SIP was designed as a multimedia protocol that could take advantage of the architecture and messages found in popular Internet applications. By using a distributed architecture—with universal resource locators (URLs) for naming and text-based messaging—SIP attempts to take advantage of the Internet model for building VoIP networks and applications. In addition to VoIP, SIP is used for videoconferencing and instant messaging.

As a protocol, SIP only defines how sessions are to be set up and torn down. It utilizes other IETF protocols to define other aspects of VoIP and multimedia sessions, such as SDP for capabilities exchange, URLs for addressing, Domain Name Systems (DNSs) for service location, and Telephony Routing Over IP (TRIP) for call routing.

Figure 3. SIP Networks

Although the IETF has made great progress defining extensions that allow SIP to work with legacy voice networks, the primary motivation behind the protocol is to create an environment that supports next-generation communications models that utilize the Internet and Internet applications.

As a protocol used in a distributed architecture, SIP allows companies to build large-scale networks that are scalable, resilient, and redundant. It provides mechanisms for interconnecting with other VoIP networks and for adding intelligence and new features on either the endpoints or the SIP proxy or redirect servers.