Investigate and implement Session Initiation Protocol (SIP) gateways

As deployments of voice over IP (VoIP) networks continue at a rapid pace, network professionals need to understand the real-world implementation issues surrounding voice gateways, which handle the many tasks involved in translating between transmission formats and protocols and act as the interface between an IP telephony network and the PSTN or PBX. Focus on the Session Initiation Protocol (SIP) in this sample chapter from Cisco Voice Gateways and Gatekeepers. SIP is an Internet Engineering Task Force (IETF) standard call control protocol 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. SIP performs four basic tasks:

• Locating users, resolving their SIP address to an IP address
• Negotiating capabilities and features among all the session participants
• Changing the session parameters during the call
• Managing the setup and teardown of calls for all users in the session

Learn how SIP works and how to configure SIP gateways. Investigate SIP call flow, SIP pros and cons, dial plan considerations, security issues, and troubleshooting tools.

Title: Cisco Voice Gateways and Gatekeepers
ISBN: 1-58705-258-X
Published: August 2006
Authors: David Mallory, Ken Salhoff, Denise Donohue
Chapter: Chapter 4: Session Initiation Protocol
Published by Cisco Press