 and infinity so that
   "*(element)" allows any number, including zero; "1*element" requires
   at least one; and "1*2element" allows one or two.


   [rule]


   Square brackets enclose optional elements; "[foo bar]" is equivalent
   to "*1(foo bar)".


   N rule


   Specific repetition: "<n>(element)" is equivalent to
   "<n>*<n>(element)"; that is, exactly <n> occurrences of (element).
   Thus 2DIGIT is a 2-digit number, and 3ALPHA is a string of three
   alphabetic characters.


   #rule


   A construct "#" is defined, similar to "*", for defining lists of
   elements. The full form is "<n>#<m> element" indicating at least <n>
   and at most <m> elements, each separated by one or more commas (",")
   and OPTIONAL linear white space (LWS). This makes the usual form of
   lists very easy; a rule such as



           ( *LWS element *( *LWS "," *LWS element ))





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   can be shown as 1# element. Wherever this construct is used, null
   elements are allowed, but do not contribute to the count of elements
   present. That is, "(element), , (element)" is permitted, but counts
   as only two elements. Therefore, where at least one element is
   required, at least one non-null element MUST be present. Default
   values are 0 and infinity so that "#element" allows any number,
   including zero; "1#element" requires at least one; and "1#2element"
   allows one or two.


   ; comment


   A semi-colon, set off some distance to the right of rule text, starts
   a comment that continues to the end of line. This is a simple way of
   including useful notes in parallel with the specifications.


   implied *LWS


   The grammar described by this specification is word-based. Except
   where noted otherwise, linear white space (LWS) can be included
   between any two adjacent words (token or quoted-string), and between
   adjacent tokens and separators, without changing the interpretation
   of a field. At least one delimiter (LWS and/or separators) MUST exist
   between any two tokens (for the definition of "token" below), since
   they would otherwise be interpreted as a single token. Note that URLs
   do NOT contain LWS.

C.1 Basic Rules

   The following rules are used throughout this specification to
   describe basic parsing constructs. The US-ASCII coded character set
   is defined by ANSI X3.4-1986.



        OCTET     =  %x00-ff ; any 8-bit sequence of data
        CHAR      =  %x00-7f ; any US-ASCII character (octets 0 - 127)
        upalpha   =  "A" | "B" | "C" | "D" | "E" | "F" | "G" | "H" | "I" |
                     "J" | "K" | "L" | "M" | "N" | "O" | "P" | "Q" | "R" |
                     "S" | "T" | "U" | "V" | "W" | "X" | "Y" | "Z"
        lowalpha  =  "a" | "b" | "c" | "d" | "e" | "f" | "g" | "h" | "i" |
                     "j" | "k" | "l" | "m" | "n" | "o" | "p" | "q" | "r" |
                     "s" | "t" | "u" | "v" | "w" | "x" | "y" | "z"
        alpha     =  lowalpha | upalpha
        DIGIT     =  "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" |



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                     "8" | "9"
        alphanum  =  alpha | DIGIT
        CTL       =  %x00-1f | %x7f ; (octets 0 -- 31) and DEL (127)
        CR        =  %d13 ; US-ASCII CR, carriage return character
        LF        =  %d10 ; US-ASCII LF, line feed character
        SP        =  %d32 ; US-ASCII SP, space character
        HT        =  %d09 ; US-ASCII HT, horizontal tab character
        CRLF      =  CR LF ; typically the end of a line


   The following are defined in RFC 2396 [9] for the SIP URI:


        unreserved  =  alphanum | mark
        mark        =  "-" | "_" | "." | "!" | "~" | "*" | "'"
                   |   "(" | ")"
        escaped     =  "%" hex hex


   SIP header field values can be folded onto multiple lines if the
   continuation line begins with a space or horizontal tab. All linear
   white space, including folding, has the same semantics as SP. A
   recipient MAY replace any linear white space with a single SP before
   interpreting the field value or forwarding the message downstream.



        LWS  =  *( SP | HT ) [CRLF] 1*( SP | HT ) ; linear whitespace


   The TEXT-UTF8 rule is only used for descriptive field contents and
   values that are not intended to be interpreted by the message parser.
   Words of *TEXT-UTF8 contain characters from the UTF-8 character set
   (RFC 2279 [23]).  The TEXT-UTF8-TRIM rule is used for descriptive
   field contents that are not quoted strings, where leading and
   trailing LWS is not meaningful. In this regard, SIP differs from
   HTTP, which uses the ISO 8859-1 character set.



        TEXT-UTF8       =  *(TEXT-UTF8char | LWS)
        TEXT-UTF8-TRIM  =  *TEXT-UTF8char *(*LWS TEXT-UTF8char)
        TEXT-UTF8char   =  %x21-7e
                        |  UTF8-NONASCII
        UTF8-NONASCII   =  %xc0-df 1UTF8-CONT
                        |  %xe0-ef 2UTF8-CONT
                        |  %xf0-f7 3UTF8-CONT
                        |  %xf8-fb 4UTF8-CONT



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                        |  %xfc-fd 5UTF8-CONT
        UTF8-CONT       =  %x80-bf


   A CRLF is allowed in the definition of TEXT-UTF8 only as part of a
   header field continuation. It is expected that the folding LWS will
   be replaced with a single SP before interpretation of the TEXT-UTF8
   value.

   Hexadecimal numeric characters are used in several protocol elements.



        HEX  =  "A" | "B" | "C" | "D" | "E" | "F"
                | "a" | "b" | "c" | "d" | "e" | "f" | DIGIT


   Many SIP header field values consist of words separated by LWS or
   special characters. Unless otherwise stated, tokens are case-
   insensitive. These special characters MUST be in a quoted string to
   be used within a parameter value.



        token                            =  1*(alphanum | "-" | "." | "!" | "%" | "*"
        | "_" | "+" | "`" | "'" | "~" )
        separators                       =  "(" | ")" | "<" | ">" | "@" |
                                            "," | ";" | ":" | "\" | <"> |
                                            "/" | "[" | "]" | "?" | "=" |
                                            "{" | "}" | SP | HT


   Comments can be included in some SIP header fields by surrounding the
   comment text with parentheses. Comments are only allowed in fields
   containing "comment" as part of their field value definition. In all
   other fields, parentheses are considered part of the field value.



        comment  =  "(" *(ctext | quoted-pair | comment) ")"
        ctext    =  < any TEXT-UTF8  excluding "("  and ")">


   A string of text is parsed as a single word if it is quoted using
   double-quote marks. In quoted strings, quotation marks (") and
   backslashes (
   ) need to be escaped.




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        quoted-string  =  ( <"> *(qdtext | quoted-pair ) <"> )
        qdtext         =  LWS | %x21 | %x23-5b | %x5d-7e
                       |  UTF8-NONASCII


   The backslash character ("\") MAY be used as a single-character
   quoting mechanism only within quoted-string and comment constructs.
   Unlike HTTP/1.1, the characters CR and LF cannot be escaped by this
   mechanism to avoid conflict with line folding and header separation.



        quoted-pair  =  "\" (%x00 - %x09 | %x0b | %x0c | %x0e - %x7f)


D IANA Considerations

   Section 4.4 describes a name space and mechanism for registering SIP
   options. Section 6.48 describes the name space for registering SIP
   warn-codes.

   SIP Header field names are registered with IANA. They do not require
   working group or working group chair review, but SHOULD be documented
   in an RFC or Internet draft. For Internet drafts, IANA is requested
   to make the draft available as part of the registration database.


        By the time an RFC is published, colliding names may have
        already been implemented.

   Headers SHOULDNOT use the X prefix notation and MUSTNOT duplicate the
   names of headers used by SMTP or HTTP unless the syntax is a
   compatible superset and the semantics are similar. Some common and
   widely used header fields MAY be assigned one-letter compact forms
   (Section 9). Compact forms can only be assigned after SIP working
   group review. In the absence of this working group, a designated
   expert reviews the request.

E Changes from RFC 2543

   In addition to editorial clarifications, this document changes or
   adds the following features to SIP as specified in RFC 2543:

        o Extensions developed by the IETF no longer use the org.ietf
          prefix.

        o Tag syntax was generalized.




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        o Via header branch parameters were extended to allow "spirals",
          where two requests that differ only in the request URI are not
          treated as copies.

        o New optional header fields, Alert-Info, Call-Info, In-Reply-
          To.

F Changes Made in Version 00

        o In Sec. 10.3.1, indicated that UAC should send both CANCEL and
          BYE after a retransmission fails.

        o Added semicolon and question mark to the list of unreserved
          characters for the user part of SIP URLs to handle tel: URLs
          properly.

        o Uniform handling of if hop count Max-Forwards: return 483.
          Note that this differs from HTTP/1.1 behavior, where only
          OPTIONS and TRACE allow this header, but respond as the final
          recipient when the value reaches zero.

        o Clarified that a forking proxy sends ACKs only for INVITE
          requests.

        o Clarified wording of DNS caching. Added paragraph on "negative
          caching", i.e., what to do if one of the hosts failed. It is
          probably not a good idea to simply drop this host from the
          list if the DNS ttl value is more than a few minutes, since
          that would mean that load balancing may not work for quite a
          while after a server is brought back on line. This will be
          true in particular if a server group receives a large number
          of requests from a small number of upstream servers, as is
          likely to be the case for calls between major consumer ISPs.
          However, without getting into arbitrary and complicated retry
          rules, it seems hard to specify any general algorithm. Might
          it be worthwhile to simply limit the "black list" interval to
          a few minutes?

        o Added optional Call-Info and Alert-Info header fields that
          describe the caller and information to be used in alerting.
          (Currently, avoided use of "purpose" qualification since it is
          not yet clear whether rendering content without understanding
          its meaning is always appropriate. For example, if a UAS does
          not understand that this header is to replace ringing, it
          would mix both local ring tone and the indicated sound URL.)
          TBD!

        o SDP "s=" lines can't be empty, unfortunately.  (Section B.7)



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        o Noted that maddr could also contain a unicast address, but
          SHOULD contain the multicast address if the request is sent
          via multicast (Section 6.47, 10.0.1).

        o Clarified that responses are sent to port in Via sent-by
          value.

        o Added "other-*" to the user URL parameter and the Hide and
          Content-Disposition headers.

        o Clarified generation of timeout (408) responses in forking
          proxies and mention the Expires header.  (Section 12.4)

        o Clarified that CANCEL and INVITE are separate transactions
          (Fig. 12). Thus, the INVITE request generates a 487
          (Transaction Terminated) if a CANCEL or BYE arrives.

        o Clarified that Record-Route SHOULD be inserted in every
          request, but that the route, once established, persists. This
          provides robustness if the called UAS crashes.

        o Emphasized that proxy, redirect, registrar and location
          servers are logical, not physical entities and that UAC and
          UAS roles are defined on a request-by-request basis. (Section
          1.4)

        o In Section 6.47, noted that the maddr and received parameters
          also need to be encrypted when doing Via hiding.

        o Simplified Fig. 12 to only show INVITE transaction.

        o Added definition of the use of Contact (Section 6.15) for
          OPTIONS.

        o Added HTTP/RFC822 headers Content-Language and MIME-Version.

        o Added note in Section A indicating that UAs need to support
          UDP.

        o Added explanation in Section 11.5 explaining what a UA should
          do when receiving an initial INVITE with a tag.

        o Clarified UA and proxy behavior for 302 responses (Section
          7.3.3).

        o Added details on what a UAS should do when receiving a tagged
          INVITE request for an unknown call leg. This could occur if
          the UAS had crashed and the UAC sends a re-INVITE or if the



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          BYE got lost and the UAC still believes to be in the call.

        o Added definition of Contact in 4xx, 5xx and 6xx to "redirect"
          to more error details.

        o Added note to forking proxy description in Section 12.4 to
          gather *-Authenticate from responses. This allows several
          branches to be authenticated simultaneously.

        o Changed URI syntax to use URL escaping instead of quotation
          marks.

        o Changed SIP URL definition to reference RFC 2806 for
          telephone-subscriber part.

        o Clarified that the To URI should basically be ignored by the
          receiving UAS except for matching requests to call legs. In
          particular, To headers with a scheme or name unknown to the
          callee should be accepted.

        o Clarified in Section 6.47.1 that maddr is to be added by any
          client, either proxy or UAC.

        o Added response code 488 to indicate that there was no common
          media at the particular destination. (606 indicates such
          failure globally.)

        o In Section 6.25, noted that registration updates can shorten
          the validity period.

        o Added note to Section 14.3 to enclose the URI in quotation
          marks. The BNF in RFC 2617 is in error.

        o Clarified that registrars use Authorization and WWW-
          Authenticate, not proxy authentication.

        o Added note in Section 6.15 that "headers" are copied from
          Contact into the new request.

        o Changed URL syntax so that port specifications have to have at
          least one digit, in line with other URL formats such as
          "http".  Previously, an empty port number was permissible.

        o In Section B, added a section on how to add and delete streams
          in re-INVITEs.

        o IETF-blessed extensions now have short names, without
          org.ietf. prefix.



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        o Cseq is unique within a call leg, not just within a call
          (Section 6.21).

        o Added IPv6 literal addresses to the SIP URL definition in
          Section 2, according to RFC 2732 [45].  Modified the IPv4
          address to limit segments to at most three digits.

        o In Section 4.2.6, modify registration procedure so that it
          explicitly references the URL comparison. Updates with shorter
          expiration time are now allowed.

        o For send-only media, SDP still must indicate the address and
          port, since these are needed as destinations for RTCP
          messages.  (Section B)

        o Changed references regarding DNS SRV records from RFC 2052 to
          RFC 2782, which is now a Proposed Standard. Integrated SRV
          into the search procedure in Section 1 and removed the SRV
          appendix. The only visible change is that protocol and service
          names are now prefixed by an underscore. Added wording that
          incorporates the precedence of maddr.

        o Allow parameters in Record-Route and Route headers.

        o In Table 2, list udp as the default value for the transport
          parameter in SIP URI.

        o Removed sentence that From can be encrypted. It cannot, since
          the header is needed for call-leg identification.

        o Added note that a UAC only copies a To tag into subsequent
          transactions if it arrives in a 200 OK to an INVITE in Section
          11. This avoids the problem that occurs when requests get
          resubmitted after receiving, say, a 407 (or possibly 500, 503,
          504, 305, 400, 411, 413, maybe even 408). Under the old rules,
          these requests would have a tag, which would force the called
          UAS to reject the request, since it doesn't have an entry for
          this tag.

        o Loop detection has been modified to take the request-URI into
          account (Section 12.3 and 6.47.6). This allows the same
          request to visit the server twice, but with different request
          URIs ("spiral").

        o Elaborated on URL comparison and comparison of From/To fields.

        o Added np-queried user parameter.




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        o Changed tag syntax from UUID to token, since there's no reason
          to restrict it to hex.

        o Added Content-Disposition header based on earlier discussions
          about labeling what to do with a message body (part).

        o Clarification: proxies must insert To tags for locally
          generated responses.

        o Clarification: multicast may be used for subsequent
          registrations.

        o Feature: Added Supported header. Needed if client wants to
          indicate things the server can usefully return in the
          response.

        o Bug: The From, To, and Via headers were missing extension
          parameters. The Encryption and Response-Key header fields now
          "officially" allow parameters consisting only of a token,
          rather than just "token = value".

        o Bug: Allow was listed as optional in 405 responses in Table 4.
          It is mandatory.

        o Added in Section 4.2.4: "A BYE request from either called or
          calling party terminates any pending INVITE, but the INVITE
          request transaction MUST be completed with a final response."

        o Clarified in Section 4.2.1: "If an INVITE request for an
          existing session fails, the session description agreed upon in
          the last successful INVITE transaction remains in force."

        o Clarified in Section 4.2.1 what happens if two INVITE requests
          meet each other on the wire, either traveling the same or in
          opposite directions:


             A UAC MUSTNOT issue another INVITE request for the
             same call leg before the previous transaction has
             completed. A UAS that receives an INVITE before it
             sent the final response to an INVITE with a lower CSeq
             number MUST return a 400 (Bad Request) response and
             MUST include a Retry-After header field with a
             randomly chosen value of between 0 and 10 seconds. A
             UA that receives an INVITE while it has an INVITE
             transaction pending, returns a 500 (Internal Server
             Error) and also includes a Retry-After header field.




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        o Expires header clarified: limits only duration of INVITE
          transaction, not the actual session. SDP does the latter.

        o The In-Reply-To header was added (Section 6.27).

        o There were two incompatible BNFs for WWW-Authenticate.  One
          defined for PGP, and the other borrowed from HTTP. For basic
          or digest:


            WWW-Authenticate: basic realm="Wallyworld"



          and for pgp:


            WWW-Authenticate: pgp; realm="Wallyworld"



          The latter is incorrect and the semicolon has been removed.

        o Added rules for Route construction from called to calling UA.

        o We now allow Accept and Accept-Encoding in BYE and CANCEL
          requests. There is no particular reason not to allow them, as
          both requests could theoretically return responses,
          particularly when interworking with other signaling systems.

        o PGP "pgp-pubalgorithm" allows server to request the desired
          public-key algorithm.

        o ABNF rules now describe tokens explicitly rather than by
          subtraction; explicit character enumeration for CTL, etc.

        o Registrars should be careful to check the Date header as the
          expiration time may well be in the past, as seen by the
          client.

        o Content-Length is mandatory; Table 4 erroneously marked it as
          optional.

        o User-Agent was classified in a syntax definition as a request
          header rather than a general header.

        o Clarified ordering of items to be signed and include realm in
          list.



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        o Allow Record-Route in 401 and 484 responses.

        o Hop-by-hop headers need to precede end-to-end headers only if
          authentication is used (Section 6).

        o 1xx message bodies MAY now contain session descriptions.

        o Changed references to HTTP/1.1 and authentication to point to
          the latest RFCs.

        o Added 487 (Request terminated) status response. It is issued
          if the original request was terminated via CANCEL or BYE.

        o The spec was not clear on the identification of a call leg.
          Section 1.3 says it's the combination of To, From, and Call-
          ID. However, requests from the callee to the caller have the
          To and From reversed, so this definition is not quite
          accurate. Additionally, the "tag" field should be included in
          the definition of call leg. The spec now says that a call leg
          is defined as the combination of local-address, remote-
          address, and call-id, where these addresses include tags.

          Text was added to Section 6.21 to emphasize that the From and
          To headers designate the originator of the request, not that
          of the call leg.

        o All URI parameters, except method, are allowed in a Request-
          URI. Consequently, also updated the description of which
          parameters are copied from 3xx responses in Sec. 6.15.

        o The use of CRLF, CR,or LF to terminate lines was confusing.
          Basically, each header line can be terminated by a CR, LF, or
          CRLF.  Furthermore, the end of the headers is signified by a
          "double return".  Simplified in Section 3 to require sending
          of CRLF, but require senders to receive CR and LF as well and
          only allow CR CR, LF LF in addition to double CRLF as a
          header-body separator.

        o Round brackets in Contact header were part of the HTTP legacy,
          and very hard to implement. They are also not that useful and
          were removed.

        o The spec said that a proxy is a back-to-back UAS/UAC. This is
          almost, but not quite, true. For example, a UAS should insert
          a tag into a provisional response, but a proxy should not.
          This was clarified.

        o Section 6.13 in the RFC begins mid-paragraph after the BNF.



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          The following text was misplaced in the conversion to ASCII:

             Even if the "display-name" is empty, the "name-addr"
             form MUST be used if the "addr-spec" contains a comma,
             semicolon or question mark.

G Changes Made in Version 01

        o Uniform syntax specification for semicolon parameters:


             Foo        =  "Foo" ":" something *( ";" foo-param )
             foo-param  =  "bar" "=" token
                       |   generic-param


        o Removed np-queried user parameter since this is now part of a
          tel URL extension parameter.

        o In Section B, noted that if the capabilities intersection is
          empty, a dummy format list still has to be returned due to SDP
          syntax constraints. Previously, the text had required that no
          formats be listed. (Brian Rosen)

        o Reorganized tables 4 and 5 to show proxy interaction with
          headers rather than "end-to-end" or "hop-by-hop".

H Changes Made in Version 02

        o Tentatively removed restriction that CANCEL requests cannot
          have Route headers. (Billy Biggs)

        o Tentatively added Also header for BYE requests, as it is
          widely implemented and a simple means to implement
          unsupervised call transfer. Subject to removal if there is
          protest. (Billy Biggs)

        o If a proxy sends a request by UDP (TCP), the spec did not
          disallow placing TCP (UDP) in the transport parameter of the
          Via field, which it should. Added a note that the transport
          protocol actually used is included.

        o No default value for the q parameter in Contact is defined.
          This is not strictly needed, but is useful for consistent
          behaviors at recursive proxies and at UAC's. Now 0.5.

        o Clarified that To and From tag values should be different to
          simplify request matching when calling oneself.



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        o Removed ability to carry multiple requests in a single UDP
          packet (Section 6.19).

        o Added note that Allow MAY be included in requests, to indicate
          requestor capabilities for the same call ID.

        o Added note to Section 6.22 indicating that registrars MUST
          include the Date header to accomodate UAs that do not have a
          notion of absolute time.

        o Added note to Section 4.2.6 emphasizing that non-SIP URIs are
          permissible.

        o Rewrote the server lookup section to be more precise and more
          like pseudo-code, with nesting instead of "gotos".

        o Removed note

             Note that the two URLs example.com and
             example.com:5060, while considered equal, may not lead
             to the same server, as the former causes a DNS SRV
             lookup, while the latter only uses the A record.
             since that is no longer the case.

        o Emphasized that proxies have to forward requests with unknown
          methods.

        o Aligned definition of call leg with URI comparison rules.

        o Required that second branch parameter be globally unique, so
          that a proxy can distinguish different branches in spiral
          scenarios similar to the following, with record-routing in
          place:

                 B  ---> P1 -------> P2 ------------> P1 ----------------> A
          BYE B   B/1      P1/2,B/1    P2/3,P1/2,B/1    P1/4,P2/3,P1/2,B/1


          Here, A/1 denotes the Via entry with host A and branch
          parameter 1. Also, this requires updating the definition of
          isomorphic requests, since the Request-URI is the same for all
          BYE that are record-routed.

        o Removed Via hiding from spec, for the following reasons:

          - complexity, particularly hidden "gotchas" that surface at
            various points (as in this instance);




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          - interference with loop detection and debugging;

          - Unlike HTTP, where via-hiding makes sense since all data is
            contained in the request or response, Via-hiding in SIP by
            itself does nothing to hide the caller or callee, as address
            information is revealed in a number of places:

            - Contact;

            - Route/Record-Route;

            - SDP, including the o= and c= lines;

            - possibly accidental leakage in User-Agent header and
              Call-ID headers.

          - Unless this is implemented everywhere, the feature is not
            likely to be very useful, without the sender having any
            recourse such as "don't route this request unless you can
            hide". It appears that almost all existing proxies simply
            ignore the Hide header.

        o Added Error-Info header field.

I Changes To Be Made

   NOTE: Almost all of these require further discussion or may already
   have been integrated into the main spec.

        o Record-Route needs to be clarified.

        o Bug: Consider a proxy that implements an ACD service. It
          proxies call requests to operators based on some complex
          logic. After receiving and INVITE, proxying it, and forwarding
          the response, the proxy eventually times out its state. Later
          on, an ACK (a late one, or one that has been wandering in the
          network) arrives. Where should the proxy send this? Ideally,
          the proxy should know where it sent the INVITE, and send it to
          the same place. This is not workable for complex forwarding
          logic. To fix this, the ACKs should go directly to the UAS.
          This will occur if a UAS inserts a Contact into the 200 OK,
          and if the UAC honors this in subsequent requests (including
          ACK). However, insertion of Contact, and honoring of Contact,
          are SHOULD, not MUST, so this "lost ACK" case is still
          possible in conditionally compliant implementations. TO fix
          this, (1) insertion of a Contact header, and (2) honoring of a
          Contact header, should be made MUST strength. (Q: is this the
          only solution???)



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        o Bug: Section 4.2.6 specifies that, for a REGISTER request, the
          user name of the Request-URI MUST be empty. While it does not
          mention anything about the "@" sign, the SIP-URL BNF is
          defined so that an "@" sign is used only in conjunction with a
          user ID. The implication that no "@" sign appears in a
          REGISTER Request-URI is never explicitly stated.

        o Issue: Usage of Content-Length. Spec is not that clear under
          what conditions its needed, particularly with TCP. This should
          be clarified. Another issue: should Content-Length just be
          made mandatory? It's not as a carryover from HTTP; do we have
          the same assumptions they do? Also, the section on Content-
          Length does not discuss behavior for requests over TCP that do
          not contain Content-Lengths (Answer: if no Content-Length is
          present in requests, the request packet must be terminated
          with a close connection. However, having the client close a
          connection is the same as sending a CANCEL, so no response is
          ever received. Now, a client *can* do this, but it makes no
          sense)

        o Issue: In REGISTER requests, should the request URI be a fully
          qualified hostname, or the domainname:


            REGISTER sip:machine.company.com SIP/2.0



          or

            REGISTER sip:company.com SIP/2.0



          It matters since registrars check if the domain name is
          there's before accepting the registration. The comparison
          might fail if the wrong thing is sent. Answer: probably a
          proxy/UAC SHOULD send it to company.com (and then use SRV to
          find the right machine), but a registrar SHOULD accept both
          fully qualified host names and domain names in comparisons. In
          fact, the registrar probably should be making this comparison
          on the To field, not the Request-URI!

        o Issue: A sends an INVITE to B, and B returns a 180 with a tag.
          Then, A sends BYE to B, without a tag in the To field. Should
          B accept this BYE or not? Answer: not sure; I think actually
          it says somewhere in the spec you are not supposed to initiate
          a new transaction until the previous completes.



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        o Bug: A INVITEs B through a proxy, B responds to A with a tag
          in the To field. Later, A sends a re-INVITE to B, with a tag
          in the To field, through the proxy. This request is malformed,
          so the proxy returns a 400 error. Normally, a proxy will
          insert a tag into the To field of a 400 response. But here, it
          cannot, since a tag is already present in the request. Thus,
          the rule should be that if a request arrives with a tag, the
          entity sending a response MUST NOT insert an additional tag,
          even if it a different entity than the one that created the
          tag in the first place.

        o Bug: The spec says that if a server receives a CANCEL for
          which it has no pending transaction, it returns an error.
          This, however, is not true for a proxy. A stateless proxy
          SHOULD forward the CANCEL as it would any non-INVITE request.

        o Table 5 lists WWW-Authenticate as a response-header allowed
          only in 401 responses. However, section 13.2 allows its reuse
          in requests containing Require: signed-response header. Also
          examples in section 14.1 have WWW-Authenticate in INVITE
          requests with no Require: present.

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

          This is also an inconsistency. If we do allow this, then WWW-
          Authenticate must be a request and response header field, and
          the Authorization header as well, since then it is included in
          a signed response.

        o Text on including Authorization in responses and WWW-
          Authenticate in requests needs to be added.

J Acknowledgments

   We wish to thank the members of the IETF MMUSIC and SIP WGs for their
   comments and suggestions. Detailed comments were provided by Brian
   Bidulock, Jim Buller, Neil Deason, Dave Devanathan, Cédric Fluckiger,
   Yaron Goland, Bernie Höneisen, Phil Hoffer, Christian Huitema, Jean
   Jervis, Gadi Karmi, Peter Kjellerstedt, Anders Kristensen, Jonathan
   Lennox, Gethin Liddell, Keith Moore, Vern Paxson, Moshe J. Sambol,
   Chip Sharp, Igor Slepchin, Robert Sparks, Eric Tremblay., and Rick
   Workman.

   This work is based, inter alia, on [46,47].




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K Authors' Addresses

   Mark Handley
   ACIRI
   electronic mail:  mjh@aciri.org

   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

   Jonathan Rosenberg
   dynamicsoft
   72 Eagle Rock Ave
   East Hanover, NJ 07936
   USA
   electronic mail:  jdrosen@dynamicsoft.com

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   Research and Experience , Vol. 4, pp. 99--120, June 1993.  ISI
   reprint series ISI/RS-93-359.

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


   Full Copyright Statement

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   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.




                           Table of Contents



   1          Introduction ........................................    2
   1.1        Overview of SIP Functionality .......................    2
   1.2        Terminology .........................................    3
   1.3        Overview of SIP Operation ...........................    3
   1.4        Definitions .........................................    4
   1.4.1      SIP Addressing ......................................    8
   1.4.2      Locating a SIP Server ...............................    9
   1.4.3      SIP Transaction .....................................   12
   1.4.4      Initiating a Session ................................   12
   1.4.5      Locating a User .....................................   13
   1.4.6      Changing an Existing Session ........................   16
   1.4.7      Registration Services ...............................   16
   1.5        Protocol Properties .................................   16
   1.5.1      Minimal State .......................................   16
   1.5.2      Lower-Layer-Protocol Neutral ........................   17
   1.5.3      Text-Based ..........................................   17
   2          SIP Uniform Resource Locators .......................   17
   2.1        SIP URL Comparison ..................................   22
   2.2        Non-SIP URLs ........................................   23
   3          SIP Message Overview ................................   23
   4          Request .............................................   25
   4.1        Request-Line ........................................   25
   4.2        Methods .............................................   25
   4.2.1      INVITE ..............................................   26
   4.2.2      ACK .................................................   29
   4.2.3      OPTIONS .............................................   30
   4.2.4      BYE .................................................   30
   4.2.5      CANCEL ..............................................   31
   4.2.6      REGISTER ............................................   32
   4.2.6.1    Where to Register ...................................   32
   4.2.6.2    REGISTER Header Fields ..............................   33
   4.2.6.3    Registering Contact Locations .......................   34
   4.2.6.4    Registration Expiration .............................   35
   4.2.6.5    List of Current Registrations .......................   35
   4.2.6.6    Removing Registrations ..............................   36
   4.3        Request-URI .........................................   36
   4.3.1      SIP Version .........................................   37
   4.4        Option Tags .........................................   37
   4.4.1      Registering New Option Tags with IANA ...............   38



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   5          Response ............................................   38
   5.1        Status-Line .........................................   39
   5.1.1      Status Codes and Reason Phrases .....................   39
   6          Header Field Definitions ............................   41
   6.1        General Header Fields ...............................   45
   6.2        Entity Header Fields ................................   46
   6.3        Request Header Fields ...............................   46
   6.4        Response Header Fields ..............................   46
   6.5        Header Field Format .................................   46
   6.6        Accept ..............................................   47
   6.7        Accept-Encoding .....................................   48
   6.8        Accept-Language .....................................   48
   6.9        Alert-Info ..........................................   49
   6.10       Allow ...............................................   49
   6.11       Also ................................................   50
   6.12       Authorization .......................................   50
   6.13       Call-ID .............................................   50
   6.14       Call-Info ...........................................   52
   6.15       Contact .............................................   52
   6.16       Content-Disposition .................................   55
   6.17       Content-Encoding ....................................   56
   6.18       Content-Language ....................................   57
   6.19       Content-Length ......................................   57
   6.20       Content-Type ........................................   58
   6.21       CSeq ................................................   58
   6.22       Date ................................................   60
   6.23       Encryption ..........................................   60
   6.24       Error-Info ..........................................   62
   6.25       Expires .............................................   62
   6.26       From ................................................   63
   6.27       In-Reply-To .........................................   65
   6.28       Max-Forwards ........................................   65
   6.29       MIME-Version ........................................   66
   6.30       Organization ........................................   66
   6.31       Priority ............................................   66
   6.32       Proxy-Authenticate ..................................   67
   6.33       Proxy-Authorization .................................   68
   6.34       Proxy-Require .......................................   68
   6.35       Record-Route ........................................   68
   6.35.1     Operation ...........................................   68
   6.35.2     Construction of Route Header ........................   69
   6.35.3     Request Destination .................................   70
   6.35.4     Syntax ..............................................   70
   6.35.5     Example .............................................   70
   6.36       Require .............................................   71
   6.37       Response-Key ........................................   72
   6.38       Retry-After .........................................   72
   6.39       Route ...............................................   73



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   6.40       Server ..............................................   73
   6.41       Subject .............................................   73
   6.42       Supported ...........................................   74
   6.43       Timestamp ...........................................   74
   6.44       To ..................................................   75
   6.45       Unsupported .........................................   76
   6.46       User-Agent ..........................................   77
   6.47       Via .................................................   77
   6.47.1     Requests ............................................   77
   6.47.2     Receiver-tagged Via Header Fields ...................   78
   6.47.3     Responses ...........................................   78
   6.47.4     User Agent and Redirect Servers .....................   79
   6.47.5     Forwarding Responses ................................   79
   6.47.6     Syntax ..............................................   79
   6.48       Warning .............................................   81
   6.49       WWW-Authenticate ....................................   83
   7          Status Code Definitions .............................   83
   7.1        Informational 1xx ...................................   83
   7.1.1      100 Trying ..........................................   84
   7.1.2      180 Ringing .........................................   84
   7.1.3      181 Call Is Being Forwarded .........................   85
   7.1.4      182 Queued ..........................................   85
   7.1.5      183 Session Progress ................................   85
   7.2        Successful 2xx ......................................   85
   7.2.1      200 OK ..............................................   85
   7.3        Redirection 3xx .....................................   85
   7.3.1      300 Multiple Choices ................................   86
   7.3.2      301 Moved Permanently ...............................   86
   7.3.3      302 Moved Temporarily ...............................   86
   7.3.4      305 Use Proxy .......................................   87
   7.3.5      380 Alternative Service .............................   87
   7.4        Request Failure 4xx .................................   87
   7.4.1      400 Bad Request .....................................   87
   7.4.2      401 Unauthorized ....................................   87
   7.4.3      402 Payment Required ................................   87
   7.4.4      403 Forbidden .......................................   88
   7.4.5      404 Not Found .......................................   88
   7.4.6      405 Method Not Allowed ..............................   88
   7.4.7      406 Not Acceptable ..................................   88
   7.4.8      407 Proxy Authentication Required ...................   88
   7.4.9      408 Request Timeout .................................   88
   7.4.10     409 Conflict ........................................   89
   7.4.11     410 Gone ............................................   89
   7.4.12     411 Length Required .................................   89
   7.4.13     413 Request Entity Too Large ........................   89
   7.4.14     414 Request-URI Too Long ............................   89
   7.4.15     415 Unsupported Media Type ..........................   89
   7.4.16     420 Bad Extension ...................................   89



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   7.4.17     480 Temporarily Unavailable .........................   90
   7.4.18     481 Call Leg/Transaction Does Not Exist .............   90
   7.4.19     482 Loop Detected ...................................   90
   7.4.20     483 Too Many Hops ...................................   90
   7.4.21     484 Address Incomplete ..............................   90
   7.4.22     485 Ambiguous .......................................   91
   7.4.23     486 Busy Here .......................................   91
   7.4.24     487 Request Terminated ..............................   91
   7.4.25     488 Not Acceptable Here .............................   92
   7.5        Server Failure 5xx ..................................   92
   7.5.1      500 Server Internal Error ...........................   92
   7.5.2      501 Not Implemented .................................   92
   7.5.3      502 Bad Gateway .....................................   92
   7.5.4      503 Service Unavailable .............................   92
   7.5.5      504 Server Time-out .................................   92
   7.5.6      505 Version Not Supported ...........................   93
   7.6        Global Failures 6xx .................................   93
   7.6.1      600 Busy Everywhere .................................   93
   7.6.2      603 Decline .........................................   93
   7.6.3      604 Does Not Exist Anywhere .........................   93
   7.6.4      606 Not Acceptable ..................................   93
   8          SIP Message Body ....................................   94
   8.1        Body Inclusion ......................................   94
   8.2        Message Body Type ...................................   94
   8.3        Message Body Length .................................   95
   9          Compact Form ........................................   95
   10         Behavior of SIP Clients and Servers .................   96
   10.1       Multicast Unreliable Transport Protocols ............   96
   10.1       Reliable Transport Protocols ........................   97
   10.2       Reliability for Requests Other Than INVITE ..........   98
   10.2.1     Unreliable Transport Protocols ......................   98
   10.2.2     Reliable Transport Protocol .........................   99
   10.3       Reliability for INVITE Requests .....................   99
   10.3.1     Unreliable Transport Protocols ......................  100
   10.3.2     Reliable Transport Protocol .........................  103
   10.4       ICMP Handling .......................................  103
   11         Behavior of SIP User Agents .........................  103
   11.1       Caller Issues Initial INVITE Request ................  103
   11.2       Callee Issues Response ..............................  104
   11.3       Caller Receives Response to Initial Request .........  104
   11.4       Caller or Callee Generate Subsequent Requests .......  104
   11.5       Receiving Subsequent Requests .......................  105
   12         Behavior of SIP Proxy and Redirect Servers ..........  106
   12.1       Redirect Server .....................................  106
   12.2       User Agent Server ...................................  107
   12.3       Proxy Server ........................................  107
   12.3.1     Proxying Requests ...................................  107
   12.3.2     Proxying Responses ..................................  108



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   12.3.3     Stateless Proxy: Proxying Responses .................  108
   12.3.4     Stateful Proxy: Receiving Requests ..................  108
   12.3.5     Stateful Proxy: Receiving ACKs ......................  109
   12.3.6     Stateful Proxy: Receiving Responses .................  109
   12.3.7     Stateless, Non-Forking Proxy ........................  109
   12.4       Forking Proxy .......................................  110
   13         SIP Authentication using HTTP Basic and Digest
   Schemes ........................................................  114
   13.1       Framework ...........................................  114
   13.2       Basic Authentication ................................  115
   13.3       Digest Authentication ...............................  115
   13.4       Proxy-Authentication ................................  116
   14         SIP Security Using PGP ..............................  117
   14.1       PGP Authentication Scheme ...........................  117
   14.1.1     The WWW-Authenticate Response Header ................  117
   14.1.2     The Authorization Request Header ....................  119
   14.2       PGP Encryption Scheme ...............................  120
   14.3       Response-Key Header Field for PGP ...................  121
   15         Examples ............................................  121
   15.1       Registration ........................................  121
   15.2       Invitation to a Multicast Conference ................  123
   15.2.1     Request .............................................  123
   15.2.2     Response ............................................  124
   15.3       Two-party Call ......................................  125
   15.4       Terminating a Call ..................................  127
   15.5       Forking Proxy .......................................  128
   15.6       Redirects ...........................................  132
   15.7       Negotiation .........................................  134
   15.8       OPTIONS Request .....................................  135
   A          Minimal Implementation ..............................  136
   A.1        Transport Protocol Support ..........................  136
   A.2        Client ..............................................  136
   A.3        Server ..............................................  137
   A.4        Header Processing ...................................  137
   B          Usage of the Session Description Protocol (SDP)
   ................................................................  138
   B.1        Configuring Media Streams ...........................  138
   B.2        Setting SDP Values for Unicast ......................  140
   B.3        Multicast Operation .................................  141
   B.4        Delayed Media Streams ...............................  141
   B.5        Adding and Deleting Media Streams ...................  142
   B.6        Putting Media Streams on Hold .......................  142
   B.7        Subject and SDP "s=" Line ...........................  142
   B.8        The SDP "o=" Line ...................................  143
   C          Summary of Augmented BNF ............................  143
   C.1        Basic Rules .........................................  145
   D          IANA Considerations .................................  148
   E          Changes from RFC 2543 ...............................  148



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   F          Changes Made in Version 00 ..........................  149
   G          Changes Made in Version 01 ..........................  156
   H          Changes Made in Version 02 ..........................  156
   I          Changes To Be Made ..................................  158
   J          Acknowledgments .....................................  160
   K          Authors' Addresses ..................................  161
   L          Bibliography ........................................  161












































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