Cisco Systems H.323/SIP Manual De Usuario

When a call is placed from a PSTN phone to a Cisco IP phone, the call is routed through a voice gateway, 
which is the demarcation point where the circuit-switched voice network meets the packet-switched data 
network. The primary responsibility of the voice gateway is to ensure that PSTN voice traffic reaches 
the data network and vice versa. You can use the voice gateway to forward an IP or PSTN call to its 
opposing network through Cisco Unified CallManager or a PBX. 

When a call is placed from an Cisco IP phone, it is routed to Cisco Unified CallManager, which is 
responsible for setting up the call, directing the call to the called device, and sending network 
information— such as the IP address, UDP port number, and communication capabilities of the called 
device—to the Cisco IP phone. After receiving the information, the Cisco IP phone sends its digitized 
voice traffic directly to the called device.

The following steps describe how Cisco IP phones and PSTN phones use Release 5.2.1 to access the 
Cisco Unified MeetingPlace Audio Server system, as shown in 

.

.

On the Cisco IP phone dial pad, the caller enters a 
dialable number to the Cisco Unified 
MeetingPlace Audio Server system that will host 
the meeting. 

By using a PSTN phone, the caller dials the number 
to the voice gateway. 

The call is immediately routed by using SSP to 
Cisco Unified CallManager.

The voice gateway routes the call to Cisco Unified 
CallManager.

Cisco IP phone

Cisco CallManager

PSTN phone

Cisco
MeetingPlace
Audio Server

Cisco MeetingPlace

H.323/SIP IP Gateway

Voice gateway

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