TR4112 00 05 04 PN4689v4 RB ritt MA rmstrong Nortel

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					                                                                             TR41.1.2/00-0508-0041


                                         DRAFT 4
STANDARDS PROJECT:            PN-4689


TITLE:                        Voice Quality Recommendations for IP Telephony


SOURCE:                       Nortel Networks
                              P.O. Box 3511 Station C
                              Ottawa, ON. K1Y 4H7
                              Canada


CONTACT:                      Roger Britt
                              Phone: (613) 763-4820
                              Fax:      (613) 763-2096
                              Internet: rbritt@nortelnetworks.com
                              Mark Armstrong
                              Phone: (613) 763-4409
                              Fax:      (613) 763-2096
                              Internet: markarm@nortelnetworks.com


DATE:                         16 May, 2000


DISTRIBUTION TO:              TIA TR-41.1.2



The contributor grants a free, irrevocable license to the Telecommunications Industry Association
(TIA) to incorporate text contained in this contribution and any modifications thereof in the creation
of a TIA standards publication; to copyright in TIA's name any standards publication even though it
may include portions of this contribution; and at TIA's sole discretion to permit others to reproduce
in whole or in part the resulting TIA standards publication.
                    PN-4689 V4.0
       (to be published as TIA/EIA/TSB-116)



                  Telecommunications

               IP Telephony Equipment

Voice Quality Recommendations for IP Telephony




 Formulated under the cognizance of TIA Subcommittee TR-41.1,
                  Multiline Terminal Systems

    With the approval of TIA Engineering Committee TR-41,
       User Premises Telecommunications Requirements
                                                                              PN-4689V4 (to be published as TIA/EIA/TSB-116)

                                                         TABLE OF CONTENTS

1.          INTRODUCTION ..................................................................................................................... 3
2.          REFERENCES .......................................................................................................................... 4
3.          DEFINITIONS, ABBREVIATIONS AND ACRONYMS ..................................................... 6
     3.1.         CODEC ................................................................................................................................. 6
     3.2.         ABBREVIATIONS AND ACRONYMS .................................................................................. 6
4.          THE E-MODEL ........................................................................................................................ 7
     4.1.      TRANSMISSION RATING FACTOR “R”.............................................................................. 8
     4.2.      IP TELEPHONY IMPAIRMENTS AND THE E-MODEL ....................................................... 9
          4.2.1. Delay ............................................................................................................................. 9
          4.2.2. Echo ............................................................................................................................ 10
          4.2.3. Speech Compression ................................................................................................... 13
          4.2.4. Packet Loss ................................................................................................................. 16
     4.3.      WHAT DOES R SOUND LIKE? .......................................................................................... 17
     4.4.      E-MODEL SYMMETRY AND PERFORMANCE ................................................................. 18
     4.5.      E-MODEL ENHANCEMENTS ............................................................................................ 19
     4.6.      E-MODEL CONVENTIONS ................................................................................................ 19
     4.7.      SUMMARY ......................................................................................................................... 19
5.          WIRELINE PSTN VOICE QUALITY BENCHMARKS ................................................... 20
     5.1.         ISDN VOICE QUALITY ..................................................................................................... 20
     5.2.         PSTN VOICE QUALITY..................................................................................................... 22
     5.3.         TOLL VOICE QUALITY ..................................................................................................... 25
     5.4.         WIRELINE PSTN VOICE QUALITY SUMMARY .............................................................. 27
6.          IP TELEPHONY VOICE QUALITY ANALYSIS .............................................................. 28
     6.1.        VOICE QUALITY ISSUES FOR IP TELEPHONY ............................................................... 30
     6.2.        VOICE QUALITY RECOMMENDATIONS FOR IP TELEPHONY....................................... 30
            6.2.1. Delay ........................................................................................................................... 30
            6.2.2. G.711 Packet Loss Concealment (PLC) ..................................................................... 37
            6.2.3. Low Bit-rate Coders .................................................................................................... 38
            6.2.4. Packet Loss ................................................................................................................. 38
            6.2.5. Transcoding................................................................................................................. 40
            6.2.6. Echo Cancellers (ECAN) ............................................................................................ 41
            6.2.7. TCLw .......................................................................................................................... 41
            6.2.8. New Loss Plan .............................................................................................................. 1




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                                           FOREWORD


                             (This foreword is not part of this standard.)

This document is a TIA/EIA Telecommunications Technical Services Bulletin (TSB) produced by
Working Group TR-41.1.2 of Committee TR-41. This TSB was developed in accordance with
TIA/EIA procedural guidelines, and represents the consensus position of the Working Group and its
parent subcommittee TR-41.1, which served as the formulating group.


The TR-41.1.2 VoIP Voice Quality Working Group acknowledges the contribution made by the
following individuals in the development of this standard.

                         Name                    Representing
               Roger Britt                 Nortel Networks              Chair/Editor
               Mark Armstrong              Nortel Networks
               Dermot Kavanagh             Nortel Networks
               Peter Melton                Cortelco Systems Inc.
               Kirit Patel                 Cisco Systems



Copyrighted parts of ITU-T Appendix I to Recommendation G.113 and Recommendation ???? are
used with permission of the ITU. The ITU owns the copyright for the ITU Recommendations.

The ??? annexes in this Standard are informative and are not considered part of this Standard.

Suggestions for improvement of this standard are welcome. They should be sent to:

                             Telecommunications Industry Association
                                    Engineering Department
                                           Suite 300
                                     250 Wilson Boulevard
                                     Arlington, VA 22201




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                                                 PN-4689V4 (to be published as TIA/EIA/TSB-116)

1. Introduction
The objective of this TSB is to provide end-to-end voice quality guidelines for North American IP
Telephony. IP Telephony introduces many impairments, some of which are familiar and some are
new. The E-Model (ITU-T Recommendation G.107) is a tool that can estimate the end-to-end voice
quality, taking the IP Telephony parameters and impairments into account. First, this TSB describes
how the E-Model handles IP Telephony impairments and then it reinforces the points by detailing
with specific E-Model scenarios, rules for good voice quality.

This TSB builds on similar work done for North American PBX private networks that was published
in TIA/EIA/TSB32-A.




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2. References
The following documents contain provisions that are referenced in this TSB. At the time of
publication, the editions indicated were valid. All standards are subject to revision, and parties to
agreements based on this Standard are encouraged to investigate the possibility of applying the most
recent editions of the standards indicated below. ANSI and TIA maintain registers of currently valid
national standards published by them.

[1]     TIA/EIA/TSB32-A-1998, Overall Transmission Plan Aspects for Telephony in a Private
        Network.

[2]     TIA/EIA-810 (12/99), Telecommunications – Telephone Terminal Equipment – Transmission
        Requirements for Narrowband Voice over IP and Voice over PCM Digital Wireline
        Telephones.

[3]     ANSI T1.521, American National Standard for Packet Loss Concealment with ITU-T
        Recommendation G.711.

[4]     ITU-T Recommendation G.107 (12/98), The E-Model, A Computational Model for use in
        Transmission Planning.

[5]     ITU-T Recommendation G.108 (2000), Conversational impacts on end-to-end speech
        transmission quality – a planning guide on effects not covered by the E-Model.

[6]     ITU-T Recommendation G.109 (1999), Definition of categories of speech transmission
        quality.

[7]     ITU-T Recommendation G.113 (02/96), Transmission impairments.

[8]     ITU-T Appendix I to Recommendation G.113 (1998), Transmission impairments – Appendix
        I: Provisional planning values for the equipment impairment factor Ie.

[9]     ITU-T Recommendation G.114 (02/96), One-way transmission time.

[10]    CCITT Recommendation G.131 (08/96), Control of talker echo.

[11]    ITU-T Recommendation G.175 (04/97), Transmission planning for private/public network
        interconnection of voice traffic.

[12]    ITU-T Recommendation G.177 (2000), Transmission planning for voiceband services over
        hybrid Internet/PSTN connections.

[13]    CCITT Recommendation G.711 (11/88), Pulse code Modulation (PCM) of voice frequencies.

[14]    ITU-T Recommendation G.712 (11/96), Transmission performance characteristics of pulse
        code modulation.

[15]    ITU-T Recommendation G.723.1 (03/96), Dual rate speech coder for multimedia
        communications transmitting at 5.3 and 6.3 kbit/s.

[16]    ITU-T Recommendation G.729 (03/96), Coding of speech at 8 kbit/s using conjugate-
        structure algebraic-code-excited linear-prediction (CS-ACELP).



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                                              PN-4689V4 (to be published as TIA/EIA/TSB-116)

[17]   ITU-T Recommendation P.861 (02/98), Objective quality measurement of telephone-band
       (300 -3400 Hz) speech codecs.

[18]   ETSI EG 201 377-1 (1999), Specification and measurement of speech transmission quality;
       Part 1: Introduction to objective comparison measurement methods for one-way speech
       quality across networks.




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3. Definitions, Abbreviations and Acronyms
For the purposes of this TSB, the following definitions apply.

    3.1. Codec
A codec is a combination of an analog-to-digital encoder and a digital-to-analog decoder operating in
opposite directions of transmission in the same equipment.

    3.2. Abbreviations and Acronyms
Abbreviations and acronyms, other than in common usage, which appear in this standard, are defined
below.

ERL     Echo Return Loss
ERLE    Echo Return Loss Enhancement
GoB     Good or Better
Ie      Equipment Impairment factor
IP      Internet Protocol
ISDN    Integrated Services Digital Network
MOS     Mean Opinion Score
NLP     Nonlinear Processor
OLR     Objective Loudness Rating
PBX     Private Branch Exchange
PCM     Pulse Code Modulation
PL      Packet Loss
PLC     Packet Loss Concealment
PoW     Poor or Worse
PSTN    Public Switched Telephone Network
QoS     Quality of Service
RLR     Receive Loudness Rating
RTP     Recommended Test Position
SLR     Send Loudness Rating
STMR    Sidetone Masking Rating
TCLw    Weighted Terminal Coupling Loss
TDM     Time Division Multiplexing
TELR    Talker Echo Loudness Rating
VAD     Voice Activity Detector
VoIP    Voice over Internet Protocol




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                                                  PN-4689V4 (to be published as TIA/EIA/TSB-116)

4. The E-Model
The objectives for this section are to:
 demonstrate the suitability of the E-Model for estimating the voice quality of IP (Internet
   Protocol) Telephony
 explain the R-scale used by the E-Model
 explain what some of the IP Telephony impairments sound like.

The E-Model is a transmission planning tool for estimating the relative user satisfaction of a
narrowband, handset conversation, as perceived by the listener. It is not intended for predicting
absolute user satisfaction. The E-Model has proven to be versatile tool that has adapted well to the
impairments of IP telephony. This document assumes that the reader is familiar with the E-Model
and the basics of following standards.

   ITU-T Recommendation G.107 (the E-Model, including a program listing)
   ITU-T Recommendation G.108 (a tutorial on the E-Model and network planning)
   ITU-T Recommendation G.109 (defines categories of speech transmission quality)
   ITU-T Recommendation G.113 (details transmission impairments)
   ITU-T Appendix I to Rec. G.113 (table of the equipment impairment factor, Ie, values)
   ITU-T Recommendation G.114 (details delay, including the expected delay for IP coders)
   ITU-T Recommendation G.131 (details talker echo)
   ITU-T Recommendation G.177 (provides guidelines for mixed IP/PSTN connections)
   TIA/EIA/TSB32-A and ETSI Guide 201 050 (a tutorial on the E-Model and a transmission
    planning guide for private networks).




               Figure 1 – Comparison of E-Model Output Scales and Categories

                    R          USER SATISFACTION                  MOS      %GOB %POW
    G.107          100
    Default         94             Very Satisfied                  4.4        98.4       0.1
     Value          90                                             4.3        97.0       0.2
                                       Satisfied
                    80                                             4.0        89.5       1.4
                            Some Users Dissatisfied
                    70                                             3.6        73.6       5.9
                             Many Users Dissatisfied
                    60                                             3.1        50.1      17.4
                          Nearly All Users Dissatisfied
                    50                                             2.6        26.6      37.7
                                Not Recommended
                     0                                             1.0          0       99.8




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    4.1. Transmission Rating Factor “R”
The output of the E-Model is a scalar called the “Rating Factor”, the “R-value”, or simply “R”. The
scale is typically from 50 to 100, where everything below 50 is clearly unacceptable and everything
above 94.15 (the maximum with the G.107 E-Model, version 19 default values) is unobtainable in
narrowband (300 to 3400 Hz) telephony. The scale on the left-hand side of Figure 1 illustrates this
point. The center scale labeled “User Satisfaction” shows the categories defined in G.109. This gives
an indication of the quality of the conversation.

It is important to note the distinction between E-Model objective results and the results of subjective
studies that are expressed using the MOS (Mean Opinion Score), %GoB (percent Good or Better) or
%PoW (percent Poor or Worse) scales. In subjective testing, subjects are requested to classify the
perceived quality into categories (for example, a five point scale that includes the classifications
excellent, good, fair, poor and bad). In each subjective experiment, the MOS scores may differ, even
for the same conditions, depending on the design of the experiment, the set of subjects, etc. E-Model
results, however, are calculated using the Impairment Factor method in which impairment values
along the speech path (such as loss, distortion, echo, delay, noise, etc.) are combined to obtain an
overall transmission rating “R”, which is objective and repeatable. While the R-value can be
deterministically converted into MOS, %GoB or %PoW scores, it is preferable to avoid confusion
and use only the R scale for all E-Model work. For reference, the MOS, %GoB and %PoW scales are
shown on the right-hand side of Figure 1.

The E-Model consists of several models, which relate specific impairment parameters to end-to-end
performance, as well as their interaction. The total end-to-end performance, taking into account all
factors, is estimated using the Impairment Factor method on the principle that transmission
impairments can be transformed into “Psychological Factors” and these factors are additive on the
“Psychological Scale”.

The equation for the transmission rating factor R is:

                                       R = Ro - Is - Id - Ie +A
Where,
 Ro, the basic signal-to-noise ratio based on send and receive loudness ratings and the circuit and
  room noise;
 Is, the sum of real-time or simultaneous speech transmission impairments, e.g., loudness levels,
  sidetone and PCM quantising distortion;
 Id, the sum of delayed impairments relative to the speech signal, e.g., talker echo, listener echo
  and absolute delay;
 Ie, the Equipment Impairment factor for special equipment, e.g., low bit-rate coding (determined
  subjectively for each codec and for each % packet loss and documented in Appendix I to G.113);
 A, the Advantage factor adds to the total and improves the R-value for new services, like satellite
  phones, to take into account the advantage of using a new service and to reflect acceptance of
  lower quality by users for such services. It is assumed that the Advantage Factor will be reduced
  over time as the service improves and the customers get use to the benefits of the new service. It
  is not recommended to use the Advantage Factor for IP telephony because it is a replacement for
  existing services, rather than a completely new service.

The Equipment Impairment factor and the Advantage factor are unique to the E-Model, but it is the
Equipment Impairment factor that makes the E-Model a powerful tool for estimating the relative user
satisfaction of IP Telephony conversations.




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                                                    PN-4689V4 (to be published as TIA/EIA/TSB-116)

    4.2. IP Telephony Impairments and the E-Model
The four main impairments of IP telephony are:
 delay, including delay variation and jitter
 echo
 speech compression
 packet loss.
The ability to handle these impairments is one of the strengths of the E-Model.


                      Figure 2 – Delay Impairment of Reference Connection


                                                                User Satisfaction
       100
                                                                        Very
                                                                    satisfactory
       90

                                                                    Satisfactory

       80
                                                                    Some users
   R                                                                                      TELR = 65 dB
                                                                    dissatisfied
       70
                                                                    Many users
                                                                    dissatisfied

       60
                                                                     Exceptional
                                                                    limiting case
       50
             0        100           200          300          400                   500

                                   One-way Delay (ms)




        4.2.1. Delay
The curve in Figure 2 plots the transmission rating factor R vs delay for the reference connection
shown in Figure 3. The right-hand side of Figure 2 includes the “User Satisfaction” scale for
reference.

Using delay as the dependent variable on the x-axis, gives a clear picture of how important delay is in
IP telephony. The reference connection curve uses the G.107 default values for all parameters, except
the variable delay. This gives the best possible performance for a narrowband handset conversation,
over this range of delay, and therefore will be used as the “relative reference” throughout this
document. The connection consists of two ideal digital telephones with G.711 codecs and some
means to vary network delay from 0 to 500 ms, as shown in Figure 3. The parametric variable, TELR
(talker echo loudness rating), shown in the Figure 2 legend, is explained in the next section.

Notice that there is a knee on the curve at about 175 ms. The region between 150 and 200 ms is
where the delay starts to affect the dynamics of a conversation. The steeper slope of the top curve
after 175 ms clearly says the dynamics of normal conversation degrades as delay increases. Why? In
a normal face-to-face conversation, after one person speaks, there is about a 200 ms break, then the
other person speaks, followed by another 200 ms break and so on. This is called turn taking. When an
extra 150 ms or more is added in each direction, then the normal turn taking rules fail and the

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PN-4689V4 (to be published as TIA/EIA/TSB-116)

conversation rhythm has to change. Often when there is added delay, one person will start talking
before the other person is finished or both people will start talking simultaneously, which causes the
conversation stop and restart. If one person dominates the conversation, then the other person will
have trouble breaking in, because the dominant talker has already started in again before the break
reaches the other person. There are other effects as well. For instance, the extra delay can also
change the message. Suppose someone asks the question, “Will you marry me?” and the answer,
“Yes.”, is delayed by some noticeable amount. The delay may be interpreted as a hesitation to reply
rather the normal operation of codecs, jitter buffers and propagation delay. The medium can distort
the message.

         Figure 3 – Block Diagram and E-Model Parameters for Reference Connection
                                                                    Side A       0 dBr     Side B
                                                                     Digital                Digital
                                                                   Telephone              Telephone




                                              Echo Path - Side A

                                              Echo Path - Side B

                                                      Abbrev.      Digital                Digital
                            Title
                                                     (Default)      Set         0 dBr      Set
            Electric Circuit Noise (at 0 dBr)     Nc (-70)                       -70
            Room Noise                            Po (35)             35                    35
            Send Loudness Rating                  SLR      (8)        8                     8
            Receive Loudness Rating               RLR      (2)        2                     2
            D-factor                              D      (3)          3                     3
            Noise Floor                           Nfor (-64)         -64                   -64
            Sidetone Masking Rating               STMR (15)           15                    15
            Equipment Impairment Factor           Ie      (0)         0                     0
            Expectation (Advantage) Factor        A       (0)         0                     0
            Mean One-Way Delay (upper)            Tu       (0)        0                     0
            Mean One-Way Delay (lower)            Tl       (0)        0                     0
            Mean One-Way Delay (upper = lower)    Tul      (0)        0        0 to 500     0
            Electrical Loss (upper)               Lu
            Electrical Loss (lower)               Ll
            Electrical Loss (upper = lower)       Lul
            Quantizing Distortion Units (upper)   qduu      (1)      0                      0
            Quantizing Distortion Units (lower)   qdul      (1)      0                      0
            Echo Return Loss                      ERL                55                     55



        4.2.2. Echo
The family of curves in Figure 4 shows the effect of echo as predicted by the E-Model for the
connection shown in Figure 7. To fully understand the meaning of the graph it is necessary to take a
step back and explain the parametric variable TELR. First, the definition (based on Figures 5 and 6
Echo Path B):

          TELR (Side B) = SLR (Side B) + Loss in bottom path + ELR or TCLw (Side A)
                              + Loss in top path + RLR (Side B)

The TELR for Side A is similar, but follows the opposite path, as highlighted by the arrows below
the block diagrams. Figure 5 shows a connection with two 2-to-4 wire conversions and an analog-to-

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                                                    PN-4689V4 (to be published as TIA/EIA/TSB-116)

digital and digital-to-analog conversion. The 2-to-4 wire conversion is called a hybrid and the amount
of echo that gets through the hybrid is called the transhybrid loss, the echo return loss or more simply
the ERL. Figure 6 shows an all digital connection. In this case the echo return loss is called weighted
terminal coupling loss or TCLw. TCLw is leakage in the analog portion of the digital set, i.e., the
analog circuits, capacitive coupling in the handset cord, mechanical coupling from the receiver to the
transmitter in the handset or acoustical coupling from the receiver to the transmitter in the handset.

TELR is the sum of the losses around the loop, from one set’s transmitter back to itself. The SLR
(send loudness rating) and RLR (receive loudness rating) are the values for the same telephone. In
Figure 5, loss pads in the upper and lower paths control the loss plan. These pads may add gain,
which increases the echo, they may add loss, which reduces the echo or they may be neutral (0 dB)
and have no affect on the echo. The loss plan for an all digital connection is determined by the
loudness ratings of the telephones and there are no additional losses in the network to allow “clear
channel” transmission. The loss plan for an analog or mixed analog/digital connection is a fine
balance between providing enough loss to attenuate the echo and maintain circuit stability, while still
being audible over a range of analog loops.

Back to Figure 4. Note that as TELR is reduced, the amount of end-to-end delay available to the
connection is reduced for a given performance quality objective on the R scale. The nominal
loudness ratings for digital telephones are SLR = 8 dB and RLR = 2 dB. TELR has to be about 65 dB
to completely remove echo, so TCLw has to be:

                         TCLw = TELR – SLR – RLR = 65 – 8 – 2 = 55 dB.

The ERL standard for echo cancellers (ECANs), ITU-T Recommendation G.168, specifies ERL >=
55 dB of echo path loss for ECANs in gateways, but the standard for digital sets, TIA-810, only
specifies TCLw >= 45 dB. The curve for TELR = 55 dB (45 + 8 +2) shows that this is a good
requirement for low delay connections like local ISDN and digital proprietary PBX telephones, but it
is not adequate for IP telephones. Clearly, IP telephones need to have TCLw >= 55 dB for minimum
echo return, just like ECANs, because they work in a long delay environment.

Figure 4 is useful for understanding the implications of double talk on the performance of ECANs. In
single talk mode, i.e., when one person is talking and the other is silent, the convolution processor
part of the echo canceller provides about 18 dB of echo return loss enhancement (ERLE), in addition
to the typical analog telephone ERL of about 12 dB, and the nonlinear processor (NLP) provides an
additional loss of at least 25 dB, for a total of 55 dB. When both people start talking at once, the NLP
drops out leaving the connection with only 30 dB of loss, ignoring the possibility of multiple echoes.
So in double talk mode, the echo performance drops from the TELR = 65 dB curve to the TELR = 40
dB curve, with a significant drop in R. This also happens with some older ECANS when a
conversation starts up and the ECAN hasn’t had a chance to converge; the ECAN is said to “leak”.
Since there is no conclusive evidence that anyone is listening when both people are talking, the point
may well be moot.




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                                 Figure 4– E-Model Prediction of Echo Impairment




                                        Figure 5– Echo Path for Analog Connection
                 Side A                      Digital PBX                             Digital PBX                              Side B
                Analog                       CODEC                                            CODEC                           Analog
               Telephone       Hybrid                                                                              Hybrid    Telephone
                                                          U dB                     U dB
                                              A                                                    A
 SLR = 11 dB                                                                                                                              RLR = 3 dB
                                   ERL =          D                   Digital                  D             ERL =
                                   14 dB                              PSTN                     D             14 dB
                                                  D                                                                                       SLR = 11 dB
  RLR = 3 dB
                                              A                                                    A
                                                          L dB                     L dB


               Echo Path A



                                                                                                                            Echo Path B



                                        Figure 6– Echo Path for Digital Connection
                               Side A       Digital PBX                         Digital PBX             Side B
                                Digital                                                                  Digital
                              Telephone       U dB                                U dB                 Telephone

               SLR = 8 dB                                                                                            RLR = 2 dB
                                                                 Digital
                                                                 PSTN
                RLR = 2 dB                                                                                           SLR = 8 dB
                             TCLw = 45 dB      L dB                                L dB            TCLw = 45 dB


                             Echo Path A



                                                                                                   Echo Path B




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            Figure 7– Block Diagram and E-Model Parameters for Echo Impairment
                                                                  Side A      0 dBr                  Side B
                                                                      IP                                IP
                                                                  Telephone                         Telephone



                                                                                      IP Intranet




                                             Echo Path - Side A

                                             Echo Path - Side B

                                                     Abbrev.
                           Title
                                                    (Default)      IP Set     0 dBr   IP Intranet   IP Set
           Electric Circuit Noise (at 0 dBr)     Nc (-70)                      -70
           Room Noise                            Po     (35)         35                               35
           Send Loudness Rating                  SLR      (8)        8                                 8
           Receive Loudness Rating               RLR       (2)       2                                 2
           D-factor                              D        (3)        3                                 3
           Noise Floor                           Nfor (-64)         -64                               -64
           Sidetone Masking Rating               STMR (15)           15                               15
           Equipment Impairment Factor           Ie       (0)        0                                 0
           Expectation (Advantage) Factor        A       (0)         0                                 0
           Mean One-Way Delay (upper)            Tu        (0)       0                                 0
           Mean One-Way Delay (lower)            Tl       (0)        0                                 0
           Mean One-Way Delay (upper = lower)    Tul      (0)        0                 0 to 500        0
           Electrical Loss (upper)               Lu       (0)
           Electrical Loss (lower)               Ll       (0)
           Electrical Loss (upper = lower)       Lul      (0)
           Quantizing Distortion Units (upper)   qduu       (1)     0.5                               0.5
           Quantizing Distortion Units (lower)   qdul      (1)      0.5                               0.5
           Echo Return Loss                      ERL              35 to 55                          35 to 55


Figure 7 shows the block diagram and E-Model parameters for echo impairment. It is the same as
Figure 3, except the parametric variable TELR ranges from 45 to 65 dB in 5 dB steps. Actually, it is
the TCLw parameter that ranges from 35 to 55 dB in 5 dB steps, but it is called ELR in the E-Model,
as shown on the bottom line of Figure 7.

        4.2.3. Speech Compression
A unique feature of the E-Model is its flexibility to deal with the impairments introduced by speech
compression and packet loss via the Equipment Impairment Factor (Ie). The provisional Equipment
Impairment Factors for several low bit rate codecs are listed in Appendix I of Rec. G.113. The Ie
values for the popular IP codecs, G.711, G.729 and G.723.1, are listed in Table 1 (in the 0% packet
loss row). Before using these Ie values in any E-Model calculations, the reader should refer to the
latest revision of Appendix I of Rec. G.113.

The Ie values in Table 1 were determined in subjective experiments with ideal software
implementations of the codecs; the performance provided by commercial codecs may vary.

As detailed in section 4.1, the Ie value is subtracted from the R-value, in effect lowering the listener
satisfaction rating. Figure 8 illustrates the point by comparing the best-case curves for three IP
codecs. Notice that codecs with larger Ie values can tolerate less one-way delay.

Figure 9 shows the block diagram with Ie as the parametric variable and delay as the dependent
variable. The jitter buffers and packetization modules are shown in the gateways without any delay
allotment. Instead all the delay is shown at the 0 dBr point. This is done only for convenience. When
modeling a real connection, the proper gateway delay would be entered in the gateway columns. The
G.711 curve is the same as the TELR = 65 dB reference curve in the previous sections. Notice that
the ECANS ERL of 55 dB includes set’s TCLw of 45 dB, as explained in the previous section. Also,
this E-Model tool ignores ERL values in outside columns, if there are ERL values in inside columns.

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    Table 1– Provisional Planning Values for the Equipment Impairment Factor Ie under
      Conditions of Packet Loss for Codecs G.711, G.729A + VAD and G.732.1 + VAD
Packet Loss        G.711         G.711 with      G.711 with    G.729A +       G.723.1 +
    %           without PLC     PLC Random       PLC Bursty      VAD            VAD
                                 Packet Loss     Packet Loss                  6.3 kbit/s
      0               0               0               0             11            15
      0.5             –               –               –             11            15
      1              25               5               5             15            19
      1.5             –               –               –             17            22
      2              35               7               7             19            24
      3              45              10              10             23            27
      4               –               –               –             26            32
      5              55              15              30              –             –
      7               –              20              35              –             –
      8               –               –               –             36            41
     10               –              25              40              –             –
     15               –              35              45              –             –
     16               –               –               –             49            55
     20               –              45              50              –             –

                          Figure 8– Speech Compression Impairment




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    Figure 9– Block Diagram and E-Model Parameters for Speech Compression Impairment
                                                         Side A           Gateway           0 dBr                    Gateway            Side B
                                                          Digital                                                                        Digital
                                                        Telephone                                                                      Telephone
                                                                          G.711                                            G.7xx
                                                                                                                     JB

                                                                                                      IP Intranet
                                                                                   JB
                                                                           G.7xx                                               G.711




                                   Echo Path - Side A

                                   Echo Path - Side B

                                           Abbrev.      Digital                                                                        Digital
                Title
                                          (Default)      Set         IP Gateway            0 dBr     IP Intranet    IP Gateway          Set
Electric Circuit Noise (at 0 dBr)      Nc (-70)                                             -70
Room Noise                             Po (35)             35                                                                            35
Send Loudness Rating                   SLR      (8)         8                                                                             8
Receive Loudness Rating                RLR      (2)         2                                                                             2
D-factor                               D      (3)           3                                                                             3
Noise Floor                            Nfor (-64)         -64                                                                           -64
Sidetone Masking Rating                STMR (15)           15                                                                            15
Equipment Impairment Factor            Ie      (0)          0            0, 11, 15                                   0, 11, 15            0
Expectation (Advantage) Factor         A       (0)          0                                                                             0
Mean One-Way Delay (upper)             Tu       (0)         0                                                                             0
Mean One-Way Delay (lower)             Tl       (0)         0                                                                             0
Mean One-Way Delay (upper = lower)     Tul      (0)         0               0             0 to 500       0                0               0
Electrical Loss (upper)                Lu                                   0                                             0
Electrical Loss (lower)                Ll                                   0                                             0
Electrical Loss (upper = lower)        Lul                                  0                                             0
Quantizing Distortion Units (upper)    qduu      (1)      0.5               0                                             0             0.5
Quantizing Distortion Units (lower)    qdul      (1)      0.5               0                                             0             0.5
Echo Return Loss                       ERL                45                55                                            55            45



    Figure 10 – Provisional Planning Values for the Equipment Impairment Factor Ie under
        Conditions of Packet Loss for Codecs G.711, G.729A + VAD and G.732.1 + VAD

                60


                50

                40


            Ie 30


                20

                10


                  0
                        0                  5                        10                    15                 20                   25
                                                                    Packet Loss %
                            G.711 without PLC                                           G.711 with PLC Random Packet Loss
                            G.711 with PLC Bursty Packet Loss                           G.729A + VAD
                            G.723.1 + VAD



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        4.2.4. Packet Loss
The provisional Equipment Impairment Factors for three IP low bit rate codecs under conditions of
packet loss are listed in Table 1. The G.711 codec is has three columns, one without Packet Loss
Concealment (PLC) and two with PLC. The two with PLC are further subdivided into random and
bursty packet loss conditions. As a point of interest, PLC algorithms are specified in Annexes A and
B of ANSI T1.521.

The most significant cell in Table 1 is the Ie-value of 25 at 1% packet loss for the G.711 without PLC
condition. Clearly, G.711 should not be used without PLC in IP Telephony. More visually
compelling is Figure 10, which is a plot of the Ie values in Table 1.

Another way to describe packet loss visually is to plot the family of curves for a given codec. Figure
11 does so for the G.711 with PLC and under the conditions of random packet loss. Think of the
performance plotted in Figure 11 as the best-case performance, with the loss plan, echo control and
all other parameters at the ideal values. Comparing Figure 11 to Figure 8, G.711 with PLC at 3 %
packet loss has about the same impairment as G.729 at 0% packet loss. Section 6 will provide further
details, but it might be clear by now that for long delay applications like IP telephony, it is preferable
to use G.711 with PLC than to use to codecs with speech compression.

Figure 12 shows the block diagram with Ie as the parametric variable and with all the delay again
concentrated in the 0 dBr column, rather than distributed appropriately.


      Figure 11 – G.711 with PLC under Conditions of Random Packet Loss Impairment




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         Figure 12 – Block Diagram and E-Model Parameters for Packet Loss Impairment
                                                        Side A       Gateway          0 dBr                    Gateway            Side B
                                                         Digital                                                                   Digital
                                                       Telephone                                                                 Telephone
                                                                     G.711                                           G.7xx
                                                                                                               JB

                                                                                                IP Intranet
                                                                               JB
                                                                      G.7xx                                              G.711




                                  Echo Path - Side A

                                  Echo Path - Side B

                                          Abbrev.      Digital                                                                   Digital
                Title
                                         (Default)      Set        IP Gateway        0 dBr     IP Intranet    IP Gateway          Set
Electric Circuit Noise (at 0 dBr)     Nc (-70)                                        -70
Room Noise                            Po (35)             35                                                                       35
Send Loudness Rating                  SLR      (8)         8                                                                        8
Receive Loudness Rating               RLR      (2)         2                                                                        2
D-factor                              D      (3)           3                                                                        3
Noise Floor                           Nfor (-64)         -64                                                                      -64
Sidetone Masking Rating               STMR (15)           15                                                                       15
Equipment Impairment Factor           Ie      (0)          0         0 to 45                                    0 to 45             0
Expectation (Advantage) Factor        A       (0)          0                                                                        0
Mean One-Way Delay (upper)            Tu       (0)         0                                                                        0
Mean One-Way Delay (lower)            Tl       (0)         0                                                                        0
Mean One-Way Delay (upper = lower)    Tul      (0)         0           0            0 to 500       0                0               0
Electrical Loss (upper)               Lu                               0                                            0
Electrical Loss (lower)               Ll                               0                                            0
Electrical Loss (upper = lower)       Lul                              0                                            0
Quantizing Distortion Units (upper)   qduu      (1)      0.5           0                                            0             0.5
Quantizing Distortion Units (lower)   qdul      (1)      0.5           0                                            0             0.5
Echo Return Loss                      ERL                45            55                                           55            45




     4.3. What Does R Sound Like?
Now that we are confident that the E-Model is a suitable tool for IP telephony, it is time to consider
what R-values are possible and what do these values sound like? First of all, it is important to
appreciate that all the possible R-values may be reached by multiple combinations of impairments.
Therefore, all R-values have many different sound characteristics. Some are listening characteristics
and some are conversation characteristics. These characteristics were discussed in general terms in
section 4.2 and will be discussed further in section 6. Table 2 is a glossary of many of the sounds that
IP telephony users will experience, along with the potential causes.

The question of what values of R are possible is one of the objectives of this TSB and it is a much
more difficult question than can be answered in this section. It will take most of section 6 to flesh out
the answer. Figures 8 and 11 give some hints about the R-axis works, as each codec has a family of
curves that simply shifts the references curve down to lower starting points on the R-axis.

The delay axis is a bit more complicated. Delay in can be partitioned as: speech coding/packetization
+ jitter compensation + network routing + propagation. Section 6 will provide the necessary delay
details, but for now it is sufficient to say that unlike the PSTN, the region below 100 ms is not well
used by IP telephony. The concept of regions will be explored further in section 5 and although
section 6 will calculate specific R-values for scenarios, do not forget that with parameters like delay
variation on the X-axis and packet loss on the Y-axis, IP Telephony is much more a game of
statistical probability than absolute R-values.




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   Table 2 – Descriptions of the Sound Characteristics caused by IP Telephony Impairments
           Description                                               Cause
Convergence echo                         A brief blast of echo at the start of a call, before the ECAN
                                         converges, or during hybrid changes due to conferencing call
                                         setups.
Double talk echo                         The ECAN’s NLP reduces loss when both people talk
                                         simultaneously, leaving only the low ERLE of the convolution
                                         processor, creating significant echo that may end the double
                                         talk or may not even be noticed because they are talking not
                                         listening.
After double talk echo                   Echo caused by double talk, but arriving after double talk is
                                         finished due to network delay (see next)
Conversation protocol issues, like Delay caused by speech coding/packetization + jitter
turn-taking, over-talking, break in compensation + network routing and propagation.
and who’s-in-charge problems        Because of the loss of simultaneity, the parties may perceive
                                    each other as inattentive, insincere, or rude. This will
                                    increase with increased delay, until turn-taking cues break
                                    down completely.
Whirlybird distortion, or waterfall      ACELP codec algorithmes.
effect
Speech Clipping at beginning VAD not switching quickly enough, ECAN not switching
and/or end of phases or words quickly enough or VAD and ECAN interfering with each
                              other.
Background noise          contrast   in No comfort noise generator, or a comfort noise generator that
silence periods                         sends a fixed noise (stationary noise) level rather than sending
                                        the sampled noise (sampling the actual noise).
Background        noise     transition Comfort noise generator switches in too slowly, hang time on
contrast                               VAD too long.
Noise pumping                            Background noise is triggering the VAD, lack of comfort
                                         noise generator or comfort noise level does not match the
                                         actual background noise level.
Dropouts and clipping                    Lack of signal caused by packet loss.
Mechanical voice artifacts               Side effects of the packet loss concealment algorithm,
                                         especially with high loss rates with G.711.
Low level tones                          Created intentionally by decoders during long bursts of packet
                                         loss.


    4.4. E-Model Symmetry and Performance
This document implies symmetry between Side A and Side B that probably does not exist in practice.
For instance, packet loss, delay and loudness ratings may be asymmetrical. Also, there is reason to
believe that the performance predicted by the E-Model is optimistic because the Ie values are based
on ideal implementations of ITU-T codecs. The performance of real codecs may be worse due to
implementation issues. Also, all jitter buffers are not equal. Many jitter buffers simply discard
packets under overload conditions. Smart jitter buffers wait for silence periods before discarding
packets. The difference is audible. Delay variation over time may also be audible in some cases.


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    4.5. E-Model Enhancements
The E-Model has gained worldwide acceptance because it is based on several existing transmission
quality models and the results of many subjective experiments conducted over a period of 50 years.
One of the E-Model’s limitations is that it is only a narrowband handset tool. Obvious enhancements
are to add headset, handsfree and wideband functionality. Further work under Q. 20/12 in ITU-T
Study Group 12 is planning to include headset and handsfree operation, but currently there is no
support for developing a planning model for wideband audio.

Work has started in ITU-T Study Group 12 on a new Recommendation (P.DIES) to provide a
detailed methodology for determining equipment impairment factors for use in G.107. The existing
methodology involves correlating the results from subjective experiments with a one number
summary called the Equipment Impairment Factor, Ie. The methodology is not well documented and
could be improved. Some believe that the existing methodology may not be adequate and may be
replaced by a new objective methodology using the successor to Recommendation P.861 (02/98) -
Test vectors for implementations of Perceptual Speech Quality Measure (PSQM).

ITU-T COM 12-70 proposes a change to the supplementary amount of equivalent circuit noise Nos in
the E-Model to include the Lombard effect. The Lombard effect involves the behavior of a talker in a
noisy environment to raise his voice only half the amount necessary to maintain the same signal-to-
noise ratio as in a quiet environment. Acknowledging this behavior requires a change to a constant in
the Nos equation from 0.008 to 0.004. This change was determined in the September 1999 ITU-T
Study Group 12 meeting and the change will be made officially in ITU-T Recommendation G.107
(05/2000).

Also changing in ITU-T Recommendation G.107 (05/2000) is the relationship between the R-value
and the number of QDUs. In G.107 (12/1998), R was constant for values of QDU below about 5, but
it is being removed in G.107 (05/2000). The slope remains the same, but it starts at QDU = 1 instead
of about QDU = 5. This information is explained in G.113 (2000).

Europe has traditionally preferred about 6 dB louder sidetone than North America. Europe has
favored a nominal STMR of about 12 dB and North America has favored a quieter nominal STMR of
about 18 dB. Currently, the E-Model incorrectly penalizes STMRs quieter than 15 dB. TIA-810
specifies STMR = 18 dB +/- 6 dB. An effort will be made to change the 15 dB threshold to 21 dB in
a future revision of G.107 to better accommodate North American sidetone preferences.


    4.6. E-Model Conventions
To obtain the consistent answers it is necessary to agree on certain conventions. These are listed
below.

   Ta = T
   Tr = 2T
   QDUminimum = 1 for all digital connections including IP telephones and gateways
   It is a relative tool rather than an absolute tool.


    4.7. Summary
The objectives of this section were met. The methods for dealing with delay, echo, speech
compression and packet loss using the E-Model were demonstrated. Also, the R-scale was explained
and feeling for what the impairments introduced by IP Telephony sound like were outlined.



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5. Wireline PSTN Voice Quality Benchmarks
IP telephony is replacing a well developed network with acceptable voice quality, called the PSTN.
Therefore, it is only natural to compare the performance of IP telephony with the benchmarks
established by the PSTN (wireline only). Since everyone has used the PSTN, everyone can relate
subjectively with terms like toll quality and PSTN quality, but what do they really mean terms the E-
Model and the R-scale. The objective of this section is to establish solid objective benchmarks for
comparing IP telephony to the PSTN. Or to put it another way, the objective is to establish definitive
relative E-Model references on the R-scale, for use by the IP telephony scenarios in the next section.

To this end three representative PSTN benchmarks will be illustrated: ISDN voice quality, PSTN
voice quality (referring to a connection with analog telephones at both ends and no speech
compression) and Toll Compression voice quality (referring to a connection with G.726 speech
compression). Then these benchmarks will be summarized as the existing PSTN region.

    5.1. ISDN Voice Quality
All digital (TDM) connections have ISDN voice quality and they have the following characteristics:

       G.711 only
       Echo control in the telephones (45 dB >= TCLw >= 40 dB)
       Nominal loudness ratings of SLR = 8 dB and RLR = 2 dB and STMR = 15 dB (note sidetone
        discussion in section 4.5)
       Delay from close to 0 ms to 100 ms (20,000 km @ 0.005 ms/km per ITU-T Recommendation
        G.114 Annex A)

Figure 13 graphically illustrates ISDN quality with delay from 0 to 100 ms as the dependent variable
and TELR of 50, 55 and 60 dB as the parametric variable. This corresponds to TCLw values of 40,
45 and 50 dB, respectively. The rational for using these values, is TIA-810 specifies a nominal
TCLw of 45 dB and a desirable TCLw of 50 dB (the yellow region). Previous digital set standards
specified TCLw values that were 5dB lower (the gray region).




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                                                  PN-4689V4 (to be published as TIA/EIA/TSB-116)

                          Figure 13 – ISDN Quality Voice Benchmark




The maximum delay for a national connection is about 25 ms @ 5000 km. The TCLw does not
matter at this distance; the R-values remain in the “very satisfied” category. However, as the delay
increases the R-value for the TCLw = 40 dB curve drops rapidly down to 80. The delay value of 100
ms was selected based on an international connection of about half the circumference of the earth
being about 20,000 km times the delay rate specified in G.114 Annex A of 0.005 ms/km.

The green region is the existing PSTN as defined by the region between the reference curve and R =
80 and between 0 and 100 ms. Figure 14 shows the connection diagram and the E-Model parameters
for the ISDN quality scenario.




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   Figure 14 – Block Diagram and E-Model Parameters for ISDN Voice Quality Benchmark
                                                                      Side A       0 dBr     Side B
                                                                       ISDN                   ISDN
                                                                      Terminal               Terminal




                                                Echo Path - Side A

                                                Echo Path - Side B

                                                        Abbrev.       ISDN                   ISDN
                               Title
                                                       (Default)     Terminal     0 dBr     Terminal
               Electric Circuit Noise (at 0 dBr)     Nc (-70)                      -70
               Room Noise                            Po (35)           35                      35
               Send Loudness Rating                  SLR      (8)       8                      8
               Receive Loudness Rating               RLR      (2)       2                      2
               D-factor                              D      (3)         3                      3
               Noise Floor                           Nfor (-64)        -64                    -64
               Sidetone Masking Rating               STMR (15)         15                      15
               Equipment Impairment Factor           Ie      (0)        0                      0
               Expectation (Advantage) Factor        A       (0)        0                      0
               Mean One-Way Delay (upper)            Tu       (0)       0                      0
               Mean One-Way Delay (lower)            Tl       (0)       0                      0
               Mean One-Way Delay (upper = lower)    Tul      (0)       0        0 to 100      0
               Electrical Loss (upper)               Lu
               Electrical Loss (lower)               Ll
               Electrical Loss (upper = lower)       Lul
               Quantizing Distortion Units (upper)   qduu      (1)      0.5                   0.5
               Quantizing Distortion Units (lower)   qdul      (1)      0.5                   0.5
               Echo Return Loss                      ERL             55,45,40               55,45,40



   5.2. PSTN Voice Quality
Mixed analog and digital (TDM) connections, without speech compression, have PSTN voice quality
and they have the following characteristics:

      G.711 only
      No echo control below 10 ms (ERL = 11 dB + 6 dB Rx loss = 17 dB)
      Echo control enabled at 10 ms (ELR = 55 dB)
      Nominal loudness ratings of SLR = 11 dB and RLR = -3 dB and STMR = 15 dB (note
       sidetone discussion in section 4.5)
      Delay from close to 0 ms to 100 ms (20,000 km @ 0.005 ms/km per ITU-T Recommendation
       G.114 Annex A)

The much lower echo control of the PSTN network before ECANs are enabled accounts for the very
rapid degradation of the quality vs delay as shown in Figure 15. Once the ECAN is enabled at 10 ms,
there is an abrupt improvement in the voice quality, due to the ECAN’s significant improvement in
ERL. ITU-T Recommendation G.131 provides guidance on when to enable the ECAN, but each
administration enables ECANs according to their own rules. For this scenario, 10ms was selected to
maintain a minimum R-value of 80 because some administration always have the ECANs enabled,
while others wait until 22 ms and the R-value drops to about 70.

The red curve with the ECAN enabled, uses the analog telephone loudness ratings, with G.711
codecs in the digital segment. Compare it to the red curve in Figure 17, which uses G.729 in the
digital segment, with an Ie of 7.

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                                                PN-4689V4 (to be published as TIA/EIA/TSB-116)


Figure 16 shows the connection diagram and the E-Model parameters for the PSTN quality scenario.

                         Figure 15 – PSTN Voice Quality Benchmark




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                                Figure 16 – Block Diagram and E-Model Parameters for PSTN Voice Quality Benchmark


                                                             Side A      Hybrid   Digital PBX                                                Digital PBX     Hybrid    Side B
                                                                                                             0 dBr
                                                             Analog               CODEC            ECAN                             ECAN           CODEC               Analog
                                                            Telephone                                                                                                 Telephone
                                                                         U dB               U dB (Nat'l                             (Nat'l U dB              U dB
                                                                                   A             & Int'l)                           & Int'l)            A
                                                                                       D                                Digital                     D

                                                                                       D                                PSTN                        D
                                                                                   A               55 dB                            55 dB               A
                                                                          L dB              L dB                                            L dB              L dB




                                       Echo Path - Side A

                                       Echo Path - Side B

                                                            Analog
                                              Abbrev.
                      Title                                   Set                 Digital PBX A2D                                   Digital PBX D2A                   Analog
                                             (Default)
                                                            (A-side)    Hybrid        w/ECAN                0 dBr    Digital PSTN       w/ECAN              Hybrid     Set
      Electric Circuit Noise (at 0 dBr)     Nc (-70)                                                         -70
      Room Noise                            Po (35)             35                                                                                                       35
      Send Loudness Rating                  SLR     (8)         11                                                                                                       11
      Receive Loudness Rating               RLR     (2)          -3                                                                                                      -3
      D-factor                              D     (3)             3                                                                                                      3
      Noise Floor                           Nfor (-64)          -64                                                                                                     -64
      Sidetone Masking Rating               STMR (15)           15                                                                                                       15
      Equipment Impairment Factor           Ie     (0)            0                        0                                                  0                          0
      Expectation (Advantage) Factor        A      (0)            0                                                                                                      0
      Mean One-Way Delay (upper)            Tu      (0)           0                                                                                                      0
      Mean One-Way Delay (lower)            Tl      (0)           0                                                                                                      0
      Mean One-Way Delay (upper = lower)    Tul     (0)           0                         0                          0 to 100               0                          0
      Electrical Loss (upper)               Lu                            0                 0                                                 0               6
      Electrical Loss (lower)               Ll                            6                 0                                                 0               0
      Electrical Loss (upper = lower)       Lul                           0                 0                                                 0               0
      Quantizing Distortion Units (upper)   qduu     (1)        0                          0.5                                               0.5                         0
      Quantizing Distortion Units (lower)   qdul     (1)        0                          0.5                                               0.5                         0
      Echo Return Loss                      ERL                          17                55                                                55              17




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                                                PN-4689V4 (to be published as TIA/EIA/TSB-116)



   5.3. Toll Compression Voice Quality
Mixed analog and digital (TDM) connections, with speech compression, have Toll Compression
voice quality and they have the following characteristics:

      G.711 and G.726 @ 32 kbit/s (Ie = 7)
      Echo control enabled (ERL = 55 dB)
      Nominal analog loudness ratings of SLR = 11 dB and RLR = -3 dB and STMR = 15 dB (note
       sidetone discussion in section 4.5)
      Nominal digital loudness ratings of SLR = 8 dB and RLR = 2 dB and STMR = 15 dB (note
       sidetone discussion in section 4.5)
      Delay from 10 ms to 100 ms (20,000 km @ 0.005 ms/km per ITU-T Recommendation G.114
       Annex A)

The gray region in Figure 17 shows the toll compression quality benchmark. The red curve matches
the analog connection details shown in Figure 18. The blue curve was calculated with digital
telephones, no hybrids, but with the ECANs enabled. The main impairment is due to the reduced
voice quality of the G.726 codec.

                   Figure 17 – Toll Compression Voice Quality Benchmark




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                    Figure 18 – Block Diagram and E-Model Parameters for Toll Compression Voice Quality Benchmark




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                                                   PN-4689V4 (to be published as TIA/EIA/TSB-116)



    5.4. Wireline PSTN Voice Quality Summary
The wireline PSTN is characterized by:

       analog telephones with low delay, good loudness ratings and poor echo control
       digital telephones with low delay, good loudness ratings and good echo control
       digital networks with low delay, low impairments and good echo control.

It is summarized by the green, “Existing PSTN” region in Figure 19 and it is bounded by the best
G.711 performance on the top, R =80 on the bottom and delay between 0 and 100 ms. Most of the
delay is available for propagation delay.

The above analysis is specific to wireline connections, but a brief mention of wireless characteristics
would set the stage for IP connections. Most wireless connections add about 100 ms one-way delay
for the handset + base station combination, which is about the same as the total wireline PSTN delay.
Also, wireless connections have Ie values between 5 and 20. These are significant impairments that
would mask the high voice quality of the existing PSTN and complicate the objective of defining IP
telephony goals and rules. However, the wireless, wireline and IP networks must work together and
this document will detail many of the challenges.


                             Figure 19 – Existing PSTN Voice Quality




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6. IP Telephony Voice Quality Analysis
Section 4 demonstrated the suitability of the E-Model as tool for estimating the relative voice and
conversation quality of IP telephony. Section 5 defined the voice quality of the existing PSTN. The
goal of this section is to define the region of voice quality that is acceptable for IP telephony and to
establish the rules to manage the impairments introduced by IP telephony.

The first step is to define the lower voice quality threshold. R = 70 is the dividing line between
“some” and “many” users dissatisfied. The maximum delay for R = 70 is about 325 ms. This is about
twice the amount of delay that causes problems in interactive conversations. The equivalent
impairment factor for R = 70 is 24, which is about the same as G.726 ADPCM at 24 kbit/s or GSM
Half-rate with its typical delay of about 100 ms. Also, as mentioned in the section 5 Summary,
wireless has significant delay and it uses speech compression, so the region between R = 70 and R=
50 will be used in practice by IP to wireless connections. That makes this region an unsuitable goal
for IP telephony, even under the banner of “Best Effort”. R = 70 is a reasonable lower threshold.

Figure 20 shows the previously established existing PSTN voice quality region in green, overlaid on
the acceptable IP voice quality region above R = 70 in yellow. The red region below R = 70 is not
recommended because it generates too many complaints. Combining the yellow and green regions
into one acceptable region does not encourage matching PSTN quality, but following the three ITU-T
categories is not appropriate either given the delay introduced by IP telephony.

Instead the R = 80 lower threshold of the existing PSTN voice quality region needs to be extended to
the intercept of the G.711/Ie = 0 curve to define a desirable voice quality goal for IP telephony. The
recommended voice quality goals for IP telephony are shown in Figure 21 with the desirable region
between R = 80 and the best-case G.711 curve and the acceptable region between R = 80, R= 70 and
the best-case G.711 curve.

                   Figure 20 – PSTN and IP Telephony Voice Quality Regions




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Figure 22 is another way of presenting the information in Figure 21. In this graph, R and Ie trade
places, so that Ie is the y-axis, R is the parametric variable with the contours defining the desirable,
acceptable and not recommended regions. Some prefer this alternative view.

                 Figure 21 – Recommended IP Telephony Voice Quality Regions




                 Figure 22 – Recommended IP Telephony Voice Quality Regions




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Figure 23 is similar to Figure 1. It summarizes the recommended IP telephony voice quality
categories and compares them to MOS, %GOB and %POW scales. This figure includes a new
category between R = 70 and R = 50 called “Reach Connection” to address connections that include
other high impairment and/or high delay technologies like wireless.


               Figure 23 – Recommended IP Telephony Voice Quality Categories

 G.107            R     Recommended IP Categories MOS                     %GOB %POW
 Default          94                              4.4                      98.4 0.1
 Value
                                    Desirable

                  80                                              4.0        89.5       1.4
                                   Acceptable
                  70                                              3.6        73.6       5.9


                  60          Reach Connection                    3.1        50.1      17.4

                  50                                              2.6        26.6      37.7
                              Not Recommended
                    0                                             1.0          0       99.8



    6.1. Voice Quality Issues for IP Telephony
   Delay
   G.711 packet loss concealment (PLC)
   Low bit-rate coders
   Packet loss
   Transcoding
   Echo Cancellers (ECAN)
   TCLw
   New loss plan


    6.2. Voice Quality Recommendations for IP Telephony

       6.2.1. Delay
Delay Rule #1: Use G.711, unless using it causes transcoding then transport the bits instead.
Delay Rule #2: Use the 1 ms speech frame size specified in H.323 for G.711 to minimize jitter buffer
delay.
Delay Rule #3: Actively minimize one-way delay.
Delay Rule #4: Believe in the E-Model, which permits longer delays for low Ie-value codecs, like
G.711 for a given R-value, see Figure 21.

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                                                           PN-4689V4 (to be published as TIA/EIA/TSB-116)


This section tries to illustrate the effects of delay without reiterating ITU-T Recommendation G.114,
the definitive document on delay. One-way delay has three components:
     propagation delay
     encoding/decoding/packetization + jitter buffer delay
     transport delay.

One assumes that any telephone can call any other telephone, therefore propagation delay must be
reserved, even though it may not always be required. As previously established in “Existing PSTN”
analysis, 100 ms is reasonable amount to reserve for propagation delay.

G.114 has a thorough analysis of encoding/decoding/packetization delay. It provides the following
formulas for calculating the minimum and maximum coder-related processing delay:

Minimum packetization delay = (N+1) x frame size + look-ahead
Maximum packetization delay = (2N+1) x frame size + look-ahead
Where, N = number of frames per packet.

Ideally, the jitter buffer should be sized to the expected delay of the transport. Eventually, transport
jitter must be reduced to zero by replacing it with QoS controls, but initially jitter buffers solve the
lost and late packets problem by introducing delay, sometimes very significant delay. There are two
ways to treat jitter buffers. One is frame-based, which means the size of the buffer is a multiple of the
speech frame size. The other is absolute based is simply the size necessary to do the buffering. The
frame based jitter buffer can increase delay dramatically, if the frame size is large. A rule of thumb
for frame-based buffers is the jitter buffer must be two times the speech frame size.

Table 3 summarizes G.114 packetization delay for many combinations of G.711 speech frames and
frames per packet, with and without the two times the speech frame size jitter buffer. The designation
row follows the format of M/N, where M = speech frame size and N = number of speech frames per
packet. H.323 specifies a 1 ms speech frame size for G.711. Table 3 shows, for example, the effect of
treating the packet as ten 1 ms frames (1/10) or one 10 ms frame (10/1). Figures 25 to 28 illustrate
the packetization and jitter buffer delay issue by overlaying bar charts of the information in Table 3
on the R vs. Delay curve for G.711. Figure 24 does not include the effects of the jitter buffer because
it is negligible with 1 ms speech frames. These figures reserve the first 100 ms for propagation delay
and clearly highlight the need for small speech frames and minimum transport jitter.

     Table 3 – G.711 Packetization and Packetization + Jitter Buffer Delay based on G.114
Designation      1/5     1/10   1/20   1/30   1/40   5/1     5/2   5/3   10/1   10/2   10/3   20/1   20/2   30/1
Speech frames     1        1      1      1      1     5       5     5     10     10     10     20     20     30
Frames/packet     5       10     20     30     40     1       2     3      1      2     3       1     2       1
Look ahead        0        0      0      0      0     0       0     0      0      0     0       0     0       0
G.114 min         6       11     21     31     41    10      15    20     20     30     40     40     60     60
G.114 max        11       21     41     61     81    15      25    35     30     50     70     60    100     90
Jitter (JB)       2        2      2      2      2    10      10    10     20     20     20     40     40     60
G.114+JB min      8       13     23     33     43    20      25    30     40     50     60     80    100    120
G.114+JB max     13       23     43     63     83    25      35    45     50     70     90    100    140    150
Note: all times in ms.

Table 4 is similar to Table 3, except it




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                      Table 4 – G.729A and G.732.1 Packetization and
                     Packetization + Jitter Buffer Delay based on G.114
               Designation             10/1   10/2   10/3    10/4     30/1
               Speech frames            10     10     10      20        30
               Frames/packet            1       2      3      4          1
               Look ahead               5       5      5      5        7.5
               G.114 min                25     35     45      55       67.5
               G.114 max                35     55     75      95       97.5
               Jitter (JB)              20     20     20      20        60
               G.114+JB min             45     55     65      75      127.5
               G.114+JB max             55     75     95     115      157.5
              Note: all times in ms.




    Figure 24 – G.711 Packetization Delay with 1 ms Speech Frame, without Jitter Buffer




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                                             PN-4689V4 (to be published as TIA/EIA/TSB-116)

Figure 25 – G.711 Packetization Delay with 5 and 20 ms Speech Frames, without Jitter Buffer




    Figure 26 G.711 Packetization Delay with 5 and 20 ms Speech Frames + Jitter Buffer




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Figure 27 – G.711 Packetization Delay with 10 and 30 ms Speech Frames, without Jitter Buffer




   Figure 28 – G.711 Packetization Delay with 10 and 30 ms Speech Frames + Jitter Buffer




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                                          PN-4689V4 (to be published as TIA/EIA/TSB-116)

Figure 29 – G.729A Packetization Delay with 10 ms Speech Frame, without Jitter Buffer




   Figure 30 – G.729A Packetization Delay with 10 ms Speech Frame + Jitter Buffer




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  Figure 31 -– G.723.1 Packetization Delay with 30 ms Speech Frame, without Jitter Buffer




      Figure 32 -– G.723.1 Packetization Delay with 30 ms Speech Frame + Jitter Buffer




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                                                    PN-4689V4 (to be published as TIA/EIA/TSB-116)



  is contained summary for IP telephony is the maximum one-way delay should be less than 200 ms
and the R-value should be greater than 70. Table 1 shows the nominal encoding/decoding + jitter
buffer delay for the G.711, G.729 and G.723.1 codecs for the practical combinations of speech frame
size and frames per packet using the following rule of thumb: Total Delay = encoding/decoding +
jitter buffer = (3 + number of frames/packet – 1) x speech frame size + look ahead + (2 x speech
frame size). On the right-hand side of the table the total delay is split one third on the send side and
two thirds on the receive according to section 5.10 of TIA/EIA/IS-810. But by studying the G.711
curve in Figure 2 it is apparent that the total one-way delay that can be tolerated by a G.711 (0%
packet loss) call is about 325 ms, before the user satisfaction drops below R = 70. TIA/EIA/IS-810
provides guidance for G.711 packet voice latency for IP telephones. It advises that the overall
telephone component be less than 100 ms. This leaves 100 ms one-way delay for the hubs, firewalls,
gateways, routers, jitter buffers and transport, etc., in the connection plus the propagation delay. The
one-way propagation delay across North America is about 25 ms, leaving 75 ms for the hardware.
However, Table 1 indicates that with a 10 ms speech frame and two frames/packet, G.711 should
having an encoding/decoding + jitter buffer delay of only 50 ms. This would leave about 125 ms for
the hardware. The equipment impairment factor (Ie), for the codecs of interest are listed in Table 2.
This information was taken from ITU-T Recommendation G.113, APPENDIX I, Table I.1. Figure 2
highlights the impairment effects of the low bit-rate coders, e.g., Ie for G.729A + VAD = 11 which is
subtracted on the R or y-axis from the G.711 or reference value. The reduced the voice quality
reduces the amount of delay available to the hardware. In fact, G.729A and G.723.1 can only tolerate
210 ms and 240 ms respectively, before the user satisfaction drops below R = 70. The impairments
and fixed delay introduced by speech compression to save bandwidth, ultimately has the effect of
reducing the amount of network and propagation delay available to the connection, to point where
bandwidth-saving low bit-rate coders are impractical. Using G.711 and optimizing delay provides the
customer with a better product.

G.114 provides minimum and maximum packetization delay guidelines for various speech coders.
The previous Figure plots the packetization delay for G.711, G.729 and G.723.1 at various speech
frame rates and with various speech frames per packet. To give a more realistic estimate of the total
delay a jitter buffer delay of two times the speech frame rate to each of the G.114 numbers. The
horizontal lines show the TIA 810 guidelines for IP telephone set delay of less than 100 ms and the
G.114 total one-way delay objective of less than 150 ms. The difference between the 150 ms yellow
line and the top of the bars indicates how much delay is available for network elements and
propagation, before interactive conversation issues begin.




        6.2.2. G.711 Packet Loss Concealment (PLC)
G.711 Rule: Use PLC with G.711.
Although PLC adds about 10 ms delay.
G.729?? and G.723.1 have packet loss concealment (PLC) algorithms, which reduce the effects of
packet loss by using information in the current packet to estimate the following packet if it doesn’t
arrive in time. Recommendation G.711 does not specify a PLC algorithm because originally, it was
intended to be used in a dedicated 64 kbit/s channel. Recently, ITU-T SG16 approved the T1.521
Annex A PLC algorithm as Appendix I of Recommendation G.711. Also, T1.521 documents a
second algorithm PLC in Annex B. However, PLC algorithms work on the receive side only and
therefore proprietary algorithms are acceptable. Figure 3 shows how necessary it is to have a good
PLC algorithm. This information was taken from ITU-T Recommendation G.113, APPENDIX I,
Table I.3 and is reproduced in Table 3. The top curve is the default settings curve for the E-Model
and the curve for G.711 with 0% packet loss. The bottom curves are G.711 without PLC for 1% and

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PN-4689V4 (to be published as TIA/EIA/TSB-116)

2% packet loss. This shows that the performance drops off very rapidly for small amounts of packet
loss. The middle curves are G.711 with PLC for the same 1% and 2% rates of packet loss. The
improvement is dramatic and necessary.




                                            Figure 3
                          G.711 Performance with Packet Loss / Scenario 1



           6.2.3. Low Bit-rate Coders
Text....



           6.2.4. Packet Loss
Packet Loss Rule: Use G.711 with PLC.

Figures 3, 4 and 5 show the random packet loss performance family of curves for the G.711 + PLC,
G.729A + VAD and G.723.1A (6.3 kbit/s) codecs, respectively. The Ie values in Table 4 were taken
from ITU-T Recommendation G.113, APPENDIX I, Table I.2 and Table I.3. The family of curves for
each codec shows consistent performance degradation with respect to packet loss on top of the Ie
impairments related to the low bit-rate codecs. This emphasizes the point made in the delay section
that low bit-rate codecs reduce the amount of delay available to the network. Taking packet loss into
consideration, G.723.1 is barely usable and G.729 is a poor second to G.711.

Packet Loss Rule: Use G.711 with PLC.




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                                    PN-4689V4 (to be published as TIA/EIA/TSB-116)

                                Table 4
% Packet Loss      G.729A + VAD        G.723.1.A + VAD     GSM EFR
                                           6.3 kbit/s
       0                  11                  15               5
      0.5                 13                  17               —
       1                  15                  19               16
      1.5                 17                  22               —
       2                  19                  24               21
       3                  23                  27               26
       4                  26                  32               —
       5                  —                   —                33
       8                  36                  41               —
      16                  49                  55               —
NOTE – Number of frames per packet:
• G.729-A + VAD: 2;
• G.723.1-A + VAD: 1.
• GSM EFR: 1




                              Figure 4
            G.729A Performance with Packet Loss / Scenario 1




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PN-4689V4 (to be published as TIA/EIA/TSB-116)




                                           Figure 5
                        G.723.1 Performance with Packet Loss / Scenario 1




       6.2.5. Transcoding
Transcoding Rule: Maximum number of transcodes = 1.

Three connection examples are shown, depicting 1, 2 and 3 transcodes. For the sake of brevity, all
required hubs, routers, PBX’s, etc. are rolled into the appropriate clouds (“IP/Packet Network” or
“Circuit Switched Network”). Also, echo cancellers are not shown (it can be assumed that sufficient
echo cancellation is provided by the end-terminals). Transcodes are between G.729A and G.711
vocoders.


                                                 1 Transcode

                          Side A                    Gateway                    Side B
                             IP                                                 Digital
                         Telephone                                            Telephone
                                                         G.7xx
                                                    JB
                                                                    Circuit
                                     IP/Packet
                                                                   Switched
                                      Network
                                                                   Network
                                                          G.711



                         G.729A      G.729A       G.729A - G.711    G.711      G.711



                                                 2 Trancodes




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                                                                          PN-4689V4 (to be published as TIA/EIA/TSB-116)
                                   Side A        Gatew ay                           Gatew ay           Side B
                                      IP                                                                  IP
                                  Telephone                                                           Telephone
                                                         G.7xx                      G.711
                                                JB
                                                                      Circuit
                                                                     Sw itched
                                                                     Netw ork                   JB
                                                          G.711                      G.7xx


                                  G.729A      G.729A - G.711          G.711      G.711 - G.729A       G.729A




                                                                  3 Transcodes

  Side A         Gatew ay                        Gatew ay                           Gatew ay                           Gatew ay        Side B
     IP                                                                                                                                   IP
 Telephone                                                                                                                            Telephone
                      G.7xx                      G.711                                       G.7xx                     G.711
                 JB                                                                 JB
                                Circuit                                                                   Circuit
                                                                     IP/Packet
                               Sw itched                                                                 Sw itched
                                                                      Netw ork
                               Netw ork                     JB                                           Netw ork               JB
                       G.711                      G.7xx                                       G.711                     G.7xx


  G.729A      G.729A - G.711    G.711         G.711 - G.729A          G.729A     G.729A - G.711           G.711      G.711 - G.729A   G.729A




             6.2.6. Echo Cancellers (ECAN)
Text....



             6.2.7. TCLw
Text....




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PN-4689V4 (to be published as TIA/EIA/TSB-116)




                                                         Figure 6

                                                 Scenario for TCLw Example



                                           42
                                                              TIA/EIA-IS-810




                                          Figure 7
                                  TCLw Scenario Performance



           6.2.8. New Loss Plan
Text....




                                            1
TIA/EIA-IS-810

7. Appendix A - VoIP End to End delay budget planning for Private Networks
                                SOURCE:        Cisco Systems
                                170 West Tasman Drive
                                San Jose, CA 95134
                                USA


CONTACTS:                     Kirit                                                            Patel
                              Phone:    (408)                                              525-1355
                              Fax:      (408)                                              525-9150
                              E-mail:   kdpatel@cisco.com
                              Michael                                                       Knappe
                              Phone: (408)                                                 527-3849
                              Fax:    (408)                                                526-0455
                              E-mail: mknappe@cisco.com



NOTE: Appendix A is contribution TIA TR-41.1.2/00-02-04. It was agreed to temporarily insert this
contribution as an Appendix. The text will be merged with the existing delay text in Section 6 before
the next meeting, but the tables will remain in this Appendix.




                                               2
                                                                                                                            TIA/EIA-IS-810

      8. Introduction


      This paper provides a more detailed view of VoIP one way end to end delay sources in a private IP
      network or intranet. End-to-end delay will be used synonymously with one-way delay in this
      document. Section 1.1 covers delay sources in an example worst case end to end private network.
      Section 1.2 and 1.3 show detailed end-to-end delay budget planning in a VoIP network for aG.729A
      vocoder and shows how the end-to-end delay is affected by the voice packet size, link speed and
      maximum data packet size. Although this document covers the delay budget planning for the G.729A
      vocoder only, the same planning rules can be applied to any other vocoder.




            8.1. VoIP End-to end delay sources overview and definitions

      Figure 1 below shows a VoIP end-to-end private network connection and lists the main delay sources
      for each section of the network. There are basically two types of delay source, fixed or variable and
      each delay source in the figure 1 is listed in one of the two categories.


                                  Figure 33: VoIP End-to-End delay sources for Private network case



                     Originating                                                                              Terminating
                                                                          Core Network
                     Voice-LAN                                                                                 Voice-LAN

    A-          Orginating               Edge                                                              Edge                    Terminating         B-
   Side         Gateway                  Router                                                            Router                   Gateway
                             L1 -Link                                                                                                                 Side
                                                       L2 -Link           Core Network        L2 -Link                  L1 -Link
                                                                            Routers




Fixed:                                            Fixed:            Fixed:               Fixed:                     Fixed:                       Fixed:
                    Fixed:
- Look ahead                                      - Serialization   - Switching          - Serialization            - Switching                  - Decoding
                    - Switching
- Encoding                                             WAN          - Progation               WAN                   Variable:                    Variable:
                    Variable:
- Buffer                                                            - Serialization                                 - Voice contention           - Dejitter buffer
                    - Voice contention
- VAD                                                               Variable:                                       - Data contention
                    - Data contention
- Packetizing                                                       - Voice contention
                                                                    - Data contention

                                                                                                                                                   KDP 2/10/2000




                                                                               3
TIA/EIA-IS-810


Delay sources definition

Vocoder Encoding
Details on the vocoder delays are from ITU-T Recommendation G.114 and also see section 5.2.1 of
PN-4689. This consists of fixed delays, look ahead, the encoding process and packetization. There is
also the additional serialization delay to transmit the packets over the 10/100 Base T link, but this is
negligible (much less than 1ms) so it is ignored.


Originating Voice-LAN
Fixed switching delay:
 through the edge switch can be significant since forwarding engines in the edge switch are not very
fast.

Variable Voice contention delay:
 is delay due to contention between voice packets for the link bandwidth. Average queue delays
caused by contention between voice packets sharing the same queuing priority can be modeled using
the queuing theory formula for fairly constant bit rate traffic sharing a single queue is:
Average voice queuing time is: tQ-av = tdls * /2*(-1)
Worst case queuing time is (95% of distribution): tQ-wo = 2*tQ-av
Where tdls is Voice packet link serialization delay
 is the link utilization of voice packets

Variable Data/voice contention delay:
is delay due to contention between voice packets and data packet, where data packet has already
started transmission. When forwarding node uses priority scheduling algorithms for differentiated
QoS between voice and data classes, than the maximum time voice packet is delayed by the data
packet is:
 tD-max = (Maximum # Data MTU bytes + 48 overhead)/(link speed kbps/8)

Important planning rule: need to use priority scheduling for voice class traffic, as well as RTP
header compression and data packet fragmentation on slow speed links to minimize the
contribution of this variable delay source.

Fixed Serialization WAN delay:
 is delay due to voice packets transmission on the WAN L2- link. The link rate can vary from 56kb/s
to OC3 and up. The formula for serialization delay is:
tV-max = (Voice packet bytes + 48 overhead)/(link speed kbps/8)

Important planning rule: in order to minimize the effect of this delay source, avoid using slow
serial links in any of the end to end network connections.

Core Network
Fixed switching delay:
Includes packet switching engine delay (see originating Voice-LAN section for details) and any other
network multiplexing equipment delays. An estimate of 1 ms of delay for each hop is used in the
calculation table in the next section.

Fixed propagation delay:
is the cumulative delay due to the physical ‘speed of light’ limitations of propagation through the
network. Details for this are contained in ITU-T Recommendation G.114. For the purpose of this
exercise a figure of 5s/km is used in the calculation of the table in the next section.


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                                                                                         TIA/EIA-IS-810


Fixed Serialization delay network:
is the same as defined earlier, but since the link rate in the core network is usually in the broadband
range, the total effect of this delay source is small enough (< 1.5 ms) that it is ignored in the
calculation table in the next section.



Variable Voice contention delay:
is same as defined earlier:
Average voice queuing time is: tQ-av = tdls * /2*(-1)
Worst case queuing time is (95% of distribution): tQ-wo = 2*tQ-av
Total core network worst case queuing time is (95% of distribution): = t Q-wo * (number of hops -1)

Since the link rate in the core network is usually in the broadband range, the tdls delay source is
small in addition , link utilization ratio for voice packet is small, that the total effect of this delay
source can be ignored in the calculation table in the next section.

Variable Data/voice contention delay:
is same as defined earlier:
 tD-max = (Maximum # Data MTU bytes + 48 overhead)/(link speed kbps/8)
Total core network maximum data MTU queuing time is: = tQ-wo * (number of hops -1)

Important planning rule: need to use priority scheduling for voice class traffic, as well as RTP
header compression and data packet fragmentation on slow speed links to minimize the
contribution of this variable delay source.

Terminating Voice-LAN

Fixed Serialization WAN:
 is delay due to voice packet transmission on the WAN L2- link. The link rate can vary from 56kb/s
to OC3 and up. The formula for serialization delay is:
tV-max = (voice packet bytes + 48 overhead)/(link speed kbps/8)

Important planning rule: in order to minimize the effect of this delay source, avoid using slow
serial links in any of the end to end network connections.

Vocoder decoder

Variable dejitter buffer delay:
Is the delay required to buffer all the variable delays in the network so that the voice packets can be
played at constant bit-rate to the decoder. The size of dejitter buffer is, vocoder encoding
compression amount plus the total variable delay in the end to end connection.

Fixed decoder:
Details on the vocoder decoding delay is detailed in ITU-T Recommendation G.114




                                                  5
TIA/EIA-IS-810



    8.2. VoIP End-to end delay budget Case 1




                                 Table 5: Case 1a - VoIP End to end delay budget
                    Case 1a: L1 = 10Mb/s; L2 = 128kb/s; Data MTU max= 128
                                                   Codec type:            G.729    G.729     G.729    G.729
                                                                          10.00    10.00     20.00    20.00
Delay type                                                        Units   Fixed   Variable   Fixed   Variable
                                                                          (ms)     (ms)      (ms)     (ms)



A-side phone
      Encoding process delay            Codec Look ahead           ms      5.0                5.0
                                          Encoding compression     ms     10.0               10.0
                                             1xbuffer              ms     10.0       ~       10.0       ~
             Packetization delay                                   ms      0.0       ~       10.0       ~
             # of Voice bytes/pkt                                 Bytes   10.0               20.0
Orginating Voice-LAN
                       Switching       1 hops, @ > 100 pps         ms     10.0               10.0
     Voice contention queuing        voice packets queuing @       ms               1.5                2.9
                                        128kb/s (Max 2*SD)
                   Data Queuing Max. data unit 128 bytes +48       ms               11.0               11.0
                                      O/H @ 128kb/s
      Serialization WAN delay Voice pkt + 48 O/H @ 128kb/s         ms      3.6       ~        4.3       ~

Core Network
                       Switching        5 hops, @ > 1k pps         ms      5.0                5.0
     Voice contention queuing        voice packets queuing @       ms               0.1                0.3
                                       1544kb/s (Max 2*SD)
                   Data Queuing 5 hops, Max data 128+48 O/H        ms               3.6                3.6
                                      @ 1544kb/s avg
               Serialization core     Voice pkt + 48 O/H @         ms      1.2       ~        1.4       ~
                                            1544kb/s
              Propogation delay         5000km @ 5us/km            ms     25.0       ~       25.0       ~
Terminating Voice-LAN
      Serialization WAN delay Voice pkt + 48 O/H @ 128kb/s         ms      3.6       ~        4.3       ~

                       Switching       1 hops, @ > 100 pps         ms     10.0               10.0
B-side phone
             Dejitter buffer delay   1 comp. delay + network       ms     10.0      16.2     10.0      17.8
                                          variable delay
                 Decoding delay                                    ms     10.0       ~       10.0       ~
                                                                          103.5     32.5     114.9     35.7
                                                        Min/Max           103.5    135.9     114.9    150.6




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                                                                                               TIA/EIA-IS-810



                                 Table 6: Case 1b - VoIP End to end delay budget

                    Case 1b: L1 = 10Mb/s; L2 = 128kb/s; Data MTU max= 512
                                                   Codec type:            G.729    G.729     G.729    G.729
                                                                          10.00    10.00     20.00    20.00
Delay type                                                        Units   Fixed   Variable   Fixed   Variable
                                                                          (ms)     (ms)      (ms)     (ms)



A-side phone
      Encoding process delay            Codec Look ahead           ms      5.0                5.0
                                          Encoding compression     ms     10.0               10.0
                                             1xbuffer              ms     10.0       ~       10.0       ~
             Packetization delay                                   ms      0.0       ~       10.0       ~
             # of Voice bytes/pkt                                 Bytes   10.0               20.0
Orginating Voice-LAN
                       Switching       1 hops, @ > 100 pps         ms     10.0               10.0
     Voice contention queuing        voice packets queuing @       ms               1.5                2.9
                                        128kb/s (Max 2*SD)
                   Data Queuing Max. data unit 512 bytes +48       ms               35.0               35.0
                                      O/H @ 128kb/s
      Serialization WAN delay Voice pkt + 48 O/H @ 128kb/s         ms      3.6       ~        4.3       ~

Core Network
                       Switching        5 hops, @ > 1k pps         ms      5.0                5.0
     Voice contention queuing        voice packets queuing @       ms               0.1                0.3
                                       1544kb/s (Max 2*SD)
                   Data Queuing 5 hops, Max data 512+48 O/H        ms               11.6               11.6
                                      @ 1544kb/s avg
               Serialization core     Voice pkt + 48 O/H @         ms      1.2       ~        1.4       ~
                                            1544kb/s
              Propogation delay         5000km @ 5us/km            ms     25.0       ~       25.0       ~
Terminating Voice-LAN
      Serialization WAN delay Voice pkt + 48 O/H @ 128kb/s         ms      3.6       ~        4.3       ~

                       Switching       1 hops, @ > 100 pps         ms     10.0               10.0
B-side phone
             Dejitter buffer delay   1 comp. delay + network       ms     10.0      48.2     10.0      49.8
                                          variable delay
                 Decoding delay                                    ms     10.0       ~       10.0       ~
                                                                          103.5     96.4     114.9     99.6
                                                        Min/Max           103.5    199.9     114.9    214.5




                                                             7
TIA/EIA-IS-810



    8.3. VoIP End-to end delay budget Case 2




                                 Table 7: Case 2a - VoIP End to end delay budget

                    Case 2a: L1 = 10Mb/s; L2 = 1544kb/s; Data MTU max= 128
                                                   Codec type:            G.729    G.729     G.729    G.729
                                                                          10.00    10.00     20.00    20.00
Delay type                                                        Units   Fixed   Variable   Fixed   Variable
                                                                          (ms)     (ms)      (ms)     (ms)



A-side phone
      Encoding process delay            Codec Look ahead           ms      5.0                5.0
                                          Encoding compression     ms     10.0               10.0
                                             1xbuffer              ms     10.0       ~       10.0       ~
             Packetization delay                                   ms      0.0       ~       10.0       ~
             # of Voice bytes/pkt                                 Bytes   10.0               20.0
Orginating Voice-LAN
                       Switching       1 hops, @ > 100 pps         ms     10.0               10.0
     Voice contention queuing        voice packets queuing @       ms               0.1                0.2
                                        128kb/s (Max 2*SD)
                   Data Queuing Max. data unit 128 bytes +48       ms               0.9                0.9
                                      O/H @ 1544kb/s
      Serialization WAN delay         Voice pkt + 48 O/H @         ms      0.3       ~        0.4       ~
                                            1544kb/s
Core Network
                       Switching        5 hops, @ > 1k pps         ms      5.0                5.0
     Voice contention queuing        voice packets queuing @       ms               0.1                0.3
                                       1544kb/s (Max 2*SD)
                   Data Queuing 5 hops, Max data 128+48 O/H        ms               3.6                3.6
                                      @ 1544kb/s avg
               Serialization core     Voice pkt + 48 O/H @         ms      1.2       ~        1.4       ~
                                            1544kb/s
              Propogation delay         5000km @ 5us/km            ms     25.0       ~       25.0       ~
Terminating Voice-LAN
      Serialization WAN delay Voice pkt + 48 O/H @ 128kb/s         ms      0.3       ~        0.4       ~

                       Switching       1 hops, @ > 100 pps         ms     10.0               10.0
B-side phone
             Dejitter buffer delay   1 comp. delay + network       ms     10.0      4.8      10.0      5.1
                                          variable delay
                 Decoding delay                                    ms     10.0       ~       10.0       ~
                                                                          96.8      9.6      107.1     10.2
                                                        Min/Max           96.8     106.4     107.1    117.3




                                                             8
                                                                                               TIA/EIA-IS-810



                                 Table 8: Case 2b - VoIP End to end delay budget

                    Case 2b: L1 = 10Mb/s; L2 = 1544kb/s; Data MTU max= 512
                                                   Codec type:            G.729    G.729     G.729    G.729
                                                                          10.00    10.00     20.00    20.00
Delay type                                                        Units   Fixed   Variable   Fixed   Variable
                                                                          (ms)     (ms)      (ms)     (ms)



A-side phone
      Encoding process delay            Codec Look ahead           ms      5.0                5.0
                                          Encoding compression     ms     10.0               10.0
                                             1xbuffer              ms     10.0       ~       10.0       ~
             Packetization delay                                   ms      0.0       ~       10.0       ~
             # of Voice bytes/pkt                                 Bytes   10.0               20.0
Orginating Voice-LAN
                       Switching       1 hops, @ > 100 pps         ms     10.0               10.0
     Voice contention queuing        voice packets queuing @       ms               0.1                0.2
                                       1544kb/s (Max 2*SD)
                   Data Queuing Max. data unit 512 bytes +48       ms               2.9                2.9
                                      O/H @ 1544kb/s
      Serialization WAN delay         Voice pkt + 48 O/H @         ms      0.3       ~        0.4       ~
                                            1544kb/s
Core Network
                       Switching        5 hops, @ > 1k pps         ms      5.0                5.0
     Voice contention queuing        voice packets queuing @       ms               0.1                0.3
                                       1544kb/s (Max 2*SD)
                   Data Queuing 5 hops, Max data 512+48 O/H        ms               11.6               11.6
                                      @ 1544kb/s avg
               Serialization core     Voice pkt + 48 O/H @         ms      1.2       ~        1.4       ~
                                            1544kb/s
              Propogation delay         5000km @ 5us/km            ms     25.0       ~       25.0       ~
Terminating Voice-LAN
      Serialization WAN delay         Voice pkt + 48 O/H @         ms      0.3       ~        0.4       ~
                                            1544kb/s
                       Switching       1 hops, @ > 100 pps         ms     10.0               10.0
B-side phone
             Dejitter buffer delay   1 comp. delay + network       ms     10.0      14.8     10.0      15.0
                                          variable delay
                 Decoding delay                                    ms     10.0       ~       10.0       ~
                                                                          96.8      29.5     107.1     30.1
                                                        Min/Max           96.8     126.3     107.1    137.2




                                                             9

				
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