Cisco SPA525G Administration Guide - Page 89

Configuring SIP, SPCP, and NAT, Session Initiation Protocol and Cisco IP Phones

Page 89 highlights

5 Configuring SIP, SPCP, and NAT The SPA and Wireless IP Phones use the following protocols: • Session Initiation Protocol (SIP)-SPA9X2, WIP310, SPA525G • Cisco Smart Phone Control Protocol (SPCP)-SPA525G only This chapter describes how to configure the phone protocols and other parameters. It contains the following sections: • Session Initiation Protocol and Cisco IP Phones, page 87 • Configuring SIP, page 89 • Configuring SPCP on the SPA525G, page 109 • Network Address Translation (NAT) and Cisco IP Phones, page 109 Session Initiation Protocol and Cisco IP Phones Cisco IP phones use Session Initiation Protocol (SIP), allowing interoperation with all ITSPs supporting SIP. SIP handles signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management controls the attributes of an end-to-end call. The following diagram shows a SIP request for connection to another subscriber in the network. In typical commercial IP telephony deployments, all calls go through a SIP proxy server. The requesting phone is called the SIP user agent server (UAS), while the receiving phone is called the user agent client (UAC). Cisco SPA and Wireless IP Phone Administration Guide 87

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5
Cisco SPA and Wireless IP Phone Administration Guide
87
Configuring SIP, SPCP, and NAT
The SPA and Wireless IP Phones use the following protocols:
Session Initiation Protocol (SIP)—SPA9X2, WIP310, SPA525G
Cisco Smart Phone Control Protocol (SPCP)—SPA525G only
This chapter describes how to configure the phone protocols and other
parameters. It contains the following sections:
Session Initiation Protocol and Cisco IP Phones, page 87
Configuring SIP, page 89
Configuring SPCP on the SPA525G, page 109
Network Address Translation (NAT) and Cisco IP Phones, page 109
Session Initiation Protocol and Cisco IP Phones
Cisco IP phones use Session Initiation Protocol (SIP), allowing interoperation with
all ITSPs supporting SIP.
SIP handles signaling and session management within a packet telephony
network.
Signaling
allows call information to be carried across network
boundaries.
Session management
controls the attributes of an end-to-end call.
The following diagram shows a SIP request for connection to another subscriber
in the network.
In typical commercial IP telephony deployments, all calls go through a SIP proxy
server. The requesting phone is called the SIP user agent server (UAS), while the
receiving phone is called the user agent client (UAC).