Specifications

Feature Summary
2
Cisco IOS Release 12.0(3)T
Figure 2 VoIP Used as a PSTN Gateway for Internet Telephone Traffic
How VoIP Processes a Telephone Call
Before configuring VoIP on your Cisco AS5300, it helps to understand what happens at an
application level when you place a call using VoIP. The general flow of a two-party voice call using
VoIP is as follows:
1 The user picks up the handset; this signals an off-hook condition to the signalling application part
of VoIP in the Cisco AS5300.
2 The session application part of VoIP issues a dial tone and waits for the user to dial a telephone
number.
3 The user dials the telephone number; those numbers are accumulated and stored by the session
application.
4 After enough digits are accumulated to match a configured destination pattern, the telephone
number is mapped to an IP host via the dial plan mapper. The IP host has a direct connection to
either the destination telephone number or a PBX that is responsible for completing the call to
the configured destination pattern.
5 The session application then runs the H.323 session protocol to establish a transmission and a
reception channel for each direction over the IP network. If the call is being handled by a PBX,
the PBX forwards the call to the destination telephone. If RSVP has been configured, the RSVP
reservations are put into effect to achieve the desired quality of service (QoS) over the IP
network.
6 The CODECs are enabled for both ends of the connection and the conversation proceeds using
RTP/UDP/IP as the protocol stack.
PSTN
408 526-4000
408 526-4001
408 526-4002
310 520-1000 310 520-1003
310 520-1001 310 520-1002
408 526-4003
Central
office
Cisco AS5300
10.1.1.1
10.1.1.2
IP
cloud
Cisco 3640
Voice port
1/0/0
10352