The Session Initiation Protocol: Providing advanced telephony services across the Internet

During the past few years, Internet telephony has evolved from a toy for the technically savvy to a technology that, in the not too distant future, may replace the existing circuit-switched telephone network. Supporting the widespread use of Internet telephony requires a host of standardized protocols to ensure quality of service (QoS), transport audio and video data, provide directory services, and enable signaling. Signaling protocols are of particular interest because they are the basis for advanced services such as mobility, universal numbers, multiparty conferencing, voice mail, and automatic call distribution. Two signaling protocols have emerged to fill this need: the ITU-T H.323 suite of protocols and session initiation protocol (SIP), developed by the Internet Engineering Task Force (IETF). In this paper we examine how SIP is used in Internet telephony. We present an overview of the protocol and its architecture, and describe how it can be used to provide a number of advanced services. Our discussion of some of SIP's strengths - its simplicity, scalability, extensibility, and modularity - also analyzes why these are critical components for an IP telephony signaling protocol. SIP will prove to be a valuable tool, not just for end-to-end IP services, but also for controlling existing phone services.