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1823 VoIP Gateway User's Guide
Chapter 6 VoIP
6.1 SIP Configuration
SIP, the Session Initiation Protocol, is a signaling protocol for Internet conferencing, telephony, presence,
events notification, and instant messaging. It is the Internet Engineering Task Force's (IETF's) standard
for multimedia conferencing over IP. It is designed to address the functions of signaling and session
management within a packet telephony network. Signaling allows call information to be carried across
network boundaries. Session management provides the ability to control the attributes of an end-to-end
call.
Session Initiation Protocol is a peer-to-peer protocol. There are four components in the SIP standard:
User Agent (UA), Proxy Server, Registrar Server, and, Redirect Server. This document describes SIP
User Agents and the call establishment between User Agents.
The SIP page appears when the SIP link is clicked under the VoIP menu. The initial screen lists the Local
SIP Parameters.
Click on the tabs to access the different SIP configuration screens:
•
Local: VoIP host/voice CODEC configuration for the gateway, and SIP proxy/registrar
configuration.
•
PhoneList: SIP phone number configuration for the gateway.
•
Remote: Buddy phone number configuration. The gateway looks up the buddy list for the dialed
phone number first. If the phone number is not listed, then the gateway sends an inquiry to the
SIP proxy. It is also suitable for peer-to-peer calling in cases where there is no SIP
Registrar/proxy.
•
CallForward: Call forwarding configuration.
•
Dial Plan:To set a particular method to filter you phone numbers in the list.
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May 2005
1823-A2-GB20-00
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