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Internet Engineering Task Force                                 MMUSIC WG
Internet Draft                    M. Handley, H. Schulzrinne, E. Schooler
ietf-mmusic-sip-02.txt                            ISI/Columbia U./Caltech
March 27, 1997
Expires: September 25, 1997


                    SIP: Session Initiation Protocol

STATUS OF THIS MEMO

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

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

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

   Distribution of this document is unlimited.

                                 ABSTRACT


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

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



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   Authors' Note

   This document is the result of a merger of the Session Invitation
   Protocol (draft-ietf-mmusic-sip-00.txt) and the Simple Conference
   Invitation Protocol (draft-ietf-mmusic-scip-00.txt), and of an
   attempt to make SIP more generic and to fit into a more flexible
   infrastructure that includes companion protocols including SDP, HTTP
   and RTSP.


   Changes

   Since version -01, the following things have changed:

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

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

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

        o Formatting closer to HTTP and RTSP.

        o Explain relationship to H.323.

1 Introduction

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

        o The session is advertised, users see the advertisement, then
         join the session address to participate.

        o Users are invited to participate in a session, which may or
         may not already be advertised.

   The Session Description Protocol (SDP) [1] together with the Session
   Announcement Protocol (SAP) [2], provide a mechanism for the former.
   This document presents the Session Initiation Protocol (SIP) to
   perform the latter. SIP MAY also use SDP to describe a session.


                      Figure omitted in ASCII version


   Figure 1: Session Lifecycle




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   We make the design decision that how a user discovers that a session
   exists is orthogonal to a session's conference control model.  Figure
   1 shows a potential place for SIP in the lifecycle of both
   lightweight sessions and in more tightly-coupled conferencing. Note
   that the Session Initiation Protocol and the Session Announcement
   Protocol may be invoked or re-invoked at later stages in a session's
   lifecycle.

   The Session Initiation Protocol is also intended to be used to invite
   servers into sessions. Examples might be where a recording server can
   be invited to participate in a live multimedia session to record that
   session, or a video-on-demand server can be invited to play a video
   stream into a live multimedia conference. In such cases we would like
   SIP to lead the server gracefully into the control protocol that
   controls the actual recording and playback.

   We also make the design decision that inviting a user to participate
   in a session is independent of quality of service (QoS) guarantees
   for that session. Such QoS guarantees (if they are required) may be
   dependent on the full membership of the session, and this may or may
   not be known to the agent performing session invitation.

   SIP offers some of the same functionality as H.323, but can also be
   used in conjunction with it. In this mode, SIP is used to locate the
   appropriate terminal, where the terminal is identified by its H.245
   address [TBD: what does this look like?]. An H.323-capable terminal
   then proceeds with a normal H.323/H.245 invitation [3].

1.1 Requirements

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC xxxx [4].

1.2 Terminology

   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 HTTP.

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

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




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   Invitation: A request sent to attempt to contact a user (or service)
        to request that they participate in a session.

   Invitee, Invited User: The person or service that the calling party
        is trying to invite to a conference.

   Location server: A program that is contacted by a client and that
        returns one or more possible locations for the user or service
        without contacting that user or service directly.

   Location service: A service used by a location server to obtain
        information about a user's possible location.

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

   Server: An application program that accepts connections in order to
        service requests by sending back responses. A server may be the
        called user agent, a proxy server, or a location server.

   User Agent, Called User Agent: The server application which contacts
        the invitee to inform them of the invitation, and to return a
        reply.

   Any given program may be capable of acting both as a client and a
   server. A typical multimedia conference controller would act as a
   client to initiate calls or to invite others to conferences and as a
   server to accept invitations.

1.3 General Requirements

   SIP is a Session Initiation Protocol. It is not a conference control
   protocol. SIP can be used to perform a search for a user or service
   and to request that that user or service participate in a session.

   Once SIP has been used to initiate a multimedia session SIP's task is
   finished. There is no concept of a SIP session (as opposed to a SIP
   search for a user or service). If whatever conference control
   mechanism is used in the session needs to add or remove a media
   stream, SIP may be used to perform this task, but again, once the
   information has been successfully conveyed to the participants, SIP
   is then no longer involved.

   SIP must be able to utilize both UDP and TCP as transport protocols.



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   From a performance point of view, UDP is preferable as it allows the
   application to more carefully control the timing of messages, it
   allows parallel searches without requiring connection state for each
   outstanding request, and allows the use of multicast.

   From a pragmatic point of view, TCP allows easier passage through
   existing firewalls, and with appropriate protocol design, allows
   common SIP, HTTP and RTSP servers.

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

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

   Unlike control protocols, there is minimal shared-state in SIP -- in
   a minimal implementation the initiator maintains all the state about
   the current attempt to locate and contact a user or service - servers
   or proxies can be stateless (although they don't have to be). All the
   state needed to get a response back from a server to the initiator is
   carried in the SIP request itself - this is also necessary for loop
   prevention.


        Whilst redesigning SIP, we have attempted to ensure that it
        has a clear interaction with the currently evolving Real-
        Time Stream Control Protocol.

1.4 Addressing

   SIP is a protocol that exchanges messages between peer user agents or
   proxies for user agents. We assume the user agent is an application
   that acts on behalf of the user it represents (thus it is sometimes
   described as a client of the user) and that is co-resident with that
   user. A proxy for a user agent serves as a forwarding mechanism or
   bridge to the actual location of the user agent. We also refer to
   such proxies as location server

   In the computer realm, the equivalent of a personal telephone number
   combines the user's login id ( mjh ) with a machine host name (
   metro.isi.edu ) or numeric network address ( 128.16.64.78 ). A user's



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   location-specific address can be obtained out-of-band, can be learned
   via existing media agents, can be included in some mailers' message
   headers, or can be recorded during previous invitation interactions.

   However, users also publish several well-known addresses that are
   relatively location-independent, such as email or web home-page
   addresses. Rather than require that users provide their specific
   network locales, we can take advantage of email and web addresses as
   being (relatively) memorable, and also leverage off the Domain Name
   Service (DNS) to provide a first stage location mechanism. Note that
   an email address ( M.Handley@cs.ucl.ac.uk ) is usually different from
   the combination of a specific machine name and login name (
   mjh@mercury.lcs.mit.edu ). SIP should allow both forms of addressing
   to be used, with the former requiring a location server to locate the
   user.

   One perceived problem of email addressing is that it is possible to
   guess peoples' addresses and thus the system of unlisted (in the
   telephone directory) numbers is more of a problem. However, this
   really only provides security through obscurity, and real security is
   better provided through authentication and call screening.

1.5 Call Setup

   Call setup is a multi-phase procedure. In the first phase, the
   requesting client tries to ascertain the address where it should
   contact the remote user agent or user agent proxy. The local client
   checks if the user address is location-specific. If so, then that is
   the address used for the remote user agent. If not, the requesting
   client looks up the domain part of the user address in the DNS. This
   provides one or more records giving IP addresses. If a new service
   (SRV) resource record [5] is returned giving a location server, then
   that is the address to contact next. If no relevant resource record
   is returned, but an A record is returned, then that is the address to
   contact next. If neither a resource record or an A record is
   returned, but an MX record is returned, then the mail host is the
   address to contact next.

   Presuming an address for the invitee is found from the DNS, the
   second and subsequent phases basically implement a request-response
   protocol.  A session description (typically using SDP format) is sent
   to the contact address with an invitation for the user to join the
   session.

   This request may be sent over a TCP connection or as a single UDP
   datagram (the format of both is the same and is described later), and
   is sent to a well-known port.




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   If a user agent or conference server is listening on the relevant
   port, it can send one of the responses below. If no server or agent
   is listening, an ICMP port-unreachable response will be triggered
   which should cause the TCP connection setup to fail or cause a UDP
   send failure on retransmissions.

1.6 Locating a User

   It is expected that a user is situated at one of several frequented
   locations. These locations can be dynamically registered with a
   location server for a site (for a local area network or
   organization), and incoming connections can be routed simultaneously
   to all of these locations if so desired. It is entirely up to the
   location server whether the server issues proxy requests for the
   requesting user, or if the server instructs the client to redirect
   the request.

   In general a reply MUST be sent by the same mechanism that the
   request was sent by. Hence, if a request was unicast, then the reply
   MUST be unicast back to the requester; if the request was multicast,
   the reply MUST be multicast to the same group to which the request
   was sent; if the request was sent by TCP, the reply MUST be sent by
   TCP.

   In all cases where a request is forwarded onwards, each host relaying
   the message SHOULD add its own address to the path of the message so
   that the replies can take the same path back, thus ensuring correct
   operation through compliant firewalls and loop-free requests. On the
   reply path, these routing headers MUST be removed as the reply
   retraces the path, so that routing internal to sites is hidden. When
   a multicast request is made, first the host making the request, then
   the multicast address itself are added to the path.

2 Notational Conventions and Generic Grammar

   Since many of the definitions and syntax are identical to HTTP/1.1,
   this specification only points to the section where they are defined
   rather than copying it. For brevity, [HX.Y] is to be taken to refer
   to Section X.Y of the current HTTP/1.1 specification (RFC 2068).

   All the mechanisms specified in this document are described in both
   prose and an augmented Backus-Naur form (BNF) similar to that used in
   RFC 2068 [H2.1]. It is described in detail in [6].

   In this draft, we use indented and smaller-type paragraphs to provide
   background and motivation.

3 Protocol Parameters



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3.1 SIP Version

   applies, with HTTP replaced by SIP.

   Applications sending Request or Response messages, as defined by this
   specification, MUST include an SIP-Version of "SIP/2.0". Use of this
   version number indicates that the sending application is at least
   conditionally compliant with this specification.

3.2 UCI: Universal Communication Identifier

   [TBD: describe all legal address formats.]

4 SIP Message

   All messages are text-based, using the conventions of HTTP/1.1
   [H4.1], except for the additional ability of SIP to use UDP. When
   sent over TCP or UDP, multiple requests can be carried in a single
   TCP connection or UDP datagram. UDP Datagrams should not normally
   exceed the path MTU in size if it is known, or 1,000 bytes if the MTU
   is unknown.

4.1 Message Types

   SIP messages consist of requests from client to server and responses
   from server to client.


     SIP-message = Request | Response     ; HTTP/1.1 messages



   Request (section 5) and response (section 6) messages use the generic
   message format of RFC 822 for transferring entities (the payload of
   the message). Both types of messages consist of a start-line, one or
   more header fields (also known as "headers"), an empty line (i.e., a
   line with nothing preceding the CRLF) indicating the end of the
   header fields, and an optional message-body.


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

     start-line      = Request-Line | Status-Line





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   In the interest of robustness, servers SHOULD ignore any empty
   line(s) received where a Request-Line is expected. In other words, if
   the server is reading the protocol stream at the beginning of a
   message and receives a CRLF first, it should ignore the CRLF.

4.2 Message Headers

   HTTP header fields, which include general-header (section ),
   request-header (section ), response-header (section ), fields, follow
   the same generic format as that given in Section 3.1 of RFC 822. Each
   header field consists of a name followed by a colon (":") and the
   field value. Field names are case-insensitive. The field value may be
   preceded by any amount of LWS, though a single SP is preferred.
   Header fields can be extended over multiple lines by preceding each
   extra line with at least one SP or HT. Applications SHOULD follow
   "common form" when generating HTTP constructs, since there might
   exist some implementations that fail to accept anything beyond the
   common forms.


     message-header = field-name ":" [ field-value ] CRLF

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



   The order in which header fields with differing field names are
   received is not significant.

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

4.3 Message Body

   The rules for when a message-body is allowed in a message differ for
   requests and responses.



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   The presence of a message-body in a request is signaled by the
   inclusion of a Content-Length or Transfer-Encoding header field in
   the request's message-headers. A message-body MAY be included in a
   request only when the request method allows an entity-body.

   For response messages, whether or not a message-body is included with
   a message is dependent on both the request method and the response
   status code (section ). All 1xx (informational) responses MUST NOT
   include a message-body. All other responses do include a message-
   body, although it may be of zero length.

4.4 Message Length

   When a message-body is included with a message, the length of that
   body is determined by one of the following (in order of precedence):

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

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

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

4.5 General Header Fields

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


     general-header = Date                     ; Section
                    | Transfer-Encoding        ; Section
                    | Via                      ; Section



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

5 Request




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


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



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

6 Response

   [H6] applies except that HTTP-Version is replaced by SIP-Version
   define some HTTP codes.

   After receiving and interpreting a request message, the recipient
   responds with an SIP response message.


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



6.1 Status-Line

   The first line of a Response message is the Status-Line , consisting
   of the protocol version followed by a numeric status code, the
   sequence number of the corresponding request and the textual phrase
   associated with the status code, with each element separated by SP
   characters. No CR or LF is allowed except in the final CRLF sequence.
   Note that the addition of a


     Status-Line = SIP-Version SP Status-Code SP seq-no SP Reason-Phrase CRLF






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6.1.1 Status Code and Reason Phrase

   The Status-Code element is a 3-digit integer result code of the
   attempt to understand and satisfy the request. These codes are fully
   defined in section10. The Reason-Phrase is intended to give a short
   textual description of the Status-Code. The Status-Code is intended
   for use by automata and the Reason-Phrase is intended for the human
   user. The 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. There are 5
   values for the first digit:

        o 1xx: Informational - Request received, continuing process

        o 2xx: Success - The action was successfully received,
         understood, and accepted

        o 3xx: Redirection - Further action must be taken in order to
         complete the request

        o 4xx: Client Error - The request contains bad syntax or cannot
         be fulfilled

        o 5xx: Server Error - The server failed to fulfill an apparently
         valid request

   The individual values of the numeric status codes defined for
   SIP/2.0, and an example set of corresponding Reason-Phrase below. The
   reason phrases listed here are only recommended -- they may be
   replaced by local equivalents without affecting the protocol. Note
   that SIP adopts many HTTP/1.1 status codes and adds SIP-specific
   status codes in the starting at 450 to avoid conflicts with newly
   defined HTTP status codes.


      Status-Code    = "100"   ; Continue
                     | "200"   ; OK
                     | "300"   ; Multiple Choices
                     | "301"   ; Moved Permanently
                     | "302"   ; Moved Temporarily
                     | "303"   ; See Other
                     | "305"   ; Use Proxy
                     | "400"   ; Bad Request
                     | "401"   ; Unauthorized
                     | "402"   ; Payment Required
                     | "403"   ; Forbidden



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                     | "404"   ; Not Found
                     | "405"   ; Method Not Allowed
                     | "406"   ; Not Acceptable
                     | "407"   ; Proxy Authentication Required
                     | "408"   ; Request Time-out
                     | "409"   ; Conflict
                     | "410"   ; Gone
                     | "411"   ; Length Required
                     | "412"   ; Precondition Failed
                     | "413"   ; Request Entity Too Large
                     | "414"   ; Request-URI Too Large
                     | "415"   ; Unsupported Media Type
                     | "500"   ; Internal Server Error
                     | "501"   ; Not Implemented
                     | "502"   ; Bad Gateway
                     | "503"   ; Service Unavailable
                     | "504"   ; Gateway Time-out
                     | "505"   ; HTTP Version not supported
                     | extension-code

      extension-code = 3DIGIT

      Reason-Phrase  = *<TEXT, excluding CR, LF>




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

6.1.2 Response Header Fields

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




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     response-header = Location              ; Section
                       | Proxy-Authenticate  ; Section
                       | Public              ; Section
                       | Retry-After         ; Section
                       | Server              ; Section
                       | Vary                ; Section
                       | WWW-Authenticate    ; Section



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

7 SIP Message Body

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

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


   INVITE 128.16.64.19/65729 SIP/2.0
   Via: SIP/2.0/UDP 239.128.16.254 16
   Via: SIP/2.0/UDP 131.215.131.131
   Via: SIP/2.0/UDP 128.16.64.19
   From: mjh@isi.edu
   To: schooler@cs.caltech.edu
   Content-type: application/sdp
   Content-Length: 187

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



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

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

   The request header above states that the request was initiated by
   mjh@isi.edu (specifically it was initiated from 128.16.64.19, as can
   be seen from the  Via header) and the user being invited is
   schooler@cs.caltech.edu.

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

   The header is terminated by an empty line and is followed by the
   session description payload.

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

8 Methods

   The following methods are defined:

   INVITE: The user or service is being invited to participate in the
        session. The session description given must be completely
        acceptable for a "200 OK" response to be given. This method MUST
        be supported by a SIP server.

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

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

   Methods that are not supported by a user agent should cause a "501
   Not Implemented" response to be returned.

9 Header Field Definitions



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   SIP header fields are similar to HTTP header fields in both syntax
   and semantics. In general the ordering of the header fields is not of
   importance (with the exception of Via fields, see below) but proxies
   MUST NOT reorder or otherwise modify header fields other than by
   adding a new Via field. This allows an authentication field to be
   added after the Via fields that will not be invalidated by proxies.
   Field names are not case-sensitive, although their values may be.

   Content-Length,  Content-Type,  To,  From header fields are
   compulsory. Other fields may be added as required. Header fields MUST
   be separated by a single linefeed character. The header MUST be
   separated from the payload by an empty line (two linefeed
   characters).

   A compact form of these header fields is also defined in section 10.9
   for use over UDP when the request has to fit into a single packet and
   size is an issue.

9.1 Accept

   See [H14.1]. This header field is used only for the  OPTIONS request
   to indicate what description formats are acceptable.

9.2 Accept-Language

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

9.3 Authentication

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


        HS: Ordering and semantics needs work. Maybe we can recycle
        the S/MIME work?

9.4 Confirm

   TBD.

9.5 Contact-Host



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   TBD.

9.6 From

   The request header MUST contain a  From request-header field,
   indicating the invitation initiator. The field MUST be machine-
   usable, as defined my mailbox in RFC 822 (as updated by RFC 1123).
   Only a single initiator and a single invited user are allowed to be
   specified in a single SIP request.

9.7 Retry-After

   The  Retry-After response-header field can be used with a 503
   (Service Unavailable) response to indicate how long the service is
   expected to be unavailable to the requesting client and with a 404
   (Not Found) or 451* (Busy) response to indicate when the called party
   may be available again. The value of this field can be either an
   HTTP-date or an integer number of seconds (in decimal) after the time
   of the response.


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



   Two examples of its use are


     Retry-After: Fri, 31 Dec 1999 23:59:59 GMT
     Retry-After: 120



   In the latter example, the delay is 2 minutes.

9.8 Reason

   TBD.

9.9 To

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

9.10 Via

   The  Via field indicates the path taken by the request so far.  This
   prevents request looping and ensures replies take the same path as



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   the requests, which assists in firewall traversal and other unusual
   routing situations. Initiators MUST add their own Path field to each
   request. This Path field MUST be the first field in the request.
   Subsequent proxies SHOULD each add their own additional Path field
   which MUST be added before any existing Path fields. When a reply
   passes through a proxy on the reverse path, that proxies Path field
   MUST be removed from the reply.

   The format for a  Via header is:

     Via = "Via" ":" 1#( sent-protocol sent-by [ ttl ] [ comment ] )
     sent-protocol     = [ protocol-name "/" ] protocol-version
                         [ "/" transport ]
     protocol-name     = "SIP" | token
     protocol-version  = token
     transport         = "UDP" | "TCP"
     sent-by           = host [ ":" port ]
     ttl               = *DIGIT



   TTL is included only if the address is a multicast address.

10 Status Code Definitions

   The response codes are consistent with, and extend, HTTP/1.1 response
   codes. Not all HTTP/1.1 response codes are appropriate, and only
   those that are appropriate are given here. Response codes not defined
   by HTTP/1.1 are marked with an asterisk, and have codes x50 upwards
   to avoid clashes with future HTTP response codes, or 6xx which are
   not used by HTTP. The default behavior for unknown response codes is
   given for each category of codes.

10.1 Informational 1xx

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

10.1.1 100 Trying

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



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10.1.2 150 Ringing

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

10.2 Successful 2xx

   The request was successful and MUST terminate a search.

10.2.1 200 OK

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

10.3 Redirection 3xx

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

10.3.1 300 Multiple Choices

   The requested resource corresponds to any one of a set of
   representations, each with its own specific location, and agent-
   driven negotiation information (section 13) is being provided so that
   the user (or user agent) can select a preferred representation and
   redirect its request to that location.

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

   If the server has a preferred choice, it SHOULD include the specific
   URL for that representation in the  Location field; user agents MAY
   use the Location field value for automatic redirection.

10.3.2 301 Moved Permanently

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



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   the request's path as the user's new location.

10.3.3 302 Moved Temporarily

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

10.3.4 350* Alternative Service

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

10.4 Request Failure 4xx

   4xx responses are definite failure responses that MUST terminate the
   existing search for a user or service. They SHOULD NOT be retried
   immediately without modification.

10.4.1 400 Bad Request

   The request could not be understood due to malformed syntax.

10.4.2 401 Unauthorized

   The request requires user authentication.

10.4.3 402 Payment Required

   Reserved for future use.

10.4.4 403 Forbidden

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

10.4.5 404 Not Found

   The server has definitive information that the user does not exist at
   the domain specified.

10.4.6 406 Not Acceptable

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

10.4.7 450* Decline



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   The user's machine was successfully contacted but the user explicitly
   does not wish to participate.

10.4.8 451* Busy

   The user's machine was successfully contacted but the user is busy,
   or the user does not wish to participate (the ambiguity is
   intentional).

10.5 Server Failure 5xx

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

10.5.1 500 Server Internal Error

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

10.5.2 501 Not implemented

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

10.5.3 503 Service Unavailable

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

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

10.6 Search Responses 6xx

   6xx responses are failure responses given whilst trying to locate the
   specified user or service. They are not definitive failures, and
   SHOULD NOT terminate the search if other possible locations remain
   untried.

10.6.1 600* Search Failure



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   The user agent or proxy server understood the user's address, but the
   request was unsuccessful in contacting the user. A proxy might return
   this error towards the initiator if an attempt to contact a server
   failed for an unknown reason.

10.6.2 601* Not known here

   The call was unsuccessful because the user or service was not known
   at the address called. This is not a definitive failure; the address
   may be valid at another server.

10.6.3 602* Not currently here

   The call was unsuccessful because although the the user or service
   was known at the address called, the user or service is not currently
   located at this address. This is not a definitive failure; the user
   may be contactable at another server.

10.6.4 603* Alternative Address

   The call was unsuccessful because the user or service is not
   available at this location, but one or more alternative non-
   definitive locations are suggested to try in addition to any that may
   already be being tried.  A 603 response MUST NOT suggest any of the
   hosts in the request's path as an alternative location.

10.7 Example: Normal Replies

   An example reply is given below. The first line of the reply states
   the SIP version number, that it is a "200 OK" reply, which means the
   request was successful. The  Via header are taken from the request,
   and entries are removed hop by hop as the reply retraces the
   request's path. A new authentication field is added by the invited
   user's agent if required. The session ID is taken directly from the
   original request, along with the request header. The original sense
   of From field is preserved (i.e, it's the session originator).

   In addition, a  Contact-host field is added giving details of the
   host the user was located on, or alternatively the relevant proxy
   contact point which should be reachable from the invitation
   initiator's host.


   SIP/2.0 200 128.16.64.19/65729
   Via: SIP/2.0/UDP 239.128.16.254 16
   Via: SIP/2.0/UDP 131.215.131.131
   Via: SIP/2.0/UDP 128.16.64.19 1
   From: mjh@isi.edu



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   To: schooler@cs.caltech.edu
   Contact-host: 131.215.131.147



   This same format is used for replies for other categories of reply,
   except that some of then may require payloads to be carried.

   If the invited user's agent requires confirmation of receipt of a
   "200 OK" reply, it may optionally add an additional Confirm: required
   header to the body of the message specifying that an acknowledgment
   is required. This is only permitted with category 2xx replies. An
   example is:


   SIP/2.0 200 128.16.64.19/65729
   Via: SIP/2.0/UDP 239.128.16.254 16
   Via: SIP/2.0/UDP 131.215.131.131
   Via: SIP/2.0/UDP 128.16.64.19
   From: mjh@isi.edu
   To: schooler@cs.caltech.edu
   Contact-host: 131.215.131.147
   Confirm: required



   In response to such a request, the invitation initiators agent should
   retransmit its request with an additional  Confirm header, with the
   value "true" or "false" stating whether the session still exists or
   no longer exists respectively (see section 7.1 for details). An
   example of an confirmation request is:


   INVITE 128.16.64.19/65729 SIP/2.0
   Via: SIP/2.0/UDP 239.128.16.254:70 16
   Via: SIP/2.0/UDP 131.215.131.131
   Via: SIP/2.0/UDP 128.16.64.19
   From: mjh@isi.edu
   To: schooler@cs.caltech.edu
   Confirm: true
   Content-type: application/sdp
   Content-Length: 187

   v=0
   o=user1 2353655765 2353687637 IN IP4 128.3.4.5
   s=Mbone Audio
   i=Discussion of Mbone Engineering Issues
   e=mbone@somewhere.com



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   c=IN IP4 224.2.0.1/127
   t=0 0
   m=audio 3456 RTP/AVP 0



   Such confirmations are still useful when TCP transport is used as
   they provide application level confirmation rather than transport
   level confirmation. If they are not used, it is possible that a "200
   OK" response may be received after the application making the call
   has timed out the call and exited.

10.7.1 Redirects

   "603 alternative address" replies and 301 and 302 moved replies
   should specify another location using the  Location field.

   An example of a "603 alternative address" reply is:


   SIP/2.0 603 128.16.64.19/65729
   Via: SIP/2.0/UDP 131.215.131.131 1
   Via: SIP/2.0/UDP 128.16.64.19
   From: mjh@isi.edu
   To: schooler@cs.caltech.edu
   Location: 239.128.16.254 16
   Content-length:0



   In this example, the proxy (131.215.131.131) is being advised to
   contact the multicast group 239.128.16.254 with a ttl of 16. In
   normal situations a server would not suggest a redirect to a local
   multicast group unless (as in the above situation) it knows that the
   previous proxy or client is within the scope of the local group.

   For unicast 603 redirects, a proxy MAY query the suggested location
   itself or send MAY the redirect on back towards the client. For
   multicast 603 redirects, a proxy SHOULD query the multicast address
   itself rather than sending the redirect back towards the client as
   multicast may be scoped and this allows a proxy within the
   appropriate scope regions to make the query.

   For 301 or 302 redirects, a proxy SHOULD send the redirect on back
   towards the client and terminate any other searches it is performing
   for the same request. Multicast 301 or 302 redirects MUST NOT be
   generated.




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10.8 Alternative Services

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

   SIP/2.0 350 128.16.64.19/32492/2
   Via: SIP/2.0/UDP 131.215.131.131
   Via: SIP/2.0/UDP 128.16.64.19
   From: mjh@isi.edu
   To: schooler@cs.caltech.edu
   Contact-host: IN IP4 131.215.131.131
   Content-type: application/sdp
   Content-length: 146

   v=0
   o=mm-server 2523535 0 IN IP4 131.215.131.131
   s=Answering Machine
   i=Leave an audio message
   c=IN IP4 128.16.64.19
   t=0 0
   m=audio 12345 RTP/AVP 0



   In this case, the answering server provides a session description
   that describes an "answering machine".  If the invitation initiator
   decides to take advantage of this service, it should send an
   invitation request to the contact host (131.215.131.131) with the
   session description provided. This request should contain a different
   session id from the one in the original request.  An example would
   be:


   INVITE 128.16.64.19/32492/3 SIP/2.0
   Via: SIP/2.0/UDP 128.16.64.19
   From: mjh@isi.edu
   To: schooler@cs.caltech.edu
   Content-type: application/sdp
   Content-length: 146

   v=0
   o=mm-server 2523535 0 IN IP4 128.16.5.31
   s=Answering Machine
   i=Leave an audio message
   c=IN IP4 128.16.64.19
   t=0 0
   m=audio 12345 RTP/AVP PCMU





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

10.8.1 Negotiation

   A "406 Not Acceptable" reply means that the user wishes to
   communicate, but cannot support the session described adequately. The
   "406 Not Acceptable" reply contains a list of reasons why the session
   described cannot be supported. These reasons can be one or more of:

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

   406.2 Incompatible Protocol: one or more protocols described in the
        request is not available.

   406.3 Incompatible Format: one or more media formats described in the
        request is not available.

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

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

   Other reasons are likely to be added later. It is hoped that
   negotiation will not frequently be needed, and when a new user is
   being invited to join a pre-existing lightweight session, negotiation
   may not be possible. If is up to the invitation initiator to decide
   whether or not to act on a "406 Not Acceptable" reply.

   A complex example of a "406 Not Acceptable" reply is:

   SIP/2.0 406 128.16.64.19/32492/5
   From: mjh@isi.edu
   To: schooler@cs.caltech.edu
   Contact-host: 131.215.131.131
   Reason: 406.1, 406.3, 406.4
   Content-Type: meta/sdp
   Content-Length: 50

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



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   m=video 2232 RTP/AVP 31



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

   Invitation initiators MAY choose to treat "406 Not Acceptable"
   replies as a failure if they wish to do so.

10.9 Compact Form

   When SIP is carried over UDP with authentication and a complex
   session description, it may be possible that the size of a request or
   reply is larger than the MTU (or default 1,000-byte limit if the MTU
   is not known).  To reduce this problem, a more compact form of SIP is
   also defined by using alternative names for common header fields.
   These short forms are NOT abbreviations, they are field names. No
   other abbreviations are allowed.


   short field name    long field name      note
   a                    Confirm             from "acknowledge"
   c                    Content-Type
   e                    Content-Encoding
   f                    From
   h                    Contact-Host
   l                    Content-Length
   m                    Location            from "moved"
   r                    Reason
   t                    To
   v                    Via


   Thus the header in section ?? could also be written:


   INVITE 128.16.64.19/65729 SIP/2.0
   p:IN IP4 UDP 239.128.16.254 1 16
   p:IN IP4 UDP 131.215.131.131 1
   p:IN IP4 UDP 128.16.64.19 1
   f:mjh@isi.edu



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   t:schooler@cs.caltech.edu
   c:application/sdp
   l:187

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



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

11 SIP Transport

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

   UDP has advantages over TCP from a performance point of view, as the
   SIP application can keep control of the precise timing of
   retransmissions, and can also make simultaneous call attempts to many
   potential locations of many users without needing to keep TCP
   connection state for each connection.

   TCP has the advantage that clients are simpler to implement because
   no retransmission timing code needs to be written and also that it is
   possible to have a single server serving SIP and HTTP with very
   little extra code.

   With UDP, all the additional reliability code is in the client. It is
   recommended that servers SHOULD implement both TCP and UDP
   functionality as the additional server code required is very small.

   Clients MAY implement either TCP or UDP transport or both as they see
   fit.

11.1 Reliability using UDP transport

   The Session Invitation Protocol is straightforward in operation. Only
   the initiating client needs to keep any state regarding the current
   connection attempt. SIP assumes no additional reliability from IP.



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   Requests or replies may be lost. A SIP client SHOULD simply
   retransmit a SIP request until it receives a reply, or until it has
   reached some maximum number of timeouts and retransmissions. If the
   reply is merely a 1xx Informational progress report, the initiating
   client SHOULD still continue retransmitting the request, albeit less
   frequently.

   When the remote user agent or server sends a final 2xx or 4xx
   response (not a 1xx report), it cannot be sure the client has
   received the response, and thus SHOULD cache the results until a
   connection setup timeout has occurred to avoid having to contact the
   user again. The server MAY also choose to cache 3xx or 6xx responses
   if the cost of obtaining the response outweighs the cost of caching
   it.

   It is possible that a user can be invited successfully, but that the
   reply that the user was successfully contacted may not reach the
   invitation initiator. If the session still exists but the initiator
   gives up on including the user, the contacted user has sufficient
   information to be able to join the session. However, if the session
   no longer exists because the invitation initiator "hung up" before
   the reply arrived and the session was to be two-way, the conferencing
   system should be prepared to deal with this circumstance.

   One solution is for the initiator to acknowledge the invitee's "200
   OK" reply. Although not required, in the case of a successful
   invitation the invited user's agent can make a confirmation request
   in its "200 OK" reply. In this case the initiator's agent sends a
   single request with a reply  Confirm: true if the request was still
   valid or a reply  Confirm: false if it was not so that a premature
   hang-up can be detected without a long timeout. Such a confirmation
   request may be retransmitted by the invited user's agent if it so
   desired. Confirmation requests can only be made with "200 OK"
   replies, and only the invitation initiator's agent may issue the
   actual confirmation.

   Only a "200 OK" reply warrants such a confirmation handshake, because
   it is the only situation where user-relevant state may be
   instantiated anywhere other than at the initiator's client. In all
   other cases, it is not necessary that state is maintained. In
   particular, when a server makes multiple proxy requests, "5xx Server
   Error" and "6xx Search Response" replies do not immediately get
   passed back to the invitation initiator, and so no end-to-end
   acknowledgment of a failed request is possible.

11.2 Reliability using TCP transport

   TCP is a reliable transport protocol, and so we do not need to define



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   additional reliability mechanisms. However, we must define rules for
   connection closedown under normal operation.

   The normal mode of operation is for the client (or proxy acting as a
   client) to make a TCP connection to the well-known port of a host
   housing a SIP server. The client then sends the SIP request to the
   server over this connection and waits for one or more replies. The
   client MAY close the connection at any time.

   The server MAY send one or more 1xx Informational responses before
   sending a single 2xx, 3xx, 4xx, 5xx or 6xx reply. The server MUST NOT
   send more than one reply, with the exception of 1xx responses. The
   server SHOULD NOT close the TCP connection until it has sent its
   final response, at which point it MAY close the TCP connection if it
   wishes to. However, normally it is the client's responsibility to
   close the connection.

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

   The same application-level confirmation rules apply for TCP as for
   UDP.

12 Searching

   A basic assumption of SIP is that a location server at the user's
   home site either knows where the user resides, knows how to locate
   the user, or at the very least knows another location server that
   possibly might have a better idea. How these servers get this
   information is outside the scope of SIP itself, but it is expected
   that many different user-location services will exist for some time.
   SIP is designed so that it does not care which location service SIP
   servers actually employ.

12.1 Proxy servers: Relaying and Redirection

   If a proxy server receives a request for a user whose location it
   does not know, and for whom it has no better idea where the user
   might be, then the server should return a "601 Not Currently Here"
   reply message.

   If the server does have an idea how to contact the user, it can
   either forward (relay) the request itself, or can redirect the
   invitation initiator to another client that is more likely to know by
   sending a 603, 301 or 302 response as appropriate. It can also
   gateway the request into some other form if some other invitation



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   protocol is in use in a region containing the invited user, though in
   doing so the server is likely to give up being stateless.

   Whether to relay the request or to redirect the request is up to the
   server itself. For example, if the server is on a firewall machine,
   then it will probably have to relay the request to servers inside the
   firewall. Additionally, if a local multicast group is to be used for
   user location, then the server is likely to relay the request.
   However, if the user is currently away from home, relaying the
   request makes little sense, and the server is more likely (though not
   compelled) to send a redirect reply. SIP is policy-free on this
   issue. In general, local searches are likely to be better performed
   by relaying whereas wide-area searches are likely to be better
   performed by redirection.

   When SIP uses UDP transport, clients and servers can make multiple
   simultaneous requests to locate a particular user at low cost. This
   greatly speeds up any search for the user, and in most cases will
   only result in one successful response. Although several simultaneous
   paths may reach the same host, successful responses arriving from
   multiple paths will not confuse the client as they should all contain
   the same successful host address. However, this does imply that paths
   with many levels of relaying should be strongly discouraged as if the
   request is fanned out at each hop and relayed many times, request
   implosions could result. Thus servers that are not the first hop
   servers in a chain of servers SHOULD NOT make multiple parallel
   requests, but should send a redirection response with multiple
   alternatives. Thus a firewall host can still perform a parallel
   search but can control the fanout of the search.

12.2 Parallel Searches: Initiator Behavior

   The session initiator may make a parallel search for a user. This can
   occur when DNS resolution results in multiple addresses, or when
   contacting a remote server results in a "603 Alternative Address"
   response containing multiple addresses to try. All such parallel
   searches for the same SIP request MUST contain the same SIP Id,
   though the sequence number (given in the  Path field) SHOULD be
   different for each of the parallel searches.

   Whilst performing a parallel search, different responses may result
   from different servers, and it is important for the initiating client
   to handle these responses correctly. In general, the following rules
   apply:

        o If a 2xx response is received, the invitation was successful,
         the user should be informed and all pending requests should be
         terminated and/or ignored.



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        o If a 4xx response is received the invitation has definitively
         failed, the user should be informed, and all pending requests
         should be terminated and/or ignored.

        o If a 3xx response is received, the search should be terminated
         and all pending requests should be terminated and/or ignored.
         However, further action MAY be taken depending on the actual
         reply without informing the user or alternatively the
         invitation MAY be regarded as having failed in which case the
         user MUST be informed.

        o If a 5xx or 6xx response is received, the particular server
         responding is removed from the parallel search and the search
         continues.  If a "603 Alternative Address" response is
         received, the search may be expanded to include those servers
         listed in the response that have not already responded. The
         user SHOULD NOT be informed unless there are no other servers
         left to try, in which case the user MUST be informed.

        o If a 1xx response is received, the search continues. The user
         MAY be informed as deemed appropriate.

12.3 Parallel Searches: Proxy Behavior

   In the same way that an Initiating Client can discover multiple
   addresses to try, a proxy server can also discover multiple addresses
   that it may try. For a proxy server to be stateless, it must not make
   multiple SIP requests because it would then be possible to return a
   5xx or 6xx response to the Initiating Client and afterwards obtain a
   definitive answer. To be able to make multiple parallel SIP requests,
   it must keep state as to the replies it has already received and MUST
   NOT return any reply other than 1xx informational replies until it
   has received a definitive reply or has no further addresses to try.

   Thus faced with DNS resolution giving multiple addresses, a proxy
   server that wishes to be stateless should only send a SIP request to
   the first address. Similarly a stateless proxy should not attempt to
   send SIP request to multiple addresses given in a "603 Alternative
   Address" response that is returned it it, but should forward such a
   response back towards the initiator.

   Proxies that wish to keep state should follow the following rules
   regarding responses obtained during a parallel search:

        o If a 2xx response is received, the invitation was successful,
         the 2xx response should be forwarded back towards the
         initiator, and all pending requests should be terminated and/or
         ignored.



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        o If a 4xx response is received the invitation has definitively
         failed, the 4xx response should be forwarded back towards the
         initiator, and all pending requests should be terminated and/or
         ignored.

        o If a 3xx response is received the invitation is regarded by
         the proxy as having failed, the 3xx response should be
         forwarded back towards the initiator, the search should be
         terminated and all pending requests should be terminated and/or
         ignored.

        o If a 5xx or 6xx response is received, the particular server
         responding is removed from the parallel search and the search
         continues.  If a "603 Alternative Address" response is
         received, the search may be expanded to include those servers
         listed in the response that have not already responded. No
         response other than a periodic "100 Trying" response should be
         send towards the initiator unless there are no other servers
         left to try, in which case a response SHOULD be sent as
         described below.

        o If a 1xx response is received, the search continues. The 1xx
         response MAY be forwarded towards the initiator as appropriate.

   If a proxy had exhausted its search and still not obtained a
   definitive response (it received only 1xx, 5xx, and 6xx responses)
   the proxy should cache these responses and return the first response
   from the following ordered list:

        1.   503 Service Unavailable;

        2.   500 Server Internal Error;

        3.   501 Not Implemented;

        4.   any other 5xx error not yet defined;

        5.   600 Search Failure;

        6.   602 Not Currently Here;

        7.   601 Not Known Here;

        8.   any other 6xx error response not yet defined.

   If a proxy has exhausted its search and the only response it has
   received has been "603 Alternative Address", then the proxy should
   send a "600 Search Failure" response if any connection attempt timed



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   out or failed, or it should send "602 Not Currently Here" if two or
   more "603 Alternative Address" responses only provide references to
   each other.

12.4 Change of Transport at a Proxy


        Editors note: this section is still incomplete. Several
        options exist for where the responsibility should lie for
        retransmissions from proxies between TCP and UDP transport.
        This section generally assumes local retransmission, but
        end-to-end transmission through a chain of proxies is also
        possible.

   It is possible that a proxy server will receiver a request using TCP
   and relay it onwards using UDP or vice-versa. SIP does not assume
   end-to-end reliability even when the initiating client is using TCP,
   but a SIP client sending a request over TCP MAY assume that the
   request has been received by the server it sent the request to.
   Retransmission of the request is then not the responsibility of the
   client. However, a called user agent SHOULD NOT assume that a 2xx
   success response has been received by the invitation initiator, even
   if all the path fields in the request indicated TCP transport because
   it cannot be certain all those TCP connections still exist. If the
   called user agent requires knowledge that the response did reach the
   invitation initiator, it MAY add a  Confirm: required field to the
   reply as it would if the response was sent using UDP.

   In the following, the term "TCP-UDP proxy" is used to mean a proxy
   that received a request using TCP and relayed it using UDP. Similarly
   a "TCP-UDP proxy" receives a reply using UDP and should relay it
   using TCP.

12.4.1 Retransmission from a TCP-UDP Proxy

   A proxy receiving a request with TCP transport and forwarding that
   request using UDP becomes responsible for retransmission of the
   request as required and for timing out the request if no answer is
   forthcoming.

12.4.2 Retransmissions arriving at a UDP-TCP Proxy

   A proxy receiving a request using UDP transport and forwarding that
   request using TCP transport may have have SIP request state
   associated with that TCP connection or SIP response state associated
   with it.

   If such a proxy receives a retransmission of the UDP request whilst



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   in the state or awaiting a response (i.e, has request state), it
   SHOULD NOT forward the duplicate request into the TCP connection
   unless the request has been modified, but instead SHOULD respond with
   a "100 Trying" response sent back towards the initiator.

   Note: This behavior is different from a UDP-UDP proxy which MUST
   forward the retransmitted request and MAY additionally respond with a
   "100 Trying" response sent back towards the initiator.

   If such a proxy receives a retransmission of the UDP request in
   response state (i.e, it has already sent a definitive response) then
   the proxy MAY retransmit that response if it has cached it.
   Alternatively if it has not cached the response, it SHOULD resend the
   request towards the called user agent, either via an existing TCP
   connection if there is one or via a new TCP connection if there is
   not, to obtain a retransmission of the response. In the latter case,
   the proxy MAY additionally respond with a "100 Trying" response sent
   back towards the initiator.

   Note: This behavior is the same as a UDP-to-UDP proxy in the same
   circumstances.

12.4.3 Confirmation arriving at a TCP-UDP Proxy

   One possible event that may occur is that whilst performing a search
   using UDP, a response may arrive that should be relayed back towards
   the initiator using TCP, but the TCP connection has been terminated
   by the initiator. In this case the proxy MUST NOT attempt to relay
   the response (by opening a TCP connection) and should terminate any
   outstanding search. In this circumstance only, if the response was a
   "200 OK" response with a  Confirm: required field, the proxy MAY
   resend the request to the Contact Host with a  Confirm: false field
   to speed hang-up discovery at the called user agent.

12.4.4 Confirmation sent from a UDP-TCP Proxy

   Normally a response that arrives at a proxy using TCP that should be
   sent back towards the initiator using UDP should be sent once, and
   should only be resent if the request is resent from the UDP proxy
   closer to the initiator. However, this does not allow for reliable
   confirmation.

13 Using Variants for Terminal Negotiation

   Redirection allows the called party to indicate several communication
   alternatives to the caller using the 300 (Multiple Choices) response,
   all reachable using a single published communication identifier.




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   The  Alternates header in the response contains the variant list.
   The response may contain an entity, typically of content type
   text/html, providing guidance to the user. The calling user agent is
   free to ignore this part and solely rely on the  Alternates header.


     SIP/2.0 300 Multiple Choices
     Date: Thu, 06 Mar 1997 10:08:55 GMT
     Alternates:
        {"hgs@erlang.cs.columbia.edu" 0.9 {mobility fixed} {class business}
          {service IP, voice-mail} {media all} {duplex full}},
        {"+12129397042" 0.8 {mobility fixed} {class business}
          {service POTS} {media audio} {duplex full}},
        {"+12129397000" 0.7 {mobility fixed} {class business}
          {service ISDN, attendant} {media audio} {duplex full}
          {language en, es, iw}},
        {"+12125551212" 0.6 {mobility mobile} {class personal}
          {service POTS} {media audio} {duplex full}}
       }
     Content-Type: text/html
     Content-Length: 283

     <html>
     You can reach <a href="http://www.cs.columbia.edu/~doe">John Doe</a> at

     <ul>
     <li><a href="sip://hgs@erlang.cs.columbia.edu">Internet telephony</a>

     <li><a href="phone://+1219397042">analog phone</a>

     <li>...

     </dl>
     </html>




13.1 Variant Description

   A variant can be described in a machine-readable way with a variant
   description [7].


     variant-description =
       "{" <"> UCI <"> communication-quality *variant-attribute "}"

     communications-quality = qvalue



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     variant-attribute = "{" "mobility"  ( "fixed" | "mobile" ) "}"
                       | "{" "class"     ( "personal" | "business" ) "}"
                       | "{" "language"  1#language-tag "}"
                       | "{" "service"   1#service-tag "}"
                       | "{" "media"     1#media-tag "}"
                       | "{" "features"  feature-list "}"
                       | "{" "description" quoted-string "}"
                       | "{" "duplex"    ( "full" | "half" | "receive-only" |
                                           "send-only" ) "}"
                       | extension-attribute

     extension-attribute = "{" extension-name extension-value "}"
     extension-name      = token
     extension-value     = *( token | quoted-string | LWS |
                              extension-specials )
     extension-specials  = <any element of tspecials except <"> and "}">

     language-tag        = <see [H3.10]>
     service-tag         = fax | IP | POTS | pager | voice-mail |
                           attendant
     media-tag           = <see SDP: audio | video | ... >
     feature-list        =



   Attributes which are unknown should be omitted. New tags for class-
   tag and  service-tag can be registered with IANA. The media tag uses
   Internet media types, e.g., audio, video, application/x-wb, etc. This
   is meant for indicating general communication capability, not the
   support for specific encodings. It should be sufficient to allow the
   caller to choose an appropriate communication address.

14 Acknowledgments

   We wish to thank the members of the IETF MMUSIC WG for their comments
   and suggestions. This work is based, inter alia, on [8,9].

15 Authors' Addresses

   Mark Handley
   USC Information Sciences Institute
   c/o MIT Laboratory for Computer Science
   545 Technology Square
   Cambridge, MA 02139
   USA
   electronic mail: mjh@isi.edu

   Henning Schulzrinne



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   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, MY 10027
   USA electronic mail: schulzrinne@cs.columbia.edu

   Eve Schooler
   Computer Science Department 256-80
   California Institute of Technology
   Pasadena, CA 91125
   USA
   electronic mail: schooler@cs.caltech.edu

16 Bibliography

   [1] M. Handley, "SDP: Session description protocol," Internet Draft,
   Internet Engineering Task Force, Nov. 1996.  Work in progress.

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

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

   [4] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," Internet Draft, Internet Engineering Task Force, Jan. 1997.
   Work in progress.

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

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

   [7] K. Holtman and A. Muntz, "Transparent Content Negotiation in
   HTTP," Internet Draft, Internet Engineering Task Force, Nov. 1997.
   Work in progress.

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

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



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