Overview - Polycom IP 320 Administrator's Manual

Overview

This chapter provides an overview of the Session Initiation Protocol (SIP)
application and how the phones fit into the network configuration.
SIP is the Internet Engineering Task Force (IETF) standard for multimedia
communications over IP. It is an ASCII-based, application-layer control
protocol (defined in RFC 3261) that can be used to establish, maintain, and
terminate calls between two or more endpoints. Like other voice over IP
(VoIP) protocols, SIP 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.
For the SoundPoint IP / SoundStation IP / VVX phones to successfully
operate as a SIP endpoint in your network, it must meet the following
requirements:
• A working IP network is established.
• Routers are configured for VoIP.
• VoIP gateways are configured for SIP.
• The latest (or compatible) SoundPoint IP / SoundStation IP / VVX phone
SIP application image is available.
• A call server is active and configured to receive and send SIP messages.
For more information on IP PBX and softswitch vendors, go to
http://www.polycom.com/techpartners1/
This chapter contains information on:
• Where SoundPoint IP / SoundStation IP / VVX Phones Fit
• Session Initiation Protocol Application Architecture
• Available Features
• New Features in SIP 3.2
To install your SoundPoint IP / SoundStation IP / VVX phones on the
network, refer to Setting up Your System
SoundPoint IP / SoundStation IP / VVX phones with the desired features,
.
on page 3-1. To configure your
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