SIP | VoIP Standards and Specifications

The IETF developed SIP in reaction to the ITU-T H.323 recommendations. The IETF believed H.323 was inadequate for evolving IP telephony requirements because its command structure was too complex and its architecture was too centralized and monolithic. SIP is an application layer control protocol that can establish, modify, and terminate multimedia sessions or calls. SIP transparently supports name mapping and redirection services, allowing the implementation of ISDN and Intelligent Network telephony subscriber services. The early implementations of SIP have been in network carrier IP-Centrex trials. IP-PBX manufacturers are in the process of developing SIP-based versions of their current product offerings.

SIP was designed as part of the overall IETF multimedia data and control architecture that supports protocols such as Resource Reservation Protocol (RSVP), RTP, Real-Time Streaming Protocol (RTSP), Session Announcement Protocol (SAP), and Session Description Protocol (SDP). Figure 1 shows SIP and its associated protocols.

Figure 1: SIP signaling protocols.

SIP provides the necessary protocol mechanisms to support the following basic functions:

• Name translation and user location—Determination of the end system to be used for communication

• Feature negotiation—Allows station users involved in a call to agree on the features supported, recognizing that not all features are available to all station users

• Call participation management—During a call, a station user can conference other station users into the call or cancel connections to conferenced parties; station users can also be transferred or placed on hold

• Call feature changes—A station user should be able to change the call characteristics during the course of the call; new features may be enabled based on call requirements or new conferenced station users

The two major components in a SIP network are User Agents and Network Servers. A User Agent Client (UAC) initiates SIP requests, and a User Agent Server (UAS) receives SIP requests and return responses on user behalf. A Registration Server receives updates regarding the current user location, and a Proxy Server receives and forwards requests to a next-hop server, which has more information regarding called party location. A Redirect Server receives requests, determines next-hop server, and returns an address to client.

SIP request messages consist of three elements: Request Line, Header, and Message Body. SIP response messages consist of three elements: Status Line, Header, and Message Body.

Figure 2 shows the basic steps for a SIP call set-up.

Figure 2: SIP call setup.