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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-03.txt    ACIRI/Columbia U./Caltech/dynamicsoft
May 29, 2001
Expires: November 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", "MUST NOT", "REQUIRED",
   "SHALL", "SHALLNOT", "SHOULD", "SHOULD NOT", "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
   5.1) initiates sessions, ACK (Section 5.1.1) confirms session
   establishment, OPTIONS (Section 8) requests information about
   capabilities, BYE (Section 6) terminates a sessions, CANCEL (Section
   5.2) cancels a pending session and REGISTER (Section 7) 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.

   Proxies MAY modify any part of the SIP message that are not
   integrity-protected, except those needed to identify call legs.
   Proxies generally do not modify the session description, but MAY do
   so.

   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 10.34) 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.

        Back-To-Back User Agent: Also known as a B2BUA, this is a
             logical entity that receives an invitation, and acts as a
             UAS to process it. In order to determine how the request
             should be answered, it acts as a UAC and initiates a call
             outwards. A B2BUA appears like a proxy, but differs in that
             it maintains complete call state and must remain in a call.
             Since it is nothing more than a concatenation of other
             logical functions, no explicit definitions are needed for



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             its behavior.

        Call: A call consists of all participants in a session invited
             by a common source. A SIP call is identified by a globally
             unique call-id (Section 10.12), and is created when a user
             agent sends an INVITE request. This INVITE request may
             generate multiple acceptances, each of which are part of
             the same call (but different call legs). Furthermore, if a
             user is invited to the same multicast session by several
             people, each of these invitations will be a unique 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 a pairwise signaling relationship
             between two SUP usage agents. A call leg is established
             when a call invitation results in a successful response. It
             is identified by the combination of the Call-ID header
             field, the local address of the participant, and the remote
             address of the other participant. For the caller, the local
             address is the From field of the INVITE, and the remote
             address is the To field of the 200 class response. For the
             callee, the local address is the To field of the 200 class
             response to the INVITE, and the remote address is the From
             field of the INVITE. SIP URIs are compared according to
             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.

        Call Stateful: A proxy is said to be call stateful when it
             retains state that persists for the duration of a call
             initiated through it. To properly manage that state, the
             proxy will normally need to receive the BYE requests that
             terminate the call.

        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



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             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. A caller
             retains this role for the duration of a call.

        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. A callee retains this role for the duration of
             a call.

        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 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
             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.)




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        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
             through the location server.

        Regular Transaction: A regular transaction is any transaction
             with a method other than INVITE, ACK, or CANCEL.

        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



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             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 10.20)
             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.

        Spiral: A spiral is a SIP request which is routed to a proxy,
             forwarded onwards, and arrives once again at that proxy,
             but this time, with a Request-URI that differs from the
             previous arrival. A spiral is not an error condition,
             unlike a loop.

        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 for the duration of a SIP transaction. Also
             known as a transaction stateful proxy. The behavior of a
             stateful proxy is further defined in Section 17.3. A
             stateful proxy is not the same as a call stateful proxy.

        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 logical entity
             that initiates a SIP transaction with a request. This role
             lasts only for the duration of that transaction. In other
             words, if a piece of software initiates a request, it acts
             as a UAC for the duration of that request. If it receives a
             request later on, it takes on the role of a User Agent



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             Server for the processing of that transaction.

        User agent server (UAS): A user agent server is a logical entity
             that responds to a SIP request, generally acting on behalf
             of some user. The response accepts, rejects or redirects
             the request. This role lasts only for the duration of that
             transaction. In other words, if a piece of software
             responds to a request, it acts as a UAS for the duration of
             that request. If it generates a request later on, it takes
             on the role of a User Agent Client for the processing of
             that transaction.

        User agent (UA): A logical entity which acts as both a user
             agent client and user agent server for the duration of a
             call.

   An application program MAY be capable of acting both as a client and
   a server. For example, a typical multimedia conference control
   application would act as a 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
    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

   The Request-URI is determined according to the rules in Section 16
   and can be derived from either the Route, Contact or To header
   fields.

   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] and can be specified
   either as a set of parameters such as network address or host name,
   protocol port and transport protocol, or as a SIP URI.

   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.

   Clients MUST re-run the above selection algorithm, re-drawing any
   random numbers involved, once per transaction rather than for each
   request, i.e., requests within the same transaction MUST be sent to
   the same network address. Thus, the same address is used for the
   request, any retransmissions, any associated CANCEL requests and ACK
   requests for non-2xx responses. However, ACKs for 2xx responses use



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   another iteration of the selection algorithm. (Indeed, in many cases,
   they may have different request URIs.)

   A stateless proxy can accomplish this, 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.

   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
   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.

   If the client is configured with the address of an outbound proxy,
   the parameters of the outbound proxy, including transport protocol
   and port, become the destination used below.

   If there is no outbound proxy, the destination is the Request-URI.
   The destination address is the maddr parameter if it exists and the
   host element if not. The transport protocol is the transport
   parameter.

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

        1.   If the destination address is a numeric IP address, the
             client contacts the server at the given address and the
             port number specified in the SIP-URI or, if not specified,
             the default port (5060).

             If the destination specifies a protocol, the client
             contacts the server using that protocol. If no protocol is
             specified, the client first tries UDP. If attempt fails, or
             if the client does not support UDP but supports other



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             protocols, it tries those protocols in some
             implementation-defined order.

             The client then skips the remaining steps.

        2.   If the destination specifies no port number or port number
             5060, the transport protocol determines the use of one of
             the following three rules:

             - If the destination 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 above
               for stateless proxies. Message length or other request
               properties do not influence the server selection. 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 a transport protocol is specified and this protocol is
               supported by the client, the procedure in the paragraph
               above is used, limited to DNS resource records with the
               transport protocol specified in the SIP-URI.

             - If the transport protocol specified is not supported by
               the client, the client gives up.

             If there are no SRV records, the next step applies.

        3.   If the destination specifies 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. If there are address records, the same rules as
             in step 2 apply.

   Clients MUST NOT cache query results except according to the rules in



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   RFC 1035 [13].

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

        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 10.43)). 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 14). 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 10.46). Each
   proxy server in the forward path of the request forwards the response
   using these Via header fields, as described in detail in Sections
   10.46.3 and 10.46.4. For datagram protocols, reliability is achieved
   using retransmission (Section 14).

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 5.1) 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



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   has received that response by sending an ACK (Section 5.1.1) 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
   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




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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


   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, 7). A location server MAY also use one or
   more other protocols, such as finger (RFC 1288 [14]), rwhois (RFC
   2167 [15]), LDAP (RFC 1777 [16]), multicast-based protocols [17] or
   operating-system dependent mechanisms to actively determine the end
   system where a user might be reachable. A location server MAY return
   several locations because the user is logged in at several hosts
   simultaneously or because the location server has (temporarily)
   inaccurate information. The SIP server combines the results to yield
   a list of a zero or more locations.




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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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   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 10.14). A SIP proxy server can
   sequentially or in parallel try the addresses until the call is
   successful (2xx response) or the callee has declined the call (6xx
   response). With sequential attempts, a proxy server can implement an
   "anycast" service.

   If a proxy server forwards a SIP request, it MUST add itself to the
   beginning of the list of forwarders noted in the Via (Section 10.46)
   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 10.46).
   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



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   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 17. For efficiency, a
   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



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   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.


        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). Although the ABNF described in Section C uses
   implicit whitespace, unescaped whitespace MUST NOT be present within
   a SIP URL. Any 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 MUST NOT 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 [18]. 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



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


   Figure 3: SIP URL syntax


             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-@



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             part as a telephone number if local restrictions on the
             name space for user name allow it.

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

        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
             [19], TCP [20], TLS [21], 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:




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



             The transport, maddr, and ttl parameters MUST NOT be used
             in 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 MUST NOT 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 MUST NOT 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.


   Examples of SIP URLs are:

     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





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                  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
   headers        --            -      -    -       o         -          o


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

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, for which there is no default
          value, 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.

        o URL parameters that have no default value are compared only if
          they are present in both URLs.

   Thus, the following URLs are equivalent:



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


   while

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


   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 [18].

   In the following locations, only SIP URLs are allowed:

        o Request-URI in a REGISTER request;

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

   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 [22]). 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.




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   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.

   Unlike HTTP, SIP MAY use UDP or other unreliable datagram protocols.
   Each such datagram carries one request or response. Datagrams,
   including all headers, SHOULD NOT 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 [23] 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
        [24]). 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 9) messages use the
   generic-message format of RFC 822 [25] 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.





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

        start-line       =  Request-Line |     ;Section 4.1
                            Status-Line        ;Section 9.1




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



   In the interest of robustness, any leading empty line(s) MUST be
   ignored. In other words, if the Request or Response message begins
   with 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 12


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 MUST NOT be enclosed in "<>".  absoluteURI
   is defined in [H3.2.1].



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



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        Request-URI   =  SIP-URL | absoluteURI
        SIP-Version   =  "SIP/2.0"


4.2 Methods

   The methods are described in detail below: REGISTER 7 for registering
   contact information, INVITE, ACK and CANCEL (Section 5.1) for setting
   up sessions, BYE (Section 6) for terminating sessions and OPTIONS
   (Section 8) for querying servers about their capabilities. SIP
   extensions may define additional methods ("extension-method").

   Proxy and redirect servers treat all methods other than INVITE and
   CANCEL, whether the method is defined in this specification or
   elsewhere, in the same way. 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 (Not Implemented) response to be returned (Section 11).
   As in HTTP, the Method token is case-sensitive.



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


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 MUST NOT 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



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        general-header   =  Accept               ; Section 10.6
                         |  Accept-Encoding      ; Section 10.7
                         |  Accept-Language      ; Section 10.8
                         |  Call-ID              ; Section 10.12
                         |  Call-Info            ; Section 10.13
                         |  Contact              ; Section 10.14
                         |  CSeq                 ; Section 10.20
                         |  Date                 ; Section 10.21
                         |  Encryption           ; Section 10.22
                         |  From                 ; Section 10.25
                         |  MIME-Version         ; Section 10.28
                         |  Organization         ; Section 10.29
                         |  Record-Route         ; Section 10.34
                         |  Require              ; Section 10.35
                         |  Supported            ; Section 10.41
                         |  Timestamp            ; Section 10.42
                         |  To                   ; Section 10.43
                         |  User-Agent           ; Section 10.45
                         |  Via                  ; Section 10.46
        entity-header    =  Allow                ; Section 10.10
                         |  Content-Disposition  ; Section 10.15
                         |  Content-Encoding     ; Section 10.16
                         |  Content-Language     ; Section 10.17
                         |  Content-Length       ; Section 10.18
                         |  Content-Type         ; Section 10.19
                         |  Expires              ; Section 10.24
        request-header   =  Alert-Info           ; Section 10.9
                         |  Authorization        ; Section 10.11
                         |  In-Reply-To          ; Section 10.26
                         |  Max-Forwards         ; Section 10.27
                         |  Priority             ; Section 10.30
                         |  Proxy-Authorization  ; Section 10.32
                         |  Proxy-Require        ; Section 10.33
                         |  Route                ; Section 10.38
                         |  Response-Key         ; Section 10.36
                         |  Subject              ; Section 10.40
        response-header  =  Error-Info           ; Section 10.23
                         |  Proxy-Authenticate   ; Section 10.31
                         |  Retry-After          ; Section 10.37
                         |  Server               ; Section 10.39
                         |  Unsupported          ; Section 10.44
                         |  Warning              ; Section 10.47
                         |  WWW-Authenticate     ; Section 10.48


   Table 3: SIP headers



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        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
   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 [18]. 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



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   SIP.  These tags are used in Require (Section 10.35), Supported
   (Section 10.41) and Unsupported (Section 10.44) header fields.

   Syntax:


        option-tag  =  token


   See Section C for the definition of token. The creator of a new SIP
   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 MUST NOT 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;




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        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 [26].

5 INVITE, ACK and CANCEL

5.1 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 MUST NOT send
   an updated session description in an ACK request if it had already
   sent a session description in the INVITE request.  If the UAC wishes
   to modify the session after the call setup has begun, it MUST
   initiate another INVITE transaction after the current one has
   completed.


        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



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   the user is already participating in, identified either by the SIP
   Call-ID or a globally unique identifier within the session
   description, with a 200 (OK) response.

   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 MUST NOT issue another INVITE request for the same call leg
   before the previous INVITE transaction has completed. A UAS that
   receives an INVITE before it sent the final response to an INVITE
   with a lower CSeq number on the same call leg 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



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   Retry-After header field specifying when the request should be
   resubmitted.


        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.

5.1.1 ACK

   The ACK request confirms that the client has received a final
   response to an INVITE request. (ACK is used only with INVITE
   requests.) Treatment of ACK for a 200 class response differs
   significantly from that of a non-200 class response. 2xx responses
   are acknowledged by client user agents, all other final responses by
   the first stateful 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 or UAC to receive a non-2xx final response. For a
   non-200 class response, the Via in the ACK that is constructed MUST
   be the same as the request being acknowledged. The ACK for a 200
   class response will contain Route headers if Record-Route headers
   were present in the response. An ACK for a non-200 class response
   never contains Route headers.  The ACK request for a 200 class
   response is forwarded as the corresponding INVITE request, based on



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   its Request-URI or Route headers, 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 Request-URI for the
   ACK is set to the top entry in the route set for a 200 class response
   (see Section 16). For a non-200 class response, the Request-URI MUST
   be the same as the Request-URI in the request being acknowledged.

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

   The ACK request for a 200 class response MAY contain a message body
   with the final session description to be used by the callee. See
   Section 5.1 for further details on the relationship between 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.  However, an ACK not destined for the proxy SHOULD NOT
   be retransmitted.


        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.

5.2 CANCEL

   The CANCEL request cancels a pending request with the same Call-ID,
   To, From, top Via header and Request-URI 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.




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   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 stateful proxy that receives a CANCEL request immediately responds
   with a 200 class response. It then generates a new CANCEL, and
   forwards the request to all destinations with pending requests. A
   stateless proxy, or a stateful proxy with no transaction state for
   the cancelled request, proxies the CANCEL request to the same set of
   destinations the original request was proxied to.

   The Request-URI, topmost Via, Call-ID, To, the numeric part of CSeq
   and From header fields in the CANCEL request are identical to those
   in the original request being cancelled, including tags. 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
   responds to the original request with a 487 (Request Terminated),
   following normal rules for response retransmissions defined in
   Section 14. 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.  If the
   UAS or redirect server has no record of the request being cancelled,
   the CANCEL is responded to with a 481.

   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



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        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.

6 BYE

   The user agent client uses BYE to indicate to the server that it
   wishes to release the call leg. A BYE request is forwarded like an
   INVITE request and MAY be issued by either caller or callee.  A BYE
   request SHOULD NOT be sent to terminate a pending call request which
   has not generated either a final response or a provisional response
   containing a To tag. A party to a call SHOULD issue a BYE request
   before releasing a call ("hanging up"). A party receiving a BYE
   request MUST cease transmitting media streams specifically directed
   at the party issuing the BYE request.

   A UAS receiving a BYE request MUST respond to any pending requests
   received for that call, including INVITE. It is RECOMMENDED that a
   487 response is generated.

   This method SHOULD be supported by user agent servers.

7 Registrars, Registrations and the REGISTER Method

   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 18.2).

7.1 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.




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        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
             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 MUST 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 [17]; 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.

7.2 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



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             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.

        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.

7.3 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.  It is RECOMMENDED that registrars authorize whether
             the entity in the From field is allowed to register
             addresses for the address-of-record in the To field.

        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



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             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
             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.

7.4 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 request and response 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.

   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. 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



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   registration.

   The registrar determines the expiration time; it may be longer or
   shorter than the one requested by the registrand. The REGISTER
   response contains the actual registration lifetime; the client MUST
   refresh at least as often and SHOULD NOT refresh more frequently. In
   general, the server SHOULD honor the expiration time offered by the
   user agent. A server MAY decide to lengthen the expiration interval
   if, for example, the refresh rate of a particular client exceeds a
   threshold.

   This behavior is different from RFC 2543, which only allowed
   registrars to decrease, but not increase, the interval.


        Allowing the registrar to set the registration interval
        protects it against excessively frequent registration
        refreshes while limiting the state that it needs to
        maintain and decreasing the chance for stale registrations
        that require proxying effort.

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

7.5 List of Current Registrations

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

7.6 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 "*". This usage is only allowed in REGISTER
   requests when a Expires header with value of zero is present.

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

8 OPTIONS

   The OPTIONS method is used to query a server 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



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   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 10.12. 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.

9 Response

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



        Response  =  Status-Line        ;  Section 9.1
                     *( general-header
                     | response-header
                     | entity-header )
                     CRLF
                     [ message-body ]   ;  Section 12


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

9.1 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


9.1.1 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



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   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
        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 11. 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.



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   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.  However, proxies SHOULD distinguish
   100 from other 1xx responses.  (The former SHOULD NOT be forwarded,
   while the latter MUST be.  See Section 17.3.)  For example, if a
   client receives an unrecognized response code of 431, it can safely
   assume that there was something wrong with its request and treat the
   response as if it had received a 400 (Bad Request) response code. In
   such cases, user agents SHOULD present to the user the message body
   returned with the response, since that message body is likely to
   include human-readable information which will explain the unusual
   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






10 Header Field Definitions




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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 Terminated
                     |   "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"  ;  Server Time-out
                     |   "505"  ;  SIP Version not supported


   Figure 7: Server error status codes


   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



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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


   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
   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
   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 MUST NOT 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 MUST NOT 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 required if the message body
   is not empty. See sections 10.18, 10.19 and 12 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 10.1) and entity header (Section 10.2) 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



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   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 MUST NOT alter any fields that are authenticated (see Section
   18.2), 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 18.2 apply. Proxies
   SHOULD NOT re-order header fields.



   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 MUST NOT 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 13 for use
   over 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.

10.1 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.

10.2 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.

10.3 Request 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                2xx          -   -   -   o   o   o
        Accept-Encoding        R           -   o   o   o   o   o
        Accept-Encoding       2xx          -   -   -   o   o   o
        Accept-Encoding       415          -   o   o   o   o   o
        Accept-Language        R           -   o   o   o   o   o
        Accept-Language       2xx          -   -   -   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
        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
        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

   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.


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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          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

   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.

10.4 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-


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   header" fields if all parties in the communication recognize them to
   be "response-header" fields. Unrecognized header fields are treated
   as "entity-header" fields.

10.5 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 [25]. Each header field consists of a
   name followed by a colon (":") and the field value. Field names are
   case-insensitive. The field value MAY be preceded by any amount of
   leading white space (LWS), though a single space (SP) is preferred.
   Header fields can be extended over multiple lines by preceding each
   extra line with at least one SP or horizontal tab (HT). Applications
   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 MUST NOT
   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



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   contained within these brackets. If the URL is not enclosed in angle
   brackets, any semicolon-delimited parameters are header-parameters,
   not URL parameters.

10.6 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
   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



10.7 Accept-Encoding

   The Accept-Encoding general-header field is similar to Accept, but
   restricts the content-codings [H3.5] 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.




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10.8 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.

   Example:


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



10.9 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>



10.10 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.



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   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


10.11 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. The Authorization field value consists of credentials
   containing the authentication information of the user agent for the
   realm of the resource being requested.

   Section 18.2 overviews the use of the Authorization header field, and
   Section 19 describes the syntax and semantics when used with HTTP
   Basic and Digest authentication.

10.12 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 SHOULD NOT alert
   the user if the user has responded previously to the Call-ID in the
   INVITE request. If the user is already a member of the conference and
   the conference parameters contained in the session description have
   not changed, a callee user agent server MAY silently accept the call,
   regardless of the Call-ID. An invitation for an existing Call-ID or
   session 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.

   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.



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   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 MUST NOT be
   reused for later calls. Use of cryptographically random identifiers
   [28] 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
   13. 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



10.13 Call-Info




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   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 [29] or LDIF [30] 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



10.14 Contact

   Among the methods discussed in this specification, the Contact
   general-header field can appear in INVITE, OPTIONS, ACK, and REGISTER
   requests, and in 1xx, 2xx, 3xx, and 485 responses. Other methods
   defined elsewhere may allow or require the use of the Contact header
   field. This is generally necessary if the recipient of this method
   needs to send requests to the originator. 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 a single Contact header indicating a
             single URI from which location the request is originating.



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             The URI SHOULD contain the address of the client itself
             (i.e., its IP address, or a FQDN for the host, or an SRV
             record with the highest priority entry beingan FQDN of that
             host). See Section 16 for usage of the Contact header for
             routing subsequent requests. 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 MUST NOT 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 single
             Contact response header field indicating a single SIP URI
             under which it is reachable most directly for future SIP
             requests, such as ACK, within the same call leg. The URI
             SHOULD contain the address of the server itself (i.e., its
             IP address, or a FQDN for the host, or an SRV record with
             the highest priority entry beingan FQDN of that host). See
             Section 16 for usage of the Contact header for routing
             subsequent requests.

             If a UA supports both UDP and TCP, it SHOULD NOT indicate a
             transport parameter in the URI.


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

        REGISTER requests and responses: See Section 7. The Contact
             header value of "*" is only used in REGISTER requests.

        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



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             transport parameters. A 300 (Multiple Choices), 301 (Moved
             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, [31]) 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



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             years) as equivalent to 2**32-1.



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

   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 <">
                  | extension-attribute

   extension-attribute = extension-name [ "=" extension-value ]


   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



10.15 Content-Disposition



        Content-Disposition   =  "Content-Disposition" ":"



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                                 disposition-type *( ";" disposition-param )
        disposition-type      =  "render" | "session" | "icon" | "alert"
                             |   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 [32]).

   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.

10.16 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



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   content codings have been applied to the entity-body, and thus what
   decoding mechanisms MUST be applied in order to obtain the media-type
   referenced by the Content-Type header field.  Content-Encoding is
   primarily used to allow a 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.

10.17 Content-Language

   See [H14.12].

10.18 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



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   assume that the request encompasses the remainder of the packet. If a
   server receives a datagram request with a Content-Length, but the
   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 12
   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.

10.19 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 MUST be present if the
   body is not empty. If the body is empty, and a Content-Length header
   is present, it indicates that the body of the specific type has zero
   length (for example, if it is an emtpy audio file).



        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



10.20 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. The sequence number MUST
   be expressible as a 32-bit unsigned integer. A server MUST echo the
   CSeq value from the request in its response. The CSeq header serves
   to order transactions within a call leg, and to provide a means to
   uniquely identify transactions.



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



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   For requests that are outside of a call leg, or for a request that
   initiates a session, the value of the sequence number is arbitrary,
   but MUST be less than 2**31. For requests which are subsequent ones
   within an existing call leg (such as a re-INVITE or BYE), the CSeq
   header MUST contain strictly monotonically increasing and contiguous
   (increasing-by-one) sequence numbers; sequence numbers do not wrap
   around. Retransmissions of the same request carry the same CSeq
   value.

   For requests outside of a call leg, ordering is irrelevant, and so
   the value of the CSeq number in requests received by a UAS is not
   important. For requests within a call leg, ordering is important.
   Therefore, a UAS MUST remember the highest sequence number for any
   request received within a call leg. The server MUST reject, using a
   400 class response, any request within a call leg with a lower
   sequence number. Any request that is received with a sequence number
   higher than the highest received so far (even it is higher by more
   than one), SHOULD be accepted.

   If a client initiates a session, and receives multiple 200 class
   responses, each establishes a separate call leg. For subsequent
   requests within each of those call legs (each of which differs only
   by the tag in the To field), the CSeq numbers increment independently
   from the other call legs. Furthermore, the CSeq numbering space is
   unique in each direction. That is, the CSeq values in requests from A
   to B are independent of the values in requests from B to A.

   The ACK request MUST contain the same CSeq numeric value as the
   INVITE request that it refers to, but with a Method of "ACK". The
   CANCEL request MUST contain the same CSeq numeric value as the
   request it cancels, but with a Method of "CANCEL".

   The Method value allows the client to distinguish the response to a
   CANCEL request from that of the request it is cancelling. 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.

   Example:



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     CSeq: 4711 INVITE



10.21 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 [33] 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.

10.22 Encryption

   The Encryption general-header field specifies that the content has
   been encrypted. Section 18 describes the overall SIP security
   architecture and algorithms. This header field is intended for end-
   to-end encryption of requests and responses. Requests are encrypted
   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.



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   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 18.

   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
   otherwise identical un-encrypted request.

10.23 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 ))


10.24 Expires

   The Expires entity-header field gives the date and time after which



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   the message content expires.

   This header field is currently defined only for the REGISTER, as
   described in Section 7, and INVITE methods.

   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



10.25 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 MUST contain the "tag" parameter. However, a
   server MUST be prepared to receive a request without a tag, in which



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   case the tag is considered to effectively have a value of zero. This
   is to maintain backwards compatibility with RFC2543, which did not
   mandate From tags. .  The server copies the From header field from
   the request to the response. The 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 MUST NOT 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 10.5.



        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;tag=887s
     From: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8



   The "tag" value MUST be globally unique and cryptographically random
   with at least 32 bits of randomness. It SHOULD differ for each call
   leg.

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

        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 10. Only
          parameters that appear in both header fields are compared.





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        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 [25] header, but with a
        URI instead of just an email address.

10.26 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



10.27 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


   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



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   recipient MUST NOT 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



10.28 MIME-Version

   See [H19.4.1].

10.29 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


10.30 Priority

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



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


   It is RECOMMENDED that the value of "emergency" only be used when



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   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 [34], with the addition of
        "emergency".

10.31 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 19 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.


        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.




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10.32 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 19 for a
   discussion of its usage.

10.33 Proxy-Require

   The Proxy-Require header field is used to indicate proxy-sensitive
   features that MUST be supported by the proxy. If a proxy 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. A UAC SHOULD attempt to retry the request,
   without using the features listed in the Unsupported header.

   See Section 10.35 for more details on the mechanics of this message
   and a usage example.



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


10.34 Record-Route

   The Record-Route header field has the following syntax:


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


   Details of its use are described in Section 16.




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10.35 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. A UAC SHOULD attempt to retry the request,
   without using the features listed in the Unsupported header.



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


   Example:

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

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




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

   Proxy and redirect servers MUST ignore features that are not
   understood. If a particular extension requires that intermediate
   devices support it, the extension MUST be tagged in the Proxy-Require
   field as well (see Section 10.33).

10.36 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:



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        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 18 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.

10.37 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.

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



        Retry-After  =  "Retry-After" ":" ( 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



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     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.

10.38 Route

   The Route header field has the following syntax:


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


   Details of its use are described in Section 16.

10.39 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].

10.40 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
   the same subject indication as the invitation.)



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


   Example:


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



10.41 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.




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        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


10.42 Timestamp

   The Timestamp general-header field describes when the client sent the
   request to the server. The client uses the current time value at the
   time of transmission, i.e., each retransmission of a request is
   likely to have a different timestamp value.

   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
   in all provisional and final responses 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) ]


   Note that there MUST NOT be any LWS between a DIGIT and the decimal
   point.

10.43 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,



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

   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 MUST NOT
   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
   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 10.25 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 15 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 MUST NOT be
   inserted.

   Section 10.25 describes how To and From header fields are compared
   for the purpose of matching requests to call legs.




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   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.

10.44 Unsupported

   The Unsupported response-header field lists the features not
   supported by the server. See Section 10.35 for a usage example and
   motivation.

   Syntax:


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


10.45 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 [H14.43].

10.46 Via




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   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.

10.46.1 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 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 17.3.1.

10.46.2 Receiver-tagged Via Header Fields

   A proxy or UAS receiving a request SHOULD check the first Via header
   field to ensure that it contains the sender's correct network
   address, as seen from that proxy. If the Via header contains a domain
   name or if it contains an IP address that differs from the packet
   source address, the proxy or UAS SHOULD add a "received" attribute to
   that Via header field.


        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.
        Similarly, a request traversing a network address
        translator (NAT) will also cause the sending address to



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        differ from the address seen by the next hop. The mechanism
        described here is unlikely to be sufficient, however, for
        allowing packets to traverse a NAT in the reverse
        direction.

   An example is:


     Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
     Via: SIP/2.0/UDP 128.59.16.1:5060 ;received=128.59.19.3



   In this example, the message originated from a multi-homed host with
   two addresses, 128.59.16.1 and 128.59.19.3. The sender guessed wrong
   as to which network interface would be used. Erlang.bell-
   telephone.com noticed the mismatch, and added a parameter to the
   previous hop's Via header field, containing the address that the
   packet actually came from.

10.46.3 Receiving Responses

   Via header fields in responses received 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. Specifically, the sent-by
             value should equal the value inserted by the proxy or UAC.
             The recevied parameter MUST NOT be used by a proxy or UAC
             to determine if the response is for a request it sent. If
             the sent-by value is not equal to the value inserted by the
             proxy or UAC, discard the message. Otherwise, remove this
             Via field.

        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 10.46.5.

10.46.4 Generating Responses

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

10.46.5 Sending Responses

   Given a destination described by a Via header field, the response is



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   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. If no
          connection is open, open a connection to the IP address in the
          received parameter, if present using the port in the sent-by
          value, or port 5060 if none is present. If the connection
          attempt fails, or if there was no received parameter, the
          server SHOULD attempt to open a connection to the address in
          the sent-by value, which may be a domain name. To do this, it
          constructs a SIP URL of the form "sip:<sent-
          by>;transport=<sent-protocol>" and then uses the procedures
          defined in [35] to determine the IP address and port to open
          the connection and send the response to.

        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 10.46.2),
          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. If this fails, e.g.,
          elicits an ICMP "port unreachable" response, send the response
          to the address in the "sent-by" parameter. The address to send
          to is determined by constructing a SIP URL of the form
          "sip:<sent-by>", and then using the DNS procedures defined in
          [35] to send the response.

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

   Note that the response to an unreliable datagram request is not
   returned to the port from which the request came, but it is always
   returned to the source IP that that request came from.

10.46.6 Syntax

   The format for a Via header field is shown in Fig. 9. 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 10.46.2).




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


   Figure 9: Syntax of Via header field



   The "branch" parameter is included by every proxy. The token MUST be
   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 17.3.1), 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 hash SHOULD also
   include any other fields the proxy uses to make a routing decision on
   the request. This is to ensure that if the request is routed back to
   the proxy, and one of those fields changes, it is treated as a
   spiral, and not a loop.  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



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        request-URI, or any other headers used for routing, to
        distinguish looped requests from requests whose request-URI
        (or whatever headers are used for routing) is changed and
        which then reach a server visited earlier.

   CANCEL and ACK requests MUST have the same branch value as the
   corresponding request they cancel or acknowledge. When a response
   arrives at the proxy it can use the branch value to figure out which
   branch the response corresponds to.


     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



10.47 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



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   MUST NOT 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.

   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.



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        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
             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 MUST NOT take any
             automated action.


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

   If the warning is caused by the session description, the status
   response SHOULD include a session description similar to that
   included in OPTIONS responses indicating the capabilities of the UAS.

   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'"



10.48 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 [H14.47] for a
   definition of the syntax, and Section 19 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-



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   use these values on the next request for that destination.

11 Status Code Definitions

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

11.1 Informational 1xx

   Informational responses indicate that the server or proxy contacted
   is performing some further action and does not yet have a definitive
   response. The client SHOULD wait for a further response from the
   server, and the server SHOULD send such a response without further
   prompting. 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



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   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.

11.1.1 100 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.

11.1.2 180 Ringing

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

11.1.3 181 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.

11.1.4 182 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.

11.1.5 183 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.

11.2 Successful 2xx

   The request was successful and MUST terminate a search.

11.2.1 200 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.




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        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.

11.3 Redirection 3xx

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

11.3.1 300 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 10.14).  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.

11.3.2 301 Moved Permanently



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   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 10.14). 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.

11.3.3 302 Moved Temporarily

   The requesting client SHOULD retry the request at the new address(es)
   given by the Contact header field (Section 10.14).  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 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 10.24) header. If there is no explicit expiration time, the
   address is only valid for this call and MUST NOT be cached for future
   calls.

11.3.4 305 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.

11.3.5 380 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.

11.4 Request Failure 4xx

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



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11.4.1 400 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".

11.4.2 401 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.

11.4.3 402 Payment Required

   Reserved for future use.

11.4.4 403 Forbidden

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

11.4.5 404 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.

11.4.6 405 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.

11.4.7 406 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.

11.4.8 407 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 10.31) containing a
   challenge applicable to the proxy for the requested resource. The
   client MAY repeat the request with a suitable Proxy-Authorization



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   header field (section 10.32). SIP access authentication is explained
   in section 18.2 and 19.

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

11.4.9 408 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.

11.4.10 409 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.

11.4.11 410 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.

11.4.12 413 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.

11.4.13 414 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.

11.4.14 415 Unsupported Media Type




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   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.  The client SHOULD retry the request, this time omitting any
   bodies not supported by the server.

11.4.15 420 Bad Extension

   The server did not understand the protocol extension specified in a
   Proxy-Require (Section 10.33) or Require (Section 10.35) header
   field.

11.4.16 480 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.

11.4.17 481 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.) A UAC receiving a 481 to a
   request sent for an existing call leg MUST consider that call leg
   terminated.

11.4.18 482 Loop Detected

   The server received a request with a Via (Section 10.46) path
   containing itself.

11.4.19 483 Too Many Hops




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

11.4.20 484 Address Incomplete

   The server received a request with a To (Section 10.43) 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
        string. It sends strings of increasing lengths, prompting
        the user for more input, until it no longer receives a 484
        status response.

11.4.21 485 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.

11.4.22 486 Busy Here

   The callee's end system was contacted successfully but the callee is



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   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.

11.4.23 487 Request Terminated

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

11.4.24 488 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.

11.5 Server Failure 5xx

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

11.5.1 500 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.

11.5.2 501 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.)

11.5.3 502 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.

11.5.4 503 Service Unavailable



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

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

11.5.5 504 Server Time-out

   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.

11.5.6 505 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.

11.5.7 513 Message Too Large

   The server was unable to process the request since the message length
   exceeded its capabilities.

11.6 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.

11.6.1 600 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



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   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.

11.6.2 603 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.

11.6.3 604 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.

11.6.4 606 Not Acceptable

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

   A 606 (Not Acceptable) response means that the user wishes to
   communicate, but cannot adequately support the session described. The
   606 (Not Acceptable) response MAY contain a list of reasons in a
   Warning header field describing why the session described cannot be
   supported. Reasons are listed in Section 10.47.  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.

12 SIP Message Body

12.1 Body Inclusion

   Requests MAY contain message bodies unless otherwise noted. In this
   specification, the CANCEL request MUST NOT 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



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   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 11.3. 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

12.2 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.

   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.

12.3 Message Body Length

   The body length in bytes SHOULD be given by the Content-Length header
   field. Section 10.18 describes the behavior in detail.

   The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
   (Note: The chunked encoding modifies the body of a message in order
   to transfer it as a series of chunks, each with its own size
   indicator.)

13 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



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   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 20.2 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
     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 MUST NOT be done to fields following an
   Authorization header.

14 Behavior of SIP Clients and Servers

14.1 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



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   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 10.46).

   To avoid response implosion, servers MUST NOT 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
   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.

14.2 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



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   open at least until the first final response arrives.

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

   If the server leaves the connection open, and if the client so
   desires it MAY re-use the connection for further SIP requests. These
   requests can be for the same transaction or call, or for totally
   different transactions or calls. There is no requirement that a
   transaction must complete before a new one is initiated on an
   existing connection. As a result, a server MUST support receiving a
   request for a new transaction on an existing connection before the
   previous transaction on the same connection has completed.

   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.

14.3 Reliability for Requests Other Than INVITE

14.3.1 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 SHOULD NOT 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



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        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
   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 10.42), 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 14.4, responses to these requests are not retransmitted
   periodically and not acknowledged via ACK.

14.3.2 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.

14.4 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.




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

        3.   The client has to be able to terminate an on-going request,
             e.g., because it is no longer willing to wait for the
             connection or search to succeed. The server will have to
             wait several 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".

14.4.1 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.  If no response
   (final or provisional) is received after sending seven request
   packets, processing continues as if a 481 response was received for
   that request (no ACK is generated, however), and a CANCEL SHOULD NOT
   be sent.

   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 entity
   generating the ACK (a UAC for 2xx responses, UAC or proxy for non-
   2xx) retransmits the ACK on receipt of a response retransmission. The
   value of a final response is not changed by the arrival of a BYE or
   CANCEL request.

   Servers SHOULD only send a single 401 or 407 status response upon
   receiving a request that is not authenticated at either the SIP,
   transport or network layer. (See Section 18.4)

   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



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   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. 10 and 11 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 14.3.




        The mechanism in Sec. 14.3 would not work well for INVITE
        because of the long delays between INVITE and a final
        response. If the 200 response were to get lost, the callee
        would believe the call to exist, but the voice path would
        be dead since the caller does not know that the callee has
        picked up. Thus, the INVITE retransmission interval would
        have to be on the order of a second or two to limit the
        duration of this state confusion. Retransmitting the
        response with an exponential back-off helps ensure that the
        response is received, without placing an undue burden on
        the network.

14.4.2 Reliable Transport Protocol

   A user agent using a reliable transport protocol such as TCP, SCTP or
   TLS MUST NOT retransmit requests, but uses the same algorithm as for
   unreliable transport protocols (Section 14.4.1) 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



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              +===========+
              *           *
              *  Initial  *
              *           *
              +===========+
                    |
                    |    -
                    |  INVITE
                    |
                    v
              *************
    T1*2^n <--*           *
    INVITE -->*  Calling  *--------+
              *           *        |
              *************        |
                |   |              |
                |   | 1xx          | 7 INVITE SENT
          +-----+   |  -           |    -
          |         |              |
          |         v              |
    status|   *************        |
      ACK |   *    Call   *        |
          |   * Proceeding*<->1xx  |
          |   *           *    -   |
          |   *************        |
          |         |              |
          +-----+   | status       |
                |   | ACK          |
               \/   v              |
              *************        |
     status<--*           *        |
      ACK  -->* Completed *<-------+
              *           *
              *************

key for transitions:   event
                       request sent





   Figure 10: State transition diagram of client for INVITE method






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                +===============+
                *               *
                *   Initial     *
                *               *
                +===============+
                        |
                        | INVITE
                        |  1xx
                       \/
      INVITE -->*****************
        1xx  <--*               *---> status change
                *  Proceeding . *<---  1xx
          +---- *               *
  termntd |     *****************
    487   |           |   |
          |           |   |
          |   failure |   |  picks up
          |   >= 300  |   |    2xx
          |   +-------+   +-----------------+
         \/   v                             v
        ***********    min(T1*2^N,T2)   ***********
 INVITE<*         *---->   status  <----*         *--> INVITE
 status>* failure *<----           ---->* success *<--   2xx
        *         *                     *         *
       ;***********    ACK              ***********
          |     |       -                   |   |
          |     +--------+------------------+   |
          |              |                      | 32s
          +--------------|----------------------+  -
          |              |
          |             \/
          |       **************
          |       * confirmed  *
          |       *            *
          |       **************
          |              | 32s
          |              |  -
          |             \/
          |       **************
          +------>* completed  *
                  *            *
     event        **************
  message sent




   Figure 11: State transition diagram of server for INVITE method

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        responses, although that is not strictly necessary.

14.5 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.

15 Behavior of SIP User Agents

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

15.1 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. It MUST
   contain a tag. 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.

   If the UAC desires to end the call before a response is received to
   the INVITE, it SHOULD send a CANCEL. This CANCEL will normally result
   in a 487 response to be returned to the INVITE, indicating successful
   cancellation. However, it is possible that the CANCEL and a 200 class
   response to the INVITE "pass on the wire". In this case, the UAC will
   receive a 2xx to the INVITE. It then terminates the call by following
   the procedures described in Section 15.4.

15.2 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.



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   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 tags. These
   become the local and remote addresses of the call leg, respectively.

15.3 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, with a distinct call leg identifier. The call leg identifier is
   defined as the combination of the remote address, local address, and
   Call-ID. The local address is the value of the From field, including
   the tag, in the 2xx responses (they will all be the same). The remote
   address is the value of the To field, including the tag. Each 2xx
   response to the INVITE will differ in the value of the tag in the To
   field.

   The UAC MUST generate an ACK request for each distinct call leg
   created by a 2xx. The Request-URI and Route headers for the ACK are
   constructed as described in Section 16.4. The To field in the ACK
   MUST contain the remote address for the call leg (which includes the
   tag). The From field in the ACK MUST contain the local address for
   the call leg. The Call-ID MUST contain the Call-ID for the call leg.
   The Via header in the ACK MUST be indentical to the one in the
   request being acknowledged, including any branch parameter. The CSeq
   number MUST be the same as the INVITE being acknowledged, but the
   CSeq method MUST be ACK. The ACK might possibly require a session
   description in the body. See Section B for guidelines.

   After acknowledging, the caller MAY choose to terminate the call leg
   with a responding UAS by sending a BYE request. Procedures for doing
   so are defined in Section 15.4.

15.4 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. It
   MAY initiate other requests as needed. A UA MUST NOT initiate a new
   INVITE transaction within a call leg while one is in progress. A UA
   MUST NOT initiate a new regular transaction while a regular
   transaction is in progress. However, a UA MAY initiate a regular
   transaction while an INVITE transaction on the same call leg is in
   progress.

   Regardless of whether the caller or callee is generating the new
   request, the header fields in the request are set as follows. For the



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   desired call leg, the To header field is set to the remote address,
   and the From header field is set to the local address (both including
   tags). A UAC copies the tag from the final response into the ACK, but
   it MUST NOT 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.

   For an INVITE, the Contact header field MAY be different than the
   Contact header field sent in a previous response or request.

   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 10.34, 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.

15.5 Receiving Subsequent Requests

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



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   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.

             If the UAS wishes to reject the INVITE, because it does not
             wish to recreate the call, it MUST respond to the request
             with a 481 status code. A UAC receiving a 481 response for
             any mid-call request (INVITE or otherwise) MUST consider
             that call terminated.

        2.   If the To, From, Call-ID, CSeq, Request-URI, and branch-ID
             in the topmost Via 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.
             It is possible for the CSeq header to be higher than the
             previous by more than one. This is not an error condition,
             and a UAS SHOULD be prepared to receive and process
             requests with CSeq values more than one higher than the



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             previous received request. A request on an existing call
             leg with a lower CSeq MUST be rejected.

16 Routing of Requests

   Record routing is the process whereby a proxy server can request to
   receive all messages between two UAs for a particular call leg. The
   proxy accomplishes this by inserting a header, called Record-Route
   into the initial request that begins the call leg. The user agents
   use these headers to construct a set of Route headers, that gets
   inserted into subsequent requests in the call leg. The Route headers
   contain a set of proxies that the request must visit on its way from
   one UA to another. Proxies use these to forward the requests, much
   like strict IP source routing.

   The process of record-routing works for any SIP request that
   initiates some kind of session. For this specification, that includes
   only INVITE. Extensions MAY identify new requests as ones which
   initiate sessions, in which case the procedures defined here apply to
   processing of those requests.

   All user agents MUST support the processing rules below which apply
   to them. As such, they MUST be able to parse and process both
   Record-Route and Route headers. Proxies MAY support the record
   routing procedures of Section 16.3, but they MUST support the route
   header procedures of Sectionsec:rr:proxy2.


        This is a change from RFC2543, where all record-route and
        route processing was optional for user agents.

   The syntax for the Route header is described in Section 10.38. The
   syntax for the Record-Route header is described in Section 10.34.

16.1 UAC Processing for initial transaction

   The UAC formulates its initial request for the session as defined in
   section 15.1. If the final response is a 200 class response, it may
   contain Record-Route headers, and may contain a Contact header.

        Contact was not mandatory in RFC2543. Thus, if the UAC is
        talking to an older UAS, the UAS might not insert the
        Contact header. Thus, this text says that the Contact "may"
        be present in the response. Note also that this may is
        lower case; it is NOT saying that a UAS compliant to this
        specification can optionally insert the Contact header into
        the 200-class response.  The UAC MUST construct a route set
        , defined as a list of URIs, in the following manner:



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        1.   The list of URIs present in the Record-Route headers in the
             200 class response are taken, if present, and their order
             is reversed.

        2.   The URI in the Contact header from the 200 class response,
             if present, is taken, and appended to the end of the list
             from the previous step.

        3.   The list of URIs resulting from the above two operations is
             referred to as the route set

   The UAC MUST store the route set for the duration of the call leg.
   NOTE that it is possible for the route set to be empty. This will
   occur if neither Record-Route headers nor a Contact header were
   present in the 200 class response. Since there may be multiple 200 OK
   responses to an INVITE request, each response constitutes a separate
   call leg, and thus has a separate route set. The UAC MUST also
   remember whether the bottom-most entry in the route set was
   constructed from a Contact header or not. This is a boolean value,
   which we refer to as CONTACT_SET.

   An ACK request for the 200 class response to an initial INVITE
   transaction MUST be formulated according to the rules of Section
   15.3, as if it were a subsequent request within the call leg. That
   is, the ACK for a 200 class response contains Route headers.

   Record-Route headers MAY be present in a provisional response to
   INVITE. In this case, the UAC can construct a route set for the
   call-leg associated with that provisional response, in the same way
   it would construct a route set for a 200 class response. This route
   set will only be needed if the UAC sends a request to the far end of
   the call leg before the initial INVITE transaction completes.

16.2 UAS Processing of initial transaction

   When a UAS receives a request for a new call session, and it responds
   with a 200 class response, it MUST copy the contents of the Record-
   Route headers from the request to the response. This includes the
   URIs, URI parameters, and any Record-Route header parameters, whether
   they are known or unknown to the UAS.

        Record-Route parameters are very useful for proxies. They
        allow the proxy to match the record-route entry in the
        response with the one in the request.  The order of the
        Record-Route headers MUST be preserved in the response.

   The UAS MUST construct a route set in the following manner:




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        1.   The list of URIs in the Record-Route headers in the initial
             request, if present, are taken, including any URI
             parameters.

        2.   The URI in the Contact header from the request, if present,
             is taken, including any URI parameters. The URI is appended
             to the bottom of the list of URIs from the previous step.

        3.   The resulting list of URIs is called the route set

   The UAS MUST store the route set for the duration of the call leg.
   It is possible for the route set to be empty. This will occur if
   neither Record-Route headers nor a Contact header were present in the
   initial request. The UAS MUST also remember whether the bottom-most
   entry in the route set was constructed from a Contact header or not.
   This is effectively a boolean value, which we refer to as
   CONTACT_SET.

   If the UAS sends provisional responses before the session is
   accepted, it SHOULD copy the Record-Route headers from the request
   into the provisional responses in the same manner described above for
   the 200 class response.

16.3 Proxy procedures for record routing a transaction

   Each proxy MAY independently decide to record-route a transaction
   that initiates a session. Amongst the methods defined in this
   specification, that includes only the INVITE transaction. However,
   extensions MAY designate new methods as ones that initiate a session
   of some sort. In that case, the procedures described here apply to
   those requests. Both the initial request that initiates the session,
   and any refreshes (such as a re-INVITE) MUST be record-routed if the
   the initial request was record routed. This means a proxy will often
   need to record-route requests that contain Route headers.  Generally,
   the choice about whether to record-route or not is a tradeoff of
   features vs. performance.  Faster request processing and higher
   scalability is achieved when proxies do not record route. However,
   provision of certain services may require a proxy to observe all
   messages for a call leg. It is RECOMMENDED that proxies do not
   automatically record route. They should do so only if specifically
   required.

   To record-route, the proxy inserts a Record-Route header into the
   request before proxying it onwards. A forking proxy MAY insert a
   different Record-Route header into each forked request. The Record-
   Route header that it inserts MUST be inserted as the first Record-
   Route header, appearing before any existing ones in the request. The
   URL in the header MUST be a SIP URL.  MUST NOT contain the transport



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   parameter.  The URL MUST have the property that when the processing
   described in [35] is followed for that URL, the result of the lookup
   is an address/port of the proxy server inserting the Record-Route.
   The URL and the proxy configuration SHOULD be such that if a request
   is received with this URL in the Request-URI, the proxy's normal
   request processing will cause it to be forwarded to one of the
   previous-hop servers that the request traversed, including the UAS.

        These two properties are important. The first one
        guarantees that subsequent requests from the called party
        are routed back to this actually proxy. The second property
        is there for robustness. It guarantees that the request URI
        always contain meaningful information, even if there are no
        Route headers that tell the proxy where to forward the
        request to next.  The URL placed into the Record-Route
        header MUST be unique for each unique Request-URI in the
        request, but must not equal the Request-URI of the request
        (they will not be equal if the proxy adds an maddr
        parameter).  The proxy MAY insert Record-Route header
        parameters into the request. These will be returned to the
        proxy in any 200 class response to the INVITE, and are
        useful for pushing state into the message.

   If a 200 class response arrives for the proxied request, the response
   will contain the entire list of Record-Route headers inserted by
   proxies along the request path. The proxy MAY modify the Record-Route
   value matching the one it inserted into the request. Like the URL in
   the request, it MUST be a SIP URL and MUST NOT contain a transport
   parameter. The URL in this Record-Route header MUST have the property
   that when the processing described in [35] is followed for that URL,
   the result of the lookup is an address/port of the proxy server that
   inserted the Record-Route. The URL in this header, and the proxy
   configuration SHOULD be such that if a request is received with this
   URL in the Request-URI, the proxy's normal request processing will
   cause it to be forwarded to the same next hop server that the
   original request was forwarded to. The URL placed into the Record-
   Route header MUST be unique for each unique Request-URI in the
   request, but must not equal the Request-URI of the request (they will
   not be equal if the proxy adds an maddr parameter).

   The four properties (two for the request, two for the response), can
   be satisfied in a number of ways. One way is that the URL inserted
   into the Record-Route in the request is nearly the same as the
   Contact header in the initial request (if present, else the From
   field), but with the maddr and port set to resolve to the proxy, and
   with a transaction identifier added to the user part of the request-
   URI (in order to meet the requirement that the URI in the Record-
   Route be different for each distinct Request-URI). Then, the proxy



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   modifies the URL in the Record-Route header in the response, setting
   it to be the URL from the Request-URI of the initial request, but
   with the maddr and port set to resolve to the proxy.

   As an example, consider a proxy at 10.0.1.1 listening on port 5061
   which receives the following request (many headers are omitted for
   brevity):


   INVITE sip:user@example.com SIP/2.0
   Via:  SIP/2.0/UDP callerspc.univ.edu
   Contact:  sip:caller@callerspc.univ.edu



   The proxy forwards this request to sip:j_user@div11.example.com, and
   record-routes:


   INVITE sip:j_user@div11.example.com SIP/2.0
   Via: SIP/2.0/UDP 10.0.1.1:5061
   Via: SIP/2.0/UDP callerspc.univ.edu
   Record-Route: <sip:caller.8jjs0@callerspc.univ.edu:5061;maddr=10.0.1.1>
   Contact:  sip:caller@callerspc.univ.edu



   The 200 response received by the proxy will look like, in part:


   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.0.1.1:5061
   Via: SIP/2.0/UDP callerspc.univ.edu
   Record-Route: <sip:caller.8jjs@callerspc.univ.edu:5061;maddr=10.0.1.1>
   Contact: sip:j_user@host32.div11.example.com



   The proxy modifies its Record-Route header in the response, resulting
   in the new response forwarded upstream:


   SIP/2.0 200 OK
   Via:  SIP/2.0/UDP callerspc.univ.edu
   Record-Route:  <sip:j_user@example.com:5061;maddr=10.0.1.1>
   Contact:  sip:j_user@host32.div11.example.com





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   The route set computed by the UAS is:


   sip:caller.8jjs@callerspc.univ.edu:5061;maddr=10.0.1.1
   sip:caller@callerspc.univ.edu



   and the route set computed by the UAC is:


   sip:j_user@example.com:5061;maddr=10.0.1.1
   sip:j_user@host32.div11.example.com



   There are other ways to meet these requirements. The proxy could
   construct a URL for the request which encodes all of the needed
   information, by placing it in the user portion, for example. A call
   stateful proxy could insert a URL into the request with the form
   sip:proxy.example.com, and not modify it in the response. This URL
   clearly satisfies the first required property (of getting routed back
   to the proxy that inserted it). The second property can be maintained
   by being call stateful, and extracting the needed parameters from
   local storage.

   When a proxy does decide to modify the Record-Route header in the
   response, one of the operations it must perform is to locate the
   Record-Route that it had inserted. If the request spiraled, and the
   proxy inserted a Record-Route in each iteration of the spiral,
   locating the correct header in the response (which must be the proper
   iteration in the reverse direction) is tricky. Note that the rules
   above dictate that a proxy insert a different URI into the Record-
   Route for each distinct Request-URI received. The two issues can be
   solved jointly. A RECOMMENDED mechanism is for the proxy to append a
   piece of data to the user portion of the URL. This piece of data is a
   hash of the transaction key for the incoming request, concatenated
   with a unique identifier for the proxy instance. Since the
   transaction key includes the Request-URI, this key will be unique for
   each distinct Request-URI. When the response arrives, the proxy
   modifies the first Record-Route whose identifier matches the proxy
   instance. The modification results in a URI without this piece of
   data appended to the user portion of the URI. Upon the next
   iteration, the same algorithm (find the topmost Record-Route header
   with the parameter) will correctly extract the next Record-Route
   header inserted by that proxy.

16.4 UA Processing of Subsequent Requests in a Call Leg



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   When a UA wishes to send another request for the call-leg (such as a
   BYE or INVITE), it follows the procedures defined in this subsection.
   The procedures here MUST also be followed for an ACK request for a
   200 response to the initial INVITE for a call leg. The procedures
   here MUST NOT be followed for a CANCEL request or an ACK request for
   a non-200 class response. This implies that CANCEL for an initial
   INVITE never contains a Route header, nor does an ACK for a non-200
   response.

   The request is constructed as specified in Section 15.4.  The UA then
   takes the list of URI in the route set. The top URI is inserted into
   the request URI of the request, including all parameters. Any URI
   parameters not allowed in the request URI MUST then be stripped. Each
   of the remaining URIs (if any) from the route set, including all URI
   parameters, are placed into a Route header into the request, in
   order. The procedures of [35] are then applied to the URI in the
   request URI, the the request is forwarded to the resulting server.

   If a UAS has a route set for a call leg, and receives a refresh for
   that call leg containing Record-Route headers (the only refresh
   defined in this specification is a re-INVITE), it MUST copy those
   headers into any 200 class response to that request. If the boolean
   variable CONTACT_SET is true, the Contact header in the request (if
   present) replaces the last entry in the route set. If the boolean
   variable CONTACT_SET is false, the UAS MUST add the URL in the
   Contact header in the re-INVITE to the bottom of the route set, and
   then set CONTACT_SET to true. If the request did not contain a
   Contact header, the route-set at the UAS remains unchanged.

   Similarly, if a UAC has a route set for a call leg, and receives a
   200 class response to a refresh it sent, the Contact header is
   examined. If not present, the route set remains unchanged. If the
   response had a Contact header, and the boolean variable CONTACT_SET
   is false, the URL in the Contact header in the response is added to
   the bottom of the route set, and CONTACT_SET is set to true. If the
   re-INVITE response had a Contact header, and CONTACT_SET is true, the
   URL in the Contact header of the re-INVITE response replaces the
   bottom value in the route set.

   The above two paragraphs allow a UA to update its Contact address
   mid-call, but proxies cannot update their route address. Once on the
   route, a proxy remains on the route for the duration of the call leg.

        Why the different treatment for Contact and Record-Route in
        a re-INVITE? It has to do with backwards compatibility.
        RFC2543 did not mandate that a proxy needs to refresh its
        record-route headers. As a result, the lack of a record-
        route in a re-INVITE cannot be interpreted to mean that the



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        proxy does not want to be included on the route any longer.
        Updating of the Contact header mid-call is useful for
        mobility applications, and is also useful for forms of
        third party call control

16.4.1 Local outbound proxies

   Special considerations exist for local outbound proxies.

   A UA which uses a local outbound proxy will send all requests without
   Route headers to that proxy. Typically, this includes initial INVITE
   requests for a call.

   If a local outbound proxy wishes to remain on the SIP messaging path
   for a call leg, it MUST record-route using the procedures above. .

   A UA which uses a local outbound proxy, and attempts to send a
   subequent request in a call leg with a route set, SHOULD use the
   procedures in Section 16.4. However, in some instances, a UA may not
   be capable of DNS, and therefore may not be able to follow those
   procedures. In this case, the UA MAY send the request to its local
   outbound proxy. In this case, it MUST NOT remove the top Route
   header. It sets the Request-URI to the same value it used for the
   initial request, and sends it to its local outbound proxy.

16.5 Proxy routing procedures

   The rules in this section MUST be followed by all proxies to
   determine the appropriate routing procedures to apply to a request.
   Proxies MAY respond to requests with Route headers for any reason (in
   order to perform proxy authorization, for example). But, if the
   request is to be proxied, the procedures here MUST be used to
   determine where the request is proxied to. The procedures in this
   section apply indepedent of the request method.  They MUST be
   followed even if the request method is unknown to the proxy.  Since
   CANCEL can never contain Route, these procedures never apply to
   CANCEL. However, they do apply to ACK requests for a "200 OK"
   response, which do contain Route headers.

   When a proxy receives a request, it MUST check for the existence of a
   Route header. If one exists, it MUST pop that Route header, and place
   it (including all URI parameters) into the Request-URI. Any URL
   parameters present in the top Route header which are not allowed in
   the request URI MUST be removed from the Request-URI by the proxy.It
   SHOULD be sent using the procedures of [35]. If no Route header is
   present, the proxy examines the Request-URI. If it contains an maddr
   parameter, and the address in maddr is not an interface that the
   proxy is listening on, the proxy SHOULD forward the request using the



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   procedures of [35] on the Request-URI. This will cause it to be
   forwarded to the address in the maddr. If the maddr is present, but
   is an interface the proxy is listening on, the proxy MUST strip the
   maddr, and then continue processing as if it were never there. If
   there was no maddr (or if it was stripped in the previous step), and
   the domain of the Request-URI is not a domain the proxy is managing,
   the procedures of [35] SHOULD be used to forward the request to that
   URI. If the domain is one the proxy is managing, the request is
   processed by whatever policies are desired by the administrator.

16.6 Pre-Loaded Route Headers

   Normally, a UA constructs the Route headers in a request from the
   route set learned through Record-Route headers. However, in some
   circumstances, it is useful for a UA to insert Route headers into an
   initial request. These headers may have been learnt by the UA through
   some out of bands means. When an initial request (initial as far as
   the UAC and UAS are concerned) contains Route headers, this is
   referred to as a "pre-loaded Route". It is equivalent to strict
   source routing in IP. Proxies will often not be able to distinguish
   this case from the case described in Section 16.3, and will therefore
   properly use these route headers to forward the request. If the
   proxies are interested in receiving subsequent messages for the call
   leg, their insertion of Record-Route as mandated by Section 16.3 will
   establish a correct route set at both UAC and UAS. This route set may
   end up being different from the pre-loaded Route used by the UAC. As
   such, a UAC that inserts a pre-loaded route set MUST follow the
   procedures of Section 16.1 in processing the response to this initial
   INVITE.

17 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.

17.1 Redirect Server

   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.

17.2 User Agent Server



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

17.3 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 MUST be stateful.
   Proxies that accept TCP connections MUST be stateful when handling
   the TCP connection. A proxy sending a request out to a multicast
   address MUST be stateful.


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

   A stateful proxy SHOULD NOT 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.

17.3.1 Proxying Requests

   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 10.46.6 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 MUST NOT forward a request to a multicast group which
   already appears in any of the Via headers.




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   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 10.46).

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

17.3.2 Proxying Responses

   A proxy only processes a response if the topmost Via field matches
   one of its addresses (see Section 10.46). A response with a non-
   matching top Via field MUST be dropped.

17.3.3 Stateless Proxy: Proxying Responses

   A stateless proxy MUST follow the procedures in Section 10.46 in
   order to determine where to forward the response to.

   A stateless proxy MUST NOT generate its own provisional responses. It
   MUST forward all provisional responses, including 100, upstream.

17.3.4 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.

17.3.5 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.

17.3.6 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



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   requests. If there is no match, the response is forwarded upstream.
   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 17.4 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.

17.3.7 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 17.4.

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

17.4 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



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   10.34).

   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 */
       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++) {



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         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);
         }

         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.



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            */
           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
   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 MUST forward provisional responses greater than
             100 upstream towards the client and SHOULD NOT forward 100



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             (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.  (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
             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.



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   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.

18 Security Considerations

18.1 Confidentiality and Privacy: Encryption

18.1.1 End-to-End Encryption

   SIP requests and responses can contain sensitive information about
   the communication patterns and communication content of individuals.
   The SIP message body MAY also contain encryption keys for the session
   itself. SIP supports three complementary forms of encryption to
   protect privacy:

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

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

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

   End-to-end encryption relies on keys shared by the two user agents
   involved in the request. Typically, the message is sent encrypted
   with the public key of the recipient, so that only that recipient can



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   read the message.

   A SIP request (or response) is end-to-end encrypted by splitting the
   message to be sent into a part to be encrypted and a short header
   that will remain in the clear. Some parts of the SIP message, namely
   the request line, the response line and certain header fields marked
   with "r" in the "proxy" column in Table 4 and 5 need to be read and
   returned by proxies and thus MUST NOT be encrypted end-to-end.
   Possibly sensitive information that needs to be made available as
   plaintext include destination address (To) and the forwarding path
   (Via) of the call. The Authorization header field MUST remain in the
   clear if it contains a digital signature as the signature is
   generated after encryption, but MAY be encrypted if it contains
   "basic" or "digest" authentication.

   Other header fields MAY be encrypted or MAY travel in the clear as
   desired by the sender. The Subject, Allow and Content-Type header
   fields will typically be encrypted. The Accept, Accept-Language,
   Date, Expires, Priority, Require, Call-ID, Cseq, and Timestamp header
   fields will remain in the clear.

   All fields that will remain in the clear MUST precede those that will
   be encrypted. The message is encrypted starting with the first
   character of the first header field that will be encrypted and
   continuing through to the end of the message body. If no header
   fields are to be encrypted, encrypting starts with the second CRLF
   pair after the last header field, as shown below. Carriage return and
   line feed characters have been made visible as "$", and the encrypted
   part of the message is outlined.


     INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP 169.130.12.5$
     To: T. A. Watson <sip:watson@bell-telephone.com>$
     From: A. Bell <sip:a.g.bell@bell-telephone.com>;tag=7abm$
     Encryption: PGP version=5.0$
     Content-Length: 224$
     Call-ID: 187602141351@worcester.bell-telephone.com$
     Content-Type: message/sip
     CSeq: 488$
     $
   *******************************************************
   * Subject: Mr. Watson, come here.$                    *
   * Content-Type: application/sdp$                      *
   * $                                                   *
   * v=0$                                                *
   * o=bell 53655765 2353687637 IN IP4 128.3.4.5$        *
   * s=Mr. Watson, come here.$                           *



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   * t=0 0$                                              *
   * c=IN IP4 135.180.144.94$                            *
   * m=audio 3456 RTP/AVP 0 3 4 5$                       *
   *******************************************************



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

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

   A Content-Type indication of "message/sip" MAY be added, but will be
   overridden after receipt.

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


     INVITE sip:watson@boston.bell-telephone.com SIP/2.0$
     Via: SIP/2.0/UDP 169.130.12.5$
     To: T. A. Watson <sip:watson@bell-telephone.com>$
     From: A. Bell <a.g.bell@bell-telephone.com>;tag=7abm$
     Encryption: PGP version=5.0$
     Content-Type: application/sdp$
     Content-Length: 107$
     Call-ID: 187602141351@worcester.bell-telephone.com$
     CSeq: 488$
     $
   *************************************************
   * $                                             *
   * v=0$                                          *
   * o=bell 53655765 2353687637 IN IP4 128.3.4.5$  *
   * c=IN IP4 135.180.144.94$                      *
   * m=audio 3456 RTP/AVP 0 3 4 5$                 *
   *************************************************






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18.1.2 Privacy of SIP Responses

   SIP requests can be sent securely using end-to-end encryption and
   authentication to a called user agent that sends an insecure
   response.  This is allowed by the SIP security model, but is not a
   good idea.  However, unless the correct behavior is explicit, it
   would not always be possible for the called user agent to infer what
   a reasonable behavior was. Thus, when end-to-end encryption is used
   by the request originator, the encryption key to be used for the
   response SHOULD be specified in the request (Section 10.36). If this
   were not done, it might be possible for the called user agent to
   incorrectly infer an appropriate key to use in the response. Thus, to
   prevent key-guessing becoming an acceptable strategy, we specify that
   a called user agent receiving a request that does not specify a key
   to be used for the response SHOULD send that response unencrypted.

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

18.1.3 Encryption by Proxies

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


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

18.1.4 Hop-by-Hop Encryption

   SIP requests and responses MAY also be protected by security
   mechanisms at the transport or network layer. No particular mechanism
   is defined or recommended here. Two possibilities are IPSEC [39] or
   TLS [21]. The use of a particular mechanism will generally need to be
   specified out of band, through manual configuration, for example.

18.2 Message Integrity and Access Control: Authentication

   Protective measures need to be taken to prevent an active attacker
   from modifying and replaying SIP requests and responses. The same
   cryptographic measures that are used to ensure the authenticity of
   the SIP message also serve to authenticate the originator of the
   message.  However, the "basic" and "digest" authentication mechanism



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   offer authentication only, without message integrity.

   Transport-layer or network-layer authentication MAY be used for hop-
   by-hop authentication. SIP also extends the HTTP WWW-Authenticate
   (Section 10.48) and Authorization (Section 10.11) header field and
   their Proxy counterparts to include cryptographically strong
   signatures. SIP also supports the HTTP "basic" and "digest" schemes
   (see Section 19) and other HTTP authentication schemes to be defined
   that offer a rudimentary mechanism of ascertaining the identity of
   the caller.

   SIP requests MAY be authenticated using the Authorization header
   field to include a digital signature of certain header fields, the
   request method and version number and the payload, none of which are
   modified between client and called user agent. The Authorization
   header field is used in requests to authenticate the request
   originator end-to-end to proxies and the called user agent, and in
   responses to authenticate the called user agent or proxies returning
   their own failure codes. If required, hop-by-hop authentication can
   be provided, for example, by the IPSEC Authentication Header.

   Generally, SIP authentication is for a specific request URI and
   realm, a protection domain. Thus, for basic and digest
   authentication, each such protection domain has its own set of user
   names and secrets. If a user agent does not care about different
   request URIs, it makes sense to establish a "global" user name,
   secret and realm that is the default challenge if a particular
   request URI does not have its own realm or set of user names.
   Similarly, SIP entities representing many users, such as PSTN
   gateways, MAY try a pre-configured global user name and secret when
   challenged, independent of the request URI.

   SIP does not dictate which digital signature scheme is used for
   authentication. As indicated above, SIP implementations MAY also use
   "basic" and "digest" authentication and other authentication
   mechanisms defined for HTTP [40]. Note that "basic" authentication
   has severe security limitations. The following does not apply to
   these schemes.

   To cryptographically sign a SIP request, the order of the SIP header
   fields is important. When an Authorization header field is present,
   it indicates that all header fields following the Authorization
   header field have been included in the signature.  Therefore, hop-
   by-hop header fields which MUST or SHOULD be modified by proxies MUST
   precede the Authorization header field as they will generally be
   modified or added-to by proxy servers.  Hop-by-hop header fields
   which MAY be modified by a proxy MAY appear before or after the
   Authorization header. When they appear before, they MAY be modified



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   by a proxy. When they appear after, they MUST NOT be modified by a
   proxy. To sign a request, a client constructs a message from the
   request method (in upper case) followed, without LWS, by the SIP
   version number, followed, again without LWS, by the request headers
   to be signed and the message body.  The message thus constructed is
   then signed.

   For example, if the SIP request is to be:

   INVITE sip:watson@boston.bell-telephone.com SIP/2.0
   Via: SIP/2.0/UDP 169.130.12.5
   Authorization: PGP version=5.0, signature=...
   From: A. Bell <sip:a.g.bell@bell-telephone.com>;tag=7abm
   To: T. A. Watson <sip:watson@bell-telephone.com>
   Call-ID: 187602141351@worcester.bell-telephone.com
   Subject: Mr. Watson, come here.
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=bell 53655765 2353687637 IN IP4 128.3.4.5
   s=Mr. Watson, come here.
   t=0 0
   c=IN IP4 135.180.144.94
   m=audio 3456 RTP/AVP 0 3 4 5



   Then the data block that is signed is:

   INVITESIP/2.0From: A. Bell <sip:a.g.bell@bell-telephone.com>;tag=7abm
   To: T. A. Watson <sip:watson@bell-telephone.com>
   Call-ID: 187602141351@worcester.bell-telephone.com
   Subject: Mr. Watson, come here.
   Content-Type: application/sdp
   Content-Length: ...

   v=0
   o=bell 53655765 2353687637 IN IP4 128.3.4.5
   s=Mr. Watson, come here.
   t=0 0
   c=IN IP4 135.180.144.94
   m=audio 3456 RTP/AVP 0 3 4 5



   Clients wishing to authenticate requests MUST construct the portion
   of the message below the Authorization header using a canonical form.



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   This allows a proxy to parse the message, take it apart, and
   reconstruct it, without causing an authentication failure due to
   extra white space, for example. Canonical form consists of the
   following rules:

        o No short form header fields;

        o Header field names are capitalized as shown in this document;

        o No white space between the header name and the colon;

        o A single space after the colon;

        o Line termination with a CRLF;

        o No line folding;

        o No comma separated lists of header values; each must appear as
          a separate header;

        o Only a single SP between tokens, between tokens and quoted
          strings, and between quoted strings; no SP after last token or
          quoted string;

        o No LWS between tokens and separators, except as described
          above for after the colon in header fields;

        o The To and From header fields always include the < and >
          delimiters even if the display-name is empty.

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

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

   The correct behavior in handling unauthenticated responses to a
   request that requires authenticated responses is described in section
   18.2.1.

18.2.1 Trusting responses

   There is the possibility that an eavesdropper listens to requests and
   then injects unauthenticated responses that terminate, redirect or
   otherwise interfere with a call. (Even encrypted requests contain



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   enough information to fake a response.)

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

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

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

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

   These attacks are prevented by having the client require response
   authentication and dropping unauthenticated responses. A server user
   agent that cannot perform response authentication responds using the
   normal Require response of 420 (Bad Extension).

18.3 Callee Privacy

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

18.4 Denial of Service



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   Attackers can spoof a Via header field to direct responses to a third
   party, using a SIP UAS or proxy to generate traffic. This attack can
   be prevented by requiring an existing security association, such as
   TLS or IPsec. This may be an appropriate solution, e.g., between
   proxy servers that exchange significant amounts of signaling traffic
   or between a user agent and its outbound proxy.

   If such a security association is not feasible, clients and proxies
   SHOULD respond to unauthenticated requests with only a single 401
   (Unauthorized) or 407 (Proxy Authentication Required) instead of
   using the normal response retransmission algorithm. Retransmitting
   the 401 or 407 status response amplifies the problem of an attacker
   using a spoofed Via header address to to direct traffic to a third
   party.

18.5 Known Security Problems

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

19 SIP Authentication using HTTP Basic and Digest Schemes

   SIP implementations MAY use HTTP's basic and digest authentication
   mechanisms (RFC 2617 [40]) 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 [40]. This
   section outlines this operation, pointing to RFC 2617 [40] for
   details, and noting the differences when used in SIP. Since RFC 2543
   is based on HTTP basic and digest as defined in RFC 2069 [41], SIP
   servers supporting RFC 2617 MUST ensure they are backwards compatible
   with RFC 2069. Procedures for this backwards compatibility are
   specified in RFC 2617.

19.1 Framework

   The framework for SIP authentication parallels that for HTTP (RFC
   2617 [40]). 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 MUST NOT, and instead MAY use the 407 response. The
   requirements for inclusion of the Proxy-Authenticate, Proxy-



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   Authorization, WWW-Authenticate, and Authorization in the various
   messages is identical to RFC 2617 [40].

   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.

19.2 Basic Authentication

   The rules for basic authentication follow those defined in [40] but
   with the words "origin server" replaced with "user agent server,
   redirect server , or registrar".

   Since SIP URIs are not hierarchical, the paragraph in [40] 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.

   Due to its weak security, the usage of basic authentication is NOT
   RECOMMENDED. However, servers SHOULD support it to handle older RFC
   2543 clients that might still use it.

19.3 Digest Authentication



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   The rules for digest authentication follow those defined in [40],
   with "HTTP 1.1" replaced by "SIP/2.0" in addition to the following
   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 [40] regarding cache operation does
             not apply to SIP.

        7.   RFC 2617 [40] 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.

19.4 Proxy-Authentication

   The use of the Proxy-Authentication and Proxy-Authorization parallel
   that as described in [40], with one difference. Proxies MUST NOT 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



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   proxy which has asked for authentication.


        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 10.31 and 10.32 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.

20 Examples

   In the following examples, we often omit the message body and the
   corresponding Content-Length and Content-Type headers for brevity.

20.1 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
         From: <sip:watson@bell-tel.com>;tag=19al
         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:



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   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=19al
         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=19al
         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
         From: <sip:jon.diligent@bell-tel.com>;tag=7eff
         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.

20.2 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



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   [6]) as the session description format.

20.2.1 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=1ija
         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



   The From request header above states that the request was initiated
   by alice@wonderland.com and addressed to bob@example.com (To 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.



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20.2.2 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 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=1ija
         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.


   S->C: SIP/2.0 200 OK
         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=1ija
         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:





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   C->S: ACK sip:bob@one.example.com SIP/2.0
         Via: SIP/2.0/UDP mouse.wonderland.com
         From: Alice <sip:alice@wonderland.com>;tag=1ija
         To: Bob <sip:bob@example.com> ;tag=3141593
         Call-ID: 602214199@mouse.wonderland.com
         CSeq: 1 ACK



20.3 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=3pcc
         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
         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=3pcc
         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



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         Via: SIP/2.0/UDP kton.bell-tel.com
         From: A. Bell <sip:a.g.bell@bell-tel.com>;tag=3pcc
         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=3pcc
         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=3pcc
         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=3pcc
         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: ...

         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.



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   Watson can only use PCMU and GSM. 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=3pcc
         To: T. Watson <sip:watson@bell-tel.com> ;tag=37462311
         Call-ID: 662606876@kton.bell-tel.com
         CSeq: 1 ACK



20.4 Terminating a Call

   To terminate a call, the caller can send a BYE request formatted as
   follows:


   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=3pcc
         To: T. A. Watson <sip:watson@bell-tel.com> ;tag=37462311
         Call-ID: 3298420296@kton.bell-tel.com
         CSeq: 2 BYE



   If the callee wishes to terminate the call, it sends a BYE request as
   well. However, this BYE request has the contents of the To field from
   the above message in the From field, and the contents of the From
   field in the above message in the To field.

20.5 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,



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   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=3pcc
         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=3pcc
         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



   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=3pcc
         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



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   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=3pcc
         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=3pcc
         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=3pcc
         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

   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=3pcc
         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



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   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=3pcc
         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=3pcc
         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=3pcc
         To:      T. Watson <sip:t.watson@ieee.org>
         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=3pcc
         To:      T. Watson <sip:t.watson@ieee.org>
         Call-ID: 31415@c.bell-tel.com



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         CSeq:    1 CANCEL



   In addition, A 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=3pcc
         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=3pcc
         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
         From:     A. Bell <sip:a.g.bell@bell-tel.com>;tag=3pcc
         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



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         From:     A. Bell <sip:a.g.bell@bell-tel.com>;tag=3pcc
         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=3pcc
         To:       T. Watson <sip:t.watson@ieee.org>;tag=35253448
         Call-ID:  31415@c.bell-tel.com
         CSeq:     2 BYE



20.6 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=3pcc
         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



   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=5h7j
         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



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   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=5h7j
         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=5h7j
         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
   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=5h7j
         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:




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   C->S: INVITE sip:alice@wonderland.com SIP/2.0
         Via: SIP/2.0/UDP h.caller.com
         From: <sip:charlie@caller.com>;tag=5h7j
         To: Alice <sip:alice@wonderland.com>
         Call-ID: 27182@caller.com
         CSeq: 2 INVITE
         Contact: sip:charlie@h.caller.com



20.7 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=3ab6
         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
         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.




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   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.

20.8 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=1gloo
         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=1gloo
         To: Bob <sip:bob@example.com> ;tag=376364382
         Call-ID: 6378@cat.wonderland.com
         Content-Length: 81
         Content-Type: application/sdp

         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




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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, Record-Route, Route 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. UAs MUST be able
   to use outbound proxies.

   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.

        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



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   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

   Implementations SHOULD NOT have built-in limits for the number of
   header instances and header field lengths, beyond imposing an overall
   message length. Header fields that identify the transaction SHOULD
   appear first in the message so that implementations that cannot
   handle the full message can at least return a status response, e.g.,
   513 (Message Too Large).

   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.


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 MUST contain zero or one SDP
   messages. Although the SDP specification allows for multiple session
   descriptions to be concatenated together into a large SDP message, an
   SDP message used with SIP MUST contain only a single session
   description.

   Proxies generally do not modify the session description, but MAY do
   so if necessary, e.g., for network address translators, and if the
   session description is not protected by a cryptographic integrity
   mechanism.



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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     -      -
   Record-Route          R     m     -     m      m
   Require               R     m     -     m      m
   Response-Key          R     -     -     e      e
   Route                 R     m     m     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.1 General Methodology

   The usage of SDP within SIP follows an "offer-answer" model. One side
   offers an SDP that provides their view of the session, and the other
   side answers the SDP with a matching one. The offer-answer model MAY
   occur in two ways. The offer MAY be placed in an INVITE, in which
   case the the answer MUST be in a 200 class response, and the ACK MUST
   NOT contain SDP. Or, the INVITE MAY contain no SDP, in which case the
   offer MUST be in the 200 class response, and the ACK MUST contain the
   answer.




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   If the offered session is not acceptable, it is rejected. If the
   offer is in the INVITE, rejection occurs by responding with a 488 or
   606 response. If the offer occurs in the 200 class response, the UAC
   MUST send an ACK request, with a valid answer. It MAY send a BYE
   request to terminate the call, or MAY generate a re-INVITE, with the
   offer in the INVITE, to change the session parameters back to an
   acceptable form.

   SDP processing does not depend on whether the offer comes in the
   INVITE or 200 class response.

B.2 Generating the initial offer

   The offer (and answer) MUST be a valid SDP, as defined by RFC 2327
   [6]. This means it MUST contain a v line, o line, s line and t line.
   Either an e line or p line MUST be present. However, it is
   RECOMMENDED that all implementations accept SDP without e, p, or s
   lines. The numeric value of the session id and version in the o line
   MUST be representable with a 64 bit signed integer.

   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 20.2 illustrates this point. For
   invitations to two-party sessions, the SDP "s=" line MAY consist of a
   single space character (0x20).


        Unfortunately, SDP does not allow to leave the "s=" line
        empty.

B.2.1 Unicast

   The offer MUST contain zero or more media sections. Zero media
   sessions implies that the offerer wishes to communicate, but that the
   streams for the session will be added at a later time through re-
   INVITEs.

   If a session description from an offerer contains a media stream
   which is listed as send (receive) only, it means that the offerer is
   only willing to send (receive) this stream, not receive (send). Media
   streams are marked as send-only or receive-only media streams using
   the SDP "a=sendonly" and "a=recvonly" attributes, respectively. If
   neither attribute is present, the default is both send and receive
   (which MAY be explicitly indicated with the "a=sendrecv" attribute).

   For recvonly and sendrecv streams, the port number and address in the



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   session description indicate where the media stream should be sent
   to. For send-only RTP streams, the address and port number indicate
   where RTCP reports are to be sent to. Specifically, RTCP reports are
   sent to the port number one higher than the number indicated. The IP
   address and port present in the offer indicate nothing about the
   source IP address and source port of RTP and RTCP packets that will
   be sent by the offerer. A port number of zero in the offer indicates
   that the stream is offered but should never be used. This has no
   useful semantics in an initial INVITE, but is allowed for reasons of
   completeness, since the response can contain a zero port indicating a
   rejected stream (Section B.3. Furthermore, existing streams can be
   terminated by setting the port to zero (Section B.4. In general, a
   port number of zero indicates that the media stream is not wanted.

   The list of payload types for each media stream conveys two pieces of
   information, namely the set of codecs that the offerer is capable of
   sending and/or receiving (depending on the direction attributes), and
   the RTP payload type numbers used to identify those codecs. If
   multiple codecs are listed, it means that the offerer is capable of
   making use of any of those codecs during the call. In other words,
   the answerer MAY change codecs in the middle of the call, without
   sending a SIP message, to make use of any of those listed. For a
   send-only stream, the offer SHOULD indicate those codecs the offerer
   is willing to send for this stream. For a receive-only stream, the
   offer SHOULD indicate those codecs the offerer is willing to receive
   for this stream. For a send and receive stream, the offer SHOULD
   indicate those codecs that the offerer is willing to send and receive
   with.

   For receive-only streams, the payload type numbers indicate the value
   of the payload type field in RTP packets the offerer is expecting to
   receive for that codec. For send-only streams, the payload type
   numbers indicate the value of the payload type field in RTP packets
   the offerer 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 offerer expects to both send and receive. This
   means that the payload type for a codec is the same in both
   directions.

   All media descriptions SHOULD contain "a=rtpmap" mappings from RTP
   payload types to encodings. If there is no "a=rtpmap", the static
   payload type table from RFC 1890 [26] is to be used.

        This allows easier migration away from static payload
        types.

   In all cases, the codecs in the m line are listed in order of
   preference, with the first codec listed being preferred. In this



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   case, preferred means that the recipient of the offer SHOULD use the
   codec with the highest preference that is acceptable to it.

   If multiple media streams of different types are present, it means
   that the offerer wishes to use those streams at the same time. A
   typical case is an audio and video stream as part of a
   videoconference.

   If multiple media streams of the same type are present, it means that
   the offerer wishes to send (and/or receive), multiple streams at the
   same time. When sending multiple streams of the same type, the source
   of the stream (such as the microphone or circuit on a gateway) is
   sent multiple times, once for each stream. Each stream MAY use
   different encodings. When receiving multiple streams of the same
   type, the streams MUST be mixed before playing them out. A typical
   usage example is a pre-paid calling card application, where the user
   can enter in a "long pound" at any time during a call to hangup and
   make a new call on the same card. This requires media from the user
   to the remote gateway, and to a system that looks for the long pound.

   There are some codecs, such as the RTP payload format for DTMF tones
   and digits [42] and comfort noise codecs, which can only encode
   specific types of media content. When one of these codecs is present
   in an offered stream that is send-only or send-and-receive, it means
   that the offerer will send using that codec only when the content of
   the media stream is of a type that can be encoded with that codec.
   When the content of the media stream cannot be encoded with that
   codec, another codec indicated in the m line can be used. If there
   are no other codecs in the m line, nothing is sent. This is useful
   for handling the case where a UA would like to send DTMF only, using
   RFC 2833, to a remote server. This is indicated with a single media
   line containing only the DTMF codec.

B.2.2 Multicast

   Construction of a session description for a multicast offer follows
   the procedures above, with a few exceptions.

   The address listed in the c line MUST be a multicast address. It
   indicates the address that the offerer wishes to receive packets on.

   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.



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B.3 Generating the answer

   The answer to an offered SDP is based on the offered SDP. If the
   answer is different in any way (different IP addresses, ports, etc.),
   the origin line MUST be different in the answer, since the answer is
   generated by a different entity. In that case, the version number in
   the o line of the answer is unrelated to the version number in the o
   line of the offer.

   For each m line in the offer, there MUST be a corresponding m line in
   the answer. The answer MUST contain exactly the same number of m
   lines as the offer. This allows for streams to be matched up based on
   their order. This implies that if the offer contained zero m lines,
   the answer MUST contain zero m lines.

   An offered stream MAY be rejected in the answer, for any reason. The
   definition of rejected is both neither offerer and answerer MUST NOT
   generate media (or RTCP packets) for that stream. To reject an
   offered stream, the port number in the corresponding stream in the
   answer is set to zero. Any media formats listed are ignored. At least
   one MUST be present, as specified by SDP.

B.3.1 Unicast

   If a stream is offered as sendonly, the corresponding stream MUST be
   marked as recvonly in the answer. Furthermore, the corresponding
   stream in the answer MUST contain at least one codec the answerer is
   willing to receive with from amongst those listed in the offer. The
   stream MAY indicate additional codecs, not listed in the
   corresponding stream in the offer, that the answerer is willing to
   receive with. The connection address and port indicate the address
   where the answerer wishes to receive RTP (RTCP will be received on
   the port which is one higher).

   If a media stream is listed as recvonly in the offer, the answer MUST
   be marked as sendonly. Furthermore, the corresponding stream in the
   answer MUST contain at least one codec the answerer is willing to
   send with from amongst those listed in the offer. The IP address and
   port indicate the address where the answerer wishes to receive RTCP
   (RTCP will be received on the port number one higher than the one
   listed in the SDP).

   If an offered media stream is listed as sendrecv (or contains no
   direction attribute, in which case it is sendrecv by default), the
   corresponding stream in the answer MAY be marked as sendonly,
   recvonly, or sendrecv. The default is sendrecv. If the stream in the
   answer is marked as sendonly, it MUST contain at least one codec the
   answerer is willing to send with from amongst those listed in the



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   offer. The IP address and port indicate the address where the
   answerer wishes to receive RTCP. If the stream in the answer is
   marked as recvonly, it MUST contain at least one codec the answerer
   is willing to receive with from amongst those listed in the offer.
   The stream MAY indicate additional codecs, not listed in the
   corresponding stream in the offer, that the answerer is willing to
   receive with. The connection address and port in the answer indicate
   the address where the answerer wishes to receive RTP and RTCP (RTCP
   will be received on the port number one higher than the one listed in
   the SDP). If the stream in the answer is marked as sendrecv, it MUST
   contain at least one codec the answerer is willing to both send and
   receive with, from amongst those listed in the offer. The stream MAY
   indicate additional codecs, not listed in the corresponding stream in
   the offer, that the answerer is willing to receive with. The
   connection address and port indicate the address where the answerer
   wishes to receive RTP (RTCP will be received on the port which is one
   higher).

   The payload type numbers for a particular codec within a stream MUST
   be the same in offer and answer. In other words, a different dynamic
   payload type number for the same codec cannot be used in each
   direction.

   In all cases, the codecs in the m line are listed in order of
   preference, with the first codec listed being preferred. In this
   case, preferred means that the recipient of the answer SHOULD use the
   codec with the highest preference that is acceptable to it.

   If the answerer has no media formats in common for a particular
   offered stream, the answerer MUST reject that media stream.

   If there are no media formats in common for all streams, the entire
   offered session is rejected.

B.3.2 Multicast

   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 offerer is acceptable to the
   answerer, the answerer can choose not to include a session
   description or MAY echo the description in the response.

   An answerer MAY 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 offerer that the answerer does not support the given
   stream or media types which were removed. An answerer MUST NOT change
   whether a given stream is send-only, receive-only, or send-and-
   receive.



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   If an answerer does not support multicast at all, it SHOULD reject
   the session description.

B.4 Modifying the session

   At any point during the call, either participant MAY issue a re-
   INVITE to modify characteristics of the session. It is fundamental to
   the operation of SIP that the exact same offer-answer procedure
   defined above is used for re-INVITE. This means that a re-INVITE MAY
   contain no SDP, so that the 200 OK to the re-INVITE contains the
   offer. In this case, the offerer SHOULD offer the same SDP it
   provided previously if it has no reason to change anything.

   The offer in a re-INVITE MAY be identical to the last SDP provided to
   the other party (which may have been provided in an offer or an
   answer), or it MAY be different. We refer to the last SDP provided as
   the "previous SDP". If the offer is the same, the answer MAY be the
   same as the previous SDP from the answerer, or it MAY be different.
   If the offered SDP is different from the previous SDP, some
   constraints are placed on its construction, discussed below.

   Nearly all aspects of the session can be modified. New streams can be
   added, existing streams can be deleted, and parameters of existing
   streams can change. When issuing an offer that modifies the session,
   the o line of the new SDP MUST be identical to that in the previous
   SDP, except that the version in the origin field MUST increment from
   the previous SDP. If the version in the origin line does not
   increment, the SDP MUST be identical to the SDP with that version
   number. The answerer MUST be prepared to receive an offer that
   contains SDP with a version that has not changed; this is effectively
   a no-op. However, the answerer MUST generate a valid answer (which
   MAY be the same as the previous SDP from the answerer, or MAY be
   different), according to the procedures defined in Section B.3.

   If an SDP is offered which is different from the previous SDP, the
   new SDP MUST have a matching media section for each media section in
   the previous SDP. In other words, if the previous SDP had N media
   lines, the new SDP MUST have at least N media lines. The ith media
   stream in the previous SDP, counting from the top, matches the ith
   media stream in the new SDP, counting from the top. This matching is
   necessary in order for the answerer to determine which stream in the
   new SDP corresponds to a stream in the previous SDP. Because of these
   requirements, the number of m lines in a stream never decreases, but
   only increases. Deleted media streams from a previous SDP MUST NOT be
   removed from a new SDP.

B.4.1 Adding a media stream




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   New media streams are created by adding additional media descriptions
   below the existing ones. New media sections MUST appear below any
   existing media sections. The rules for formatting this media section
   are identical to those described in Section B.2.

   When the answerer receives an SDP with more media descriptions than
   the previous SDP from the offerer, the answerer knows that new media
   streams are being added. These can be rejected or accepted by placing
   a matching media description in the answer. The procedures for
   constructing the new media description in the answer are described in
   Section B.3.

B.4.2 Removing a media stream

   Existing media streams are removed by creating a new SDP with the
   port number for that stream set to zero. Otherwise, the media
   description SHOULD be formatted identically to the corresponding
   stream in the previous SDP.

   A stream that is offered with a port of zero MUST be marked with port
   zero in the answer. Otherwise, the media description for the removed
   stream SHOULD be formatted identically to the corresponding stream in
   the previous SDP.

B.4.3 Modifying a media stream

   Nearly all characteristics of a media stream can be modified.

   The port number for a stream MAY be changed. To do this, the offerer
   creates a new media description, with the port number in the m line
   different from the corresponding stream in the previous SDP. If only
   the port number is to be changed, the rest of the media stream
   description SHOULD remain unchanged.

   The corresponding media stream in the answer MAY be the same as the
   stream in the previous SDP from the answerer, or MAY be different.
   If the updated stream is accepted by the answerer, the answerer
   SHOULD begin sending traffic for that stream to the new port
   immediately. This implies that the offerer MUST be prepared to
   receive media on the new port the instant it makes the offer.

   To change the IP address where media is sent to, the same procedure
   is followed for changing the port number. The only difference is that
   the connection line is updated, not the port number.

   The list of codecs used in the session MAY be changed. To do this,
   the offerer creates a new media description, with the list of media
   formats in the m line different from the corresponding stream in the



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   previous SDP. This list MAY include new codecs, and MAY remove codecs
   present from the previous SDP. When a new codec is used with a
   dynamic payload type number, it MUST NOT reuse a dynamic payload type
   number used previously in the session.

   The corresponding media stream in the answer is formulated as
   described in Section B.3. If the new list of codecs for a stream
   changes the choice of which codec is used, the new codec SHOULD be
   used immediately. That means the offerer MUST be prepared to receive
   media with a new codec as soon as it sends the offer, and the
   answerer MUST be prepared to receive media with a new codec as soon
   as it sends the answer.

   The media type (audio, video, etc.) for a stream MAY be changed. This
   is particularly useful for changing between voice and fax in a single
   stream, which are both separate media types. To do this, the offerer
   creates a new media description, with a new media type, in place of
   the description in the previous SDP which is to be changed. The IP
   address and port for the stream MAY change, or MAY remain the same.
   However, the list of payload type numbers for the new codecs MUST be
   different than any used previously for this stream.

   The corresponding media stream in the answer is formulated as
   described in Section B.3. Assuming the stream is acceptable, the
   answerer SHOULD begin sending with the new media type and codecs as
   soon as it receives the offer.

   The transport for a stream MAY be changed. The process for doing this
   is identical to changing the port, excepting the transport is
   updated, not the port.

   Any other attributes in a media description MAY be updated in an
   offer.

B.4.4 Putting a media stream 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 offers the other an updated SDP. This SDP has the connection
   address set to zero (0.0.0.0) for those streams that are to be put on
   hold. The treament of this is no different than for any other change
   in address, with the exception that the 0.0.0.0 address is
   effectively interpreted as "dev/null", and no media is sent.
   Specifically, this means that a stream is placed "on hold" separately
   in each direction. Each stream is placed "on hold" independently. The
   recipient of an offer for a stream on-hold SHOULD NOT automatically
   return an answer with the corresponding stream on hold. An SDP with
   all streams "on hold" is referred to as held SDP



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        Certain third party call control scenarios do not work when
        a UA responds to held SDP with held SDP.

   Typically, when a user "presses" hold, the UA will generate a re-
   INVITE with all streams in the SDP indicating an address of 0.0.0.0,
   and it will also locally mute, so that no media is sent to the far
   end.

B.5 Example

   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). The offered SDP is:


   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 as the answer:

   v=0
   o=bob 2890844730 2890844730 IN IP4 host.example.com
   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



   At some point later, Bob decides to change the port where he will
   receive the audio stream (from 47920 to 6400), and at the same time,



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   add an additional audio stream as receive only, using the RTP payload
   format for events. Bob offers the following SDP in the INVITE:


   v=0
   o=bob 2890844730 2890844731 IN IP4 host.example.com
   s=New board design
   t=0 0
   c=IN IP4 host.example.com
   m=audio 8864 RTP/AVP 110
   a=rtpmap:110 telephone-events
   a=recvonly
   m=audio 6400 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



   Alice accepts the additional media stream, and so generates the
   following answer:


   v=0
   o=alice 2890844526 2890844527 IN IP4 host.anywhere.com
   s=New board design
   t=0 0
   c=IN IP4 host.anywhere.com
   m=audio 4520 RTP/AVP 110
   a=rtpmap:110 telephone-events
   a=sendonly
   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



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 [25] and RFC 2234 [43]. Implementors will need to be
   familiar with the notation in order to understand this specification.
   The augmented BNF includes the following constructs:



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        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.


   (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]





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   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 ))


   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





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   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" |
                     "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



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   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 [22]). 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
                        |  %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.



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        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.



        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 10.47 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



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   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 SHOULD NOT use the X prefix notation and MUST NOT 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 13). 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.

        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. 14.4.1, 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.



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        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.

        o Noted that maddr could also contain a unicast address, but
          SHOULD contain the multicast address if the request is sent
          via multicast (Section 10.46, 14.1).

        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 17.4)

        o Clarified that CANCEL and INVITE are separate transactions
          (Fig. 11). Thus, the INVITE request generates a 487 (Request
          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



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          UAS roles are defined on a request-by-request basis. (Section
          1.4)

        o In Section 10.46, noted that the maddr and received parameters
          also need to be encrypted when doing Via hiding.

        o Simplified Fig. 11 to only show INVITE transaction.

        o Added definition of the use of Contact (Section 10.14) 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 15.5 explaining what a UA should
          do when receiving an initial INVITE with a tag.

        o Clarified UA and proxy behavior for 302 responses (Section
          11.3.3).

        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
          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 17.4 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 10.46.1 that maddr is to be added by any
          client, either proxy or UAC.




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        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 10.24, noted that registration updates can shorten
          the validity period.

        o Added note to Section 19.3 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 10.14 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.

        o Cseq is unique within a call leg, not just within a call
          (Section 10.20).

        o Added IPv6 literal addresses to the SIP URL definition in
          Section 2, according to RFC 2732 [44].  Modified the IPv4
          address to limit segments to at most three digits.

        o In Section 7, 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.




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        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
          15. 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 17.3 and 10.46.6). 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.

        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.



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          It is mandatory.

        o Added in Section 6: "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 5.1: "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 5.1 what happens if two INVITE requests
          meet each other on the wire, either traveling the same or in
          opposite directions:


             A UAC MUST NOT 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.

        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 10.26).

        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.



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        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.

        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 10).

        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.




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          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. 10.14.

        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.
          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



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          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 Added "or UAS" in description of received headers in Section
          10.46.1. This makes the response algorithm work even if the
          last IP address in the Via is incorrect.

        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.

        o Removed ability to carry multiple requests in a single UDP
          packet (Section 10.18).

        o Added note that Allow MAY be included in requests, to indicate
          requestor capabilities for the same call ID.

        o Added note to Section 10.21 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 7 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



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             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);

          - 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



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            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 Made in Version 03

        o Description of Route and Record-Route moved to separate
          section, Section 16, which is new. All UAs must now support
          this mechanism (Section A).

        o Removed status code 411, since it cannot occur (Jonathan
          Rosenberg, James Jack).

        o Rewrote Record-Route section to reflect new mechanism. In
          particular, requests from callee to caller now use the same
          path as in the opposite direction, without substituting the
          From header field values. The maddr parameter is now optional.

        o Disallowed SIP URLs that only have a password, without a user
          name. The prototype from RFC 1738 also doesn't allow this.

        o Allow registrar to set the expiration time.

        o CSeq (Section 10.20) is counted within a call leg, not a call.

        o Removed wording that connection closing is equivalent to
          CANCEL or 500. This does not work for connections that are
          used for multiple transactions and has other problems.

        o Cleaned up CSeq section. Removed text about inserting CSeq
          method when it is absent. Clarified that CSeq increments for
          all requests, not just invite. Clarified that all out of order
          requests, not just out of order INVITE, are rejected with a
          400 class response. Clarified the meaning of "initial"
          sequence number. Clarified that after a request forks, each
          200 OK is a separate call leg, and thus, separate CSeq space.
          Clarified that CSeq numbers are independent for each direction
          of a call leg.

        o Massive reorganization and cleanup of the SDP section.
          Introduced the concept of the offer-answer model. Clarified
          that set of codecs in m line are usable all at the same time.
          Inserted size restriction on representation of values in o
          line. Explicitly describe forked media. New media lines for
          adding streams appear at the bottom of the SDP (used to say
          append).



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        o Removed Also.

        o Added text to Require and Proxy-Require sections, making it a
          SHOULD to retry the request without the unsupported extension.

        o Added text to section on 415, saying that UAC SHOULD retry the
          request without the unsupported body.

        o Added text to section on CANCEL and ACK, clarifying much of
          the behavior.

        o Modified Content-Type to indicate that it can be present even
          if the body is empty.

        o From tags mandatory

        o Old text said that if you hang up before sending an ACK, you
          need not send the ACK. That is wrong. Text fixed so that an
          ACK is always sent.

        o Old text said that if you never got a response to an INVITE,
          the UAC should send both an INVITE and CANCEL. This doesn't
          make sense. Rahter, it should do nothing and consider the call
          terminated.

        o Added text that says pending requests are responded to with a
          487 if a BYE is received.

        o Updated section 2.2, so that its clear that Contact is not
          used with BYE.

        o Clarified Via processing rules. Added text on handling loops
          when proxies route on headers besides the request URI. Added
          text on handling case when sent-by contains a domain name.
          Added text to 6.47 on opening TCP connections to send
          responses upstream.

        o Clarified that a 1xx with an unknown xx is not the same as the
          100 response.

        o Removed usage of Retry-After in REGISTER.

        o Clarified usage of persistent connections.

        o Clarified that servers supporting HTTP basic or digest in
          rfc2617 MUST be backwards compatible with RFC 2069.

        o Clarified that ACK contains the same branch ID as the request



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          its acknowledging.

        o Added definitions for spiral, B2BUA.

        o Rephrased definitions for UAC, UAS, Call, call-leg, caller,
          callee, making them more concrete.

        o URL comparison ignores parameters not present in both URLs
          only for unknown parameters.

        o Clarified that * in Contact is used only in REGISTER with
          Expires header zero. Mentioned * case in section on Contact
          syntax.

        o Removed text that says a UA can insert a Contact in 2xx that
          indicates the address of a proxy. Not likely to work in
          general.

        o Removed SDP text about aligning media streams within a media
          type to handle certain crash and restart cases.

        o Receiving a 481 to a mid-call request terminates that call
          leg. Agreed upon at IETF 49.

        o Introduced definitin of regular transaction - non-INVITE
          excepting ACK and CANCEL.

        o Clarified rules for overlapping transactions.

        o Forking proxies MUST be stateful (used to say SHOULD). Proxies
          that send requests on multicast MUST be stateful (used to say
          nothing)

        o Text added recommending that registrars authorize that entity
          in From field can register address-of-record in the To field.

        o Forwarding of non-100 provisionals upstream in a proxy changed
          from SHOULD to MUST.

        o Removed PGP.

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



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   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 [45,46].

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

L Bibliography

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




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   [4] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," Request for Comments 2326, Internet Engineering
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   [5] M. Handley, C. Perkins, and E. Whelan, "Session announcement
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   [9] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
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   [14] D. Zimmerman, "The finger user information protocol," Request
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   [15] S. Williamson, M. Kosters, D. Blacka, J. Singh, and K. Zeilstra,
   "Referral whois (rwhois) protocol V1.5," Request for Comments 2167,
   Internet Engineering Task Force, June 1997.




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   [16] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access
   protocol," Request for Comments 1777, Internet Engineering Task
   Force, Mar. 1995.

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

   [18] A. Vaha-Sipila, "URLs for telephone calls," Request for Comments
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   [19] J. Postel, "User datagram protocol," Request for Comments 768,
   Internet Engineering Task Force, Aug. 1980.

   [20] J. Postel, "DoD standard transmission control protocol," Request
   for Comments 761, Internet Engineering Task Force, Jan. 1980.

   [21] T. Dierks and C. Allen, "The TLS protocol version 1.0," Request
   for Comments 2246, Internet Engineering Task Force, Jan. 1999.

   [22] F. Yergeau, "UTF-8, a transformation format of ISO 10646,"
   Request for Comments 2279, Internet Engineering Task Force, Jan.
   1998.

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

   [24] J. C. Mogul and S. E. Deering, "Path MTU discovery," Request for
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   [25] D. Crocker, "Standard for the format of ARPA internet text
   messages," Request for Comments 822, Internet Engineering Task Force,
   Aug. 1982.

   [26] H. Schulzrinne, "RTP profile for audio and video conferences
   with minimal control," Request for Comments 1890, Internet
   Engineering Task Force, Jan.  1996.

   [27] D. Meyer, "Administratively scoped IP multicast," Request for
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   [29] F. Dawson and T. Howes, "vcard MIME directory profile," Request
   for Comments 2426, Internet Engineering Task Force, Sept. 1998.



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   [30] G. Good, "The LDAP data interchange format (LDIF) - technical
   specification," Request for Comments 2849, Internet Engineering Task
   Force, June 2000.

   [31] P. Hoffman, L. Masinter, and J. Zawinski, "The mailto URL
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   in internet messages: The content-disposition header," Request for
   Comments 1806, Internet Engineering Task Force, June 1995.

   [33] R. Braden and Ed, "Requirements for internet hosts - application
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   [34] J. Palme, "Common internet message headers," Request for
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   [35] H. Schulzrinne and J. Rosenberg, "SIP: Session initiation
   protocol -- locating SIP servers," Internet Draft, Internet
   Engineering Task Force, Mar. 2001.  Work in progress.

   [36] R. Rivest, "The MD5 message-digest algorithm," Request for
   Comments 1321, Internet Engineering Task Force, Apr. 1992.

   [37] H. Alvestrand, "IETF policy on character sets and languages,"
   Request for Comments 2277, Internet Engineering Task Force, Jan.
   1998.

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   extensions (MIME) part two: Media types," Request for Comments 2046,
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   [39] R. Atkinson, "Security architecture for the internet protocol,"
   Request for Comments 1825, Internet Engineering Task Force, Aug.
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   A. Luotonen, and L. Stewart, "HTTP authentication: Basic and digest
   access authentication," Request for Comments 2617, Internet
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   [41] J. Franks, P. Hallam-Baker, J. Hostetler, P. Leach, A. Luotonen,
   E. Sink, and L. Stewart, "An extension to HTTP : Digest access
   authentication," Request for Comments 2069, Internet Engineering Task
   Force, Jan. 1997.




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   [42] H. Schulzrinne and S. Petrack, "RTP payload for DTMF digits,
   telephony tones and telephony signals," Request for Comments 2833,
   Internet Engineering Task Force, May 2000.

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   specifications:  ABNF," Request for Comments 2234, Internet
   Engineering Task Force, Nov.  1997.

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   IPv6 addresses in URL's," Request for Comments 2732, Internet
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   packet-switched teleconferencing system," Journal of Internetworking:
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   the Internet," in European Workshop on Interactive Distributed
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   1996.


   Full Copyright Statement

   Copyright (c) The Internet Society (2001). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
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   The limited permissions granted above are perpetual and will not be
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   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION



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   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   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 ......................................    9
   1.4.2      Locating a SIP Server ...............................   10
   1.4.3      SIP Transaction .....................................   13
   1.4.4      Initiating a Session ................................   13
   1.4.5      Locating a User .....................................   14
   1.4.6      Changing an Existing Session ........................   17
   1.4.7      Registration Services ...............................   17
   1.5        Protocol Properties .................................   17
   1.5.1      Minimal State .......................................   17
   1.5.2      Lower-Layer-Protocol Neutral ........................   18
   1.5.3      Text-Based ..........................................   18
   2          SIP Uniform Resource Locators .......................   18
   2.1        SIP URL Comparison ..................................   23
   2.2        Non-SIP URLs ........................................   24
   3          SIP Message Overview ................................   24
   4          Request .............................................   26
   4.1        Request-Line ........................................   26
   4.2        Methods .............................................   27
   4.3        Request-URI .........................................   27
   4.3.1      SIP Version .........................................   29
   4.4        Option Tags .........................................   29
   4.4.1      Registering New Option Tags with IANA ...............   30
   5          INVITE, ACK and CANCEL ..............................   31
   5.1        INVITE ..............................................   31
   5.1.1      ACK .................................................   33
   5.2        CANCEL ..............................................   34
   6          BYE .................................................   36
   7          Registrars, Registrations and the REGISTER Method
   ................................................................   36
   7.1        Where to Register ...................................   36
   7.2        REGISTER Header Fields ..............................   37
   7.3        Registering Contact Locations .......................   38
   7.4        Registration Expiration .............................   39



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   7.5        List of Current Registrations .......................   40
   7.6        Removing Registrations ..............................   40
   8          OPTIONS .............................................   40
   9          Response ............................................   41
   9.1        Status-Line .........................................   41
   9.1.1      Status Codes and Reason Phrases .....................   41
   10         Header Field Definitions ............................   43
   10.1       General Header Fields ...............................   46
   10.2       Entity Header Fields ................................   46
   10.3       Request Header Fields ...............................   46
   10.4       Response Header Fields ..............................   48
   10.5       Header Field Format .................................   49
   10.6       Accept ..............................................   50
   10.7       Accept-Encoding .....................................   50
   10.8       Accept-Language .....................................   51
   10.9       Alert-Info ..........................................   51
   10.10      Allow ...............................................   51
   10.11      Authorization .......................................   52
   10.12      Call-ID .............................................   52
   10.13      Call-Info ...........................................   53
   10.14      Contact .............................................   54
   10.15      Content-Disposition .................................   57
   10.16      Content-Encoding ....................................   58
   10.17      Content-Language ....................................   59
   10.18      Content-Length ......................................   59
   10.19      Content-Type ........................................   60
   10.20      CSeq ................................................   60
   10.21      Date ................................................   62
   10.22      Encryption ..........................................   62
   10.23      Error-Info ..........................................   63
   10.24      Expires .............................................   63
   10.25      From ................................................   64
   10.26      In-Reply-To .........................................   66
   10.27      Max-Forwards ........................................   66
   10.28      MIME-Version ........................................   67
   10.29      Organization ........................................   67
   10.30      Priority ............................................   67
   10.31      Proxy-Authenticate ..................................   68
   10.32      Proxy-Authorization .................................   69
   10.33      Proxy-Require .......................................   69
   10.34      Record-Route ........................................   69
   10.35      Require .............................................   70
   10.36      Response-Key ........................................   70
   10.37      Retry-After .........................................   71
   10.38      Route ...............................................   72
   10.39      Server ..............................................   72
   10.40      Subject .............................................   72
   10.41      Supported ...........................................   72



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   10.42      Timestamp ...........................................   73
   10.43      To ..................................................   73
   10.44      Unsupported .........................................   75
   10.45      User-Agent ..........................................   75
   10.46      Via .................................................   75
   10.46.1    Requests ............................................   76
   10.46.2    Receiver-tagged Via Header Fields ...................   76
   10.46.3    Receiving Responses .................................   77
   10.46.4    Generating Responses ................................   77
   10.46.5    Sending Responses ...................................   77
   10.46.6    Syntax ..............................................   78
   10.47      Warning .............................................   80
   10.48      WWW-Authenticate ....................................   82
   11         Status Code Definitions .............................   83
   11.1       Informational 1xx ...................................   83
   11.1.1     100 Trying ..........................................   84
   11.1.2     180 Ringing .........................................   84
   11.1.3     181 Call Is Being Forwarded .........................   84
   11.1.4     182 Queued ..........................................   84
   11.1.5     183 Session Progress ................................   84
   11.2       Successful 2xx ......................................   84
   11.2.1     200 OK ..............................................   84
   11.3       Redirection 3xx .....................................   85
   11.3.1     300 Multiple Choices ................................   85
   11.3.2     301 Moved Permanently ...............................   85
   11.3.3     302 Moved Temporarily ...............................   86
   11.3.4     305 Use Proxy .......................................   86
   11.3.5     380 Alternative Service .............................   86
   11.4       Request Failure 4xx .................................   86
   11.4.1     400 Bad Request .....................................   87
   11.4.2     401 Unauthorized ....................................   87
   11.4.3     402 Payment Required ................................   87
   11.4.4     403 Forbidden .......................................   87
   11.4.5     404 Not Found .......................................   87
   11.4.6     405 Method Not Allowed ..............................   87
   11.4.7     406 Not Acceptable ..................................   87
   11.4.8     407 Proxy Authentication Required ...................   87
   11.4.9     408 Request Timeout .................................   88
   11.4.10    409 Conflict ........................................   88
   11.4.11    410 Gone ............................................   88
   11.4.12    413 Request Entity Too Large ........................   88
   11.4.13    414 Request-URI Too Long ............................   88
   11.4.14    415 Unsupported Media Type ..........................   88
   11.4.15    420 Bad Extension ...................................   89
   11.4.16    480 Temporarily Unavailable .........................   89
   11.4.17    481 Call Leg/Transaction Does Not Exist .............   89
   11.4.18    482 Loop Detected ...................................   89
   11.4.19    483 Too Many Hops ...................................   89



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   11.4.20    484 Address Incomplete ..............................   90
   11.4.21    485 Ambiguous .......................................   90
   11.4.22    486 Busy Here .......................................   90
   11.4.23    487 Request Terminated ..............................   91
   11.4.24    488 Not Acceptable Here .............................   91
   11.5       Server Failure 5xx ..................................   91
   11.5.1     500 Server Internal Error ...........................   91
   11.5.2     501 Not Implemented .................................   91
   11.5.3     502 Bad Gateway .....................................   91
   11.5.4     503 Service Unavailable .............................   91
   11.5.5     504 Server Time-out .................................   92
   11.5.6     505 Version Not Supported ...........................   92
   11.5.7     513 Message Too Large ...............................   92
   11.6       Global Failures 6xx .................................   92
   11.6.1     600 Busy Everywhere .................................   92
   11.6.2     603 Decline .........................................   93
   11.6.3     604 Does Not Exist Anywhere .........................   93
   11.6.4     606 Not Acceptable ..................................   93
   12         SIP Message Body ....................................   93
   12.1       Body Inclusion ......................................   93
   12.2       Message Body Type ...................................   94
   12.3       Message Body Length .................................   94
   13         Compact Form ........................................   94
   14         Behavior of SIP Clients and Servers .................   95
   14.1       Multicast Unreliable Transport Protocols ............   95
   14.2       Reliable Transport Protocols ........................   96
   14.3       Reliability for Requests Other Than INVITE ..........   97
   14.3.1     Unreliable Transport Protocols ......................   97
   14.3.2     Reliable Transport Protocol .........................   98
   14.4       Reliability for INVITE Requests .....................   98
   14.4.1     Unreliable Transport Protocols ......................   99
   14.4.2     Reliable Transport Protocol .........................  100
   14.5       ICMP Handling .......................................  103
   15         Behavior of SIP User Agents .........................  103
   15.1       Caller Issues Initial INVITE Request ................  103
   15.2       Callee Issues Response ..............................  103
   15.3       Caller Receives Response to Initial Request .........  104
   15.4       Caller or Callee Generate Subsequent Requests .......  104
   15.5       Receiving Subsequent Requests .......................  105
   16         Routing of Requests .................................  107
   16.1       UAC Processing for initial transaction ..............  107
   16.2       UAS Processing of initial transaction ...............  108
   16.3       Proxy procedures for record routing a transaction
   ................................................................  109
   16.4       UA Processing of Subsequent Requests in a Call Leg
   ................................................................  112
   16.4.1     Local outbound proxies ..............................  114
   16.5       Proxy routing procedures ............................  114



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   16.6       Pre-Loaded Route Headers ............................  115
   17         Behavior of SIP Proxy and Redirect Servers ..........  115
   17.1       Redirect Server .....................................  115
   17.2       User Agent Server ...................................  115
   17.3       Proxy Server ........................................  116
   17.3.1     Proxying Requests ...................................  116
   17.3.2     Proxying Responses ..................................  117
   17.3.3     Stateless Proxy: Proxying Responses .................  117
   17.3.4     Stateful Proxy: Receiving Requests ..................  117
   17.3.5     Stateful Proxy: Receiving ACKs ......................  117
   17.3.6     Stateful Proxy: Receiving Responses .................  117
   17.3.7     Stateless, Non-Forking Proxy ........................  118
   17.4       Forking Proxy .......................................  118
   18         Security Considerations .............................  123
   18.1       Confidentiality and Privacy: Encryption .............  123
   18.1.1     End-to-End Encryption ...............................  123
   18.1.2     Privacy of SIP Responses ............................  126
   18.1.3     Encryption by Proxies ...............................  126
   18.1.4     Hop-by-Hop Encryption ...............................  126
   18.2       Message Integrity and Access Control:
   Authentication .................................................  126
   18.2.1     Trusting responses ..................................  129
   18.3       Callee Privacy ......................................  130
   18.4       Denial of Service ...................................  130
   18.5       Known Security Problems .............................  131
   19         SIP Authentication using HTTP Basic and Digest
   Schemes ........................................................  131
   19.1       Framework ...........................................  131
   19.2       Basic Authentication ................................  132
   19.3       Digest Authentication ...............................  132
   19.4       Proxy-Authentication ................................  133
   20         Examples ............................................  134
   20.1       Registration ........................................  134
   20.2       Invitation to a Multicast Conference ................  135
   20.2.1     Request .............................................  136
   20.2.2     Response ............................................  137
   20.3       Two-party Call ......................................  138
   20.4       Terminating a Call ..................................  140
   20.5       Forking Proxy .......................................  140
   20.6       Redirects ...........................................  145
   20.7       Negotiation .........................................  147
   20.8       OPTIONS Request .....................................  148
   A          Minimal Implementation ..............................  148
   A.1        Transport Protocol Support ..........................  149
   A.2        Client ..............................................  149
   A.3        Server ..............................................  149
   A.4        Header Processing ...................................  150
   B          Usage of the Session Description Protocol (SDP)



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   ................................................................  150
   B.1        General Methodology .................................  151
   B.2        Generating the initial offer ........................  152
   B.2.1      Unicast .............................................  152
   B.2.2      Multicast ...........................................  154
   B.3        Generating the answer ...............................  155
   B.3.1      Unicast .............................................  155
   B.3.2      Multicast ...........................................  156
   B.4        Modifying the session ...............................  157
   B.4.1      Adding a media stream ...............................  157
   B.4.2      Removing a media stream .............................  158
   B.4.3      Modifying a media stream ............................  158
   B.4.4      Putting a media stream on hold ......................  159
   B.5        Example .............................................  160
   C          Summary of Augmented BNF ............................  161
   C.1        Basic Rules .........................................  164
   D          IANA Considerations .................................  166
   E          Changes from RFC 2543 ...............................  167
   F          Changes Made in Version 00 ..........................  167
   G          Changes Made in Version 01 ..........................  174
   H          Changes Made in Version 02 ..........................  175
   I          Changes Made in Version 03 ..........................  177
   J          Acknowledgments .....................................  179
   K          Authors' Addresses ..................................  180
   L          Bibliography ........................................  180


























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