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.