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.

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.


Preparing the Network for VoIP

When adding voice or video to an existing network, you should examine several things in advance to provide the high level of availability users expect in their phone system:

• What features are needed?—Power for IP phones, voice VLANs on the switches, network redundancy for high availability, security for voice calls, and Quality of Service (QoS) settings.

• The physical plant—Cabling at least CAT-5.

• Electrical power for the IP phones—Use either inline power from Catalyst switch or power patch panel. Need uninterruptible power supply (UPS) with auto-restart, monitoring, and 4-hour response contract. May need generator backup. Maintain correct operating temperatures.

• Bandwidth—Commit no more than 75 percent of bandwidth. Consider all types of traffic—voice, video, and data. Have more than enough bandwidth if possible. Include both voice and callcontrol traffic in your planning.

• Network management—Need to monitor and proactively manage the network so that it does not go down.

Network and Bandwidth Considerations

The network requirements for VoIP include:

• Maximum delay of 150–200 ms (one-way)
• No more than 1 percent packet loss
• Maximum average jitter of 30 ms
• Bandwidth of 21–106 kbps per call, plus about 150 bps per phone for control traffic

A formula to use when calculating bandwidth needed for voice calls is as follows:

(Packet payload + all headers) * Packet rate per second


Auxiliary (or Voice) VLANs

Cisco switches can be configured to dynamically place IP telephones into a VLAN separate from the data VLANs. They can do this even when the phone and PC are physically connected to the same switch port. This is called an auxiliary VLAN or a voice VLAN. Voice VLANs allow phones to be dynamically placed in a separate IP subnet from hosts, to have QoS (using 802.1Q/p headers) and security policies applied, and makes troubleshooting easier.