Voice and Video in A Campus Network
Voice and Video in A Campus Network
by Denise Donohue
Chapter 8
Voice and Video in a Campus Network
Voice over IP (VoIP) has become common in the business world, and now Video over IP is becoming more integrated into networks. Neither should be added to a network without advance planning to ensure good voice and video quality. Some benefits of converging voice, video, and data networks include:
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Consolidating network expenses: Only one wire and one switch port are needed per user. One network to provision and manage. More efficient bandwidth use: Bandwidth can be used for data when there is not an active voice/video session. Lower telephony costs: Internal calls use the data network, rather than the PSTN. Innovative services: Ability to unify a companys various methods of communication. For service providers, the ability to sell new services: Can lead to increased revenue, flexibility in pricing, and access to new communication devices.
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Voice, video, and data have different network requirements. Voice requirements include low bandwidth, little delay, small amounts of jitter (variable delay), small amounts of packet loss, a highly available network, and PoE. Security requirements are about average, but management is highly important. Video requirements depend on whether it is a one-way stream or an interactive video session. One-way streams use a fairly steady amount of bandwidth but in interactive sessions the bandwidth varies widely. Typical requirements include high bandwidth, little delay, small amounts of jitter, and little packet loss. High availability is not as important, and PoE is not needed. Security and management needs are medium.
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by Denise Donohue
Data requirements typically include high bandwidth, but delay and jitter are not crucial. A highly available network is needed, but PoE is not. Data security should be high, with medium management levels.
IP phones: Provide voice and applications to the user Cisco Unified Communications Manager (UCM): Serves as an IP PBX. Registers phones, controls calls Voice gateways: Translates between PSTN and IP calls and provides backup to the Cisco UCM (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 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 and Presence
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2010 Pearson Education, Inc. All rights reserved. This publication is protected by copyright. Please see page 112 for more details.
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by Denise Donohue
WAN
User PC
IP Phone
PSTN
VoIP traffic consists of two types: voice bearer and call control signaling. Voice bearer traffic is carried over the UDPbased Real Time Protocol (RTP). Call control uses one of several different protocols to communicate between the phone and UCM and between the UCM and the voice gateways.
What features are needed?: Power for IP phones, security for voice calls, and Quality of Service (QoS) to control bandwidth, delay, jitter, and packet loss. The physical plant: Cabling at least CAT-5.
2010 Pearson Education, Inc. All rights reserved. This publication is protected by copyright. Please see page 112 for more details.
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by Denise Donohue
Electrical power for the IP phones: Use either PoE from Catalyst switch or power inline module, or a power brick. 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 call-control traffic in your planning. Network management: Monitor and proactively manage the network so that it is always available. Need voice VLANs on the switches and DHCP for the phones. High availability: Provide redundant hardware links. Need uninterruptible power supply (UPS) with auto-restart, monitoring, and four-hour response contract. Might need generator backup. Maintain correct operating temperatures.
Maximum delay of 150200 ms (one-way) No more than 1 percent packet loss Maximum average jitter of 30 ms Bandwidth of 21106 kbps per call, plus approximately 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
Voice VLANs
Cisco switches can be configured to dynamically place IP telephones into a Voice, or auxiliary, VLAN separate from the data VLANs. They can do this even when the phone and PC are physically connected to the same switch port. Voice VLANs enable 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 make troubleshooting easier.
2010 Pearson Education, Inc. All rights reserved. This publication is protected by copyright. Please see page 112 for more details.
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by Denise Donohue
Cisco IP phones have a small internal switch that places an 802.1q tag on voice traffic and marks the Class of Service (CoS) bits in the tag. Data traffic is sent untagged over the native VLAN. The switch port does not actually become a trunk and still can be configured as an access port. It is considered a multi-VLAN access port.
Prioritizes access to resources so that critical traffic can be served Allows good management of network resources
2010 Pearson Education, Inc. All rights reserved. This publication is protected by copyright. Please see page 112 for more details.