Linksys SPA932 Manual De Usuario

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Linksys SPA9x2 Phone Administration Guide
Cisco Confidential--First Draft
9
Session Initiation Protocol and SPA9x2 Phones
Introducing Linksys SPA9x2 Phones
Session Initiation Protocol and SPA9x2 Phones
Linksys SPA9x2 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). 
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). 
The Internet protocol 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 the UDP protocol. See 
.
SIP Over TCP 
To guarantee state-oriented communications, SPA9x2 phones can use TCP as the transport 
protocol for SIP. This protocol is “guaranteed delivery”, which assures that lost packets are 
retransmitted. TCP also guarantees that the SIP packages are received in the same order that 
they were sent. 
SIP UA
SIP UA
SIP Proxy
SIP Proxy
RTP
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SIP Proxy