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Chapter 4
Configuring the PA-VXA, PA-VXB, and PA-VXC
How VoIP Handles a Typical Telephone Call
Before configuring VoIP on your Cisco 7200 series router, Cisco 7200 VXR router, Cisco 7301 router,
Cisco 7401ASR router, or Cisco 7500 series router, 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.
2.
3.
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Configuration Tasks
To configure VoIP on the Cisco 7200 series routers, Cisco 7200 VXR routers, Cisco 7301 router,
Cisco 7401ASR routers, or Cisco 7500 series routers, you need to perform the following steps:
Configure your IP network to support real-time voice traffic. Fine-tuning your network to adequately
Step 1
support VoIP involves a series of protocols and features geared toward quality of service (QoS). To
configure your IP network for real-time voice traffic, you need to take into consideration the entire scope
of your network, and then select and configure the appropriate QoS tool or tools:
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OL-3592-02
The user picks up the handset; this signals an off-hook condition to the signaling application part of
VoIP in the Cisco 7200 series routers, Cisco 7200 VXR routers, Cisco 7301 router, Cisco 7401ASR
routers, or Cisco 7500 series routers.
The session application part of VoIP issues a dial tone and waits for the user to dial a telephone
number.
The user dials the telephone number; those numbers are accumulated and stored by the session
application.
After enough digits are accumulated to match a configured destination pattern, the telephone
number is mapped to an IP host through 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.
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 Resource Reservation Protocol (RSVP) has
been configured, the RSVP reservations are put into effect to achieve the desired quality of service
over the IP network.
The codecs are enabled for both ends of the connection and the conversation proceeds using
Realtime Transport Protocol/User Datagram Protocol/Internet Protocol (RTP/UDP/IP) as the
protocol stack.
Any call-progress indications (or other signals that can be carried in-band) are cut through the voice
path as soon as an end-to-end audio channel is established. Signaling that can be detected by the
voice ports (for example, in-band DTMF digits after the call setup is complete) is also trapped by
the session application at either end of the connection and carried over the IP network encapsulated
in Real Time Conferencing Protocol (RTCP) using the RTCP extension mechanism.
When either end of the call hangs up, the RSVP reservations are torn down (if RSVP is used) and
the session ends. Each end becomes idle, waiting for the next off-hook condition to trigger another
call setup.
RSVP
Multilink PPP with interleaving
RTP header compression
T1/E1 Digital Voice Port Adapter Installation and Configuration
Configuring Voice over IP
4-7
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Troubleshooting
