Nortel 9110 User manual

Remote Gateway 9100 Series Network Engineering Guidelines

Product Release 1.6 Standard 1.6 June 2005

555-8421-103

Printed in Canada.

All information contained in this document is subject to change without notice. Nortel reserves the

right to make changes to equipment design or program components, as progress in engineering,

manufacturing methods, or other circumstances may warrant.

*Nortel, the Nortel logo, the Globemark, Unified Networks, Meridian 1 PBX, Communication Server

1000 (CS 1000), and Communication Server 2100 (CS 2100) are trademarks of Nortel.

2 Remote Gateway 9100 Series Network Engineering Guidelines

Standard 1.6 June 2005

Remote Gateway 9100 Series Network Engineering Guidelines 3

Standard 1.6 June 2005

Contents

In this document

About this document 4

Introduction to packetized voice 5

The E-model to measure audio quality and user satisfaction 7

Compressed audio quality under ideal conditions 8

Compressed audio quality under packet loss conditions 11

Evaluating your network 13

Bandwidth usage table 26

Quality of service issues 28

Ordering ISDN lines 35

4 Remote Gateway 9100 Series Network Engineering Guidelines

Standard 1.6 June 2005

About this document

The Remote Gateway 9100 Series Network Engineering Guidelines

(NTP 555-8421-103) is written for individuals who are responsible for the

installation, configuration, and day-to-day management of the Remote Gateway

9100 Series units and the Reach Line Card (RLC).

This document describes the network engineering guidelines for the Remote

Gateway 9150 unit, Remote Gateway 911x series unit (9110 and 9115), Digital

Telephone IP Adapter unit (internal and external), and the RLC.

This document provides:

a clearer understanding of how you should plan your network

a detailed description of Nortel's patented QoS Transitioning Technology

instructions on how to order ISDN lines

Terminology

Throughout this document, the term “host PBX” refers to the following Nortel

PBXs:

Meridian 1 PBX

Communication Server 1000 (CS 1000)

Communication Server 2100 (CS 2100)

Unless otherwise specified, the term “Remote Gateway 9100 Series units” refers

to the following products:

Remote Gateway 9110 unit

Remote Gateway 9115 unit

Remote Gateway 9150 unit

Digital Telephone IP Adapter unit (Internal and External)

The product name “Remote Gateway 9100 Series” refers to all of the “Remote

Gateway 9100 Series units” as well as the RLC.

Remote Gateway 9100 Series Network Engineering Guidelines 5

Standard 1.6 June 2005

Introduction to packetized voice

The Public Switched Telephone Network (PSTN) has evolved into a stable, high

quality communications medium. When you take a telephone off-hook, dial tone

is immediate. Connection time is quite fast and clarity of the audio is excellent.

With the development of packetized communications such as Voice over

Internet Protocol (VoIP), the same high quality experience can be realized -

when engineered properly.

To correctly engineer a VoIP system, you must use subjective testing results

performed by numerous companies and independent labs to understand the

factors that affect the quality of the VoIP. The main factors are categorized as

follows:

audio levels

The PSTN has been built around clear rules of Loss Level Planning – the

level of transmitted and received audio. The audio levels are such that the

conversation is comfortable and highly intelligible. You can converse

without having to strain to hear the caller.

extra audio (echo, clicks, pops)

Extra audio can be created in many ways. The most common extra audio is

echo, or a reflection of your own speech. In traditional PSTN telephony,

echo is present, but the Loss Level Planning reduces the amplitude, and the

small amount of echo delay makes it unperceivable. With VoIP systems, the

inherent audio processing and network delays can make the echo

perceivable, if the echo is not cancelled/managed properly.

Clicks and pops can be heard when packet loss is occurring on the network.

The severity of the click or pop is determined by the size and duration of

the packet loss experienced, as well as the audio compression and

decompression scheme (Codec) chosen. A Codec compresses audio,

reducing the amount of bandwidth required to have a conversation.

— G.711 (64 Kbps data rate - equivalent to a wired telephone [no

compression])

— G.726 (32 Kbps data rate - reduces the required bandwidth by one half

in trade for a slightly lower audio fidelity)

— G.729A (8 Kbps data rate - high level of compression in trade for lower

audio fidelity)

6 Remote Gateway 9100 Series Network Engineering Guidelines

Standard 1.6 June 2005

missing audio (drop-outs, reduced fidelity)

Missing audio is often a result of packet loss on the network. Reduced

audio fidelity is a subjective factor that varies greatly by system user. The

G.711 and G.726 Codecs provides wire line quality when the network

conditions are ideal. Packet loss with high compression Codecs such as

G.729A lead to an even lower audio fidelity.

conversational delay

When speaking to a person face to face, your speech audio is transmitted

and received instantaneously. You also have the added benefit of being able

to see the other person, and therefore can use body language, expressions

and gestures to assist in the communication. With telephony, it is the tone,

phrasing and timing of speech that is transmitted. VoIP systems can affect

the fidelity of the audio but the greatest impact can be in the area of

conversational delay. For example, assuming that you are speaking, your

audio must be compressed, packetized, transmitted across a network,

decompressed and transmitted to the final destination. These processes all

take a small measurable amount of time. If the delay is too great, the

conversation can be awkward, with each talker speaking over one another.

trans/dual encoding

Transcoding is a term used when audio is processed through one Codec,

decompressed then passed through another Codec. Each time the audio is

compressed (G.726 or G.729A), the audio fidelity reduces.

With these factors in mind and the industry subjective testing results as a

reference, you can engineer a high quality VoIP system. One of the best

references available to “rate” the quality and factors that impair VoIP systems is

the International Telephony Union G.107 Recommendations (ITU-G.107). This

document and the subsequent G.108 and G.113 documents have been developed

by industry leaders, including Nortel.

Many of the ITU recommendations have been “rolled” into the TIA TSB116

document that clearly outlines the methods of engineering an excellent

packetized voice system.

Remote Gateway 9100 Series Network Engineering Guidelines 7

Standard 1.6 June 2005

The E-model to measure audio quality and

user satisfaction

The ITU developed what is called the E-model. The E-model uses the factors

described in “Introduction to packetized voice” on page 5 to derive a quality

value corresponding to the quality experienced by the VoIP system users.

The following graph shows the E-model rating, R value, versus the traditional

Mean Opinion Score (MOS) values.

8 Remote Gateway 9100 Series Network Engineering Guidelines

Standard 1.6 June 2005

Compressed audio quality under ideal

conditions

The following graph shows the Remote Gateway 9100 Series Codecs graphed

against the R value and conversational delay.

Note: T is one-way delay in milliseconds (ms).

To fully use this graph, you must know the Remote Gateway 9100 Series

inherent delay values. The critical values are as follows:

Codec

Encoding/

Decoding

BRI

Serialization

911x V.32

Serialization

G.711 - 64 Kbps 30 ms 25 ms -

G.726 - 32 Kbps 30 ms 12 ms -

G.729A - 8 Kbps 35 ms 10 ms 90 ms

Remote Gateway 9100 Series Codecs - No Impair ments

100

0 100 200 300 400 500

G.711

G.726

G.729A

Exceptional

limiting case

Very

satisfactory

Satisfactory

Some users

dissatisfied

Many users

dissatisfied

User Satisfaction

Remote Gateway 9100 Series Network Engineering Guidelines 9

Standard 1.6 June 2005

There are additional delays experienced that are a function of the customer

network:

end-to-end delay

The time it takes to pass an Internet Protocol (IP) packet from the RLC to

the Remote Gateway 9150 unit (or the opposite direction)

jitter buffer depth

If the network has low jitter, the jitter buffer can be reduced. The

configurable values in the Remote Gateway 9100 Series product are 30, 60,

or 90 ms.

Conversational delay calculation examples

Remote Gateway 9100 Series units over IP

When using the Remote Gateway 9100 Series units over IP, the controllable

delay factors are the jitter buffer and the network delay. For example, assuming

you are using G.711 with a 60 ms jitter buffer:

30 ms (G.711 encoding/decoding) + 20 ms (one-way network delay) + 60 ms

(jitter buffer) = 110 ms one-way delay

Remote Gateway 9150 unit over PSTN

In the following example, the RLC and Remote Gateway 9150 unit is

configured to use G.729A to reduce bandwidth requirements:

35 ms (G.729A encoding/decoding) + 10 ms (serialization delay) + 60 ms (jitter

buffer) = 105 ms one-way delay

Remote Gateway 911x series unit over PSTN

The Remote Gateway 911x series unit uses a V.32 modem emulation to provide

connectivity over the PSTN. In this configuration, you must use G.729A.

35 ms (G.729A encoding/decoding) + 90 ms (serialization delay) + 60 ms (jitter

buffer) = 185 ms one-way delay

10 Remote Gateway 9100 Series Network Engineering Guidelines

Standard 1.6 June 2005

Remote Gateway 9150 unit to Remote Gateway 9150 unit

When a customer environment uses multiple Remote Gateway 9150 units, the

audio must be treated by each individual Remote Gateway 9150 unit. Therefore,

the delay each Remote Gateway 9150 unit experiences is added together. For

example, assuming one Remote Gateway 9150 unit over IP (G.711)

communicating to another Remote Gateway 9150 unit over BRI (G.729A):

30 ms (G.711 encoding/decoding) + 20 ms (one-way network delay) + 60 ms

(jitter buffer) + 35 ms (G.729A encoding/decoding) + 10 ms (serialization delay)

+ 60 ms (jitter buffer) = 215 ms one-way delay

Notes:

1. Since the audio has been compressed by the G.729A Codec, each user

experiences the quality provided by the G.729A Codec.

2. When calls are made from Remote Gateway 9150 unit to Remote Gateway

9150 unit as described, it is possible that the Echo Cancellers in the Remote

Gateway 9100 Series product can cause periods of audio clipping when the

conversational delay is large. To minimize this possibility, you must

manage network delay and jitter so that the jitter buffer depth can be

reduced as much as possible.

As you have calculated, when using IP connectivity, the network performance

can affect the jitter buffer depth and network delay, thus increasing the

conversational delay. The network can also have periods of high jitter or packet

loss that can greatly affect the audio quality.

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