============================================================
1. Question 1/XII - Future programme of work
(continuation of Question 1/XII studied in 1985-88)

Considering

(a)   that the evolution of new technology and services raises issues
related to telephone transmission quality on a continuing basis;

(b) that there is no systematic way of identifying and introducing such
issues into the programme of work unless they are related to existing
questions under study,

1. What are major factors in telecommunications which should influence
the future programme of study in the area of telephone transmission
quality?

2. What new Questions important to administrations should be considered
for study in future Plenary periods?

2. Question 2/XII - Hands-free telephony
(merging and continuation of Questions 2/XII and 17/XII studied in 1985-
88)

Considering

(a) that the desirable transmission characteristics of conventional
hands-free telephones are described in P.34;

(b) that desirable transmission characteristics of Group Audio Terminals
are given in P.30;

(c) that advances in technology have allowed for the development of a
range of hands-free communication terminals with a variety of
loudspeaker and microphone arrangements, with complex acoustic echo
control (various combinations of voice-switching, suppression and
cancellation techniques) and likely to include wide bandwidth,

the following points should be studied:

1. What improvements can be brought in the modelling of speech
production and particularly perception on the physical aspect (head
diffraction effects) and on the psychoacoustical aspect (two-ears
listening) in order to improve the determination of loudness for hands-
free terminals? What can be gained by the use of head and torso
simulators?

2. What further conditions should be satisfied by hands-free terminals
equipped with complex acoustical echo controllers?

3. What measuring methods should be applied, what kind of signals should
be used and what values should be proposed in order to improve the
existing methods of assessing the performances of A-E-C equipped
terminals (especially to take into account the effect of noise,
acoustical environment, double-talk, network degradation, ...)

4. What other improvements can be brought in the design of hands-free
terminals: choice of microphones and loudspeakers characteristics and
arrangements (depending on the service using hands-free telephony like
hands- free mobile communication, etc.), introduction of signal
processing techniques like dereverberation, noise-guard, acoustical
arrays, etc. in order to minimize the degrading effects due to the
acoustical environment?

============================================================
Question 3/XII - Definitions in the field of telephonometry and of
characteristics of international connections and circuits
(continuation of Question 35/XII studied in 1985-88)

Considering

(a) that there is a need for uniformity of terminology in the field of
telephonometry and of characteristics of international connections and
circuits, including transmission planning;

(b) that it appears desirable to collect the pertinent definitions in
one general Recommendation on terminology;

(c) that new concepts and terms are evolving,

the following should be studied:

What definitions should be given to terms used in the field of
telephonometry and of characteristics of international connections and
circuits?
Note 1 - This field includes such topics as general transmission
performance, transmission impairment, hypothetical reference
connections, propagation time, echo and stability.

Note 2 - Coordination with some other Study Groups (in particular with
Study Groups XI, XV and CMV may be necessary.

Note 3 - Study on this Question should be based on Recommendations G.100
and P.10.

Note 4 - Attention is drawn also to Recommendations of the A-Series.

============================================================
Question 4/XII - Updating of the CCITT telephonometric and transmission
planning Handbooks
(continuation of Question 36/XII studied in 1985-88)

Considering that

i. new testing methodologies are continually evolving; and

ii. that Recommendations are being:

a) generated;

b) updated; and

c) replaced,

iii.the telephonometric and transmission planning Handbooks will need
constant revision,

the following Question should be studied:

What revisions may be drafted into the telephonometric and transmission
planning Handbooks to take account of recent changes?

============================================================
Question 5/XII - Speech synthesis/recognition systems
(continuation of Question 5/XII studied in 1985-88)

Considering

that speech synthesis/recognition systems will be exploited to control
the access to the telephone network, to data bases or other functions
through the telephone network, agrees to study the subjective
acceptability of such systems, from the viewpoint of the performance of
single devices as well as of the whole interactive system/service;

decides to put the following Question, the study of which should be
organized as follows:

synthesis and recognition systems should be studied in the
telecommunication environment taking into account particularly the
characteristics normally found there, for example:

1) bandwidth and/or bit rate,

2) loss and level of signals,

3) distortion,

4) noise.

The Question can be segregated into the following categories:

1) Voice synthesis in telephony

Definition of synthesis
Vocabulary size
Intelligibility
Naturalness
Listener effort

2) Voice recognition in telephony

Vocabulary size
Input format (isolated words or continuous speech)
Correct recognition and rejection ratio
Robustness to background and circuit noise and distortions
Speaker dependency
Language/Dialect dependency
Training time and procedure
Recognition time

3) Specific items which should be studied are:

a) Which characteristics can be quantitatively measured and which
assessment procedures are suggested?

b) Can acceptability ranges be recommended?

c) Can standardized speech data bases be established to enable the
testing of recognizers and synthesizers in the telephony environment?

d) How will administrations deal with the multi-language problem?

Study Group XII asks if Study Group II might wish to study:

4) Synthesis/recognition interactive services

Input format (syntactic requirements)
Error correction, ease and time required
Response time
Feedback response mode (audio/visual)
Overall friendliness
Applications
ANNEX 1

(to Question 5/XII)

List of documents for Question 5/XII, study period 1985-1988

COM XII-15, June 1985 (British Telecom): Early contribution to propose
new question on speech recognition and synthesis: method for assessing
isolated word speaker dependent recognition systems.

Annex B to the reply to Question 18/XII, in Report COM XII-R 12,
September 1986 (Liaison Officer between Study Group XII and Study Group
XVIII): Status report on Study Group XVIII/8 (Speech processing)

Annex to the reply to Question 5/XII in Report COM XII-R 12, September
1986 (CSELT, Italy): Subjective assessment of automatic voice answering
devices

COM XII-148, February 1987 (France): An "objective" evaluation of
difficulty in understanding voice synthesis devices

COM XII-176, June 1987 (Sweden): Subjective quality assessment of
synthetic speech
ANNEX 2

(to Question 5/XII)

Preliminary reply to Question 5/XII, in COM XII-R 29,
February 1988, 2.4
ANNEX 3

(to Question 5/XII)

Contribution COM XII-176 with the following amendments:

-Page 2, between second and third paragraphs, add the following
paragraph: "An average listening level was expected to be within the
preferred range."

-Page 3, replace last but one paragraph by: "20 subjects participated in
the test. The speech was presented to them monaurally over headphones at
a comfortable listening level (approximately 80 dB sound level as
measured on an artificial ear)."

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Question 6/XII - Harmonization of G.100-Series of Recommendations
(continuation of Question 37/XII studied in 1985-88)

Considering that

(a) when planning the (national and international) circuits, connections
and networks for the transmission and switching of telephone and other
voice-band signals, some problems may arise from the present lack of
harmonization of the Recommendations concerning the apportionment of
impairments to the national systems and international circuits as well
as to the two-wire and four-wire sections; further, as a consequence of
the fact that Recommendations were drafted at different periods in time
and with emphasis on the corresponding technologies (earlier, purely
analogue and later on, mixed analogue/digital transmission and
switching), the Recommendations - though being themselves correctly and
well formulated - may convey sometimes slightly different meanings,
view-points and general guidances; these facts are in the future no
longer admissible as more and more correlations are being established
between the transmission quality parameters in the national and
international circuits in the evolving networks;

(b) revisions to existing Recommendations will be made in the future,
and new Recommendations will be proposed;

(c) changes to the tables published in Volume III of the Blue Book will
be required,

The CCITT decides to study the following Questions:

1. What modifications and/or rearrangements are necessary to the present
texts of the G.100-Series Recommendations to adequately harmonize them?

2. What modifications and/or rearrangements are necessary to the present
text of the tables summarizing the Recommendations concerning line
transmission, in order that they are harmonized with the G-Series
Recommendations?

============================================================
Question 7/XII - Models for predicting transmission quality from
objective measurements
(continuation of Question 7/XII studied in 1985-88)

Considering

(a) that according to CCITT Recommendation P.74, "Conversation tests"
are preferable to other methods for subjective determination of
transmission quality;

(b) that such tests cannot be carried out on a sufficiently large scale
to cover the whole range of all degrading factors that are of interest
and combinations of more than two factors at a time are very rarely
feasible;

(c) that Supplement No. 3, to CCITT Recommendation P.11 contains
descriptions of models used respectively by Bellcore, BT, CNET and NTT
for assessing the effects of customer satisfaction of certain ranges of
degrading factors;

(d) that, in principle, it is desirable that the following degrading
factors should be taken properly into account by such models:

Factor 1. Transmission loss

2. Circuit noise

3. Room noise

4. Attenuation/frequency distortion

5. Sidetone

6. Non-linear distortions

7. Echo

8. Propagation time

9. Delay distortion

10. Mutilation by voice switching;

CCITT decides to study the following Question (the Question consists of
three parts):

Part 1 - What changes can be made to the models to incorporate
additional degradations?

Part 2 - How well do all these models predict the results of
"Conversation tests" conducted with real subjects?

Part 3 - To what extent are the predictions from the different models
compatible with each other?

Note 1 - In studying the Question, comparisons have to be made:

i) between the results of "Conversation tests" as described in
Recommendation P.74 conducted by different administrations;

ii) between predictions by different models, and

iii) between a particular set of "Conversation test" results and a
particular model.

These comparisons have been hampered because uniform methods have not
been used for expressing the objective characteristics of the
connections. It is strongly recommended that as much use as possible be
made of methods defined in CCITT Recommendations and relating to
loudness ratings, especially P.64 and P.76.

Note 2 - A great deal of information helpful in studying this Question
appears in contributions to other Questions, especially 9/XII
(Sidetone), 13/XII (Non-linear distortion) and 18/XII (Quantizing
distortion).

Note 3 - Annex 1 contains the text of the reply to the Question after
study during 1985-1988. Annex 2 contains a list of contributions which
have become available for use in the study of the Question.

Note 4 - It is intended that the model or models finally recommended
shall be used to update the numerical values now given provisionally in
Recommendation P.11. The model or models may also be used for studying
the Questions mentioned in Note 2.
ANNEX 1

(to Question 7/XII)

Refer to the list of documentation in Report COM XII-R 29,
February 1988, sections 4.1 and 4.2
ANNEX 2

Refer to the reply to Question 7/XII in Report COM XII-R 29,
February 1988, section 4.4
ANNEX 3

(to Question 7/XII)

It seemed to be urgent to prepare a Recommendation for the quickly
developing land mobile radio-telephone service. The aim was to have
transmission quality equal or at least very similar to the PSTN. On the
other hand due to the limited [25 kHz, 30 kHz] radio frequency bandwidth
16 kbit/s voiceband coding is necessary. To meet simultaneously the
above mentioned requirement was up to now not possible. Therefore
further study is necessary which must be supported by


============================================================
Question 8/XII - Improvement of the methods for the determination of
loudness ratings
(New Question)

Considering

(a) that the methodology for the determination of loudness ratings has
been defined in a number of CCITT P-Recommendations;

(b) that following the world-wide use of loudness ratings it seems that
some of the methods are not perfect so far as the accuracy,
reproducibility, comparability and simplicity are concerned;

(c) that there is widespread usage of headsets in telephony;
(d) that there have been recent advances in both headset technology and
complexity;

(e) that the results of subjective and objective measurements on
headsets have been studied in the last Plenary period but further
analysis is required;

(f) that the IEC-711 occluded ear simulator has been adopted within
Recommendation P.51 for ear insert receivers, but other types of headset
receivers do not have recommended test methods.

PART A - HANDSETS

The following Questions should be studied:

a) What improvement should be made to Recommendation P.78 for the
determination of loudness ratings for special purposes?

Note 1 - Although the network planning mainly depends on the objective
measured data of loudness ratings, however, subjective determination
will still be necessary in some cases, for example, the determination of
SLRs of certain types of carbon sets, or a new design of telephone set
with non-linear amplitude characteristics.

Note 2 - A method using the direct balance technique against IRS for the
subjective determination of loudness ratings is described in Supplement
No. 17 Volume V of the Blue Book.

b) What improvement should be made to Recommendations P.64 and P.75 for
the determination of sensitivity/frequency characteristics of a local
telephone system including telephones using non-linear techniques and/or
noise cancelling microphones.

Note 3 - Annex 1 summarizes the progress of the study of the previous
Question 8/XII (measuring the efficiency of a microphone or a receiver),
especially for the carbon microphones.

Note 4 - Study of the difference between SMJ and SmJ, i.e., the sending
SFC measured by real mouth and by artificial mouth, LM, and the earphone
coupling loss, LE are covered by the present Question.

Note 5 - The artificial mouth and the artificial ear used for the
determination of sensitivity/frequency characteristics are studied under
Question 12/XII, and the algorithm used for the calculation of loudness
ratings is studied under Question 15/XII.

c) What improvement should be made to Recommendation P.65 on the
instrumentation for the objective measurement of loudness ratings?

Note 6 - Description of some existing instruments for the objective
measurement of loudness losses (including LR, R25E etc.) is given in
section 5 of the Handbook on Telephonometry.

Note 7 - During the study period 1985-1988 modifications have been made
to some types of the instruments mentioned above. These points can be
found in a number of contributions of the study period.

PART B - HEADSETS

What precautions should be recommended when measuring the performance of
special types of headsets?

Examples of special types of headsets:

i) headsets with noise cancelling microphones;

ii) headsets with intra-concha, circumaural and non-contact types of
receivers;
iii) headsets with either dual receivers or "dual receive-path"
receivers.
ANNEX 1

(to Question 8/XII)

Reply to Question 8/XII in Report COM XII-R 27, February 1988,
sections B.2.2.2 and B.2.2.3

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Question 9/XII - Sidetone
(continuation of Question 9/XII studied in 1984-88; new wording)

Considering

(a) that from the point of view of the talker in quiet surroundings a
sidetone rating method STMR (Sidetone Masking Rating - Recommendations
P.76, P.79) that treats the human sidetone as a masking threshold yields
results that correlate well with the subjective effects of sidetone;

(b) that from the point of view of the subscriber considered as a
listener in noisy conditions the STMR method can, if associated with the
sidetone sensitivity measured with room noise, be used to derive a
listener sidetone rating (LSTR) that correlates well with subjective
effects of listener sidetone;

(c) that for various connection conditions values for STMR and LSTR are
recommended in G.121 and P.11;

(d) that digital exchanges or systems can present an unfavourable
impedance to the telephone set and increase sidetone levels and short
delay echo signals on some connections;

(e) that in order to achieve adequately high values of LSTR under
adverse connection and/or room noise conditions some manufacturers and
administrations will make use of noise cancelling microphones requiring
positional discipline in their use,

1. Are the values of STMR and LSTR recommended in P.11 and G.121
sufficiently comprehensive for the control of sidetone under variety of
connection conditions or does the information need to be supplemented?

2. What design criteria can be recommended to ensure that satisfactory
sidetone and/or short delay talker echo levels are maintained during
telephone conversations?

3. In what circumstances or under what conditions is it possible to
recommend the use of noise cancelling microphones, special acoustic or
electronic techniques in telephone handsets?
ANNEX

(to Question 9/XII)

Reply to Question 9/XII, section B.3.3 in Report COM XII-R 27, February
1988

============================================================
Question 10/XII - Speech transmission characteristics for digital
handset telephones
(continuation of Question 10/XII studied in 1985-88; new wording)

Considering

(a) that we now have Recommendation P.31 concerning SLR, RLR, STMR,
LSTR, to send and receive frequency responses for digital sets,

the following Question should be studied:

1. What are the desirable speech transmission characteristics for
digital telephone sets? Contributions should address the headings listed
hereafter but the headings under 1.1, 1.2, 2.1, 2.2 and 3.1 should only
be addressed if a revision of Recommendation P.31 seems necessary.

1. Sending
1.1 Sending frequency characteristics
1.2 Sending loudness rating
1.3 Distortion
1.4 Noise
1.5 Linearity

2. Receiving
2.1 Receiving frequency characteristics
2.2 Receiving loudness rating
2.3 Distortion
2.4 Noise
2.5 Linearity

3. Sidetone
3.1 Sidetone masking rating
3.2 Distortion

4. Telephone acoustic loss
4.1 Echo
4.2 Stability

5. Delay

6. Other
6.1 Discrimination against out-of-band input signals
6.2 Spurious out-of-band signals

2. How can Recommendation P.35 Part B about handset dimensions be
improved.

Note - Test methods are dealt with under Question 38/XII. Wideband
telephony are dealt with under Question 17/XII.
ANNEX

(to Question 10/XII)

Report on digital telephones
(study period 1985-1988)

1.Send and Receive LR:s for Digital Sets

1.1Provisional Recommendation from the previous study period

In the annex to Question 10/XII 1.3 a provisional recommendation on LR:s
for digital sets was given.

1.2 Contributions

-COM XII-18 (Special Rapporteur for Question 19/XII) - "On the optimum
CRE and LR"

The document is summarized and discussed in COM XII-R 4.

-COM XII-72 (BT) "Preferred loudness ratings for digital telephones"

-COM XII-108 (NTT) "Desirable SLR and RLR values for digital telephone
sets"

-COM XII-110 (NTT) "Allowable LR for digital telephone sets considering
echo effect under the mixed analogue digital environment"

These contributions are summarized and discussed in COM XII-R 11.

-COM XII-169 (BT) "Loudness ratings for digital telephones"

The contribution is summarized and discussed in COM XII-R 16.

-COM XII-179 (Norway) "Transmission aspects for digital telephony"
-COM XII-R 20 Annex 1 to reply to Q.10 (Norway) "Comments on
transmission aspects for digital telephony" (i.e. COM XII-179)

-COM XII-R 20 Annex 2 to reply to Question 10 (BT) "Comments on COM XII-
179"

The contributions are reported in COM XII-R 20.

Working Party XII/2 - Delayed D.52 (ATT) - "Send loudness ratings for
digital telephones"

The main finding in Working Party XII/2-D.52 was that linear microphone
equipped telephones produced an average 2.8 dB higher active speech
level on the telephone line as measured with an SV-6 speech voltmeter
conforming to Recommendation P.56. Although the study was done for sets
having equal objectively measured EARS-based Transmit Objective Loudness
Ratings (TOLRs), there is every reason to believe that similar results
would hold for carbon and linear sets having equal Send Loudness Ratings
(SLRs) based on Recommendation P.79.

The implication is that linear microphone equipped telephones should
have about 3 dB different (less loud) SLRs in order to produce the same
speech levels on the telephone line, and hence the same subjective
loudness for the far end listener, as carbon transmitter equipped
telephones. This subject is still under study by the EIA, and no
official U.S. position has been taken on the matter. But any
recommendation concerning the sending level for digital telephones which
is derived from grade-of-service models and experiences based on carbon
transmitter equipped telephones measured using swept sine waves may want
to allow for such differences being taken into account.

The findings above may influence the implementation of recommended LR
values. More information is needed before more precise conclusions can
be drawn.

1.3 Discussion

From the previous study period the following ranges were provisionally
recommended:

nominal SLR within the range 3 to 5 dB

nominal RLR within the range 0 to 2 dB

These values were mainly based on an investigation by BT showing an
optimum for OLR around 5 dB. During this study period several
administrations have expressed their concern about transmission
degradation from echo. This degradation will decrease with increased
OLR. As the optimum around 5 dB is fairly flat the provisionally
recommended ranges were changed to the following short term objective:

nominal SLR within the range 5 to 11 dB

nominal RLR within the range -1 to 5 dB

The range has also been increased to allow for compatibility with the
existing analogue network because of differences in the transmission
plans in different countries.

However, as a long term objective the following values have been agreed
upon:

nominal SRL = +8 dB

nominal RLR = +2 dB

COM XII-179 is in a form of a standard (NET 5) as an outcome of work
within Europe which contains tolerance limits for SRL and RLR. However,
it has been agreed upon not to include such tolerance limits in the
proposed recommendation. The draft Recommendation (P.31) as agreed upon
in the October meeting 1987 (COM XII-R 20) is found in section 5.

At the meeting in June 1985 (COM XII-R 4) some problems with LR:s for
digital telephones connected to PABX:es were reported. To achieve
compatibility with analogue telephones in a mixed analogue and digital
network as low LR values as SLR = +5 dB and RLR = -4 dB sometimes had
been chosen. The largest difference relative to the present proposed
recommendation seems to be at receiving. One way out of the problem, so
that the customers can achieve receiving levels as they are used to, may
be to allow for a volume control for instance as allowed for in the
proposed NET 5 standard (COM XII-179 amended by Annexes 1 and 2 of the
report of Q.10 in COM XII-R 20) see 4.2.2.2.

2. Sidetone characteristics for digital sets

2.1 Suggestion from the previous study period

In the annex to Question 10/XII 3, a suggestion for STRE was given.

2.2 Contributions

-COM XII-115 (BNR) "Sidetone characteristics for digital sets"

COM XII-116 (NTT) "Desirable STMR for digital telephone sets"

The contributions are summarized and discussed in COM XII-R 11.

-COM XII-169 (BT) "Loudness ratings for digital telephones"

The contribution is summarized and discussed in COM XII-R 16.

-COM XII-179 (Norway) "Transmission aspects for digital telephony"

-COM XII-R 20 Annex 1 to reply to Q.10 (Norway) "Comments on
transmission aspects for digital telephony" (i.e. COM XII-179)

-COM XII-R 20 Annex 2 to reply to Q.10 (BT) "Comments on COM XII-179"

-Delayed Contribution D.46 (October meeting 1987) (Special Rapporteur on
Question 9/XII) "Sidetone for digital telephony"

The contribution is discussed in COM XII-R 20.

-COM XII-226 (Sweden) "The effect of delayed sidetone on the overall
quality of a telephone connection" (New contribution)

The contribution presents the results from a conversation test where the
following parameters were varied

Overall Echo Loudness Rating (OELR) = 0, 10, 20, 25 and 30 dB

STMR = 15 dB and the electrical path disconnected

Delay times = 0, 4, 16 and 32 dB

The investigation shows:

The delayed sidetone is marked by the direct sidetone at STMR = 15 dB
when the delayed sidetone is weak (OELR = 30 dB).

If the electrical path is disconnected the weak delayed sidetone (OELR =
30 dB) has an obvious influence on the quality at the longest delay,
i.e. 32 ms.

At OELR = 20 dB the delayed speech is disturbing and increasing with the
delay and the disturbing efect is higher with the direct sidetone
switched off.

At OELR = 10 dB the delayed sidetone is dominating over the direct
sidetone and the degradation is considerable already at a 4 ms delay.

2.3 Discussion

The question of sidetone is discussed more in detail in connection with
Question 9/XII.

From our previous study period a suggestion for a STRE value was given.
However, it has been agreed that STMR and LSTR are more adequate
measures for sidetone effects at talking and listening respectively.

A compromise value STMR around 15 dB with respect to both quiet and
noisy room conditions was suggested in COM XII-R 11. With respect to the
study connected with Question 9/XII and what people are used to with
present analogue telephones a nominal range of 10-12 dB was agreed upon
at the Working Party XII/2 meeting in October 1987 (COM XII-R 20).
(Manufacturing tolerances should not be given.) However, at the Working
Party XII/2 meeting in Geneva, January 1988 it was agreed to include
also the value +15 dB in the recommended range which means 10-15 dB
which takes into account the results of our agreement to include LSTR as
a factor. It should also be noted that a certain sidetone level will
also mask the talker echo as reported in 3.2.2 above.

As suggested in the work with Q.9 it was also agreed at the Working
Party XII/2 meeting in October 1987 (COM XII-R 20) that the concept of
listener sidetone should also be adapted and that a LSTR > 15 dB should
be recommended. The proposed recommendation for STMR and LSTR are
included in Recommendation P.31 (see section 5 below).

3. Send and Receive frequency responses

3.1 Suggestion from the previous study period

In the annex to Question 10/XII 2 a suggestion for frequency responses
was given as reference for further study.

3.2 Contributions

-Working Party XII/2-Delayed D.1 (LM Ericsson) (Working Party XII/2
meeting June 1985) "On effect of mains frequency noise and room noise
with digital telephones"

This contribution was summarized and discussed in COM XII-R 4.

-COM XII-54 (NTT) "Preliminary study on the desirable sending
sensitivity frequency characteristics for digital telephone sets"

This contribution is summarized and discussed in COM XII-R 9.

-Working Party XII/2-Temporary Document 19 (Special Rapporteur on
Question 10/XII) (Working Party XII/2 meeting September 1986)
"Corrections on the frequency response limits in COM XII-1 page 61" ( 2
in the annex to Question 10/XII).

This contribution is considered in COM XII-R 11.

-COM XII-164 (NTT) "Sending and receiving frequency responses for
digital telephone sets"

-COM XII-168 (BT) "Desirable sending frequency response characteristics
for digital telephones"

This contributions are summarized and discussed in COM XII-R 16.

-COM XII-179 (Norway) "Transmission aspects for digital telephony"

-COM XII-R 20 Annex 1 to reply to Q.10 (Norway) "Comments on
transmission aspects for digital telephony" (i.e. COM XII-179)

-COM XII-R 20 Annex 2 to reply to Q.10 (BT) "Comments on COM XII-179"

-COM XII-229 (Sweden) "Desirable sending frequency response of telephone
sets"

These contributions are reported in COM XII-R 20.

3.3 Discussion

From the previous study period we had a suggestion for frequency
characteristic limits as shown in the annex to Q.10.

At the meeting in April 1987 (COM XII-R 16) the earlier proposed limits
were considered to be too narrow and should be opened up. However, wide
limits will give very little guidance of the preferred frequency
characteristics. At the October meeting 1987 (COM XII-R 20) it was
therefore agreed to recommend frequency responses instead of giving
upper and lower limits. Manufacturing tolerances should not be included.

COM XII-229 makes a review and comparison of available information. It
makes the following conclusion assuming a substantially flat response
for the receiving side, for the sending side a recommended range of
response curves might cover linear slopes between 0 and +3 dB/octave
within the transmission band. The slope +3 dB corresponds to a level
difference of about 12 dB between the upper and lower band limit. The
same difference is obtained by a response curve which is flat below 1
kHz and raises linearly towards 12 dB at 4 kHz. This type of response
curve is expected to give at least as good quality as a constant slope
within the whole band.

Based on available contributions and the agreement at the last Working
Party XII/2 meeting a draft Recommendation for sending and receiving
frequency characteristics is given in section 6.

4. Telephone acoustic loss (TAL) and to TAL related echo

4.1 Contributions

-COM XII-97 (BT) "Considerations of echo caused by the acoustic loss
path of telephone sets"

-COM XII-109 (BT) "Measurement of acoustic loss"

These contributions are summarized and discussed in COM XII-R 11.

-COM XII-165 (Norway) "Assessment of talker echo caused by the acoustic
echo-path in a 4-wire telephone set"

-COM XII-166 (BT) "Measurements of the acoustic loss of handset
telephone sets on human and artificial ears"

-COM XII-Delayed D.25 (BT) (Working Party XII/2, April 1987)
"Measurement of HAL using a simple simulation"

These contributions are summarized and discussed in COM XII-R 16.

-COM XII-179 (Norway) "Transmission aspects for digital telephony"

-COM XII-Delayed D.38 (BT) (Working Party XII/2 October 1987)
"Suggestions for amendments to draft Recommendation on digital telephone
testing"

-COM XII-Delayed D.39 (BT) (Working Party XII/2 October 1987) "A human
head simulator for handset acoustic loss determination"

-COM XII-218 (BNR) "Measurement of acoustic echo path loss in handsets"

These documents are summarized and discussed in COM XII-R 20.

-COM XII-Delayed D.48 (BNR) "Measurements of acoustic echo path loss in
handsets" is a revised version of COM XII-218. The revision is that the
echo path is also quoted in LR figures.

4.2 Discussion

In a 4-wire telephone in a fully digital network the coupling between
the earphone and the microphone will be the main and perhaps only path
that will contribute with a loss that surpresses echo. Therefore it is
important to reach a recommendation on the loss that can be achieved and
which should be recommended. This will also determine if extra echo
surpression will be needed.

In the study contributions have addressed the different sources for
coupling:

-acoustic coupling

-electric coupling

-sesmic coupling (in the handset)

All of these need consideration when designing a handset. It has been
shown that in carefully designed handsets the acoustic coupling will
dominate. It has been agreed that the coupling should be measured
simulating real use. For this purpose a measuring method has to be
defined. A real use simulator will also be needed to facilitate the
measurements. Contributions on coupling loss measured on human heads has
been reported as well as a suggestion of a simulator. At present some
more information is needed to arrive at recommendations but the
sugestion in COM XII-Delayed D.39 of

-a human head simulator and of

-a telephone acoustic loss > 46 dB
should be considered in the forthcoming study. It is also suggested that
the question of a human head simulator should be studied in connection
with Question 12/XII.

5. Others

5.1 Contributions

-COM XII-167 (BT) "Suitable characteristics to be specified for digital
sets"

This contribution is discussed in COM XII-R 16.

-COM XII-179 (Norway) "Transmission aspects for digital telephony"

-COM XII-R 20 Annex 1 to reply to Q.10 "Comments to transmission aspects
for digital telephony" (i.e. COM XII-179)

-COM XII-R 20 Annex 2 to reply to Q.10 (BT) "Comments to COM XII- 179"

The contributions are reported in COM XII-R 20.

5.2 Discussion

Relevant parts of COM XII-167 have been incorporated in the proposed new
question in section 7 below.

The other three contributions form the present proposal for the NET 5
standard and contain parts related to all the headings given in the
proposed new question in section 7 and should therefore be remembered in
the forthcoming study of the new question.

============================================================
Question 11/XII - Transmission degradation introduced by interaction
between voice operated devices
(continuation of Question 11/XII studied in 1985-1988; new wording)

Considering

- that international network configurations can contain a combination of
voice operated devices such as:

- acoustical or electrical echo control devices, speech interpolation
devices, DCME, packetizing devices, etc.

the following Questions should be studied:

1. What are the network configurations of particular interest needing
testing priorities?

2. How is the transmission quality of a network configuration impaired
by:

-such devices, acting in tandem, or in association with each other, and
by

-the interaction of these devices with other relevant connection
characteristics (loss, noise (stationary or changing), echo path loss
and delay, etc. ...),

3. What are the most appropriate testing methodologies able to evaluate
and compare the transmission degradation induced in the different
network configurations?

4. What are the maximum overall performance degradations acceptable for
these network configurations in conjunction with subscriber expectations
(see for example Note 1, 2.2 in Recommendation G.131)?

5. What other considerations and possible limitations can be introduced
by voice band data transmission, if any?

Note - A liaison will have to be established with the Questions dealing
with transmission quality evaluation of equipment listed in the wording
of Q.11/XII.

============================================================
Question 12/XII - Artificial mouths and ears
(continuation of Question 12/XII studied in 19851988; new wording)

Considering

(a) that the increasing spread of oddly shaped earcaps which, besides
not coupling sometimes adequately to the human ear, also are measured
with difficulty according to the current procedures.

(b) that the appearance of intra-concha, circumaural, non-contact type
of earphones in telecommunications equipment, like operator headsets.

(c) that the results of the international experiment on the evaluation
of the human mouth radiation.

Administrations are requested to contribute on the following subjects:

1. a) Do the administrations consider that there is a need for a type of
artificial ear reproducing the acoustical, mechanical and geometrical
characteristics of the human ear?

b) If so, for what purpose should such a device be used?

c) What characteristics should it exhibit and how should these
characteristics be realized?

2. a) Do the administrations consider that there is a need for an
artificial ear for measuring without leakage intra-concha, circumaural
and non-contact type earphones, used in telecommunications equipment?

b) If so, what characteristics should it exhibit and how should these
characteristics be realized?

3. a) Should the artificial mouth be included in a HATS (Head and Torso
Simulator) in order to better reproduce the radiation characteristics of
the human mouth?

b) If so, for what purpose should such a device be used (hands free
sets, group audio terminals (GATs), operator headsets, telephone
handsets)?

c) What characteristics should it have and how should they be
implemented?

Note 1 - The annex describes the status of Question 12/XII at the end of
study period 1985-1988.

Note 2 - The results of the international experiment on the
characterization of the mouth radiation are summarized in COM XII-230
(1987). Machine readable diskettes with all the analytical data of the
experiment can be obtained from the Special Rapporteur of Q.12/XII.

Note 3 - Close collaboration with IEC is highly desirable.
ANNEX

(to Question 12/XII)

Refer to final status report on this Question in
Report COM XII-R 27, section B.4, February 1988

============================================================
Question 13/XII - Methods for the evaluation of non-linear distortions
(continuation of Question 13/XII studied in 19851988; new wording)

Non-linear distortion, in its most general sense, occurs in any system
where the output is not linearly related to the input. Currently, there
are several different methods for objectively quantifying non-linear
distortion some of which are only applicable with a particular input
signal (e.g. harmonic distortion). The following Questions should
therefore be studied:

1. What objective methods are available for evaluating the subjective
effect of non-linear distortion on speech transmission quality?

2. What are the applications/limitations of these methods, in terms of
predicting the subjective perception of non-linear distortion and can a
single method be recommended?

3. What test signals are appropriate for these methods?
(Attention is drawn to Recommendation P.50 which specifies a CCITT
artificial voice and to Question AA/XII)?

The results of the study should be applicable to the testing of
individual pieces of equipment, or in conjunction with models being
developed under Question 7/XII, as a more general tool for use in
transmission planning of complete connection.

Notes - Points to consider when evaluating the various methods for
measuring non-linear distortion include:

1. Correlation with subjectively derived results for a wide variety of
non-linear processes, including digital encoding schemes.

2. Capability of operating on actual hardware. This implies that the
method must be able to take into account any time delay existing between
input and output samples (at least up to delays typically encountered).

3. Capability of including other network impairments (e.g. frequency
response, circuit noise absolute listening level etc.) in the
calculation model.
ANNEX

(to Question 13/XII)

During the 1985-1988 study period, three different objective methods for
evaluating non-linear distortion were proposed. Since it was not
possible to gain agreement on a single method, each of these methods has
been retained for further study. The methods are:

1) A method from BNR based on the Coherence Function

(See COM XII-46 "Objective evaluation of non-linear distortion effects
on voice transmission quality" and COM XII-175 "Re-evaluation of the
objective method for measurement of non- linear distortion")

2) A method from NTT based on the Cepstral Distance

(See COM XII-7 "Proposal of study items on objective quality evaluation
method for voiceband codecs", COM XII-8 "Proposal of objective quality
measure for voiceband codecs" and COM XII-86 "Objective speech quality
estimation of non-linear telecommunication devices by LPC Cepstrum
distance measure")

3) A method from France based on the Information Index (See COM XII221
"Calculation of transmission performance from objective measurements by
the information index method").

============================================================
Question 14/XII - Application for the artificial voice
(New Question)

Considering

(a) that the progress of the last study period concerning the artificial
voice generation, which has been included in the new Recommendation
P.50;

(b) that the results of the international data collection of speech
samples;

(c) that the probability of applications of Recommendation P.50 to
objective speech quality evaluation for new digital technologies, i.e.,
objective evaluation of low bit rate coders, DCME, echo control devices
and loud-speaking telephones, are urgently required.

Administrations are requested to contribute on the following subjects:

1. Where can the newly standardized artificial voice be applied for the
objective evaluation of telecommunication equipment. What are the
limitations in any given applications?

2. Which characteristics of real speech are still missing?

3. How can silent periods be inserted into the artificial voice to
simulate actual conversation?

4. How can the artificial voice be applied to the calibration of the
digital speech voltmeter?

5. Does the application of the artificial voice as provisionally
recommended in P.50 have any influence on the conventional load defined
in Recommendation G.227? If yes, what modification is necessary in the
text of that Recommendation?

============================================================
Question 15/XII - Loudness rating, algorithm and application rules
(continuation of Question 15/XII studied in
1985-1988 new wording)

Considering

(a) that the concept of loudness rating will be widely adopted by
administrations for their transmission planning and assessment of speech
transmission quality;

(b) that a basis for the application of loudness ratings has been given
in the P- and G-Series Recommendations (particularly Recommendation
P.79);

(c) that the telephone networks will grow more complex and be fitted
with more varied features:

-what changes and additions are needed in the loudness rating
calculating rules in order to provide administrations with viable
planning and evaluation tools?

============================================================
Question 16/XII - Impedance strategy in the local network
(continuation of Question 16/XII studied during the 1985-1988 study
period; new working)

Considering

- that a range of equipment is used to interwork with the local network
and that such equipment incorporates 2-wire to 4-wire conversion with
associated impedances and possible adaptive balancing then the following
Question should be studied.

What is the optimum impedance strategy for future customer equipment,
local lines and network digital switching and multiplex equipment?

In particular the following points should be considered:

a) the echo and stability performance of connections;

b) the sidetone performance offered to customers;

c) the effect on loudness rating specifications;

d) the specification of the levels of information tones and
announcements;

e) the use of adaptive balancing techniques in 2-wire 4-wire conversion
devices;

f) the tolerances of specified impedances.
ANNEX 1

(to Question 16/XII)

List of documents submitted to the Question

COM XII-1Que3tions allocated to Study Group XII, 1985-1988 study period

COM XII-2 (Ellemtel) Sidetone, impedance matching and permissible end-
end loss

COM XII-21 (CTNE) Optimum network for the impedance of subscriber
equipment and the hybrid transformer balance impedance in digital local
exchanges

COM XII-22 (LM Ericsson) Sidetone and nominal impedance of local
exchanges

COM XII-44 (Netherlands) Sidetone, impedances and the transmission plan

COM XII-45 (India) Comments on 2.2 of Annex 4 of COM XII-32

COM XII-75 (Sweden) Sidetone problems in PABXs

COM XII-76 (Sweden) Subjective tests on talker and listener sidetone

COM XII R 6, R 13, R 18Reports of the Working Party XII/4 meetings
ANNEX 2

(to Question 16/XII)

COM XII-215 (from Working Party XI/4), Annex 2 to Part B.3,
1984-1988 study period

============================================================
Question 17/XII - Actual and preferred speech levels in telephone
connections
(New Question)

Considering

(a) that a new speech level measurement technique has been recommended
(Recommendation P.56);

(b) that the speech levels assumed as a design rule for FDM carrier
systems (see Recommendation G.223) may not represent the actual speech
levels found in most telephone networks of today;

(c) that PCM equipment reacts to the peak values of signals, and that
the relation between peak and RMS values may vary with the different
pre- emphasis used for the telephone set response curves as well as the
different subscriber line attenuation distortions encountered;

(d) that subscribers expect received speech in recorded announcements to
be within a range of levels they have found to be comfortable and giving
good intelligibility for normal telephone speech;

(e) that systems for reproducing recorded announcements are used to
provide network services;

(f) that systems for recording and reproducing voice messages left by
one customer for receipt by another at a later time are used or will be
used to provide network services such as voice mail;

(g) that the level of the speech signals produced by these systems is
adjustable and must be set to the proper level;

(h) that customer has a preferred listening level that depends on, among
other things, room and circuit noise;

(i) that the speech announcement or playback systems may be located some
distance (electrically) from the receiving customers and the loudness
loss of the electrical path may change from call to call,

The CCITT has decided to study:

1. What are the actual speech levels (RMS and peak values), activity
factors and distributions found in typical telephone networks?

2. What is the preferred speech level for announcements, recorded
messages and synthetic speech systems?

3. What is the preferred speech level for normal telephone connections?

4. What reference points are used to establish the level?

5. What measurement techniques and devices are recommended for measuring
the speech level?

Note 1 - Annex 1 provides guidance for this study.

Note 2 - Attention is drawn to Supplement No. 5, Volume III.2 for
information on measurements conducted in study periods 1968-72 and 1973-
76.

Note 3 - Attention is drawn to Question 5/XII with respect to synthetic
speech systems.
ANNEX 1

(to Question 17/XII)

To simplify calculations when designing carrier systems the CCITT has
adopted a conventional value to represent the mean absolute power level
(at a zero relative level point) of the speech plus signalling currents
of -15 dBmA of which -16 dBmO is assumed for the speech signals. With an
assumed speech activity of approximately 25 per cent the corresponding
active speech power could be -10 dBmO. However, actual speech power
distribution have a standard deviation of several dB and some
information suggests that the mean of the actual distribution and the
preferred value of speech level based on user opinion may be
substantially lower. For example, information in contributions from
Bellcore permit the following estimates:

a) Preferred active speech level

Preferred active speech level at 0 dB RLR -23 dBm
(for noise level of -56 dBmp) (TD WP XII/3)

Short-term objectives for RLR referred to 1-6 dB
0 dBr international switching point
(CCITT Recommendation G.129)

Preferred active speech level at 0 dBr -17 to -22 dBm
international switching point

b) Actual speech level

Mean active speech power at 0 dB RLR -9-OLR dBm
Supplement 3, Volume V

Short-term objective for SLR referred to 7 to 15 dB
O dBr international switching point

Corresponding mean active speech level at-16 to -24 dBm
0 dBr international switching point

These results would suggest that speech announcements should be in the
range from -16 to -24 dBm at a 0 dBr international switching point.

To facilitate the study of this Question administrations are invited to
submit contributions on the following points:

The preferred speech level at 0 dB RLR as a function of the circuit
noise level.

The actual mean active speech level and standard deviation at a 0 dBr
international switching point.

The mean active speech levels used for speech announcements in their
networks referred to a 0 dBr switching point.

============================================================
Question 18/XII - Transmission performance of digital systems
(continuation of Question 18/XII studied in
1985-1988)

Considering

(a) that many new types of digital processes are being considered for
introduction into telephone networks resulting in mixed analogue/digital
connections;

(b) that distortions (e.g. quantization, circuit noise, etc.) due to
these processes can significantly affect overall transmission
performance of telephone connections;

(c) that the MNRU (Modulated Noise Reference Unit) of Recommendation
P.70 has been adopted as the reference system in terms of which the
transmission performance of digital processes for telephony should be
expressed;

(d) that Supplement No. 14 provides guidance on the use of the MNRU in
conducting subjective tests using the system of Recommendation P.70; and

(e) that methods of Supplement No. 14 are most suitable for evaluating
the relative transmission performance of waveform digital processes
(e.g., PCM, ADPCM),

asks the following Questions:

1. What changes can be made in Recommendation P.70 and Supplement No.
14,

a) to improve their application for waveform digital processes intended
for telephony (e.g. synchronous tandeming of PCM and ADPCM systems);

b) to enable their application for wideband processes (e.g. 7 kHz
programme) and low bit rate (< 16 kbit/s) processes;

c) to enable their application for non-waveform digital processes.

2. In what terms (for example QN and QW determined using the system of
Recommendation P.70) should the subjective effect of distortion of
digital processes be expressed and what magnitude should be recommended
for different digital processes to ensure satisfactory transmission
performance?

3. What objectively measured values can be recommended for different
digital processes?

Note - Objective methods are being studied under Question 13/XII.
Attention is also drawn on Question 14/XII.

============================================================
Question 19/XII - Recommended values for loudness ratings
(continuation of Question 19/XII studied during the 1985-88 study
period; new wording)

Considering

that transmission planning based on loudness ratings will be widely
adopted for networks which are growing increasingly complex,

the following Question should be studied:

What changes and additions involving loudness ratings are necessary in
the G-Series Recommendations, especially in G.111 and G.121, in order to
ensure an adequate overall transmission quality?

============================================================
Question 20/XII - Wideband telephony
(New Question)

Considering

(a) that current telephone connections generally provide a bandwidth of
about 300 to 3 400 kHz;

(b) that this bandwidth results from historical decisions regarding
efficient design of telephone set transducers and circuits, analogue
lines, signalling apparatus, and time and frequency division multiplex
carrier systems;
(c) that as the telecommunications network evolves to an end-to-end
digital transmission capability many of the elements limiting the
bandwidth to 3.1 kHz will disappear;

(d) that the CCITT has approved a Recommendation G.722 "7 kHz Audio
Coding within 64 kbit/s" by accelerated procedures;

(e) that G.722 is a unique, world-wide standard for coding 7 kHz at 64
kbit/s;

(f) that subjective tests have shown that G.722 is markedly superior to
G.711 A-law or m5 -law PCM coding at the same 64 kbit/s rate;

(g) that end-to-end digital communications using 64 kbit/s G.722 ADPCM
would utilize the same switching and transmission capacities as 64
kbit/s G.711 PCM;

(h) that the bandwidth of a voice connection in an end-to-end digital
connection will be determined by the transducers, codecs and analogue
circuitry in the telephone sets;

(i) that low cost, small, wide bandwidth, high quality receiver and
microphone transducers are available;

(j) that current handset and hands free telephone set transmission
objectives generally provide for a 300 to 3 400 Hz bandwidth;

(k) that current methods and instruments for objectively measuring
handset and hands free telephones generally may be limited to the same
bandwidth, and that;

(l) that the existing body of subjective and objective methods for
assessing the performance of telephone connections are based upon the
use of 3.1 kHz telephones,

it is proposed that the CCITT study:

1. how existing definitions of transmission parameters used to define
the performance of telephone connections, e.g., loudness ratings, echo
return loss, circuit noise, etc., should be revised to provide for an
increase in the bandwidth from 3.1 kHz to 7 kHz;

Note 1 - The transmission performance of 3.1 kHz bandwidth digital
telephones is studied under Question 10/XII, and the corresponding test
methods are studied under Question 30/XII.

2. how the existing transmission objectives for handset and hands free
telephones should be revised to accommodate wideband operation;

Note 2 - Methods for determination of loudness ratings are studied under
Question 8/XII.

3. how the existing subjective and objective methods and instruments for
measuring the transmission performance of telephone sets should be
revised to accommodate wide bandwidth;

4. how the existing subjective and objective methods for assessing the
performance of telephone connections should be revised to accommodate
wideband operation, and

Note - See Recommendation P.70 which incorporates requirements for a 7
kHz MNRU to be used when testing wideband digital processes.

5. the need for new subjective and/or objective tests designed to
establish the degree of improvement in performance of connections
employing wideband telephones as compared to connections using
conventional sets.

============================================================
Question 21/XII - Relative level at the boundary between national
systems and the international chain
(New Question)

Considering

- that the Blue Book (Recommendation G.121) uses both a 0 dB point and
the conventional VASP as a reference point for sending and receiving
loudness ratings, the following Question should be studied,

Should Study Group XII convert all documentation to use the 0 dB point
as the reference on the international chain rather than the present
VASP?

If so, what changes will be needed to the G-Series Recommendations?

============================================================
Question 22/XII - International telephone conference
(continuation of Question 32/XII studied in 1984- 1988)

Recommendations E.151 and G.172 contain relevant information with
respect to the network arrangements and equipment used to establish
voice telephone conferences. Considering these existing Recommendations.

What should be recommended in respect of transmission performance to
enable international telephone conference calls to be satisfactorily
accomplished?

In particular,

a) What transmission characteristics should be recommended in respect to
the bridging equipment to be used?

b) What transmission restrictions should be recommended for the other
facilities used in connections for international conference calls?

c) What considerations should be given with respect to the transmission
of non-speech signals (e.g. voice band data, facsimile, graphics, slow
scan television, etc.) on international telephone conference
connections?

Note 1 - Characteristics of interest for the bridging equipment include:

- insertion loss;

- automatic gain control;

- features for control of stability and echo;

- features for control of noise accumulation;

- special requirements for sending non-speech signals.

Note 2 - In Part B, consideration should be given to:

-application rules for using switched gain (loss) devices (e.g., loud-
speaking telephones, echo suppressors, bridging equipment, etc.) in
tandem;

-the relationship between the transmission performance of customer
premises and bridging equipment;

-the use of satellite circuits;

-the consequences of multiple star conference connections, especially
those employing more than a single satellite circuit.

============================================================
Question 23/XII - Coupling of hearing aids to telephone receivers
(continuation of Question 23/XII studied in 1984-88; new wording)

Considering

(a) that due to deregulation on the telephone set market many
administrations and license giving bodies have a strong need for methods
to evaluate how signals are transferred from a telephone set to a
hearing aid;

(b) that in real life very often magnetic or acoustic coupling
techniques are used;

(c) that there is a growing interest in magnetic loop systems;

(d) that in-the-ear hearing aids are becoming increasingly available;

recommends that the following should be studied:

-What parameters should be defined to describe the coupling?

-How are these parameters measured?

-What values or intervals of the parameters can be recommended?

-How can Recommendation P.37 be extended to include the essential
information on description of couplings, measuring methods and
recommended values for important parameters to ensure that hearing
impaired subscribers can communicate using the telephone network?

============================================================
Question 24/XII - Integration of mobile systems into the public
switched network
(merging of Questions 24/XII and 33/XII. Continuation of Question 24/XII
studied during the 1984-88 study period, new wording)

Considering

(a) that the number of automatic land mobile and maritime mobile systems
connected (directly or through a national telephone network) to the
international telephone network, will increase;

(b) that speech processing equipment, such as automatic speech volume
control, companders, low-bit-rate coding, etc. used in some mobile
systems, may interact adversely with equipment in the international
telephone network;

(c) that for technical and economical reasons, mobile systems will not
conform to existing CCITT Recommendations;

(d) that existing CCITT Recommendations may be insufficient to ensure
satisfactory transmission quality of international connections including
mobile systems,

the following Questions should be studied:

1. What Recommendations should be given for

- loudness ratings;

- quantizing distortion;

- echo;

- one-way delay;

- stability;

for mobile systems?

2. What assessment method should be used and what values should be
recommended for noise in mobile systems?

3. Aspects on non-speech signal transmission.

4. Requirements of special service like roaming.

Note - A preliminary draft Recommendation G.173 is given in Supplement
No. 30, Volume III.1 of the Blue Book.
ANNEX 1

(to Question 24/XII)

Reply given to the Question in Report COM XII-R 30,
section 3.4, February 1988
ANNEX 2

(to Question 24/XII)

Reply given to Question 33/XII in Report COM XII-30,
section 9, February 1988

ANNEX 3

(to Question 24/XII)

Documentation received during the 1985-1988 period
1.COM XII-R 6-E, pages 8-19

2.COM XII-R 13-E, pages 9-14

3.COM XII-R 18-E, pages 11-15 and 106-108

4.COM XII-68 (European Joint Experts Group on low-bit-rate coding for
mobile radio)

Subjective testing methodology for the evaluation of low-bit-rate codes
for mobile radio.

5.COM XII-73 (British Telecom)

Delay aspects of low-bit-rate speech coding below 32 kbit/s

6.COM XII-79 (Norway)

Some considerations on medium-bit-rate speech coding for digital mobile
radio.

7.COM XII-94 (LM Ericsson)

Delay in low-bit-rate codecs.

8.COM XII-97 (British Telecom)

Considerations of echo caused by the acoustic loss path of telephone
sets.

9.COM XII-129 (Study Group XI)

Coordination of work on the mobile systems between CCITT Study Groups.

10.COM XII-134 (Study Group XV)

Reply from Working Party XV/2 to Working Party XII/3 on companders.

11.COM XII-147 (Swedish Administration)

Report on subjective test on candidate codecs for mobile radio.

12.COM XII-165 (Norway)

Assessment of talker echo caused by the acoustic echo-path in a 4-wire
telephone set. Results from a conversational experiment.

13.COM XII-183 (NTT)

Transmission speech quality in land mobile radio.

============================================================
Question 25/XII - Transmission impairments in the evolving mixed
analogue/digital and ISDN networks
(continuation of Question 25/XII studied in 1985-1988; revised wording)

Considering

(a) that the telephone network will be a mixture of analogue and digital
circuits and exchanges for a significantly long period before it becomes
wholly digital;

(b) that ISDN and mixed ISDN-PSTN connections are expected to become
available in the next few years;

(c) that increasingly sophisticated digital processes (codecs, DCME, and
packet voice, etc.) are expected to be used in the mixed
analogue/digital, ISDN and ISDN-PSTN connections;

(d) that these processes will introduce impairments (e.g., quantizing
distortion, speech clipping and delay) that will impact on the quality
of the services carried on these connections;

(e) that an increase can be expected in the number of sophistication of
non-speech services over the switched telephone network in advance of
the introduction of extensive integrated services digital network, and
that such non-speech services could be affected by the increased
impairments,

the CCITT has decided to study:

1. What changes are necessary to the hypothetical reference connections
to model the changes and effects foreseen in the consideranda?

2. What Recommendations can be made to control the effect these changes
could have on telephone services including non-speech services carried
by the switched telephone network?

Specific points for particular study are:

a) the effect of introducing digital processing techniques for telephony
other than those described in Recommendations G.711, G.721, G.722 and
G.72Z. For example:

-16 kbit/s codecs;

-non waveform codecs;

- DCME (Digital Circuit Multiplication Equipment);

- packetized voice systems.

3. What impairments are introduced by the new digital processes and what
can be recommended concerning allowable level of these impairments on
connections conveying speech, voice band data and other non-speech
signals?

b) the use of circuits for non-telephony applications including
alternative use of an established connection for various non-speech
services. For example:

- voice band data transmission;

- Telefax;

- Videotex.

c) Recommendation G.113 provides for speech services information on the
number of units of quantizing distortion produced by various digital
processes. Can additional processes be identified and, if so, can the
appropriate number of units of quantizing distortion be specified?

d) can a planning rule (similar but not equal to a qdu) be identified
which indicates the impact of a digital process on voice band data and
other non-speech signals. Note in G.113, section 4, these important
points concerned with this aspect of the Question:

"From the point of view of developing a simple planning rule which can
be used to assess the effects of digital processes on voice band data
performance, several points are important:

1) Impairments (especially transients) other than those customarily
measured for speech performance are important for measuring voice band
data performance.

2) A simple measure of the steady-state impairments (e.g., signal-to-
total noise ratio) may not prove to be a satisfactory basis for a voice
band data planning rule. A planning rule may have to take the transient
impairments into account.

3) Modem type and speed must be taken into account. Thus, unlike the
planning rules for speech, rules for voice band data may turn out to be
modem-specific.

4) The type of data service may influence the extent to which certain
kinds of data errors and thus certain impairments are important. Thus
the planning rules may be service- specific.

5) Only an objective measurement method taking these first four points
into account is likely to provide a successful basis for deriving useful
planning rules.

6) Such a measurement method does not exist at present."

e) Can the planning rule in G.113, which is now applicable only to mixed
analogue/digital connections, be extended to cover ISDN and mixed ISDN-
PSTN connections?

f) Can the planning rule in G.113 be revised to incorporate the effects
of bit errors on the qdu assignment made to digital processes and of all
noise type impairments (total noise) (e.g., quantizing, circuit, inter-
modulation, impulse, etc.) and, if so, what additivity laws for the
impairments apply, how is the total noise allocated to portions of the
connections, and what are the end-to-end objectives for the total noise?
(Note that relevant information on this part of the Question is
contained in COM XII-221, COM XIIR 29 and in COM XII-185. Note also
Recommendations Q.551 and Q.553 from Study Group XI may contain helpful
information concerning this study and that Study Group IV is studying
similar questions and should be kept advised of our work.) (Note that
the effect of delay and delay allocation rules are being studied in
Question 27/XII.)

g) What is the effect of frame slips and random bit errors on speech and
voice band data signals?

h) What Recommendations can be made with regard to planning rules for
the introduction of DCME and packetized voice systems in the world-wide
networks.

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Question 26/XII - Setting objectives for mixed analogue/digital circuits
(continuation of Question 26/XII studied in 1985-1988; reviewed wording)

Considering

(a) that there exists a draft Recommendation G.1xx;

(b) that still for a long time there will exist mixed analogue/digital
circuits on international and national networks during the period of
transition to all-ditigal networks;

(c) that mixed analogue/digital circuits may have a complex structure
and they may be composed by using PCM equipment and group codecs
(G.795), transmultiplexers (G.793, G.794), modems (G.941, V.37),
transcoders (G.761), equipment with different coding methods other than
PCM ones;

(d) that the existing draft text of Recommendation G.1xx sets objectives
for mixed analogue/digital circuits formed by using only PCM equipment;

(e) that the transmission quality of mixed analogue/digital channels
should not be impaired considerably,

the CCITT has decided to study the following Questions:

1. What parameters should be considered for mixed analogue/digital
circuits of different structures, using various analogue/digital
conversion methods, and with digital telephone sets on each and as well
as with a digital telephone set on one end and an analogue telephone set
on the other end?

2. What numerical values of the parameters should be set including
planning rules?

3. What rules of addition of impairments should be used depending on the
type of analogue/digital conversion?

4. What new Recommendations on objectives for mixed analogue/digital
circuits should be drawn up?

5. What additions and amendments should be introduced into the existing
CCITT Recommendations?

6. How should draft Recommendation G.136 (Supplement No. 29, Volume
III.1 of the Blue Book) be completed, using the study results of other
Questions?

Note 1 - Study of this Question is interest to Study Groups IV, XII and
XVIII.

Note 2 - The problem of the "total noise design" should also be studied
with regard to the addition law of the "circuit noises (analogue)" and
the "MOS equivalent to noise due to quantizing distortion".

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Question 27/XII - Talker echo, propagation time and stability in
telephone networks, ISDN and interconnection with ISDN
(merger of Questions 6/XII and 27/XII studied in 1985-1988; new wording)

How should Recommendations be revised to take account of transmission
planning aspects of talker echo, propagation time, and stability in
telephone networks?

Specific points for particular study are:

a) the effect of propagation delay on transmission performance under
varying echo conditions;

b) the allocation of the maximum values of mean one-way propagation
times specified in Recommendation G.114 among international circuits and
national extensions;

c) the updating of Table 1/G.114 to include delay values for new devices
or systems;

d) echo control in ISDN and PSTN connections and in ISDN-PSTN
interconnections;

e) clarification of the stability clauses in G.122;

f) echo control in private network-PSTN interconnections;

g) echo control for packetized speech;

h) echo control for connections with non-linear or non-stationary
conditions in the echo path, e.g. low rate encoding, adaptive balancing,
etc.;

i) the adequacy of the values in Table 3, Supplement No. 2, for circuits
with short propagation times;

j) the effect of irregular sensitivity/frequency characteristics on the
calculation of echo loss according to G.122.

Note 1 - Recommendation G.114 deals with mean one-way propagation time.

Recommendation G.122 deals with the influence of national networks on
stability and echo in international connections. Recommendation G.131
deals with stability and echo in an international connection.

Recommendations G.161, G.164 and G.165 deal with echo control devices.

Supplement No. 2 gives the basis for talker echo control on
international connections.

Note 2 - Echo control may be required in wholly 4-wire connections (e.g.
ISDN) due to echo through acoustical feedback paths. If control is
required, it is necessary to determine whether echo cancellers
(conforming to Recommendation G.165) are effective in controlling the
echo caused by acoustical feedback in the handset of digital telephones.

Note 3 - It will be necessary to define the test conditions of any
experimental investigations of the effect of echo and/or delay on
transmission performance. Annex 1 provides details. It would be
desirable that subjective tests be conducted in a manner to facilitate
the extension of the existing opinion models.

Note 4 - Study Group XI should be kept informed of progress in
clarification of the stability clauses of Recommendation G.122.

Note 5 - Study Group XV is developing Recommendations for acoustic echo
controllers and should be kept informed of the progress on this
Question.

Note 6 - Study Group XVIII should be kept informed of progress in echo
control for ISDN connections, ISDN-PSTN interconnections and for
packetized speech.

Note 7 - Annex 2 provides information on the ISDN-PSTN interworking echo
control issues that are of concern to Study Group XVIII. COM XII-204 is
also relevant. (Note to Secretariat - For Annex 2 use COM XII-203).

Note 8 - Useful information on the concerns of Study Group XI, and on
the technical issues involved in clarification of the stability clauses
in Recommendation G.122 is contained in COM XII-211, and COM XII-215).

Note 9 - Other relevant information to the study of the Question can be
found in COM XII-R 5, R 12, R 17, R 29.
ANNEX 1

(to Question 27/XII)

Annex 2 to Question 6/XII in COM XII-1, 1985-1988 study period


ANNEX 2

(to Question 27/XII)

COM XII-203, 1984-1988 study period

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Question 28/XII - Listener echo (receive end echo) in the public
switched telephone networks
(continuation of Question 28/XII studied in 1985-1988)

How should Recommendations be revised to take account of transmission
planning aspects of listener echo (receive end echo) in the public
switched telephone networks?

Note 1 - Listener echo can cause bit errors in data transmission over
telephone networks. Listener echo in voice transmission is mainly
exhibited by hollowness. The increased use of four-wire equipment, such
as digital exchanges and transmission equipment, in telephone networks
may increase the occurrence of listener echo unless proper measures are
taken. In particular, the number of 2w-4w-2w loops in a connection, and
the loss in each loop, will influence the listener echo received at the
far-end of the connection.

Note 2 - Recommendation G.122 contains information on listener echo and
the number of loops in a connection. The adequacy of this information
and any additional information or planning rules need to be studied.

Note 3 - Supplement No. 3 and Annex 2 provide useful information on
listener echo.
ANNEX 1

(to Question 28/XII)

COM XII-173, 1984-1988 study period
ANNEX 2

(to Question 28/XII)

COM XII-208, 1984-1988 study period

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Question 29/XII - Transmission plan aspects of the interworking between
PSTN and ISDN in the evolving network
(continuation of Question 29/XII studied during the 1985-1988 study
period; new wording)

Considering

(a) that digital telephones, PBXs, remote line concentrators, local and
main exchanges are either already in service or are planned for the
PSTN;

(b) that intense planning is underway for ISDNs and trials are occurring
in some administrations;

(c) that the mixed analogue/digital PSTNs must interwork with ISDNs to
preserve the objectives of the overall transmission plan,

The CCITT has decided to study the following Questions:

What new Recommendations or changes to existing Recommendations must be
made concerning PSTNs or connections to PSTNs to ensure that voice and
voice band data interconnection between mixed analogue/digital PSTNs and
ISDNs comply with the objectives of the transmission plan?

Voice or voice band data connections may be established from digital
telephones or modems in the ISDN and terminate in the conventional
manner in a PSTN or in a private network subtending the PSTN. On an end-
to-end basis the objectives of the transmission plan must be satisfied
with regard to loss/level and the control of stability and echo.

Specific points for study are:

a) For digital interconnection below a digital local exchange in a PSTN,
what should be the location, means of control and range of values of
loss insertion devices taking into account interworking with ISDNs?

b) What special considerations concerning noise and level must be taken
into consideration in interconnecting PSTN and ISDN or in providing
voice service on ISDN?

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Question 30/XII - Methods for evaluating the transmission performance of
digital telephone sets
(continuation of Question 38/XII studied in 1984-88)

Considering

(a) that digital telephone sets are already under development for public
and private extension networks, and are beginning to appear on the
market;

(b) that the desirable transmission performance characteristics of
digital telephone sets are under study in Question 10/XII;

(c) that current digital telephone sets are based on the use of 64
kbit/s PCM encoding (A or m5 -Law), and other encoding schemes are
currently being used or developed by some administrations;

(d) that only a provisional Recommendation P.66 exists for verifying
their performance characteristics,

The CCITT decides to study the following Questions:

1. What further methods can be specified for evaluating the transmission
performance of digital telephone sets to complete the Recommendation
P.66 and how should it be amended to meet the needs for future
development?

2. What digital access point and interface should be used for testing
purposes?