Nortel 9110 User manual

Remote Gateway 9100 Series Network Engineering Guidelines
Product Release 1.6 Standard 1.6 June 2005
555-8421-103
Copyright © 2000–2005 Nortel. All Rights Reserved.
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
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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
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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.

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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)

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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.

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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.

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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
50
60
70
80
90
100
0 100 200 300 400 500
T
R
G.711
G.726
G.729A
Exceptional
limiting case
Very
satisfactory
Satisfactory
Some users
dissatisfied
Many users
dissatisfied
User Satisfaction

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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

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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.