Polycom 1725-11530-310 User Manual

2 - 1

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 

conferencing 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 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 phone SIP 

application image is available.

• A call server is active and configured to receive and send SIP messages.

http://www.polycom.com/techpartners1/

 .

This chapter contains information on:

To install your SoundPoint IP / SoundStation IP phones on the network, refer 

to 

. To configure your SoundPoint IP / 

SoundStation IP phones with the desired features, refer to