VoIP in a Campus Network
Many companies are integrating Voice over IP (VoIP) into their networks.
Figure 7-1 shows some components of a VoIP system, which can include the
following:
■ IP phones—Provide voice and applications to the user.
■ Voice gateways—Translates between PSTN and IP calls and provides
backup to the Cisco CallManager (IP PBX, or Call Agent).
■ Gatekeepers—An optional component that can do call admission
control, allocate bandwidth for calls, and resolve phone numbers into
IP addresses.
■ Cisco CallManager—Serves as an IP PBX. Registers phones, controls
calls.
■ Video conferencing unit—Allows voice and video in the same phone
call.
■ Multipoint control unit—Allows multiple participants to join an audio
and/or video conference call.
■ Application server—Provides services such as Unity voice mail.
Figure 7-1 Some Components of a VoIP System
WAN
PSTN
Unity Server
CallManager
Server
Power over
Ethernet Switch
User PC IP Phone Voice and WAN
Gateway
Video Conferencing
Camera
Voice and data have different network requirements. Although TCP data
adjusts to dropped packets, packet loss is one of the biggest enemies of voice
transmissions and is often caused by jitter and congestion. Jitter (variable
delay) causes buffer over- and under-runs. Congestion at the interface can be
caused by traffic from a fast port being switched to exit out a slower port,
which causes the transmit buffer to be overrun.
VoIP traffic consists of two types: voice bearer and call control signaling.
Voice bearer traffic is carried over the UDP-based Real Time Protocol (RTP).
Call control uses one of several different protocols to communicate between
the phone and CallManager and between the CallManager and the voice
gateways.