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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                                        Jonathan Rosenberg
                                                             dynamicsoft
                                                     Henning Schulzrinne
                                                             Columbia U.
                                                       Gonzalo Camarillo
                                                                Ericsson
                                                           Alan Johnston
                                                                Worldcom
                                                            Jon Peterson
                                                                 Neustar
                                                           Robert Sparks
                                                             dynamicsoft
                                                            Mark Handley
                                                                   ACIRI
                                                            Eve Schooler
                                                                    AT&T



draft-ietf-sip-rfc2543bis-05.txt
October 26, 2001
Expires: April 2002


                    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
   telephone calls, multimedia distribution and multimedia conferences.

   SIP invitations used to create sessions carry session descriptions
   which allow participants to agree on a set of compatible media types.
   SIP makes use of elements called proxy servers to help route requests
   to the users current location, assist in firewall traversal, and
   provide features to users. SIP also provides a registration function
   that allows them to upload their current location for use by proxy
   servers.  SIP runs ontop of several different transport protocols.


1 Introduction




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   There are many applications of the Internet that require the creation
   and management of a session, where a session is considered an
   exchange of data between an association of participants. The
   implementation of these services is complicated by the practices of
   participants; users may move between endpoints, they may be
   addressable by multiple names, and they may communicate in several
   different media - sometimes simultaneously. Numerous protocols have
   been authored that carry various forms of real-time multimedia
   session data such as voice, video, or text messages. SIP works in
   concert with these protocols by enabling Internet endpoints (called
   "user agents") to discover one another and to agree on a
   characterization of a session they would like to share.  For locating
   prospective session participants, SIP relies on an infrastructure of
   network hosts (called "proxy servers") to which user agents can send
   registrations, invitations to sessions and other requests. SIP is an
   agile, general-purpose tool for creating, modifying and terminating
   sessions that works independently of underlying transport protocols
   and without dependency on the type of session that is being
   established.

2 Overview of SIP Functionality

   The Session Initiation Protocol (SIP) is an application-layer control
   protocol that can establish, modify and terminate multimedia sessions
   (conferences) such as Internet telephony calls. SIP can also invite
   participants to already existing sessions. A SIP entity issuing an
   invitation for an already existing session does not necessarily have
   to be a member of the session to which it is inviting. Media can be
   added to (and removed from) an existing session. SIP transparently
   supports name mapping and redirection services, which supports
   personal mobility [1] - users can maintain a single externally
   visible identifier (SIP URI) regardless of their network location.

   SIP supports five facets of establishing and terminating multimedia
   communications:

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

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

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

        Session setup: "ringing", establishment of session parameters at
             both called and calling party;




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        Session handling: including transfer and termination of
             sessions, modifying session parameters, and invoking
             services.

   SIP is not a vertically integrated communications system. SIP is
   rather a component of the overall IETF multimedia data and control
   architecture which incorporates 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 multimedia sessions via multicast
   and the session description protocol (SDP) (RFC 2327 [6]) for
   describing multimedia sessions. Therefore, SIP should be used in
   conjunction with other protocols in order to provide complete
   services to the users. However, the basic functionality and operation
   of SIP does not depend on any of these protocols.

   SIP does not provide services. SIP rather provides primitives that
   can be used to implement different services. For example, SIP can
   locate a user and deliver an opaque object to his current location.
   If this primitive is used to deliver a session description written in
   SDP, for instance, the parameters of a session can be agreed between
   endpoints. If the same primitive is used to deliver a photo of the
   caller as well as the session description, a "caller ID" service can
   be easily implemented. As this example shows, a single primitive is
   typically used to provide several different services. Consequently,
   generality is more important than efficiency when designing SIP
   primitives.

   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 initiate a session that uses some other
   conference control protocol. SIP does not allocate multicast
   addresses and does not reserve network resources.

3 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.

4 Overview of Operation

   This section will introduce the basic operations of the SIP protocol
   using simple examples. Note that this section is tutorial in nature
   and does not contain any normative statements.



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   The first example will show the basic functions of SIP: location of
   an end point, signaling a desire to communicate, negotiation of
   session parameters to establish the session, and teardown of the
   session once established.

   Figure 1 shows a typical example of a SIP message exchange between
   two users, Alice and Bob. (Each message is labeled with the letter
   "F" and a number for reference by the text.) In this example, Alice
   uses a SIP application on her PC (referred to as a softphone) to call
   Bob on his SIP phone over the Internet. Also shown are two SIP proxy
   servers which act on behalf of Alice and Bob to facilitate the
   session establishment. This typical arrangement is often referred to
   as the "SIP trapezoid" as shown by the geometric shape of the dashed
   lines in Figure 1.


   Alice "calls" Bob using his SIP identity, a type of Uniform Resource
   Identifier (URI) called a SIP URI and defined in Section 21.1. It has
   a similar form to an email address, typically containing a username
   and a host name. In this case it is sip:bob@biloxi.com, where
   biloxi.com is the domain of Bob's SIP service provider (which can be
   an enterprise, retail provider, etc). Alice also has a SIP URI of
   sip:alice@atlanta.com. Alice might have typed in Bob's URI or perhaps
   clicked on a hyperlink or an entry in an address book.

   SIP is based on an HTTP-like request/response transacton model. Each
   transaction consists of a request that invokes a particular "Method",
   or function, on the server, and at least one response. In this
   example, the transaction begins with Alice's softphone sending an
   INVITE request addressed to Bob's SIP URI. INVITE is an example of a
   SIP method which specifies the action that the requestor (Alice)
   wants the server (Bob) to take. The INVITE request contains a number
   of header fields. Header fields are additional named attributes which
   provide additional information about a message. The ones present in
   an INVITE include a unique identifier for the call, the destination
   address, Alice's address, and information about the type of session
   that Alice wishes to establish with Bob. The INVITE (message F1 in
   Figure 1) might look like this:


     INVITE sip:bob@biloxi.com SIP/2.0
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact: <sip:alice@10.1.3.3>
     Content-Type: application/sdp



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                 atlanta.com  . . . biloxi.com
             .      proxy              proxy     .
           .                                       .
   Alice's  . . . . . . . . . . . . . . . . . . . .  Bob's
  softphone                                        SIP Phone
     |                |                |                |
     |    INVITE F1   |                |                |
     |--------------->|    INVITE F3   |                |
     |  100 Trying F2 |--------------->|    INVITE F5   |
     |<---------------|  100 Trying F4 |--------------->|
     |                |<-------------- | 180 Ringing F6 |
     |                | 180 Ringing F7 |<---------------|
     | 180 Ringing F8 |<---------------|     200 OK F9  |
     |<---------------|    200 OK F10  |<---------------|
     |    200 OK F11  |<---------------|                |
     |<---------------|                |                |
     |                       ACK F12                    |
     |------------------------------------------------->|
     |                   Media Session                  |
     |<================================================>|
     |                       BYE F13                    |
     |<-------------------------------------------------|
     |                     200 OK F14                   |
     |------------------------------------------------->|
     |                                                  |



   Figure 1: SIP session setup example with SIP trapezoid


     Contact-Length: 142

     (Alice's SDP not shown)



   The first line of the text-encoded message contains the method name
   (INVITE). The lines which follow are a list of header fields.  This
   example contains a minimum required set. The headers are briefly
   described below:

   Via contains the IP address (10.1.3.3), port number (5060), and
   transport protocol (UDP) on which Alice is expecting to receive
   responses to this request.



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   To contains a display name (Bob) and a SIP URI (sip:bob@biloxi.com)
   that the request was originally directed towards.

   From also contains a display name (Alice) and a SIP URI
   (sip:alice@atlanta.com) that indicate the originator of the request.
   This header field also has a tag parameter which contains a
   pseudorandom string (1928301774) which was added to the URI by the
   softphone. It is used for identification purposes.

   Call-ID contains a globally unique identifier for this call,
   generated by the combination of a pseudorandom string and the
   softphone's IP address. The combination of the To, From, and Call-ID
   completely define a peer-to-peer SIP relationship betwee Alice and
   Bob, and is referred to as a "dialog".

   CSeq or Command Sequence contains an integer and a method name. The
   CSeq number is incremented for each new request, and is a traditional
   sequence number.

   Contact contains a SIP URI which represents a direct route to reach
   or contact Alice, usually composed of a username at an IP address.
   While the Via header field is used to tell other elements where to
   send the response, the Contact header field tells other elements
   where to send future requests for this dialog.

   Content-Type contains a description of the message body (not shown).

   Content-Length contains an octet (byte) count of the message body.

   The complete set of SIP header fields is defined in Section 22.

   The details of the session, type of media, codec, sampling rate, etc.
   are not described using SIP. Rather, the body of a SIP message
   contains a description of the session, encoded in some other protocol
   format.  One such format is Session Description Protocol (SDP) [6].
   This SDP message (not shown in the example) is carried by the SIP
   message in an analogous way that a document attachment is carried by
   an email message, or a web page is carried in an HTTP message.

   Since the softphone has no knowledge of Bob's exact location, or how
   to locate the SIP server in the biloxi.com domain, the softphone
   sends the INVITE to the SIP server that serves Alice's domain,
   atlanta.com.  The IP address of the atlanta.com SIP server could have
   been configured in Alice's softphone, or it could have been
   discovered by DHCP, for example.

   The atlanta.com SIP server is a type of SIP server known as a proxy
   server. A proxy server receives SIP requests and forwards them on



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   behalf of the requestor. In this example, the proxy server receives
   the INVITE request and generates a 100 Trying response which is sent
   back to Alice's softphone. The 100 Trying response indicates that the
   INVITE has been received and that the proxy is working on her behalf
   to try to route the INVITE to the destination.  Responses in SIP use
   a numerical three digit code followed by a descriptive phrase. This
   response contains the same To, From, Call-ID, and CSeq as the INVITE,
   which allows Alice's softphone to correlate this response to the sent
   INVITE. The atlanta.com proxy server locates the proxy server at
   biloxi.com, possibly by performing a DNS (Domain Name Service) lookup
   to find the SIP server which serves the biloxi.com domain. This is
   described in Section 24. As a result, it obtains the IP address of
   the biloxi.com proxy server and forwards, or proxies, the INVITE
   request there. Before forwarding the request, the atlanta.com proxy
   server adds an additional Via header field which contains its own IP
   address (the INVITE already contains Alice's IP address in the first
   Via). The biloxi.com proxy server receives the INVITE and responds
   with a 100 Trying response back to the Atlanta.com proxy server to
   indicate that it has received the INVITE and is processing the
   request. The proxy server consults a database, generically called a
   location service, which contains the current IP address of Bob. (We
   shall see in the next section how this database can be populated.)
   The biloxi.com proxy server adds another Via header with its own IP
   address to the INVITE and proxies it to Bob's SIP phone.

   Bob's SIP phone receives the INVITE and begins to alert Bob to the
   incoming call from Alice so that Bob can decide whether or not to
   answer the call - i.e. Bob's phone rings. Bob's SIP phone sends an
   indication of this in a 180 Ringing response. This response is routed
   back thorough the two proxies in the reverse direction. Each proxy
   uses the Via header to figure out where to send the response, and
   removes its own address from the top. As a result, although DNS and
   location service lookups were required to route the initial INVITE,
   the 180 Ringing response can be returned to the caller without
   lookups, or without state being maintained in the proxies. This also
   has the desirable property that each proxy that sees the INVITE will
   also see all responses to the INVITE.

   When Alice's softphone receives the 180 Ringing response, it passes
   this information to Alice, perhaps using an audio ringback tone, or
   just by displaying or flashing a message on Alice's screen.

   In this example, Bob decides to answer the call. When he picks up the
   handset his SIP phone sends a 200 OK response to indicate that the
   call has been answered. The 200 OK contains a message body containing
   the SDP media description of the type of session that Bob is willing
   to establish with Alice. As a result, there is a two-phase exchange
   of SDP messages; Alice sent one to Bob, and Bob sent one back to



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   Alice.  This two-phase exchange provides basic negotiation
   capabilities, and is based on a simple offer/answer model, If Bob did
   not wish to answer the call, or was busy on another call, an error
   response would have been sent instead of the 200 OK, which would have
   resulted in no media session being established. The complete list of
   SIP response codes is in Section 23. The 200 OK (message F9 in Figure
   1) might look like this:


     SIP/2.0 200 OK
     Via: SIP/2.0/UDP 10.2.1.1:5060;branch=4b43c2ff8.1
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact: <sip:bob@10.4.1.4>
     Content-Type: application/sdp
     Contact-Length: 131

     (Bob's SDP not shown)



   The first line of the response contains the response code (200) and
   the reason phrase (OK). The remaining lines contain header fields.
   The Via header fields, To, From, Call- ID, and CSeq are all copied
   from the INVITE request.  (Note that there are three Via headers -
   one added by Alice's SIP phone, one added by the atlanta.com proxy,
   and one added by the biloxi.com proxy.) Also note that Bob's SIP
   phone has added a tag parameter to the To header field. This tag will
   be incorporated by both User Agents into the dialog and will be
   included in all future requests and responses in this call. The
   Contact header field contains a URI at which Bob can be directly
   reached at his SIP phone. The Content-Type and Content-Length refer
   to the not shown message body which contains Bob's SDP media
   information.

   In additon to DNS and location service lookups shown in this example,
   proxy servers can make arbitrarily complex "routing decisions" in
   order to decide where to send a request. For example, if Bob's SIP
   phone returned a 486 Busy Here response, the biloxi.com proxy server
   could proxy the INVITE to Bob's voicemail server. A proxy server can
   also send an INVITE to a number of locations at the same time.  This
   type of parallel search is known as "forking".

   In this case, the 200 OK is routed back through the two proxies and



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   is received by Alice's softphone which then stops the ringback tone
   and indicates that the call has been answered. Finally, an
   acknowledgement message, ACK, is sent by Alice to Bob to confirm the
   reception of the final response (200 OK). Note that in this example,
   the ACK is sent directly from Alice to Bob, bypassing the two
   proxies. This is due to the fact that through the INVITE/200 OK
   exchange, the two SIP user agents have learned each other's IP
   address through the Contact header fields, something which was not
   known when the initial INVITE was sent. The lookups performed by the
   two proxies are no longer needed, so they drop put of the call flow.
   This completes the INVITE/200/ACK three way handshake used to
   establish SIP sessions, and is the end of the transaction. Full
   details on session setup is in Section 13.

   Alice and Bob's media session has now begun, and they begin sending
   media packets using the format agreed to in the exchange of SDP. In
   general, the end-to-end media packets will take a different path from
   the SIP signaling messages.

   During the session, either Alice or Bob may decide to change the
   characteristics of the media session. This is accomplished by sending
   a re-INVITE containing a new media description. If the change is
   acceptable to the other party, a 200 OK is sent which is itself
   responded to with an ACK. This re-INVITE will reference the existing
   dialog so the other party knows that it is to modify an existing
   session instead of establishing a new session. If the change is not
   acceptable, an error response, such as a 406 Not Acceptable response
   is sent, which also receives an ACK. However, the failure of the re-
   INVITE does not cause the existing call to fail - the session
   continues using the previously negotiated characteristics.  Full
   details on session modification is in Section 14.

   At the end of the call, Bob disconnects (hangs up) first, and
   generates a BYE message. This BYE is routed directly to Alice's
   softphone, again bypassing the proxies. Alice confirms receipt of the
   BYE with a 200 OK response, which terminates the session and the BYE
   transaction. Note that no ACK is sent - an ACK is only sent in
   response to a response to an INVITE request. The reasons for this
   special handling for INVITE will be discussed later, but relate to
   the reliability mechanisms in SIP, the length of time it can take for
   a ringing phone to be answered, and forking. For this reason, request
   handling in SIP is often classified as either INVITE or non- INVITE,
   referring to all other methods besides INVITE. Full details on
   session termination is in Section 15.

   Full details of all the messages shown in the example of Figure 1 are
   shown in Section 25.2.




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   In some cases, it may be useful for proxies in the SIP signaling path
   see all the messaging between the two endpoints for the duration of
   the session. For example, if the biloxi.com proxy server wished to
   remain in the SIP messaging path beyond the initial INVITE, it would
   add to the INVITE a required routing header field known as Record-
   Route containing a URI which resolves to the proxy. This information
   would be received by both Bob's SIP phone and (due to the Record-
   Route header field being passed back in the 200 OK) Alice's softphone
   and stored for the duration of the dialog. This would then result in
   the ACK, BYE, and 200 OK to the BYE being received and proxied by the
   biloxi.com proxy server. Each proxy can independently decide to
   receive subsequent messaging, and that messaging will go through all
   proxies that elected to receive it. A common use of this capability
   is in firewall traversal or mid-call feature implementation.

   Registration is another common operation in SIP. Registration is one
   way in which the biloxi.com server can learn the current location of
   Bob. Upon initialization, and at periodic intervals, Bob's SIP phone
   sends REGISTER messages a server in the biloxi.com domain known as a
   SIP registrar. The REGISTER messages associate Bob's SIP URL
   (sip:bob@biloxi.com) with the machine he is currently logged in at
   (conveyed as a SIP URL in the Contact header). The registrar writes
   this association, also called a binding, to a database, called the
   location service , where it can be used by the proxy in the
   biloxi.com domain. Often, a registrar server for a domain is co-
   located with the proxy for that domain. It is an important concept
   that the distinction between types of SIP servers are logical, not
   physical.

   Bob is not limited to registering from a single device. For example,
   both his SIP phone at home and the one in the office could send in
   registrations. This information is stored together in the location
   service, and allows a proxy to perform various types of searches to
   locate Bob. Similarly, more than one user can be registered on a
   single device at the same time.

   The location service is just an abstract concept. It generally
   contains information that allows a proxy to input a URI and get back
   a translated URI that tells the proxy where to send the request.
   Registrations are one way to create this information, but not the
   only way. Arbitrarily complex mapping functions can be programmed, at
   the discretion of the administrator.

   Finally, it is important to note that in SIP, registration is used
   for routing incoming SIP requests and has no role in authorizing
   outgoing requests. Authorization and authentication are handled in
   SIP either on a request by request, challenge/response mechanism, or
   using a lower layer scheme as discussed in Section 20.



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   The complete set of SIP message details for this registration example
   is in Section 25.2.

   Additional operations in SIP include querying for the capabilities of
   a SIP server or client using OPTIONS, and canceling a pending request
   using CANCEL will be introduced in later sections.

5 Structure of the Protocol

   The SIP protocol is structured as a layered protocol, which means
   that its behavior is described in terms of a set of fairly
   independent processing stages, with only a loose coupling between
   each stage. The structuring of the protocols into layers is for the
   purpose of presentation and conciseness; it allows the grouping of
   functions common across elements into a single place. It does not
   dictate an implementation in any way. When we say that an element
   "contains" a layer, that means it is compliant to the set of rules
   defined by that layer.

   Not every element specified by the protocol contains every layer.
   Furthermore, the elements specified by SIP are logical elements, not
   physical ones. A physical realization can choose to act as different
   logical elements, perhaps even on a transaction by transaction basis.

   The lowest layer of the SIP protocol is its syntax and encoding. Its
   encoding is specified using a BNF. The complete BNF is specified in
   Section 26. However, a basic overview of the structure of a SIP
   message can be found in Section 7. This section introduces enough of
   an understanding of the format of a SIP message to facilitate
   understanding the remainder of the protocol.

   The next higher layer is the transport layer. This layer defines how
   a client takes a request, and physically sends it over the network,
   and how a response is sent by a server, and then received by a
   client. All SIP elements contain a transport layer. The transport
   layer is described in Section 19.

   The next higher layer is the transaction layer. Transactions are a
   fundamental component of SIP. A transaction is a request, sent by a
   client transaction (using the transport layer), to a server
   transaction, along with all responses to that request sent from the
   server transaction back to the client. The transaction layer handles
   retransmissions, matching of responses to requests, and timeouts. Any
   task that a UAC wishes to accomplish takes place using a series of
   transactions. Discussion of transactions can be found in Section 17.
   User agents contain a transaction layer, as do stateful proxies.
   Stateless proxies do not contain a transaction layer.




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   The transaction layer has a client component (referred to as a client
   transaction), and a server component (referred to as a server
   transaction), each of which are represented by an FSM that is
   constructed to process a particular request. The layer on top of the
   transaction layer is called the transaction user (TU), of which there
   are several types. When a TU wishes to send a request, it creates a
   client transaction instance and passes it the request, along with the
   destination IP address, port, and transport to send the request to.

   SIP provides the ability for a transaction to be canceled by the
   client which initiated it. When a client cancels a transaction, it
   requests that the server give up on further processing, revert to the
   state that existed before the transaction was initiated, and generate
   a specific error response to that transaction. This is done with a
   CANCEL request, which constitutes its own transaction, but references
   the transaction to be cancelled. Cancellation is described in Section
   9.

   There are several different types of transaction users. A UAC
   contains a UAC core, a UAS contains a UAS core, and a proxy contains
   a proxy core. The behavior of the UAC and UAS cores depend largely on
   the method. However, there are some common rules for all methods.
   These rules are captured in Section 8. The primarily deal with
   construction of a request, in the case of a UAC, and processing of
   that request, and generation of a response, in the case of a UAS.

   UAC and UAS core behavior for the REGISTER method is described in
   Section 10. Registrations play an important role in SIP. In fact, a
   UAS that handles a REGISTER is given a special name - a registrar -
   and it is described in that section.

   UAC and UAS core behavior for the OPTIONS method, used for
   determining the capabilities of a UAC, are described in Section 11.

   Certain other requests are sent within a dialog peer-to-peer SIP
   relationship between a two user agents that persists for some time.
   The dialog facilitates sequencing of messages between the user
   agents, and proper routing of requests between both them. One way to
   setup a dialog is with the INVITE method. When a UAC sends a request
   that is within the context of a dialog, it follows the common UAC
   rules as discussed in Section 8, but also the rules for mid-dialog
   requests. Section 12 discusses dialogs, and presents the procedures
   for their construction, and maintenance, in addition to construction
   of requests within a dialog.

   The most important method in SIP is the INVITE method, which is used
   to establish a session between participants. A session is a
   collection of participants, and streams of media between them, for



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   the purposes of communication. Section 13 discusses how sessions are
   initiated, resulting in one or more SIP dialogs. Section 14 discusses
   how characteristics of that session are modified, through the use of
   an INVITE request within a dialog. Finally, section 15 discusses how
   a session is terminated.

   The procedures of Sections 8, 10, 11, 12, 13, 14, and 15 deal
   entirely with the UA core. Section 16 discusses the proxy element,
   which facilitates routing of messages between user agents.

6 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: A back-to-back user agent (B2BUA) is a
             logical entity that receives a request, and processes it as
             a UAS. In order to determine how the request should be
             answered, it acts as a UAC and generates requests. Unlike a
             proxy server, it maintains dialog state, and must
             participate in all requests sent on the dialogs it has
             established. Since it is a concatenation of a UAC and UAS,
             no explicit definitions are needed for its behavior.

        Call: A call is an informal term that refers to a dialog between
             peers, generally set up for the purposes of a multimedia
             conversation.

        Call leg: Another name for a dialog.

        Call stateful: A proxy is call stateful if it retains state for
             a dialog from the initiating INVITE to the terminating BYE
             request. A call stateful proxy is always stateful, but the
             converse is not true.

        Client: A client is any network element that sends SIP requests,
             and receives SIP responses. Clients may or may not interact
             directly with a human user. User agent clients and proxies
             are clients.

        Conference: A multimedia session (see below) that contains
             multiple participants.

        Dialog: A dialog is a peer-to-peer SIP relationship between a



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             UAC and UAS that persists for some time. A dialog is
             established by SIP messages, such as a 2xx response to an
             INVITE request. A dialog is identified by a call
             identifier, local address, and remote address. A dialog was
             formerly known as a call leg in RFC 2543.

        Downstream: A direction of message forwarding within a
             transaction which refers to the direction that requests
             flow from the 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.

        Informational Response: Same as a provisional response.

        Initiator, calling party, caller: The party initiating a session
             with an INVITE request. A caller retains this role from the
             time it sends the INVITE until the termination of any
             dialogs established by the INVITE.

        Invitation: An INVITE request.

        Invitee, invited user, called party, callee: The party that
             receives an INVITE request for the purposes of establishing
             a new session. A callee retains this role from the time it
             receives the INVITE until the termination of the dialog
             established by that INVITE.

        Isomorphic request or response: Two requests are defined to be
             isomorphic for the purposes of this document if they have
             the same values for the Call-ID, To, From, CSeq, Request-
             URI and the top-most Via header. Two responses are
             isomorphic if they have the same values for the Call-ID,
             To, From, CSeq and top Via header. A message which is
             isomorphic to another is also known as a retransmission.

        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). It is an abstract database, sometimes
             referred to as a location server. The contents of the
             database can be populated in many ways, including being
             written by registrars.

        Loop: A request that arrives at a proxy, is forwarded, and later
             arrives back at the same proxy. When it arrives the second



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             time, its Request-URI is identical to the first time, and
             other headers that affect proxy operation are unchanged, so
             that the proxy would make the same processing decision on
             the request it made the first time around. Looped requests
             are errors, and the procedures for detecting them and
             handling them are described by the protocol.

        Method: The method is the primary function that a request is
             meant to invoke on a server. The method is carried in the
             request message itself. Example methods are INVITE and BYE.

        Outbound proxy: A proxy that receives all requests from a
             client, even though it is not the server resolved by the
             Request-URI. The outbound proxy sends these requests, after
             any local processing, to the address indicated in the
             Request-URI, or to another outbound proxy.

        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 entity that acts as both a
             server and a client for the purpose of making requests on
             behalf of other clients. A proxy server primarily plays to
             role of routing, which means its job is to ensure that a
             request is passed on to another entity that can further
             process the request. Proxies are also useful for enforcing
             policy and for firewall traversal. A proxy interprets, and,
             if necessary, rewrites parts of a request message before
             forwarding it.

        Registrar: A registrar is a server that accepts REGISTER
             requests, and places the information it receives in those
             requests into the location service for the domain it
             handles.

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




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

        Server: A server is a network element that receives requests in
             order to service them, and sends back responses to those
             requests.  Examples of servers are proxies, user agent
             servers, redirect servers, and registrars.

        Sequential search: In a sequential search, a proxy server
             attempts each contact address in sequence, proceeding to
             the next one only after the previous has generated a non-
             2xx final response.

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

        (SIP) transaction: A SIP transaction occurs between a client and
             a server and comprises all messages from the first request
             sent from the client to the server up to a final (non-1xx)
             response sent from the server to the client, and the ACK
             for the response in the case the response was a 2xx. The
             ACK for a 2xx response is a separate transaction.

        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, differs in a way which will result in a
             different processing decision than the original request.
             Typically, this means that it has 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 the
             client or server transaction state machines defined in this
             specification when it processes requests. A stateless proxy
             forwards every request it receives downstream and every
             response it receives upstream.

        Stateful proxy: A logical entity that maintains the client and
             server transaction state machines defined by this



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             specification during the processing of a request. Also
             known as a transaction stateful proxy. The behavior of a
             stateful proxy is further defined in Section 16. A stateful
             proxy is not the same as a call stateful proxy.

        Transaction User (TU): The layer of protocol processing that
             resides above the transaction layer. Transaction users
             include the UAC core, UAS core, and proxy core.

        Upstream: A direction of message forwarding within a transaction
             which refers to the direction that responses flow from the
             user agent server to 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 creates a new request, and then uses the client
             transaction state machinery to send it. The role of UAC
             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 transaction. If it
             receives a request later on, it takes on the role of a User
             Agent Server for the processing of that transaction.

        UAC Core: The set of processing functions required of a UAC that
             reside above the transaction and transport layers.

        User agent server (UAS): A user agent server is a logical entity
             that generates a response to a SIP request.  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 transaction. If it generates a
             request later on, it takes on the role of a User agent
             client for the processing of that transaction.

        UAS Core: The set of processing functions required at a UAS that
             reside above the transaction and transport layers.

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

   The role of UAC and UAS as well as proxy and redirect servers are
   defined on a transaction-by-transaction 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.



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   Similarly, the same software can act as a proxy server for one
   request and as a redirect server for the next request.

   Proxy, location and registrar servers defined above are logical
   entities; implementations MAY combine them into a single application
   program.

7 SIP Messages

   SIP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [11]).

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

   Both Request (section 7.1) and Response (section 7.2) messages use
   the generic-message format of RFC 822 [12]. Both types of messages
   consist of a start-line, one or more header fields (also known as
   "headers"), an empty line indicating the end of the header fields,
   and an optional message-body.



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


   The start-line, each message-header line, and the empty line MUST be
   terminated by a carriage-return line-feed sequence (CRLF). Note that
   the empty line MUST be present even if the message-body is not.

   Except for the above difference in character sets, much of SIP's
   message and header field syntax is 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]).

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

7.1 Requests

   SIP Requests are distinguished by having a Request-Line for a start-
   line. A 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 end-of-line CRLF sequence. No LWS is allowed in any of
   the elements.



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                      Method Request-URI SIP-Version

        o Method

          This specification defines six methods : REGISTER for
          registering contact information, INVITE, ACK and CANCEL for
          setting up sessions, BYE for terminating sessions and OPTIONS
          for querying servers about their capabilities. SIP extensions
          may define additional methods.

        o Request-URI

          The Request-URI is a SIP URL as described in Section 21.1 or a
          general URI (RFC 2396 [9]).  It indicates the user or service
          to which this request is being addressed.  The Request-URI
          MUST NOT contain unescaped spaces or control characters and
          MUST NOT be enclosed in "<>".

          SIP servers MAY support Request-URIs with schemes other than
          "sip", for example the "tel" URI scheme of RFC 2806 [13].  It
          MAY translate non-SIP URIs using any mechanism at its
          disposal, resulting in either a SIP URI or some other scheme.

        o 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 send upper-case.


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

7.2 Responses

   SIP Responses are distinguished by having a Status-Line for a start-
   line.  A Status-Line, consists of the protocol version 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.

                   SIP-version Status-Code Reason-Phrase




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

   The first digit of the Status-Code defines the class of response.
   The last two digits do not have any categorization role. For this
   reason, any response with a status code between 100 and 199 is
   referred to as a "1xx response", any response with a status code
   between 200 and 299 as a "2xx response", and so on. 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.

   Full definitions of these classes and each registered code appear in
   Section 23.

7.3 Header Fields

   SIP header fields are similar to HTTP header fields in both syntax
   and semantics. In particular, SIP header fields follow the [H4.2]
   definitions of syntax for message-header, the rules for extending
   header fields over multiple lines, the use of multiple message-header
   fields with the same field-name, and the rules regarding ordering of
   header fields.

7.3.1 Header Field Format

   Header fields follow the same generic header format as that given in
   Section 3.1 of RFC 822 [12]. Each header field consists of a field



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   name followed by a colon (":") and the field value.
                          field-name: field-value

   Note that the formal grammar for a message-header specified in
   Section 26 allow for an arbitrary amount of whitespace on either side
   of the colon. No space before the colon and a single space (SP)
   between the colon and the field-value is preferred. That is,

   Subject:            lunch
   Subject      :      lunch
   Subject            :lunch
   Subject: lunch


   are all valid, and equivalent, but the last is the preferred form.

   Header fields can be extended over multiple lines by preceding each
   extra line with at least one SP or horizontal tab (HT). The line
   break and the whitespace at the beginning of the next line are
   treated as a single SP character. Thus the following are equivalent:

   Subject: I know you're there, pick up the phone and talk to me!
   Subject: I know you're there,
            pick up the phone
            and talk to me!



   The relative order of header fields with different field names is not
   significant. The relative order of those with the same field name is
   important.  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.

   Implementations MUST be able to process multiple header fields with
   the same name in any combination of the single-value-per-line or
   comma-separated value forms.

   The following blocks of headers are valid and equivalent:

   Route: sip:alice@atlanta.com
   Subject: Lunch
   Route: sip:bob@biloxi.com
   Route: sip:carol@chicago.com



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   Route: sip:alice@atlanta.com, sip:bob@biloxi.com
   Route: sip:carol@chicago.com
   Subject: Lunch

   Subject: Lunch
   Route: sip:alice@atlanta.com, sip:bob@biloxi.com, sip:carol@chicago.com



   Each of the following blocks is valid but not equivalent to the
   others:

   Route: sip:alice@atlanta.com
   Route: sip:bob@biloxi.com
   Route: sip:carol@chicago.com

   Route: sip:bob@biloxi.com
   Route: sip:alice@atlanta.com
   Route: sip:carol@chicago.com

   Route: sip:alice@atlanta.com,sip:carol@chicago.com,sip:bob@biloxi.com



   The format of a header field-value is defined per header-name. It
   will always be either an opaque sequence of TEXT-UTF8 octets, or a
   combination of whitespace, tokens, separators, and quoted strings.
   Many of them will adhere to the general form of a value followed by a
   semi-colon separated sequence of parameter-name, parameter-value
   pairs:
        field-name: field-value *(;parameter-name=parameter-value)

   When comparing headers, field names are always case-insensitive.
   Unless otherwise stated in the definition of a particular header
   field, field values, parameter names, and parameter values (tokens in
   general) are case-insensitive. Unless specified otherwise, values
   expressed as quoted strings are case-sensitive.

   The following are equivalent:

   Contact: <sip:alice@atlanta.com>;expires=3600
   CONTACT: <sip:alice@atlanta.com>;ExPiReS=3600

   Contact-Disposition: session;handling=optional
   contact-disposition: Session;HANDLING=OPTIONAL






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   The following are not equivalent;

   Warning: 370 devnull "Choose a bigger pipe"
   Warning: 370 devnull "CHOOSE A BIGGER PIPE"



7.3.2 Header Field Classification

   Some header fields only make sense in requests or responses. These
   are called Request Header Fields and Response Header fields
   respectively.  Those header fields that can appear in either a
   request or response are called General Header Fields. If a header
   appears in a message not matching its category (such as a request
   header in a response), it MUST be ignored. Section 22 defines the
   classification of each header.

7.3.3 Compact Form

   SIP provides a mechanism to represent common header fields in an
   abbreviated form. This may be useful when messages would otherwise
   become to large to be carried on the transport available to it
   (exceeding the MTU when using UDP for example). These compact forms
   are defined in Section 22. A compact form MAY be substituted for the
   longer form of a header name at any time without changing the
   semantics of a the message. Multiple header fields in a message with
   the same header name MAY appear with an arbitrary mix of its long and
   short field name form. Implementations MUST accept both the long and
   short forms of each header name.

7.4 Bodies

   Requests, including new requests defined in extensions to this
   specification, MAY contain message bodies unless otherwise noted.

   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.

7.4.1 Message Body Type

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




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   The "multipart" MIME type defined in RFC 2046 [14] MAY be used within
   the body of the message.

   Implementations that send requests containing multipart message
   bodies MUST be able to send a session description as a non-multipart
   message body if the remote implementation requests this through an
   Accept header field.

7.4.2 Message Body Length

   The body length in bytes is provided by the Content-Length header
   field. Section 22.14 describes the necessary contents of this header
   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.)

7.5 Framing SIP messages

   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 [15] 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
        [16]). Thus, another reasonable value would be a message
        size of 950 bytes, to accommodate packet headers within the
        SLIP MTU without fragmentation.

   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.

   Likewise, Implementations processing SIP messages over stream
   oriented transports MUST ignore noise between messages.

8 General User Agent Behavior




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   A user agent represents an end system. It contains a User Agent
   Client (UAC), which generates requests, and a User Agent Server (UAS)
   which responds to them.  A UAC is capable of generating a request
   based on some external stimulus (the user clicking a button, or a
   signal on a PSTN line), and processing a response.  A UAS is capable
   of receiving a request, and generating response, based on user input,
   external stimulus, the result of a program execution, or some other
   mechanism.

   When a UAC sends a request, it will pass through some number of proxy
   servers, which forward the request towards the UAS. When the UAS
   generates a response, the response is forwarded towards the UAC.

   UAC and UAS procedures depend strongly on two factors. First, whether
   the request or response is inside or outside of a dialog, and second,
   based on the method of a request. Dialogs are discussed thoroughly in
   Section 12; they represent a peer-to-peer relationship between user
   agents, and are established by specific SIP methods, such as INVITE.

   In this section, we discuss the method independent rules for UAC and
   UAS behavior when processing of requests that are outside of a
   dialog. This includes, of course, the requests which themselves
   establish a dialog.

8.1 UAC Behavior

8.1.1 Generating the Request

   A valid SIP request formulated by a UAC MUST at a minimum contain the
   following headers: To, From, CSeq, Call-ID, and Via; all of these
   headers are mandatory in all SIP messages. These five headers are the
   fundamental building blocks of a SIP message, as they jointly provide
   for most of the critical message routing services including the
   addressing of messages, the routing of responses, ordering of
   messages, and the unique identification of transactions.

   Examples of requests send outside of a dialog include an INVITE to
   establish a session (Section 13) and an OPTIONS to query for
   capabilities (Section 11).

8.1.1.1 To

   The To general-header field first and foremost specifies the desired
   "logical" recipient of the request, or the address of record of the
   user or resource that is the target of this request. This may or may
   not be the ultimate recipient of the request. The To header MAY
   contain a SIP URI, but it may also make use of other URI schemes (for
   example as the tel URL [13]) when appropriate. The To header field



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   allows for a display name; this is meant to contain a descriptive
   version of the URI, and is intended to be displayed to a user
   interface.

   A UAC may learn how to populate the To header field for a particular
   request in a number of ways. Usually the user will suggest the To
   header field through a human interface, perhaps inputting the URI
   manually or selecting it from some sort of address book.

   A request outside of a dialog MUST NOT contain a tag; the tag in the
   To field of a request identifies the peer of the dialog. Since no
   dialog is established, no tag is present.

   For further information on the To header see Section 22.37.

   The following is an example of valid To header:

     To: Carol <sip:carol@chicago.com>



8.1.1.2 From

   The From general-header field indicates the logical identity of the
   initiator of the request, possibly the user's address of record. Like
   the To field, it contains a URI and optionally a display name. It is
   used by SIP elements to determine processing rules to apply to a
   request (for example, automatic call rejection). As such, it is very
   important that the URI not contain IP addresses or host names, since
   these are not logical names.

   The From header field allows for a display name; this is meant to
   contain a descriptive version of the URI, and is intended to be
   displayed to a user interface. A UAC SHOULD use the display name
   "Anonymous" if the identity of the client is to remain hidden.

   Usually the value that populates the From header field in requests
   generated by a particular user agent is pre-provisioned by the user
   or by the administrators of the user's local domain. If a particular
   user agent is used by multiple users, it might have switchable
   profiles that include a URI corresponding to the identity of the
   profiled user. Recipients of requests can authenticate the originator
   of a request in order to ascertain that they are who their From
   header field claims they are (see Section 20.2 for more on
   authentication).

   The From field MUST contain a new "tag" parameter, chosen by the UAC.
   See Section 21.3 for details on choosing a tag.



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   For further information on the From header see Section 22.20.

   Examples:


     From: "Bob" <sip:bob@biloxi.com> ;tag=a48s
     From: sip:+12125551212@server.phone2net.com;tag=887s
     From: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8



8.1.1.3 Call-ID

   The Call-ID general-header field acts as a unique identifier to group
   together series of messages. It is always the same for all requests
   and responses sent by either UA in a dialog. It is also the same in
   each registration from a UA within a single boot cycle.

   In a new request created by a UAC outside of any dialog, unless
   overridden by method specific behavior, it MUST be selected by the
   UAC as a a globally unique identifier over space and time; all SIP
   user agents must have a means to guarantee that the Call-ID headers
   they produce will not be inadvertently generated by any other user
   agent.

   Use of cryptographically random identifiers [17] in the generation of
   Call-IDs 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, and reduces the
        likelihood of unintentional Call-ID collisions.

   No provisioning or human interface is required for the selection of
   the Call-ID header field value for a request.

   For further information on the Call-ID header see Section 22.8.

   Example:


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



8.1.1.4 CSeq




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   The Cseq header serves as a way to identify and order transactions.
   It consists of a sequence number and a method. The method MUST match
   that of the request. The sequence number value is arbitrary, but MUST
   be expressible as a 32-bit unsigned integer and MUST be less than
   2**31.

   As long as it follows the above guidelines, a client may use any
   mechanism it would like to select CSeq header field values.

   For further information on the CSeq header see Section 22.16.

   Example:


     CSeq: 4711 INVITE



8.1.1.5 Via

   The Via header is used to determine the transport to use for sending
   a request, and for identifying the IP address and port where the
   response is to be sent. Rules for setting and using the values in
   this header are described in Section 19.

   For further information on the Via header see Section 22.40.

8.1.1.6 Contact

   The Contact header provides a SIP URI that can be used to contact
   that specific instance of the user agent for subsequent requests. The
   Contact header MUST be present in any request that can result in the
   establishment of a dialog. For the methods defined in this
   specification, that includes only the INVITE request. For these
   requests, the scope of the Contact is the dialog. That is, the
   Contact header refers to the URL that the UA would like to receive
   requests at, for requests that are part of that dialog only. Only a
   single URI MUST be present.

   For further information on the Contact header, see Section 22.10.

8.1.1.7 Request-URI

   The initial Request-URI of the message SHOULD be set to the value of
   the URI in the To field. One notable exception is the REGISTER
   method; behavior for setting the Request-URI of register is given in
   Section 10. Another exception is the case of pre-existing Route
   headers; in that case, the procedures of Section 12.2.1.1 as they



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   pertain to the Request- URI are followed, even though there is no
   dialog.

8.1.1.8 Supported and Require

   If the UAC supports extensions to SIP that can be applied by the
   server to the response, the UAC SHOULD include a Supported header in
   the request listing the option tags for those extensions.

   The option-tags listed MUST only refer to extensions defined in
   standards track RFCs. This is to prevent servers from insisting that
   clients implement non-standard, vendor defined features in order to
   receive service. Extensions defined by experimental and informational
   RFCs are explicitly excluded from usage with the Supported header in
   a request, since they too are often used to document vendor defined
   extensions.

   If the UAC wishes to insist that a UAS understand an extension that
   the UAC will apply to the request in order to process the request, it
   MUST insert a Require header into the request listing the option tag
   for that extension. If the UAC wishes to apply an extension to the
   request and insist that a proxy understand that extension, it MUST
   insert a Proxy-Require header into the request listing the option tag
   for that extension.

8.1.1.9 Additional Message Components

   After a new request has been created, the headers described above
   have been properly constructed, any additional optional headers are
   added, as are any headers specific to the method.

   SIP requests MAY contain a MIME-encoded message-body. Regardless of
   the type of body that a request contains, certain headers must be
   formulated to characterize the contents of the body. For further
   information on these headers see Section 7.4.

8.1.2 Sending the Request

   The destination for the request is then computed. This can be a
   preconfigured IP address, port and transport of an outbound proxy, or
   it can be determined through DNS procedures applied to the Request-
   URI. These procedures are described in Section 24, which yield an
   ordered set of address, port and transports to attempt. The UAC
   SHOULD follow the procedures defined there for stateful elements,
   trying each address until a server is contacted. Each try constitutes
   a new transaction, and therefore a new client transaction MUST be
   constructed for each.




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8.1.3 Processing Responses

   Responses are first processed by the transport layer, and then passed
   up to the transaction layer. The transaction layer performs its
   processing, and then passes it up to the TU. The majority of response
   processing in the TU is method specific. However, there are some
   general behaviors independent of the method.

8.1.3.1 Unrecognized Responses

   A UAC MUST treat any response they do not recognize as being
   equivalent to the x00 response code of that class, and MUST be able
   to process the x00 response code for all classes. For example, if a
   UAC 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.

8.1.3.2 Vias

   If more than one Via header field is present in a response, the UAC
   SHOULD discard the message.

        The presence of additional Via header fields that precede
        the originator of the request suggests that the message was
        misrouted or possibly corrupted.

8.1.3.3 Processing 3xx responses

   Upon receipt of a redirection response (e.g. a 3xx response status
   code), clients SHOULD use the URI(s) in the Contact header field to
   formulate a new request.

   To do that, 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.

   In all other respects, requests sent upon receipt of a redirect
   response SHOULD re-use the headers and bodies of the original
   request.

   The Contact values present in redirection responses SHOULD NOT be
   cached across calls, as they may not represent the most desirable
   location for a particular destination address.

8.1.3.4 Processing 4xx responses

   Certain 4xx response codes require specific UA processing,



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   independent of the method.

   If a 401 or 407 response is received, the UAC SHOULD follow the
   authorization procedures of Section 20.2.2 and Section 20.2.3 to
   retry the request with credentials.

   If a 413 response is received (Section 23.4.11), it means that the
   request contained a body that was longer than the UAS was willing to
   accept. If possible, the UAC SHOULD retry the request, either
   omitting the body or using one of a smaller length.

   If a 415 response is received (Section 23.4.13), it means the request
   contained media types not supported by the UAS. The UAC SHOULD retry
   sending the request, this time only using content with types listed
   in the Accept header in the response, with encodings listed in the
   Accept-Encoding header in the response, and with languages listed in
   the Accept-Language in the response.

   If a 420 response is received (Section 23.4.14), it means the request
   contained a Require or Proxy-Require header listing an option-tag for
   a feature not supported by a proxy or UAS. The UAC SHOULD retry the
   request, this time omitting any extensions listed in the Unsupported
   header in the response.

   In all of the above cases, retrying the request is accomplished by
   creating a new request with the appropriate modifications. This new
   request SHOULD have the same value of the Call-ID, To, and From of
   the previous request, but the CSeq should contain a new sequence
   number that is one higher than the previous.

   With other 4xx responses, a retry may or may not be possible
   depending on the method and the use case.

8.2 UAS Behavior

   When a request outside of a dialog is processed by a UAS, there are a
   set of processing rules which are followed, independent of the
   method. Section 12 gives guidance on how a UAS can tell whether a
   request is inside or outside of a dialog.

8.2.1 Authentication/Authorization

   A UAS MAY authenticate the originator of a request, and this process
   may require the server to issue a challenge for credentials. The
   required behavior is independent of the method of the request, and is
   detailed in Section 20.2.

8.2.2 Method Inspection



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   Once a request is authenticated (or no authentication was desired),
   the UAS MUST inspect the method of the request. If the UAS does not
   support the method of a request it MUST generate a 405 (Method Not
   Allowed) response.  Procedures for generation of responses are
   described in Section 8.2.7. The UAS MUST also add an Allow header to
   the 405 response. The Allow header field MUST list the set of methods
   supported by the UAS generating the message.

   The Allow header is presented in Section 22.5.

   If the method is one supported by the server, processing continues.

8.2.3 Header Inspection

   If a UAS does not understand a header field in a request (i.e. the
   header is not defined in this specification or in any supported
   extension), the server MUST ignore that header and continue
   processing the message. A UAS SHOULD ignore any malformed headers
   which are not necessary for processing requests.

8.2.3.1 To and Request-URI

   The To header field identifies the original recipient of the request
   designated by the user identified in the From field. The original
   recipient may or may not be the UAS processing the request, do to
   call forwarding or other proxy operations. A UAS MAY apply any policy
   it wishes in determination of whether to accept requests when the To
   field is not the identity of the UAS. However, it is RECOMMENDED that
   a UAS accept requests even if they do not recognize the URI scheme
   (e.g., a tel: URI) in the To header, or if the To header does not
   address a known or current user of this UAS. If, on the other hand,
   the UAS decides to reject the request, it SHOULD generate a response
   with a 403 status code and send it to the server transaction for
   transmission.

   However, the Request-URI identifies the UAS that is to process the
   request. If the Request-URI does not identify an address that the UAS
   is willing to accept requests for, it SHOULD reject the request with
   a 404 (Not Found) response. If the Request-URI does not provide
   sufficient information for the UAS to determine whether it is willing
   to process the request, it SHOULD return a 485 (Ambiguous) response.
   This response SHOULD contain a Contact header field containing URIs
   of new addresses to be tried.

   Typically, a UA which uses the REGISTER method to bind its address of
   record to a specific contact address, will see requests whose
   Request-URI equals those contact addresses.




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8.2.3.2 Require

   Assuming the UAS decides that it is the proper element to process the
   request, it examines the Require header field, if present.

   The Require general-header field is used by UAC to tell UAS about SIP
   extensions that the UAC expects the UAS to support in order to
   properly process the request. If a UAS does not understand an option
   listed in a Require header field, it MUST respond by generating a
   response with status code 420 (Bad Extension). The UAS MUST add a
   Unsupported, and list in it those options it does not understand
   amongst those in the Require header of the request. Upon receipt of
   the 420 the client SHOULD retry the request, this time without using
   those extensions listed in the Unsupported header in the response.

   Example:

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

   UASS->UAC:   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.

8.2.4 Content Processing

   Assuming the UAS understands any extensions required by the client,
   the UAS examines the body of the message, and the headers that
   describe it. If there are any bodies whose type (indicated by the
   Content-Type), language (indicated by the Content-Language) or
   encoding (indicated by the Content-Encoding) are not understood, and
   that body part is not optional (as indicated by the Content-
   Disposition) header, the UAS MUST reject the request with a 415
   (Unsupported Media Type) response. The response MUST contain a Accept



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   header listing the types of all bodies it understands, in the event
   the request contained bodies of types not supported by the UAS. If
   the request contained content encodings not understood by the UAS,
   the response MUST contain an Accept-Encoding header listing the
   encodings understood by the UAS. If the request contained content
   with languages not understood by the UAS, the response MUST contain
   an Accept-Language header indicating the languages understood by the
   UAS.

   Beyond these checks, body handling is method and type specific.

   For further information on the processing of Content-specific headers
   see Section 7.4.

8.2.5 Applying Extensions

   A UAS that wishes to apply some extension when generating the
   response MUST only do so if support for that extension is indicated
   in the Supported header in the request. If the desired extension is
   not supported, the server SHOULD rely only on baseline SIP and any
   other extensions supported by the client. To ensure that the SHOULD
   can be fulfilled, any specification of a new extension MUST include
   discussion of how to gracefully return to baseline SIP when the
   extension is not present. In rare circumstances, where the server
   cannot process the request without the extension, the server MAY send
   a 421 (Extension Required) response. This response indicates that the
   proper response cannot be generated without support of a specific
   extension. The needed extension(s) MUST be included in a Require
   header in the response. This behavior is NOT RECOMMENDED, as it will
   generally break interoperability.

   Any extensions applied to a non-421 response MUST be listed in a
   Require header included in the response. Of course, the server MUST
   NOT apply extensions not listed in the Supported header in the
   request. As a result of this, the Require header in a response will
   only ever contain option tags defined in standards track RFCs.

8.2.6 Processing the Request

   Assuming all of the checks in the previous subsections are passed,
   the UAS processing becomes method specific. Section 10 deals with the
   REGISTER request, section 11 deals with the OPTIONS request, section
   13 deals with the INVITE request, and section 15 deals with the BYE
   request.

8.2.7 Generating the Response

   When a UAS wishes to construct a response to a request, it follows



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   these procedures. Additional procedures may be needed depending on
   the status code of the response and the circumstances of its
   construction. These additional procedures are documented elsewhere.

   The From field of the response MUST equal the From field of the
   request. The Call-ID field of the response MUST equal the Call-ID
   field of the request. The Cseq field of the response MUST equal the
   Cseq field of the request. The Via headers in the response MUST equal
   the Via headers in the request, and MUST maintain the same ordering.

   If a request contained a To tag in the request, the To field in the
   response MUST equal that of the request. However, if the To field in
   the request did not contain a tag, the URI in the To field in the
   response MUST equal the URI in the To field in the request.
   Additionally, the UAS MUST add a tag to the To field in the response.
   This serves to identify the UAS that is responding, possibly
   resulting in a component of a dialog ID. The same tag MUST be used
   for all responses to that request, both provisional and final.
   Procedures for generation of tags are defined in Section 21.3.

8.3 Redirect Servers

   In some architectures it may be desirable to reduce the processing
   load on proxy servers that are responsible for routing requests by
   relying on redirection.  Redirection allows servers to push routing
   information for a request back in a response to the client, thereby
   taking themselves out of the loop of further messaging for this
   transaction while still aiding in locating the target of the request.
   When the originator of the request receives the redirection it will
   send a new request based on the routing information it has received.
   By propagating routing information from the core of the network to
   its edges, redirection allows for considerable network scalability.

   A redirect server is logically constituted of a server transaction
   layer and a transaction user that has access to a location service of
   some kind (see Section 10 for more on registrars and location
   services). This location service is effectively a database containing
   mappings between a single URI and a set of one or more alternative
   locations at which the target of that URI can be found.

   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 from the location service and either 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 an entire SIP transaction. It is the
   responsibility of clients to detect forwarding loops between redirect



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   servers.

   When a redirect server returns a 3xx response to a request, it
   populates the list of (one or more) alternative locations into
   Contact headers. An "expires" parameter to the Contact header may
   also be supplied to indicate the lifetime of the Contact data.

   The Contact header field contains URIs giving the new locations or
   user names to try, or may simply specify additional transport
   parameters. A 301 or 302 response may also give the same location and
   username that was targeted by the initial request 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.

   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 informational
   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
   URIs. For example, it could contain URL's for phones, fax, or irc (if
   they were defined) or a mailto: (RFC 2368, [18]) URL.

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

   Redirect servers MUST ignore features that are not understood
   (including unrecognized headers, Required extensions, or even method
   names) and proceed with the redirection of the session in question.
   If a particular extension requires that intermediate devices support
   it, the extension MUST be tagged in the Proxy-Require field as well
   (see Section 22.28).

9 Canceling a Request

   The previous section has discussed general UA behavior for generating
   requests, and processing responses, for requests of all methods. In
   this section, we discuss a general purpose method, called CANCEL.

   The CANCEL request, as the name implies, is used to cancel a previous
   request sent by a client. Specifically, it asks the user agent server
   to cease processing the request, and generate an error response to



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   that request. CANCEL has no effect on a request that has already been
   responded to. Because of this, it is most useful to CANCEL requests
   which can take a long time to respond to. For this reason, CANCEL is
   most useful for INVITE requests, which can take a long time to
   generate a response. In that usage, a UAS that receives a CANCEL
   request for an INVITE, but has not yet sent a response, would "stop
   ringing", and then respond to the INVITE with a specific error
   response (a 487).

   Cancel requests can be constructed and sent by any type of client,
   including both proxies and user agent servers. Section 15 discusses
   under what conditions a UAC would CANCEL an INVITE request, and
   Section 16 discusses proxy usage of INVITE.

   Because a stateful proxy can generate its own CANCEL, a stateful
   proxy also responds to a CANCEL, rather than simply forwarding a
   response it would receive from a downstream element. For that reason,
   CANCEL is referred to as a "hop-by-hop" request, since it is
   responded to at each stateful proxy hop.

9.1 Client Behavior

   The following procedures are used to construct a CANCEL request. The
   Request-URI, Call-ID, To, the numeric part of CSeq and From header
   fields in the CANCEL request MUST be identical to those in the
   request being cancelled, including tags. A CANCEL constructed by a
   client MUST have only a single Via header, whose value matches the
   top Via in the request being cancelled. Using the same values for
   these headers allows the CANCEL to be matched with the request it
   cancels (Section 9.2 indicates how such matching occurs). However,
   the method part of the Cseq header MUST have a value of CANCEL. This
   allows it to be identified and processed as a transaction in its own
   right (See Section 17).

   Once the CANCEL is constructed, the client SHOULD check whether any
   response (provisional or final) has been received for the request
   being cancelled (herein referred to as the "original request"). The
   CANCEL request MUST NOT be sent if no provisional response has been
   received, rather, the client MUST wait for the arrival of a
   provisional response before sending the request. If the original
   request has generated a final response, the CANCEL SHOULD NOT be
   sent, as it is an effective no-op, since CANCEL has no effect on
   requests which have already generated a final response. When the
   client decides to send the CANCEL, it creates a client transaction
   for the CANCEL, and passes it the CANCEL request along with the
   destination address, port and transport. The destination address,
   port, and transport for the CANCEL MUST be identical to those used to
   send the original request.



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        If it was allowed to send the CANCEL before receiving a
        response for the previous request the server could receive
        the CANCEL before the original request.

   Note that both the transaction corresponding to the original request
   and the CANCEL transaction will complete independently. However, a
   UAC canceling a request cannot rely on receiving a 487 (Request
   Terminated) response for the original request, as an RFC 2543-
   compliant UAS will not generate such a response. If there is no final
   response for the original request in 64*T1 seconds for an INVITE
   transaction, and T3 seconds for a non-INVITE transaction, the client
   SHOULD then consider the original transaction cancelled and SHOULD
   destroy the client transaction handling the original request.

9.2 Server Behavior

   The CANCEL method requests that the TU at the server side cancel a
   pending request with the same Call-ID, To, From, top Via header and
   Request-URI and CSeq (sequence number only) header field values.

   The processing of a CANCEL request at a server depends on the type of
   server. A stateless proxy will forward it, a stateful proxy might
   respond to it and generate some CANCEL requests of its own, and a UAS
   will respond to it. See Section 16.8 for proxy treatment of CANCEL.

   When a UAS receives a CANCEL, it looks for any server transactions
   which were created by requests with the same To, From, Call-ID, Cseq
   numeric value, Request-URI and top Via header. If no matching
   transactions are found, the CANCEL is responded to with a 481 (Call
   Leg/Transaction Does Not Exist).  If the transaction for the original
   request still exists, the behavior of the UAS on receiving a CANCEL
   request depends on whether it has already sent a final response for
   original request. If it has, the CANCEL request has no effect on the
   processing of the original request, no effect on any session state,
   and no effect on the responses generated for the original request. If
   the UAS has not issued a final response for the original request, it
   immediately responds to the original request with a 487 (Request
   Terminated).

   The CANCEL request itself is answered with a 200 (OK) response in
   either case. Once the response is constructed it is passed to the
   server transaction for the CANCEL request.

10 Registrations

10.1 Overview of Usage

   SIP is a protocol that offers a discovery capability. For one user to



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   initiate a session with another, SIP must discover the current
   host(s) that the called user is reachable at. This discovery process
   is accomplished by SIP proxy servers, which are responsible for
   receiving a request, determining where to send it based on knowledge
   of the location of the user, and then sending it there. To do this,
   proxies consult an abstract service known as a location service ,
   which provides address bindings for a particular domain. These
   address bindings map an incoming SIP URL, sip:bob@Biloxi.com , for
   example, to one or more SIP URLs which are somehow "closer" to the
   desired user, sip:bob@engineering.Biloxi.com , for example.
   Ultimately, a proxy will consult a location service which maps a
   received URL to the current host(s) that a user is logged in to.

   There are many ways by which the contents of the location service can
   be established. One way is administratively. In the above example,
   Bob is known to be a member of the engineering department through
   access to a corporate database. SIP provides a mechanism, however,
   for a user agent to explicitly create a binding in the location
   service of a proxy. This mechanism is known as registration.

   The process of registration entails sending a REGISTER message to a
   special type of UAS known as a registrar. The registrar acts as a
   front end to the location service for a domain, reading and writing
   mappings based on the contents of the REGISTER messages. This
   location service will then be consulted by a proxy server that is
   responsible for routing requests for that domain.

   SIP does not mandate a particular mechanism for implementing the
   location service. The only requirement is that a registrar for some
   domain MUST be capable of reading and writing data to the location
   service, and a proxy for that domain MUST be capable of reading that
   same data. A registrar MAY be co-located with a particular SIP proxy
   server for the same domain, allowing usage of an in memory database
   for the location service. Usage of a shared database is another
   implementation choice. The choice depends entirely on the
   architectural requirements (redundancy, scalability, etc) of a
   particular deployment.

   Registration creates bindings in a location service for a particular
   domain that associate an "address of record" URI with one or more
   "contact addresses".  This means that when a proxy for that domain
   receives a request whose request URI matches the address of record,
   the proxy will forward the request to the contact addresses
   registered to that address of record. Generally, it only makes sense
   to register an address of record at a location service for a domain
   when requests for that address of record would be routed to that
   domain. In most cases, this means that the domain of the registration
   will need to match the domain in the URI of the address of record.



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   The most important usage of the registration mechanism is to inform a
   proxy of the mapping between the address of record and the current
   host on which the UA resides. However, the registration process is a
   general mechanism for establishing bindings, and can be used for
   other purposes (for example, to set up call forwarding).


10.2 Construction of the REGISTER request

   Several operations can be performed with a REGISTER method with
   respect to a registrar. One of these is the basic registration
   operation that is described above, which provides a new binding
   between an address of record and one or more contact addresses.
   Registration on behalf of a particular address of record may be
   performed by a third party if they are authorized to do so. A client
   may also remove previous bindings, or query to determine which
   bindings are currently in place for an address of record.

   Aside from the exceptions noted in this and the following sections,
   the construction of the REGISTER method, and behavior of clients
   sending a REGISTER is identical to the general UAC behavior described
   in Section 8.1 and Section 17.1. Regardless of the operation that is
   performed by a REGISTER, the following header fields MUST be
   formulated as follows:

        Request-URI: The Request-URI names the domain of the location
             service that the registration is meant for (e.g.
             "chicago.com"). The user name MUST be empty.

        To: The To header field contains the address of record whose
             registration is to be created or modified. Note that the
             initial To header field and the Request-URI field SHOULD
             therefore be different in a REGISTER message.

        From: The From header field contains the address of record of
             the person responsible for the registration, which MAY be
             identical to the value of the To header field. For third-
             party registrations the From header field and To header
             field are different.

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

             If different Call-IDs were used for overlapping
             REGISTER messages coming from the same client, the
             registrar might have trouble determining their
             ordering.



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        Contact: REGISTER requests MAY contain one or more Contact
             header fields. Contact addresses are presented in the
             Contact header fields of REGISTER requests.

   Note that user agents MUST NOT send a new registration (containing
   new Contact header fields, as opposed to a retransmission) until they
   have received a response from the registrar for the previous one.

   The following optional Contact header parameters also contain
   behavior specific to the registration process.

        action: The "action" parameter has been deprecated.  UACs SHOULD
             NOT use the "action" parameter.

        expires: The "expires" parameter indicates how long the UAC
             would like the binding to be 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 binding is
             valid. Implementations MAY treat values larger than 2**32-1
             (4294967295 seconds or 136 years) as equivalent to 2**32-1.

10.2.1 Adding Bindings with REGISTER

   For a simple registration, a REGISTER request sent to a registrar
   includes contact addresses to which requests should be forward for
   the originating user's address of record. The address of record
   itself (i.e. 'sip:carol@chicago.com') MUST populate the To header of
   the REGISTER. The Contact header fields of the request typically
   contain SIP URIs that identify particular SIP endpoints (i.e.
   'sip:carol@cube2214a.chicago.com'), but they MAY use any URI scheme;
   this way a SIP UA can choose to register telephone numbers (with the
   tel URL, [13]) or email addresses (with a mailto URL, [18]) as
   Contacts for an address of record.

   For example, if Carol, whose address of record is to register with
   the registrar associated with the location service of chicago.com.
   This location service would then be accessed by a proxy server that
   receives requests targeting users in the chicago.com domain, and
   hence new requests for Carol's address of record will be routed to
   her SIP endpoint.

   Once a client has established bindings at a registrar, it MAY send
   subsequent registrations containing new bindings or modifications to
   pre-existing bindings as necessary. The 2xx response to the REGISTER
   message will contain (in Contact header fields) a complete list of
   bindings that have been registered for this address of record at this
   registrar.



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                                                   bob
                                                 +----+
                                                 | UA |
                                                 |    |
                                                 +----+
                                                    |
                                                    |3)INVITE
                                                    |   carol@chicago.com
           chicago.com        +--------+            V
           +---------+ 2)Store|Location|4)Query +-----+
           |Registrar|=======>| Service|<=======|Proxy|sip.chicago.com
           +---------+        +--------+=======>+-----+
                 A                      5)Resp      |
                 |                                  |
                 |                                  |
       1)REGISTER|                                  |
                 |                                  |
              +----+                                |
              | UA |<-------------------------------+
     cube2214a|    |                            6)INVITE
              +----+                    carol@cube2214a.chicago.com
               carol























   Figure 2: REGISTER example

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10.2.1.1 Setting the Expiration Interval of Contact Addresses

   When a client sends a REGISTER request, it MAY suggest an expiration
   interval that indicates how long the client would like the
   registration to be valid (although as is detailed in Section 10.3,
   the registrar has the ultimate say).

   There are two ways in which a client can suggest an expiration
   interval for a binding: through an Expires header, or an "expires"
   Contact header parameter. The latter allows expiration intervals to
   be suggested on a per-binding basis when more than one binding is
   given in a single REGISTER, whereas the former suggests an expiration
   interval for all Contact header fields that do not contain the
   "expires" parameter.

   If neither mechanism for expressing a suggested expiration time is
   present in a REGISTER, a default suggestion of one hour is assumed.

10.2.1.2 Setting Preference among Contact Addresses

   If more than one Contact is sent in a REGISTER, then the registering
   UA intends to associate all of the URIs given in these Contact
   headers with the address of record present in the To field. This list
   can be prioritized with the "q" mechanism.

        q: The "q" parameter indicates a relative preference for the
             particular Contact header field compared to other bindings
             present in this REGISTER message or existing within the
             location service of the registrar. For an example of how a
             proxy server uses "q" values, see Section 16.5.

10.2.2 Removing Bindings with REGISTER

   Registrations are removed from the registrar through an expiration
   process; registrations are soft state and need to be refreshed
   periodically. A client may attempt to influence the expiration
   intervals selected by the registrar as described in Section 10.2.1.

   A registering user agent requests the immediate removal of a binding
   by specifying an expiration interval of "0" for that contact address
   in a REGISTER. It is RECOMMENDED that user agents support this
   mechanism so that bindings can be removed (for whatever reason)
   before their expiration interval has passed.

   The REGISTER-specific Contact header field value of "*" applies to
   all registrations, but it MUST only be used when the Expires header
   is present with a value of "0".




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        Use of the "*" Contact header field value allows a
        registering user agent to remove all of its bindings
        expediently.

10.2.3 Fetching Bindings with REGISTER

   If no Contact headers are present in a REGISTER, then the UA is not
   in fact registering any new bindings, and the list of bindings is
   therefore left unchanged. As noted above, in a successful response to
   this REGISTER message, the complete list of existing bindings is
   returned, and thus a REGISTER without Contact headers serves as a
   fetch operation.

10.2.4 Refreshing Registrations

   When a 2xx response has been received by the client for a REGISTER
   request, the client MUST determine when each of the bindings
   enumerated in the response needs to be refreshed. This may include
   bindings that were registered in previous REGISTER transactions.

   Since the list of bindings returned in the response to a REGISTER may
   contain bindings that were not included in this REGISTER transaction,
   the client must correlate Contact header fields in the response with
   the Contact header fields it sent in the request in order to
   establish proper expiration timers. This correlation should be
   performed in accordance with the URI comparison rules given in
   Section 21.1.4.

   The registering UA MUST re-register each contact address at least as
   often as the mandated expiration interval. A REGISTER that refreshes
   a binding SHOULD have the same Call-ID as the request which created
   the binding. The CSeq header SHOULD have a numeric sequence number
   that is one higher than the value sent in the last request with the
   same Call-ID.

   Note that a UA MUST must update its expiration timers for refreshing
   each binding every time it receives a response to a registration
   request.

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

10.2.5 Discovering a Registrar

   Depending on the policy of their administrative domain, SIP UAs can
   be configured with the address of a local registrar. Some UAs may be
   equipped with protocol tools (outside the scope of SIP) that allow
   them to discover their local registrar dynamically.



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   Note that as an alternate means of discovering a registrar if no
   local registrar is configured in the user agent, clients MAY register
   via multicast. 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 (see [19]),
   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 (see [20]); 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.

   If a SIP UA knows of an appropriate registrar it SHOULD attempt to
   register with this server periodically - management of registration
   intervals is detailed below.

10.3 Processing of REGISTER at the Registrar

   A registrar is a UAS that responds to a REGISTER request, and stores
   the information gathered from that request in a location service that
   is in turn accessible to proxy servers within its administrative
   domain. A registrar handles requests as a UAS (in conformity with
   Section 8.2 and Section 17.2) but it accepts only the REGISTER method
   and generates only the responses detailed in this section. Note that
   the REGISTER method also does not support the Record-Route or Route
   header, and that proxy servers MUST NOT add Record-Route headers to
   REGISTER requests.

   A registrar must know (through provisioning or some other mechanism)
   the set if administrative domain(s) for which its associated location
   service(s) are responsible. REGISTER requests MUST be processed by a
   registrar in the order that they are received.

   Upon the arrival of a REGISTER message, the registrar MUST inspect
   the Request-URI to determine whether it has access to a location
   service responsible for the domain to which this request is
   addressed. If this message is for some other administrative domain,
   then if the registrar can act as a proxy server, it SHOULD forward
   the request to the addressed domain (following the general behavior
   for proxying messages described in Section 16).

   When a registrar receives a REGISTER message, it is RECOMMENDED that
   the registrar authenticate the user agent client. Mechanisms for the



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   authentication of SIP user agents are described in Section 20.2;
   registration behavior in no way overrides the generic authentication
   framework for SIP. If no authentication mechanism is available, the
   registrar MAY take the From address as the asserted identity of the
   originator of the request.

   Once the identity of the registering user has been ascertained, it is
   RECOMMENDED that the registrar determine if the authenticated user
   agent is authorized to request and/or modify registrations for this
   address of record. For example, a registrar might consult a
   authorization database (directly or through an appropriate protocol)
   that maps credentials or other tokens of identity resulting from
   authentication to one or more addresses of record for which this
   identity is responsible.


        Note that in architectures that support third-party
        registration, one entity may be responsible for updating
        the registrations associated with multiple addresses of
        record.

   When the registrar has determined that the client is permitted to
   make the request, the registrar MUST extract the address of record
   from the To header field of the REGISTER. Note that the registrar
   MUST extract the entire To header field URI in order to use it as an
   index in the location service.

   Next, the registrar MUST query its location service (the repository
   of previously registered bindings) for the set of bindings associated
   with this address of record. If the address of record is not valid
   for this administrative domain (for example, because the username is
   not assigned), then the registration attempt fails (see below). A
   full URI comparison (as described in Section 21.1.4) MUST be
   performed to determine whether a given binding matches this address
   of record.

   The registrar now MUST extract all the Contact header fields from the
   REGISTER message (note that there may be no Contact header field).

   Each contact address in a REGISTER MUST now be compared to all
   existing registrations at this location service according to the
   rules in Section 21.1.4. Note that URIs other than SIP URIs in
   contact addresses MUST be compared according to the standard URI
   equivalency rules for the URI schema in question.

   If a match is found among pre-existing registrations, the registrar
   MUST copy all parameters associated with the current Contact header
   field from the REGISTER message into the pre-existing binding in its



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   location service (overwriting with changed values any existing
   parameters as necessary, with the exception of "expires"). Expiration
   intervals for this contact address MUST also be reset, based on any
   suggested expiration in the REGISTER (remember that this can be "0").

   If no match is found among the set of pre-existing registrations, the
   registrar MUST create a new binding in its location service between
   the address of record and the current Contact header field. All
   Contact header field parameters are copied verbatim into this new
   binding (again with the exception of "expires"). An expiration
   interval MUST be selected by the registrar, taking into account any
   suggested expiration for this contact address in the REGISTER.


        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 likelihood of registrations
        going stale.

   The expiration interval mandated by the registrar may be either
   longer or shorter than the interval suggested by the sender of the
   REGISTER, though the registrar SHOULD abide by the registering
   client's suggestion.


        A server MAY decide to lengthen the expiration interval if
        the refresh rate of a particular client exceeds a
        threshold, for example.

   After the expiration interval selected by the registrar for a binding
   has passed, if the binding has not been refreshed (increasing the
   expiration interval), the registrar SHOULD silently discard the
   binding.

   Once all bindings in the location service have been updated to
   reflect any changes present to contact addresses in the REGISTER
   message, the registrar MUST remove any bindings that expire
   immediately.


        The REGISTER might have set the expiration interval for
        some bindings to "0" to remove them before their expiration
        interval passes.

   Finally, the registrar must generate a response. If the address of
   record given in the To header field of the REGISTER method is valid
   for its administrative domain, then a 200 response MUST be sent,



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   which MUST contain a complete list (within Contact header fields) of
   the currently valid bindings in the location service associated with
   the address of record contained in the To field of the REGISTER
   request. This list MAY be empty (in which case the 200 would not
   contain any Contact headers).

   In a successful response to a REGISTER, wherein the bindings for this
   address of record are enumerated as described above, the registrar
   MUST supply an expiration interval for each contact address in either
   an "expires" parameter of a Contact header or an Expires header. This
   interval specifies the expiration interval that has been mandated by
   the registrar (taking into account the registering UA's suggestion).

   If the registration failed because the address of record contained in
   the To field of the REGISTER is not valid for this domain, then a 404
   MUST be sent.

11 Querying for Capabilities

   The SIP method OPTIONS allows a client to query another client or
   server as to its capabilities. This allows a client to discover
   information about the methods, content types, extensions, codecs etc.
   supported without actually "ringing" the other party. For example,
   before a client inserts a Require header field into an INVITE listing
   an option that it is not certain the destination UAS supports, the
   client can query the destination UAS with an OPTIONS to see if this
   option is returned in a Supported header field.

   The target of the OPTIONS request is identified by the Request-URI,
   which could identify another User Agent or a SIP Server.
   Alternatively, a server receiving an OPTIONS request with a Max-
   Forwards header value of 0 MAY respond to the request regardless of
   the Request-URI.


        This behavior is common with HTTP/1.1.

   An OPTIONS request sent as part of an established dialog does not
   have any impact on the dialog.

11.1 Construction of OPTIONS Request

   An OPTIONS request is constructed using the standard rules for a SIP
   request as discussed Section 8.1.1.

   A Contact header field MAY be present in an OPTIONS.





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        OPEN ISSUE #197: What is the semantic of this Contact

   An Accept header field SHOULD be included to indicate the type of
   message body the UAC wishes to receive in the response.

   Example OPTIONS request:


     OPTIONS sip:carol@chicago.com SIP/2.0
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=23411513a6
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: <sip:carol@chicago.com>
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 63104 OPTIONS
     Contact: <sip:alice@10.1.3.3>
     Accept: application/sdp
     Contact-Length: 0



11.2 Processing of OPTIONS Request

   The response to an OPTIONS is constructed using the standard rules
   for a SIP response as discussed in Section 8.2.7.  The response code
   chosen is the same that would have been chosen had the request been
   an INVITE. That is, a 200 (OK) would be returned if the UAS is ready
   to accept a call, a 486 (Busy Here) would be returned if the UAS is
   busy, etc. This allows an OPTIONS request to be used to determine the
   basic state of a UAS, which can be an indication of whether the UAC
   will accept an INVITE request.

   Note that this use of OPTIONS has limitations due the differences in
   proxy handling of OPTIONS and INVITE requests. While a forked INVITE
   can result in multiple 200 OK responses being returned, a forked
   OPTIONS will only result in a single 200 OK response, since it is
   treated by proxies using the non-INVITE handling. See Section 13.2.1
   for the normative details.

   Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
   fields SHOULD be present in a 200 OK response to an OPTIONS request.

   A Contact header field MAY be present in a 200 OK response.

   A Warning header field MAY be present.

   A message body MAY be sent, the type of which is determined by the
   Accept header in the OPTIONS request.



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   Example OPTIONS response (corresponding to the request in Section
   11.1):


     SIP/2.0 200 OK
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=23411513a6
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: <sip:carol@chicago.com>;tag=93810874
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 63104 OPTIONS
     Contact: <sip:carol@10.3.6.6>
     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
     Accept: application/sdp
     Accept-Encoding: gzip
     Accept-Language: en
     Supported: foo
     Content-Type: application/sdp
     Contact-Length: 274

     v=0
     o=carol 28908764872 28908764872 IN IP4 10.3.6.6
     s=-
     t=0 0
     c=IN IP4 10.3.6.6
     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



12 Dialogs

   A key concept for a user agent is that of a dialog. A dialog
   represents a peer- to-peer SIP relationship between a two user agents
   that persists for some time.  The dialog facilitates sequencing of
   messages between the user agents, and proper routing of requests
   between both them. The dialog represents a context in which to
   interpret SIP messages. The previous section discussed method
   independent UA processing for requests and responses outside of a
   dialog. This section discusses how those requests and responses are
   used to construct a dialog, and then how subsequent requests and
   responses are sent within a dialog.



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   A dialog is identified at each UA with a dialog ID, which consists of
   a Call-ID value, a local URI and local tag (together called the local
   address), and a remote URI and remote tag (together called the remote
   address).  The dialog ID at each UA involved in the dialog is not the
   same. Specifically, the local URI and local tag at one UA are
   identical to the remote URI and remote tag at the peer UA. The tags
   are opaque tokens that facilitate the generation of unique dialog
   IDs.

   A dialog ID is also associated with all responses, and with any
   request that contains a tag in the To field. The rules for computing
   the dialog ID of a message depend on whether the entity is a UAC or
   UAS. For a UAC, the Call-ID value of the dialog ID is set to the
   Call-ID of the message, the remote address is set to the To field of
   the message, and the local address is set to the From field of the
   message (these rules apply to both requests and responses). As one
   would expect, for a UAS, the Call-ID value of the dialog ID is set to
   the Call-ID of the message, the remote address is set to the From
   field of the message, and the local address is set to the To field of
   the message.

   A dialog contains certain pieces of state needed for further message
   transmissions within the dialog. This state consists of the Call-ID,
   a local sequence number (used to order requests from the UA to its
   peer), a remote sequence number (used to order requests from its peer
   to the UA), and a route set, which is an ordered list of URIs. The
   route set is the set of servers that need to be traversed to send a
   request to the peer. A dialog can also be in the "early" state, which
   occurs when it is created with a provisional response, and then
   transition to the "established" state when the final response comes.

12.1 Creation of a Dialog

   Dialogs are created through the generation of non-failure responses
   to requests with specific methods. Within this specification, only
   the 2xx and 1xx responses to INVITE establish a dialog. A dialog
   established by a non-final response to a request is called an early
   dialog. Extensions MAY define other means for creating dialogs.
   Section 13 gives more details that are specific to the INVITE method.
   Here, we describe the process for creation of dialog state that is
   not dependent on the method.

12.1.1 UAS

   When a UAS responds to a request with a response that establishes a
   dialog (such as a 2xx to INVITE), the UAS MUST copy all Record-Route
   headers from the request into the response, and MUST maintain the
   order of those headers. This includes the URIs, URI parameters, and



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   any Record-Route header parameters, whether they are known or unknown
   to the UAS. The UAS MUST add a Contact header field to the response.
   The Contact header field contains an address where the UAS would like
   to be contacted for subsequent requests in the dialog (which includes
   the ACK for a 2xx response in the case of an INVITE). Generally, the
   host portion of this URI is the IP address of the host, or its FQDN.
   The URI provided in the Contact header MUST be a SIP URL.

   The UAS then constructs the state of the dialog. This state MUST be
   maintained for the duration of the dialog. First, the route set MUST
   be computed by following these steps:

        1.   The list of URIs in the Record-Route headers in the
             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.


             Contact was not mandatory in RFC2543. Thus, if the UAS
             is talking to an older UAC, the UAC might not have
             inserted the Contact header.

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


        These rules clearly imply that a UA MUST be able to parse
        and process Record-Route header fields. This is a change
        from RFC2543, where all record-route and route processing
        was optional for user agents.

   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
   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. This
   is needed in order for the UA to determine whether the bottom most
   value can be updated from subsequent requests; if it was constructed
   from a Contact, it can be updated.

   The remote sequence number sequence number MUST be set to the value
   of the sequence number in the Cseq header of the request. The local
   sequence number MUST be empty. The call identifier component of the
   dialog ID MUST be set to the value of the Call-ID in the request. The
   local address component of the dialog ID MUST be set to the To field
   in the response to the request (which therefore includes the tag),



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   and the remote address component of the dialog ID MUST be set to the
   From field in the request. A UAS MUST be prepared to receive a
   request without a tag in the From field, in which case the tag is
   considered to effectively have a value of null.

        This is to maintain backwards compatibility with RFC2543,
        which did not mandate From tags.

12.1.2 UAC

   When a UAC receives a response that establishes a dialog, it
   constructs the state of the dialog. This state MUST be maintained for
   the duration of the dialog. First, the route set MUST be computed by
   following these steps:

        1.   The list of URIs present in the Record-Route headers in the
             response are taken, if present, including all URI
             parameters, and their order is reversed.

        2.   The URI in the Contact header from the response, if
             present, is taken, including all URI parameters, 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

   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
   response. The UAC 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. This
   is needed in order for the UA to determine whether the bottom most
   value can be updated from subsequent requests; if it was constructed
   from a Contact, it can be updated.

   The local sequence number sequence number MUST be set to the value of
   the sequence number in the Cseq header of the request. The remote
   sequence number MUST be empty (it is established when the UA sends a
   request within the dialog). The call identifier component of the
   dialog ID MUST be set to the value of the Call-ID in the request. The
   local address component of the dialog ID MUST be set to the From
   field in the request, and the remote address component of the dialog
   ID MUST be set to the To field of the response. A UAC MUST be
   prepared to receive a response without a tag in the To field, in
   which case the tag is considered to effectively have a value of null.

        This is to maintain backwards compatibility with RFC2543,
        which did not mandate To tags.



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12.2 Requests within a Dialog

   Once a dialog has been established between two UAs either of them MAY
   initiate new transactions as needed within the dialog. However, a
   dialog imposes some restrictions on the use of simultaneous
   transactions.

   A TU MUST NOT initiate a new regular transaction within a dialog
   while a regular transaction is in progress (in either direction)
   within that dialog.


        OPEN ISSUE #113: Should we relax the constraint on non-
        overlapping regular transactions?

   A refresh request sent within a dialog is defined as a request that
   can modify the route set of the dialog. For dialogs that have been
   established with an INVITE, the only refresh request defined is re-
   INVITE (see Section 14). Other extensions may define different
   refresh requests for dialogs established in other ways.

        Note that an ACK is NOT a refresh request.

12.2.1 UAC Behavior

12.2.1.1 Generating the Request

   A request within a dialog is constructed by using many of the
   components of the state stored as part of the dialog.

   The To header field of the request MUST be set to the remote address,
   and the From header field MUST be set to the local address (both
   including tags, assuming the tags are not null).

   The Call-ID of the request MUST be set to the Call-ID of the dialog.
   Requests within a dialog MUST contain strictly monotonically
   increasing and contiguous CSeq sequence numbers (increasing-by-one)
   in each direction. Therefore, if the local sequence number is not
   empty, the value of the local sequence number MUST be incremented by
   one, and this value MUST placed into the Cseq header. If the local
   sequence number is empty, an initial value MUST be chosen using the
   guidelines of Section 8.1.1.4. The method field in the Cseq header
   MUST match the method of the request.


        With a length of 32 bits, a client could generate, within a
        single call, one request a second for about 136 years
        before needing to wrap around.  The initial value of the



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        sequence number is chosen so that subsequent requests
        within the same call will not wrap around. A non-zero
        initial value allows clients to use a time-based initial
        sequence number. A client could, for example, choose the 31
        most significant bits of a 32-bit second clock as an
        initial sequence number.

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

   The UAC takes the list of URI in the route set inserted into the
   request URI of the request, including all URI 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, MUST be placed into a Route header field into the
   request, in order.

   A TU SHOULD follow the rules just mentioned to build the Request-URI
   of the request, regardless of whether the UA uses an outbound proxy
   server or not. However, in some instances, a UA may not be willing or
   capable of sending the request to the top element in the route set is
   not capable of DNS, and therefore may not be able to follow those
   procedures.  In these cases, the UA MAY send the request to a local
   outbound server. In this case, it MUST NOT remove the top Route
   header.

        In dialogs created by an INVITE, if the UA is the caller,
        it sets the Request-URI to the same value it used for the
        initial request, and sends it to its local outbound server.

   Bug#161: Which Request-URI does the callee use?

   A UAC SHOULD include a Contact header in any refresh requests within
   a dialog, and unless there is a need to change it, the URI SHOULD be
   the same as used in previous requests within the dialog. As discussed
   in Section 12.2.2, a Contact header in a refresh request updates the
   route set. This allows a UA to provide a new contact address, should
   its address change during the duration of the dialog.

   However, requests that are not refresh requests do not affect the
   route set for the dialog.

   Once the request has been constructed, the address of the server is
   computed and the request is sent, using the same procedures for
   requests outside of a dialog (Section 8.1.1).

12.2.1.2 Processing the Responses




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   The UAC will receives responses to the request from the transaction
   layer.

   The behavior of a UAC that receives a 3xx response for a request sent
   within a dialog is the same as if the request would have been sent
   outside a dialog. This behavior is described in Section  13.2.2.

        Note however that when the UAC tries alternative locations
        it still uses the route set for the dialog to build the
        Route header of the request.

   If a UAC has a route set for a dialog, and receives a 2xx response to
   a refresh it sent, the Contact header field of the response is
   examined.  If not present, the route set remains unchanged. If the
   response had a Contact header field, and the boolean variable
   CONTACT_SET is false, the URL in the Contact header field in the
   response is added to the bottom of the route set , and CONTACT_SET is
   set to true. If the refresh request response had a Contact header
   field, and CONTACT_SET is true, the URL in the Contact header field
   of the response to the refresh request replaces the bottom value in
   the route set is responded with a non-2xx final response the route
   set remains unchanged as if no refresh request had been issued.

   If the response for the a request within a dialog is a 481
   (Call/Transaction Does Not Exist) or a 408 (Request Timeout) the UAC
   SHOULD terminate the dialog.

        For INVITE initiated dialogs terminating the dialog
        consists of sending a BYE.

12.2.2 UAS behavior

   The UAS will receive the request from the transaction layer. If the
   request has a tag in the To header field, the UAS core computes the
   dialog identifier corresponding to the request and compares it with
   existing dialogs.  If there is a match, this is a mid-dialog request.
   In that case, the same processing rules for requests outside of a
   dialog, discussed in Section 8.2, are applied by the UAS once the
   request is received from the transaction layer.

   Requests that do not change in any way the state of a dialog may be
   received within a dialog (e.g., an OPTIONS request). They are
   processed as if they had been received outside the dialog.

   Requests within a dialog MAY contain Record-Route and Contact header
   fields. However, requests that are not refresh requests do not update
   the route set for the dialog. This specification only defines one
   refresh request: re-INVITE (see Section  14).



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   Special rules apply when updated Record-Route or Contact header
   fields are received inside a refresh request. If a UAS has a route
   set for a dialog, and receives a refresh for that dialog containing
   Record-Route header fields, it MUST copy those header fields into any
   2xx response to that request. If the boolean variable CONTACT_SET is
   true, the Contact header field in the request (if present) replaces
   the last entry in the route set the UAS MUST add the URL in the
   Contact header field 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 field, the route-set at the UAS remains unchanged.

   If the remote sequence number is empty, it MUST be set to the value
   of the sequence number in the Cseq header in the request. If the
   remote sequence number was not empty, but the sequence number of the
   request is lower than the remote sequence number, the request is out
   of order and MUST be rejected with a 500 response. If the remote
   sequence number was not empty, and the sequence number of the request
   is greater than the remote sequence number, the request is in order.
   It is possible for the CSeq header to be higher than the remote
   sequence number 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 previous received request.  The
   UAS MUST then set the remote sequence number to the value of the
   sequence number in the Cseq header in the request.

12.3 Termination of a Dialog

   Dialogs can end in several different ways, depending on the method.
   When a dialog is established with INVITE, it is terminated with a
   BYE. No other means to terminate a dialog are described in this
   specification, but extensions can define other ways.

13 Initiating a Session

13.1 Overview

   When a user agent client desires to initiate a session (for example,
   audio, video, or a game), it formulates an INVITE request. The INVITE
   request asks a server to establish a session. This request is
   forwarded by proxies, eventually arriving at one or more UAS which
   can potentially accept the invitation. These UAS's will frequently
   need to query the user about whether to accept the invitation. After
   some time, those UAS can accept the invitation (meaning the session
   is to be established) by sending a 2xx response. If the invitation is
   not accepted, a 3xx,4xx,5xx or 6xx response is sent, depending on the
   reason for the rejection. Before sending a final response, the UAS
   can also send a provisional response (1xx) to advise the UAC of
   progress in contacting the called user.



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   After possibly receiving one or more provisional responses, the UA
   will get one or more 2xx responses or one non-2xx final response.
   Because of the protracted amount of time it can take to receive final
   responses to INVITE, the reliability mechanisms for INVITE
   transactions differ from those of other requests (like OPTIONS). Once
   it receives a final response, the UAC needs send an ACK for every
   final response it receives. The procedure for sending this ACK
   depends on the type of response. For final responses between 300 and
   699, the ACK processing is done in the transaction layer, and follows
   one set of rules (See Section 17). For 2xx responses, the ACK is
   generated by the UAC core.

   A 2xx response to an INVITE establishes a session, and it also
   creates a dialog between the UA that issued the INVITE and the UA
   that generated the 2xx response. Therefore, when multiple 2xx
   responses are received from different remote UAs (because the INVITE
   forked), each 2xx establishes a different dialog. All these dialogs
   are part of the same call.

   This section provides details on the establishment of a session using
   INVITE.

13.2 Caller Processing

13.2.1 Creating the Initial INVITE

   Since the initial INVITE represents a request outside of a dialog,
   its construction follows the procedures of Section 8.1.1. Additional
   processing is required for the specific case of INVITE.

   An Allow header field (Section  22.5) SHOULD be present in the
   INVITE. It indicates what methods can be invoked within a dialog, on
   the UA sending the INVITE, for the duration of the dialog. For
   example, a UA capable of receiving INFO requests within a dialog [21]
   SHOULD include an Allow header listing the INFO method.

   A Supported header field (Section  22.35) SHOULD be present in the
   INVITE. It enumerates all the extensions understood by the UAC.

   An Accept (Section  22.1) header field MAY be present in the INVITE.
   It indicates which content-types are acceptable to the UA, in both
   the response received by it, and in any subsequent requests sent to
   it within dialogs established by the INVITE. The Accept header is
   especially useful for indicating support of various session
   description formats.

   The UA MAY add an Expires header field (Section  22.19) to limit the
   validity of the invitation. If the time indicated in the Expires



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   header field is reached and no final answer for the INVITE has been
   received the UAC core SHOULD generate a CANCEL request for the
   original INVITE.

   A UAC MAY also find useful to add, among others, Subject (Section
   22.34), Organization (Section  22.24) and User-Agent (Section  22.39)
   header fields. They all contain useful information related to the
   INVITE.

   The UAC MAY choose to add a message body to the INVITE. Section
   8.1.1.9 deals with how to construct the header fields- Content-Type
   among others- needed to describe the message body.

   There are special rules for message bodies that contain a session
   description - their corresponding Content-Disposition is "session".
   SIP uses an offer/answer model where one UA sends a session
   description, called the offer, which contains a proposed description
   of the session. The offer indicates the desired communications means
   (audio, video, games), parameters of those means (such as codec
   types) and addresses for receiving media from the offerer. The other
   UA responds with another session description, called the answer,
   which indicates which communications means are accepted, the
   parameters which apply to those means, and addresses for receiving
   media from the answerer. The offer/answer model can be mapped into
   the INVITE transaction in two ways. The first, which is the most
   intuitive, is that the INVITE contains the offer, the 2xx response
   contains the answer, and no session description is provided in the
   ACK. In this model, the UAC is the offerer, and the UAS is the
   answerer.  A second model is that the INVITE contains no session
   description, the 2xx response contains the offer, and the ACK
   contains the answer. In this model, the UAS is the offerer, and the
   UAC is the answerer. The second model is useful for gateways from
   H.323v1 to SIP, where the H.323 media characteristics are not known
   until the call is established. This is also useful for sessions that
   use third-party call control. As a result of these models, if the
   INVITE contains a session description, the ACK MUST NOT contain one.
   Conversely, if the caller chooses to omit the session description in
   the INVITE, the ACK MUST contain one (if a 2xx response is received).
   2xx responses to an INVITE MUST always contain a session description.
   All user agents that support INVITE MUST support both models.

   The Session Description Protocol (SDP) [6] MUST be supported by all
   user agents as a means to describe sessions, and its usage for
   construction offers and answers MUST follow the procedures defined in
   [22].

   Note that the restrictions of the offer-answer model (session
   description only in the INVITE OR in the ACK, but not in both) just



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   described only apply to bodies whose Content-Disposition header field
   is "session".  Therefore, it is possible that both the INVITE and the
   ACK contain a body message (e.g., the INVITE carries a photo
   (Content-Disposition:  render) and the ACK a session description
   (Content-Disposition:  session) ).

        If the Content-Disposition header field is missing, bodies
        of Content-Type application/sdp imply the disposition
        "session", while other content types imply "render".

   Once the INVITE has been created, the UAC follows the procedures
   defined for sending requests outside of a dialog (Section 8). This
   results in the construction of a client transaction that will
   ultimately send the request and deliver responses to the UAC.

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

13.2.2 Processing INVITE Responses

   Once the INVITE has been passed to the INVITE client trasaction, the
   UAC waits for responses for the INVITE. Responses are matched to
   their corresponding INVITE because they have the same Call-ID, the
   same From header field, the same To header field, excluding the tag,
   and the same CSeq. Rules for comparisons of these headers are
   described in Section 22.

13.2.2.1 1xx responses

   Zero, one or multiple provisional responses may arrive before one or
   more final responses are received. Provisional responses for an
   INVITE request can create "early dialogs". If a provisional response
   has a tag in the To field, and if the dialog ID of the response does
   not match an existing dialog, one is constructed using the procedures
   defined in Section 12.1.0.2.

   The early dialog will only be needed if the UAC needs to send a
   request to its peer within the dialog before the initial INVITE
   transaction completes.  Header fields present in a provisional
   response are applicable for the duration of the early dialog (e.g.,
   an Allow header field in a provisional response contains the methods
   that can be used in the early dialog).

13.2.2.2 3xx responses




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   A 3xx response may contain a Contact header field providing new
   addresses where the callee might be reachable.  Depending on the
   status code of the 3xx response (see Section 23.3) the UAC MAY choose
   to try those new addresses.

13.2.2.3 4xx, 5xx and 6xx responses

   A single non-2xx final response may be received for the INVITE. 4xx,
   5xx and 6xx responses may contain a Contact header field indicating
   the location where additional information about the error can be
   found.

   All early dialogs are considered terminated upon reception of the
   non-2xx final response.

   After having received the non-2xx final response the UAC core
   considers the INVITE transaction completed. The INVITE client
   transaction handles generation of ACKs for the response (see Section
   17).

13.2.2.4 2xx responses

   Multiple 2xx 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
   dialog, with a distinct dialog identifier.

   If the dialog identifier in the 2xx response matches the dialog
   identifier of an existing dialog, the dialog MUST be transitioned to
   the "established", and the route set for the dialog MUST be
   recomputed based on the 2xx response using the procedures of Section
   12.1.0.2. Otherwise, a new established dialog is constructed in the
   same fashion.


        The route set only is recomputed for backwards
        compatibility. RFC 2543 did not mandate mirroring of
        Record-Route headers in a 1xx, only 2xx. However, we cannot
        update the entire state of the dialog, since mid-dialog
        requests may have been sent within the early call leg,
        modifying the sequence numbers, for example.

   The UAC core MUST generate an ACK request for each 2xx received from
   the transaction layer. The header fields of the ACK are constructed
   in the same way as for any request sent within a dialog (see Section
   12) with the exception of the CSeq. The sequence number of the CSeq
   header field MUST be the same as the INVITE being acknowledged, but
   the CSeq method MUST be ACK. If the INVITE did not contain an offer,



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   the 2xx will contain one, and therefore the ACK MUST carry an answer
   in its body.

   Once the ACK has been constructed, the procedures of Section 24 are
   used to send it. However, the request is passed to the transport
   layer directly for transmission, rather than a client transaction.
   This is because the UAC core handles retransmissions of the ACK, not
   the transaction layer. The ACK MUST be passed to the client transport
   every time a retransmission of the 2xx final response that triggered
   the ACK arrives.

   The UAC core considers the INVITE transaction completed 62*T1 seconds
   after the reception of the first 2xx response. At this point all the
   early dialogs that have not transitioned to established dialogs are
   terminated. Once the INVITE transaction is considered completed by
   the UAC core, no more new 2xx responses are expected to arrive.

   If, after acknowledging any 2xx response to an INVITE, the caller
   does not want to continue with that dialog, then the caller MUST
   terminate the dialog by sending a BYE request as described in Section
   15.

13.3 Callee Processing

13.3.1 Processing of the INVITE

   The UAS core will receive INVITE requests from the transaction layer.
   It first performs the request processing procedures of Section 8.2,
   which are applied for both requests inside and outside of a dialog.

   Assuming these processing states complete without generating a
   response, the UAS core performs the additional processing steps:

        1.   If the request is an INVITE that contains an Expires header
             field the UAS core inspects this header field. If the
             INVITE has already expired a 487 response is generated.

        2.   If the request has no tag in the To the UAS core checks
             ongoing transactions. If the To, From, Call-ID, CSeq
             exactly match (including tags) those of any request
             received previously, but the branch-ID in the topmost Via
             is different from those received previously, the UAS core
             SHOULD generate a 482 (Loop detected) response and pass it
             to the server transaction.

             The same request that was generated by the UAC has
             arrived to the UAS more than once following different
             paths. The UAS processes the request that was received



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             first and responds with 482 (Loop detected) to the
             rest of them.

             If no match is found, the request does not belong to any
             existing dialog. If the request is an INVITE the UAS core
             follows the procedures described in this section.

        3.   If the request is a mid-dialog request, the method-
             independent processing described in Section 12.2.2 is first
             applied. It might also modify the session; Section 14
             provides details.

        4.   If the request has a tag in the To header field but the
             dialog identifier does not match any of the existing
             dialogs, the UAS may have crashed and restarted, or may
             have received a request for a different (possibly failed)
             UAS. The UAS MAY either accept or reject the request.
             Accepting the request provides robustness, so that dialogs
             can persist even through crashes. UAs wishing to support
             this capability must choose monotonically increasing CSeq
             sequence numbers even across reboots. This is because
             subsequent requests from the crashed-and-rebooted UA
             towards the other UA need to have a CSeq sequence 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 request, because it does
             not wish to recreate the dialog, it MUST respond to the
             request with a 481 (Call/Transaction Does Not exist) status
             code and pass that to the server transaction.

   Processing from here forward assumes that the INVITE is outside of a
   dialog, and is thus for the purposes of establishing a new session.

   The INVITE may contain a session description, in which case the UAS
   is being presented with an offer for that session. It is possible
   that the user is already a participant in that session, even though
   the INVITE is outside of a dialog. This can happen when a user is
   invited to the same multicast conference by multiple other



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   participants. If desired, the UAS 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. If
   the user is already a member of the session and the session
   parameters contained in the session description have not changed, the
   UAS MAY silently accept the INVITE

   The INVITE may not contain a session description at all, in which
   case the UAS is being asked to participate in a session, but the UAC
   has asked that the UAS provide the offer of the session.

   The callee can indicate progress, accept, redirect, or reject the
   invitation. In all of these cases, it formulates a response using the
   procedures described in Section  8.2.7.

13.3.1.1 Progess

   The UAS may not be able to answer the invitation immediately, and
   might choose to indicate some kind of progress to the caller (for
   example, an indication that a phone is ringing). This is accomplished
   with a provisional response between 101 and 199. These provisional
   responses establish early dialogs and therefore follow the procedures
   of Section 12.1.0.1 in addition to those of Section 8.2.7. A UAS MAY
   send as many provisional responses as it likes. Each of these MUST
   indicate the same dialog ID. SIP, however, does not guarantee that
   these provisional responses are reliably delivered to the UAC.

13.3.1.2 The INVITE is redirected

   If the UAS decides to redirect the call, a 3xx response is sent. A
   300 (Multiple Choices), 301 (Moved Permanently) or 302 (Moved
   Temporarily) response SHOULD contain a Contact header field
   containing URIs of new addresses to be tried. The response is passed
   to the INVITE server transaction, which will deal with its
   retransmissions.

13.3.1.3 The INVITE is rejected

   A common scenario occurs when the callee is currently not willing or
   able to take additional calls at this end system. A 486 (Busy Here)
   SHOULD be returned in such scenario. If the UAS knows that no other
   end system will be able to accept this call a 600 (Busy Everywhere)
   response SHOULD be sent instead. However, it is unlikely that a UAS
   will be able to know this in general, and thus this response will not
   usually be used. The response is passed to the INVITE server
   transaction, which will deal with its retransmissions.

13.3.1.4 The INVITE is accepted



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   The UAS core generates a 2xx response. This response establishes a
   dialog, and therefore follows the procedures of Section 12.1.0.1 in
   addition to those of Section 8.2.7.

   A 2xx response to an INVITE SHOULD contain the Allow header field and
   the Supported header field, 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.

   If the INVITE request contained an offer, the 2xx MUST contain an
   answer.  If the INVITE did not contain an offer, the 2xx MUST contain
   an offer.

   Once the response has been constructed it is passed to the INVITE
   server transaction. Note, however, that the INVITE server transaction
   does not retransmit 2xx responses to an INVITE. Therefore, it is
   necessary to pass periodically the response to the server transaction
   until the ACK arrives.  The 2xx response is resubmitted to the server
   transaction with an interval that starts at T1 seconds and doubles
   for each retransmission until it reaches T2 seconds (T1 and T2 are
   defined in Section 17). Response retransmissions cease when an ACK
   request is received with the same dialog ID as the response. This is
   independent of whatever transport protocols are used to send the
   response.


        Since 2xx is retransmitted end-to-end, there may be hops
        between UAS and UAC which are UDP. To ensure reliable
        delivery across these hops, the response is retransmitted
        periodically even if the transport at the UAS is reliable.

   If the server retransmits the 2xx response for 64*T1 seconds without
   receiving an ACK, it considers the dialog completed, the session
   terminated, and therefore it SHOULD send a BYE.

14 Modifying an Existing Session

   A successful INVITE request (see Section 13) establishes both a
   dialog between two user agents and a session (using the offer/answer
   model). Section  12 explains how to modify an existing dialog using a
   refresh request (e.g., changing the route set of the dialog). This
   section describes how to modify the actual session. This modification
   can involve changing addresses or ports, adding a media stream,
   deleting a media stream, and so on. This is accomplished by sending a
   new INVITE request within the same dialog that established the
   session. An INVITE request sent within an existing dialog is known as
   a re-INVITE.



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        Note that a single re-INVITE can modify at the same time
        the dialog and the parameters of the session.

   Either the caller or callee can modify an existing session.

14.1 UAC Behavior

   The same offer-answer model that applies to session descriptions in
   INVITEs (Section  13.2.1) applies to re-INVITEs. As a result, a UAC
   that wants to add a media stream, for example, will create a new
   offer that contains this media stream, and send that in an INVITE
   request to its peer. It is important to note that the full
   description of the session, not just the change, is sent. This
   maintains the idempotency of SIP, supports stateless session
   processing in various elements, and supports failover and recovery
   capabilities. Of course, a UAC MAY send a re-INVITE with no session
   description, in which case the response to the re-INVITE will contain
   the offer.

   If the session description format has the capability for version
   numbers, the offerer SHOULD indicate that the version of the session
   description has changed.

   The To, From, Call-ID, CSeq, and Request-URI of a re-INVITE are set
   following the same rules as for regular requests within an existing
   dialog, described in Section  12.

   Note that, as opposed to initial INVITEs (see Section  13), re-
   INVITEs contain tags in the To header field and are sent using the
   route set for the dialog. Therefore, a single final (2xx or non-2xx)
   response is received for re-INVITEs.

   Note that a UAC MUST NOT initiate a new INVITE transaction within a
   dialog while another transaction (INVITE or non-INVITE) is in
   progress. However, a UA MAY initiate a regular transaction within an
   early dialog - while an INVITE transaction is in progress.

   If a re-INVITE is responded with a non-2xx final response the session
   parameters MUST remain unchanged, as if no re-INVITE had been issued.

   The rules for transmitting a re-INVITE and for generating an ACK  for
   a 2xx response to re-INVITE are the same as for an INVITE (Section
   13.2.1).

14.2 UAS Behavior

   Section  13.3.1 describes the steps to follow in order to distinguish
   incoming re-INVITEs from incoming initial INVITEs. This Section



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   describes the procedures to follow upon reception of a re-INVITE for
   an existing dialog.

   A UAS that receives a second INVITE before it sent the final response
   to a first INVITE with a lower CSeq sequence number on the same
   dialog MUST return a 500 response to the second INVITE and MUST
   include a Retry-After header field with a randomly chosen value of
   between 0 and 10 seconds. Similarly, a UAS the receives an INVITE on
   a dialog while an INVITE it had sent on that dialog is in progress
   MUST return a 500 response to the received INVITE and MUST include a
   Retry-After header field with a randomly chosen value of between 0
   and 10 seconds.

   If a user agent receives a re-INVITE for an existing dialog it MUST
   check any version identifiers in the session description or, if there
   are no version identifiers, the content of the session description to
   see if it has changed. If the 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 a UAS generates a 2xx response and never receives an ACK, it
   SHOULD generate a re-INVITE itself with an offer equal to the last
   session description sent to the peer. The purpose of this is to
   ensure that both caller and callee have a consistent view of the
   session parameters.

   A UAS providing an offer in a 2xx (because the INVITE did not contain
   an offer) MUST offer the same session description as last provided to
   the peer, with the exception of being able to change the IP
   address/port if so desired.

        Under error conditions (e.g., the UAS has crashed and
        restarted) the session description in the 2xx response for
        an empty re-INVITE may be different than the one in use at
        that moment. If the new session description is not
        acceptable for the UAC it SHOULD then send a BYE (after
        ACKing the 2xx response).

15 Terminating a Session

   Terminating a session is done either with the BYE request, or the
   CANCEL request, depending on the state of the dialog. Either caller
   or callee can terminate, and may do so for any reason. Sections 13



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   and 12 document some cases where call termination is normative
   behavior. As a general rule, if a UA decides that the session is to
   be terminated, it MUST follow the procedures here to initiate
   signaling action to convey that.

        Note that both the session and the dialog between both user
        agents will be terminated.

   When a UAC sends an INVITE request to create a session, if a 1xx
   response with a tag in the To field is received, an early dialog is
   created. When a 2xx response is received, the dialog becomes
   established. For either state of the dialog, if the UAC desires to
   terminate the session, the UAC SHOULD follow the procedures described
   in Section 15.1.1 to terminate the session. If the callee for a new
   session wishes to terminate the dialog, it uses the procedures of
   Section 15.1.1, but MUST NOT do so until it has receive an ACK or
   until the server transaction times out.

        This does not mean a user can't hang up right away; it just
        means that the software in their phone needs to maintain
        state for a short while in order to properly clean up.


        OPEN ISSUE #202: Is this the right solution.

   If the UAC desires to end the session before any type of dialog has
   been created, it SHOULD send a CANCEL for the INVITE request that
   requested establishment of the session that is to be terminated. The
   UAC constructs and sends the CANCEL following the procedures
   described in Section 9. 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 2xx
   response to the INVITE "pass on the wire". In this case, the UAC will
   receive a 2xx to the INVITE. It SHOULD then terminate the call by
   following the procedures described in Section 15.1.1.

15.1 Terminating a Dialog with a BYE

15.1.1 UAC Behavior

   A user agent client uses BYE request, sent within a dialog, to
   indicate to the server that it wishes to terminate the session. This
   will also terminate the dialog. A BYE request MAY be issued by either
   caller or callee. A BYE request SHOULD NOT be sent before the
   creation of a dialog (either early or established). In that case the
   UAC SHOULD follow the procedures described in Section 9 instead.

        Proxies ensure that a CANCEL request is routed in the same



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        way as the INVITE was. However, a proxy performing load
        balancing may route a BYE without a Route header field in a
        different way than the INVITE, since both requests have
        different CSeq sequence numbers.

   The To, From, Call-ID, CSeq, and Request-URI of a BYE are set
   following the same rules as for regular requests sent within a
   dialog, described in Section  12.

   Once the BYE is constructed, it creates a new non-INVITE client
   transaction, and passes it the BYE request. The user agent SHOULD
   stop sending media as soon as the BYE request is passed to the client
   transaction.

15.1.2 UAS Behavior

   A UAS core receiving a BYE request checks to see if it matches an
   existing dialog. If the BYE does not match an existing dialog, the
   UAS core SHOULD generate a 481 response and pass that to the server
   transaction.

   A UAS core receiving a BYE request for an existing dialog MUST follow
   the procedures of Section 12.2.2 to process the request. Once done,
   the UAS MUST cease transmitting media streams for the session being
   terminated. The UAS core MUST generate a 2xx response to the BYE, and
   MUST pass that to the server transaction for transmission.

   The UAS MUST still respond to any pending requests received for that
   dialog, (which can only be an INVITE). It is RECOMMENDED that a 487
   (Request Terminated) response is generated to those pending requests.

16 Proxy Behavior

16.1 Overview

   SIP proxies are elements that route SIP requests to user agent
   servers and SIP responses to user agent clients. A request may
   traverse several proxies on its way to a UAS. Each will make routing
   decisions, modifying the request before forwarding it to the next
   element.  Responses will route through the same set of proxies
   traversed by the request in the reverse order.

   It is important to note that being a proxy is a logical role for a
   SIP element.  When a request arrives, an element that can play the
   role of a proxy must first decide if it needs to respond to the
   request on its own. For instance, the request could be malformed or
   the element may need credentials from the client before acting as a
   proxy.  The element MAY respond with any appropriate error code.



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   When responding directly to a request, the element is playing the
   role of a UAS and MUST behave as described in Section 8.2.

   A proxy can operate in either a stateful or stateless mode for each
   new request.

   When stateless, a proxy acts as a simple forwarding element.  It
   forwards each request downstream to a single element determined by
   making a routing decision based on the request. It simply forwards
   every response it receives upstream. A stateless proxy discards
   information about a message once it has been forwarded.

   On the other hand, a stateful proxy remembers information
   (specifically, transaction state) about each incoming request and any
   requests it sends as a result of processing the incoming request. It
   uses this information to affect the processing of future messages
   associated with that request. A stateful proxy MAY chose to "fork" a
   request, routing it to multiple destinations. Any request that is
   forwarded to more than one location MUST be handled statefully. Any
   request processed using TCP (or any other mechanism that is
   inherently stateful), MUST be handled statefully.

   Much of the processing involved when acting statelessly or statefully
   for a request is identical. The next several subsections are written
   from the point of view of a stateful proxy. The last section calls
   out those places where a stateless proxy behaves differently.

16.2 Stateful Proxy

   When stateful, a proxy is purely a SIP transaction processing engine.
   Its behavior is modeled here in terms of the Server and Client
   Transactions defined in Section 17.  A stateful proxy has a server
   transaction associated with one or more client transactions by a
   higher layer proxy processing component (see figure 3), known as a
   proxy core. An incoming request is processed by a server transaction.
   Requests from the server transaction are passed to a proxy core.  The
   proxy core determines where to route the request, choosing one or
   more next-hop locations.  An outgoing request for each next-hop
   location is processed by its own associated client transaction. The
   proxy core collects the responses from the client transactions and
   uses them to send responses to the server transaction.

   A stateful proxy creates a new server transaction for each new
   request received. Any retransmissions of the request will then be
   handled by that server transaction per Section 17.

   Note that this is a model of proxy behavior, not of software. An
   implementation is free to take any approach that replicates the



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   external behavior this model defines.





                                                      +--+
                                                      | C|
                                                      | l|
                                                      | i|
                                                      | e|
                                                      | n|
                                                      +-t+



                     +----------------------+          +---+
                     |                      |          | C |
           +--+      |                      |          | l |
           |  |      |                      |          | i |
           |  |      |     Proxy            |          | e |
           | S|      |     "higher" layer   |          | n |
           | e|      |                      |          +-t-+
           | r|      |                      |
           | v|      |                      |
           | e|      |                      |
           | r|      |                      |           +---+
           |  |      |                      |           |   |
           |  |      |                      |           | C |
           |  |      |                      |           | l |
           +--+      +----------------------+           | i |
                                                        | e |
                                                        | n |
                                                        | t |
                                                        |   |
                                                        +---+



   Figure 3: Stateful Proxy Model



   For all new requests, including any with unknown methods, an element
   intending to proxy the request MUST:

        1.   Validate the request (Section 16.3)




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        2.   Make a routing decision (Section 16.4)

        3.   Forward the request to each chosen destination (Section
             16.5)

        4.   Process all responses (Section 16.6)

16.3 Request Validation

   Before an element can proxy a request, it MUST verify the message's
   validity. A valid message must pass the following checks:

        1.   Reasonable Syntax

        2.   Max-Forwards

        3.   Loop Detection

        4.   Proxy-Require

        5.   Proxy-Authorization

   If any of these checks fail, the element MUST behave as a user agent
   server (see Section 8.2) and respond with an error code.

        1.   Reasonable Syntax check

             The request MUST be well-formed enough to be handled with a
             server transaction. Any components involved in the
             remainder of these Request Validation steps or the Request
             Processing section MUST be well-formed. Any other
             components, well-formed or not, SHOULD be ignored. For
             instance, an element SHOULD NOT reject a request because of
             a malformed Date header field.

             This protocol is designed to be extended. Future extensions
             may define new methods and header fields at any time.  An
             element MUST NOT refuse to proxy a request because it
             contains a method or header field it does not know about.

        2.   Max-Forwards check

             The Max-Forwards header (Section 22.22) is used to limit
             the number of elements a SIP request can traverse.

             If the request does not contain a Max-Forwards header
             field, this check is passed.




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             If the request contains a Max-Forwards header field with a
             field value greater than zero, the check is passed.

             If the request contains a Max-Forwards header field with a
             field value of zero (0), the element MUST NOT forward the
             request. If the request was for OPTIONS, the element MAY
             act as the final recipient and respond per Section 11.
             Otherwise, the element MUST return a 483 (Too many hops)
             response.

        3.   Loop Detection check

             An element MUST check for forwarding loops before
             forwarding a request.  If the request contains a Via header
             field value with A sent-by value that equals a value placed
             into previous requests by the proxy, the request has been
             forwarded by this element before. The request has either
             looped or is legitimately spiraling through the element.
             To determine if the request has looped, the element MUST
             perform the branch parameter calculation described in
             Section 3 on this message and compare it to the parameter
             received in that Via field value.  If the parameters match,
             the request has looped. If they differ, the request is
             spiraling, and processing continues. If a loop is detected,
             the element MUST return a 482 (Loop Detected) response.

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

        4.   Proxy-Require check

             Future extensions to this protocol may introduce features
             that require special handling by proxies. Endpoints will
             include a Proxy-Require header in requests that use these
             features, telling the proxy it should not process the
             request unless the feature is understood.

             If the request contains a Proxy-Require header (Section
             22.28) with one or more option-tags this element does not
             understand, the element MUST return a 420 (Bad Extension)
             response.  The response MUST include an Unsupported
             (Section 22.38) header field listing those option-tags the
             element did not understand.

        5.   Proxy-Authorization check

             If an element requires credentials before forwarding a
             request, the request MUST be inspected as described in



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             Section 20.2.3.  That section also defines what the element
             must do if the inspection fails.

16.4 Making a Routing Decision

   At this point, the proxy must decide where to forward the request.
   This can be modeled as computing a set of destinations for the
   request.  This set will either be predetermined by the contents of
   the request or will be obtained from an abstract location service.
   Each destination is represented as a URI and an optional IP address,
   port and transport. This combination is referred to as a "next-hop
   location".

   First, the proxy core checks the received request for Route headers.
   If any Route header fields are present in the request, the element
   MUST use the URL (including all of its parameters) from the topmost
   Route header field as only next hop URI in the destination set, with
   no IP address, port and transport set for that next hop. The
   destination set is complete, containing only this URL, and the proxy
   MUST proceed to the Request Processing of Section 16.5.

   The Route mechanism is used to control the path a request takes
   through SIP elements, much like strict IP source routing. The UAC
   will insert Route header fields (see Section 12), usually based on
   information provided by proxies through Record-Route header fields
   (see Section 6).

   Assuming there were no Route headers in the received request, the
   proxy checks the Request-URI of the received request. If it has an
   maddr parameter, and that parameter does not indicate an interface
   the proxy is listening on, the Request-URI MUST be placed into the
   destination set as the only next hop URI, with no IP address, port
   and transport set for that next hop, and the proxy MUST proceed to
   Section 16.5. If the maddr parameter was present, but did indicate an
   interface the proxy is listening on, the proxy MUST strip the maddr
   and continue processing as if no maddr were present.


        OPEN ISSUE #213: Do we strip just the maddr, or the port
        and transport as well?


        OPEN ISSUE #218: Are we really sure this ordering of
        precedence of Route, maddr, and domain is correct?? It is
        not yet clear. This needs resolution asap finally, since it
        affects things like loose source routing, outbound proxy
        processing at a UA, and so on.




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   If the domain of the Request-URI indicates a domain this element is
   not responsible for, it SHOULD set the next hop URI to the Request-
   URI, and leave the IP address, port and transport of the next hop
   empty. That next hops MUST be placed into the destination set as the
   only next hop, and the element MUST proceed to the task of Request
   Processing (Section 16.5.


        There are many circumstances in which a proxy might receive
        a request for a domain it is not responsible for. A
        firewall proxy handling outgoing calls (the way HTTP
        proxies handle outgoing requests) is an example of where
        this is likely to occur.

   If the destination set for the request has not been predetermined as
   described above, this implies that the element is responsible for the
   domain in the Request-URI, and the element MAY use whatever mechanism
   it desires to determine where to send the request. Any of these
   mechanisms can be modeled as accessing an abstract Location Service.
   This may consist of obtaining information from a location service
   created by a SIP Registrar, reading a database, consulting a presence
   server, utilizing other protocols, or simply performing an
   algorithmic substitution on the Request-URI. The output of these
   mechanisms is used to construct the destination set.

   Any information in or about the request or the current environment of
   the element MAY be used in the construction of the destination set.
   For instance, different sets may be constructed depending contents or
   presence of header fields and bodies, the time of day of the
   request's arrival, the interface on which the request arrived,
   failure of previous requests, or even the element's current level of
   utilization.

   As potential destinations are located through these services, their
   next hops are added to the destination set. Next-hop locations may
   only be placed in the destination set once. If a next-hop location is
   already present in the set (based on the definition of equality for
   the URI type and equality of the optional parameters), it MUST NOT be
   added again.

   A proxy MAY continue to add destinations to the set after beginning
   Request Processing. It MAY use any information obtained during that
   processing to determine new locations. For instance, a proxy may
   choose to incorporate contacts obtained in a redirect response (3xx
   class) into the destination set. If a proxy uses a dynamic source of
   information while building the destination set (for instance, if it
   consults a SIP Registrar), it SHOULD monitor that source for the
   duration of processing the request. New locations SHOULD be added to



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   the destination set as they become available. As above, any given URI
   MUST NOT be added to the set more than once.


        Allowing a URI to be added to the set only once reduces
        unnecessary network traffic, and in the case of
        incorporating contacts from redirect requests prevents
        infinite recursion.

   An example trivial location service is achieved by configuring an
   element with a default outbound destination. All requests are
   forwarded to this location. The Request-URI of the request is placed
   in the destination set with the optional next-hop IP address, port
   and transport parameters set to the default outbound destination. The
   destination set is complete, containing only this URI, and the
   element proceeds to the task of Request Processing.

   If the Request-URI indicates a resource at this proxy that does not
   exist, the proxy MUST return a 404 (Not Found) response.

   If the destination set remains empty after applying all of the above,
   the proxy MUST return an error response, which SHOULD be the 480
   (Temporarily Unavailable) response.

16.5 Request Processing

   As soon as the destination set is non-empty, a proxy MAY begin
   forwarding the request.  A stateful proxy MAY process the set in any
   order. It MAY process multiple destinations serially, allowing each
   client transaction to complete before starting the next. It MAY start
   client transactions with every destination in parallel. It also MAY
   arbitrarily divide the set into groups, processing the groups
   serially and processing the destinations in each group in parallel.

   A common ordering mechanism is to use the qvalue parameter of
   destinations obtained from Contact header fields (see Section 22.10).
   Destinations are processed from highest qvalue to lowest.
   Destinations with equal qvalues may be processed in parallel.

   A stateful proxy must have a mechanism to maintain the destination
   set as responses are received and associate the responses to each
   forwarded request with the original request. For the purposes of this
   model, this mechanism is a "response context" created by the proxy
   layer before forwarding the first request.

   For each destination, the proxy forwards the request following these
   steps:




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        1.   Make a copy of the received request

        2.   Update the Request-URI

        3.   Add a Via header field value

        4.   Update the Max-Forwards field if present

        5.   Update the Route header field if present

        6.   Optionally add a Record-route header field value

        7.   Optionally add additional headers

        8.   send the new request

   Each of these steps is detailed below:

        1.   Copy request

             The proxy starts with a copy of the received request.  The
             copy MUST initially contain all of the header fields from
             the received request. Only those fields detailed in the
             processing described below may be removed.  The copy SHOULD
             maintain the ordering of the header fields as in the
             received request. The proxy MUST NOT reorder field values
             with a common field name (See Section 7.3.1).


             An actual implementation need not perform a copy; the
             primary requirement is that the processing of each
             next hop begin with the same request.

        2.   Request-URI

             The Request-URI in the copy's start line MUST be replaced
             with the URI for this destination. If the URI contains any
             parameters not allowed in a Request-URI, they MUST be
             removed.

             This is the essence of a proxy's role. This is the
             mechanism through which a proxy routes a request toward its
             destination.

        3.   Via

             The proxy MUST insert a Via header field into the copy
             before the existing Via header fields. The Via header



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             maddr, ttl, and sent-by components will be set when the
             request is processed by the transport layer (Section 19).
             The Via headers ensure that responses will follow the same
             set of elements that the request traversed.

             The proxy MUST include a "branch" parameter (Section 22.40)
             in the Via header. When the path of a request through one
             or more forking proxies is graphed, the result is a tree.
             The branch parameter identifies the "branch" each request
             was forwarded on.  The branch parameter value MUST be
             unique for each client transaction to which the request is
             forwarded.  The precise format of the branch.  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. The first part is used to
             detect loops and distinguish loops from spirals. The second
             is used to match responses to requests.

             Loop detection is performed by verifying that those fields
             having an impact on the routing decision have not changed.
             The value placed in the this part of the branch parameter
             SHOULD reflect all of those fields (which include any
             Proxy-Require and Proxy-Authorization headers). 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 (Section 3).  A common way to create this value
             is to compute 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, in addition to any Proxy-Require and Proxy-
             Authorization fields that may be present.  The algorithm
             used to compute the hash is implementation-dependent, but
             MD5 [23], expressed in hexadecimal, is a reasonable choice.
             (Note that base64 is not permissible for a token.)

             In order to correctly match responses to requests (Section
             17.1.3), the value SHOULD also contain a part that is a
             globally unique function of of the branch on which this
             request will be forwarded.  One example is a hash of a
             sequence number, local IP address and request-URI of the
             request

             For example: 7a83e5750418bce23d5106b4c06cc632.1


             The "branch" parameter MUST depend on all information
             used for routing decisions, including the incoming



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             request-URI and any header values affecting the
             routing choices. This is necessary to distinguish
             looped requests from requests whose routing parameters
             have changed before returning to this server.

             Note that the request method MUST NOT be included in the
             calculation of the branch parameter. In particular, CANCEL
             and ACK requests MUST have the same branch value as the
             corresponding request they cancel or acknowledge. The
             branch parameter is used in correlating those requests at
             server handling them (see Section 17.2.3 and 9.2).

        4.   Max-Forwards

             If the copy contains a Max-Forwards header field, the proxy
             must decrement its value by one (1).

        5.   Route

             If the copy contains a Route header field, the proxy must
             remove the first (topmost) value. Note that this value was
             placed in the destination set and then into the Request-URI
             of this copy in previous steps.

        6.   Record-Route

             If this proxy wishes to request to remain on the path of
             future requests in this dialog, it MUST insert a Record-
             Route header value (Section refsec:record-route) into the
             copy before any existing Record-Route header values.  See
             Section 12 for details on whether this request will be
             honored. Each proxy in the path of a request makes this
             request independently the presence of a Record-Route header
             does not obligate this proxy to add a value.

             If the request is honored, the information the proxy places
             in the Record-Route header value will be used at the
             endpoints to construct Route headers. As shown in the
             processing steps above, Route headers determine forwarding
             destinations much like strict IP source routing.

             The URL placed in the Record-Route header value MUST be a
             SIP URL. This URL MAY be different for each destination the
             request is forwarded to. The URL SHOULD NOT contain the
             transport parameter unless the proxy has knowledge (such as
             in a private network) that the next downstream element that
             will be in the path of subsequent requests supports that
             transport.



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             The URL this proxy provides will be used by some other
             element to make a routing decision. This proxy, in
             general, has no way to know what the capabilities of
             that element are, so it must restrict itself to the
             mandatory elements of a SIP implementation: SIP URLs
             and UDP transports.

             The URL placed in the Record-Route header value MUST
             resolve to this element when the server location procedures
             of Section 24 are applied to it. This ensures subsequent
             requests are routed back to this element.

             The URL placed in the Record-Route header value SHOULD be
             such that if a subsequent 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
             elements, including the originating client, traversed by
             the original request.  This improves robustness, ensuring
             that the Request-URI contains enough information to forward
             subsequent requests to a reasonable destination even in the
             absence of Route headers.

             The URL placed in the Record-Route header value MUST vary
             with the Request-URI in the received request.  A request
             may legitimately pass through this proxy more than once on
             the way to its final destination (this is called a
             spiraling request). The Request-URI will be different each
             time the request passes through. If this proxy places the
             same URL in the Record-Route header field each time,
             subsequent requests will be rejected as looped requests. It
             is insufficient to simply copy the Request-URI from each
             request into the Record-Route header. Some modification,
             such as adding an maddr parameter, is necessary.

             URLs satisfying the above paragraphs can be constructed in
             many ways.  One way is to use a URL that 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 URL in the Record-Route be
             different for each distinct Request-URI). A call stateful
             proxy could use a URL of the form sip:proxy.example.com and
             use information from the stored call state to meet the
             requirements.

             The proxy MAY include Record-Route header parameters in the
             value it provides. These will be returned in some responses



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             to the request (200 responses to INVITE for example) and
             may be useful for pushing state into the message.

             The Record-Route process is designed to work for any SIP
             request that initiates a dialog. The only such request in
             this specification is INVITE. Extensions to the protocol
             MAY define others, and the mechanisms described here will
             apply.  The request that initiates a dialog and all
             refreshes (re-INVITE for example) MUST have Record-Route
             header values added to them if the proxy wishes to remain
             in the request path. This means a proxy will often need to
             record-route requests that contain Route headers.  Section
             12 describes how this will affect a dialog.


             Including Record-Route even when Route headers already
             exist in a request improves robustness in the presence
             of a preloaded Route header field and recovery from
             endpoint failure.

             If a proxy needs to be in the path of any type of dialog
             (such as one straddling a firewall), it SHOULD add a
             Record-Route header value to every request with a method it
             doesn't understand.

             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 in a dialog. It
             is RECOMMENDED that proxies do not automatically record
             route. They should do so only if specifically required.

        7.   Adding Additional Headers

             The proxy MAY add any other appropriate headers to the copy
             at this point.

        8.   Forward Request

             A stateful proxy creates a new client transaction for this
             request as described in Section 17.1.  If the next-hop
             location used in building this request contains the
             optional addressing parameters, the transaction is
             instructed to send the request based on those parameters.
             Otherwise, the proxy uses the procedures of Section 24 to
             compute an ordered set of addresses from the Request-URI,
             and as described there, attempts to contact the first one



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             by instructing the client transaction to send the request
             there. If this fails, the stateful proxy continues down the
             list. Each attempt is a new client transaction, and
             therefore represents a new branch, so that the processing
             described above for each branch would need to be repeated.
             This results in a requirement to use a different branch ID
             parameter for each attempt.

16.6 Response Processing

   When a response is received by an element, it first tries to locate a
   client transaction (Section 17.1.3) matching the response. If none is
   found, the element MUST process the response (even if it is an
   informational response) as a stateless proxy (described below). If a
   match is found, the response is handed to the client transaction.


        Forwarding responses for which a client transaction (or
        more generally any knowledge of having sent an associated
        request) is not found improves robustness. In particular,
        it ensures that "late" 2xx class responses to INVITE
        requests are forwarded properly.

   As client transactions pass responses to the proxy layer, the
   following processing MUST take place:

        1.   Find the appropriate response context

        2.   Remove the topmost Via

        3.   Add the response to the response context

        4.   Check to see if this response should be forwarded

   The following processing MUST be performed on each response that is
   forwarded. Note that more than one response to each request will
   likely be forwarded - each provisional and one final at the least.

        1.   Aggregate authorization header fields if necessary

        2.   Forward the response

        3.   Generate any necessary CANCEL requests

   If no final response has been forwarded after every client
   transaction associated with the response context has been terminated,
   the proxy must choose and forward the "best" response from those it
   has seen so far.



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   Each of the above steps are detailed below:

        1.   Find Context

             The proxy locates the "response context" it created before
             forwarding the original request using the key described in
             Section 16.5. The remaining processing steps take place in
             this context.

        2.   Via

             The proxy removes the topmost Via field value from the
             response.  The address in this value necessarily matches
             the proxy since the response matched a client transaction
             above. The branch parameter from this value can be used to
             determine which branch the response corresponds to.

             If no Via field values remain in the response, the response
             was meant for this element and MUST NOT be forwarded. The
             remainder of the processing described in this section is
             not performed on this message.  This will happen, for
             instance, when the element generates CANCEL requests as
             described in Section sec:proxy-response-processing-cancel.

        3.   Add response to context

             Final responses received are stored in the response context
             until a final response is generated on the server
             transaction associated with this context. The response may
             a candidate for the best final response to be returned on
             that server transaction. Information from this response may
             be needed in forming the best response even if this
             response is not chosen.

             If the proxy chooses to recurse on a 3xx class response, it
             MUST NOT add the response to the response context

        4.   Check response for forwarding

             Until a final response has been sent on the server
             transaction, the following responses MUST be forwarded
             immediately:

             - Any provisional response other than 100 Trying

             - Any 2xx response

             If a 6xx response is received, it is not immediately



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             forwarded, but the stateful proxy SHOULD cancel all pending
             transactions as described in Section 9.


             This is a change from RFC2543, which mandated that the
             6xx be forwarded immediately. The problem with this is
             that it is possible for a 2xx to arrive on another
             branch, in which case the proxy would have to forward
             that in the case of an INVITE transaction. The result
             is that the UAC could receive a 6xx followed by a 2xx,
             which should never be allowed to happen. So, instead,
             upon receiving a 6xx, a proxy will CANCEL, which will
             generally result in 487s to all outstanding client
             transactions, and then at that point the 6xx is
             forwarded upstream.

             After a final response has been sent on the server
             transaction, the following responses MUST be forwarded
             immediately:

             - Any 2xx class response to an INVITE request

             A stateful proxy MUST NOT immediately forward any other
             responses. In particular, a stateful proxy MUST NOT forward
             any 100 Trying response. Those responses that are
             candidates for forwarding later as the "best" response have
             been gathered as described in step "Add Response to
             Context".

             Any response chosen for immediate forwarding MUST be
             processed as described in steps "Aggregate authorization
             headers" through "Record-Route".

        5.   Choosing the best response

             A stateful proxy MUST send a final response to a response
             context's server transaction if no final responses have
             been immediately forwarded by the above rules and all
             client transactions in this response context have been
             terminated.

             The stateful proxy MUST choose the "best" final response
             among those received and stored in the response context.

             If there are no final responses in the context, the proxy
             MUST send a 408 (Request Timeout) response to the server
             transaction.




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             Otherwise, the proxy MUST forward one of the responses from
             the lowest response class stored in the response context.
             The proxy MAY select any response within that lowest class.
             The proxy SHOULD give preference to responses that provide
             information affecting resubmission of this request, such as
             401, 407, 415, 420, and 484.

             A proxy which receives a 503 response SHOULD NOT forward it
             upstream unless it can determine that any subsequent
             requests it might proxy will also generate a 503. In other
             words, forwarding a 503 means that the proxy knows it
             cannot service any requests, not just the one for the
             Request-URI in the request which generated the 503.

             The forwarded response MUST be processed as described in
             steps "Aggregate authorization headers" through "Record-
             Route".

             For example, if a proxy forwarded a request to 4 locations,
             and received 503, 407, 501, and 404 responses, it may
             choose to forward the 407 response.

             The tag in the To header field serves to distinguish
             responses at the UAC. If the forwarded response did not
             have one, it MUST NOT be inserted into the response by the
             proxy.

        6.   Aggregate authorization headers

             If the selected response is a 401 or 407, the proxy MUST
             collect any WWW-Authenticate and Proxy-Authenticate header
             fields from all other 401 and 407 responses received so for
             in this response context and add them to this response
             before forwarding.

             This is necessary because any or all of the destinations
             the request was forwarded to may have requested
             credentials. The client must receive all of those
             challenges and supply credentials for each of them when it
             retries the request. Motivation for this behavior is
             provided in Section 20.

        7.   Record-Route

             If the selected response contains a Record-Route header
             field value originally provided by this proxy, the proxy
             MAY chose to rewrite the value before forwarding the
             response.  This allows the proxy to provide different URLs



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             for itself to the next upstream and downstream elements. A
             proxy may choose to use this mechanism for any reason. For
             instance, it is useful for multi-homed hosts.

             The new URL provided by the proxy MUST satisfy the same
             constraints on URLs placed in Record-Route header fields in
             requests (see Section 6) with the following modifications:

             The URL SHOULD NOT contain the transport parameter unless
             the proxy has knowledge that the next upstream (as opposed
             to downstream) element that will be in the path of
             subsequent requests supports that transport.

             The URL placed in the Record-Route header value SHOULD be
             such that if a subsequent 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 element
             (as opposed to some previous element) as the originally
             forwarded request.

             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.

        8.   Forward response

             After performing the processing described in steps
             "Aggregate authorization headers" through "Record-Route",



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             the proxy may perform any feature specific manipulations on
             the selected response.  Unless otherwise specified, the
             proxy MUST NOT remove the message body or any header values
             other than the Via header value discussed in Section
             refsec:proxy-response-processing-via.  The proxy MUST pass
             the response to the server transaction associated with the
             response context. This will result in the response being
             sent to the location now indicated in the topmost Via field
             value. If the server transaction is no longer available to
             handle the transmission, the element MUST forward the
             response statelessly by sending it to the server transport.

             Even after forwarding a final response, the proxy MUST
             maintain the response context until all of its associated
             transactions have been terminated.

        9.   Generate CANCELs

             OPEN ISSUE #7: If CANCEL is restricted to INVITE only, this
             behavior must restrict itself to INVITE requests.

             OPEN ISSUE #122: The MUST below reflects list discussion,
             but the question of how strong this requirement should be
             was not formally closed.

             If the forwarded response was a final response, the proxy
             MUST generate a CANCEL request for all pending client
             transactions associated with this response context. A proxy
             SHOULD also generate a CANCEL request for all pending
             client transactions associated with this response context
             when it receives a 6xx response. A pending client
             transaction is one that has received a provisional
             response, but no final response and has not had an
             associated CANCEL generated for it. Generating CANCEL
             requests is described in Section 9.1.

16.7 Handling transport errors

   If the transport layer notifies a proxy of an error when it tries to
   forward a request (see Section 19.4), the proxy MUST behave as if the
   forwarded request received a 400 response.

   If the proxy is notified of an error when forwarding a response, it
   drops the response. The proxy SHOULD NOT cancel any outstanding
   client transactions associated with this response context due to this
   notification.





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        If a proxy cancels its outstanding client transactions, a
        single malicious or misbehaving client can cause all
        transactions to fail through its Via header field.

16.8 CANCEL Processing

   A stateful proxy may generate a CANCEL to any other request it has
   generated at any time. For instance, it may choose to generate
   CANCELs based on having a transaction exceed the time specified in
   the Expire header of certain requests, or as a result of any logic it
   applies while forwarding requests. A proxy MUST cancel any pending
   client transactions associated with a response context when it
   receives a matching CANCEL request.


        OPEN ISSUE #185: Should generating CANCEL at a proxy based
        on Expires in INVITE be deprecated?

   While a CANCEL request is handled in a stateful proxy by its own
   server transaction, a new response context is not created for it.
   Instead, the proxy layer searches its existing response contexts for
   the server transaction handling the request associated with this
   CANCEL.  If a matching response context is found, the element MUST
   immediately return a 200 OK response to the CANCEL request. In this
   case, the element is acting as a user agent server as defined in
   Section 8.2.  Furthermore, the element MUST generate CANCEL requests
   for all pending client transactions in the context as described in
   Section 9.

   If a response context is not found, the element does not have any
   knowledge of the request to apply the CANCEL to. It MUST forward the
   CANCEL request statelessly (it may have statelessly forwarded the
   associated request previously).

16.9 Stateless proxy

   When acting statelessly, a proxy is a simple message forwarder.  Much
   of the processing performed when acting statelessly is the same as
   when behaving statefully. The differences are detailed here.

   A stateless proxy does not have any notion of a transaction, or of
   the response context used to describe stateful proxy behavior.
   Instead, the stateless proxy takes messages, both requests and
   responses, directly from the transport layer (See section 19).  As a
   result, stateless proxies do not retransmit messages on their own.
   They do, however, forward all retransmission they receive (they do
   not have the ability to distinguish a retransmission from the
   original message). Furthermore, when handling a request statelessly,



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   an element MUST NOT generate its own 100 Trying (or any other
   provisional) response.

   A stateless proxy must validate a request as described in Section
   16.3

   A stateless proxy must make a routing decision as described in
   Section 16.4 with the following exception:

        o A stateless proxy MUST choose one and only one destination
          from the destination set. This choice MUST only rely on fields
          in the message and time-invariant properties of the server. In
          particular, a retransmitted request MUST be forwarded to the
          same destination each time it is processed. Furthermore,
          CANCEL and non-Routed ACK requests MUST generate the same
          choice as their associated INVITE.

   A stateless proxy must process the request before forwarding as
   described in Section 16.5 with the following exceptions:

        o The branch parameter on the inserted Via header field MUST be
          the same each time a retransmitted request is forwarded. Thus
          for a stateless proxy, the branch parameter calculation MUST
          only depend on message parameters affecting the routing of the
          request which are invariant on retransmission.

        o The request is sent directly to the transport layer instead of
          through a client transaction. If the next-hop destination
          parameters don't provide an explicit destination, the element
          applies the procedures of Section 24 to the Request-URI to
          determine where to send the request.

   Stateless proxies MUST NOT perform special processing for CANCEL
   requests.  They are processed by the above rules as any other
   requests.

   Response processing as described in Section 16.6 does not apply to a
   proxy behaving statelessly. When a response arrives at a stateless
   proxy, the proxy inspects the address in the first (topmost) Via
   header value.  If that address matches the proxy, the proxy MUST
   remove that value from the response and forward the result to the
   location indicated in the next Via header value. Unless specified
   otherwise, the proxy MUST NOT remove any other header values or the
   message body. If the address does not match the proxy, the message
   MUST be silently discarded.

17 Transactions




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   SIP is fundamentally a transactional protocol. This means that
   interactions between components take place in a series of independent
   message exchanges. Specifically, a SIP transaction consists of a
   single request, and any responses to that request (which include zero
   or more provisional responses and one or more final responses). In
   the case of a transaction where the request was an INVITE (known as
   an INVITE transaction), the transaction also includes the ACK only if
   the final response was not a 2xx response. If the response was a 2xx,
   the ACK is not considered part of the transaction.

        The reason for this separation is rooted in the importance
        of delivering all 200 OK responses to an INVITE to the UAC.
        To deliver them all to the UAC, the UAS alone takes
        responsibility for retransmitting them, and the UAC alone
        takes responsibility for acknowledging them with ACK. Since
        this ACK is retransmitted only by the UAC, it is
        effectively considered its own transaction.

   Transactions have a client side and a server side. The client side is
   known as a client transaction, and the server side, as a server
   transaction. The client transaction sends the request, and the server
   transaction sends the response. The client and server transactions
   are logical functions that are embedded in any number of elements.
   Specifically, they exist within user agents and stateful proxy
   servers. Consider the example of Section 4. In this example, the UAC
   executes the client transaction, and its outbound proxy executes the
   server transaction. The outbound proxy also executes a client
   transaction, which sends the request to a server transaction in the
   inbound proxy. That proxy also executes a client transaction, which
   in turn, sends the request to a server transaction in the UAS. This
   is shown pictorially in Figure 4.


   A stateless proxy does not contain a client or server transaction.
   The transaction exists between the UA or stateful proxy on one side
   of the stateless proxy, and the UA or stateful proxy on the other
   side. As far as SIP transactions are concerned, stateless proxies are
   effectively transparent. The purpose of the client transaction is to
   receive a request from the element the client is embedded in (call
   this element the "Transaction User" or TU; it can be a UA or a
   stateful proxy), and reliably deliver the request to that server
   transaction. The client transaction is also responsible for receiving
   responses, and delivering them to the TU, filtering out any
   retransmissions or disallowed responses (such as a response to ACK).
   In the case of an INVITE transaction, that includes generation of the
   ACK request for any final response excepting a 2xx response.

   Similarly, the purpose of the server transaction is to receive



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 +---------+        +---------+        +---------+        +---------+
 |      +-+|Request |+-+   +-+|Request |+-+   +-+|Request |+-+      |
 |      |C||------->||S|   |C||------->||S|   |C||------->||S|      |
 |      |l||        ||e|   |l||        ||e|   |l||        ||e|      |
 |      |i||        ||r|   |i||        ||r|   |i||        ||r|      |
 |      |e||        ||v|   |e||        ||v|   |e||        ||v|      |
 |      |n||        ||e|   |n||        ||e|   |n||        ||e|      |
 |      |t||        ||r|   |t||        ||r|   |t||        ||r|      |
 |      | ||        || |   | ||        || |   | ||        || |      |
 |      |T||        ||T|   |T||        ||T|   |T||        ||T|      |
 |      |r||        ||r|   |r||        ||r|   |r||        ||r|      |
 |      |a||        ||a|   |a||        ||a|   |a||        ||a|      |
 |      |n||        ||n|   |n||        ||n|   |n||        ||n|      |
 |      |s||Response||s|   |s||Response||s|   |s||Response||s|      |
 |      +-+|<-------|+-+   +-+|<-------|+-+   +-+|<-------|+-+      |
 +---------+        +---------+        +---------+        +---------+
    UAC               Outbound           Inbound              UAS
                      Proxy               Proxy







   Figure 4: Transaction relationships


   requests from the transport layer, and deliver them to the TU. The
   server transaction filters any request retransmissions from the
   network. The server transaction accepts responses from the TU, and
   delivers them to the transport layer for transmission over the
   network. In the case of an INVITE transaction, it absorbs the ACK
   request for any final response excepting a 2xx response.

   The 2xx response, and the ACK for it, have special treatment. This
   response is retransmitted only by a UAS, and its ACK generated only
   by the UAC. This end-to-end treatment is needed so that a caller
   knows the entire set of users that have accepted the call. Because of
   this special handling, retransmissions of the 2xx response are
   handled by the UA core, not the transaction layer. Similarly,
   generation of the ACK for the 2xx is handled by the UA core. Each
   proxy along the path merely forwards each 2xx response to INVITE, and
   its corresponding ACK.



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17.1 Client transaction

   The client transaction provides its functionality through the
   maintenance of a state machine.

   The TU communicates with the client transaction through a simple
   interface. When the TU wishes to initiate a new transaction, it
   creates a client transaction, and passes it the SIP request to send,
   a value for timer C (described below), and an IP address, port, and
   transport to send it to. The client transaction begins execution of
   its state machine. Valid responses are past up to the TU from the
   client transaction.

   There are two types of client transaction state machines, depending
   on the method the request passed by the TU. One handles client
   transactions for INVITE request. This type of machine is referred to
   as an INVITE client transaction. Another type handles client
   transactions for all requests except INVITE and ACK. This is referred
   to as a non-INVITE client transaction. There is no client transaction
   for ACK. If the TU wishes to send an ACK, it passes one directly to
   the transport layer for transmission.

   The INVITE transaction is different from those of other methods
   because of its extended duration. Normally, human input is required
   in order to respond to an INVITE. The long delays expected for
   sending a response argue for a three way handshake. Requests of other
   methods, on the other hand, are expected to completely rapidly. In
   fact, because of its reliance on just a two way handshake, TUs SHOULD
   respond immediately to non-INVITE requests. Protocol extensions which
   require longer durations for generation of a response (such as a new
   method that does require human interaction) SHOULD instead use two
   transactions - one to send the request, and another in the reverse
   direction to convey the result of the request.

17.1.1 INVITE Client Transaction

17.1.1.1 Overview of INVITE Transaction

   The INVITE transaction consists of a three-way handshake. The client
   transaction sends an INVITE, the server transaction sends responses,
   and the client transaction sends an ACK. For unreliable transports
   (such as UDP), the client transaction will retransmit requests at an
   interval that starts at T1 seconds and doubles after every
   retransmission. The request is not retransmitted over reliable
   transports. After receiving a 1xx response, any retransmissions cease
   altogether, and the client waits for further responses. The server
   transaction can send additional 1xx responses, which are not
   transmitted reliably. Eventually, the server transaction decides to



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   send a final response. For unreliable transports, that response is
   retransmitted periodically, and for reliable transports, its sent
   once. For each final response that is received at the client
   transaction, the client transaction sends an ACK, the purpose of
   which is to quench retransmissions of the response.

17.1.1.2 Formal Description


   The state machine for the INVITE client transaction is shown in
   Figure 5. The initial state, "calling", MUST be entered when the TU
   initiates a new client transaction with an INVITE request. The client
   transaction MUST pass the request to the transport layer for
   transmission (see Section 19). If an unreliable transport is being
   used, the client transaction SHOULD start timer A with a value of T1,
   and SHOULD NOT start timer A when a reliable transport is being used
   (Timer A controls request retransmissions). For any transport, the
   client transaction MUST start timer B with a value of 64*T1 seconds
   (Timer B controls transaction timeouts).

   When timer A fires, the client transaction SHOULD retransmit the
   request by passing it to the transport layer, and SHOULD reset the
   timer with a value of 2*T1. When the timer fires 2*T1 seconds later,
   the request SHOULDbe retransmitted again (assuming the client
   transaction is still in this state). This process SHOULDcontinue, so
   that the request is retransmitted with intervals that double after
   each transmission. These retransmissions SHOULDonly be done while the
   client transaction is in the "calling" state.

   The default value for T1 is 500ms. T1 is an estimate of the RTT
   between the client and server transactions. The optional RTT
   estimation procedure of Section 17.3 MAY be followed, in which case
   the resulting estimate MAY be used instead of 500ms. If no RTT
   estimation is used, other values MAYbe used in private networks where
   it is known that RTT has a different value. On the public Internet,
   T1 MAY be chosen larger, but SHOULD NOT be smaller.

   If the client transaction is still in the "calling" when timer B
   fires, the client transaction SHOULD inform the TU that a timeout has
   occurred. The client transaction MUST NOT generate an ACK. The value
   of 64*T1 is equal to the amount of time required to send seven
   requests in the case of an unreliable transport.

   If the client transaction receives a provisional response while in
   the "calling" state, it transitions to the "proceeding" state. Upon
   entering this state, the client transaction MUST start timer C with
   the value provided by the TU when the client transaction was created.
   This timeout dictates how long the client transaction waits for a



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   final response before giving up (i.e., roughly how long does it "let
   the phone ring"). In the "proceeding" state, the client transaction
   SHOULD NOT retransmit the request any longer. Furthermore, the
   provisional response MUST be passed to the TU. Any further
   provisional responses MUST be passed up to the TU while in the
   "proceeding" state. When timer C fires, the client transaction MUST
   transition to the terminated state, and it MUST inform the TU of the
   timeout.

   When in either the "calling" or "proceeding" states, reception of a
   response with status code from 300-699 MUST cause the client
   transaction to transition to "completed". The client transaction MUST
   pass the received response up to the TU, and it MUST generate an ACK
   request, even if the transport is reliable (guidelines for
   constructing the ACK from the response are given in Section 17.1.1.3)
   and then pass the ACK to the transport layer for transmission. The
   ACK MUST be sent to the same address, port and transport that the
   original request was sent to. The client transaction SHOULD start
   timer D when it enters the "completed" state, with a value of T3
   seconds for unreliable transports, and zero seconds for reliable
   transports. T3 is the total amount of time that the server
   transaction can remain in the "completed" state when unreliable
   transports are used. For the default values of the timers below, this
   is 16 seconds.


        OPEN ISSUE #210: Timer D should be based on the values of
        the timers selected at the server, but these values aren't
        known by the client. We could alternatively specify an
        absolute minimum.

   Any retransmissions of the final response that are received while in
   the "completed" state SHOULD cause the ACK to be re-passed to the
   transport layer for retransmission, but the newly received response
   MUST NOT be passed up to the TU. A retransmission of the response is
   defined as any response which would match the same client
   transaction, based on the rules of Section 17.1.3.

   If timer D fires while the client transaction is in the "completed"
   state, the client transaction MUST move to the terminated state, and
   it MUST inform the TU of the timeout.

   When in either the "calling" or "proceeding" states, reception of a
   2xx response MUST cause the client transaction to enter the
   terminated state, and the response MUST be passed up to the TU. The
   handling of this response depends on whether the TU is a proxy core
   or a UAC core. A UAC core will handle generation of the ACK for this
   response, while a proxy core will always forward the 200 OK upstream.



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                               |INVITE from TU
             Timer A fires     |INVITE sent
             Reset A,          V             Timer B fires
             INVITE sent +-----------+       t.o. to TU
               +---------|           |---------------+
               |         |  Calling  |               |
               +-------->|           |-------------->|
                         +-----------+ 2xx           |
            300-699         |  |       2xx to TU     |
            ACK sent        |  |1xx                  |
            +---------------+  |1xx to TU            |
            |                  |                     |
            |  1xx             V       Timer C fires |
            |  1xx to TU  -----------+ t.o. to TU    |
            |  +---------|           |-------------->|
            |  |         |Proceeding |               |
            |  +-------->|           |-------------->|
            |            +-----------+ 2xx           |
            |       300-699    |       2xx to TU     |
            |       ACK sent,  |                     |
            |       resp. to TU|                     |
            |                  |                     |      NOTE:
            |  300-699         V                     |
            |  ACK sent  +-----------+               |  transitions
            |  +---------|           |               |  labeled with
            |  |         | Completed |               |  the event
            |  +-------->|           |               |  over the action
            |            +-----------+               |  to take
            |              ^   |                     |
            |              |   | Timer D fires       |
            +--------------+   | -                   |
                               |                     |
                               V                     |
                         +-----------+               |
                         |           |               |
                         | Terminated|<--------------+
                         |           |
                         +-----------+










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   The differing treatment of 200 OK between proxy and UAC is the reason
   that handling of it does not take place in the transaction layer.

   The client transaction MUST be destroyed the instant it enters the
   terminated state. This is actually necessary to guarantee correct
   operation. The reason is that 2xx responses to an INVITE are treated
   differently; each one is forwarded by proxies, and the ACK handling
   in a UAC is different. Thus, each 2xx needs to be passed to a proxy
   core (so that it can be forwarded) and to a UAC core (so it can be
   acknowledged). No transaction layer processing takes place. Whenever
   a response is received by the transport, if the transport layer finds
   no matching client transaction (using the rules of Section 17.1.3,
   the response is passed directly to the core. Since the matching
   client transaction is destroyed by the first 2xx, subsequent 2xx will
   find no match and therefore be passed to the core.

17.1.1.3 Construction of the ACK Request

   The ACK request constructed by the client transaction MUST contain
   values for the Call-ID, From, and Request-URI which are equal to the
   values of those headers in the request that created the client
   transaction (call this the "original request"). The To field in the
   ACK MUST equal the To field in the response being acknowledged, and
   will therefore usually differ from the To field in the original
   request by the addition of the tag parameter. The ACK MUST contain a
   single Via header, and this MUST be equal to the top Via header of
   the original request. The ACK request MUST NOT contain any Route
   headers. The CSeq header in the ACK MUST contain the same value for
   the sequence number as was present in the original request, but the
   method parameter MUST be equal to "ACK".

   These rules for construction of ACK only apply to the client
   transaction. A UAC core which generates an ACK for 2xx MUST instead
   follow the rules described in Section 13.

   For example, consider the following request:


   INVITE sip:bob@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP 10.1.3.3
   To: Bob <sip:bob@biloxi.com>
   From: Alice <sip:alice@atlanta.com>;tag=88sja8x
   Call-ID: 987asjd97y7atg@10.1.3.3
   CSeq: 986759 INVITE



   The ACK request for a non-2xx final response to this request would



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   look like:


   ACK sip:bob@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP 10.1.3.3
   To: Bob <sip:bob@biloxi.com>;tag=99sa0xk
   From: Alice <sip:alice@atlanta.com>;tag=88sja8x
   Call-ID: 987asjd97y7atg@10.1.3.3
   CSeq: 986759 ACK



17.1.2 non-INVITE Client Transaction

17.1.2.1 Overview of the non-INVITE Transaction

   non-INVITE transactions do not make use of ACK. They are a simple
   request-response interaction. For unreliable transports, requests are
   retransmitted at an interval which starts at T1, and doubles until it
   hits T2. If a provisional response is received, retransmissions
   continue for unreliable transports, but at an interval of T2. The
   server transaction retransmits the last response it sent (which can
   be a provisional or final response) only when a retransmission of the
   request is received. This is why request retransmissions need to
   continue even after a provisional response, they are what ensure
   reliable delivery of the final response.

   Unlike an INVITE transaction, a non-INVITE transaction has no special
   handling for the 2xx response. The result is that only a single 2xx
   response to a non-INVITE is ever delivered to a UAC.

17.1.2.2 Formal Description


   The state machine for the non-INVITE client transaction is shown in
   Figure 6. It is very similar to the state machine for INVITE.

   The "Trying" state is entered when the TU initiates a new client
   transaction with a request. When entering this state, the client
   transaction SHOULD set Timer F to fire in T3 seconds. The request
   MUST be passed to the transport layer for transmission. If an
   unreliable transport is in use, the client transaction MUST set timer
   E to fire in T1 seconds. If timer E fires while still in this state,
   the timer is reset, but this time with a value of MIN(2*T1, T2). When
   the timer fires again, it is reset to a MIN(4*T1, T2). This process
   continues, so that retransmissions occur with an exponentially
   increasing inverval that caps at T2. The default value of T2 is 4s,
   and it represents the amount of time a non-INVITE server transaction



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   will take to respond to a request, if it does not respond
   immediately. For the default values of T1 and T2, this results in
   intervals of 500 ms, 1 s, 2 s, 4 s, 4 s, 4s, etc.

   If Timer F fires while the client transaction is still in the
   "Trying" state, the client transaction SHOULD inform the TU about the
   timeout, and then it SHOULDenter the "Terminated" state. If a
   provisional response is received while in the "Trying" state, the
   response MUST be passed to the TU, and then the client transaction
   SHOULD move to the "Proceeding" state. If a final response (status
   codes 200-699) is received while in the "Trying" state, the response
   MUST be passed to the TU, and the client transaction MUST transition
   to the "Completed" state.

   If Timer E fires while in the "Proceeding" state, the request MUST be
   passed to the transport layer for retransmission, and Timer E MUST be
   reset with a value of T2 seconds. If timer F fires while in the
   "Proceeding" state, the TU MUST be informed of a timeout, and the
   client transaction MUST transition to the terminated state. If a
   final response (status codes 200-699) is received while in the
   "Proceeding" state, the response MUST be passed to the TU, and the
   client transaction MUST transition to the "Completed" state.

   Once the client transaction enters the "Completed" state, it MUST set
   Timer K to fire in T4 seconds for unreliable transports, and zero
   seconds for reliable transports. The "Completed" state exists to
   buffer any additional response retransmissions that may be received
   (which is why the client transaction remains there only for
   unreliable transports). T4 represents the amount of time the network
   will take to clear messages between client and server transactions.
   The default value of T4 is 5s. A response is a retransmission when it
   matches the same transaction, using the rules specified in Section
   17.1.3. If Timer K fires while in this state, the client transaction
   MUST transition to the "Terminated" state.


        OPEN ISSUE #211: This special treatment for reliable
        transports, where the state machine transactions directly
        to terminated, is new.

   Once the transaction is in the terminated state, it MUST be
   destroyed. As with client transactions, this is needed to ensure
   reliability of the 2xx responses to INVITE.

17.1.3 Matching Responses to Client Transactions

   When the transport layer in the client receives a response, it has to
   figure out which client transaction will handle the response, so that



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                              |Request from app
                              |send request
          Timer E             V             Timer F
          send request  +-----------+       t.o. to TU
              +---------|           |-------------------+
              |         |  Trying   |                   |
              +-------->|           |                   |
                        +-----------+                   |
           200-699         |  |                         |
           resp. to TU     |  |1xx                      |
           +---------------+  |resp. to TU              |
           |                  |                         |
           |   Timer E        V       Timer F           |
           |   send req +-----------+ t.o.to TU         |
           |  +---------|           |------------------>|
           |  |         |Proceeding |                   |
           |  +-------->|           |-----+             |
           |            +-----------+     |1xx          |
           |              |      ^        |resp to TU   |
           | 200-699      |      +--------+             |
           | resp. to TU  |                             |
           |              |                             |
           |              V                             |
           |            +-----------+                   |
           |            |           |                   |
           |            | Completed |                   |
           |            |           |                   |
           |            +-----------+                   |
           |              ^   |                         |
           |              |   | Timer K                 |
           +--------------+   | -                       |
                              |                         |
                              V                         |
        NOTE:           +-----------+                   |
                        |           |                   |
    transitions         | Terminated|<------------------+
    labeled with        |           |
    the event           +-----------+
    over the action
    to take








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   the processing of Sections 17.1.1 and 17.1.2 can take place.

   A response matches a client transaction through a comparison process
   with fields in the request that created the transaction.
   Specifically, the From, Call-ID, CSeq, and the topmost Via header
   MUST match the same fields in the request, using the matching
   operations for those headers defined in Section 22. If the To field
   in the request had a tag, the To field in the response MUST match the
   To field in the request, as described in Section 22.37. However, if
   the To field in the request did not contain a tag, the To field in
   the response MUST match that in the request, except that the tag MUST
   NOT be considered as part of the matching process. This is needed
   since a UAS will add a tag to the To field of the response.

17.1.4 Handling Transport Errors

   When the client transaction sends a request to the transport layer to
   be sent, the following procedures are followed if the transport layer
   indicates a failure.

   The client transaction SHOULD inform the TU that a transport failure
   has occurred, and the client transaction SHOULD transition directly
   to the terminated state.

17.2 Server Transaction

   The server transaction is responsible for the delivery of requests to
   the TU, and the reliable transmission of responses. It accomplishes
   this through a state machine. Server transactions are created by the
   core when a request is received, and transaction handling is desired
   for that request (this won't always be the case).

   As with the client transactions, the state machine depends on whether
   the received request is an INVITE request or not.

17.2.1 INVITE Server Transaction


   The state diagram for the INVITE server transaction is shown in
   Figure 7.

   When a server transaction is constructed with a request, it enters
   the "Proceeding" state. The server transaction MUST generate a 100
   response (not any status code - the specific value of 100) unless it
   knows that the TU will generate a provisional or final response
   within 200 ms, in which case it MAY generate a 100 response. This
   provisional response is needed to rapidly quench request
   retransmissions in order to avoid network congestion. The request



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   MUST be passed to the TU.

   The TU passes any number of provisional responses to the server
   transaction. So long as the server transaction is in the "Proceeding"
   state, each of these MUST be passed to the transport layer for
   transmission. They are not sent reliably (they are not
   retransmitted), and do not cause a change in the state of the server
   transaction. If a request retransmission is received while in the
   "Proceeding" state, the most recent provisional response that was
   received from the TU MUST be passed to the transport layer for
   retransmission. A request is a retransmission if it matches the same
   server transaction based on the rules of Section 17.2.3.

   If, while in the "proceeding" state, the TU passes a 2xx Response to
   the server transaction, the server transaction MUST pass this
   response to the transport layer for transmission. It is not
   retransmitted by the server transaction; retransmissions of 2xx
   responses are handled by the TU. The server transaction MUST then
   transition to the "terminated" state.

   While in the "Proceeding" state, if the TU passes a response with
   status code from 300 to 699 to the server transaction, the response
   MUST be passed to the transport layer for transmission, and the state
   machine MUST enter the "Completed" state. For unreliable transports,
   timer G is set to fire in T1 seconds, and is not set to fire for
   reliable transports.


        This is a change from RFC2543, where responses were always
        retransmitted, even over reliable transports.

   When the "Completed" state is entered, timer H MUST be set to fire in
   64*T1 seconds, for all transports. Timer H determines when the server
   transaction gives up retransmitting the response. Its value is chosen
   to equal Timer B, the amount of time a client transaction will
   continue to retry sending a request. If timer G fires, the response
   is passed to the transport layer once more for retransmission, and
   timer G is set to fire in MIN(2*T1, T2) seconds. From then on, when
   timer G fires, the response is passed to the transport again for
   transmission, and timer G is reset with a value that doubles, unless
   that value exceeds T2, in which case it is reset with the value of
   T2. This is identical to the retransmit behavior for requests in the
   "Trying" state of the non- INVITE client transaction. Furthermore,
   while in the "completed" state, if a request retransmission is
   received, the server SHOULD pass the response to the transport for
   retransmission.

   If an ACK is received while the server transaction is in the



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                                  |INVITE
                                  |pass to TU, send 100
               INVITE             V
               send response+-----------+
                   +--------|           |--------+101-199 from TU
                   |        | Proceeding|        |send response
                   +------->|           |<-------+
                            +-----------+
               300-699 from TU |     |2xx from TU
               send response   |     |send response
                               |     +-------------------+
                               |                         |
               INVITE          V          Timer G fires  |
               send response+-----------+ send response  |
                   +--------|           |--------+       |
                   |        | Completed |        |       |
                   +------->|           |<-------+       |
                            +-----------+                |
                               |     |                   |
                           ACK |     |                   |
                           -   |     +------------------>+
                               |           Timer H fires |
                               V           fail to TU    |
                            +-----------+                |
                            |           |                |
                            | Confirmed |                |
                            |           |                |
                            +-----------+                |
                                  |                      |
                                  |Timer I fires         |
                                  |-                     |
                                  |                      |
                                  V                      |
                            +-----------+                |
                            |           |                |
                            | Terminated|<---------------+
                            |           |
                            +-----------+










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   "Completed" state, the server transaction MUST transition to the
   "confirmed" state. As Timer G is ignored in this state, any
   retransmissions of the response will cease.

   If timer H fires while in the "Completed" state, it implies that the
   ACK was never received. In this case, the server transaction MUST
   transition to the terminated state, and MUST indicate to the TU that
   a transaction failure has occurred.

   The purpose of the "confirmed" state is to absorb any additional ACK
   messages that arrive, triggered from retransmissions of the final
   response. When this state is entered, timer I is set to fire in T4
   seconds for unreliable transports, and zero seconds for reliable
   transports. Once timer I fires, the server MUST transition to the
   "Terminated" state.

   Once the transaction is in the terminated state, it MUST be
   destroyed. As with client transactions, this is needed to ensure
   reliability of the 2xx responses to INVITE.

17.2.2 non-INVITE Server Transaction


   The state machine for the non-INVITE server transaction is shown in
   Figure 8.

   The state machine is initialized in the "Trying" state, and is passed
   a request other than INVITE or ACK when initialized. This request is
   passed up to the TU. Once in the "Trying" state, any further request
   retransmissions are discarded. A request is a retransmission if it
   matches the same server transaction, using the rules specified in
   Section 17.2.3.

   While in the "Trying" state, if the TU passes a provisional response
   to the server transaction, the server transaction MUST enter the
   "Proceeding" state. The response MUST be passed to the transport
   layer for transmission. Any further provisional responses that are
   received from the TU while in the "Proceeding" state MUST be passed
   to the transport layer for transmission. If a retransmission of the
   request is received while in the "Proceeding" state, the most
   recently sent provisional response MUST be passed to the transport
   layer for retransmission. If the TU passes a final response (status
   codes 200-699) to the server while in the "Proceeding" state, the
   transaction MUST enter the "Completed" state, and the response MUST
   be passed to the transport layer for transmission.

   When the server transaction enters the "Completed" state, it MUST set
   Timer J to fire in T3 seconds for unreliable transports, and zero



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   seconds for reliable transports. While in the "Completed" state, the
   server transaction MUST pass the final response to the transport
   layer for retransmission whenever a retransmission of the request is
   received. Any other final responses passed by the TU to the server
   transaction MUST be discarded while in the "Completed" state. The
   server transaction remains in this state until Timer J fires, at
   which point it MUST transition to the "Terminated" state.

   The server transaction MUST be destroyed the instant it enters the
   "Terminated" state.

17.2.3 Matching Requests to Server Transactions

   When an INVITE or ACK request is received from the network by the
   server, it has to be matched to an existing INVITE transaction. The
   INVITE request matches a transaction if the Request-URI, To, From,
   Call-ID, CSeq, and top Via header match those of the INVITE request
   which created the transaction. The ACK request matches a transaction
   if the Request-URI, From, Call-ID, CSeq method (not the number), and
   top Via header match those of the INVITE request which created the
   transaction, and the To field of the ACK matches the To field of the
   response sent by the server transaction (which then includes the
   tag). Matching is done based on the matching rules defined for each
   of those headers. The usage of the tag in the To field helps
   disambiguate ACK for 2xx from ACK for other responses at a proxy
   which may have forwarded both responses (which can occur in unusual
   conditions).

   For all other request methods, a request is matched to a transaction
   if the Request-URI, To, From, Call-ID and Cseq (including the method)
   and top Via header match those of the request which created the
   transaction. Matching is done based on the matching rules defined for
   each of those headers.

   Because the matching rules include the Request-URI, the server cannot
   match a response to a transaction. When the TU passes a response to
   the server, it must inform the TU which transaction the response is
   for.

17.3 RTT Estimation

   Most of the timeouts used in the transaction state machines derive
   from T1, which is an estimate of the RTT between the client and
   server transactions. This subsection defines optional procedures that
   a client can use to build up estimates of the RTT to a particular IP
   address. To perform this procedure, the client MUST maintain a table
   of variables for each destination IP address to which an RTT estimate
   is being made.



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                                  |Request received
                                  |pass to TU
                                  V
                            +-----------+
                            |           |
                            | Trying    |-------------+
                            |           |             |
                            +-----------+             |200-699 from TU
                                  |                   |send response
                                  |1xx from TU        |
                                  |send response      |
                                  |                   |
               Request            V      1xx from TU  |
               send response+-----------+send response|
                   +--------|           |--------+    |
                   |        | Proceeding|        |    |
                   +------->|           |<-------+    |
                            +-----------+             |
                                  |                   |
                                  |                   |
                                  |200-699 from TU    |
                                  |send response      |
               Request            V                   |
               send response+-----------+             |
                   +--------|           |             |
                   |        | Completed |-------------+
                   +------->|           |
                            +-----------+
                                  |
                                  |Timer J fires
                                  |-
                                  |
                                  V
                            +-----------+
                            |           |
                            | Terminated|
                            |           |
                            +-----------+












Internet Draft                    SIP                   October 26, 2001


        OPEN ISSUE #212: Is destination IP address the right index
        for an RTT estimate? How about Request-URI?

   If a client wishes to measure RTT for a particular IP address, it
   MUST include a Timestamp header into a request containing the time
   when the request is initially created and passed to a new client
   transaction, which transmits the request. If a 100 response (not any
   1xx, only the 100 response) is received before the client transaction
   generates a retransmission, an RTT estimate is made. This is
   consistent with the RFC 2988 requirements on TCP for using Karn's
   algorithm in RTT estimation.

   The estimate, called R, is made by computing the difference between
   the current time and the value of Timestamp header in the 100
   response. The value of R is applied to the estimation of RTO as
   described in Section 2 of RFC 2988 [24], with the following
   differences. First, the initial value of RTO is 500 ms for SIP, not 3
   s as is used for TCP. Second, there is no minimum value for the RTO,
   as there is for TCP, if SIP is being run on a private network. When
   run on the public Internet, the minimum is 500 ms, as opposed to 1 s
   for TCP. This difference is because of the expected usage of SIP in
   private networks where rapid call setup times are service critical.
   Once RTO is computed, the timer T1 is set to the value of RTO, and
   all other timers scale proportionally as described above.

18 Reliability of Provisional Responses

   Placeholder.

   Reliability of provisional responses will be incorporated into bis.
   This is a heads up on that.

19 Transport

   The transport layer is responsible for the actual transmission of
   requests and responses over network transports. This includes
   determination of the connection to use for a request or response, in
   the case of connection oriented transports.

   The transport layer is responsible for managing any persistent
   connections (for transports like TCP, TLS and SCTP) including ones it
   opened, as well as ones opened to it. This includes connections
   opened by the client or server transports, so that connections are
   shared between client and server transport functions. It is
   RECOMMENDED that connections be kept open for some implementation
   defined time after the last message was sent or received over that
   connection. This time SHOULD be at least 16 seconds in order to
   ensure with high probability that responses can be sent over the same



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   connection a request was sent.

   All SIP elements MUST support UDP at a minimum.

19.1 Clients

19.1.1 Sending Requests

   The client side of the transport layer is responsible for sending the
   request and receiving responses. The user of the transport layer
   passes the client transport the request, an IP address, port,
   transport, and possibly TTL for multicast destinations.

   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.) Requests sent to
   multicast groups SHOULD be scoped to ensure that they are not
   forwarded beyond the administrative domain to which they were
   targeted. This scooping MAY be done with either TTL or administrative
   scopes [19], depending on what is implemented in the network.

   It is important to note that the layers above the transport layer do
   not operate differently for multicast as opposed to unicast requests.
   This means that SIP treats multicast more like anycast, assuming that
   there is a single recipient generating responses to requests. If this
   is not the case, the first response will end up "winning", based on
   the client transaction rules. Any other responses from different UA
   will appear as retransmissions and be discarded. This limits the
   utility of multicast to cases where an anycast type of function is
   desired, such as registrations.


        OPEN ISSUE #7: This is a proposed resolution to whether or
        not multicast should be removed entirely.

   Before a request is sent, the client transport MUST insert a value of
   the sent-by field into the Via header. This field contains an IP
   address or host name, and port. In certain cases discussed in Section
   19.2.2, this IP address and port are used to construct a SIP URL for
   sending the response. The transport layer MUST be prepared to receive
   incoming connections (and receive responses sent over such
   connections) on any IP addresses and ports that this SIP URL might
   resolve to using the procedures defined in Section 24. The transport
   layer MUST also be prepared to receive an incoming connection on the
   source IP address that the request was sent from, and port number in
   the sent-by field. The client transport MUST also be prepared to



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   receive the response on the same connection used to send the request.

   For unreliable unicast transports, the client transport MUST be
   prepared to receive responses on the source IP address that the
   request is sent from (as responses are sent back to the source
   address), but the port number in the sent-by field. Furthermore, as
   with reliable transports, in certain cases the IP address and port
   are used to construct a URL for sending the response. The client
   transport MUST be prepared to receive responses on any IP
   address/port combinations that this SIP URL might resolve to using
   the procedures of Section 24.

   For multicast, the client transport MUST be prepared to receive
   responses on the same multicast group and port that the request is
   sent to.

   If a request is destined to an IP address, port, and transport to
   which an existing connection is open, it is RECOMMENDED that this
   connection be used to send the request, but another connection MAY be
   opened and used.

   If a request is sent using multicast, it is sent to the group
   address, port, and TTL provided by the transport user. If a request
   is sent using unicast unreliable transports, it is sent to the IP
   address and port provided by the transport user.

19.1.2 Receiving Responses

   When a response is received, the client transport examines the top
   Via header. If the value of the sent-by parameter in that header does
   not correspond to a value that the client transport is configured to
   insert into requests, the response MUST be rejected.

   If there are any client transactions in existence, the client
   transport uses the matching procedures of Section 17.1.3 to attempt
   to match the response to an existing transaction. If there is a
   match, the response MUST be passed to that transaction. Otherwise,
   the response MUST be passed to the core (whether it be stateless
   proxy, stateful proxy, or UA) for further processing. Handling of
   these "stray" responses is dependent on the core (a stateless proxy
   will forward all responses, for example).

19.2 Servers

19.2.1 Receiving Requests

   When the server transport receives a request over any transport, it
   MUST examine the value of the sent-by parameter in the top Via header



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   field. If the host portion of the sent-by parameter contains a domain
   name, or if it contains an IP address that differs from the packet
   source address, the server MUST add a "received" attribute to that
   Via header field. This attribute MUST contain the source address that
   the packet was received from. This is to assist the server transport
   layer in sending the response, since it must be sent to the source IP
   address that the request came from.

   Consider a request received by the server transport which looks like,
   in part:


     INVITE sip:bob@Biloxi.com SIP/2.0
     Via: SIP/2.0/UDP bobspc.biloxi.com:5060



   The request is received with a source IP address of 1.2.3.4. Before
   passing the request up, the transport would add a received parameter,
   so that the request would look like, in part:


     INVITE sip:bob@Biloxi.com SIP/2.0
     Via: SIP/2.0/UDP bobspc.biloxi.com:5060



   Next, the client transport attempts to match the request to the
   client transaction. It does so using the matching rules described in
   Section 17.2.3. If a matching server transaction is found, the
   request is passed to that transaction for processing. If no match is
   found, the request is passed to the core, which may decide to
   construct a new server transaction for that request.

19.2.2 Sending Responses

   The server transport uses the value of the top Via header in order to
   determine where to send a response. It MUST follow the following
   process:

        o If the "sent-protocol" is a reliable transport protocol such
          as TCP, TLS or SCTP, the response MUST be sent using the
          existing connection to the source of the original request that
          created the transaction, if that connection is still open.
          This does require the server transport to maintain an
          association between server transactions and transport
          connections. If that connection is no longer open, the server
          MAY open a connection to the IP address in the received



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          parameter, if present, using the port in the sent-by value, or
          the default port for that transport, if no port is specified
          (5060 for UDP and TCP, 5061 for TLS and SSL).  If that
          connection attempt fails, the server SHOULD construct a SIP
          URL of the form "sip:<sent-by host>;transport=<sent-protocol>"
          and then use the procedures defined in Section 24 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 (for unreliable unicast transports), if the top Via
          has a received parameter, 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 Section 24 to send the
          response.

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

19.3 Framing

   In the case of message oriented transports (such as UDP), if the
   message has a Content-Length header, the message body is assumed to
   contain that many bytes. If there are additional bytes in the
   transport packet below the end of the body, they MUST be discarded.
   If the transport packet ends before the end of the message body, this
   is considered an error. If the message is a response, it MUST be
   discarded. If its a request, the element SHOULD generate a 400 class
   response. If the message has no Content-Length header, the message
   body is assumed to end at the end of the transport packet.

   In the case of stream oriented transports (such as TCP), the
   Content-Length header indicates the size of the body. The Content-
   Length header MUST be used with stream oriented transports.

19.4 Error Handling




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   Error handling is independent of whether the message was a request or
   response.

   If the transport user asks for a message to be sent over an
   unreliable transport, and the result is an ICMP error, the behavior
   depends on the type of ICMP error. A host, network, port or protocol
   unreachable errors, or parameter problem errors SHOULD cause the
   transport layer to inform the transport user of a failure in sending.
   Source quench and TTL exceeded ICMP errors SHOULD be ignored.

   If the transport user asks for a request to be sent over a reliable
   transport, and the result is a connection failure, the transport
   layer SHOULD inform the transport user of a failure in sending.

20 Security Considerations

   The fundamental security issues confronting SIP are: preserving the
   confidentiality and integrity of messaging, preventing replay attacks
   or message spoofing, ensuring the privacy of the participants in a
   session, and preventing denial of service attacks.

   SIP messages frequently contain sensitive information about their
   senders not just what they have to say, but with whom they
   communicate, when they communicate and for how long, and from where
   they participate in sessions. Many applications and their users
   require that this sort of private information be hidden from any
   parties that do not need to know it.

   Encryption provides the best means to preserve the confidentiality of
   signaling it can also guarantee that messages are not modified by any
   malicious intermediaries. However, SIP requests and responses cannot
   be encrypted end-to-end (that is, between a pair of distinct user
   agents who share encryption keys) in their entirety because message
   fields such as the Request-URI, Route and Via need, in most network
   architectures, to be visible to proxies so that SIP requests are
   routed correctly. Note that proxy servers need to modify signaling as
   well (adding Via headers) in order for SIP to function. Proxy servers
   must therefore be a part of trust relationships in SIP networks.

   Note that there are also less direct ways in which private
   information can be divulged. If a user or service chooses to be
   reachable at an address that is guessable from the person's name and
   organizational affiliation (which describes most addresses of
   record), the traditional method of ensuring privacy by having an
   unlisted "phone number" is compromised. A user location service can
   infringe on the privacy of the recipient of a session invitation by
   divulging their specific whereabouts to the caller; an implementation
   consequently SHOULD be able to restrict, on a per-user basis, what



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   kind of location and availability information is given out to certain
   classes of callers.

   SIP entities also have a need to identify one another in a secure
   fashion. Ordinarily a SIP UA asserts an identity for the initiator of
   a request in the From header field, but in many systems this
   information is controlled directly by the end user, and thus spoofing
   the contents of the From is trivial. When a SIP endpoint asserts the
   identity of its user to a peer user agent or to a proxy server, that
   identity should in some way be verifiable. A cryptographic
   authentication mechanism is provided in SIP to address this
   requirement.

   The most comprehensive mechanisms for securing SIP messages
   (providing confidentiality and integrity guarantees for signaling as
   well as authentication) make use of transport or network layer
   encryption.  encryption encrypts the entire SIP request or response
   on the wire so that packet sniffers or other eavesdroppers cannot see
   who is calling whom.

   Note that the security of SIP signaling itself has no bearing on the
   security of protocols used in concert with SIP such as RTP, or with
   any MIME types carried as SIP bodies, such as SDP. Any media
   associated with a session can be encrypted end-to-end without any of
   the problems associated with encrypting SIP signaling. Media
   encryption is outside the scope of this document.

20.1 Transport and Network Layer Security

   SIP requests and responses MAY be protected by security mechanisms at
   the transport or network layer. No particular mechanism is
   recommended by this document, but two popular alternatives are
   briefly examined: protection at the transport layer can be afforded
   by TLS [25], and network layer security is provided by IPSec [26].

   Transport or network layer security encrypts signaling traffic,
   guaranteeing message confidentiality and integrity (note however that
   the originator and recipient of a session may be deducible by
   observers performing a network traffic analysis). The keys used to
   establish encrypt traffic can also be used to verify an asserted
   identity in many architectures, and therefore provide a means of
   authentication.

   IPSec is a network layer protocol essentially, a secure replacement
   for traditional IP (Internet Protocol). IPSec is most suited to VPN
   (virtual private network) architectures in which a set of SIP hosts
   (mingled user agents and proxy servers) or bridged administrative
   domains have a trust relationship with one another.



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   TLS is a transport protocol and hence, like TCP and UDP, TLS can be
   specified as the desired transport protocol within a Via header field
   or a SIP-URI. TLS is most suited to architectures in which a chain of
   trust joins together a set of hosts (e.g. Alice trusts her local
   proxy server, which in turn trust Bob's local proxy server, which Bob
   trusts, hence Bob and Alice can communicate securely).

   TLS must be tightly coupled with a SIP application. Note that
   transport mechanisms are specified on a hop-by-hop basis in SIP, and
   that in some networks TLS might be used for only certain portions of
   the signaling path.

   It is RECOMMENDED that SIP endpoints support TLS as a secure
   transport for SIP.

20.2 SIP Authentication

   SIP provides a stateless challenged-based mechanism for
   authentication.  Any time that a proxy server or user agent receives
   a request, they MAY challenge the initiator of the request to provide
   assurance of their identity. Once the originator has been identified,
   the recipient of the request SHOULD ascertain whether or not this
   user is authorized to make the request in question. No authorization
   systems are recommended or discussed in this document.

   The "basic" and "digest" authentication mechanisms described in this
   section provide message authentication only, without message
   integrity or confidentiality. Protective measures above and beyond
   authentication need to be taken to prevent active attackers from
   modifying and/or replaying SIP requests and responses.

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

20.2.1 Framework

   The framework for SIP authentication closely parallels that of HTTP
   (RFC 2617 [27]). 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
   identity of a user agent client. Additionally, registrars and
   redirect servers MAY make use of 401 (Unauthorized) responses for
   authentication, but proxies MUST NOT, and instead MAY use the 407
   (Proxy Authentication Required) response. The requirements for
   inclusion of the Proxy-Authenticate, Proxy- Authorization, WWW-
   Authenticate, and Authorization in the various messages are identical
   to those described in RFC 2617 [27].



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   Since SIP does not have the concept of a canonical root URL, the
   notion of protection spaces is interpreted differently in 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:bob@biloxi.com SIP/2.0
      WWW-Authenticate:  Basic realm="business"



   and


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



   Generally, SIP authentication is for a specific request 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 (e.g. an
   INVITE to 'sip:10.3.6.6'). 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.

20.2.2 User to User Authentication

   When a UAS receives a request from a UAC, the UAS MAY authenticate
   the originator before the request is processed. If no credentials (in
   the Authorization header field are provided in the request, the UAS
   can challenge the originator to provide credentials by rejecting the
   request with a 401 (Unauthorized) status code.

   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.

   An example of the WWW-Authenticate in a 401 challenge is:


      WWW-Authenticate:  Basic realm="business"



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   When the originating UAC receives the 401 it SHOULD, if it is able,
   re-originate the request with the proper credentials. The UAC may
   require input from the originating user before proceeding. The
   content of the "realm" parameter of the WWW-Authenticate header
   SHOULD be displayed to the user. Once authentication credentials have
   been supplied (either directly by the user, or discovered in a
   keyring), user agents SHOULD cache the credentials for a given value
   of the Request-URI and "realm" and attempt to re-use these values on
   the next request for that destination.

   Any 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.

   An example of the Authorization header is:


      Authorization: Basic QWxhZGRpbjpvcGVuIHNlc2FtZQ==



   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.

20.2.3 Proxy to User Authentication

   Similarly, when a UAC sends a request to a proxy server, the proxy
   server MAY authenticate the originator before the request is
   processed. If no credentials (in the Proxy-Authorization header
   field) are provided in the request, the UAS can challenge the
   originator to provide credentials by rejecting the request with a 407
   (Proxy Authentication Required) status code. The proxy MUST populate
   the 407 (Proxy Authentication Required) message with a Proxy-
   Authenticate header applicable to the proxy for the requested
   resource.

   The use of the Proxy-Authentication and Proxy-Authorization parallel
   that described in [27], with one difference. Proxies MUST NOT add the
   Proxy-Authorization header. 407 (Proxy Authentication Required)
   responses MUST be forwarded upstream towards the UAC following the
   procedures for any other response. It is the client's responsibility
   to add the Proxy-Authorization header containing credentials for the
   realm of the proxy which has asked for authentication.




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        If a proxy were to resubmit a request with a Proxy-
        Authorization header field, it would need to increment the
        CSeq in the new request. However, this would mean that the
        UAC which submitted the original request would discard a
        response from the UAS, as the CSeq value would be
        different.

   When the originating UAC receives the 407 it SHOULD, if it is able,
   re-originate the request with the proper credentials. It should
   follow the same procedures for the display of the "realm" parameter
   that are given above for responding to 401.

   Any user agent that wishes to authenticate itself to a proxy server
   -- usually, but not necessarily, after receiving a 407 response --
   MAY do so by including an Proxy-Authorization header field with the
   request. 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.

   A Proxy-Authorization header field applies only to the proxy whose
   realm is identifier in the "realm" parameter (this proxy may
   previously have demanded authentication using the Proxy-Authenticate
   field). When multiple proxies are used in a chain, the Proxy-
   Authorization header field MUST NOT be consumed by any proxy whose
   realm does not match the "realm" parameter specified in the Proxy-
   Authorization header.

   Note that if an authentication scheme is used in the Proxy-
   Authorization that does not support realms, a proxy server MUST
   attempt to parse all Proxy-Authorization headers to determine whether
   or not one of them has what it considers to be valid credentials.
   Because this is potentially very time consuming in large networks,
   proxy servers SHOULD use an authentication scheme that supports
   realms in the Proxy-Authorization header.

   It is also possible that a 401 or 407 response will contain several
   challenges, from a mixture of proxies and user agent servers, if the
   request was forked. If at least one user agent responds to a request
   with a challenge, than a 401 should be used; otherwise a 407 should
   be used. When resubmitting its request in response to the challenge,
   the UAC needs to include an Authorization for each WWW-Authenticate
   and Proxy- Authorization for each Proxy-Authenticate.

   See [H14.34] for a definition of the syntax of Proxy- Authentication
   and Proxy-Authorization.




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20.2.4 Authentication Schemes

   SIP implementations MAY use HTTP's basic and digest authentication
   mechanisms ([27]) to provide a rudimentary form of security. This
   section overviews usage of these mechanisms in SIP.  The scheme usage
   is almost completely identical to that for HTTP [27]. This section
   outlines this operation, pointing to RFC 2617 ([27]) for details and
   noting the differences that arise when using SIP. Since RFC 2543 is
   based on HTTP basic and digest as defined in RFC 2069 [28], SIP
   servers supporting RFC 2617 MUST ensure they are backwards compatible
   with RFC 2069. Procedures for this backwards compatibility are
   specified in RFC 2617.

20.2.4.1 HTTP Basic

   The rules for basic authentication follow those defined in [27] but
   with the words "origin server" replaced with "user agent server,
   redirect server , or registrar".

   Since SIP URIs are not hierarchical, the paragraph in [27] 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.

20.2.4.2 HTTP Digest

   The rules for digest authentication follow those defined in [27],
   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:





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             digest-uri-value  =  Request-URI ; as defined in
             Section 26


        4.   The example procedure for choosing a nonce based on Etag
             does not work for SIP.

        5.   The text in RFC 2617 [27] regarding cache operation does
             not apply to SIP.

        6.   RFC 2617 [27] 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 for whom that the server is willing
             to accept forwarded or direct calls.

   RFC2543 did not allow usage of the Authentication-Info header (it
   effectively used RFC 2069). However, we now allow usage of this
   header, since it provides integrity checks over the bodies and
   provides mutual authentication. RFC2617 [27] defines mechanisms for
   backwards compatibility using the qop attribute in the request. These
   mechanisms MUST be used by a server to determine if the client
   supports the new mechanisms in RFC 2617 that were not specified in
   RFC 2069.

20.3 SIP Encryption

   No mechanism is currently specified for encrypting entire SIP
   messages end-to-end for the purpose of confidentiality. This is a
   hard problem because network intermediaries (like proxy servers) need
   to view certain headers in order to route messages correctly, and if
   these intermediaries are excluded from security associations then SIP
   messages will essentially be unroutable.

   That much said, SIP messages carry MIME bodies and the MIME standard
   includes mechanisms for securing MIME contents to ensure both
   integrity and confidentiality (including the 'multipart/encrypted'
   MIME type, see [29]), but detailed description of the use of secure
   MIME types are outside the scope of this document. Implementors
   should note, however, that there may be rare network intermediaries
   (not typical proxy servers) that rely on viewing or modifying the
   bodies of SIP messages (especially SDP), and that secure MIME may
   prevent these sorts of intermediaries from functioning.

        This applies particularly to certain types of firewalls.



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   End-to-end encryption relies on keys shared by the two user agents
   involved in the request. Typically, the message is sent encrypted
   with the public key of the recipient, so that only that recipient can
   read the message. SIP does not define any mechanism for end-to-end
   key exchange.


        Note that the PGP mechanism for encrypting the headers and
        bodies of SIP messages described in RFC2543 has been
        deprecated.

20.4 Denial of Service

   Denial of service attacks focus on rendering a particular network
   element unavailable, usually by directing an excessive amount of
   network traffic at its interfaces. A distributed denial of service
   attack allows one network user to cause multiple network hosts to
   flood a target host with a large amount of network traffic.

   In many architectures SIP proxy servers face the public Internet in
   order to accept requests from worldwide IP endpoints. When the host
   on which a SIP proxy server is operating is routable from the public
   Internet, it should be deployed in an administrative domain with
   secure routing policies (blocking source-routed traffic, preferably
   filtering ping traffic).

   SIP creates a number of potential opportunities for distributed
   denial of service attacks that must be recognized and addressed by
   the implementors and operators of SIP systems.

   Floods of messages directed at proxy servers can lock up proxy server
   resources and prevent desirable traffic from reaching its
   destination.  There is a computational expense associated with
   processing a SIP transaction at a proxy server, and that expense is
   greater for stateful proxy servers that it is for stateless proxy
   servers. Therefore stateful proxies are more susceptible to flooding
   than stateless proxy servers.

   Attackers can create bogus requests that contain a falsified Via
   header field which identifies a targeted host as the originator of
   the message and then send this message to a large number of SIP
   network elements, thereby using hapless SIP UAs or proxies to
   generate denial of service traffic aimed at the target.

   Similarly, attackers might use falsified Route headers in a request
   that identify the target host and then send such messages to forking
   proxies that will amplify messaging sent to the target.  Record-Route
   could be used to similar effect when the attacker is certain that the



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   SIP dialog initiated by the request will result in numerous
   transactions originating in the backwards direction.

   One could prevent one's host from being commandeered for such an
   attack by disallowing requests that do not make use of a persistent
   security association established through a transport or network layer
   security instrument such as TLS or IPsec. This could be an
   appropriate security solution for two proxy servers that trust one
   another and exchange significant amounts of signaling traffic with
   one another, or between a user agent and its outbound proxy.

   Both TLS and IPSec can also make use of bastion hosts at the edges of
   administrative domains that participate in the security associations
   to aggregate secure tunnels and sockets. These bastion hosts can also
   take the brunt of denial of service attacks, ensuring that SIP hosts
   within the administrative domain are not encumbered with superfluous
   messaging.

   If such a persistent security association is not feasible, user
   agents and proxy servers SHOULD challenge questionable requests with
   only a single 401 (Unauthorized) or 407 (Proxy Authentication
   Required) forgoing the normal response retransmission algorithm.

        Retransmitting the 401 or 407 status response amplifies the
        problem of an attacker using a falsified header (such as
        Via) to direct traffic to a third party.

   A number of denial of service attacks open up if REGISTER requests
   are not properly authenticated and authorized by registrars.
   Attackers could de-register some or all users in an administrative
   domain, thereby preventing these users from being invited to new
   sessions. An attacker could also register a large number of contacts
   designating the same host for a given address of record in order to
   use the registrar and any associated proxy servers as amplifiers in a
   denial of service attack. Attackers might also attempt to deplete
   available memory and disk resources of a registrar by registering
   huge numbers of bindings.

   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.

   The use of multicast to transmit SIP requests can greatly increase
   the potential for denial of service attacks.



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21 Common Message Components

   There are certain components of SIP messages that appear in various
   places within SIP messages (and sometimes, outside of them), which
   merit separate discussion.

21.1 SIP Uniform Resource Locators

   A SIP URL identifies a communications resource. Like all URLs, SIP
   URLs may be placed in web pages, email messages or printed
   literature. They contain sufficient information to initiate and
   maintain a communication session with the resource.

   Examples of communications resources include

        o a user of an online service

        o an appearance on a multiline phone

        o a mailbox on a messaging system

        o a PSTN phone number at a gateway service

        o a group (such as "sales" or "helpdesk") in an organization

21.1.1 SIP URL components

   The "sip:" scheme follows the guidelines in RFC 2396 [9].  It uses a
   form similar to the mailto URL, allowing the specification of SIP
   request-header fields and the SIP message- body.  This makes it
   possible to specify the subject, media type, or urgency of sessions
   initiated by using a URL on a web page or in an email message.  The
   formal syntax for a SIP URL is presented in Section 26.  Its general
   form is
            sip:user:password@host:port;url-parameters?headers
   These tokens, and some of the tokens in their expansion, have the
   following meanings.

        user: The identifier of a particular resource at the host being
             addressed.

             Note that "host" as used here may, and frequently does,
             refer to a domain.

             The "userpart" of a URL consists of this user field, the
             password field and the @ sign following them. The userpart
             of a URL is optional and MAY be absent when the destination
             host does not have a notion of users or when the host



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             itself is the resource being identified. If the @ sign is
             present in a SIP URL, the user field MUST NOT be empty.

             If the host being addressed is capable of processing
             telephone numbers, an Internet telephony gateway for
             instance, a telephone- subscriber field defined in RFC 2806
             [13] MAY be used to populate the user field. There are
             special escaping rules for encoding telephone-subscriber
             fields in SIP URLs described in Section 21.1.2.

        password: A password associated with the user

             While the SIP URL syntax allows this field to be present,
             its use 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. For instance, transporting a PIN number in
             this field exposes the PIN.

        host: The entity hosting the SIP resource

             The host part contains either a fully-qualified domain name
             or numeric IPv4 or IPv6 address. Using the fully-qualified
             domain name form is RECOMMENDED whenever possible.

        port: The port number where the request is to be sent.

        URL parameters: Parameters affecting a request constructed from
             the URL.

             URL parameters are added after the hostport component and
             are separated by semi-colons. This extensible mechanism
             includes the transport, maddr, ttl, user, and method
             parameters.

             The transport parameter determines the transport mechanism
             to be used for sending SIP messages. SIP can use any
             network transport protocol.  Parameter names are defined
             for UDP [30], TCP [31], TLS [25], and SCTP [32].

             The maddr parameter indicates the server address to be
             contacted for this user, overriding any address derived
             from the host field. Section 24 describes the proper
             interpretation of the transport, maddr and hostport in
             order to obtain the destination address, port and transport
             for sending a request.





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             The maddr field can be used as a simple form of loose
             source routing. It allows a URL to specify a specific
             proxy that must be traversed en-route to the
             destination. This capability is useful for a roaming
             user that is forced to use an outbound proxy, but
             wishes to force requests through their home proxy.

             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 alice@atlanta.com using multicast to 239.255.255.1
             with a ttl of 15, the following URL would be used:


               sip:alice@atlanta.com;maddr=239.255.255.1;ttl=15



             The set of valid telephone-subscriber strings is a subset
             of valid user strings. The user URL parameter exists to
             distinguish telephone numbers from user names that happen
             to look like telephone numbers.  If the user string
             contains a telephone number formatted as a telephone-
             subscriber, the user parameter value "phone" SHOULD be
             present. Even without this parameter, recipients of SIP
             URLs MAY interpret the pre-@ part as a telephone number if
             local restrictions on the name space for user name allow
             it.

             The method of the SIP request constructed from the URL can
             be specified with the method parameter.

             Since the url-parameter mechanism is extensible, SIP
             elements MUST silently ignore any url-parameters that they
             do not understand.

        Headers: Headers to be included in a request constructed from
             the URL.

             Headers fields in the SIP request can be specified with the
             "?" mechanism within a SIP URL. The header names and values
             are encoded in ampersand separated hname = hvalue pairs.
             The special hname "body" indicates that the associated
             hvalue is the message-body of the SIP request.

   Table 1 summarizes the use of SIP URL components based on the context
   in which the URL appears. The external column describes URLs



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   appearing anywhere outside of a SIP message, for instance on a web
   page or business card.  Entries marked "m" are mandatory, those
   marked "o" are optional, and those marked "-" are not allowed.
   Elements processing URLs SHOULD ignore any disallowed components if
   they are present.  The second column indicates the default value of
   an optional element if it is not present. "--" indicates that the
   element is either not optional, or has no default value.

   SIP URLs in Contact header fields have different restrictions
   depending on the context in which the header field appears. One set
   applies to messages that establish and maintain dialogs (INVITE and
   its 200 OK response). The other applies to registration and
   redirection messages (REGISTER, its 200 OK response, and 3xx class
   responses to any method).

   OPEN ISSUE #203: maddr is disallowed in To/From, but not port. Should
   port be disallowed?

   OPEN ISSUE #204: Password is disallowed in From, but not To. Why?

   OPEN ISSUE #205: Should we allow method and header URL components in
   registration/redirect Contacts. What do they mean?


                                                             dialog
                                               reg./redir.  Contact/
                  default  Req.-URI  To  From    Contact    R-R/Route  external
   user           --          o      o    o         o           o         o
   password       --          o      o    -         o           o         o
   host           --          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           o         o
   ttl-param      1           o      -    -         o           -         o
   transp.-param  udp         o      -    -         o           o         o
   other-param    --          o      o    o         o           o         o
   headers        --          -      -    -         o           -         o


   Table 1: Use and default values of URL components  for  SIP  headers,
   Request-URI and references


21.1.2 Character escaping requirements

   SIP follows the requirements and guidelines of RFC 2396 when defining
   the set of characters that must be escaped in a SIP URL, and uses its



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   ""%" HEX HEX" mechanism for escaping. From RFC 2396:


        The set of characters actually 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.

   For each component, the set of valid BNF expansions defines exactly
   which characters may appear unescaped. All other characters MUST be
   escaped.

   For example, "@" is not in the set of characters in the user
   component, so the user "j@s0n" must have at least the @ sign encoded,
   as in "j%40s0n".

   Expanding the hname and hvalue tokens in Section 26 show that all URL
   reserved characters in header names and values MUST be escaped.

   The telephone-subscriber subset of the user component has special
   escaping considerations. The set of characters not reserved in the
   RFC 2806 [13] description of telephone-subscriber contains a number
   of characters in various syntax elements that need to be escaped when
   used in SIP URLs. Any characters occurring in a telephone-subscriber
   that do not appear in an expansion of the BNF for the user rule MUST
   be escaped.

21.1.3 Example SIP URLs


     sip:alice@atlanta.com
     sip:alice:secretword@atlanta.com;transport=tcp
     sip:alice@atlanta.com?subject=project
     sip:+1-212-555-1212:1234@gateway.com;user=phone
     sip:1212@gateway.com
     sip:alice@10.1.1.1
     sip:atlanta.com;method=REGISTER?to=alice
     sip:alice;day=tuesday@atlanta.com



   The last example URL above has a user field value of
   "alice;day=tuesday". The escaping rules defined above allow a
   semicolon to appear unescaped in this field. Note, however, that for



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   the purposes of this protocol, the field is opaque.  The apparent
   structure in that value is only useful to the entity responsible for
   the resource.

21.1.4 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.  (OPEN ISSUE #100 : There is a
          proposal to make only quoted string comparisons case-
          sensitive.)

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

        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 For two URLs to be equal, the user, password, host, and port
          components must match. A URL omitting the optional port
          component will match a URL explicitly declaring port 5060. A
          URL omitting the user component will not match a URL that
          includes one. A URL omitting the password component will not
          match a URL that includes one.

        o URL url-parameter components are compared as follows

          - Any url-parameter appearing in both URLs must match.

          - A user, transport, ttl, or method url-parameter appearing in
            only one URL must contain its default value or the URLs do
            not match.

          - All other url-parameters appearing in only one URL are
            ignored when comparing the URLs.

        o URL header components are never ignored. Any present header
          component MUST be present in both URLs and match for the URLs
          to match. The matching rules are defined for each header in
          Section sec:header-fields.

   The URLs within each of the following sets are equivalent:



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   sip:alice@%61tlanta.com
   sip:alice@AtLanTa.CoM;Transport=udp




   sip:carol@chicago.com
   sip:carol@chicago.com;newparam=5
   sip:carol@chicago.com;security=on




   sip:biloxi.com;transport=tcp;method=REGISTER?to=sip:bob
   sip:biloxi.com;method=REGISTER;transport=tcp?to=sip:bob




   sip:alice@atlanta.com?subject=project
   sip:alice@atlanta.com?priority=urgent&subject=project



   The URLs within each of the following sets are not equivalent:


   SIP:ALICE@AtLanTa.CoM;Transport=udp               (different usernames)
   sip:alice@AtLanTa.CoM;Transport=UDP




   sip:bob@biloxi.com                       (different port and transport)
   sip:bob@biloxi.com:6000;transport=tcp




   sip:carol@chicago.com                      (different header component)
   sip:carol@chicago.com?Subject=next




   sip:bob@phone21.boxesbybob.com     (even though that's what
   sip:bob@10.4.1.4                    phone21.boxesbybob.com resolves to)




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   Note that equality is not transitive:

        o sip:carol@chicago.com and sip:carol@chicago.com;security=on
          are equivalent

        o sip:carol@chicago.com and sip:carol@chicago.com;security=off
          are equivalent

        o sip:carol@chicago.com;security=on and
          sip:carol@chicago.com;security=off are not equivalent

   Comparing URLs is a major part of comparing several SIP headers (see
   Section 22).

21.2 Option Tags

   Option tags are unique identifiers used to designate new options
   (extensions) in SIP.  These tags are used in Require (Section 22.30),
   Proxy-Require (Section 22.28, Supported (Section 22.35) and
   Unsupported (Section 22.38) header fields.  Note that these options
   appear as parameters in those headers in an  option-tag = token  form
   (see Section 26 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) (See Section 27).

   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.

21.3 Tags

   The "tag" parameter is used in the To and From fields of SIP
   messages. It serves as a general mechanism to identify a particular
   instance of a user agent for a particular SIP URI.

   As proxies can fork requests, the same request can reach multiple
   instances of a user (mobile and home phones, for example). Since each
   can respond, there needs to be a means for the originator of a
   session to distinguish the responses. Tag fields in the To and From
   disambiguate these multiple instances of the same user.




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   This situation also arises with multicast requests.

   When a tag is generated by a UA for insertion into a request or
   response, it MUST be globally unique and cryptographically random
   with at least 32 bits of randomness. A property of this selection
   requirement is that a UA will place a different tag into the From
   header of an INVITE as it would place into the To header of the
   response to the same INVITE. This is needed in order for a UA to
   invite itself to a session, a common case for "hairpinning" of calls
   in PSTN gateways.

   Besides the requirement for global uniqueness, the algorithm for
   generating a tag is implementation specific. Tags are helpful in
   fault tolerant systems, where a dialog is to be recovered on an
   alternate server after a failure. A UAS can select the tag in such a
   way that a backup can recognize a request as part of a dialog on the
   failed server, and therefore determine that it should attempt to
   recover the dialog and any other state associated with it.

22 Header Fields

   The general syntax for header fields is covered in Section 7.3. This
   section lists the full set of header fields along with notes on
   syntax, meaning, and usage. Throughout this section, we use [HX.Y] to
   refer to Section X.Y of the current HTTP/1.1 specification RFC 2617
   [27]. Examples of each header field are given.

   Information about header fields in relation to methods and proxy
   processing is summarized in Tables 2 and 3.

   The "where" column describes the request and response types in which
   the header field can be used. Values in this column are:

        R: refers to header fields that can be used in requests.

        r: designates a 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.

        c: indicates a header field is copied from the request to the
             response.

   The "proxy" column describes the operations a proxy may perform on a
   header.

        c: indicates that a proxy can add (concatenate) comma-separated
             elements to the header




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        m: indicates that a proxy can modify the header

        a: indicates that a proxy can add the header if not present

        r: indicates that a proxy must be be able to read the header.
             Headers that need to be read cannot be encrypted.

   The next six columns relate to the presence of a header field in a
   method, with the contents indicating:

        o: for optional

        m: for mandatory

        m*: indicates a header that SHOULD be sent, but servers need to
             be prepared to receive messages without that header field.

        *: indicates that the header fields are required if the message
             body is not empty. See sections 22.14, 22.15 and 7.4 for
             details.

        -: for not applicable.

   "Optional" means thata 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 (The exception to this rule is the Require header
   field discussed in 22.30). A "mandatory" 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 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.


   A compact form of some common header fields is also defined for use
   when overall message size is an issue.

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



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       Header field          where   proxy ACK BYE CAN INV OPT REG
       ____________________________________________________________
       Accept                  R            -   o   -   m*  o   o
       Accept                 2xx           -   -   -   m*  o   o
       Accept                 415           -   o   -   o   o   o
       Accept-Encoding         R            -   o   -   m*  o   o
       Accept-Encoding        2xx           -   -   -   m*  o   o
       Accept-Encoding        415           -   o   -   o   o   o
       Accept-Language         R            -   o   -   m*  o   o
       Accept-Language        2xx           -   -   -   m*  o   o
       Accept-Language        415           -   o   -   o   o   o
       Alert-Info              R      am    -   -   -   o   -   -
       Alert-Info             180     am    -   -   -   o   -   -
       Allow                   R            o   o   o   o   o   o
       Allow                  2xx           -   o   o   m*  m*  o
       Allow                   r            -   o   o   o   o   o
       Allow                  405           -   m   m   m   m   m
       Authentication-Info    2xx           -   o   -   o   o   o
       Authorization           R            o   o   o   o   o   o
       Call-ID                 c       r    m   m   m   m   m   m
       Call-Info                      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                  o   o   -   o   o   o
       Content-Encoding                     o   o   -   o   o   o
       Content-Language                     o   o   -   o   o   o
       Content-Length                  r    m*  m*  m*  m*  m*  m*
       Content-Type                         *   *   -   *   *   *
       CSeq                    c       r    m   m   m   m   m   m
       Date                            a    o   o   o   o   o   o
       Error-Info           300-699         -   o   o   o   o   o
       Expires                              -   -   -   o   -   o
       From                    c       r    m   m   m   m   m   m
       In-Reply-To             R            -   -   -   o   -   -
       Max-Forwards            R      rm    o   o   o   o   o   o
       MIME-Version                         o   o   o   o   o   o
       Organization                   am    -   -   -   o   o   o


   Table 2: Summary of header fields, A--O

22.1 Accept

   The Accept header follows the syntax defined in [H14.1]. The
   semantics are also identical, with the exception that if no Accept


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


   Table 3: Summary of header fields, P--Z; (1):  copied  with  possible
   addition of tag

   header is present, the server SHOULD assume a default value of
   application/sdp

   Example:


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



22.2 Accept-Encoding

   The Accept-Encoding 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].

   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


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   no encoding, is permissible. If this header is not present, the
   default value is identity. This differs slightly from the HTTP
   definition, which indicates that when not present, any encoding can
   be used, but the identity encoding is preferred.

   Example:


     Accept-Encoding: gzip



22.3 Accept-Language

   The Accept-Language header follows the syntax defined in [H14.4]. The
   rules for ordering the languages based on the "q" parameter apply to
   SIP as well.

   The Accept-Language header is used in requests to indicate the
   preferred languages for reason phrases, session descriptions or
   status responses carried as message bodies in the response. If no
   Accept-Language header field is present in a request, the server
   assumes all languages are acceptable to the client.

   Example:


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



22.4 Alert-Info

   When present in an INVITE request, the Alert-Info header field
   specifies an alternative ring tone to the UAS. When present in a 180
   (Ringing) response, the Alert-Info header field specifies an
   alternative ringback tone to the UAC. A typical usage is for a proxy
   to insert this header to provide a distinctive ring feature.

   The Alert-Info header can introduce security risks. These risks, and
   the ways to handle them, are discussed in Section 22.9 which
   discusses the Call-Info header, as the risks are identical.

   In addition, a user SHOULD be able to disable this feature
   selectively.


        This helps prevent disruptions that could result from the



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        use of this header by untrusted elements.

   Example:

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



22.5 Allow

   The Allow header field lists the set of methods supported by the user
   agent generating the message.

   All methods, including ACK and CANCEL, understood by the UA MUST be
   included in the list of methods in the Allow header, when present.
   The absence of an Allow header MUST NOT be interpreted to mean that
   the UA sending the message supports no methods. Rather, it implies
   that the UA is not providing any information on what methods it
   supports.

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

   Example:

     Allow: INVITE, ACK, OPTIONS, CANCEL, BYE



22.6 Authentication-Info

   The Authentication-Info header provides for mutual authentication
   with HTTP Digest. A UAS MAY include this header in a 2xx response to
   a request that was successfully authenticated using digest based on
   the Authorization header.

   Syntax and semantics follow those specified in RFC2617 [27].

   Example:

     Authentication-Info: nextnonce="47364c23432d2e131a5fb210812c"



22.7 Authorization

   The Authorization header field contains authentication credentials of
   a UA. Section 20.2.2 overviews the use of the Authorization header



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   field, and Section 20.2.4 describes the syntax and semantics when
   used with HTTP Basic and Digest authentication.

   Note that this header field, along with Proxy-Authorization breaks
   the general rules about multiple header fields. Although not a
   comma-separated list, this header field may be present multiple
   times, and MUST NOT be combined into a single header using the usual
   rules described in Section 7.3.

   Example:

     Authorization: Digest username="Alice", realm="Bob's Friends",
      nonce="84a4cc6f3082121f32b42a2187831a9e",
      response="7587245234b3434cc3412213e5f113a5432"



22.8 Call-ID

   The Call-ID 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.Call-IDs are case- sensitive
   and are simply compared byte-by-byte.

   The compact form of the Call-IDheader field is i.

   Examples:

     Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@biloxi.com
     i:f81d4fae-7dec-11d0-a765-00a0c91e6bf6@10.4.1.4



22.9 Call-Info

   The Call-Info 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 [33] or LDIF [34] formats). Additonal
   tokens can be registered using IANA and the procedures in Section 27.

   Usage of the Call-Info header can pose a security risk. If a callee
   fetches the URLs provided by an malicious caller, the callee may be



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   at risk for displaying inappropriate or offensive content, dangerous
   or illegal content, and so on. Therefore, it is RECOMMENDED that a UA
   only render the information in the Call-Info header if it can verify
   the authenticity of the element which originated the header, and
   trusts that element. This need not be the peer UA; a proxy can insert
   this header into requests.

   The use of this header is important in converged applications.

   Example:

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



22.10 Contact

   The Contact header field provides a URL whose meaning depends on the
   the type of request or response it is in.

   Parameters defined for Contact include "q" and "expires". Additional
   parameters may be defined in other specifications.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.

   The compact form of the Contact header field is m (for "moved").

   Examples:


     Contact: "Mr. Watson" <sip:watson@worcester.bell-telephone.com>
        ;q=0.7; expires=3600,
        "Mr. Watson" <mailto:watson@bell-telephone.com> ;q=0.1
     m: <sip:bob@10.5.1.5>






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22.11 Content-Disposition

   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 [35]).

   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. The parameter has defined values of
   "optional" and "required". 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.

   Example:

     Content-Disposition: session



22.12 Content-Encoding

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



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   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. 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.

   The compact form of the Content-Encoding header field is e.

   Examples:

     Content-Encoding: gzip
     e: tar



22.13 Content-Language

   See [H14.12].

   Example:

     Content-Language: fr



22.14 Content-Length

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

   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.

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

   The short form of the header is l.

   Examples:

     Content-Length: 349
     l: 173



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22.15 Content-Type

   The Content-Type 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).

   The short form of the header is c.

   Examples:

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



22.16 CSeq

   A CSeq header field in a request contains a single decimal sequence
   number and the request method. The sequence number MUST be
   expressible as a 32-bit unsigned integer. The CSeq header serves to
   order transactions within a dialog, and to provide a means to
   uniquely identify transactions, and to differentiate between new
   requests and request retransmissions.

   Example:


     CSeq: 4711 INVITE



22.17 Date

   The Date header field contains an RFC 1123 date (see [H14.18]). Note
   that unlike HTTP/1.1, SIP only supports the most recent RFC 1123 [36]
   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.



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        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.

   Example:


     Date: Sat, 13 Nov 2001 23:29:00 GMT



22.18 Error-Info

   The Error-Info header field provides a pointer to additional
   information about the error status response.


        SIP UACs have user interface capabilities ranging from pop
        up windows and audio on PC softclients to audio-only on
        "black" phones or endpoints connected via gateways. Rather
        than forcing a server generating an error to choose between
        sending an error status code with a detailed reason phrase
        and playing an audio recording, the Error-Info header field
        allows both to be sent.  The UAC then has the choice of
        which error indicator to render to the caller.

   A UAC MAY treat a SIP URL in an Error-Info header field as if it were
   a Contact in a redirect and generate a new INVITE, resulting an a
   recorded announcement session being established. A non-SIP URL MAY be
   rendered to the user.

   Examples:

     SIP/2.0 404 The number you have dialed is not in service
     Error-Info: <sip:not-in-service-recording@atlanta.com>



22.19 Expires

   The Expires header field gives the date and time after which the
   message (or content) expires. The precise meaning of this is method
   dependent.

   Note that the expiration time in an INVITE 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



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   limits on the session duration, however.

   The value of this field can be either a date (see the Date header
   field) 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.

   Examples:

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



22.20 From

   The From header field indicates the initiator of the request. (Note
   that this may be different from the initiator of the dialog. Requests
   sent by the callee to the caller use the callee's address in the From
   header field.)

   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.

   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 7.3.1.

   The short form of the header is f.

   Examples:


     From: "A. G. Bell" <sip:agb@bell-telephone.com> ;tag=a48s
     From: sip:+12125551212@server.phone2net.com;tag=887s
     f: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8



22.21 In-Reply-To

   The In-Reply-To header field enumerates the Call-IDs that this call
   references or returns. These Call-IDs may have been cached by the
   client then included in this header in a return call.





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        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.

   Example:

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



22.22 Max-Forwards

   The Max-Forwards 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.

   The Max-Forwards value is a decimal integer indicating the remaining
   number of times this request message is allowed to be forwarded. This
   count is decremented by each server that forwards the request.

   Example:

     Max-Forwards: 6



22.23 MIME-Version

   See [H19.4.1].

   Example:

     MIME-Version: 1.0



22.24 Organization

   The Organization header field conveys the name of the organization to
   which the entity issuing the request or response belongs.


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




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   Example:

     Organization: Boxes by Bob



22.25 Priority

   The Priority header field indicates the urgency of the request as
   perceived by the client. Defined values include "non-urgent",
   "normal", "urgent", and "emergency".

   It is RECOMMENDED that the value of "emergency" only be used when
   life, limb or property are in imminent danger. Otherwise, there are
   no semantics defined for this header field.


        These are the values of RFC 2076 [37], with the addition of
        "emergency".

   Examples:


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


   or

     Subject: Weekend plans
     Priority: non-urgent



22.26 Proxy-Authenticate

   The Proxy-Authenticate header field consists of a challenge that
   indicates the authentication scheme and parameters applicable to the
   proxy for this Request-URI.

   The syntax for this header and use is defined in [H14.33]. See 20.2.3
   for further details on its usage.  Example:


      Proxy-Authenticate: Digest realm="Carrier SIP",
       domain="sip:ss1.carrier.com",
       nonce="f84f1cec41e6cbe5aea9c8e88d359",
       opaque="", stale=FALSE, algorithm=MD5



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22.27 Proxy-Authorization

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

   See [H14.34] for a definition of the syntax, and section 20.2.3 for a
   discussion of its usage.

   Note that this header field, along with Authorization breaks the
   general rules about multiple header fields. Although not a comma-
   separated list, this header field may be present multiple times, and
   MUST NOT be combined into a single header using the usual rules
   described in Section 7.3.1.

   Example:


   Proxy-Authorization: Digest username="Alice", realm="Atlanta ISP",
      nonce="c60f3082ee1212b402a21831ae",
      response="245f23415f11432b3434341c022"



22.28 Proxy-Require

   The Proxy-Require header field is used to indicate proxy-sensitive
   features that must be supported by the proxy.  See Section 22.30 for
   more details on the mechanics of this message and a usage example.

   Example:

     Proxy-Require: foo



22.29 Record-Route

   The Record-Route is inserted by proxies in a request to force future
   requests in the session to route through the proxy.

   Details of its use with the Route header field are described in
   Section 16.4.

   Example:




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     Record-Route: <sip:bob@biloxi.com;maddr=10.1.1.1>,
      <sip:bob@biloxi.com;maddr=10.2.1.1>



22.30 Require

   The Require 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. Although an optional
   header, the Require MUST NOT be ignored if it is present.


        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.

   Example:

     Require: com.example.billing



22.31 Retry-After

   The Retry-After 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.

   Examples:




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



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

22.32 Route

   The Route is used to force routing for a request through the listed
   set of proxies. Details of its use with the Record-Route header field
   are described in Section 13.

   Example:

     Route: <sip:bob@biloxi.com;maddr=10.1.1.1>, <sip:bob@10.4.1.4>



22.33 Server

   The Server 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].

   Example:

     Server: HomeProxy v2



22.34 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.)

   The short form of the header is s.

   Example:

     Subject: Need more boxes
     s: Tech Support



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22.35 Supported

   The Supported header field enumerates all the extensions upported by
   the client or server. If empty, it means that no extensions are
   supported.

   Example:

     Supported: foo, bar



22.36 Timestamp

   The Timestamp header field describes when the client sent the request
   to the server. The use of the Timestamp is covered in Section 13.

   Example:

     Timestamp: 54



22.37 To

   The To header field specifies the logical recipient of the request.

   The optional "display-name" is meant to be rendered by a human-user
   interface. The "tag" parameter serves as a general mechanism to
   distinguish multiple instances of a user identified by a single SIP
   URL.

   See Section 13 for details of the "tag" parameter.

   Section 22.20 describes how To and From header fields are compared
   for the purpose of matching requests to dialogs.  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 short form of the header is t.

   The following are examples of valid To headers:

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




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22.38 Unsupported

   The Unsupported header field lists the features not supported by the
   server. See Section 22.30 for a usage example and motivation.

   Example:

     Unsupported: foo



22.39 User-Agent

   The User-Agent header field contains information about the client
   user agent originating the request. The syntax and semantics are
   defined in [H14.43].

   Example:

     User-Agent: Softphone Beta1.5



22.40 Via

   The Via field indicates the path taken by the request so far and
   indicate the path that should be followed in routing responses.

   The Via header field contains 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. The Via header field can also contains parameters such as
   "maddr", "ttl", "received", and "branch"whose meaning and use are
   described in other sections.

   The short form of the header is v.

   Example:


     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-



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   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.

   Another example:

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



22.41 Warning

   The Warning header field is used to carry additional information
   about the status of a response. Warning headers are sent with
   responses and contain a three digit warning code, host name, and
   warning text.

   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 [38].

   The first digit of warning codes beginning with "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.




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

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

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

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

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

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

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

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

        370 Insufficient bandwidth: The bandwidth specified in the
             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:





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



22.42 WWW-Authenticate

   The WWW-Authenticate header field consists of a challenge that
   indicates the authentication scheme and parameters applicable for
   this Request-URI.

   The syntax for this header and use is defined in [H14.47]. See 20.2.2
   for further details on its usage.

   Example:

     WWW-Authenticate: Digest realm="Bob's Friends",
       domain="sip:boxesbybob.com",
       nonce="f84f1cec41e6cbe5aea9c8e88d359",
       opaque="", stale=FALSE, algorithm=MD5



23 Response Codes

   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.

23.1 Provisional 1xx

   Provisional responses indicate that the server or proxy contacted is
   performing some further action and does not yet have a definitive
   response. A server typically sends a 1xx response if it expects to
   takemore than 200 ms to obtain a final response. Note that 1xx
   responses are not transmitted reliably, that is, they do not cause
   the client to send an ACK.

   Provisional (1xx) responses MAY contain message bodies, including
   session descriptions.

   Provisional responses are also known as informational responses.

23.1.1 100 Trying



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   This response indicates that the request has been received by the
   next hop server and that some unspecified action is being taken on
   behalf of this call (e.g., a database is being consulted). This
   response stops retransmissions of an INVITE by a UAC.

23.1.2 180 Ringing

   The user agent receiving the INVITE is trying to alert the user. This
   response MAY be used to initiate local ringback.

23.1.3 181 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.

23.1.4 182 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.

23.1.5 183 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, header fields, or message body MAY be used to convey
   more details about the call progress.

23.2 Successful 2xx

   The request was successful.

23.2.1 200 OK

   The request has succeeded. The information returned with the response
   depends on the method used in the request.

23.3 Redirection 3xx

   3xx responses give information about the user's new location, or
   about alternative services that might be able to satisfy the call.

23.3.1 300 Multiple Choices




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   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 MAY include a message body 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 choices SHOULD also be listed as Contact fields (Section 22.10).
   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.

23.3.2 301 Moved Permanently

   The user can no longer be found at the address in the Request-URI and
   the requesting client SHOULD retry at the new address given by the
   Contact header field (Section 22.10). 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.

23.3.3 302 Moved Temporarily

   The requesting client SHOULD retry the request at the new address(es)
   given by the Contact header field (Section 22.10).  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. In the second approach, UAs MAY also use the Contact
   information for the To header field, as well as a new Call-ID value.

   The duration of the redirection can be indicated through an Expires
   (Section 22.19) header. If there is no explicit expiration time, the
   address is only valid for this call and MUST NOT be cached for future
   calls.

23.3.4 305 Use Proxy



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   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.

23.3.5 380 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.

23.4 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.

23.4.1 400 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".

23.4.2 401 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.

23.4.3 402 Payment Required

   Reserved for future use.

23.4.4 403 Forbidden

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

23.4.5 404 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.

23.4.6 405 Method Not Allowed



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   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.

23.4.7 406 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.

23.4.8 407 Proxy Authentication Required

   This code is similar to 401 (Unauthorized), but indicates that the
   client MUST first authenticate itself with the proxy. SIP access
   authentication is explained in section 20 and 20.2.3.

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

23.4.9 408 Request Timeout

   The server could not produce a response within a suitable amount of
   time, for example, if it could not determine the location of the user
   in time. The client MAY repeat the request without modifications at
   any later time.

23.4.10 410 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.

23.4.11 413 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.

23.4.12 414 Request-URI Too Long



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

23.4.13 415 Unsupported Media Type

   The server is refusing to service the request because the message
   body of the request is in a format not supported by the server for
   the requested method. The server SHOULD return a list of acceptable
   formats using the Accept, Accept-Encoding and Accept-Language header
   fields. UAC processing of this response is described in Section
   8.1.3.4.

23.4.14 420 Bad Extension

   The server did not understand the protocol extension specified in a
   Proxy-Require (Section 22.28) or Require (Section 22.30) header
   field.  The server SHOULD include a list of the unsupported
   extensions in an Unsupported header in the response. UAC processing
   of this response is described in Section 8.1.3.4.

23.4.15 421 Extension Required

   The UAS needs a particular extension to process the request, but this
   extension is not listed in a Supported header in the request.
   Responses with this status code MUST contain a Require header listing
   the required extensions.

   In general, a UAS SHOULD NOT use this response when it wishes to
   apply an extension to a request. The end result will often be no
   service at all, and a break in interoperability. Rather, servers
   SHOULD process the request using baseline SIP capabilities and any
   extensions supported by the client.

23.4.16 480 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). 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



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   valid forwarding location for that user.

23.4.17 481 Call/Transaction Does Not Exist

   This status indicates that the UAS received a request that does not
   match any existing dialog or transaction.

23.4.18 482 Loop Detected

   The server has detected a loop (Section 3).

23.4.19 483 Too Many Hops

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

23.4.20 484 Address Incomplete

   The server received a request with a 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.

23.4.21 485 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>






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        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.

23.4.22 486 Busy Here

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

23.4.23 487 Request Terminated

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

23.4.24 488 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.

23.5 Server Failure 5xx

   5xx responses are failure responses given when a server itself has
   erred.

23.5.1 500 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.

23.5.2 501 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



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   any user. (Proxies forward all requests regardless of method.)

23.5.3 502 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.

23.5.4 503 Service Unavailable

   The server is currently unable to handle the request due to a
   temporary overloading (i.e., congestion) 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.

   A client (proxy or UAC) receiving a 503 SHOULD attempt to forward the
   request to an alternate server. It SHOULD NOT forward any other
   requests to that server for the duration specified in the Retry-After
   header, if present.

   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.

23.5.5 504 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.

23.5.6 505 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.

23.5.7 513 Message Too Large




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

23.6 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.

23.6.1 600 Busy Everywhere

   The callee's end system was contacted successfully but the callee is
   busy and does not wish to take the call at this time. The response
   MAY indicate a better time to call in the Retry-After header. If the
   callee does not wish to reveal the reason for declining the call, the
   callee uses status code 603 (Decline) instead. This status response
   is returned only if the client knows that no other end point (such as
   a voice mail system) will answer the request. Otherwise, 486 (Busy
   Here) should be returned.

23.6.2 603 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.

23.6.3 604 Does Not Exist Anywhere

   The server has authoritative information that the user indicated in
   the Request-URI does not exist anywhere.

23.6.4 606 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 22.41.  It is hoped that
   negotiation will not frequently be needed, and when a new user is
   being invited to join an already existing conference, negotiation may
   not be possible. It is up to the invitation initiator to decide
   whether or not to act on a 606 (Not Acceptable) response.

24 Locating a SIP Server



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        NOTE: Usage of SRV records is still under discussion with
        IESG, and therefore this section is likely to change in
        subsequent versions of bis.

   The SIP URI provides a way to identify a communications resource. For
   this URI to be useful in a SIP element, a mechanism is necessary to
   take this URI and determine the IP address, port, and transport of
   one or more servers that message destined for this URI should be sent
   to. We refer to the combination of an IP address, port, and transport
   as a next hop next hop can be configured to be the same for all URIs.
   In this case, the next hop is referred to as a outbound proxy
   commonly used in a user agent which is required to send all requests
   to a specific server for policy processing or firewall traversal, for
   example. The outbound proxy can be configured by any mechanism,
   including DHCP [39].

   When the next hop is not configured, a mechanism is needed to
   determine one or more next hops from the URI. Section 24.1 provides
   an algorithm which can be used to determine an ordered list of next
   hops. Typically, the URI that is used is from the Request-URI of a
   request, in order to determine where to send that request. However,
   in certain circumstances (which are documented in Section 19.2.2), a
   URI may have been extracted from a response in order to determine
   where to send the response.

   Once the ordered list of next hops is computed, they are used
   according to the procedures of Section 24.2.

24.1 Computing the List of Next Hops

   The algorithm for computing the list of next hops begins by setting
   three variables. The first variable is called the target address
   maddr parameter of the URI, if present. If not present, it MUST be
   set to the host element of the URI. The next variable is called the
   target port set to the port element of the URI if present, else the
   target port MUST remain empty. The target transport MUST be set to
   the headertransport element of the URI if present, else the target
   transport MUST remain empty.

   The algorithm begins by examining the target address. If it contains
   a numeric IP address, the procedures of Section 24.1.1 MUST be
   followed. Otherwise, the target transport is examined. If it is
   empty, and the target port is either empty or contains a value of
   5060, the procedures of Section 24.1.2 MUST be followed. If the
   target transport is not empty, and the target port is empty, the
   procedures of Section 24.1.2 MUST be followed if the target transport
   is UDP. If the target transport and target port are not empty, but
   the target port contains the default port for the target transport



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   (5060 for UDP, TCP, and SCTP, 5061 for TLS), the procedures of
   Section 24.1.2 MUST also be followed. Otherwise, the procedures of
   Section 24.1.3 MUST be followed. Effectively, this case occurs when
   the target port and target transport don't "match", taking into
   account their defaults if empty.

24.1.1 Numeric Destination Address

   The addresses of the next hops are all the same, and MUST be equal to
   the value of the target address.

   If the target transport is specified, and the element supports that
   transport, there is only a single next hop, using the target
   transport. If the target transport is not specified, the number of
   next hops is equal to the number of transports the element supports.
   The first next hop MUST be UDP, and the ordering of the remaining
   transports is at the discretion of the element.

   For each next hop, the port number is equal to the target port, if
   specified, otherwise the default port for that transport of that next
   hop.

   For example, consider the SIP URI sip:joe@1.2.3.4 present in the
   Request-URI of a request. A UAC wishes to use this URI to determine
   the set of next hops. The UAC supports UDP and TLS. It applies the
   algorithm in this section, and ends up with the following ordered
   list of IP address, port, transport:


   {1.2.3.4, 5060, UDP}
   {1.2.3.4, 5061, TLS}



24.1.2 SRV Resolution of Host Name

   DNS SRV records are retrieved according to RFC 2782 [40]. The service
   identifier for DNS SRV records is "_sip". If the target transport is
   not empty, only records for that transport are retrieved. (If the
   element does not support the transport specified, the lookup fails.)
   If the target transport is empty, the element retrieves records for
   all transport protocols it supports. The results of all queries are
   merged and then sorted according to priority, independent of the
   transport protocol. If this list is empty, follow the procedure in
   Section 24.1.3.

   Note that the behavior above differs slightly from that described in
   RFC 2782. There, A records are consulted if the query for one



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   transport protocol fails; here, we only abandon the SRV lookup if
   none of the transport protocols supported by the client yield an
   answer.

   Clients MUST NOT cache query results except according to the rules in
   RFC 1035 [41].

24.1.3 Address Record Resolution of Host Name

   When the target address is not a numeric IP, and there is a target
   port which does not match the default port for the target transport,
   SRV records are not used. This is because SRV will normally provide
   ports, so if one is provided that is not a default, this would seem
   to imply the the URL is trying to explicitly identify the
   destination, rather than using SRV.

   In this case, 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.

   The DNS address records are kept sorted in the order returned by the
   DNS server. For each address, the port is set to the target port. For
   each address, the transport is set to the target transport if not
   empty, otherwise, the target transport MUST be UDP for the first
   address, and is at the discretion of the implementation for the
   others.


        OPEN ISSUE #221: Selection of transports for the case when
        multiple A records are returned requires more work.

   Clients MUST NOT cache query results except according to the rules in
   RFC 1035 [41].

24.2 Contacting the Next Hops

   The algorithms of the previous section will result in an ordered list
   of next hops. This section describes how that list is used.

   If the ordered list was obtained through SRV, servers are contacted
   as specified in the "Usage rules" section of RFC 2782 [40], which
   describes procedures for using the weight field to randomly select
   servers amongst those of equal priority.

   The SIP element takes the ordered list, and it tries to contact each
   next hop in turn, until a server responds. If contacting a next hop
   results in a failure, as defined in the next paragraph, the element



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   moves to the next next hop in the list, until the list is exhausted.
   If the list is exhausted, then the element gives up.

   Failures SHOULD be detected through network failure indications or
   timeouts. If the element sending the message is a client sending a
   request using a client transaction, the client transaction will
   report any transport layer failures. If the element sending the
   message is a client sending a request directly to the transport
   layer, the transport layer will report any failures (See Section
   19.4). In either case, the client SHOULD try the next address. This
   will involve creating a new client transaction for it in the former
   case. The new request MUST have a new branch ID in the Via header.
   Note also that the new destination might be with a different
   transport, which might require a change in other parts of the Via
   header.

   Response failures are handled by the transport layer itself, which
   may retry the response to the next next hop. See Section 19.2.2.

   Failures can be detected through timeouts only if the element is a
   client sending a request through the client transaction. In that
   case, if a timeout is reported by the client transaction, the client
   SHOULD try the next next hop in the list.


        OPEN ISSUE #219: It might be easier to encapsulate the SRV
        processing in one place, at the transport layer, rather
        than the behavior being dependent on client v. server. This
        can only be done if merging of srv records across
        transports is deprecated, along with failures based on
        timeouts.

   Once a next hop is successfully contacted, that same next hop address
   MUST be used for all subsequent messages that share the same Call-ID.
   More specifically, once a request is delivered successfully to a
   particular next hop, all subsequent requests with the same Call-ID
   MUST be delivered to that next hop. Once a response is delivered
   successfully to a particular next hop, all subsequent responses with
   the same Call-ID MUST be delivered to that next hop. However, if that
   next hop fails, the selection algorithms MUST be re-run for the top.


        This is a change from RFC2543, which only used the same
        address for requests within a transaction. Broadening the
        scope to Call-ID helps, for example, ensure that requests
        with credentials after a challenge are delivered to the
        same server that issued the challenge.




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   A stateless proxy can accomplish this, for example, by using the
   modulo N of a hash of the Call-ID value as the uniform random number
   described in the weighting algorithm of RFC 2782 [40]. Here, N is the
   sum of weights within the priority class.


        OPEN ISSUE #220: This stateless selection algorithm doesn't
        work if there are failures.

25 Examples

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

25.1 Registration

   Bob registers on start-up. The message flow is shown in Figure 9.




biloxi.com         Bob's
 registrar       softphone
    |                |
    |   REGISTER F1  |
    |<---------------|
    |    200 OK F2   |
    |--------------->|



   Figure 9: SIP Registration Example





   F1 REGISTER Bob -> Registrar

     REGISTER sip:registrar.biloxi.com
     Via: SIP/2.0/UDP 10.4.1.4:5060
     To: Bob <sip:bob@biloxi.com>
     From: Bob <sip:bob@biloxi.com>;tag=456248
     Call-ID: 843817637684230@phone21.boxesbybob.com
     CSeq: 1826 REGISTER
     Contact: <sip:bob@10.4.1.4>
     Expires: 7200
     Contact-Length: 0



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   The registration expires after two hours. The registrar responds with
   a 200 OK:



   F2 200 OK Registrar -> Bob

     SIP/2.0 200 OK
     Via: SIP/2.0/UDP 10.4.1.4:5060
     To: Bob <sip:bob@biloxi.com>
     From: Bob <sip:bob@biloxi.com>;tag=456248
     Call-ID: 843817637684230@phone21.boxesbybob.com
     CSeq: 1826 REGISTER
     Contact: <sip:bob@10.4.1.4>
     Expires: 7200
     Contact-Length: 0




25.2 Session Setup

   This example contains the full details of the example session setup
   in Section 4. The message flow is shown in Figure 1.



   F1 INVITE Alice -> atlanta.com proxy

     INVITE sip:bob@biloxi.com SIP/2.0
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact: <sip:alice@10.1.3.3>
     Content-Type: application/sdp
     Contact-Length: 142

     (Alice's SDP not shown)





   F2 100 Trying atlanta.com proxy -> Alice

     SIP/2.0 100 Trying



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     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact-Length: 0





   F3 INVITE atlanta.com proxy -> biloxi.com proxy

     INVITE sip:bob@biloxi.com SIP/2.0
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact: <sip:alice@10.1.3.3>
     Content-Type: application/sdp
     Contact-Length: 142

     (Alice's SDP not shown)





   F4 100 Trying biloxi.com proxy -> atlanta.com proxy

     SIP/2.0 100 Trying
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact-Length: 0





   F5 INVITE biloxi.com proxy -> Bob

     INVITE sip:bob@10.4.1.4 SIP/2.0



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     Via: SIP/2.0/UDP 10.2.1.1:5060;branch=4b43c2ff8.1
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact: <sip:alice@10.1.3.3>
     Content-Type: application/sdp
     Contact-Length: 142

     (Alice's SDP not shown)





   F6 180 Ringing Bob -> biloxi.com proxy

     SIP/2.0 180 Ringing
     Via: SIP/2.0/UDP 10.2.1.1:5060;branch=4b43c2ff8.1
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact-Length: 0





   F7 180 Ringing biloxi.com proxy -> atlanta.com proxy

     SIP/2.0 180 Ringing
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact-Length: 0








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   F8 180 Ringing atlanta.com proxy -> Alice

     SIP/2.0 180 Ringing
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact-Length: 0





   F9 200 OK Bob -> biloxi.com proxy

     SIP/2.0 200 OK
     Via: SIP/2.0/UDP 10.2.1.1:5060;branch=4b43c2ff8.1
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact: <sip:bob@10.4.1.4>
     Content-Type: application/sdp
     Contact-Length: 131

     (Bob's SDP not shown)





   F10 200 OK biloxi.com proxy -> atlanta.com proxy

     SIP/2.0 200 OK
     Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact: <sip:bob@10.4.1.4>
     Content-Type: application/sdp
     Contact-Length: 131

     (Bob's SDP not shown)



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   F11 200 OK atlanta.com proxy -> Alice

     SIP/2.0 200 OK
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 INVITE
     Contact: <sip:bob@10.4.1.4>
     Content-Type: application/sdp
     Contact-Length: 131

     (Bob's SDP not shown)





   F12 ACK Alice -> Bob

     ACK sip:bob@10.4.1.4 SIP/2.0
     Via: SIP/2.0/UDP 10.1.3.3:5060
     To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     From: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 314159 ACK
     Contact-Length: 0



   The media session between Alice and Bob is now established.

   Bob hangs up first. Note that Bob's SIP phone maintains its own CSeq
   numbering space, which, in this example, begins with 231. Also not
   that since Bob is making the request, the To and From URLs and tags
   have been swapped.



   F13 BYE Bob -> Alice

     BYE sip:alice@10.1.3.3 SIP/2.0
     Via: SIP/2.0/UDP 10.4.1.4:5060
     From: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     To: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 231 BYE
     Contact-Length: 0



Various Authors                                             [Page 170]

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   F14 200 OK Alice -> Bob

     SIP/2.0 200 OK
     Via: SIP/2.0/UDP 10.4.1.4:5060
     From: Bob <sip:bob@biloxi.com>;tag=a6c85cf
     To: Alice <sip:alice@atlanta.com>;tag=1928301774
     Call-ID: a84b4c76e66710@10.1.3.3
     CSeq: 231 BYE
     Contact-Length: 0



   The SIP Call Flows document [42] contains further examples of SIP
   messages.

   ;; This buffer is for notes you don't want to save, and for Lisp
   evaluation.  ;; If you want to create a file, first visit that file
   with C-x C-f, ;; then enter the text in that file's own buffer.

26  Augmented BNF for the SIP Protocol

   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 [12] 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:



        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.



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   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]


   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




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   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.

26.1 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"



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        alphanum  =  alpha | DIGIT
        CTL       =  %x00-1f | %x7f ; (octets 0 -- 31) and DEL (127)
        CR        =  %d13 ; US-ASCII CR, carriage return character
        LF        =  %d10 ; US-ASCII LF, line feed character
        SP        =  %d32 ; US-ASCII SP, space character
        HT        =  %d09 ; US-ASCII HT, horizontal tab character
        CRLF      =  CR LF ; typically the end of a line


   The following are defined in RFC 2396 [9] for the SIP URI:


        unreserved  =  alphanum | mark
        mark        =  "-" | "_" | "." | "!" | "~" | "*" | "'"
                   |   "(" | ")"
        escaped     =  "%" hex hex


   SIP header field values can be folded onto multiple lines if the
   continuation line begins with a space or horizontal tab. All linear
   white space, including folding, has the same semantics as SP. A
   recipient MAY replace any linear white space with a single SP before
   interpreting the field value or forwarding the message downstream.
   This is intended to behave exactly as HTTP 1.1 as described in
   RFC2615 [8].



        LWS  =  *( SP | HT ) [CRLF] 1*( SP | HT ) ; linear whitespace


   To separate the header name from the rest of value, a colon is used,
   which, by the above rule allows whitespace before, but no line break,
   and whitespace after, including a linebreak. The HCOLON defines this
   construct.



        HCOLON  =  *( SP | HT ) ":" LWS


   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 [11]). 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.



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        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.
   Some elements (authentication) force hex alphas to be lower case.


        LHEX  =  digit | "a" | "b" | "c" | "d" | "e" | "f"


   Others allow mixed upped and lower case


        hex  =  LHEX | "A" | "B" | "C" | "D" | "E" | "F"


   Many SIP header field values consist of words separated by LWS or
   special characters. Unless otherwise stated, tokens are case-
   insensitive. These special characters MUST be in a quoted string to
   be used within a parameter value.



        token                            =  1*(alphanum | "-" | "." | "!" | "%" | "*"
        | "_" | "+" | "`" | "'" | "~" )
        separators                       =  "(" | ")" | "<" | ">" | "@" |
                                            "," | ";" | ":" | "\" | <"> |
                                            "/" | "[" | "]" | "?" | "=" |
                                            "{" | "}" | SP | HT


   When tokens are used or separators are used between elements,
   whitespace is often allowed before or after these characters:




Various Authors                                             [Page 175]

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        MINUS    =  LWS "-" LWS ; minus
        DOT      =  LWS "." LWS ; period
        PERCENT  =  LWS "%" LWS ; percent
        BANG     =  LWS "!" LWS ; exclamation
        PLUS     =  LWS "+" LWS ; plus
        STAR     =  LWS "*" LWS ; askerisk
        TILDE    =  LWS " LWS ; tilde
        EQUAL    =  LWS "=" LWS ; equal
        LPAREN   =  LWS "(" LWS ; left parenthesis
        RPAREN   =  LWS ")" LWS ; right parenthesis
        LANGLE   =  LWS "<" LWS ; left angle bracket
        RAQUOT   =  ">" LWS ; right angle quote
        LAQUOT   =  LWS "<"; left angle quote
        RANGLE   =  LWS ">" LWS ; right angle bracket
        BAR      =  LWS "|" LWS ; vertical bar
        ATSIGN   =  LWS "@" LWS ; atsign
        COMMA    =  LWS "," LWS ; comma
        SEMI     =  LWS ";" LWS ; semicolon
        COLON    =  LWS ":" LWS ; colon
        DQUOT    =  LWS <"> LWS ; double quotation mark
        LDQUOT   =  LWS <">; open double quotation mark
        RDQUOT   =  <"> LWS ; close double quotation mark
        LBRACK   =  LWS "{" LWS ; left square bracket
        RBRACK   =  LWS "}" LWS ; right square bracket


   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  =  LPAREN *(ctext | quoted-pair | comment) RPAREN
        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  =  ( LWS <"> *(qdtext | quoted-pair ) <"> )
        qdtext         =  LWS | %x21 | %x23-5b | %x5d-7e
                          |                                         UTF8-NONASCII





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   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)




        SIP-URL          =  "sip:" [ userinfo "@" ] hostport
                            url-parameters [ headers ]
        userinfo         =  [ user | telephone-subscriber [ ":" password ]]
        user             =  *( unreserved | escaped | user-unreserved )
        user-unreserved  =  "                                               " | "=" | "+" | "$" | "," | ";" | "?" | "/"
        password         =  *( unreserved | escaped |
                            "                                               " | "=" | "+" | "$" | "," )
        hostport         =  host [ ":" port ]
        host             =  hostname | IPv4address | IPv6reference
        hostname         =  *( domainlabel "." ) toplabel [ "." ]
        domainlabel      =  alphanum
                            | alphanum *( alphanum | "-" ) alphanum
        toplabel         =  alpha | alpha *( alphanum | "-" ) alphanum




        IPv4address    =  1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
        IPv6reference  =  "[" IPv6address "]"
        IPv6address    =  hexpart [ ":" IPv4address ]
        hexpart        =  hexseq | hexseq "::" [ hexseq ] | "::" [ hexseq ]
        hexseq         =  hex4 *( ":" hex4)
        hex4           =  1*4HEX
        port           =  1*DIGIT




        url-parameters    =  *( ";" url-parameter)
        url-parameter     =  transport-param | user-param | method-param
                             |ttl-param | maddr-param | other-param
        transport-param   =  "transport="
                             ( "udp" | "tcp" | "sctp" | "tls"
                             | other-transport)
        other-transport   =  token
        user-param        =  "user=" ( "phone" | "ip" | other-user)



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        other-user        =  token
        method-param      =  "method=" Method
        ttl-param         =  "ttl=" ttl
        maddr-param       =  "maddr=" host
        other-param       =  pname [ "=" pvalue ]
        pname             =  1*paramchar
        pvalue            =  1*paramchar
        paramchar         =  param-unreserved | unreserved | escaped
        param-unreserved  =  "[" | "]" | "/" | ":" | "                   " | "+" | "$"




        headers         =  "?" header *( "                             " header )
        header          =  hname "=" hvalue
        hname           =  1*( hnv-unreserved | unreserved | escaped )
        hvalue          =  *( hnv-unreserved | unreserved | escaped )
        hnv-unreserved  =  "[" | "]" | "/" | "?" | ":" | "+" | "$"




        SIP-message   =  Request | Response
        Request       =  Request-Line
                         *( message-header )
                         CRLF
                         [ message-body ]
        Request-Line  =  Method SP Request-URI SP SIP-Version CRLF
        Request-URI   =  SIP-URL | absoluteURI
        SIP-Version   =  "SIP/2.0"




        message-header
                         =  Accept
                         |  Accept-Encoding
                         |  Accept-Language
                         |  Alert-Info
                         |  Allow
                         |  Authentication-Info
                         |  Authorization
                         |  Call-ID
                         |  Call-Info
                         |  Contact
                         |  Content-Disposition
                         |  Content-Encoding
                         |  Content-Language



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                         |  Content-Length
                         |  Content-Type
                         |  CSeq
                         |  Date
                         |  Error-Info
                         |  Expires
                         |  From
                         |  In-Reply-To
                         |  Max-Forwards
                         |  MIME-Version
                         |  Organization
                         |  Priority
                         |  Proxy-Authenticate
                         |  Proxy-Authorization
                         |  Proxy-Require
                         |  Record-Route
                         |  Require
                         |  Retry-After
                         |  Route
                         |  Server
                         |  Subject
                         |  Supported
                         |  Timestamp
                         |  To
                         |  Unsupported
                         |  User-Agent
                         |  Via
                         |  Warning
                         |  WWW-Authenticate




        Method            =  "INVITE" | "ACK" | "OPTIONS" | "BYE"
                             | "CANCEL" | "REGISTER" | extension-method
        extension-method  =  token
        option-tag        =  token
        Response
                          =  Status-Line
                             *( message-header )
                             CRLF
                             [ message-body ]




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



Various Authors                                             [Page 179]

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                        =  Informational
                       |   Redirection
                       |   Success
                       |   Client-Error
                       |   Server-Error
                       |   Global-Failure
                       |   extension-code
        extension-code  =  3DIGIT




        Reason-Phrase
                        =  *<TEXT-UTF8,  excluding CR, LF>
        Informational
                        =  "100"                            ;  Trying
                       |   "180"                            ;  Ringing
                       |   "181"                            ;  Call Is Being Forwarded
                       |   "182"                            ;  Queued
                       |   "183"                            ;  Session Progress




        Success  =  "200"  ;  OK




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




        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



Various Authors                                             [Page 180]

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                     |   "410"  ;  Gone
                     |   "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




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




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




        Accept            =  "Accept" HCOLON
                             #( media-range [ accept-params ] )
        media-range       =  ( "*/*"
                             | ( type LWS "/" "*" LWS )
                             | ( type SLASH subtype )
                             ) *( SEMI parameter )
        accept-params     =  SEMI "q" EQUAL qvalue *( accept-extension )
        accept-extension  =  SEMI token [ EQUAL ( token | quoted-string ) ]







Various Authors                                             [Page 181]

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        Accept-Encoding  =  "Accept-Encoding" HCOLON
                            1#( codings [ SEMI "q" EQUAL qvalue ] LWS )
        codings          =  ( content-coding | "*" )
        content-coding   =  token
        qvalue           =  ( "0" [ "." 0*3DIGIT ] )
                            | ( "1" [ "." 0*3("0") ] )




        Accept-Language  =  "Accept-Language" HCOLON
                            1#( language-range [ SEMI "q" EQUAL qvalue ] )
        language-range   =  ( ( 1*8ALPHA *( MINUS 1*8ALPHA ) ) | "*" )




        Alert-Info     =  "Alert-Info" HCOLON #
                          ( LAQUOT URI RAQUOT *( COLON generic-param ))
        generic-param  =  token [ EQUAL ( token | host |
                          quoted-string ) ]




        Allow  =  "Allow" HCOLON 1#Method




        Authorization     =  "Authorization" HCOLON credentials
        credentials       =  LWS "Digest" digest-response
        digest-response   =  1#( username | realm | nonce | digest-uri
                             | dresponse | [ algorithm ] | [cnonce]
                             | [opaque] | [message-qop]
                             | [nonce-count] | [auth-param] )
        username          =  "username" EQUAL username-value
        username-value    =  quoted-string
        digest-uri        =  "uri" EQUAL digest-uri-value
        digest-uri-value  =  request-uri ; As specified by HTTP/1.1
        message-qop       =  "qop" EQUAL qop-value
        cnonce            =  "cnonce" EQUAL cnonce-value
        cnonce-value      =  nonce-value
        nonce-count       =  "nc" EQUAL nc-value
        dresponse         =  "response" EQUAL request-digest
        request-digest    =  LDQUOT 32LHEX RDQUOT





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        AuthenticationInfo  =  "Authentication-info" HCOLON 1#( digest | nextnonce )
        nextnonce           =  "nextnonce" EQUAL nonce-value
        callid              =  token [ ATSIGN token ]




        Call-ID  =  ( "Call-ID" | "i" ) HCOLON callid




        Call-Info   =  "Call-Info" HCOLON # ( LAQUOT URI RAQUOT
                       *( SEMI info-param) )
        info-param  =  "purpose" EQUAL ( "icon" | "info"
                       | "card" | token ) | generic-param


        Contact       =  ( "Contact" | "m" ) HCOLON
                         (STAR | (1# (( name-addr | addr-spec )
                         *( SEMI contact-params ))))
        name-addr     =  [ display-name ] LAQUOT addr-spec RAQUOT
        addr-spec     =  SIP-URL | URI
        display-name  =  LWS (*token | quoted-string)



   contact-params    =   "q"                         EQUAL  qvalue
                     |   "action"                    EQUAL  "proxy" | "redirect"
                     |   "expires"                   EQUAL  delta-seconds |
                         LDQUOT SIP-date RDQUOT
                     |   contact-extension
   contact-extension =   generic-param
   qvalue             =  ( "0" [ "." 0*3DIGIT ] )
                         | ( "1" [ "." 0*3("0") ] )



        delta-seconds  =  1*DIGIT




        Content-Disposition   =  "Content-Disposition" HCOLON
                                 disposition-type *( SEMI disposition-param )
        disposition-type      =  "render" | "session" | "icon" | "alert"
                                 | disp-extension-token
        disposition-param     =  "handling" EQUAL



Various Authors                                             [Page 183]

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




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




        Content-Language  =  "Content-Language" HCOLON 1#language-tag
        language-tag      =  primary-tag *( MINUS subtag )
        primary-tag       =  1*8ALPHA
        subtag            =  1*8ALPHA




        Content-Length  =  ( "Content-Length" | "l" ) HCOLON 1*DIGIT




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




        CSeq  =  "CSeq" HCOLON 1*DIGIT Method




        Date          =  "Date" HCOLON SIP-date
        SIP-date      =  rfc1123-date
        rfc1123-date  =  wkday COMMA SP date1 SP time SP "GMT"
        date1         =  2DIGIT SP month SP 4DIGIT
                         ; day month year (e.g., 02 Jun 1982)
        time          =  2DIGIT ":" 2DIGIT ":" 2DIGIT
                         ; 00:00:00 - 23:59:59
        wkday         =  "Mon" | "Tue" | "Wed"
                         | "Thu" | "Fri" | "Sat" | "Sun"
        month         =  "Jan" | "Feb" | "Mar" | "Apr"
                         | "May" | "Jun" | "Jul" | "Aug"



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                         | "Sep" | "Oct" | "Nov" | "Dec"




        Error-Info  =  "Error-Info" HCOLON #
                       ( LAQUOT URI RAQUOT
                       *( SEMI generic-param ))




        Expires     =  "Expires" HCOLON ( SIP-date | delta-seconds )
        From        =  ( "From" | "f" ) HCOLON
                       ( name-addr | addr-spec )
                       *( SEMI from-param )
        from-param  =  tag-param | generic-param
        tag-param   =  "tag" EQUAL token




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




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




        MIME-Version  =  "MIME-Version" HCOLON 1*DIGIT "." 1*DIGIT




        Organization  =  "Organization" HCOLON TEXT-UTF8-TRIM




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





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        Proxy-Authenticate  =  "Proxy-Authenticate" HCOLON 1#challenge
        challenge           =  LWS "Digest" digest-challenge
        digest-challenge    =  1#( realm | [ domain ] | nonce |
                               [ opaque ] | [ stale ] | [ algorithm ] |
                               [ qop-options ] | [auth-param] )
        realm               =  "realm" EQUALS realm-value
        realm-value         =  quoted-string
        domain              =  "domain" EQUAL LDQUOT URI
                               ( 1*SP URI ) RDQUOT
        URI                 =  absoluteURI | abs_path
        nonce               =  "nonce" EQUAL nonce-value
        nonce-value         =  quoted-string
        opaque              =  "opaque" EQUAL quoted-string
        stale               =  "stale" EQUAL ( "true" | "false" )
        algorithm           =  "algorithm" EQUAL ( "MD5" | "MD5-sess" |
                               token )
        qop-options         =  "qop" EQUAL LDQUOT 1#qop-value RDQUOT
        qop-value           =  "auth" | "auth-int" | token




        Proxy-Authorization  =  "Proxy-Authorization" HCOLON credentials




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




        Record-Route  =  "Record-Route" HCOLON 1#
                         ( name-addr *( SEMI rr-param ))
        rr-param      =  generic-param
        Require       =  "Require" HCOLON 1#option-tag




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






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        Route  =  "Route" HCOLON 1# ( name-addr
                  *( SEMI rr-param ))




        Server           =  "Server" HCOLON 1*( product | comment )
        product          =  token [SLASH product-version]
        product-version  =  token




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




        Supported  =  ( "Supported" | "k" ) HCOLON 0#option-tag




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




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




        Unsupported  =  "Unsupported" HCOLON 1#option-tag




        User-Agent  =  "User-Agent" HCOLON 1*( product | comment )




        Via               =  ( "Via" | "v" ) HCOLON



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                             1#( sent-protocol sent-by
                             *( SEMI via-params ) [ comment ] )
        via-params        =  via-hidden | via-ttl | via-maddr
                             | via-received | via-branch
                             | via-extension
        via-hidden        =  "hidden"
        via-ttl           =  "ttl" EQUAL ttl
        via-maddr         =  "maddr" EQUAL host
        via-received      =  "received" EQUAL host
        via-branch        =  "branch" EQUAL token
        via-extension     =  generic-param
        sent-protocol     =  protocol-name SLASH protocol-version
                             SLASH transport
        protocol-name     =  "SIP" | token
        protocol-version  =  token
        transport         =  "UDP" | "TCP" | "TLS" | "SCTP"
                             | other-transport
        sent-by           =  host [ COLON port ]
        ttl               =  1*3DIGIT                               ; 0 to 255




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




        WWW-Authenticate  =  "WWW-Authenticate" HCOLON challenge


27 IANA Considerations

   All new or experimental method names, header field names, and status
   codes used in SIP applications SHOULD be registered with IANA in
   order to prevent potential naming conflicts. It is RECOMMENDED that
   new "option- tag"s and "warn-code"s also be registered. Before IANA
   registration, new protcol elements SHOULD be characterized in an
   Internet- Draft or, preferably, an RFC.

   For Internet-Drafts, IANA is requested to make the draft available as



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   part of the registration database.

        By the time an RFC is published, colliding names may have
        already been implemented.

   When a registration for either a new header field, new method or new
   status code is created based on an Internet-Draft, and that
   Internet-Draft becomes an RFC, the person that performed the
   registration MUST notify IANA to change the registration to point to
   the RFC instead of the Internet-Draft.

   Registrations should be sent to iana@iana.org

27.1 Option Tags

   Option tags are used in headers such as Require, Supported, Proxy-
   Require and Unsupported in support of SIP compatibility mechanisms
   for extensions. For more on the use of option tags in these headers
   see Section 21.2. The option tag itself is a string that is
   associated with a particular SIP option (e.g. an extension) in order
   to identify the option in signaling between SIP endpoints.

   When registering a new SIP option with IANA, 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 Section 26) 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

        o Contact information (postal and email address)





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        This procedure has been borrowed from RTSP [4] and the RTP
        AVP [44].

27.2 Warn-Codes

   Warning codes provide information supplemental to the status code in
   SIP response messages when the failure of the transaction results
   from a Session Description Protocol (SDP, [6]). New "warn-code"
   values can be registered with IANA as they arise.

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

   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.

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

27.3 Header Field Names

   Header field names do not require working group or working group
   chair review prior to IANA registration, but SHOULD be documented in
   an RFC or Internet- Draft before IANA is consulted.

   The following information needs to be provided to IANA in order to
   register a new header field name:

        o The name and email address of the individual performing the
          registration.

        o The name of the header field being registered.

        o A compact form version for that header field, if one is
          defined.

        o The name of the draft or RFC where the header field is
          defined.

        o A copy of the draft or RFC where the header field is defined.

   Header fields SHOULD NOT use the X prefix notation and MUST NOT
   duplicate the names of header fields used by SMTP or HTTP unless the
   syntax is a compatible superset and the semantics are similar. Some



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   common and widely used header fields MAY be assigned one-letter
   compact forms (Section 7.3.3). Compact forms can only be assigned
   after SIP working group review. In the absence of this working group,
   a designated expert reviews the request.

27.4 Method and Response Codes

   Because the status code space is limited, they do require working
   group or working group chair review, and MUST be documented in an RFC
   or Internet draft. The same procedures apply to new method names.

   The following information needs to be provided to IANA in order to
   register a new response code or method:

        o The name and email address of the individual performing the
          registration.

        o The number of the response code or name of the method being
          registered.

        o The default reason phrase for that status code, if applicable.

        o The name of the draft or RFC where the method or status code
          is defined.

        o A copy of the draft or RFC where the method or status code is
          defined.

28 Changes Made in Version 00

        o Indicated that UAC should send both CANCEL and BYE after a
          retransmission fails.

        o Added semicolon and question mark to the list of unreserved
          characters for the user part of SIP URLs to handle tel: URLs
          properly.

        o Uniform handling of if hop count Max-Forwards: return 483.
          Note that this differs from HTTP/1.1 behavior, where only
          OPTIONS and TRACE allow this header, but respond as the final
          recipient when the value reaches zero.

        o Clarified that a forking proxy sends ACKs only for INVITE
          requests.

        o Clarified wording of DNS caching. Added paragraph on "negative
          caching", i.e., what to do if one of the hosts failed. It is
          probably not a good idea to simply drop this host from the



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          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 22.40.

        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.

        o Clarified that CANCEL and INVITE are separate transactions
          (Fig. 7). 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
          UAS roles are defined on a request-by-request basis. (Section
          6)




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        o In Section 22.40, noted that the maddr and received parameters
          also need to be encrypted when doing Via hiding.

        o Simplified Fig. 7 to only show INVITE transaction.

        o Added definition of the use of Contact (Section 22.10) for
          OPTIONS.

        o Added HTTP/RFC822 headers Content-Language and MIME-Version.

        o Added note in minimal section indicating that UAs need to
          support UDP.

        o Added explanation explaining what a UA should do when
          receiving an initial INVITE with a tag.

        o Clarified UA and proxy behavior for 302 responses.

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

        o Added response code 488 to indicate that there was no common
          media at the particular destination. (606 indicates such
          failure globally.)




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        o In Section 22.19, noted that registration updates can shorten
          the validity period.

        o Added note to enclose the URI for digest 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 22.10 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 SDP section, 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 22.16).

        o Added IPv6 literal addresses to the SIP URL definition,
          according to RFC 2732 [45].  Modified the IPv4 address to
          limit segments to at most three digits.

        o 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.

        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 and removed the SRV appendix. The
          only visible change is that protocol and service names are now
          prefixed by an underscore. Added wording that incorporates the
          precedence of maddr.

        o Allow parameters in Record-Route and Route headers.

        o In Table 1, list udp as the default value for the transport
          parameter in SIP URI.



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        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. 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. 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 2.
          It is mandatory.

        o Added: "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."




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        o Clarified: "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 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.

        o There were two incompatible BNFs for WWW-Authenticate.  One
          defined for PGP, and the other borrowed from HTTP. For basic
          or digest:


            WWW-Authenticate: basic realm="Wallyworld"



          and for pgp:


            WWW-Authenticate: pgp; realm="Wallyworld"



          The latter is incorrect and the semicolon has been removed.

        o Added rules for Route construction from called to calling UA.

        o We now allow Accept and Accept-Encoding in BYE and CANCEL
          requests. There is no particular reason not to allow them, as
          both requests could theoretically return responses,



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          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 2 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.

        o 1xx message bodies MAY now contain session descriptions.

        o Changed references to HTTP/1.1 and authentication to point to
          the latest RFCs.

        o Added 487 (Request terminated) status response. It is issued
          if the original request was terminated via CANCEL or BYE.

        o The spec was not clear on the identification of a call leg.
          Section 1.3 says it's the combination of To, From, and Call-
          ID. However, requests from the callee to the caller have the
          To and From reversed, so this definition is not quite
          accurate. Additionally, the "tag" field should be included in
          the definition of call leg. The spec now says that a call leg
          is defined as the combination of local-address, remote-
          address, and call-id, where these addresses include tags.

          Text was added to Section 6.21 to emphasize that the From and
          To headers designate the originator of the request, not that
          of the call leg.

        o All URI parameters, except method, are allowed in a Request-



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          URI. Consequently, also updated the description of which
          parameters are copied from 3xx responses in Sec. 22.10.

        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 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.

29 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 SDP section, noted that if the capabilities intersection is
          empty, a dummy format list still has to be returned due to SDP
          syntax constraints. Previously, the text had required that no
          formats be listed. (Brian Rosen)

        o Reorganized tables 2 and 3 to show proxy interaction with
          headers rather than "end-to-end" or "hop-by-hop".




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30 Changes Made in Version 02

        o Added "or UAS" in description of received headers in Section
          22.40. 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 22.14).

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

        o Added note to Section 22.17 indicating that registrars MUST
          include the Date header to accomodate UAs that do not have a
          notion of absolute time.

        o Added note emphasizing that non-SIP URIs are permissible in
          REGISTER.

        o Rewrote the server lookup section to be more precise and more
          like pseudo-code, with nesting instead of "gotos".

        o Removed note

             Note that the two URLs example.com and
             example.com:5060, while considered equal, may not lead
             to the same server, as the former causes a DNS SRV
             lookup, while the latter only uses the A record.



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             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
            recourse such as "don't route this request unless you can
            hide". It appears that almost all existing proxies simply
            ignore the Hide header.




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        o Added Error-Info header field.

31 Changes Made in Version 03

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

        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 22.16) 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).

        o Removed Also.

        o Added text to Require and Proxy-Require sections, making it a



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          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
          its acknowledging.

        o Added definitions for spiral, B2BUA.



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        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 definition 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.

32 Changes Made in Version 04

        o Removed Unsupported as a request header from Table 3.

        o Clarified SDP procedures for changing IP address and port.
          Specifically, spelled out the duration for which a UA needs to
          received media on the old port and address.

        o Added text in the SDP session which recommends that the
          answerer use the same ordering of codecs as used on the offer,
          in order to help ensure symmetric codec operation under normal



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          conditions.

        o Fixed bug in the example in the SDP section, where the new
          media line was listed at the top. Should have been the bottom.

        o Authorization credentials are cached based on the URL of the
          To header, not the entire To header as 10.48 implied.

        o Section 10.31, on Proxy-Authenticate, indicated that a server
          responds with a 401 if the client guessed wrong. This is
          incorrect. It should be 407.

        o Section 10.14, removed motivational text about Contact
          allowing an INVITE to be routed directly between end systems,
          since its confusing. Some have interpreted to mean that
          Record-Route is ignored when Contact is present.

        o Added reference to SCTP RFC.

        o Updated 2.2 to allow non-SIP URLs in OPTIONS and 2xx to
          OPTIONS.

        o Fixed example in 20.5. Added ACK for 487, and added To tag to
          487 response.

        o Clarified further URL comparisons. Its only URL parameters
          without defaults that are ignored if not present in both URLs.

        o Section 1.5.2, UDP mandatory for all. TCP is a SHOULD for UA,
          MUST for proxy, registrar, redirect servers.

        o Brought syntax for Contact, Via, and the SIP URL into
          alignment between the text and postscript versions.

        o Updated the text in section 6 which said that the ordering of
          header fields follows HTTP, with the exception of Via, where
          order matters. However, the HTTP spec says that order matters,
          so this sentence is redundant and confusing. The sentence was
          removed.

        o Added e lines to SDP examples in the Examples section.

        o Rewrote Allow discussion, more formally defining its semantics
          and usage cases.

        o Updated text on 604 status, to indicate that its based on the
          Request-URI, not the To.




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        o Added response registrations to IANA considerations. Provided
          more details on registration process.

        o Clarified that only a UAS rejects a request because the To tag
          doesn't match a local value.

        o Clarified that stateless proxies need to route based on static
          criteria only.

        o Proxy and UAC CANCEL generation upon 2xx, 6xx if it forked is
          now a SHOULD; used to be a MAY.

        o Added text saying that a UAS SHOULD send a BYE if it never
          gets an ACK for a 2xx establishing a call leg.

        o Added text saying that a UAS SHOULD send a re-INVITE if it
          never gets an ACK for a 2xx to a re-INVITE.

        o Added text on 503 processing, indicating that a client should
          try a different server when receiving a 503, and that a proxy
          shouldn't forward a 503 upstream unless it can't service any
          other requests.

        o Removed motivational text in Section 10.43 on Via headers
          since its not consistent with the text before it.

        o Changed IPSec reference to RFC2401, from RFC1825.

        o Updated retransmission defininition in 17.3.4 to be consistent
          with the rest of the spec.

        o Softened the language for insertion of the transport param in
          the record-route. Specifically, it can be inserted in private
          networks where it is known apriori that the specific transport
          is supported.

        o Updated definition of B2BUA.

        o Added text to section on 420 processing, which mandates that
          the client retry the request without extensions listed in the
          Unsupported header in the response.

        o Allow Authentication-Info header to be used for HTTP digest.

33 Changes Made in Version 05

        o Updated Table 2 to reflect that Error-Info is a response
          header in 3xx-6xx responses (it was previously listed as a



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          request header).

        o Removed WWW-Authenticate as a request header from Table 3.
          Authentication of responses is now done according to RFC2617.

        o Updated the Accept, Accept-Encoding and Accept-Language
          sections. More details on precise semantics for the various
          requests and responses is now provided. Presence of these
          headers is now a SHOULD for INVITE and 2xx to INVITE when a
          non-default value is present. Extra emphasis is placed on
          including the Accept-Language in INVITE and 2xx in order to
          support internationalization. Usage of these three headers in
          CANCEL has been removed since it makes no sense.

        o Generalized local outbound processing rules in Section 16.4.1
          to cover the case where the UAS is using a local outbound
          proxy which was not in the initial call setup path.

        o Updated record-routing section, so that a proxy can insert a
          transport param if it knows that the proxy on one side
          supports the specific transport (the previous text required
          the proxy to know whether the proxies on both sides supported
          the specific transport).

        o Added Authentication-Info to Section 10.

        o Clarified the meaning of Table 2 for responses.

        o Updated Table 1 to reflect that maddr is no longer mandatory
          in Record-Route.

        o Updated Table 3 so that header fields in responses to ACK are
          never listed as optional, mandatory, etc. - only not
          applicable. This is because responses to ACK are not allowed.
          Also improved wording in Section 5.1.1 to clarify that there
          MUST NOT be responses to ACK.

        o Updated SRV procedures. Old text said to treat a failure to
          contact a server as a 4xx, which would stop the SRV
          processing. But, this is not so. Sentence was stricken.

        o Updated 12.1 to clarify that 2xx INVITE responses MUST contain
          session descriptions.

        o Changed User-Agent to a request header in Table 3.

        o Updated SDP section, so that a UA cannot change the SDP when
          it gets a re-INVITE with no SDP.



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        o Clarified Appendix B that a unicast offer MUST have a unicast
          response.

        o Clarified that any request can be record-routed, but it may
          not be used by the UA, depending on the method.

        o non-2xx responses to INVITE no longer retransmitted over TCP.

        o Removed lower bound on T1 and T2 in private networks, which
          can use lower values. Furthermore, T1 can be smaller on the
          public Internet if proper RTT estimation is used.

        o UAS Cannot send a BYE for a call leg until it receives ACK, in
          order to eliminate a race condition between BYE and 200 OK.

        o Support of CR or LF alone as line terminators, as opposed to
          CRLF, is no longer required.

        o Client behavior on receipt of a 3xx to re-INVITE is now
          specified, and it is no longer forbidden to generate a 3xx.
          This is needed to maintain the idempotency of INVITE, as a
          proxy might redirect without knowing its a 3xx.

        o CANCEL cannot be sent before a 1xx is received, in order to
          eliminate race condition between request and CANCEL.

        o Termination of the client and server transactions is now based
          entirely on timeouts, rather than retransmission counters, in
          order to unify TCP and UDP behavior. Timeout values scale as a
          function of the RTT estimate, defined as T1. For reliable
          transports, many of these timers are now set to zero. Many
          timeouts differ than in bis-04.

        o Added a working RTT estimation algorithm using the Timestamp
          header, and specified it to be compliant to RFC 2988.

        o UAS accepting requests with unknown schemes in the URI in the
          To field is now a RECOMMENDED instead of SHOULD. This reflects
          the fact that processing a request when the To field doesn't
          match is a matter of policy.

        o Bodies are now allowed in any request and response, including
          CANCEL, although there may not be any semantics associated
          with that.

        o Supporting of INVITE without SDP is now a MUST (no strength
          was previously specified).




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        o Registration procedures for visiting, which had a few
          sentences in bis-04, have been removed. Roaming is a complex
          issue, and should be treated elsewhere.

        o Bis-04 mandated that a 2xx response to REGISTER contain
          expires Contact parameters indicating the expiration time of a
          contact. This behavior has now been made consistent with
          requests, so that the expiration time of a contact is the same
          in either case:  the expires param is used first if present,
          then the Expires header if present, else one hour for SIP
          URLs.

        o Action parameter in contact registrations is deprecated.

        o 2xx to REGISTER MUST contain current contacts. This was just a
          SHOULD in bis-04.

        o Multicast operation radically changed. Now, the treatment is
          no different than unicast. That is, only the first non-1xx
          response to a multicast request will be used. This is a
          natural consequence of the layering now applied to the
          protocol. This still enables anycast types of functions,
          mirroring the real usage of registrar discovery.

        o To completely separate transport rules from transaction rules,
          the rule in bis-04 that said a UAC SHOULD keep a connection
          opened until a response is received, has been turned into a
          timer recommendation. Specifically, the spec now says that it
          is RECOMMENDED that connections be kept opened for a minimum
          interval of sufficient duration to guarantee, with high
          probability, that responses are sent over the same connections
          as a request.

        o Re-use of existing connections for new requests to the same
          address and port is now RECOMMENDED, it was only a MAY in
          bis-04.

        o Modification of headers below the Authorization header by
          proxies is no longer disallowed, since the only mechanism that
          used Authorization in that way, PGP, has been deprecated
          previously.

        o Authentication of registrations now RECOMMENDED; no strength
          was defined previously.

        o Registering of new headers with IANA is now SHOULD; no
          strength was defined previously.




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        o Proxy aggregation of challenges now a SHOULD; no strength was
          defined previously.

        o Server support of basic authentication downgraded from SHOULD
          to MAY.

        o UAC resubmitting requests with credentials after a challenge
          upgraded from MAY to SHOULD.

        o TLS is now RECOMMENDED as the transport layer security for SIP
          signaling.

        o UA recursion on a redirect is now SHOULD; no strength was
          assigned previously.

        o UA reuse of headers in a recursed request is now SHOULD; no
          strength was assigned previously.

        o Security considerations added for Call-Info and Alert-Info.

        o Proxies no longer forward a 6xx immediately on receiving it.
          Instead, they CANCEL pending branches immediately. This avoids
          a potential race condition that would result in a UAC getting
          a 6xx followed by a 2xx. In all cases except this race
          condition, the result will be the same - the 6xx is forwarded
          upstream.

        o The term call-leg has been eliminated from the spec; a more
          generic term, dialog, is used in its place.

        o For SRV processing, subsequent requests with the same Call-ID
          (as opposed to the same transaction in bis-04) are sent to the
          same server.

        o SRV processing generalized to deal with the fact that the
          default port is transport dependent.

        o Per IESG request, draft-ietf-sip-serverfeatures has been
          integrated into bis.

        o Per IESG request, draft-ietf-sip-100rel will be integrated
          into bis. This is marked with a placeholder in this draft.

        o The BNF has been converted from implicit LWS to explicit LWS.

        o Caching of responses in a proxy to avoid redoing location
          server lookups used to be a SHOULD. Caching behavior for
          responses is now fully encapsulated in the transaction



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          processing.

        o Proxy usage of SRV in processing Route headers upgraded from
          SHOULD to MUST.

34 Acknowledgments

   We wish to thank the members of the IETF MMUSIC and SIP WGs for their
   comments and suggestions. Detailed comments were provided by Brian
   Bidulock, Jim Buller, Neil Deason, Dave Devanathan, Cédric Fluckiger,
   Yaron Goland, Bernie Höneisen, Phil Hoffer, Christian Huitema, Jean
   Jervis, Gadi Karmi, Peter Kjellerstedt, Anders Kristensen, Jonathan
   Lennox, Gethin Liddell, Keith Moore, Vern Paxson, Moshe J. Sambol,
   Chip Sharp, Igor Slepchin, Robert Sparks, Eric Tremblay., and Rick
   Workman.

   Brian Rosen provided the compiled BNF.

   This work is based, inter alia, on [46,47].

35 Authors' Addresses

   Authors addresses are listed alphabetically for the editors, the
   writers, and then the original authors of RFC 2543.

   Jonathan Rosenberg
   dynamicsoft
   72 Eagle Rock Ave
   East Hanover, NJ 07936
   USA
   electronic mail:  jdrosen@dynamicsoft.com

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

   Gonzalo Camarillo
   Ericsson
   Advanced Signalling Research Lab.
   FIN-02420 Jorvas
   Finland
   electronic mail:  Gonzalo.Camarillo@ericsson.com

   Alan Johnston



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   WorldCom
   100 South 4th Street
   St. Louis, MO 63102
   USA
   electronic mail:  alan.johnston@wcom.com

   Jon Peterson
   NeuStar, Inc
   1800 Sutter Street, Suite 570
   Concord, CA 94520
   USA
   electronic mail:  jon.peterson@neustar.com

   Robert Sparks
   dynamicsoft, Inc.
   5100 Tennyson Parkway
   Suite 1200
   Plano, Texas 75024
   USA
   electronic mail:  rsparks@dynamicsoft.com

   Mark Handley
   ACIRI
   electronic mail:  mjh@aciri.org

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

36 Bibliography

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   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
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   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.



Various Authors                                             [Page 215]

Internet Draft                    SIP                   October 26, 2001


   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
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.




                           Table of Contents



   1          Introduction ........................................    1
   2          Overview of SIP Functionality .......................    2
   3          Terminology .........................................    3
   4          Overview of Operation ...............................    3
   5          Structure of the Protocol ...........................   11
   6          Definitions .........................................   13
   7          SIP Messages ........................................   18
   7.1        Requests ............................................   18
   7.2        Responses ...........................................   19
   7.3        Header Fields .......................................   20
   7.3.1      Header Field Format .................................   20
   7.3.2      Header Field Classification .........................   23
   7.3.3      Compact Form ........................................   23
   7.4        Bodies ..............................................   23
   7.4.1      Message Body Type ...................................   23
   7.4.2      Message Body Length .................................   24
   7.5        Framing SIP messages ................................   24
   8          General User Agent Behavior .........................   24
   8.1        UAC Behavior ........................................   25
   8.1.1      Generating the Request ..............................   25
   8.1.1.1    To ..................................................   25
   8.1.1.2    From ................................................   26
   8.1.1.3    Call-ID .............................................   27
   8.1.1.4    CSeq ................................................   27
   8.1.1.5    Via .................................................   28
   8.1.1.6    Contact .............................................   28
   8.1.1.7    Request-URI .........................................   28
   8.1.1.8    Supported and Require ...............................   29
   8.1.1.9    Additional Message Components .......................   29
   8.1.2      Sending the Request .................................   29
   8.1.3      Processing Responses ................................   30
   8.1.3.1    Unrecognized Responses ..............................   30
   8.1.3.2    Vias ................................................   30
   8.1.3.3    Processing 3xx responses ............................   30



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   8.1.3.4    Processing 4xx responses ............................   30
   8.2        UAS Behavior ........................................   31
   8.2.1      Authentication/Authorization ........................   31
   8.2.2      Method Inspection ...................................   31
   8.2.3      Header Inspection ...................................   32
   8.2.3.1    To and Request-URI ..................................   32
   8.2.3.2    Require .............................................   33
   8.2.4      Content Processing ..................................   33
   8.2.5      Applying Extensions .................................   34
   8.2.6      Processing the Request ..............................   34
   8.2.7      Generating the Response .............................   34
   8.3        Redirect Servers ....................................   35
   9          Canceling a Request .................................   36
   9.1        Client Behavior .....................................   37
   9.2        Server Behavior .....................................   38
   10         Registrations .......................................   38
   10.1       Overview of Usage ...................................   38
   10.2       Construction of the REGISTER request ................   40
   10.2.1     Adding Bindings with REGISTER .......................   41
   10.2.1.1   Setting the Expiration Interval of Contact
   Addresses ......................................................   43
   10.2.1.2   Setting Preference among Contact Addresses ..........   43
   10.2.2     Removing Bindings with REGISTER .....................   43
   10.2.3     Fetching Bindings with REGISTER .....................   44
   10.2.4     Refreshing Registrations ............................   44
   10.2.5     Discovering a Registrar .............................   44
   10.3       Processing of REGISTER at the Registrar .............   45
   11         Querying for Capabilities ...........................   48
   11.1       Construction of OPTIONS Request .....................   48
   11.2       Processing of OPTIONS Request .......................   49
   12         Dialogs .............................................   50
   12.1       Creation of a Dialog ................................   51
   12.1.1     UAS .................................................   51
   12.1.2     UAC .................................................   53
   12.2       Requests within a Dialog ............................   54
   12.2.1     UAC Behavior ........................................   54
   12.2.1.1   Generating the Request ..............................   54
   12.2.1.2   Processing the Responses ............................   55
   12.2.2     UAS behavior ........................................   56
   12.3       Termination of a Dialog .............................   57
   13         Initiating a Session ................................   57
   13.1       Overview ............................................   57
   13.2       Caller Processing ...................................   58
   13.2.1     Creating the Initial INVITE .........................   58
   13.2.2     Processing INVITE Responses .........................   60
   13.2.2.1   1xx responses .......................................   60
   13.2.2.2   3xx responses .......................................   60
   13.2.2.3   4xx, 5xx and 6xx responses ..........................   61



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   13.2.2.4   2xx responses .......................................   61
   13.3       Callee Processing ...................................   62
   13.3.1     Processing of the INVITE ............................   62
   13.3.1.1   Progess .............................................   64
   13.3.1.2   The INVITE is redirected ............................   64
   13.3.1.3   The INVITE is rejected ..............................   64
   13.3.1.4   The INVITE is accepted ..............................   64
   14         Modifying an Existing Session .......................   65
   14.1       UAC Behavior ........................................   66
   14.2       UAS Behavior ........................................   66
   15         Terminating a Session ...............................   67
   15.1       Terminating a Dialog with a BYE .....................   68
   15.1.1     UAC Behavior ........................................   68
   15.1.2     UAS Behavior ........................................   69
   16         Proxy Behavior ......................................   69
   16.1       Overview ............................................   69
   16.2       Stateful Proxy ......................................   70
   16.3       Request Validation ..................................   72
   16.4       Making a Routing Decision ...........................   74
   16.5       Request Processing ..................................   76
   16.6       Response Processing .................................   82
   16.7       Handling transport errors ...........................   87
   16.8       CANCEL Processing ...................................   88
   16.9       Stateless proxy .....................................   88
   17         Transactions ........................................   89
   17.1       Client transaction ..................................   92
   17.1.1     INVITE Client Transaction ...........................   92
   17.1.1.1   Overview of INVITE Transaction ......................   92
   17.1.1.2   Formal Description ..................................   93
   17.1.1.3   Construction of the ACK Request .....................   96
   17.1.2     non-INVITE Client Transaction .......................   97
   17.1.2.1   Overview of the non-INVITE Transaction ..............   97
   17.1.2.2   Formal Description ..................................   97
   17.1.3     Matching Responses to Client Transactions ...........   98
   17.1.4     Handling Transport Errors ...........................  100
   17.2       Server Transaction ..................................  100
   17.2.1     INVITE Server Transaction ...........................  100
   17.2.2     non-INVITE Server Transaction .......................  103
   17.2.3     Matching Requests to Server Transactions ............  104
   17.3       RTT Estimation ......................................  104
   18         Reliability of Provisional Responses ................  106
   19         Transport ...........................................  106
   19.1       Clients .............................................  107
   19.1.1     Sending Requests ....................................  107
   19.1.2     Receiving Responses .................................  108
   19.2       Servers .............................................  108
   19.2.1     Receiving Requests ..................................  108
   19.2.2     Sending Responses ...................................  109



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   19.3       Framing .............................................  110
   19.4       Error Handling ......................................  110
   20         Security Considerations .............................  111
   20.1       Transport and Network Layer Security ................  112
   20.2       SIP Authentication ..................................  113
   20.2.1     Framework ...........................................  113
   20.2.2     User to User Authentication .........................  114
   20.2.3     Proxy to User Authentication ........................  115
   20.2.4     Authentication Schemes ..............................  117
   20.2.4.1   HTTP Basic ..........................................  117
   20.2.4.2   HTTP Digest .........................................  117
   20.3       SIP Encryption ......................................  118
   20.4       Denial of Service ...................................  119
   21         Common Message Components ...........................  121
   21.1       SIP Uniform Resource Locators .......................  121
   21.1.1     SIP URL components ..................................  121
   21.1.2     Character escaping requirements .....................  124
   21.1.3     Example SIP URLs ....................................  125
   21.1.4     SIP URL Comparison ..................................  126
   21.2       Option Tags .........................................  128
   21.3       Tags ................................................  128
   22         Header Fields .......................................  129
   22.1       Accept ..............................................  131
   22.2       Accept-Encoding .....................................  132
   22.3       Accept-Language .....................................  133
   22.4       Alert-Info ..........................................  133
   22.5       Allow ...............................................  134
   22.6       Authentication-Info .................................  134
   22.7       Authorization .......................................  134
   22.8       Call-ID .............................................  135
   22.9       Call-Info ...........................................  135
   22.10      Contact .............................................  136
   22.11      Content-Disposition .................................  137
   22.12      Content-Encoding ....................................  137
   22.13      Content-Language ....................................  138
   22.14      Content-Length ......................................  138
   22.15      Content-Type ........................................  139
   22.16      CSeq ................................................  139
   22.17      Date ................................................  139
   22.18      Error-Info ..........................................  140
   22.19      Expires .............................................  140
   22.20      From ................................................  141
   22.21      In-Reply-To .........................................  141
   22.22      Max-Forwards ........................................  142
   22.23      MIME-Version ........................................  142
   22.24      Organization ........................................  142
   22.25      Priority ............................................  143
   22.26      Proxy-Authenticate ..................................  143



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   22.27      Proxy-Authorization .................................  144
   22.28      Proxy-Require .......................................  144
   22.29      Record-Route ........................................  144
   22.30      Require .............................................  145
   22.31      Retry-After .........................................  145
   22.32      Route ...............................................  146
   22.33      Server ..............................................  146
   22.34      Subject .............................................  146
   22.35      Supported ...........................................  147
   22.36      Timestamp ...........................................  147
   22.37      To ..................................................  147
   22.38      Unsupported .........................................  148
   22.39      User-Agent ..........................................  148
   22.40      Via .................................................  148
   22.41      Warning .............................................  149
   22.42      WWW-Authenticate ....................................  151
   23         Response Codes ......................................  151
   23.1       Provisional 1xx .....................................  151
   23.1.1     100 Trying ..........................................  151
   23.1.2     180 Ringing .........................................  152
   23.1.3     181 Call Is Being Forwarded .........................  152
   23.1.4     182 Queued ..........................................  152
   23.1.5     183 Session Progress ................................  152
   23.2       Successful 2xx ......................................  152
   23.2.1     200 OK ..............................................  152
   23.3       Redirection 3xx .....................................  152
   23.3.1     300 Multiple Choices ................................  152
   23.3.2     301 Moved Permanently ...............................  153
   23.3.3     302 Moved Temporarily ...............................  153
   23.3.4     305 Use Proxy .......................................  153
   23.3.5     380 Alternative Service .............................  154
   23.4       Request Failure 4xx .................................  154
   23.4.1     400 Bad Request .....................................  154
   23.4.2     401 Unauthorized ....................................  154
   23.4.3     402 Payment Required ................................  154
   23.4.4     403 Forbidden .......................................  154
   23.4.5     404 Not Found .......................................  154
   23.4.6     405 Method Not Allowed ..............................  154
   23.4.7     406 Not Acceptable ..................................  155
   23.4.8     407 Proxy Authentication Required ...................  155
   23.4.9     408 Request Timeout .................................  155
   23.4.10    410 Gone ............................................  155
   23.4.11    413 Request Entity Too Large ........................  155
   23.4.12    414 Request-URI Too Long ............................  155
   23.4.13    415 Unsupported Media Type ..........................  156
   23.4.14    420 Bad Extension ...................................  156
   23.4.15    421 Extension Required ..............................  156
   23.4.16    480 Temporarily Unavailable .........................  156



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   23.4.17    481 Call/Transaction Does Not Exist .................  157
   23.4.18    482 Loop Detected ...................................  157
   23.4.19    483 Too Many Hops ...................................  157
   23.4.20    484 Address Incomplete ..............................  157
   23.4.21    485 Ambiguous .......................................  157
   23.4.22    486 Busy Here .......................................  158
   23.4.23    487 Request Terminated ..............................  158
   23.4.24    488 Not Acceptable Here .............................  158
   23.5       Server Failure 5xx ..................................  158
   23.5.1     500 Server Internal Error ...........................  158
   23.5.2     501 Not Implemented .................................  158
   23.5.3     502 Bad Gateway .....................................  159
   23.5.4     503 Service Unavailable .............................  159
   23.5.5     504 Server Time-out .................................  159
   23.5.6     505 Version Not Supported ...........................  159
   23.5.7     513 Message Too Large ...............................  159
   23.6       Global Failures 6xx .................................  160
   23.6.1     600 Busy Everywhere .................................  160
   23.6.2     603 Decline .........................................  160
   23.6.3     604 Does Not Exist Anywhere .........................  160
   23.6.4     606 Not Acceptable ..................................  160
   24         Locating a SIP Server ...............................  160
   24.1       Computing the List of Next Hops .....................  161
   24.1.1     Numeric Destination Address .........................  162
   24.1.2     SRV Resolution of Host Name .........................  162
   24.1.3     Address Record Resolution of Host Name ..............  163
   24.2       Contacting the Next Hops ............................  163
   25         Examples ............................................  165
   25.1       Registration ........................................  165
   25.2       Session Setup .......................................  166
   26          Augmented BNF for the SIP Protocol .................  171
   26.1       Basic Rules .........................................  173
   27         IANA Considerations .................................  188
   27.1       Option Tags .........................................  189
   27.2       Warn-Codes ..........................................  190
   27.3       Header Field Names ..................................  190
   27.4       Method and Response Codes ...........................  191
   28         Changes Made in Version 00 ..........................  191
   29         Changes Made in Version 01 ..........................  198
   30         Changes Made in Version 02 ..........................  199
   31         Changes Made in Version 03 ..........................  201
   32         Changes Made in Version 04 ..........................  203
   33         Changes Made in Version 05 ..........................  205
   34         Acknowledgments .....................................  210
   35         Authors' Addresses ..................................  210
   36         Bibliography ........................................  211





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