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📄 rfc2543.txt

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   12.3.7     Stateless, Non-Forking Proxy ........................  100   12.4       Forking Proxy .......................................  100   13         Security Considerations .............................  104   13.1       Confidentiality and Privacy: Encryption .............  104   13.1.1     End-to-End Encryption ...............................  104   13.1.2     Privacy of SIP Responses ............................  107   13.1.3     Encryption by Proxies ...............................  108   13.1.4     Hop-by-Hop Encryption ...............................  108   13.1.5     Via field encryption ................................  108   13.2       Message Integrity and Access Control:              Authentication ......................................  109Handley, et al.             Standards Track                     [Page 5]RFC 2543            SIP: Session Initiation Protocol          March 1999   13.2.1     Trusting responses ..................................  112   13.3       Callee Privacy ......................................  113   13.4       Known Security Problems .............................  113   14         SIP Authentication using HTTP Basic and Digest              Schemes .............................................  113   14.1       Framework ...........................................  113   14.2       Basic Authentication ................................  114   14.3       Digest Authentication ...............................  114   14.4       Proxy-Authentication ................................  115   15         SIP Security Using PGP ..............................  115   15.1       PGP Authentication Scheme ...........................  115   15.1.1     The WWW-Authenticate Response Header ................  116   15.1.2     The Authorization Request Header ....................  117   15.2       PGP Encryption Scheme ...............................  118   15.3       Response-Key Header Field for PGP ...................  119   16         Examples ............................................  119   16.1       Registration ........................................  119   16.2       Invitation to a Multicast Conference ................  121   16.2.1     Request .............................................  121   16.2.2     Response ............................................  122   16.3       Two-party Call ......................................  123   16.4       Terminating a Call ..................................  125   16.5       Forking Proxy .......................................  126   16.6       Redirects ...........................................  130   16.7       Negotiation .........................................  131   16.8       OPTIONS Request .....................................  132   A          Minimal Implementation ..............................  134   A.1        Client ..............................................  134   A.2        Server ..............................................  135   A.3        Header Processing ...................................  135   B          Usage of the Session Description Protocol (SDP)......  136   B.1        Configuring Media Streams ...........................  136   B.2        Setting SDP Values for Unicast ......................  138   B.3        Multicast Operation .................................  139   B.4        Delayed Media Streams ...............................  139   B.5        Putting Media Streams on Hold .......................  139   B.6        Subject and SDP "s=" Line ...........................  140   B.7        The SDP "o=" Line ...................................  140   C          Summary of Augmented BNF ............................  141   C.1        Basic Rules .........................................  143   D          Using SRV DNS Records ...............................  146   E          IANA Considerations .................................  148   F          Acknowledgments .....................................  149   G          Authors' Addresses ..................................  149   H          Bibliography ........................................  150   I          Full Copyright Statement ............................  153Handley, et al.             Standards Track                     [Page 6]RFC 2543            SIP: Session Initiation Protocol          March 19991 Introduction1.1 Overview of SIP Functionality   The Session Initiation Protocol (SIP) is an application-layer control   protocol that can establish, modify and terminate multimedia sessions   or calls. These multimedia sessions include multimedia conferences,   distance learning, Internet telephony and similar applications. SIP   can invite both persons and "robots", such as a media storage   service.  SIP can invite parties to both unicast and multicast   sessions; the initiator does not necessarily have to be a member of   the session to which it is inviting. Media and participants can be   added to an existing session.   SIP can be used to initiate sessions as well as invite members to   sessions that have been advertised and established by other means.   Sessions can be advertised using multicast protocols such as SAP,   electronic mail, news groups, web pages or directories (LDAP), among   others.   SIP transparently supports name mapping and redirection services,   allowing the implementation of ISDN and Intelligent Network telephony   subscriber services. These facilities also enable personal mobility.   In the parlance of telecommunications intelligent network services,   this is defined as: "Personal mobility is the ability of end users to   originate and receive calls and access subscribed telecommunication   services on any terminal in any location, and the ability of the   network to identify end users as they move. Personal mobility is   based on the use of a unique personal identity (i.e., personal   number)." [1]. Personal mobility complements terminal mobility, i.e.,   the ability to maintain communications when moving a single end   system from one subnet to another.   SIP supports five facets of establishing and terminating multimedia   communications:   User location: determination of the end system to be used for        communication;   User capabilities: determination of the media and media parameters to        be used;   User availability: determination of the willingness of the called        party to engage in communications;   Call setup: "ringing", establishment of call parameters at both        called and calling party;Handley, et al.             Standards Track                     [Page 7]RFC 2543            SIP: Session Initiation Protocol          March 1999   Call handling: including transfer and termination of calls.   SIP can also initiate multi-party calls using a multipoint control   unit (MCU) or fully-meshed interconnection instead of multicast.   Internet telephony gateways that connect Public Switched Telephone   Network (PSTN) parties can also use SIP to set up calls between them.   SIP is designed as part of the overall IETF multimedia data and   control architecture currently incorporating protocols such as RSVP   (RFC 2205 [2]) for reserving network resources, the real-time   transport protocol (RTP) (RFC 1889 [3]) for transporting real-time   data and providing QOS feedback, the real-time streaming protocol   (RTSP) (RFC 2326 [4]) for controlling delivery of streaming media,   the session announcement protocol (SAP) [5] for advertising   multimedia sessions via multicast and the session description   protocol (SDP) (RFC 2327 [6]) for describing multimedia sessions.   However, the functionality and operation of SIP does not depend on   any of these protocols.   SIP can also be used in conjunction with other call setup and   signaling protocols. In that mode, an end system uses SIP exchanges   to determine the appropriate end system address and protocol from a   given address that is protocol-independent. For example, SIP could be   used to determine that the party can be reached via H.323 [7], obtain   the H.245 [8] gateway and user address and then use H.225.0 [9] to   establish the call.   In another example, SIP might be used to determine that the callee is   reachable via the PSTN and indicate the phone number to be called,   possibly suggesting an Internet-to-PSTN gateway to be used.   SIP does not offer conference control services such as floor control   or voting and does not prescribe how a conference is to be managed,   but SIP can be used to introduce conference control protocols. SIP   does not allocate multicast addresses.   SIP can invite users to sessions with and without resource   reservation.  SIP does not reserve resources, but can convey to the   invited system the information necessary to do this.1.2 Terminology   In this document, the key words "MUST", "MUST NOT", "REQUIRED",   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",   and "OPTIONAL" are to be interpreted as described in RFC 2119 [10]   and indicate requirement levels for compliant SIP implementations.Handley, et al.             Standards Track                     [Page 8]RFC 2543            SIP: Session Initiation Protocol          March 19991.3 Definitions   This specification uses a number of terms to refer to the roles   played by participants in SIP communications. The definitions of   client, server and proxy are similar to those used by the Hypertext   Transport Protocol (HTTP) (RFC 2068 [11]). The terms and generic   syntax of URI and URL are defined in RFC 2396 [12]. The following   terms have special significance for SIP.   Call: A call consists of all participants in a conference invited by        a common source. A SIP call is identified by a globally unique        call-id (Section 6.12). Thus, if a user is, for example, invited        to the same multicast session by several people, each of these        invitations will be a unique call. A point-to-point Internet        telephony conversation maps into a single SIP call. In a        multiparty conference unit (MCU) based call-in conference, each        participant uses a separate call to invite himself to the MCU.   Call leg: A call leg is identified by the combination of Call-ID, To        and From.   Client: An application program that sends SIP requests. Clients may        or may not interact directly with a human user.  User agents and        proxies contain clients (and servers).   Conference: A multimedia session (see below), identified by a common        session description. A conference can have zero or more members        and includes the cases of a multicast conference, a full-mesh        conference and a two-party "telephone call", as well as        combinations of these.  Any number of calls can be used to        create a conference.   Downstream: Requests sent in the direction from the caller to the        callee (i.e., user agent client to user agent server).   Final response: A response that terminates a SIP transaction, as        opposed to a provisional response that does not. All 2xx, 3xx,        4xx, 5xx and 6xx responses are final.   Initiator, calling party, caller: The party initiating a conference        invitation. Note that the calling party does not have to be the        same as the one creating the conference.   Invitation: A request sent to a user (or service) requesting        participation in a session. A successful SIP invitation consists        of two transactions: an INVITE request followed by an ACK        request.Handley, et al.             Standards Track                     [Page 9]RFC 2543            SIP: Session Initiation Protocol          March 1999   Invitee, invited user, called party, callee: The person or service        that the calling party is trying to invite to a conference.   Isomorphic request or response: Two requests or responses are defined        to be isomorphic for the purposes of this document if they have        the same values for the Call-ID, To, From and CSeq header        fields. In addition, isomorphic requests have to have the same        Request-URI.   Location server: See location service.   Location service: A location service is used by a SIP redirect or        proxy server to obtain information about a callee's possible        location(s). Location services are offered by location servers.        Location servers MAY be co-located with a SIP server, but the        manner in which a SIP server requests location services is        beyond the scope of this document.   Parallel search: In a parallel search, a proxy issues several        requests to possible user locations upon receiving an incoming        request.  Rather than issuing one request and then waiting for        the final response before issuing the next request as in a        sequential search , a parallel search issues requests without        waiting for the result of previous requests.

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