The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard call control protocol, based on research at Columbia University by Henning Schulzrinne and his team. The first SIP RFC, number 2543, was published in 1999. Since then, much work has been done, and numerous RFCs have been published to solidify and extend SIP capabilities.
SIP is designed to provide signaling and session management for voice and multimedia connections over packet-based networks. It is a peer-to-peer protocol with intelligent endpoints and distributed call control, such as H.323. Gateways that use SIP do not depend on a call agent, although the protocol does define several functional entities that help SIP endpoints locate each other and establish a session.
In this chapter you will learn
Part I: Voice Gateways and Gatekeepers
Gateways and Gatekeepers
Part II: Gateways
Media Gateway Control Protocol
H.323
Session Initiation Protocol
Circuit Options
Connecting to the PSTN
Connecting to PBXs
Connecting to an IP WAN
Dial Plans
Digit Manipulation
Influencing Path Selection
Configuring Class of Restrictions
SRST and MGCP Gateway Fallback
DSP Resources
Using Tcl Scripts and VoiceXML
Part III: Gatekeepers
Deploying Gatekeepers
Gatekeeper Configuration
Part IV: IP-to-IP Gateways
Cisco Multiservice IP-to-IP Gateway
Appendix A. Answers to Chapter-Ending Review Questions
Index