Configure Sip And Nat; Sip And Cisco Unified Ip Conference Phone 8831 For Third-Party Call Control - Cisco 8831 Administration Manual

Unified ip conference phone for third-party call control
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Configure SIP and NAT

The Cisco Unified IP Conference Phone 8831 for Third-Party Call Control uses the following protocol:
This chapter describes how to configure the SIP phone protocol:
SIP and Cisco Unified IP Conference Phone 8831 for Third-Party
Call Control
The Cisco Unified IP Conference Phone 8831 for Third-Party Call Control uses Session Initiation
Protocol (SIP), which allows interoperation with all IT service providers that support SIP. SIP is an
IETF-defined signaling protocol that controls voice communication sessions in an IP network.
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.
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).
SIP message routing is dynamic. If a SIP proxy receives a request from a UAS for a connection but
cannot locate the UAC, the proxy forwards the message to another SIP proxy in the network. When the
UAC is located, the response is routed back to the UAS, and a direct peer-to-peer session is established
between the two UAs. Voice traffic is transmitted between UAs over dynamically-assigned ports using
Real-time Protocol (RTP).
RTP transmits real-time data such as audio and video; it does not guarantee real-time delivery of data.
RTP provides mechanisms for the sending and receiving applications to support streaming data.
Typically, RTP runs on top of UDP.
Session Initiation Protocol (SIP)
SIP and Cisco Unified IP Conference Phone 8831 for Third-Party Call Control, page 3-1
Configure SIP, page 3-4
Configure NAT Support Parameters, page 3-10
Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide
3
C H A P T E R
3-1

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