📄 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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