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Cisco AVVID Network Infrastructure Enterprise Quality of Service Design

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Chapter 1 Overview

Why is Quality of Service Required for AVVID?

Delay

Delay (or latency) is the amount of time it takes a packet to reach the receiving endpoint after being
transmitted from the sending endpoint. This time period is termed the "end-to-end delay" and can be
broken into two areas: fixed network delay and variable network delay. Fixed network delay includes
encoding/decoding time (for voice and video), as well as the finite amount of time required for the
electrical/optical pulses to traverse the media en route to their destination. Variable network delay
generally refers to network conditions, such as congestion, that may affect the overall time required for
transit.

In data networks carrying voice, there are three types of delay:

Packetization delay, which is the amount of time that it takes to sample and encode the analog voice
signals and turn them in to packets.

Serialization delay, which is the amount of time that it takes to place the bits of the data packets onto
the physical media.

Propagation delay, which is the amount of time it takes to transmit the bits of a packet across the
physical wire.

Delay Variation

Delay variation (or jitter) is the difference in the end-to-end delay between packets. For example, if one
packet required 100 ms to traverse the network from the source-endpoint to the destination-endpoint and
the following packet required 125 ms to make the same trip, then the delay variation is calculated as
25 ms.

Each end station in a VoIP or Video over IP conversation has a jitter buffer. Jitter buffers are used to
smooth out changes in arrival times of data packets containing voice. A jitter buffer is dynamic and can
adjust for up to a 30 ms average change in arrival times of packets. If you have instantaneous changes
in arrival times of packets that are outside of the capabilities of a jitter buffer's ability to compensate
you will have jitter buffer over-runs and under-runs.

A jitter buffer under-run occurs when arrival times of packets increases to the point where the jitter
buffer has been exhausted and contains no packets to be processed by the DSPs when it is time to
play out the next piece of voice or video.

A jitter buffer over-run occurs when packets containing voice or video arrive faster than the jitter
buffer can dynamically resize itself to accommodate. When this happens packets are dropped and
when it is time to play out the voice or video samples contained in the dropped packets quality is
degraded.

Quality of Service Requirements for Voice

When addressing the QoS needs of voice traffic, keep the following in mind:

Loss should be no more than 1%.

One-way latency should be no more than 150-200 ms.

Average jitter should be no more than 30 ms.

21-106 kbps of guaranteed priority bandwidth is required per call (depending on the sampling rate,
codec and Layer 2 overhead).

150 bps (+ Layer 2 overhead) per phone of guaranteed bandwidth is required for Voice Control
traffic.