Bandwidth Management

Bandwidth management with Cisco CallManager call admission control enables you to control the audio and video quality of calls over a WAN link by limiting the number of calls that are allowed on that link simultaneously. In a packet-switched network, audio and video quality can begin to degrade when a link carries too many active calls and the bandwidth is oversubscribed. Call admission control regulates audio and video quality by limiting the number of calls that can be active on a particular link at the same time. Call admission control does not guarantee a particular level of audio or video quality on the link (this is the job of QoS), but it does allow you to regulate the amount of bandwidth that active calls on the link consume.

The actual bandwidth required is more than just the speed of the video call. The speed of the video call is just the payload, but the final packet also includes some amount of overhead for header information that encapsulates the payload into RTP segments, User Datagram Protocol (UDP) frames, IP packets, and finally a Layer 2 transport medium (such as Ethernet frames, ATM cells, or Frame Relay frames). References to the video call bandwidth should include the sum of the video call speed and all packetization overhead (RTP, UDP, IP, and Layer 2).

Video Call Bandwidth Requirement

A typical video call consists of two media channels: one for the video stream and one for the audio stream. These channels are referred to as logical channels in the H.323 protocol, and each logical channel is negotiated separately. For the call to succeed, Cisco CallManager checks that audio is successfully signaled; if video cannot be negotiated, the call will be an audio-only call. The audio channel consists of the actual audio bit rate. This bit rate is dictated by the audio codec in use. In the case of a G.711 codec, the bit rate is 64 kbps, whereas in the case of a G.729 codec, it is 8 kbps. For an audio channel, only the pure audio data is considered, not the packetization overhead.

The bit rate available for the video channel depends on the negotiated audio codec and the video call speed. The video channel bit rate is the speed of the video call minus the bit rate of the codec used for the audio channel. In the case of a 384-kbps video call with an audio channel that uses G.711, the bit rate left for the video channel is 320 kbps (384 kbps minus 64 kbps). In the case of an audio channel using the G.729 codec, the payload of the same video call (384 kbps) leaves 376 kbps. Table 28-3 lists possible video call speeds, their possible audio channel codecs, and the associated video channel bit rates.

Table 28-3. Video Call Speeds and the Associated Audio and Video Codecs

Video Call Speed

Audio Codec and Rate

Video Codec and Rate

128 kbps

G.711 at 64 kbps

H.261 or H.263 at 64 kbps

128 kbps

G.729 at 8 kbps

H.261 or H.263 at 120 kbps

128 kbps

G.728 at 16 kbps

H.261 or H.263 at 112 kbps

384 kbps

G.729 at 8 kbps

H.261 or H.263 at 376 kbps

384 kbps

G.711 at 64 kbps

H.261 or H.263 at 320 kbps

768 kbps

G.729 at 8 kbps

H.261 or H.263 at 760 kbps

768 kbps

G.711 at 64 kbps

H.261 or H.263 at 704 kbps

1.472 Mbps

G.729 at 8 kbps

H.261 or H.263 at 1.464 Mbps

1.472 Mbps

G.711 at 64 kbps

H.261 or H.263 at 1.408 Mbps

7 Mbps

G.729 at 8 kbps

Wideband at 7 Mbps (minus 8 kbps for the audio stream)

7 Mbps

G.711 at 64 kbps

Wideband at 7 Mbps (minus 64 kbps for the audio stream)

For example, as shown in Figure 28-3, a 384-kbps video call may be G.711 at 64 kbps (for audio) plus 320 kbps (for video). If the audio codec for a video call is G.729 (at 8 kbps), the video rate increases to maintain a total bandwidth of 384 kbps. If the call involves an H.323 endpoint, the H.323 endpoint might use less than the total video bandwidth that is available. An H.323 endpoint can always choose to send at less than the maximum bit rate for the call.

Figure 28-3. Video Call Bandwidth Requirement Examples

Note

None of these values include packetization overhead.

 

Calculating the Total Bandwidth

The two main factors that influence bandwidth requirements for video calls are the media channels and the bandwidth used per call.

Actual Bandwidth Used Per Video Call

To calculate the exact overhead ratio for video, it is recommended that you add about 20 percent to the video call speed regardless of which type of Layer 2 medium the packets are traversing. The additional 20 percent gives plenty of headroom to allow for the differences among Ethernet, ATM, Frame Relay, PPP, High-Level Data Link Control (HDLC), and other transport protocols and also some cushion for the bursty nature of video traffic. Keep in mind that the amount of bandwidth consumed by the video correlates directly to the amount of movement on the video. Video streams with constant motion of the person speaking and the background (such as speaking in front of an ocean backdrop) can potentially double the amount of bandwidth required. The 20 percent rule is a general guide, but your actual results might vary.

Table 28-4 shows the recommended bandwidth values to use for some of the more popular video call speeds, incorporating this 20 percent margin.

Table 28-4. Video Call Speeds and the Associated Audio and Video Codecs

Video Call Speed Requested by Endpoint

Actual Bandwidth Required on the Link

128 kbps

153.6 kbps

256 kbps

307.2 kbps

384 kbps

460.8 kbps

512 kbps

614.4 kbps

768 kbps

921.6 kbps

1.5 Mbps

1.766 Mbps

7 Mbps

8.4 Mbps

 

Call Admission Control in Cisco CallManager

As shown in Figure 28-4, the location configuration settings for Cisco CallManagerbased call admission control have also been enhanced, compared to earlier Cisco CallManager releases not supporting video, to provide for accounting of video bandwidth on a per-call and aggregate basis. The location setting for call admission control defines the overall bandwidth allowed for all video calls to a certain location. That is the video call speed for all video calls. To allow five video calls with a bandwidth of 384 kbps for each video call (defined in the Cisco CallManager regions), the value to enter in the Cisco CallManager location is 1920 kbps.

Figure 28-4. Cisco CallManager Enhanced Location Configuration

For video calls, the negotiated bandwidth for a video-enabled device typically includes both audio and video; for example, a 384-kbps video call comprises 64-kbps audio and 320-kbps video channels. For voice-only calls, the region uses the same setting that is used for the audio channel in video calls. The negotiated bandwidth for an IP telephony device includes the "real" audio bandwidth including IP overhead; for example, a G.711 64-kbps audio call uses 80 kbps, and this value has to be entered in the Cisco CallManager location configuration.

H.323 gatekeepers have slightly different bandwidth measurements. The H.323 specification dictates that the bandwidth values must be entered as twice the codec bit rate. For example, a 384-kbps video call would be entered as 768 kbps in the gatekeeper. A G.711 audio-only call would be entered as 128 kbps in the gatekeeper.

Note

Call admission control behavior changed in Cisco CallManager Release 3.2(2)c and Cisco IOS Software Release 12.2(2)XA. Before that release, Cisco CallManager asked for bit rate plus Layer 3 overhead, and Cisco IOS gateways asked for 64 kbps, regardless of the type of call.


Call Admission Control Within a Cluster

Part I: Cisco CallManager Fundamentals

Introduction to Cisco Unified Communications and Cisco Unified CallManager

Cisco Unified CallManager Clustering and Deployment Options

Cisco Unified CallManager Installation and Upgrades

Part II: IPT Devices and Users

Cisco IP Phones and Other User Devices

Configuring Cisco Unified CallManager to Support IP Phones

Cisco IP Telephony Users

Cisco Bulk Administration Tool

Part III: IPT Network Integration and Route Plan

Cisco Catalyst Switches

Configuring Cisco Gateways and Trunks

Cisco Unified CallManager Route Plan Basics

Cisco Unified CallManager Advanced Route Plans

Configuring Hunt Groups and Call Coverage

Implementing Telephony Call Restrictions and Control

Implementing Multiple-Site Deployments

Part IV: VoIP Features

Media Resources

Configuring User Features, Part 1

Configuring User Features, Part 2

Configuring Cisco Unified CallManager Attendant Console

Configuring Cisco IP Manager Assistant

Part V: IPT Security

Securing the Windows Operating System

Securing Cisco Unified CallManager Administration

Preventing Toll Fraud

Hardening the IP Phone

Understanding Cryptographic Fundamentals

Understanding the Public Key Infrastructure

Understanding Cisco IP Telephony Authentication and Encryption Fundamentals

Configuring Cisco IP Telephony Authentication and Encryption

Part VI: IP Video

Introducing IP Video Telephony

Configuring Cisco VT Advantage

Part VII: IPT Management

Introducing Database Tools and Cisco Unified CallManager Serviceability

Monitoring Performance

Configuring Alarms and Traces

Configuring CAR

Using Additional Management and Monitoring Tools

Part VIII: Appendix

Appendix A. Answers to Review Questions

Index

show all menu



Authorized Self-Study Guide Cisco IP Telephony (CIPT)
Cisco IP Telephony (CIPT) (Authorized Self-Study) (2nd Edition)
ISBN: 158705261X
EAN: 2147483647
Year: 2004
Pages: 329
Similar book on Amazon

Flylib.com © 2008-2017.
If you may any questions please contact us: flylib@qtcs.net