Internet Engineering Task Force                                MMUSIC WG
Internet Draft                    M. Handley, H. Schulzrinne, E. Schooler
ietf-mmusic-sip-02.txt                              Handley/Schulzrinne/Schooler
draft-ietf-mmusic-sip-03.txt                           ISI/Columbia U./Caltech
March 27,
July 31, 1997
Expires: September 25, 1997 January 20, 1998

                    SIP: Session Initiation Protocol

STATUS OF THIS MEMO

   This document is an Internet-Draft. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
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   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. scheme. In particular, it aims to enable user
         mobility by relaying and redirecting invitations to a
         user's current location.

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

   Authors' Note

   This document is the result of a merger

1 Introduction

1.1 Overview of the SIP Functionality

   The Session Invitation Initiation Protocol (draft-ietf-mmusic-sip-00.txt) (SIP) is an application-layer
   protocol that can establish and the Simple Conference
   Invitation Protocol (draft-ietf-mmusic-scip-00.txt), control multimedia sessions or calls.
   These multimedia sessions include multimedia conferences, distance
   learning, Internet telephony and of an
   attempt to make similar applications. SIP more generic and to fit into a more flexible
   infrastructure that includes companion protocols including SDP, HTTP can
   initiate both unicast and RTSP.

   Changes

   Since version -01, multicast sessions; the following things initiator does not
   necessarily have changed:

        o CAPABILITIES to OPTIONS for closer alignment with HTTP be a member of the session. Media and
         RTSP;

        o Path
   participants can be added to Via for closer alignment with HTTP an existing session. SIP can be used to
   "call" both persons and RTSP;

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

        o Formatting closer
   storage device to HTTP and RTSP.

        o Explain relationship record an ongoing conference or to H.323.

1 Introduction

   There are two basic ways invite a video-
   on-demand server to locate play a video into a conference. (SIP does not
   directly control these services, however; see RTSP [4].)

   SIP transparently supports name mapping and redirection services,
   allowing the implementation of telephony services such as selective
   call forwarding, selective call rejection, conditional and
   unconditional call forwarding, call forwarding busy, call forwarding
   no response. SIP may use multicast to participate in a multimedia
   session:

        o The session try several possible callee
   locations at the same time.

   SIP supports personal mobility telecommunications intelligent network
   services, this is advertised, defined as:  "Personal mobility is the ability of
   end users see to originate and receive calls and access subscribed
   telecommunication services on any terminal in any location, and the advertisement, then
         join
   ability of the session address network to participate.

        o Users are invited identify end users as they move. Personal
   mobility is based on the use of a unique personal identity (i.e.,
   'personal number')." [1].  Personal mobility complements terminal
   mobility, i.e., the ability to participate in maintain communications when moving a session, which may
   single end system from one network to another.

   SIP supports some or
         may not already be advertised.

   The Session Description Protocol (SDP) [1] together with all of four facets of establishing multimedia
   communications:

        1.   user location: determination of the Session
   Announcement Protocol (SAP) [2], provide a mechanism end system to be used
             for communication;

        2.   user capabilities: determination of the former.
   This document presents the Session Initiation Protocol (SIP) media and media
             parameters to
   perform be used;

        3.   user availability: determination of the latter. SIP MAY also use SDP to describe a session.

                      Figure omitted in ASCII version

   Figure 1: Session Lifecycle
   We make willingness of the design decision that how a user discovers that a session
   exists is orthogonal
             called party to a session's conference control model.  Figure
   1 shows a potential place for SIP engage in the lifecycle communications;

        4.   call setup ("ringing", establishment of call parameters at
             both
   lightweight sessions called and in more tightly-coupled conferencing. Note
   that the Session Initiation Protocol calling party)
        In particular, SIP can be used to locate a user and determine
        the Session Announcement
   Protocol appropriate end system, leaving the actual call
        establishment to other protocols such as H.323.

   SIP 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 terminate and transfer a recording server call. SIP can
   be invited to participate in also
   initiate multi-party calls using a live multimedia session to record that
   session, multipoint control unit (MCU) or
   fully-meshed interconnection instead of multicast.

        These features are for further study.

   SIP is not a video-on-demand server conference control protocol, but can be invited used to
   introduce conference control protocols to play a video
   stream into a live multimedia conference. In such cases we would like session.

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

1.2 Finding Multimedia Sessions

   There are two basic ways to locate and playback.

   We also make the design decision that inviting a user to participate in a multimedia
   session:

   Advertisement: The session is independent of quality of service (QoS) guarantees
   for that session. Such QoS guarantees (if they advertised, potential participants see
        the advertisement, then join the session address to participate.

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

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

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

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

   SIP can invite users to conferences with and without resource
   reservation. SIP does not reserve resources, but may convey to the
   invited system the information necessary to do this. Quality-of-
   service guarantees, if required, may depend on knowing the full
   membership of the session, and session; this information may or may not be known
   to the agent performing session invitation.

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

1.1 Requirements

   The [7].

1.3 Terminology

   In this document, 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 2119 [8] and
   indicate requirement levels for compliant SIP implementations.

1.4 Definitions

   This specification uses a number of terms to refer to the roles
   played by participants in SIP communications. The definitions of
   client, server and proxy are similar to those used by HTTP. the Hypertext
   Transport Protocol (HTTP) [9].

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

   Initiator:

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

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

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

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

   Location server: A program that is contacted by a  ->  client and
        that returns one or more possible locations for of the user or service
        without contacting that user called party
        or service directly. service. Location servers may be invoked by SIP redirect and
        proxy servers and may be Co-located with a SIP server.

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

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

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

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

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

   User Agent, Called User Agent: The server user agent servers. An
        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 act as a client both server and client.

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

   (SIP) transaction: A SIP transaction occurs between a  ->  client and
        a  ->  server and comprises all messages from the first request
        sent from the client to invite others the server up to conferences and as a  ->  final (non-1xx)
        response sent from the server to accept invitations.

1.3 General Requirements

   SIP the client. A transaction is
        for a Session Initiation Protocol. It is not single call (identified by a conference control
   protocol. SIP  Call-ID, Section 6.11).
        There can only be used to perform one pending transaction between a search server and
        client for a each call id.

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

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

   Once SIP has been used client and
   a server. For example, a typical multimedia conference control
   application would act as a client to initiate calls or to invite
   others to conferences and as a multimedia session SIP's task is
   finished. user agent server to accept
   invitations.

1.5 Protocol Properties

1.5.1 Minimal State

   There is no concept of a an ongoing SIP session (as opposed to a SIP
   search that lasts for the
   duration of the conference or call. Rather, a user single conference
   session or service). If whatever call may involve one or more SIP request-response
   transactions. For example, a conference control
   mechanism is used in the session needs protocol may use SIP
   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.

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

1.5.2 Transport-Protocol Neutral

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

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

   From a pragmatic point of view,  TCP allows easier passage through existing
   firewalls, and with appropriate given the similar protocol design, allows common
   servers for SIP, HTTP and RTSP servers. the Real Time Streaming Protocol (RTSP)
   [4].

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

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

1.5.3 Text-Based

   SIP is text based. This allows easy implementation in languages such
   as TCL Tcl and Perl, allows easy debugging, and most importantly, makes
   SIP flexible and extensible. As SIP is only primarily 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

1.6 SIP -- in
   a minimal implementation the initiator maintains all the state about
   the current attempt to locate Addressing

   SIP uses two kinds of address identifiers, host-specific addresses
   and contact a host-independent addresses form user@host , where 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 any
   alphanumeric identifier and 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 form of host depends on the currently evolving Real-
        Time Stream Control Protocol.

1.4 Addressing address
   type. Note that SIP is does not distinguish between the two and can,
   while inviting a protocol that exchanges messages user, map repeatedly between peer user agents or
   proxies for user agents. We assume the two address types.

   For a host-specific address, the user agent part is an application
   that acts on behalf of the operating-system
   user it represents (thus it is sometimes
   described as a client of the user) and that name. The host part is co-resident with that
   user. A proxy for either a user agent serves as domain name having a forwarding mechanism DNS A
   (address) record, 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 ). address. Examples include:

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

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

   However, users also publish several well-known

   Host-independent addresses that are
   relatively location-independent, such contain a moniker (such as email a civil name)
   or web home-page
   addresses. Rather than require that users provide their specific
   network locales, we can take advantage of email user name and web domain name that may not map into a single host.
   [1]

   Host-independent addresses may use any unambiguous user name,
   including aliases, identifying the called party as
   being (relatively) memorable, and also leverage off the Domain Name
   Service (DNS) user part of
   the address. They may use a domain name having an MX [10], SRV [11]
   or A [12] record for the host part.  These addresses may have
   different degrees of location- and provider-independence and are
   often chosen to be mnemonic. In many cases, the host-independent SIP
   address can be the same as a user's electronic mail address, but this
   is not required. SIP can thus leverage off the domain name system
   (DNS) to provide a first stage first-stage location mechanism. Note that
   an email address ( mechanisms. Examples of
   host-independent names include

     M.Handley@cs.ucl.ac.uk ) is usually different from
     H.G.Schulzrinne@ieee.org
     info@ietf.org

   An address can designate an individual (possibly located at one of
   several end systems), the combination first available person from a group of
   individuals or a specific machine name and login name (
   mjh@mercury.lcs.mit.edu ). SIP should allow both forms whole group. The form of addressing
   to be used, with the former requiring address, e.g.,
_________________________
  [1] We avoid the term  location-independent  ,  since
the  address  may  indeed refer to a location server specific location,
e.g., a company department.

   sales@example.com , is not sufficient, in general, to locate determine the
   user.

   One perceived problem
   intent of email addressing is the caller.

   If a user or service chooses to be reachable at an address that it is possible to
   guess peoples' addresses
   guessable from the person's name and thus organizational affiliation, the system
   traditional method of ensuring privacy by having an unlisted (in the
   telephone directory) numbers "phone"
   number is more of a problem. compromised. However, this
   really only provides security through obscurity, and real security is
   better provided through unlike traditional telephony, SIP
   offers authentication and access control mechanisms and can avail
   itself of lower-layer security mechanisms, so that client software
   can reject unauthorized or undesired call screening.

1.5 Call Setup attempts.

1.7 Locating a SIP Server

   Call setup is a multi-phase procedure. In the first phase, the
   requesting may proceed in several phases. A SIP client tries to ascertain MUST follow
   the address where it should
   contact following steps to resolve the remote user agent or user agent proxy. The local part of a callee address. If
   a client
   checks if the user only supports TCP or UDP, but not both, the respective
   address type is location-specific. omitted.

        1.   If so, then that there is a SRV DNS resource record [11] of type sip.udp
             , contact the address used for the remote user agent. If not, the requesting
   client looks up the domain part listed SIP servers in order of preference
             value using UDP as a transport protocol at the user address port number
             listed in the DNS. This
   provides one or more records giving IP addresses. DNS resource record.

        2.   If there is a new service
   (SRV) SRV DNS resource record [5] is returned giving [11] of type sip.tcp
             , contact the listed SIP servers in order of preference
             value using TCP as a location server, then
   that transport protocol at the port number
             listed in the DNS resource record.

        3.   If there is a DNS MX record [10], contact the address hosts listed
             in their order of preference at the default port number
             (TBD).  For each host listed, first try to contact next. If no relevant resource record the
             server using UDP, then TCP.

        4.   Finally, check if there is returned, but an a DNS CNAME or A record is returned, then that is for the address
             given host and try to contact next. If neither a resource record SIP server at the one or an A record is
   returned, but an MX record is returned,
             more addresses listed, again trying first UDP, then TCP.

        5.   If all of the mail host is above methods fail, the
   address to caller MAY contact next.

   Presuming an address for the invitee is found from
             SMTP server at the DNS, user's host and use the
   second SMTP  EXPN
             command to obtain an alternate address and subsequent phases basically implement repeat the steps
             above. As a request-response
   protocol.  A last resort, a client MAY choose to deliver the
             session description (typically using SDP format) is sent to the contact address with an invitation for callee using electronic mail.

   If a server is found using one of the user to join methods below, 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.

   If a user agent or conference server is listening other
   methods are not tried. A client SHOULD rely 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 "Port
   Unreachable" messages rather than time-outs to fail or cause a UDP
   send failure on retransmissions.

1.6 Locating a User

   It is expected determine that a user
   server is situated not reachable 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 particular address. A client MAY cache
   the result of these locations if so desired. It is entirely up to the
   location server whether reachability steps, but SHOULD start at the server issues proxy requests for
   beginning of the
   requesting user, or if sequence when the cached address fails.

   Implementation note for socket-based programs: For TCP, connect()
   returns ECONNREFUSED if there is no server instructs at the client to redirect designated address;
   for UDP, the request.

   In general a reply MUST socket should be sent by bound to the same mechanism destination address using
   connect() rather than sendto() or similar.

        This sequence is modeled after that described for SMTP,
        where MX records are to be checked before A records [13].

1.8 SIP Transactions

   Once the
   request was sent by. Hence, if host part has been resolved to a request was unicast, then SIP server, the reply
   MUST be unicast back client
   sends one or more SIP requests to that server and receives one or
   more responses from the requester; if server. If the invitation is SIP request was multicast,
   the reply MUST be multicast to is
   an invitation, it contains a session description, for example written
   in SDP format, that provides the same group called party with enough information
   to which the request
   was sent; if join the session.

   If TCP is used, request was sent by TCP, the reply MUST be sent by
   TCP.

   In all cases where and responses within a request is forwarded onwards, each host relaying single SIP transaction
   are carried over the message SHOULD add its own address to same TCP connection. Thus, the path of client SHOULD
   maintain the message so
   that connection until a final response has been received.
   Several SIP requests from the replies can take same client to the same path back, thus ensuring correct
   operation through compliant firewalls and loop-free requests. On server may use
   the
   reply path, these routing headers MUST be removed as same TCP connection or may open a new connection for each
   request. If the reply
   retraces client sent the path, so that routing internal to sites is hidden. When
   a multicast request sends via unicast UDP, the
   response is made, first sent to the host making source address of the request, then UDP request. If the
   request is sent via multicast address itself are added to the path.

2 Notational Conventions and Generic Grammar

   Since many of UDP, the definitions and syntax are identical to HTTP/1.1,
   this specification only points response is directed to the section where they are defined
   rather than copying it.
   same multicast address and destination port. For brevity, [HX.Y] UDP, reliability is
   achieved using retransmission (Section 11).

        Need motivation why we ALWAYS want to be taken to refer
   to Section X.Y have a multicast
        return.

   The SIP message format and operation is independent of the current HTTP/1.1 specification (RFC 2068).

   All the mechanisms specified in this document are described transport
   protocol.

   The basic message flow is shown 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 Fig. 1 and smaller-type paragraphs to provide
   background Fig. 2, for proxy and motivation.

3 Protocol Parameters

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
   redirect modes, respectively.

1.9 Locating a User

   A callee may move between a 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 different end systems over TCP or UDP, multiple requests
   time.  These locations can be carried in dynamically registered with a location
   server, typically for a single
   TCP connection or UDP datagram. UDP Datagrams should not normally
   exceed the path MTU in size if it is known, administrative domain, or 1,000 bytes if the MTU
   is unknown.

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

   ====> SIP messages consist request
   ----> non-SIP protocols

   Figure 1: Example of requests from client to SIP proxy server and responses
   from

   server to client.

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

   Request (section 5) and response (section 6) messages may use other protocols, such as finger [14], rwho,
   multicast-based protocols or operating-system dependent mechanism to
   actively determine the generic
   message format of RFC 822 for transferring entities (the payload of
   the message). Both types of messages consist end system where a user is reachable. The
   location services yield a list of a start-line, one zero or more header fields (also known as "headers"), an empty line (i.e., a
   line with nothing preceding the CRLF) indicating the end possible locations,
   possibly even sorted in order of the
   header fields, and an optional message-body.

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

     start-line      = Request-Line | Status-Line
   In the interest likelihood of success.

   The location server can be part of robustness, servers SHOULD ignore any empty
   line(s) received where a Request-Line is expected. In other words, if the SIP server is reading or the SIP server
   may use a different protocol stream (e.g., finger [14] or LDAP [15]) to map
   addresses. A single user may be registered at different locations,
   either because she is logged in at several hosts simultaneously or
   because the beginning of a
   message and receives location server has (temporarily) inaccurate information.

   The action taken on receiving a CRLF first, it should ignore list of locations varies with the CRLF.

4.2 Message Headers

   HTTP header fields, which include general-header (section ),
   request-header (section ), response-header (section ), fields, follow
   type of SIP server. A SIP redirect server simply returns the same generic format list to
   the client sending the request as that given in Section 3.1 of RFC 822. Each
   header field consists of a name followed by a colon (":") and  Location headers (Section 6.17). A
   SIP proxy server can sequentially try the
   field value. Field names are case-insensitive. The field value may be
   preceded by any amount of LWS, though a single SP addresses until the call is preferred.
   Header fields
   successful (2xx response) or the callee has declined the call (40x
   response). Alternatively, the server may issue several requests in
   parallel. A proxy server can be extended over multiple lines by preceding each
   extra line with at least only issue more than one SP sequential or HT. Applications SHOULD follow
   "common form" when generating HTTP constructs, since there might
   exist some implementations that fail to accept anything beyond
   parallel connection request if it is the
   common forms.

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

     field-name     = token
     field-value    = *( field-content | LWS )
     field-content  = <the OCTETs making up first in the field-value
                      and consisting chain of either *TEXT or combinations hosts
                                            +....... cs.columbia.edu .......+
                                            :                               :
                                            : (~~~~~~~~~~)                  :
                                            : ( location )                  :
                                            : ( service  )                  :
                                            : (~~~~~~~~~~)                  :
                                            :   ^      |                    :
                                            :   |   hgs@play                :
                                            :  2|     3|                    :
                                            :   |      |                    :
                                            : henning  |                    :
   +.. cs.tu-berlin.de ..+ 1: INVITE        :   |      |                    :
   :                     :    henning@cs.col:   |      |                    :
   : cz@cs.tu-berlin.de =======================>  tune                      :
   :         ^ |        <.......................                            :
   :         . |         : 4: 302 Moved     :                               :
   +...........|.........+    hgs@play      :                               :
             . |                            :                               :
             . | 5: INVITE hgs@play.cs.columbia.edu                6: ring  :
             . ==================================================> play     :
             .....................................................          :
               7: 200 OK                    :                               :
                                            +...............................+

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

   Figure 2: Example of token, tspecials, and quoted-string>

   The order SIP redirect server

   noted in which the  Via header fields with differing field names are
   received to do so. If it is not significant.

   Multiple message-header fields with the same field-name may be
   present in first, it must
   issue a message if and only if the entire field-value for that
   header field is defined as redirect response.

   If a comma-separated list (i.e., #(values) ).
   It MUST be possible proxy server forwards a SIP request, it SHOULD add itself to combine the multiple header fields into one
   "field-name:  field-value" pair, without changing
   end of the semantics list of forwarders noted in the message, by appending each subsequent field-value  Via (Section 6.31)
   headers. A proxy server also notes whether it is attempting to the first,
   each separated by a comma. reach
   several possible locations at once ("connection forking"). The order in which header fields with  Via
   trace ensures that replies can take the same field-name are received is therefore significant to path back, thus ensuring
   correct operation through compliant firewalls and loop-free requests.
   On the
   interpretation of reply path, each host most remove its Via, so that routing
   internal information is hidden from the combined field value, callee 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.

   The presence of a message-body in outside networks.
   When a multicast request is signaled by made, first the
   inclusion of a Content-Length or Transfer-Encoding header field in host making the request's message-headers. A message-body MAY be included request,
   then the multicast address itself are added to the path.

   As discussed in Section 1.6, a
   request only when SIP address may designate a group
   rather than an individual. A client indicates using the  Reach
   request method allows an entity-body.

   For response messages, header whether it wants to reach the first available
   individual or not treat the address as a message-body is included with group, to be invited as a message whole.
   The default is dependent on both to attempt to reach the request method and first available callee.  If
   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 address is included with designated as a message, group address, a proxy server MUST
   return the length of that
   body is determined by one list of the following (in order individuals instead of precedence):

        1.   Any response message which MUST NOT include a message-body
             (such as the 1xx responses) is always terminated by attempting to connect to
   these.

        Otherwise, the
             first empty line after proxy cannot report errors and call status
        appropriately.

2 SIP Uniform Resource Locators

   SIP URLs are used within SIP messages to indicate the header fields, regardless originator and
   recipient 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 SIP 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 specify redirection addresses. A
   SIP URL may be extended reliably only embedded in
   combination with web pages or other hyperlinks to indicate
   that a change in the protocol version. However, new user or
   experimental header fields service may be given the semantics of general
   header fields if all parties in the communication recognize them called. Within SIP messages, an email
   address could have been used, but this would have made it more
   difficult to
   be general-header fields.

5 Request
   The Request-Line begins with a method token, followed by the
   Request-URI and the protocol version, gateway between SIP and ending other protocols 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 other
   addressing schemes.

   For greater functionality, because interaction with some resources
   may be either  INVITE require message headers or  CAPABILITY. The request ID may
   be any URL encoded string that can be guaranteed message bodies to be globally
   unique for specified as well
   as the duration of SIP address, the request. Using sip URL scheme is extended to allow setting
   SIP request-header fields and the initiator's IP-
   address, process id, SIP  message-body.

   A SIP URL follows the guidelines of RFC 1630 [16,17] 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, takes the recipient
   responds with an SIP response message.

     Response
   following form:

        SIP-URL            =    short-sip-url | full-sip-url
        full-sip-url       =    "sip://" user [ ":" password ] "@" host
                                url-parameters [ headers ]
        short-sip-url      =    user [ ":" password ] "@" host : port
        user               = Status-Line    ; Section
                *( general-header  defined in RFC 1738 [18]
        host               =    ; Section  defined in RFC 1738
        port               =    *digit
        url-parameters     =    *( ";" url-parameter)
        url-parameter      =    transport-param | response-header      ; Section
                                ttl-param | entity-header maddr-param
        transport-param    =    "transport=" ( "udp" | "tcp" )
        ttl-param          =    "ttl=" ttl
        ttl                =    1*3DIGIT                                     ; Section
                CRLF
                [ message-body ] 0 to 255
        maddr-param        =    "maddr=" maddr
        maddr              =    ; Section

6.1 Status-Line

   The first line of  dotted decimal multicast address
        headers            =    "?" header *( "                            " header )
        header             =    hname "=" hvalue
        hname              =    *urlc
        hvalue             =    *urlc
        urlc               =    ;  defined in [17]

   Thus a Response message is the Status-Line , consisting
   of the protocol version followed by SIP URL can take either a numeric status code, the
   sequence number of short form or a full form. The short
   form MAY only be used within SIP messages where the corresponding request scheme (SIP) can
   be assumed. In all other cases, and when parameters are required to
   be specified, 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. full form MUST be used.

   Note that the addition of a

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

6.1.1 Status Code and Reason Phrase all URL reserved characters must be encoded. The Status-Code element special
   hname  "body" indicates that the associated  hvalue is a 3-digit integer result code of the
   attempt to understand and satisfy message-
   body of the SIP  INVITE request. These codes Within sip URLs, the characters
   "?",  "=",  "&" are fully
   defined in section10. The Reason-Phrase is intended to give a short
   textual description reserved.

   Examples of the Status-Code. The Status-Code is intended
   for use by automata short and the Reason-Phrase is intended for the human
   user. full form SIP URLs with identical address are:

     j.doe@big.com
     sip://j.doe@big.com
     sip://j.doe:secret@big.com;transport=tcp
     sip://j.doe@big.com?subject=project

   The client is not required  password parameter allows 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 easily specify a call-back address
   on a secure web page, but carries the first digit:

        o 1xx: Informational - Request received, continuing process

        o 2xx: Success - The action was successfully received,
         understood, same security risks as all
   URL-based passwords and accepted

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

        o 4xx: Client Error - The request contains bad syntax used under special
   circumstances where security requirements are low or cannot
         be fulfilled

        o 5xx: Server Error - The server failed all transport
   paths are secured.

   Within a SIP message, URLs are used to fulfill an apparently
         valid request

   The individual values of indicate the numeric status codes defined for
   SIP/2.0, source and an example set
   intended destination of corresponding Reason-Phrase below. The
   reason phrases listed here a request, redirection addresses and the
   current destination of a request. Normally all these fields will
   contain SIP URLs. When additional parameters are only recommended -- they not required, the
   short form SIP URL can be used unambiguously.

   In some circumstances a non-SIP URL may be
   replaced used in a SIP message. An
   example might be making a call from a telephone which is relayed by local equivalents without affecting a
   gateway onto the protocol. Note
   that internet as a SIP adopts many HTTP/1.1 status codes and adds SIP-specific
   status codes in request. In such a case, the starting at 450 to avoid
   source of the call is really the telephone number of the caller, and
   so a SIP URL is inappropriate and a phone URL might be used instead.
   Thus where SIP specifies user addresses it allows these addresses to
   be URLs.

   Clearly not all URLs are appropriate to be used in a SIP message as a
   user address. It is unlikely, for example, that HTTP or FTP URLs are
   useful in this context. The correct behavior when an unknown scheme
   is encountered by a SIP server is defined in the context of each of
   the header fields that use a SIP URL.

   SIP URLs can define specific parameters of the request, including the
   transport mechanism (UDP or TCP) and the use of multicast to make a
   request. These parameters are added after the  host and are separated
   by semi-colons. For example, to specify to call j.doe@big.com using
   multicast to 239.255.255.1 with a ttl of 15, the following URL would
   be used:

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

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

3 SIP Message Overview

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

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

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

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

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

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

        start-line         =    Request-Line | Status-Line

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

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

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

4 Request

   The  Request message format is shown below:

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

   Table 1: SIP headers

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

4.1 Request-Line

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

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

4.1.1 Methods

   The following methods are defined:

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

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

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

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

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

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

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

        Beyond its use as a simple location service, this method is
        needed if there are several SIP servers on a single host,
        so that some cannot use the default port number. Each such
        server would register with a server for the administrative
        domain.

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

        BYE and REGISTER are experimental and need to be discussed.

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

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

4.1.2 Request-URI

   The  Request-URI field is a SIP URL as described in Section 2 or a
   general URI. It indicates the user or service that this request is
   being addressed to. Unlike the  To field, the  Request-URI field may
   be re-written by proxies. For example, a proxy may perform a lookup
   on the contents of the  To field to resolve a username from a mail
   alias, and then use this username as part of the  Request-URI field
   of requests it generates.

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

4.1.3 SIP Version

   Both request and response messages include the version of SIP in use,
   and basically follow [H3.1], with HTTP replaced by SIP. To be
   conditionally compliant with this specification, applications sending
   SIP messages MUST include a  SIP-Version of "SIP/2.0".

5 Response

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

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

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

5.1 Status-Line

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

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

5.1.1 Status Codes and Reason Phrases

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

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

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

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

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

   4xx: Client Error -- the request contains bad syntax or cannot be
        fulfilled at this server;
   5xx: Server Error -- the server failed to fulfill an apparently valid
        request;

   6xx: Global Failure - the request is invalid at any server.

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

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

6 Header Field Definitions

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

   To,  From, and  Call-ID header MUST be present in each request with
   method  INVITE. The  Content-Type and Content-Length headers are
   required when there is a valid message body (of non-zero length)
   associated with the message (Section 8).

   A server MUST understand the  PEP-Require header.

   Other headers may be added as required; a server MAY ignore headers
   that it does not understand. A compact form of these header fields is
   Status-Code       =    "100"                         ;  Trying
                    |     "180"                         ;  Ringing
                    |     "200"                         ;  OK
                    |     "300"                         ;  Multiple Choices
                    |     "301"                         ;  Moved Permanently
                    |     "302"                         ;  Moved Temporarily
                    |     "303"                         ;  See Other
                    |     "305"                         ;  Use Proxy
                    |     "380"                         ;  Alternative Service
                    |     "400"                         ;  Bad Request
                    |     "401"                         ;  Unauthorized
                    |     "402"                         ;  Payment Required
                    |     "403"                         ;  Forbidden
                    |     "404"                         ;  Not Found
                    |     "405"                         ;  Method Not Allowed
                    |     "407"                         ;  Proxy Authentication Required
                    |     "408"                         ;  Request Timeout
                    |     "409"                         ;  Conflict
                    |     "410"                         ;  Gone
                    |     "411"                         ;  Length Required
                    |     "412"                         ;  Precondition Failed
                    |     "413"                         ;  Request Message Body Too Large
                    |     "414"                         ;  Request-URI Too Large
                    |     "415"                         ;  Unsupported Media Type
                    |     "420"                         ;  Bad Extension
                    |     "480"                         ;  Temporarily not available
                    |     "500"                         ;  Internal Server Error
                    |     "501"                         ;  Not Implemented
                    |     "502"                         ;  Bad Gateway
                    |     "503"                         ;  Service Unavailable
                    |     "504"                         ;  Gateway Timeout
                    |     "505"                         ;  SIP Version not supported
                    |     "600"                         ;  Busy
                    |     "603"                         ;  Decline
                    |     "604"                         ;  Does not exist anywhere
                    |     "606"                         ;  Not Acceptable
                    |     extension-code
   extension-code    =    3DIGIT
   Reason-Phrase     =    *<TEXT,  excluding CR, LF>

   Figure 3: Status Codes

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

6.1 General Header Fields

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

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

6.2 Entity Header Fields

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

6.3 Request Header Fields

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

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

6.4 Response Header Fields

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

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

6.5 Header Field Format

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

   Each header field consists of a name followed by a colon (":") and
   the field value. Field names are case-insensitive. The field value
   may be preceded by any amount of leading white space (LWS), though a
   single space (SP) is preferred. Header fields can be extended over
   multiple lines by preceding each extra line with at least one  SP or
   horizontal tab (HT). Applications SHOULD follow HTTP status codes.

      Status-Code "common form"
   when generating these constructs, since there might exist some
   implementations that fail to accept anything beyond the common forms.

        message-header    = "100"   ; Continue
                     | "200"   ; OK    field-name ":" [ field-value ] CRLF
        field-name        =    token
        field-value       =    *( field-content | "300"   ; LWS )
        field-content     =    < the OCTETs  making up the field-value
                                and consisting of either *TEXT or combinations
                                of token, tspecials, and quoted-string>

   The order in which header fields are received is not significant if
   the header fields have different field names. Multiple Choices
                     | "301"   ; Moved Permanently
                     | "302"   ; Moved Temporarily
                     | "303"   ; header fields
   with the same field-name may be present in a message if and only if
   the entire field-value for that header field is defined as a comma-
   separated list (i.e., #(values) ). It MUST be possible to combine the
   multiple header fields into one "field-name: field-value" pair,
   without changing the semantics of the message, by appending each
   subsequent field-value to the first, each separated by a comma. The
   order in which header fields with the same field-name are received is
   therefore significant to the interpretation of the combined field
   value, and thus a proxy MUST NOT change the order of these field
   values when a message is forwarded.

   Field names are not case-sensitive, although their values may be.

6.6 Accept

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

   Example:

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

6.7 Accept-Language

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

   Example:

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

6.8 Allow

   See Other
                     | "305"   ; Use Proxy
                     | "400"   ; Bad Request
                     | "401"   ; Unauthorized
                     | "402"   ; Payment Required
                     | "403"   ; Forbidden
                     | "404"   ; Not Found
                     | "405"   ; Method Not Allowed
                     | "406"   ; Not Acceptable
                     | "407"   ; Proxy [H14.7].

6.9 Authorization

   See [H14.8].

6.10 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

   Authentication fields provide a digital signature of the remaining
   fields for authentication purposes. They are not supported
                     | extension-code

      extension-code yet defined The use
   of authentication headers is optional. If used, authentication
   headers MUST be added to the header after the  Via fields and before
   the rest of the fields.

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

6.11 Call-ID

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

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

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

        Call-ID    = 3DIGIT

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

   Example:

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

6.12 Content-Length

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

        Content-Length = *<TEXT, excluding CR, LF>

   SIP status codes are extensible. SIP applications are not required "Content-Length" ":" 1*DIGIT

   An example is

     Content-Length: 3495

   Applications SHOULD use this field to
   understand indicate the meaning size of all registered status codes, though such
   understanding the
   message-body to be transferred, regardless of the media type of the
   entity. Any  Content-Length greater than or equal to zero is obviously desirable. However, applications MUST
   understand a valid
   value. If no body is present in a message, then the class Content-Length
   header MAY be omitted or set to zero.  Section 8 describes how to
   determine the length of any status code, as indicated by the first
   digit, message body.

6.13 Content-Type

   The  Content-Type entity-header field indicates the media type of the
   message-body sent to the recipient.

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

   An example of the field is

     Content-Type: application/sdp

6.14 Date

   See [H14.19].

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

6.15 Expires

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

   This header field is currently defined only for the  REGISTER and treat any unrecognized response as being equivalent
   INVITE methods.  For  REGISTER, it is a response-header field and
   allows the server to indicate when the
   x00 status code of that class, client has to re-register. For
   INVITE, it is a request-header with which the exception that callee can limit the
   validity of an
   unrecognized response MUST NOT be cached. For invitation. (For example, if an
   unrecognized status code of 431 a client wants to limit
   how long a search should take at most or when a conference being
   invited to is received by time-limited. A user interface may take this is as a
   hint to leave the client, it invitation window on the screen even if the user is
   not currently at the workstation.)

   The value of this field can
   safely assume that there was something wrong with be either an  HTTP-date or an integer
   number of seconds (in decimal), measured from the receipt of the
   request.

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

   Two example of its request use are

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

6.16 From

   Requests MUST and
   treat responses SHOULD contain a  From header field,
   indicating the response invitation initiator. The field MUST be a SIP URL as if it had received
   defined in Section 2. Only a 400 status code. In such
   cases, single initiator and a single invited
   user agents SHOULD present are allowed to the user the entity returned
   with the response, since that entity be specified in a single SIP request.  The sense
   of  To and  From header fields is likely maintained from request to include human-
   readable information which will explain
   response, i.e., if the unusual status.

6.1.2 Response Header Fields

   The response-header fields allow  From header is sip://bob@example.edu in the
   request recipient to pass
   additional information about the response which cannot then it is MUST also be placed sip://bob@example.edu in the Status-Line server response
   to that request.

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

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

        From = Location              ; Section
                       | Proxy-Authenticate  ; Section
                       | Public              ; Section
                       | Retry-After         ; Section
                       | Server              ; Section ( "From" | Vary                ; Section "f" ) ":" *1( ( SIP-URL | WWW-Authenticate    ; Section

   Response-header field names URL ) [ comment
        ] )

   Example:

     From: mjh@isi.edu (Mark Handley)

6.17 Location

   The  Location response header can be extended reliably only in
   combination used with a change in 2xx or 3xx response
   codes to indicate a new location to try. It contains a SIP URL giving
   the protocol version. However, new location or
   experimental header fields MAY be given the semantics of response-
   header fields if all parties in the communication recognize them username to
   be response-header fields. Unrecognized header fields are treated as
   entity-header fields.

7 try, or may simply specify addition
   transport parameters. For example, a "301 Moved Permanently" response
   SHOULD contain a  Location field containing the SIP Message Body

   The session description payload gives details of URL giving the session
   new location and username to try. However, a "302 Moved Temporarily"
   MAY give simply the user
   is same location and username that was being invited tried
   but specify additional transport parameters such as a multicast
   address to join. Its Internet media type MUST be given by
   the "Content-type:" try or a change of transport from UDP to TCP or vice
   versa.

   A user agent or redirect server sending a definitive, positive
   response (2xx), SHOULD insert a  Location response header field, and indicating
   the payload length in bytes
   MUST SIP address under which it is reachable most directly for future
   SIP requests. This may be given by the  Content-length header field. If address of the payload has
   undergone any encoding (such as compression) then this MUST be
   indicated by server itself or that of
   a proxy (e.g., if the  Content-encoding: header field, otherwise Content-
   encoding: MUST be omitted.

   The example below host is behind 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 firewall).

        Location              =    ( "Location" | "m" ) ( SIP-URL | URL )
                                   *( ";" location-params )
        extension-name       =     token
        extension-value      =     *( token | quoted-string | LWS | extension-specials)
        extension-specials   =      < any element of Mbone Engineering Issues
   e=mbone@somewhere.com
   c=IN IP4 224.2.0.1/127
   t=0 0
   m=audio 3456 RTP/AVP 0  tspecials except <"> >
        language-tag         =     <  see [H3.10] >
        service-tag          =     "fax" | "IP" | "PSTN" | "ISDN" | "pager" | "voice-mail
                                   | "attendant"
        media-tag            =      < see SDP: "audio" | "video" | ...
        feature-list         =      to be determined

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

   Examples:

     Location: sip://hgs@erlang.cs.columbia.edu ;service=IP,voice-mail
               ;media=audio ;duplex=full ;q=0.7
     Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed;
               language=en,es,iw ;q=0.5
     Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
               duplex=send-only;media=text; q=0.1

   Attributes which are unknown should be omitted. New tags for class-
   tag and  service-tag can be registered with IANA. The first line above states that this media tag uses
   Internet media types, e.g., audio, video, application/x-wb, etc. This
   is a SIP version 2.0 request. meant for indicating general communication capability, sufficient
   for the caller to choose an appropriate address.

6.18 Organization

   The via Organization request-header fields give conveys the hosts along the path from invitation
   initiator (the first element name of the list) towards the invitee. In the
   example above, the message was last multicast
   organization to which 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 callee belongs. It may be seen from inserted by
   proxies at the  Via header) boundary of an organization and may be used by client
   software to filter calls.

6.19 PEP

   This corresponds to the user being invited is
   schooler@cs.caltech.edu.

   In this case, the session description (as stated  PEP header in the Content-type
   header) is a Session Description Protocol (SDP). "Protocol Extension
   Protocol" defined in RFC XXXX. The header Protocol Extension Protocol (PEP)
   is terminated by an empty line extension mechanism designed to accommodate dynamic extension
   of applications such as SIP clients and is followed servers by the
   session description payload.

   If required, the session description can be encrypted using public
   key cryptography, software
   components.  The  PEP general header declares new headers and then can also carry private session keys for whether
   an application must or may understand them. Servers MUST parse this
   field and MUST return "420 Bad Extension" when there is a PEP
   extension of strength "must" (see RFC XXXX) that they do not
   understand.

6.20 Priority

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

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

   Example:

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

6.21 Proxy-Authenticate

   See [H14.33].

6.22 Proxy-Authorization

   See [H14.34].

6.23 Public

   See [H14.35].

6.24 Reach

   The  Reach request header field allows the case, four random bytes are added client to indicate whether
   it wants to reach the
   beginning of group identified by the session description before encryption and are
   removed after decryption but before parsing.

8 Methods

   The following methods are defined:

   INVITE: The user part of the
   address (value "all") or service the first available individual (value
   "first"). If not present, a value of "first" is being invited implied. The "do-
   not-forward" request prohibits proxies from forwarding the call to participate in
   another individual (e.g., the
        session. The session description given must be completely
        acceptable for a "200 OK" response call is personal or the caller does not
   want to be given. This method MUST
        be supported by a SIP server.

   OPTIONS: The user or service is being queried as shunted to its capabilities.
        A server that believes it can contact the user (such as a user
        agent where secretary if the user line is logged in and has been recently active)
        MAY respond to this request with a capability set. Support busy.)  Section 1.6
   describes the behavior of
        this method is OPTIONAL.

   Methods that are not supported by a proxy server SHOULD servers when resolving group aliases.

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

   Example:

     Reach: first, do-not-forward

        HS: This header is experimental.

6.25 Retry-After

   The  Retry-After response header field can be 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 used with a "501
   Not Implemented" "503
   Service Unavailable" response to indicate how long the service is
   expected to be returned.

9 Header Field Definitions
   SIP header fields are similar unavailable to HTTP header fields in both syntax
   and semantics. In general the ordering of the header fields is not of
   importance (with requesting client and with a "404
   Not Found" or "451 Busy" response to indicate when the exception called party
   may be available again. The value 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 this field to can be
   added either an
   HTTP-date or an integer number of seconds (in decimal) after the Via fields that will not be invalidated by proxies.
   Field names time
   of the response.

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

   Two examples of its use are not case-sensitive, although their values may be.

   Content-Length,  Content-Type,  To,  From

     Retry-After: Mon, 21 Jul 1997 18:48:34 GMT
     Retry-After: 120
   In the latter example, the delay is 2 minutes.

6.26 Sequence

   The  Sequence header fields are
   compulsory. Other fields may field MAY be added as required. Header fields MUST
   be separated by a SIP client making a
   request if it needs to distinguish responses to several consecutive
   requests sent with the same  Call-ID. A  Sequence field contains a
   single linefeed character. The header decimal sequence number chosen by the requesting client.
   Consecutive different requests made with the same  Call-ID MUST be
   separated from
   contain strictly monotonically increasing sequence numbers although
   the payload by an empty line (two linefeed
   characters). sequence space MAY NOT be contiguous. A compact form of these header fields is also defined in section 10.9
   for use over UDP when the request has server responding to fit into a single packet and
   size is an issue.

9.1 Accept

   See [H14.1]. This
   request containing a sequence number MUST echo the sequence number
   back in the response.

        Sequence = "Sequence" ":" 1*DIGIT

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

   Example:

     Sequence: 4711

6.27 Server

   See [H14.39].

6.28 Subject

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

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

   Example:

     Subject: Tune in - they are acceptable.

9.2 Accept-Language

   See [H14.4]. talking about your work!

6.29 To

   The  Accept-Language  To request header can be used to allow field specifies the client to indicate to invited user, with the
   same SIP URL syntax as the  From field.

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

   If a SIP server in which language it would
   prefer receives a request destined  To a URL indicating a
   scheme other than SIP and that is unknown to receive reason phrases. This may also be used as it, the server returns a hint
   "400 bad request" response.

   Example:

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

6.30 User-Agent

   See [H14.42].

6.31 Via

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

   In the call request path, an initiator MUST add its own  Via field to (e.g., for
   selecting a human operator).

9.3 Authentication

   Authentication fields provide a digital signature of each
   request. This  Via field MUST be the remaining
   fields for authentication purposes. They are not yet defined The use
   of authentication headers is optional. If used, authentication
   headers first field in the request. Each
   subsequent client or proxy that sends the message onwards MUST add
   its own additional  Via field, which MUST be added to the header after before any
   existing  Via fields. Additionally, if the message goes to a
   multicast address, an extra  Via fields and field is added before all the rest of others
   giving the fields.

        HS: Ordering multicast address and semantics needs work. Maybe we can recycle TTL.

   In the S/MIME work?

9.4 Confirm

   TBD.

9.5 Contact-Host
   TBD.

9.6 From

   The request header MUST contain return path,  Via fields are processed by a  From request-header field,
   indicating proxy or client
   according to the invitation initiator. The following rules:

        o If the first  Via field MUST be machine-
   usable, as defined my mailbox in RFC 822 (as updated by RFC 1123).
   Only a single initiator the reply received is the client's
         or server's local address, remove the  Via field and process
         the reply.

        o If the first  Via field in a single invited user reply you are allowed going to be
   specified in send is a single SIP request.

9.7 Retry-After

   The  Retry-After response-header
         multicast address, remove that  Via field can be used with before sending to the
         multicast address.

   These rules ensure that a 503
   (Service Unavailable) response client or proxy server only has to indicate how long check
   the service is
   expected first  Via field in a reply to see if it needs processing.

   When a reply passes through a proxy on the reverse path, that proxies
   Via field MUST be unavailable to removed from the requesting client and with reply.

   The format for a 404
   (Not Found) or 451* (Busy) response  Via header is:

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

   The "ttl" parameter is included only if the called party
   may be available again. address is a multicast
   address. The value of "fanout" parameter indicates that this proxy has
   initiated several connection attempts and that subsequent proxies
   should not do the same.

   Example:

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

6.32 Warning

   The  Warning response-header field can be either an
   HTTP-date or an integer number of seconds (in decimal) after is used to carry additional
   information about the time status of the a response.

     Retry-After Warning headers are sent
   with responses using:

        Warning          = "Retry-After"    "Warning" ":" 1#warning-value
        warning-value    =    warn-code SP warn-agent SP warn-text
        warn-code        =    2DIGIT
        warn-agent       =    ( HTTP-date | delta-seconds host [ ":" port ] )

   Two examples | pseudonym
                              ;  the name or pseudonym of its use are

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

   In the latter example, server adding
                              ;  the delay is 2 minutes.

9.8 Reason

   TBD.

9.9 To Warning header, for use in debugging
        warn-text        =    quoted-string

   A response may carry more than one  Warning header.

   The  To request-header field specifies  warn-text should be in a natural language and character set that
   is most likely to be intelligible to the invited user, with human user receiving the
   same syntax
   response. This decision may be based on any available knowledge, such
   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 location of the same path as cache or user, the requests, which assists in firewall traversal and other unusual
   routing situations. Initiators MUST add their own Path field to each
   request. This Path  Accept-Language field MUST be in a
   request, the first  Content-Language field in a response, etc. The default
   language is English and the request.
   Subsequent proxies SHOULD each default character set is ISO- 8859-1.

   Any server may add their own additional Path field
   which MUST  Warning headers to a response. New Warning
   headers should be added before after any existing Path fields. When  Warning headers. A proxy
   server MUST NOT delete any  Warning header that it received with a reply
   passes through
   response.

   When multiple  Warning headers are attached to a proxy on response, the reverse path, user
   agent SHOULD display as many of them as possible, in the order that proxies Path field
   MUST be removed from
   they appear in 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 response. If it is included only if not possible to display all of
   the address is a multicast address.

10 warnings, the user agent should follow these heuristics:

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

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

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

   Example:

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

6.33 WWW-Authenticate

   See [H14.46].

7 Status Code Definitions

   The response codes are consistent with, and extend, HTTP/1.1 response
   codes. Not all HTTP/1.1 response codes are appropriate, and only
   those that are appropriate are given here. Response codes not defined
   by HTTP/1.1 are marked with an asterisk, and have codes x50 x80 upwards to avoid clashes with future HTTP
   response codes, or 6xx which are
   not used by HTTP. codes. Also, SIP defines a new class, 6xx. The default
   behavior for unknown response codes is given for each category of
   codes.

10.1

7.1 Informational 1xx

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

10.1.1

7.1.1 100 Trying

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

10.1.2 150

7.1.2 180 Ringing

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

10.2

7.2 Successful 2xx

   The request was successful and MUST terminate a search.

10.2.1

7.2.1 200 OK

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

10.3

7.3 Redirection 3xx

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

10.3.1

7.3.1 300 Multiple Choices

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

   The response SHOULD include an entity containing a list of resource
   characteristics and location(s) from which the user or user agent can
   choose the one most appropriate. The entity format is specified by
   the media type given in the Content- Type  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

   The choices SHOULD include the specific
   URL for that representation in also be listed as  Location fields (Section 6.17).
   Unlike HTTP, the SIP response may contain several  Location field; user fields.
   User agents MAY use the  Location field value for automatic redirection.

10.3.2
   redirection or MAY ask the user to confirm a choice.

7.3.2 301 Moved Permanently

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

10.3.3

7.3.3 302 Moved Temporarily

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

10.3.4 350*

7.3.4 380 Alternative Service

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

10.4
   response.

7.4 Request Failure 4xx

   4xx responses are definite failure responses that MUST terminate the
   existing search for from a user or service. They particular
   server.  The client SHOULD NOT be retried
   immediately retry the same request without modification.

10.4.1
   modification (e.g., adding appropriate authorization). However, the
   same request to a different server may be successful.

7.4.1 400 Bad Request

   The request could not be understood due to malformed syntax.

10.4.2

7.4.2 401 Unauthorized

   The request requires user authentication.

10.4.3

7.4.3 402 Payment Required

   Reserved for future use.

10.4.4

7.4.4 403 Forbidden

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

10.4.5

7.4.5 404 Not Found

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

10.4.6 406 specified in the  Request-URI.

7.4.6 405 Method Not Acceptable Allowed

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

10.4.7 450* Decline allowed for the
   address identified by the  Request-URI. The user's machine was successfully contacted response MUST include an
   Allow header containing a list of valid methods for the indicated
   address.

7.4.7 407 Proxy Authentication Required

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

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

7.4.8 408 Request Timeout

   The client did not wish produce a request within the time that the server
   was prepared to participate.

10.4.8 451* Busy wait. The user's machine client MAY repeat the request without
   modifications at any later time.

7.4.9 420 Bad Extension

   The server did not understand the protocol extension specified with
   strength "must".

7.4.10 480 Temporarily Unavailable

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

10.5 this host),
   thus, this response does terminate any searches.

7.5 Server Failure 5xx

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

10.5.1

7.5.1 500 Server Internal Error

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

10.5.2

7.5.2 501 Not implemented

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

10.5.3 503 Service Unavailable

7.5.3 502 Bad Gateway

   The server is currently unable to handle the request due to server, while acting as a
   temporary overloading gateway 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 proxy, received an invalid
   response as it would for a 500 response.

   Note: The existence of from the 503 status code does not imply that a upstream 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
   The user agent or proxy server understood the user's address, but the
   request was unsuccessful accessed in contacting the user. A proxy might return
   this error towards the initiator if an attempt attempting 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;
   fulfill the address
   may be valid at another server.

10.6.3 602* Not currently here request.

7.5.4 503 Service Unavailable

   The call was unsuccessful because although the the user or service
   was known at the address called, the user or service server is not currently
   located at this address. This is not unable to handle the request due to a definitive failure;
   temporary overloading or maintenance of the user
   may be contactable at another server.

10.6.4 603* Alternative Address The call was unsuccessful because the user or service implication
   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 this is a temporary condition which will be being tried.  A 603 response MUST NOT suggest any alleviated after
   some delay. If known, the length of the
   hosts delay may be indicated in a
   Retry-After header. If no  Retry-After is given, the request's path client SHOULD
   handle the response as an alternative location.

10.7 Example: Normal Replies

   An example reply is given below. it would for a 500 response.

   Note: The first line existence of the reply states
   the SIP version number, 503 status code does not imply that it is a "200 OK" reply, which means
   server must use it when becoming overloaded. Some servers may wish to
   simply refuse the
   request was successful. connection.

7.5.5 504 Gateway Timeout

   The  Via header are taken server, while acting as a gateway, did not receive a timely
   response from the request,
   and entries are removed hop by hop as upstream server (e.g., a location server) it
   accessed in attempting to complete the reply retraces request.

7.6 Global Failures

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

7.6.1 600 Busy

   The callee's end system was contacted successfully but the callee is added by
   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 invited
   user's agent if required. The session ID is taken directly from
   callee does not wish to reveal the
   original request, along with reason for declining the request header. call, the
   callee should use status code 680 instead.

7.6.2 603 Decline

   The original sense
   of From field is preserved (i.e, it's callee's machine was successfully contacted but the session originator).

   In addition, user
   explicitly does not wish to participate. The response may indicate a  Contact-host field is added giving details of
   better time to call in the
   host  Retry-After header.

7.6.3 604 Does not exist anywhere

   The server has authoritative information that the user was located on, or alternatively the relevant proxy
   contact point which should be reachable from indicated in
   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
   To: schooler@cs.caltech.edu
   Contact-host: 131.215.131.147

   This same format is used for replies To request field does not exist anywhere. Searching for other categories of reply,
   except that some of then may require payloads to be carried.

   If the invited user
   elsewhere will not yield any results.

7.6.4 606 Not Acceptable

   The user's agent requires confirmation of receipt of a
   "200 OK" reply, it may optionally add an additional Confirm: required
   header to the body was contacted successfully but some aspects of the message specifying
   session profile (the requested media, bandwidth, or addressing style)
   were not acceptable.

   A "606 Not Acceptable" reply means 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 the user wishes to such a request,
   communicate, but cannot adequately support the invitation initiators agent should
   retransmit its request with an additional  Confirm header, with session described. The
   "604 Not Acceptable" reply MAY contain a list of reasons in a Warning
   header describing why the
   value "true" session described cannot be supported.
   These reasons can be one or "false" stating whether more of:

   606.1 Insufficient Bandwidth: The bandwidth specified in the session still exists
        description or
   no longer exists respectively (see section 7.1 for details). An
   example of an confirmation defined by the media exceeds that known to be
        available.

   606.2 Incompatible Protocol: One or more protocols described in the
        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
   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 available.

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

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

   606.5 Unicast not available: The site where the call and exited.

10.7.1 Redirects

   "603 alternative address" replies and 301 and 302 moved replies
   should specify another location using user is located does
        not support unicast communication (usually due to the  Location field.

   An example presence
        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) 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 advised invited to
   contact the multicast group 239.128.16.254 with a ttl of 16. In
   normal situations join a server would pre-existing lightweight session, negotiation
   may not suggest a redirect to a local
   multicast group unless (as in the above situation) it knows that the
   previous proxy or client be possible. It is within the scope of the local group.

   For unicast 603 redirects, a proxy MAY query up to the suggested location
   itself invitation initiator to decide
   whether or send MAY the redirect not to act on back towards the client. For
   multicast 603 redirects, a proxy SHOULD query the multicast address
   itself rather than sending "606 Not Acceptable" reply.

8 SIP Message Body

   The session description body gives details of the redirect back towards session the client as
   multicast may user is
   being invited to join. Its Internet media type MUST be scoped and this allows a proxy within given by the
   appropriate scope regions to make
   Content-type header field, and the query.

   For 301 or 302 redirects, a proxy SHOULD send body length in bytes MUST be given
   by the redirect on back
   towards  Content-Length header field. If the client and terminate body has undergone any other searches it is performing
   for the same request. Multicast 301 or 302 redirects
   encoding (such as compression) then this MUST NOT be
   generated.

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, indicated by the
   Content-encoding header field, otherwise Content-encoding MUST be
   omitted.

   If required, the answering server provides a session description
   that describes an "answering machine".  If can be encrypted using public
   key cryptography, and then can also carry private session keys for
   the invitation initiator
   decides to take advantage of session. If this service, it should send an
   invitation request is the case, four random bytes are added to the contact host (131.215.131.131) with
   beginning of the session description provided. This request should contain before encryption and are
   removed after decryption but before parsing.

8.1 Body Inclusion

   For a different
   session id from the one in request message, 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
   Invitation initiators can choose to treat presence of a "350 Alternative Service"
   reply as body is signaled by the
   inclusion of a failure if they wish to do so.

10.8.1 Negotiation  Content-Length header. 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 body may be included in a list of reasons why
   request only when the session
   described cannot be supported. These reasons can be one request method allows one.

   For response messages, whether or more of:

   406.1 Insufficient Bandwidth: not a body is included is dependent
   on both the bandwidth specified in request method and the session
        description or defined by response message's response code.
   All 1xx informational responses MUST NOT include a body. All other
   responses MAY include a payload, although it may be of zero length.

8.2 Message Body Length
   If no body is present in a message, then the media exceeds that known to  Content-Length header
   MAY be
        available.

   406.2 Incompatible Protocol: one omitted or more protocols described set to zero. When a body is included, its length in the
        request
   bytes is not available.

   406.3 Incompatible Format: one or more media formats described indicated in the
        request  Content-Length header and is not available.

   406.4 Multicast not available: determined by
   one of the site where following:

        1.   Any response message which MUST NOT include a body (such as
             the user 1xx responses) is located
        does not support multicast.

   406.5 Unicast not available: the site where always terminated by the user is located does
        not support unicast communication (usually due to first empty
             line after the presence
        of a firewall).

   Other reasons header fields, regardless if any  entity-
             header fields are likely to present.

        2.   Otherwise, a  Content-Length header MUST be added later. It is hoped that
   negotiation will not frequently 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 needed, and when a new user used for SIP.

9 Examples

9.1 Invitation

9.1.1 Request

   The example below is
   being invited to join a pre-existing lightweight session, negotiation
   may not be possible. If is up to the invitation request message en route from initiator to decide
   whether or not to act on a "406 Not Acceptable" reply.

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

   C->S: INVITE schooler@vlsi.cs.caltech.edu SIP/2.0 406 128.16.64.19/32492/5
         Via: SIP/2.0/UDP 239.128.16.254 16
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19
         From: mjh@isi.edu (Mark Handley)
         Subject: SIP will be discussed, too
         To: schooler@cs.caltech.edu
   Contact-host: 131.215.131.131
   Reason: 406.1, 406.3, 406.4
   Content-Type: meta/sdp (Eve Schooler)
         Call-ID: 62729-27@oregon.isi.edu
         Content-type: application/sdp
         Content-Length: 50 187

         v=0
   s=Lets talk
   b=CT:128
         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 131.215.131.131 224.2.0.1/127
         t=0 0
         m=audio 3456 RTP/AVP 7 0 13
   m=video 2232 RTP/AVP 31

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

   The  Via fields give the original request specified 256 kb/s total
   bandwidth, and hosts along the reply 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 only 128 kb/s the request was initiated by
   mjh@isi.edu the host 128.16.64.19 schooler@cs.caltech.edu is available. being
   invited; the message is currently being routed to
   schooler@vlsi.cs.caltech.edu

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

   The
   original request specified GSM audio, H.261 video, header is terminated by an empty line and WB whiteboard. is followed by a
   message body containing the session description.

9.1.2 Reply

   The audio coding and whiteboard are not available, but called user agent, directly or indirectly through proxy servers,
   indicates that it is alerting ("ringing") the called party:

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

   A sample reply
   states that DVI, PCM or LPC audio could be supported in order of
   preference. to the invitation is given below. The first line of
   the reply also states the SIP version number, that multicast is not available.
   In such a case, it might be appropriate to set up is a transcoding
   gateway "200 OK" reply,
   which means the request was successful. The  Via headers are taken
   from the request, and re-invite entries are removed hop by hop as the reply
   retraces the user.

   Invitation initiators path of the request. A new authentication field MAY choose to treat "406 Not Acceptable"
   replies as a failure be
   added by the invited user's agent if they wish to do so.

10.9 Compact Form

   When SIP required. The  Call-ID is carried over UDP taken
   directly from the original request, along with authentication and a complex
   session description, it may be possible that the size remaining fields
   of a the request or
   reply message. The original sense of  From field is
   preserved (i.e., it is larger than the MTU (or default 1,000-byte limit if session initiator).

   In addition, 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 gives details of the host where the
   user was located, or alternatively the relevant proxy contact point
   which should be written:

   INVITE 128.16.64.19/65729 reachable from the caller's host.

   S->C: SIP/2.0
   p:IN IP4 UDP 200 OK
         Via: SIP/2.0/UDP 239.128.16.254 1 16
   p:IN IP4 UDP
         Via: SIP/2.0/UDP 131.215.131.131 1
   p:IN IP4 UDP
         Via: SIP/2.0/UDP 128.16.64.19 1
   f:mjh@isi.edu
   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
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19
         Location: sip://es@jove.cs.caltech.edu

   For two-party Internet phone calls, the response must contain a
   description of Mbone Engineering Issues
   e=mbone@somewhere.com
   c=IN where to send data to, for example the reply from
   schooler to mjh :

   S->C: SIP/2.0 200 OK
         Via: SIP/2.0/UDP 239.128.16.254 16
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19 1
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19
         Location: sip://es@jove.cs.caltech.edu
         Content-Length: 102

         v=0
         o=schooler 4858949 4858949 IN IP4 224.2.0.1/127 192.1.2.3
         t=0 0
         m=audio 3456 5004 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
         c=IN IP4 131.215.131.147

   The caller confirms the invitation by sending 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 to the
   SIP application can keep control of
   location named in 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  Location header:

   C->S: CONNECTED schooler@jove.cs.caltech.edu SIP/2.0
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19

9.1.3 Aborting a Call

   If the advantage that clients are simpler to implement because
   no retransmission timing code needs caller wants to be written and also that abort a pending call, it is
   possible to have sends a single server serving SIP and HTTP  BYE request.

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

9.1.4 Redirects

   Replies with very
   little extra code.

   With UDP, all response codes "301 Moved Permanently" or "302 Moved
   Temporarily" SHOULD specify another location using the additional reliability code is in  Location
   field.

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

   In this example, the client. It proxy located at 131.215.131.131 is
   recommended that servers SHOULD implement both TCP and UDP
   functionality as being
   advised to contact 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 multicast group 239.128.16.254 with a ttl of
   16 and UDP transport

   The Session Invitation Protocol is straightforward transport. In normal situations, a server would not
   suggest a redirect to a local multicast group unless, as in operation. Only the initiating client needs to keep any state regarding above
   situation, it knows that the current
   connection attempt. SIP assumes no additional reliability from IP.

   Requests previous proxy 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 is within the
   scope of timeouts and retransmissions. the local group. If the
   reply request is merely redirected to a 1xx Informational progress report, multicast
   group, a proxy server SHOULD query the multicast address itself
   rather than sending the redirect back towards the initiating client SHOULD still continue retransmitting as multicast
   may be scoped; this allows a proxy within the request, albeit less
   frequently.

   When appropriate scope
   regions to make the remote user agent or server sends query.

9.1.5 Alternative Services

   An example of a final 2xx or 4xx
   response (not "350 Alternative Service" reply is:

   S->C: SIP/2.0 350 Alternative Service
         Via: SIP/2.0/UDP 131.215.131.131
         Via: SIP/2.0/UDP 128.16.64.19
         From: mjh@isi.edu
         To: schooler@cs.caltech.edu
         Call-ID: 62729-27@128.16.64.19
         Location: recorder@131.215.131.131
         Content-type: application/sdp
         Content-length: 146

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

   In this case, the answering server provides a 1xx report), session description
   that describes an "answering machine". If the invitation initiator
   decides to take advantage of this service, it cannot be sure should send an
   invitation request to the client has
   received answering machine at 131.215.131.131 with
   the response, session description provided (modified as appropriate for a
   unicast session to contain the appropriate local address and thus port for
   the invitation initiator). This request SHOULD cache the results until contain a
   connection setup timeout has occurred to avoid having to contact different
   Call-ID from the
   user again. The server one in the original request. An example would be:

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

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

   Invitation initiators MAY also choose to cache 3xx or 6xx responses treat a "350 Alternative Service"
   reply as a failure if the cost of obtaining the response outweighs the cost they wish to do so.

9.1.6 Negotiation

   An example of caching
   it.

   It is possible that a user can be invited successfully, but that the "606 Not Acceptable" reply that the user was successfully contacted may is:

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

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

   In this example, the invitation initiator "hung up" before original request specified 256 kb/s total
   bandwidth, and the reply arrived 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 session was to reply
   states that DVI, PCM or LPC audio could be two-way, the conferencing
   system should supported in order of
   preference. The reply also states that multicast is not available.
   In such a case, it might be prepared appropriate to deal with this circumstance.

   One solution is for set up a transcoding
   gateway and re-invite the initiator user.

9.2 OPTIONS Request

   A caller Alice can use an  OPTIONS request to acknowledge the invitee's "200
   OK" reply. Although not required, in find out the case
   capabilities of a successful
   invitation potential callee Bob, without "ringing" the invited user's agent can make a confirmation request
   in its "200 OK" reply.
   designated address. 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 case, Bob indicates that a premature
   hang-up he can be detected without
   reached at three different addresses, ranging from voice-over-IP to a long timeout. Such
   PSTN phone to a confirmation
   request may be retransmitted by the invited user's agent if it so
   desired. Confirmation requests can only be made pager.

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

   S->C: SIP/2.0 200 OK
         Location: sip://bob@host.example.com ;service=IP,voice-mail
                   ;media=audio ;duplex=full ;q=0.7
         Location: phone://1-415-555-1212 ; service=ISDN;mobility=fixed;
                   language=en,es,iw ;q=0.5
         Location: phone://1-800-555-1212 ; service=pager;mobility=mobile;
                   duplex=send-only;media=text; q=0.1

   Alternatively, Bob could have returned a description of

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

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

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

10 Compact Form

   When SIP is carried over UDP with "200 OK"
   replies, authentication and only the invitation initiator's agent a complex
   session description, it may issue be possible that the
   actual confirmation.

   Only size of a "200 OK" request or
   reply warrants such is larger than the MTU. To reduce this problem, a confirmation handshake, because
   it more compact
   form of SIP is also defined by using alternative names for common
   header fields.  These short forms are NOT abbreviations, they are
   field names. No other abbreviations are allowed.

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

   Thus the only situation where user-relevant state may header in section 9.1 could also 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" written:

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

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

   Mixing short field names and "6xx Search Response" replies do long field names is allowed, but not immediately get
   passed back to the invitation initiator,
   recommended. Servers MUST accept both short and so no end-to-end
   acknowledgment of long field names for
   requests. Proxies MUST NOT translate a failed request between short and long
   forms if authentication fields are present.

11 SIP Transport

   SIP is possible.

11.2 Reliability using TCP transport defined so it can use either UDP or TCP is as a reliable transport protocol, and so we do not need to define
   protocol.

11.1 Achieving Reliability For UDP Transport

11.1.1 General Operation

   SIP assumes no 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 from IP. Requests or more 1xx Informational responses before
   sending replies
   may be lost. A SIP client SHOULD simply retransmit a single 2xx, 3xx, 4xx, 5xx or 6xx reply. The server MUST NOT
   send more than one reply, SIP request
   periodically with the exception timer T1 (default value of 1xx responses. The
   server SHOULD NOT close the TCP connection T1: once a second) until
   it has sent its
   final receives a response, at which point or until it MAY close has reached a set limit on the TCP connection if it
   wishes to. However, normally it
   number of retransmissions. The default limit is the client's responsibility to
   close the connection.

   If the server leaves the connection open, 20.

   SIP requests and if replies are matched up by the client so
   desires it may re-use the connection for further SIP requests or for
   requests from using 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
   Call-ID header field; thus, a location server can only have one outstanding
   request per call at the user's
   home site either knows where the user resides, knows how to locate
   the user, any given time.

        HS: A transaction or at the very least knows another location server that
   possibly might have a better idea. How these servers get request ID would remove this
   information is outside
        limitation.

   If the scope of SIP itself, but it is expected
   that many different user-location services will exist for some time.
   SIP reply 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 provisional response, the initiating client SHOULD
   continue retransmitting the request, albeit less frequently, using
   timer T2. The default retransmission interval T2 is 5 seconds.

   After the server receives a request for a user whose location sends a final response, it
   does not know, cannot be sure the client
   has received the response, and thus SHOULD cache the results for whom it has no better idea where at
   least 30 seconds to avoid having to, for example, contact the user
   might be, then the or
   user location server should return again upon receiving a "601 Not Currently Here"
   reply message.

   If retransmission.

11.1.2 INVITE

   Special considerations apply for the server does have  INVITE method.

        1.   After receiving an idea how to contact invitation, considerable time may elapse
             before the user, it server can
   either forward (relay) determine the request itself, or can redirect outcome. For example,
             the
   invitation initiator to another client that is more likely to know by
   sending a 603, 301 called party may be "rung" or 302 response as appropriate. It extensive searches may be
             performed, so delays can also
   gateway reach several tens of seconds.

        2.   It is possible that the request into some other form if some other invitation
   protocol is in use in a region containing request reaches the invited user, though in
   doing so
             callee and the server callee is likely to give up being stateless.

   Whether willing to relay take the request or to redirect call, but that
             the request final response (200 OK, in this case) is up lost on the
             way to the
   server itself. For example, if caller. If the server is session still exists but the
             initiator gives up on a firewall machine,
   then it will probably have to relay including the request to servers inside user, the
   firewall. Additionally, if a local multicast group is contacted
             user has sufficient information to be used for
   user location, then the server is likely able to relay join the request.
             session. However, if the user is currently away from home, relaying session no longer exists because
             the
   request makes little sense, invitation initiator "hung up" before the reply arrived
             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 session was to be better performed
   by relaying whereas wide-area searches are likely to two-way, the conferencing system
             should be better
   performed by redirection.

   When SIP uses UDP transport, clients and servers can make multiple
   simultaneous requests prepared to locate deal with this circumstance.

        3.   If a particular telephony user at low cost. This
   greatly speeds up any search for interface is modeled or if we need to
             interface to the user, and in most cases will
   only result in one successful response. Although several simultaneous
   paths may reach PSTN, the same host, successful responses arriving from
   multiple paths caller will not confuse provide "ringback",
             a signal that the client as they should all contain callee is being alerted. Once the same successful host address. However, this does imply callee
             picks up, the caller needs to know so that paths
   with many levels of relaying should be strongly discouraged as if it can enable
             the
   request is fanned out at each hop voice path and relayed many times, request
   implosions stop ringback.  The callee's response to
             the invitation could result. Thus servers that are not get lost. Unless 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 is
             transmitted reliably, the fanout of caller will continue to hear
             ringback while the search.

12.2 Parallel Searches: Initiator Behavior callee assumes that the call exists.

        4.   The session initiator may make a parallel search client has to be able to terminate an on-going request,
             e.g., because it is no longer willing to wait for a user. This can
   occur when DNS resolution results in multiple addresses, the
             connection or when
   contacting a remote server results in a "603 Alternative Address"
   response containing multiple addresses search to try. All such parallel
   searches for succeed. One cannot rely on the same SIP
             absence of request MUST contain the same SIP Id,
   though retransmission, since the sequence number (given in server would
             have to continue honoring the  Path field) SHOULD be
   different request for each several request
             retransmission periods, that is, possible tens of the parallel searches.

   Whilst performing a parallel search, different responses may result
   from different servers, and it seconds
             if only one or two packets can be lost.

   The first problem is important for the initiating client solved by indicating progress to handle these responses correctly. In general, the following rules
   apply:

        o If caller: the
   server returns a 2xx provisional response indicating it is received, searching or
   ringing the invitation was successful, user.

   The server retransmits the user should be informed and all pending requests should be
         terminated and/or ignored.

        o If a 4xx final response is received at intervals of T3 (default
   value of T3 = 2 seconds) until it receives a  CONNECTED request for
   the invitation same  Call-ID or until it has definitively
         failed, retransmitted the user should be informed, and all pending requests
         should be terminated and/or ignored.

        o If a 3xx final response 10
   times. The  CONNECTED request is received, acknowledged only once. If the search should be terminated
   request is syntactically valid and all pending requests should be terminated and/or ignored.
         However, further action MAY be taken depending on the actual
         reply without informing  Request-URI matches that in
   the user or alternatively  INVITED request with the
         invitation same  Call-ID, the server answers with
   status code 200, otherwise with status code 400.

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

11.2 Connection Management for TCP

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

   The client MAY be regarded as having failed in close the connection at any time. Closing the
   connection before receiving a final response signals that the client
   wishes to abort the request.

   The server SHOULD NOT close the TCP connection until it has sent its
   final response, at which case point it MAY close the
         user MUST be informed.

        o If a 5xx or 6xx response TCP connection if it
   wishes to. However, normally it is received, the particular client's responsibility to
   close the connection.

   If the server
         responding is removed from leaves the parallel search connection open, and if the search
         continues.  If a "603 Alternative Address" response is
         received, the search client so
   desires it may be expanded to include those servers
         listed in re-use the response connection for further SIP requests or for
   requests from the same family of protocols (such as HTTP or stream
   control commands).

12 Behavior of SIP Servers

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

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

   We define an "A--B proxy" as a proxy that receives SIP requests over
   transport protocol A and issues requests, acting as a SIP client,
                           +===========+
                           |  Initial  |
                           +===========+
                                 |
                                 |
                                 |    -
                                 |  ------
                                 |  INVITE
                     +------v    v
                    T1     +-----------+
                  ------   |  Calling  |-------------------+
                  INVITE   +-----------+                   |
                     +------| |  |                         |
             +----------------+  |                         |
             |                   |                         |
             |                   |                         |
             |                   |                         |
             |                   |                         |
             |       +------v    v    v-----|              |
             |      T2     +-----------+   1xx             |
             |    ------   |  Ringing  |   ---             |
             |    INVITE   +-----------+    -              |
             |       +------|    |  | |-----+              |
             |                   |  +--------------+       |
             |     2xx           |                 | >=300 |
             |  ---------        |    2xx          | ----- |
             |  CONNECTED        | ---------       |   -   |
             |                   | CONNECTED       |       |
             +----------------+  |                 |       |
                     +------v |  v                 v       v
                    2xx    +-----------+         +-----------+
                 --------- | Connected |         |  Failure  |
                 CONNECTED +-----------+         +-----------+
                     +------|

              event
             -------
             message

   Figure 4: State transition diagram of client for  INVITE method

   using transport protocol B. If not already responded. The
         user SHOULD NOT be informed unless there are no other servers
         left stated explicitly, rules apply 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
   any combination of transport protocols. For conciseness, we only
   describe behavior with UDP and TCP, but the same way that an Initiating Client can discover multiple
   addresses to try, a proxy rules apply for any
   unreliable datagram or reliable protocol, respectively.

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

               event
              -------
              message

   Figure 5: State transition diagram of server can also discover multiple addresses
   that it may try. For a proxy for  INVITE method

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

12.1 Redirect Server

   A redirect server to be stateless, it must does not make
   multiple issue any 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 of its own. It can
   return any reply other than 1xx informational replies until it
   has received a definitive reply response that accepts, refuses or has no further addresses to try.

   Thus faced with DNS resolution giving multiple addresses, redirects the request.
   After receiving a proxy request, a redirect server that wishes to proceeds through the
   following steps:

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

        2.   Once the first address. Similarly a stateless proxy should not attempt server has gathered the list of alternative
             locations or has decided to
   send accept or refuse the call, it
             returns the final response.  This ends the SIP request to multiple addresses given transaction.

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

12.2 Proxies Issuing Single Unicast Requests

   Proxies in this category issue at most a "603 Alternative
   Address" response single unicast request for
   each incoming SIP request, that is returned it it, but should is, they do not "fork" requests.
   Servers may choose to always operate in the mode described in Section
   12.3.

12.2.1 UDP--UDP Proxy Server

   The UDP--UDP server can forward such a
   response back towards the initiator.

   Proxies that wish request and any responses. It
   does not have to keep maintain any state should follow for the following rules
   regarding responses obtained during a parallel search:

        o If a 2xx response SIP transaction. UDP
   reliability is received, the invitation was successful, assured by the 2xx response should be forwarded back towards next redirect server in the
         initiator, and all pending requests should be terminated and/or
         ignored.

        o If server
   chain.

12.2.2 UDP--TCP Proxy Server

   A proxy server issuing a 4xx response is received the invitation has definitively
         failed, the 4xx response should be forwarded back towards single request over TCP maintains state for
   the
         initiator, and all pending requests should be terminated and/or
         ignored.

        o whole SIP transaction indexed by the  Call-ID.

   If it receives a 3xx UDP retransmission of the same request for an
   existing session, it retransmits the last response is received from the invitation is regarded by
   TCP side.  Any changes in the proxy as having failed, message body compared to the 3xx response should be
         forwarded back towards last
   request for the initiator, Call-ID are silently ignored. (Otherwise, the search should be
         terminated proxy
   would have to remember and all pending requests should be terminated and/or
         ignored.

        o If compare the message body; this also
   violates the notion of a 5xx or 6xx response is received, SIP transaction. TBD) The server SHOULD
   cache the final response for a particular server
         responding is removed from  Call-ID after the parallel search and SIP
   transaction on the search
         continues.  If a "603 Alternative Address" response is
         received, TCP side has completed.

   After the search may be expanded to include those servers
         listed in cache entry has been expired, the response that have not already responded. No
         response other than server cannot tell
   whether an incoming request is actually a periodic "100 Trying" response should be
         send towards retransmission of an older
   request, where the initiator unless there are no other servers
         left to try, in which case a response SHOULD be sent TCP side has terminated. It will treat it as
         described below.

        o If a 1xx response is received, new
   request.

12.3 Proxy Server Issuing Several Requests

   All requests carry the search continues. The 1xx
         response MAY be forwarded towards same  Call-ID. For unicast, each of the initiator as appropriate.

   If
   requests has a proxy had exhausted its search and still not obtained different (host-dependent)  Request-URI. For
   multicast, a
   definitive response (it received only 1xx, 5xx, and 6xx responses) single request is issued, likely with a host-independent
   Request-URI. A client receiving a multicast query does not have to
   check whether the proxy should cache these responses and return host part of the first  Request-URI matches its own host
   or domain name. To avoid response
   from the following ordered list:

        1.   503 Service Unavailable;

        2.   500 Server Internal Error;

        3.   501 Not Implemented;

        4.   any other 5xx error implosion, servers SHOULD NOT
   answer multicast requests with a 404 (Not Found) status code.
   Servers MAY decide not yet defined;

        5.   600 Search Failure;

        6.   602 Not Currently Here;

        7.   601 Not Known Here;

        8.   any other 6xx error to answer multicast requests if their response not yet defined.

   If a proxy has exhausted its search and
   would be 5xx.

   The server MAY respond to the only response request immediately with a "100 Trying"
   response; otherwise it has
   received has been "603 Alternative Address", then MAY wait until either the proxy should
   send a "600 Search Failure" first response if any connection attempt timed
   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
   its requests or 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 retransmission interval.

   The following pseudo-code describes the behavior of proxies is also
        possible.

   It is possible that a proxy server will receiver
   issuing several requests in response to an incoming request. The
   function request(a) sends a SIP request using TCP to address a.
   await_response() waits until a response is received and relay it onwards using UDP or vice-versa. SIP does not assume
   end-to-end reliability even when returns the initiating
   response. request_close(a) closes the TCP connection to client with
   address a; this is using TCP,
   but optional. response(s, l, L) sends a SIP response to
   the client sending a request over TCP MAY assume that with status s and list of locations L, with l entries.
   ismulticast() returns 1 if the location is a multicast address and
   zero otherwise. The variable timeleft indicates the
   request amount of time
   left until the maximum response time has been received by expired. The variable
   recurse indicates whether the server it sent will recursively try addresses
   returned through a 3xx response. A server MAY decide to recursively
   try only certain addresses, e.g., those which are within the request to.
   Retransmission of same
   domain as the proxy server. Thus, an initial multicast request is then not the responsibility may
   trigger additional unicast requests.

     int N = 0;            /* number of the
   client. However, a called user agent SHOULD NOT assume that a 2xx
   success connection attempts */
     address_t address[];  /* list of addresses */
     location[];           /* list of locations */
     int heard = 0;        /* number of sites heard from */
     int class;            /* class of status code */
     int best = 1000;      /* best response so far */
     int timeleft = 120;   /* sample timeout value */
     int loc = 0;          /* number of locations */
     struct {              /* response */
       int status;         /* response status */
       char *location;     /* redirect locations */
       address_t a;        /* address of respondent */
     } r;
     int i;

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

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

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

       if (class == 2) {
         best = r.status;
         break;
       }
       else if (class == 3) {
             /* A server may optionally recurse.  The server MUST check whether
              * it has been received by tried this location before and whether the invitation initiator, even
   if all location is
              * part of the Via path fields in of the request indicated TCP transport because
   it cannot incoming request.  This check is
              * omitted here for brevity. Multicast locations MUST NOT be certain all those TCP connections still exist. If the
   called user agent requires knowledge that
          * returned to the response did reach client if the
   invitation initiator, server is not recursing.
          */
         if (recurse) {
           multicast = 0;
           N++;
           request(r.location);
         } else if (!ismulticast(r.location)) {
           locations[loc++] = r.location;
           best = r.status;
         }
       }
       else if (class == 4) {
         if (best >= 400) best = r.status;
       }
       else if (class == 5) {
         if (best >= 500) best = r.status;
       }
       else if (class == 6) {
         best = r.status;
         break;
       }
     }
     /* We haven't heard anything useful from anybody. */
     if (best == 1000) {
       best = 404;
     }
     if (best/100 != 3) locs = 0;
     response(best, locs, locations);

   When operating in this mode, a proxy server MUST ignore any responses
   received for  Call-IDs that it MAY add does not have a  Confirm: required field pending transaction
   for. (If server were to forward responses not belonging to a current
   transaction using the
   reply as it  Via field, the requesting client would get
   confused if the response was sent it has just issued another request using UDP.

   In the following, the term "TCP-UDP proxy" is used same Call-
   ID.)

13 Security Considerations

13.1 Confidentiality

   Unless SIP transactions are protected by lower-layer security
   mechanisms such as SSL , an attacker may be able to mean a proxy
   that received a request using TCP eavesdrop on call
   establishment and relayed it using UDP. Similarly
   a "TCP-UDP proxy" receives a reply using UDP invitations and, through the  Subject header field
   or the session description, gain insights into the topic of
   conversation.

13.2 Integrity

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

13.3 Access Control

   SIP requests are not authenticated unless the SIP  Authorization and should relay it
   using TCP.

12.4.1 Retransmission from a TCP-UDP Proxy

   A proxy receiving a request with TCP transport
   WWW-Authenticate headers are being used. The strengths and forwarding that
   request using UDP becomes responsible for retransmission weaknesses
   of these authentication mechanisms are the
   request same as required and for timing HTTP.

13.4 Privacy

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

A Summary of Augmented BNF

   In this specification we use the request if no answer Augmented Backus-Naur Form notation
   described in [19]. For quick reference, the following 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 brief
   summary of the main features of this ABNF.

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

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

   *term
        zero of more instances of  term;

   3*term
        three or more instances of  term;

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

   [ term ]
        term is optional;

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

   term1 | term2
        either  term1 or SIP response state associated
   with it.

   If such  term2 may appear but not both;

   #term
        a proxy receives comma separated list of  term;

   2#term
        a retransmission comma separated list of the UDP request whilst  term containing at least 2 items;

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

   Common Tokens

   Certain tokens are used frequently in the state or awaiting a response (i.e, has request state), BNF this document, and not
   defined elsewhere. Their meaning is well understood but we include it
   SHOULD NOT forward the duplicate request into the TCP connection
   unless
   here for completeness.

        CR       =    %d13            ;  carriage return character
        LF       =    %d10            ;  line feed character
        CRLF     =    CR LF           ;  typically the request has been modified, but instead SHOULD respond with end of a "100 Trying" response sent back towards the initiator.

   Note: This behavior is different from line
        SP       =    %d32            ;  space character
        TAB      =    %d09            ;  tab character
        LWS      =    *( SP | TAB)    ;  linear whitespace
        DIGIT    =    "0" .. "9"      ;  a UDP-UDP proxy which MUST
   forward single decimal digit

   Changes

   Since version -01, the retransmitted request following things have changed:

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

        o Initial author's note removed; dated.

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

        o Conference control (tight vs. loose) seems less and MAY additionally respond with a
   "100 Trying" response sent back towards the initiator.

   If less
         appropriate. All share some state such a proxy receives a retransmission as notions of the UDP request in
   response state (i.e,
         membership; some (ITU versions) tend to keep it has already sent in a definitive response) then
   the proxy MAY retransmit that response if it has cached central
         server, others distribute 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 Some state is synchronized at
         larger timescales than other. (After all, even a server won't
         know if there is one or via a new participant disconnects from the network until TCP connection
         keep-alive, if there is
   not, any, kicks in.)

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

        o Terminology: user always reminds me of controlled substances;
         thus, this term is avoided where better terminology exists.
         Since this protocol sits at the same as a UDP-to-UDP proxy in boundary between traditional
         Internet and telephony services, some of the same
   circumstances.

12.4.3 Confirmation arriving at a TCP-UDP Proxy

   One possible event terminology
         familiar in that may occur realm is that whilst performing a search
   using UDP, introduced.

        o Terminology: user location server replaced by redirect server,
         since a response proxy server may arrive that should be relayed back towards also invoke user location. Also, the initiator
         actual user location server (e.g., an LDAP, ULS or similar
         directory) may be invoked using TCP, but protocols other than SIP.

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

        o Defined SIP transaction, provisional and final responses.

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

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

        o Removed redundancies and contradictions from request and
         response with a  Confirm: required field, definitions (space vs. SP, duplicate CRLF definition,
         recursive request-header, ...).

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

        o Re-engineered the request invitation reliability mechanism to the Contact Host with use a  Confirm: false field
         separate confirmation message.

        o Tentative increase of MTU to speed hang-up discovery at 1500 bytes, as per discussion in
         Stevens.

        o Added  Reach,  Organization,  Subject, Priority,
         Authorization,  WWW-Authentication headers for improved call
         handling. WWW "basic" authentication isn't great, but it is
         widely deployed and probably sufficient for giving out
         "private" telephone numbers, particularly those where the called user agent.

12.4.4 Confirmation sent from a UDP-TCP Proxy

   Normally
         callee incurs a response that arrives at charge.  (I want to be able to give somebody a proxy using TCP
         password to call my 800 number via an Internet gateway;
         authenticating who that should be
   sent back towards the initiator using UDP should be sent once, and
   should only be resent if the request person is resent from requires that I modify a
         script on my server to add another distinguished name to the UDP proxy
   closer
         list of allowable callees.)

        o Renamed  Reason to  Warning (to align with HTTP) header since
         the initiator. However, this does response line already offers a failure reason.
         Unfortunately, listing several failures is not allow for reliable
   confirmation.

13 Using Variants for Terminal Negotiation

   Redirection allows all that helpful
         since the called calling party to indicate several communication
   alternatives to cannot determine which of the caller using media
         within the 300 (Multiple Choices) response,
   all reachable using a single published communication identifier.

   The  Alternates header in description causes the response contains difficulty or whether it was
         the variant list.
   The response may contain an entity, typically set of content type
   text/html, providing guidance to media as a whole, but it may give the user. The calling user agent is
   free
         some indication as to ignore what's going on.

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

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

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

        o CAPABILITIES to OPTIONS for closer alignment with a variant
   description [7].

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

     communications-quality = qvalue
     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 <"> HTTP 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
         RTSP;

        o Path to Via for class-
   tag and  service-tag can be registered closer alignment with IANA. The media tag uses HTTP and RTSP;

        o Content type meta changed to application, since "meta" doesn't
         exist as a top-level Internet media types, e.g., audio, video, application/x-wb, etc. This
   is meant type.

        o Formatting closer to HTTP and RTSP.

        o Explain relationship to H.323.

B Open Issues

   RELIABLE: How to provide reliability?

   BYE: Use of BYE method?

   REGISTER: Use of REGISTER method?

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

   TRANSACTION: Should we have a transaction id in addition to a call
        ID? Call-IDs are for indicating general communication capability, not the
   support end system, but a transaction ID is for
        a single SIP exchange. This is useful for specific encodings. It should be sufficient to allow the
   caller to choose an appropriate communication address.

14 Internet telephony,
        where a single call may trigger several transactions.

C Acknowledgments

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

15 [23,24].

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

D Authors' Addresses

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

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

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

16

E Bibliography

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

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

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

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

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

   [2]

   [6] M. Handley, "Sap: Handley and V. Jacobson, "SDP: Session announcement description protocol,"
   Internet Draft, Internet Engineering Task Force, Nov. 1996. Mar. 1997.  Work in
   progress.

   [3]

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

   [4]

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

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

   [5]

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

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

   [6]

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

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

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

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

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

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

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

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

   [7] K. Holtman

   [20] J. Mogul and A. Muntz, "Transparent Content Negotiation in
   HTTP," S. Deering, "Path MTU discovery,"  RFC 1191,
   Internet Draft, Engineering Task Force, Nov. 1990.

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

   [22] D. Crocker, "Standard for the format of ARPA internet text
   messages," STD 11, RFC 822, Internet Engineering Task Force, Nov. 1997.
   Work in progress.

   [8] Aug.
   1982.

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

   [9]

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

                           Table of Contents

   1          Introduction ........................................    2
   1.1        Overview of SIP Functionality .......................    2
   1.2        Finding Multimedia Sessions .........................    3
   1.3        Terminology .........................................    4
   1.4        Definitions .........................................    4
   1.5        Protocol Properties .................................    6
   1.5.1      Minimal State .......................................    6
   1.5.2      Transport-Protocol Neutral ..........................    6
   1.5.3      Text-Based ..........................................    6
   1.6        SIP Addressing ......................................    6
   1.7        Locating a SIP Server ...............................    8
   1.8        SIP Transactions ....................................    9
   1.9        Locating a User .....................................    9
   2          SIP Uniform Resource Locators .......................   12
   3          SIP Message Overview ................................   14
   4          Request .............................................   15
   4.1        Request-Line ........................................   16
   4.1.1      Methods .............................................   17
   4.1.2      Request-URI .........................................   18
   4.1.3      SIP Version .........................................   18
   5          Response ............................................   18
   5.1        Status-Line .........................................   19
   5.1.1      Status Codes and Reason Phrases .....................   19
   6          Header Field Definitions ............................   20
   6.1        General Header Fields ...............................   22
   6.2        Entity Header Fields ................................   22
   6.3        Request Header Fields ...............................   22
   6.4        Response Header Fields ..............................   22
   6.5        Header Field Format .................................   23
   6.6        Accept ..............................................   23
   6.7        Accept-Language .....................................   24
   6.8        Allow ...............................................   24
   6.9        Authorization .......................................   24
   6.10       Authentication ......................................   24
   6.11       Call-ID .............................................   24
   6.12       Content-Length ......................................   25
   6.13       Content-Type ........................................   25
   6.14       Date ................................................   26
   6.15       Expires .............................................   26
   6.16       From ................................................   27
   6.17       Location ............................................   27
   6.18       Organization ........................................   29
   6.19       PEP .................................................   29
   6.20       Priority ............................................   29
   6.21       Proxy-Authenticate ..................................   29
   6.22       Proxy-Authorization .................................   29
   6.23       Public ..............................................   30
   6.24       Reach ...............................................   30
   6.25       Retry-After .........................................   30
   6.26       Sequence ............................................   31
   6.27       Server ..............................................   31
   6.28       Subject .............................................   31
   6.29       To ..................................................   32
   6.30       User-Agent ..........................................   32
   6.31       Via .................................................   32
   6.32       Warning .............................................   33
   6.33       WWW-Authenticate ....................................   34
   7          Status Code Definitions .............................   34
   7.1        Informational 1xx ...................................   35
   7.1.1      100 Trying ..........................................   35
   7.1.2      180 Ringing .........................................   35
   7.2        Successful 2xx ......................................   35
   7.2.1      200 OK ..............................................   35
   7.3        Redirection 3xx .....................................   35
   7.3.1      300 Multiple Choices ................................   35
   7.3.2      301 Moved Permanently ...............................   36
   7.3.3      302 Moved Temporarily ...............................   36
   7.3.4      380 Alternative Service .............................   36
   7.4        Request Failure 4xx .................................   36
   7.4.1      400 Bad Request .....................................   36
   7.4.2      401 Unauthorized ....................................   37
   7.4.3      402 Payment Required ................................   37
   7.4.4      403 Forbidden .......................................   37
   7.4.5      404 Not Found .......................................   37
   7.4.6      405 Method Not Allowed ..............................   37
   7.4.7      407 Proxy Authentication Required ...................   37
   7.4.8      408 Request Timeout .................................   37
   7.4.9      420 Bad Extension ...................................   37
   7.4.10     480 Temporarily Unavailable .........................   38
   7.5        Server Failure 5xx ..................................   38
   7.5.1      500 Server Internal Error ...........................   38
   7.5.2      501 Not implemented .................................   38
   7.5.3      502 Bad Gateway .....................................   38
   7.5.4      503 Service Unavailable .............................   38
   7.5.5      504 Gateway Timeout .................................   39
   7.6        Global Failures .....................................   39
   7.6.1      600 Busy ............................................   39
   7.6.2      603 Decline .........................................   39
   7.6.3      604 Does not exist anywhere .........................   39
   7.6.4      606 Not Acceptable ..................................   39
   8          SIP Message Body ....................................   40
   8.1        Body Inclusion ......................................   40
   8.2        Message Body Length .................................   40
   9          Examples ............................................   41
   9.1        Invitation ..........................................   41
   9.1.1      Request .............................................   41
   9.1.2      Reply ...............................................   42
   9.1.3      Aborting a Call .....................................   43
   9.1.4      Redirects ...........................................   44
   9.1.5      Alternative Services ................................   44
   9.1.6      Negotiation .........................................   45
   9.2        OPTIONS Request .....................................   46
   10         Compact Form ........................................   47
   11         SIP Transport .......................................   48
   11.1       Achieving Reliability For UDP Transport .............   48
   11.1.1     General Operation ...................................   48
   11.1.2     INVITE ..............................................   49
   11.2       Connection Management for TCP .......................   50
   12         Behavior of SIP Servers .............................   50
   12.1       Redirect Server .....................................   53
   12.2       Proxies Issuing Single Unicast Requests .............   53
   12.2.1     UDP--UDP Proxy Server ...............................   53
   12.2.2     UDP--TCP Proxy Server ...............................   53
   12.3       Proxy Server Issuing Several Requests ...............   54
   13         Security Considerations .............................   56
   13.1       Confidentiality .....................................   56
   13.2       Integrity ...........................................   56
   13.3       Access Control ......................................   56
   13.4       Privacy .............................................   56
   A          Summary of Augmented BNF ............................   57
   B          Open Issues .........................................   60
   C          Acknowledgments .....................................   60
   D          Authors' Addresses ..................................   61
   E          Bibliography ........................................   61