ion, and the ability of the network to identify end users as they move. Called parties can redirect callers to any number of locations that are represented by URLs. H.323 can do this to a limited degree, but it does not allow a Video Communications: The Whole Picture gatekeeper to proxy a request to multiple servers. SIP supports establishment and termination of multimedia communications as follows: 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 respond Call setup-Ringing or establishing call parameters for called and calling parties Call handling-Such as transfer and termination of calls SIP is part of the IETF multimedia data and control architecture. As such, it incorporates RSVP (RFC 2205) for reserving network resources, RTP (RFC 1889) for transporting real-time data and providing QoS feedback, RTSP (RFC 2326) for controlling delivery of streaming media, SAP for advertising multimedia sessions via multicast, and SDP (RFC 2327) for describing multimedia sessions. However, SIP's functionality and operation do not specifically depend upon these protocols, and can be used in conjunction with other call setup and signaling protocols. An end system can use SIP exchanges as a protocol-independent method to determine, from a given address, the appropriate end system address and protocol. SIP can be used, for example, to determine that the callee is reachable via the PSTN, to indicate the phone number to be called, and even to suggest a viable Internet-to-PSTN gateway. Similarly, SIP could be used to determine that a party can be reached via H.323, to obtain the H.245 gateway and user address, and then to use H.225 to establish the call. Although SIP does not offer conference control services (e.g., floor control or voting) and does not prescribe how to manage a conference, it can be used to introduce conference control protocols. SIP does n