Session Initiation Protocol

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

  • How SIP works
  • SIP call flow
  • SIP pros and cons
  • Dial plan considerations
  • How to implement SIP gateways
  • Some ways to secure SIP gateways
  • Allowing H.323 to SIP connections
  • Troubleshooting tools

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

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Cisco Voice Gateways and Gatekeepers
Cisco Voice Gateways and Gatekeepers
ISBN: 158705258X
EAN: 2147483647
Year: 2004
Pages: 218
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