110857247 3 GSM Speech Quality Influence Factors Troubleshooting Methods and Tools Deliverables 20110730

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HUAWEI TECHNOLOGIES CO., LTD. www.huawei.com Huawei Confidential 英文标题:40-47pt 副标题:26-30pt 字体颜色:反白内部使用字体 : FrutigerNext LT Medium 外部使用字体 : Arial 中文标题:35-47pt 字体:黑体副标题:24-28pt 字体颜色:反白

字体:细黑体

Prepared by: GUL Network I&V and Maintenance

Department

Reviewed by: Qi Haofeng

GSM Speech Quality:

Influence Factors + Troubleshooting Methods and

Tools + Deliverables

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This document mainly discusses the main factors that affect the speech quality

of a GSM network, principles of improving the functions related to speech

quality, and suggested values of some key parameters. In addition, this

document lists the deliverables (see the attachments) that field engineers

should submit when reporting speech quality problems or evaluating the

speech quality, including drive test information, counter information, and

guides to related tools (see the operation guide). This document aims to quickly

locate and solve speech quality problems and to evaluate the speech quality

and prevent speech quality problems based on the collected information about

the existing network.

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R&D Support

For speech quality problems, we can provide trainings and 7x24 hour technical support.



List of R&D support engineers

Name

Employee ID

Phone

Yang Zhengjie (Wireless Network)

00127669 See the phone book.

Yang Chunjie 00119951 See the phone book.

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• Evaluation Standards and Principles of Speech

Quality (MOS)

• Statistics and Analysis of Factors Affecting the MOS

• Subjective Speech Problem Handling

• Voice-Related Key Parameters:



Quality Parameters



Codec Parameters



Handover Parameters



AoIP Parameters

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Subjective Evaluation

This method indicates that many people compare the original voice sample with the degraded file

processed by the system by their own subjective perceptions, then mark the mean opinion score

(MOS) (ITU-T 800) value (full score: five points), and finally obtain the average value.

Objective Evaluation

This method indicates that the score is obtained through comparing the degraded voice file after

transmission with the original voice sample file by using a certain algorithm, such as PAMS (ITU-T

P 861) and PESQ (ITU-T P 862.1).

Parameter Evaluation

This method indicates that the voice after transmission is not evaluated, and the original voice is

not obtained. Instead, the voice after transmission is evaluated through some parameters of

wireless transmission network, which has a promising prospect in wireless network, such as

RXQUAL, VQI of Huawei, and SQI of E///.

Currently, carriers all over the world treat the speech quality as the key indicator for network

acceptance. Among them, PESQ algorithm is the widely-used scoring standard. In this algorithm,

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PESQ: Perceptual evaluation of speech quality

This method is used for E2E network speech quality test. It is to compare the

original voice sample on the transmitting end in the network (narrowband) with

the distorted degraded voice file received on the receiving end, evaluate the

difference between the two signals through complex signal processing, and

finally obtain the speech quality value using the PESQ algorithm.

After the PESQ algorithm is processed, the following four metrics are obtained:

• PESQ RAW SCORE

(the raw score)

• P.862.1 (the score is obtained through the P.862.1 mapping mode based on the raw score)

• PESQ-LQ (the score is obtained through the Psytechnics mapping mode)

• PESQ-Ie (The score is obtained through the mutilation factor of instrumental models defined by

P.834)

Among them, the value of

P.862.1 is widely regarded as the reference value in voice

evaluation.

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

Evaluation Standards and Principles of Speech

Quality (MOS)



Statistics and Analysis of Factors Affecting the

MOS



Subjective Speech Problem Handling



Voice-Related Key Parameters:



Quality Parameters



Codec Parameters



Handover Parameters



AoIP Parameters

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HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential 颜色: R153 G0 B0 内部使用字体 : FrutigerNext LT Medium 外部使用字体 : Arial 中文标题:30-32pt 颜色: R153 G0 B0 字体:黑体英文正文:20-22pt 子目录 (2-5级) :18pt 颜色:黑色内部使用字体 : FrutigerNext LT Regular 外部使用字体 : Arial 中文正文:18-20pt 子目录(2-5级):18pt 颜色:黑色字体:细黑体配色参考方案：建议同一页面内不超过四种颜色，以下是 13组配色方案，同一页面内只选择一组使用。（仅供参考）

Statistics and Analysis of Factors Affecting the MOS

Voice quality Voice quality

Code

Code Bit error (frame

erasure) Bit error (frame

erasure) HOHO Directly-related factors Directly-related factors Indirectly-related factors Indirectly-related factors T ra ffi c T ra ffi c F u ll/ h a lf ra te F u ll/ h a lf ra te H ig h /lo w c o d in g r a te H ig h /lo w c o d in g r a te A ir In te rfa c e Q u a lit y A ir In te rfa c e Q u a lit y T ra ffi c B u s y T h re s h o ld T ra ffi c B u s y T h re s h o ld R a te a d ju s t T h re s h o ld R a te a d ju s t T h re s h o ld F 2 H H O T h re s h o ld F 2 H H O T h re s h o ld T h re s h o ld S e lf-A d a p tiv e T h re s h o ld S e lf-A d a p tiv e S p e e c h v e rs io n S p e e c h v e rs io n Frequent HOs, PingPong HOs, and unreasonable HOs Frequent HOs, PingPong HOs, and unreasonable HOs T o o lo w P N R u le T o o lo w P N R u le In a p p ro p ria te N e ig h b o rin g C e ll In a p p ro p ria te N e ig h b o rin g C e ll T o o S m a ll H O H y s te re s is T o o S m a ll H O H y s te re s is F ra m e T h e ft o f P h y s ic a l M e s s a g e s F ra m e T h e ft o f P h y s ic a l M e s s a g e s Parameters, algorithms, and optimizing strategies Parameters, algorithms, and optimizing strategies

The channel is normal. The channel is normal.

In te rfe re n c e In te rfe re n c e A lg o rit h m s A lg o rit h m s E n g in e e rin g n e tw o rk o p tim iz a tio n E n g in e e rin g n e tw o rk o p tim iz a tio n 3 .5 -G e n e ra tio n P o w e r C o n tr o l 3 .5 -G e n e ra tio n P o w e r C o n tr o l D T X D T X V A D V A D A n ti-In te rfe re n c e s o lu tio n A n ti-In te rfe re n c e s o lu tio n In te rm o d u la tio n In te rfe re n c e Q u ic k T ro u b le s h o o te r In te rm o d u la tio n In te rfe re n c e Q u ic k T ro u b le s h o o te r T O P O p tim iz a tio n T O P O p tim iz a tio n H O O p tim iz a tio n P a c k e t H O O p tim iz a tio n P a c k e t C a ll d ro p C a ll d ro p L o n g C a ll D ro p T im e r L o n g C a ll D ro p T im e r C o B C C H R e s id e n t S tr a te g y C o B C C H R e s id e n t S tr a te g y D is c a rd e d P a c k e ts C o m p e n s a tio n D is c a rd e d P a c k e ts C o m p e n s a tio n

The speech quality is mainly related to three factors: code, bit error, and handover (HO). The coding factor benefits the speech quality. The bit error and handover factors,

however, damage the speech quality.

To optimize the speech quality, you need to select reasonable codes and reduce the effect of the bit error rate (BER) and handovers on the speech quality.

The prerequisite is that the channel is normal.

The methods for improving the call drop rate and handover success rate usually damage the speech quality and the experience of subscribers. Therefore, strategies that optimize the speech quality may affect the call drop rate and handover success rate.

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The MOS handling process is as follows:

MOS fails to reach the standard.

Make clear the actual situation of the problem (including average MOS or deterioration conditions of the percentage of high scores, testing methods,

and instruments).

001

After the optimization and adjustment, verify whether the

problem is solved through drive test. 001 End Yes No Trouble-shoot the occupation of voice version and encoding

rate.

002

Trouble-shoot the percentage of the number of handover times and that

of MOS dotting.

003

Trouble-shoot the data configuration and transmission quality.

004

Trouble-shoot the TC recording and air interface frame data.

005

Trouble-shoot the air interface quality.

006

Analyze the preceding factors and specify the reasons that lead to the

MOS problems.

007

After the preceding troubleshooting, optimization & adjustment, and verification, if the MOS problems are

still not solved, perform the escalation processing.

008

During the processing, perform the troubleshooting from the easier to the more advanced following the dashed in red.

In existing networks, two methods are available for the MOS acceptance standard: One is that the average MOS for the drive test of the entire network shall reach a value. The other is that the proportion of high scores in the MOS shall be larger than the required value, or the proportion of low scores shall be lower than a certain value, and comparison between the two drive test data (such as migration and version upgrade) shall be performed. No matter which method is used, when the MOS does not reach the standard, troubleshooting is carried out based on factors affecting the MOS in the process.

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1. Problem specifying

 Specify the details about the MOS problem, including the current test value, target value, and the gap between

the two.

 Specify the standard of the MOS appraisal. Specify whether it is MS-MS, or MS-PSTN, whether the MOS

appraisal value is the overall average score or percentage of high scores, and whether the up-link and down-link are appraised separately.

 Contrast the test MOS values by using instruments or terminals, it is found that there is no change.

 Contrast the test time frame (start from what time point and to what time point the test ends), test route, and

test period (to ensure the comparability of MOS tests, the tests shall be performed on the same day in different weeks).

2. Speech version and coding proportion analysis

There is a large difference for the MOS in different speech versions. In normal conditions, the sequence for the MOS baseline performance is: FAMR > EFR > HAMR > FR > HR. For example, the MOS for the EFR in the MS-MS test can be 4.0, whereas that for the HR will be 3.0. Therefore, for MOS problems incurred before and after migration, contrast the occupation proportion of each speech version in the drive test, and check whether the proportion of half rate is increased.

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3. Handover times and proportion analysis

 If the MOS problem occurs before or after migration (upgrade), check whether the ratio between the number

of MOS dotting and that of handover times is changed. The larger this ratio is, the smaller the effect made by the handover on the overall MOS is.

 Analyze whether ping-pong handover exists in the drive test data, or whether the handover times in counters

are too much (usually, the number of handover times in each call is within 1 in existing networks). If such case exists, modify the corresponding parameter configuration to reduce the effect of handover on the MOS. If the PN for the PBGT (better cell) handover is added, and the PBGTSTAT (s) parameter is set to 5s and the

PBGTLAST (s) parameter is set to 4s, the judgment time for handover is delayed, and the handover is

reduced.

4. Comparative analysis of data configuration and transmission

 If the MOS problem occurs before or after migration (upgrade), check whether the data configuration and

transmission mode are changed, for example, whether parameters including cell handover and power control are changed, and whether cells or frequency band is added or reduced. In addition, trouble-shoot the radio frequency channels in the area where problems occur, and check whether KPIs in the traffic transmission are incorrect, which affect the MOS test result.

 For new-built network, check whether alarm information is displayed on the NEs such as the BTS, BSC, and

transmission in the test, which affects the test result. Check whether the network KPIs are abnormal. Usually, KPIs include TCH call drop rate, success rate of wireless handover, and TCH congestion rate.

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5. Analysis on TC recording and Um interface frame capture

Mainly check whether there is any problem on the TC and Probe terminals and the area where the

problem occurs.

6. Um interface quality analysis

For comparison before and after migration (upgrade), analyze based on the quality before and after

migration (upgrade), and check whether quality deterioration exists in all areas or part of areas. If

quality deterioration exists in some areas (including cases that call drop or handover failure occurs

due to poor quality of Um interfaces), perform analysis based on the drive test data, and make

clear whether the poor quality is caused by cases such as interference, poor coverage, and

missing cross and neighboring cells.

MOS

Data Wave Mode Comparison Files

Whether Difference Exists in Wave Mode Comparision

MOS Compared with the Original Sample

TC recording file

Probe file

UpIn vs. UpOut DnIn vs. DnOut

Original sample vs. MS uplink voice data MS downlink voice data vs. Recording (degraded) file

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Voice services are

key services in a

GSM network. The

quality of voice

services is

determined by many

factors. To report

MOS problems or

speech problems,

you need to report all

factors related to

voice during the drive

test.

For details about the

distributed

troubleshooting, see

the Guide to Locating

and Isolating GSM

Speech Problems.

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Good speech quality first depends on good network quality. The quality of the entire network, however, should be evaluated by counters. For details, see the following table.

For more detailed feedback information, see the attachment "Checklist for Data Provided for

Speech Quality Problems".

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In a site, it is required that the average value of MS-MS MOS should be

greater than 3.50 according to the drive test result of the entire network after the

migration. The value, however, is only 3.35 according to the drive test result.

Therefore, the value of MOS fails to reach the standard.

According to the statistics and analysis of the drive test information, the ratio of half rate channels

reaches 75%. This is the main factor that affects the overall MOS. The details are as follows:

Through parameter configuration and counter

analysis, it is discovered that the problem is

caused by that the values of TCH Traffic Busy

Threshold of many cells are set to be too small

(30%). After adjusting the values of this parameter

and performing another drive test, the ratio of

occupied half rate channels is reduced to 47%

and the value of MOS reaches 3.52, which

exceeds the acceptance standard.

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Typical Case 2:

In a site, it is required that more than 95% MS-PSTN MOS values should be greater than 2.7 in the

drive test of the entire network after the migration. According to the drive test result, only 90% MS-PSTN MOS values are greater than 2.7. Therefore, the MOS value fails to reach the standard.

A: According to the analysis, no known speech problem exists in the BSS after the migration. The reason that the MOS value of the existing network fails to reach the standard is that the quality over the Um interface is low, many handovers occur, and the ratio of half rate channels is high.

B. After optimizing the concentric handover parameters, adjusting the Assign Optimum Layer and the Pref.

Subcell in HO of Intra-BSC parameters and their thresholds, and optimizing the number of handovers, the ratio

between traffic and number of successful handovers rises from 68.4 to 71.9. In addition, the ratio between number of MOS values and number of handovers in the drive test rises from 1.86 to 2.76. Therefore, the ratio of MOS values that are grater than 2.7 of the entire network rises about 4%.

C. After optimizing cells one by one and expanding the capacity of busy cells, the ratio of half rate channels in the test is reduced from 46% to about 35%. This improves the overall MOS.

D. After optimizing problem sites one by one and take optimization measures at a low carrier-to-interference ratio (CIR), the ratio of Um interfaces whose quality is at level 0 to level 4 rises 2%.

After taking a series of optimization measures, the MOS value in the drive test is improved obviously. The ratio of MOS values that are greater than 2.7 rises about 10% and reaches over 95%.

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The MOS value of a migrated site is 0.1 less than that of

the original network.



According to the analysis, the smaller MOS value is

mainly caused by abnormal low MOS values. Based on

the problem location and isolation process and the

analysis of the TC recording over the A interface on the

BSC and on the core network, engineers find that the

TC recording is normal on both areas. However, packet

loss occurs on the downlink recording data before the

data enters the A interface and the core network, as

shown in the figure. It is concluded that the PSTN

causes the low MOS value.



The DT data after processing on the PSTN shows that

the MOS values are better than those of the original

network.

Typical Case 3:

The upper figure: Downlink voice sample on the A interface

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

The TFO fails to be established sometimes when the TFO of a Huawei BSC interconnects with that



Upon receiving the speech version of the FAMR from the Ericsson BSC during a TFO negotiation,

the Huawei BSC decides that the TFO frame type from the Ericsson BSC is AMR_TFO_16k and

enters the TFO establishment process normally. However, the Huawei BSC keeps receiving TFO

frames of AMR_TFO_8+8k from the Ericsson BSC, resulting in failures to establish the TFO.



As defined by the GSM protocols, when the HAMR speech version (excluding 7.95 kbit/s) is used

during the TFO negotiation, the TFO frame type must be AMR_TFO_8+8k. Therefore, Huawei

needs to change the frame search mode to resolve the problem.

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

Evaluation Standards and Principles of Speech

Quality (MOS)



Statistics and Analysis of Factors Affecting the

MOS



Subjective Speech Problem Handling



main application methods on the BSS side are speech loopback test

and TC recording. The speech loopback function can define the NE

where the problem occurs, while the TC recording function can

determine whether the problem is from the TC and the specific

changes. For the analysis methods of loopback tests and the TC

recording file analysis, see the attachment Operation Guide for Speech

Tests.

The following mainly introduces the handling methods of subjective

speech problems such as one-way audio, noise, echo, crosstalk, and

voice make-and-break.

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1. One-way audio

One party of the two call parties cannot hear the voice from the peer end, or both the two parties cannot hear the voice on the peer end, which is presented as one-way audio or no audio. When one-way audio occurs, mute also occurs.

Handling process:

 Specify the area scope where the problem occurs and the specific situations. Confirm it is uplink one-way audio (the party

who holds the MS cannot hear any voice, but the one who is on the PSTN side can hear the voice) or downlink one-way audio (the party holds the MS can hear voices, but the one who is on the PSTN side cannot).

 Enable the way audio detection function (confirm whether the current version supports it or not first), analyze the

one-way detection logs of the whole day, and find out the suspicious resources for dialing test. Perform the dialing test on the site where the problem occurs. For detailed dialing test procedures, see the attachment Operation Guide for Speech Tests. During the dialing test, perform TC recording and single user tracking. Perform loopback when the problem reoccurs, and confirm the NE where the problem occurs.

 Analyze the trunk performance measurement of the A interface, and fond out abnormal occupation timeslots (Rules: The A

interface has 31 timeslots in total, while the average busy hour of the 31 timeslots is less than 30s, and the number of timeslots whose average busy hour is less than 30s is at least 28. However, networks charged by second are excluded, which needs special treatment). Combined with specified CIC dialing test, hardware connection of interfaces, and data configuration, check whether there are problems such as crossed pair on the A interface or incorrect connection of lines (In TDM transmission mode, if the E1 line on the A interface is not configured with the SS7 signaling link, or the E1 line on the Abis interface is not configured with the RSL and OML links, the E1 line on the corresponding port is incorrectly connected, or no alarm is generated even if crossed pair are made (as long as it is not suspended). However, when users occupy this port, one-way audio or no audio occurs).

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2. Noise

During the call, abnormal voices such as bubbles, clicks, and metallic sounds occur. When it is at its worst, only noise can be heard and the normal speech cannot be heard completely. Usually, noises can be divided into two types: noise in normal conversation and handover noise. However, noises are mainly caused by bit errors, including bit errors caused by frequency interference, voice processing software, and equipment hardware.

Handling process: Specify the area scope where the problem occurs and the specific situations.

 Select the site where the problem occurs for dialing test. For detailed dialing test procedures, see the

attachment Operation Guide for Speech Tests. During the dialing test, perform TC recording and single user tracking. Perform loopback when the problem reoccurs, and confirm the NE where the problem occurs.

 Check the alarm and transmission connection line of the site where the problem occurs to see whether there is

any looseness or damage. Check transmission indexes in the traffic statistics, for example, whether problems including packet loss, jitter, and too-long delay exist.

3. Speech make-and-break

Speech make-and-break mainly presents like this: there is a sense of pause in the call, and listeners may miss half a word or several words. When the make-and-break is obvious, it may affect the normal conversation.

For the troubleshooting procedures, see the noise handling process.

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4. Echo

Echoes are mainly divided into two categories: acoustics echoes and electrical echoes. Echoes caused by MS calling MS are called acoustics echoes, whereas echoes caused by MS calling PSTN are called electrical echoes.

Handling process:

 Check whether the handsfree function is enabled or the headset mode is used. Then, check whether the echo

is disappeared or lowered after the handsfree function is disabled or the volume is lowered.

 Acoustics echoes are usually caused by the noncompliance of isolation of terminals to the protocol

requirements. During the test, adjust the volume of the MS on the peer end. If the echo volume heard on the local end is obviously changed, it indicates that the echo is produced by the MS on the peer end. You can change another MS for re-test.

Usually, acoustics echoes are strongly relevant to MSs. The solution to acoustics echoes: Enable the AEC function on the BSC side to help MSs to further eliminate echoes.

 Electrical echoes usually caused by configuration or engineering problems. For example, the call routing data

configuration is incorrect, hybrid coils on the fixed network side do not meet the relevant telecom standards, and the produced echo volume exceeds the processing capability of the echo canceler.

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5. Crosstalk

In the call, not only the speech of the called party can be heard, but also a third-party speech can be heard, or the speech of the called party cannot be heard, instead, a third-party speech is heard. The common reasons for

crosstalk are Um interface crosstalk, core network crosstalk, incorrect data configuration, and abnormal connection.

Handling process:

 Encrypt the Um interface: Enabling the Um interface encryption is the root solution to Um interface crosstalk.  In BSS data configuration, the configured value of the T3109 timer must be larger than the value configured in

RLT.

 When the MSC equipment is not Huawei equipment, enable the Call Re-establishment switch on the BSC

side, and set the call re-establishment timer to 45s.

 Record information such as routing, equipment resources, and transmission about each crosstalk and

analyze them one by one. If it is found that all crosstalks occur in long distance or cross-network (a China Mobile subscriber calls a China Unicom subscriber) calls, basically it can be concluded that the crosstalk has something to do with the core network, and the core network engineers need to participate in the fault location.

 According to the customers' complaint information, draw the CDR from the MSC and find out the

corresponding CIC to perform the designated dialing test to check whether the CIC timeslot appears regularly. If it is regularly appears, trouble shoot the hardware connection or data configuration. Meanwhile, check the data configuration and E1 connection of the problem points.

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The table on the

right side lists the

information on

handling subjective

speech problems. It

should be collected

and submitted after

the completion of

the test.

For details, see the

attachment

Checklist for Data

Provided for Speech

Quality Problems

(deliverables).

Subjective Speech Problem Handling

No. Feedback on Speech Quality Problems Output Description

1

Detailed descriptions of the problem, including the scenario in which the problem occurs and the probability that the problem occurs

For example, record whether noises periodically occur (namely whether the noise occurs once every x second (s) or every x minute (s)), and whether noises persist during the calls.

2 Tracing signaling of a single user and descriptions of the

calling and called MSs

For example, the information about the TEMS of the calling MS is as follows: The MSISDN is 13913140397 and the IMSI is 460512300000397.

3 Log data about the TEMS test Record the test log data when the problem occurs.

4 TC recording files and Um interface frame data captured

by the Probe

The files and data need to be configured before the test. Probe is a test tool of Huawei and is used to capture the information about the Um interface frames. The TC recording files are in the format of *.dat.

5 Loopback test results about the problem List of loopback test results of interfaces (see the

following table))

6

BSS version information and information about core network vendors, configuration data and information about test sites

For BSC6900V900R011C00SP720, it is a CME configuration file and a MML configuration file in the format of *.txt. For versions earlier than BSC6900V900R011C00SP720, it is a *.dat file. For the BTS3012, you need to specify whether the new DTRUs or old DTRUs are used. For the BTS3900, you need to specify the types of TRXs.

7 Information about the engineering parameter table within

the test area

The information is in the format of *.cel that is supported by the Nastar and TEMS.

8 Transmission mode on the entire network The transmission mode is all-TDM, all-IP, or hybrid.

9 BTS log, alarm log, one-way audio log, and related alarm

information about the problem site Logs and alarm information

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A user complains that the speech quality is bad under a BTS at a site. The case is that everything is normal in the outdoor coverage area of this cell, but uplink quality make-and-break and noise occur in the indoor coverage areas of this BTS.

A. Perform TC recording and make analysis at the problem site, and find that the voice make-and-break exists before the voice enters into the TC. For

the speech frame structure at the break-and-make points, the corresponding frames are all No-Data frames, namely, one or more No-Data frames

appear before two continuous speech frames without the transition of

SID frame. This usually is caused by the failure in BTS decoding.

B. During the test at the problem site, the uplink noise is extremely serious and continuous. Through the TRX loopback, the calling party can hear its own speech with break, it demonstrated that the voice make-and-break problem exists between the Um interface and the BTS DSP.

C. According to the test log analysis, it is concluded that the quality of the Um interface in the area where the problem occurs is poor, and the proportion of uplink quality 5, 6, and 7 is 64%, which is the major cause for voice break-and-make.

After the adjustment of optimization measures for uplink low CIR, the speech problem on this site disappears after several times of verification.

RxQual 0~4 5~7 RxLev AVG

Whole 90.82% 9.18% -79.7

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Typical Case 2:

A leader of a customer of a office complains that one-way audio exists in a certain probability during the call, and asks Huawei to solve the problem as soon as possible.

A. Perform the dialing test at the problem site, and perform speech loopback when the problem reoccurs.

 After the calling party A (external network) has a conversation with the called party B (under the problem site), A cannot hear B, which is

uplink one-way audio.

 Enable the remote loopback of the A interface on the B side, A can hear his/her own voice, indicating that the problem does not exist on

the MSC side or on routing nodes after the A interface. Enable the local loopback of the A interface on the B side, B cannot hear his/her own voice, indicating that the problem exists on routing nodes before the A interface on the B side.

 Enable the BTS speech loopback, B can hear his/her own voice, indicating that the problem exists between the Abis interface and the A

interface, namely the BSC.

B. Analyze the TC recording file, and find that when the one-way audio occurs, the call works properly when the uplink voice data enters into the TC. However, when the uplink voice data goes out of the TC, no voice data is available.

Upon analysis, it is concluded that the BTS sends abnormal frames, which leads to the TC scheduling

memory error, leading to one-way audio. This problem is solved after the code optimization.

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Typical Case 2:

A user of an overseas office complains that one-way audio and no audio problems occur with a high reoccurrence probability. Location process:

Because the Abis interface of this office adopts the HDLC transmission, and does not support the one-way audio detection function, the one-way

audio detection cannot be enabled. Upon the analysis on the trunk performance measurement of the A interface, the average occupied duration of

the three ports under the three BSCs is less than 30s, which is far below the average occupied duration of all BSCs. The distribution of occupied

duration has obvious time intervals. In addition, there are more than 28 timeslots that the average occupied duration is less than 30s for each port.

Perform CIC dialing test for specified A interface, the one-way audio occurs. Check the transmission, and find that the E1 lines of the A interface on the three ports are incorrectly connected. After the transmission is adjusted, the one-way audio does not reoccur.

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HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential 颜色: R153 G0 B0 内部使用字体 : FrutigerNext LT Medium 外部使用字体 : Arial 中文标题:30-32pt 颜色: R153 G0 B0 字体:黑体英文正文:20-22pt 子目录 (2-5级) :18pt 颜色:黑色内部使用字体 : FrutigerNext LT Regular 外部使用字体 : Arial 中文正文:18-20pt 子目录(2-5级):18pt 颜色:黑色字体:细黑体配色参考方案：建议同一页面内不超过四种颜色，以下是 13组配色方案，同一页面内只选择一组使用。（仅供参考） Page 29 Office Average Call Duration Shortest Call Duration in Normal Cases Number of BSCs Thailand 167 62 3 Chengdu 68 36 49 Shantou 80 44 11 Shijiazhuang 68 32 16 Hangzhou 87 30 63

1. In normal cases, the average call duration of all evaluation offices is more than 60s (Upon the analysis on 136 BSCs, there is no super short call caused by crossed pair).

2. In abnormal cases, the average call duration is less than 29s (Thailand).

3. Upon the analysis on the A interface occupation measurement for 10 BSCs in Nigeria, a large number of trunk occupied durations of the A interface on the port are less than 30s, with the shortest one is 12.69s. This may be relevant to the strategy of charging by second in Africa. Therefore, in Africa, the case that checking the A

interface connection based on the situation that the A interface trunk occupies the super short call may be altered according to the actual situation.

Typical Case 2 (continued)

Thailand Chengdu Shantou Shijiazhuang Hangzhou

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An AoIP office reports that metallic sounds occur in the existing network, consequently, some users ask to cancel the network service, provide cause analysis, and immediately solve the problem.

Analyze the TC recording and Probe file, there is no noise in the uplink UM interface voice on the calling side and in the uplink voice before entering into the core network on the BSC side. However, after the

voice passes through the core network, and when enters into the BSC downlink (the called side uses the EFR speech version), the noise appear. This, as a result, can be determined that the noise is caused during the processing of core network.

The core network confirmed that in some cases, the DSP cannot complete the call processing with 20 ms, and need to re-process it 120 ms later. In the 120 ms, the DSP will send the previous data again and again, causing the metallic sounds (which is complained by users) acoustically. After the core network engineers optimize the scheduling algorithm of the internal DSP, re-test the problem

message on site.

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

Evaluation Standards and Principles of Speech

Quality (MOS)



Statistics and Analysis of Factors Affecting the

MOS



AoIP-related parameters

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Parameters

For details about the mapping versions that support voice-related features, see the Reference List of

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

Description of the Enhanced Interference Cancellation Combining (EICC) Function:

Among signals received by dual antennas, the interference is related to both the

space (between antennas) and the time. The EICC function considers both the

relationship between interference and space and the relationship between interference

and time. In this way, it suppresses interference more effectively and improves the

voice quality.

Suggested Parameter Settings:



STIRC Allowed: Yes when serious interference exists.

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VQE: mainly includes four sub-features, such as AEC, ALC, ANR, and ANC.

Description of the Acoustic Echo Cancellation (AEC) Function

This function cancels the acoustic echo generated during the call. It determines

whether the signals input by the local and remote ends are echo to the local end. If

the signals received by the local end are echo from the remote end, this function

attenuates the echo and replaces the echo with comfortable noise. If the signals

received by the local end are voice of the speaker, this function keeps the voice

unchanged.

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Description of the Adaptive Level Control (ALC) Function:

This function controls the level automatically. It evaluates the voice level of the input signals and controls

the gain of the input signals. Specifically, it adjusts the output voice signals to the target level and

ensures that the level of the signals is stable and the signals can be understood. Therefore, the hearer

thinks that the volume is proper and has a good experience to the voice.

Suggested Parameter Settings:

Description of the Adaptive Noise Reduction (ANR) Function:

This function is mainly used to reduce the background noise in the voice without damaging the voice. In

this way, it makes the voice acceptable to the hearer.

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Description of the Automatic Noise Compensation (ANC) Function:

This function compensates the noise automatically. It evaluates the level of

the background noise at the local end and the voice level at the remote end.

When the background nose at the local end is great, this function turns up

the volume of the voice input by the remote end. This improves the

signal-to-noise ratio between the voice at the remote end and the background signal-to-noise at

the local end. Therefore, the hearer at the local end can hear the voice of the

speaker at the remote end clearly.

Suggested Parameter Settings:

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Quality Parameters:



Suggested Parameter Settings:



Currently, 3.5-generation power control algorithm is widely promoted globally. It

is required to enable both III power control algorithm switch and III power control

optimization algorithm switch. The 3rd-generation power control algorithm,

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Description of the Counter Function:

Radio Link Timeout: This counter defines the time of radio link connection failure for downlink links. The criterion is that whether the SACCH measurement report can be correctly decoded.

SACCH Multi-Frames: This counter defines the time of radio link connection failure for uplink links. The criterion is that whether the SACCH measurement report can be correctly decoded.

Suggested Parameter Settings:

Note: The suggested parameter values are to end the call in the case that the UM interface quality is bad, avoiding continuous impact of continuous bad quality on the speech MOS values. These suggested values may affect the call drop rate. Therefore, you are advised to use them only when you handle speech

problems. For other KPI handling, see the parameter baselines.

TC CRC Check:

According to GSM specifications, the BSC performs CRC check for each uplink data (TRAU frame) from the BTS. If the TRAU frame fails to pass the CRC check, the BSC regards it as an invalid frame and smoothens it. This avoids the noise caused by parameter

transmission errors and improves the speech quality.



AoIP-related parameters

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Voice-Related Key Parameters – Summary of Codec Parameters

Codec Parameters Parameter Name Commend Value Parameter Description

Speech versions FAMR/EFR/HAMR/FR/HR AMR Adjusts the speech coding mode on the uplink and downlink according to changes in Um interface quality, thereby improving the speech quality.

Enhancement of speech quality

TFO Switch ON Reduces the impact of speech transcoding on the speech quality, improving the speech quality.

RTPSWITCH OFF

Specifies whether to enable the delay function to be implemented between the BTS where the calling MS is located and the BTS where the called MS is located.

TFOOptSwitch OFF When the speech versions on the two sides are inconsistent, establish the TFO after optimization.

RATSCCHENABLED OFF Specifies whether to enable the RATSCCH procedure during a call setup. In the RATSCCH procedure, the rate set of AMR calls can be dynamically adjusted during a call to improve speech quality.

EPLC Switch OFF Compensates the packets that are lost during the transmission.

AMR Uplink Adaptive threshold allowed YES Reduce the impact of inaccurate estimation of Signal-to-Noise Ratio (SNR) or the changes in channel conditions following the time on the Adaptive Multi Rate (AMR).

Voice Quality report switch YES Uses the voice quality index (VQI) to monitor the speech quality on the network in real time. TrFO Switch YES Reduces the impact of TC coding and encoding on the speech quality, improving the speech quality.

Speech Channel Alarm Threshold 10 Specifies the threshold for reporting the speech channel alarm. If the number of one-way audio that occurs in an hour on the BSC exceeds this threshold, the speech channel alarm is reported.

Speech Channel Resume Alarm Threshold 6 Specifies the threshold for reporting the speech channel resume alarm. If the number of one-way audio that occurs in an hour on the BSC is smaller than this threshold, the speech channel resume alarm is reported.

TCMUTEDETECTFLAG ON Specifies whether to enable the class-1 one-way audio detection function.

MUTECHECKCLASS1PERIOD 5 Specifies the class-1 one-way audio detection period. If the FER within the period specified by this parameter exceeds the value of Exceptional Frame Threshold(%), you can infer that one-way audio occurs.

EXCEPFRAMETHRES 25 Specifies the threshold for the proportion of the number of bad frames to the total number of TRAU frames. If the FER exceeds this threshold within the value of Period of Mute Detect Class1(s), one-way audio may occur.

MUTECHECKCLASS2SWITCH ON Specifies whether to enable the way audio and no audio detection function to improve the accuracy of one-way report.

DETECTFRAMEPERIOD 2 Specifies the period for sending the TRAU test frame after the class-2 one-way audio detection function is enabled. One TRAU test frame is sent in each period until the response from the peer end is received or the timer expires.

MUTECHECKPEIROD 4 Specifies the class-2 one-way audio detection period. Channel allocation

CHALLOCSTRATEGY CAPABILITY Allocates the channel with good quality, improving the speech quality.

TCHBUSYTHRES 60 Improves properly the proportion of full rate occupation, improving the speech quality. TCHTRICBUSYOVERLAYTHR 70 Improves properly the proportion of full rate occupation, improving the speech quality. TCHTRIBUSYUNDERLAYTHR 60 Improves properly the proportion of full rate occupation, improving the speech quality.

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110857247 3 GSM Speech Quality Influence Factors Troubleshooting Methods and Tools Deliverables 20110730

Prepared by: GUL Network I&V and Maintenance

Department

Reviewed by: Qi Haofeng

GSM Speech Quality:

Influence Factors + Troubleshooting Methods and

Tools + Deliverables

This document mainly discusses the main factors that affect the speech quality

of a GSM network, principles of improving the functions related to speech

quality, and suggested values of some key parameters. In addition, this

document lists the deliverables (see the attachments) that field engineers

should submit when reporting speech quality problems or evaluating the

speech quality, including drive test information, counter information, and

guides to related tools (see the operation guide). This document aims to quickly

locate and solve speech quality problems and to evaluate the speech quality

and prevent speech quality problems based on the collected information about

the existing network.

R&D Support

For speech quality problems, we can provide trainings and 7x24 hour technical support.

List of R&D support engineers

Name

Employee ID

Phone

• Evaluation Standards and Principles of Speech

Quality (MOS)

• Statistics and Analysis of Factors Affecting the MOS

• Subjective Speech Problem Handling

• Voice-Related Key Parameters:

Quality Parameters

Codec Parameters

Handover Parameters

AoIP Parameters

Page 5

Subjective Evaluation

This method indicates that many people compare the original voice sample with the degraded file

processed by the system by their own subjective perceptions, then mark the mean opinion score

(MOS) (ITU-T 800) value (full score: five points), and finally obtain the average value.

Objective Evaluation

This method indicates that the score is obtained through comparing the degraded voice file after

transmission with the original voice sample file by using a certain algorithm, such as PAMS (ITU-T

P 861) and PESQ (ITU-T P 862.1).

Parameter Evaluation

This method indicates that the voice after transmission is not evaluated, and the original voice is

not obtained. Instead, the voice after transmission is evaluated through some parameters of

wireless transmission network, which has a promising prospect in wireless network, such as

RXQUAL, VQI of Huawei, and SQI of E///.

Currently, carriers all over the world treat the speech quality as the key indicator for network

acceptance. Among them, PESQ algorithm is the widely-used scoring standard. In this algorithm,

Page 6

PESQ: Perceptual evaluation of speech quality

This method is used for E2E network speech quality test. It is to compare the

original voice sample on the transmitting end in the network (narrowband) with

the distorted degraded voice file received on the receiving end, evaluate the

difference between the two signals through complex signal processing, and

finally obtain the speech quality value using the PESQ algorithm.

After the PESQ algorithm is processed, the following four metrics are obtained:

• PESQ RAW SCORE

(the raw score)

• P.862.1 (the score is obtained through the P.862.1 mapping mode based on the raw score)

• PESQ-LQ (the score is obtained through the Psytechnics mapping mode)

• PESQ-Ie (The score is obtained through the mutilation factor of instrumental models defined by

P.834)

Among them, the value of

P.862.1

is widely regarded as the reference value in voice

evaluation.

Contents



Evaluation Standards and Principles of Speech

Quality (MOS)



Statistics and Analysis of Factors Affecting the

MOS



Subjective Speech Problem Handling



Voice-Related Key Parameters:

Quality Parameters

Codec Parameters

Handover Parameters