Implementing IP Telephony
In the enterprise, IP telephony is deployed to replace a PBX. A typical PBX
contains a switching function (the “brains”) and cards that attach extensions
(station cards) and connect to the outside world (line cards). Figure 2-4
shows the evolution from an old PBX to a modern distributed IP telephony
solution.
Figure 2-4 Evolution from PBX to IP Telephony
CCNP ONT
Chapter 2: Cisco VoIP [ 261 ]
Switching Engine
Network
Router with FXO
PC with
Softphone
Call Manager
Router with FXS
( )
( )
PBX
Other Resources
Station Card
Connections to desktop phones
External Line Card
Connections to PSTN
A Cisco Call Manager takes the place of the “brains” and helps end stations
understand how to reach each other. CCM also oversees the dial plan,
produces utilization reports, and determines functionality. CCM is typically
deployed in a cluster, so that the system does not rely on one machine.
Note
Cisco Call Manager Express runs on a router and can be used for small offices. Routers are also
deployed as backup call managers (this is called Survivable Remote Site Telephony or SRST),
so being disconnected from a remote CCM does not disable a branch phone system.
IP phones and soft phones connect directly to the network, whereas legacy
phones connect to the network through FXS ports on routers. Routers operating
this way are called gateways. Think of the network and gateways as
being equivalent to the station cards in an old PBX.
Routers with external connections, such as FXO ports, are also called gateways.
In this scenario, however, the router takes the place of an external line card.
Telephony deployments follow one of four models:
■ Single Site—One office uses a CCM cluster to handle local phones.
■ Multisite with centralized call processing—One CCM cluster at headquarters
handles local and remote phones. Branch offices typically are
set up with SRST.
■ Multisite with distributed call processing—Each site has a CCM
cluster.
■ Clustering over WAN—The CCM cluster is distributed between locations.
One other piece, not shown or discussed so far, is Call Admission Control
(CAC). Usually data is described as “better to degrade service than to deny
service,” which is to say that when more users need service, everyone goes
slower. But the voice world has never said that one more user would cause
quality to go down. In fact, voice engineers would say “It’s better to deny
service than to degrade service.”
The problem is, how do you limit the number of calls going across a VoIP
network? Intuitively, there is nothing to prevent one more person from
calling. This is where CAC comes in. CAC is a tool that tracks the number
of calls and—when it reaches a threshold value—prevents another call. CAC
is an important part of an IP telephony solution.