Abstract
Internet telephony offers the opportunity to design a global multimedia communications system that may eventually replace the existing telephony infrastructure. We describe the upper-layer protocol components that are specific to Internet telephony services: the Real-Time Transport Protocol (RTP) to carry voice and video data, and the Session Initiation Protocol (SIP) for signaling. We also mention some complementary protocols, including the Real Time Streaming Protocol (RTSP) for control of streaming media, and the Wide Area Service Discovery Protocol (WASRV) for location of telephony gateways.
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