Operation Guide for Speech Tests

Operation Guide for Speech Tests Confidentiality

Operation Guide for Speech Tests

I. Capturing Frames by Probe over the Um Interface

1.1 Configuration of Frame Captuure by Probe over the Um Interface

During the assessment test of speech quality, if users want to locate the low mean

opinion score (MOS) problem, users need to select to record the 0x7143 and 0x7144

packets of the AMR during the hardware configuration. In this way, after the test, the .gen

file recorded by Probe 2.2 includes the speech frame data to facilitate users' analysis.

For details about how to configure the hardware, see the following parts:

1. In the Device Config dialog box, click Filter.

2. The Filter interface is displayed. Select UL (Tx) vocoder packet

characteristics(0x7143) and DL (Rx) vocoder packet characteristics(0x7144)

under the AMR node, as shown in the following figure.

2023-4-17 Huawei Confidential of 34


Through the preceding configurations, 7143 and 7144 packets including speech frames

will be recorded in the .gen file from the beginning of the test.

If two user equipment (UEs) are used for the test, the AMR attributes of the two UEs should be

configured respectively according to the preceding procedure.

After the preceding configurations, the type and data are displayed in the WCDMA AMR

window during the test and playback processes.



1.2Generation of Log Data After Frames Are Captured by Probe over the

Um Interface

After the Probe test, the corresponding .gen file (the log file of Probe) is generated. Then,

users should export the .gen file saved by Probe 2.2 as a .log file.

A. Start the Probe program and then choose Logfile > Export Data. A dialog box is

displayed, as shown in the following figure.

B. Select Log(Voice FRAME) from the drop-down list of Export As and then click Next.

A dialog box is displayed, as shown in the following figure.



C. Select the .gen file to be exported and the path under which the exported .log file is

saved. Then click Next. A dialog box is displayed, as shown in the following figure.



D. Click Start in the Log Wizard – Step Three dialog box. The .log file is exported.

II. TC Recording

2.1 Operations for TC Recording on the BSC6000

Configuration of TC Recording on the LMT

Function introduction: With this function, the TC can record the calls initiated by a

maximum of four specified MSs (two calling MSs and two called MSs, the four MSs

cannot be distributed on the same DSP) in real time and save the recorded data to the

specified place of the GBAM.

Operations

A. Configure the authorization code.

If the BSC version is BSC6000V900R008C12SP15 or a later version, the recording

configuration parameters can be displayed on the LMT only after you enter the

authorization code. If the BSC version is earlier than the BSC6000V900R008C12SP15,



perform step B.

The method for entering the authorization code is as follows:

1. Obtain the authorization code.

Field engineers are responsible for sending the configuration data of the BSC to be

recorded to the R&D personnel. The R&D personnel are responsible for providing the

authorization code to field engineers. The format of the authorization code is as follows:

gQAA8wAACv0JCAAA.

2. Enter the authorization code.

Write the authorization code on a piece of paper. Note that the authorization code is

case-sensitive.

On the local maintenance terminal (LMT), choose System > System Configuration.

A dialog box is then displayed, as shown in the following figure.

Press Ctrl+Shift+F12. In this case, no change occurs on the dialog box. Then, type the

authorization code as written on the paper and ensure that every letter is entered

correctly. Pay special attention to the differences between 0 and O and differences

between 1, i, and l. Then, click Yes in the displayed box (the box may not be displayed in

some scenarios). Then, the System Configuration dialog box disappears automatically.

If the dialog box still exists, it indicates that the entered authorization code is incorrect



and needs to be entered again.

B. On the Management Tree tab page, click BSC6000, and then press Shift+Ctrl+F12

to enter the super user mode. Right-click BSC6000 and then choose Configure BSC

Attributes from the shortcut menu, as shown in the following figure.

C. In the displayed Configure BSC Attributes dialog box, click the Test Parameters

tab. On the Test Parameters tab page, set TC Record Switch to Yes.



D. Configure the phone numbers of the calling and called MSs used for the recording.

The phone numbers here refers to the MSISDNs of the calling and called MSs. Note that

the calling and called MSs to be tested should be recorded by the TC.

The following part introduces how to add the country code (CC) to the phone number.

Take the CC of China (86) as an example. Before the test, use the calling MS to make a

call to the called MS and then check whether "86 + phone number of the calling MS" is

displayed on the called MS. If displayed, enter 86 before the phone number of the MS on

the LMT. That is, enter 861398888007 during the configuration if the actual phone

number of the calling MS is 1398888007. If not displayed, enter the phone number of the

calling MS directly on the LMT. This requirement is not applicable to the called MS.



After the TC recording starts, check whether a .dat file named after the last four numbers

of the calling or called MS exists in the \\xxx.xxx.xxx.xxx\BSC6000\data\mtndata\tcfile

directory. Take the 030813013-20090606171322-5320.data file as an example. 5320 is

the last four numbers of the calling or called MS.

If the preceding TC recording fails, modify the configurations during the TC recording by

using "phone number of the calling MS", "86 + phone number of the calling MS", "phone

number of the called MS", and "86 + phone number of the called MS".

In addition, to facilitate the comparison of Probe data and TC recording file, synchronize

the time on the GBAM of the BSC with the time on Probe.

2.2 Operations for TC Recording on the BSC6900

A. Run the following command to add the phone number of the MS to be recorded:

ADD CALLRESRECORD: RECINDEX=2, USERIDTYPE=BYMSISDN,

MSISDN="13908881006", RECORDTYPE=TC;

Wherein, RECINDEX indicates the index number of the MS to be recorded, and the value



ranges from 0 to 3.

MSISDN indicates the phone number of the MS to be recorded.

RECINDEX is the same as that of the BSC6000, ranging from 0 to 3. Note that only four

recording files are allowed for a BSC.

B. Enter LST CALLRESRECORD in the Command Input(F5) box to query the phone

number of the MS to be recorded, as shown in the following figure.

C. Enter RMV CALLRESRECORD in the Command Input(F5) box to remove the phone

number of the MS to be recorded, as shown in the following figure.

Enter the index number of the MS to remove the phone number of the MS. For example,

0 indicates that the phone number of the MS 0 is removed.

After the recording, remove the phone number of the recorded MS. Otherwise, the recording continues

and thus a great amount of data is generated.

2.3 Data Obtaining After the TC Recording

After the TC recording configuration is complete, a .dat file of the call is generated during

the test. The .dat file is saved in the 10.161.xx.xxx \bsc6000\ data\ mtndata\ tcFile

directory on the GBAM of the BSC. Wherein, 10.161.xx.xxx indicates the IP address of

the BSC. The generated .dat file is named after the last four numbers of the calling or

called MS.

Take the 20080914140338-011300016-6063.dat file as an example. 200809140338



indicates the specific time when the recording is started. 011300016 indicates the slot

No. and channel No. of the DSP occupied by the call. 6063 indicates the last four number

of the tested MS. Note that the 20080914140338-011300016-6063.dat file records the

information of the MS that is being called or originates the call to the MS with the last four

numbers of 6063.

III. Operations for Speech Loopback Test

3.1 Loopback on the BSC6000

3.1.1. Remote Speech Loopback

A. Timeslot Loopback on the Interface Board

The timeslot loopback on the interface board can be performed on the GE(O)IUA,

GE(O)IUB, GE(O)IUT, or GE(O)IUP. Loopback Mode can be set to Remote Loopback

or Local Loopback.

If Loopback Mode is set to Remote Loopback, the signals received on a timeslot are



sent out without being processed. During the loopback, the interface board receives

signals transmitted on the timeslot.

If Loopback Mode is set to Local Loopback, the signals that should have been sent are

received. The interface board receives signals transmitted on the timeslot during the

loopback.

Method of querying the loopback status: Click Query to query the loopback status over

relevant ports.

During speech problem location, the timeslot loopback over the Abis interface is

commonly performed. The timeslot loopback over the A interface can be replaced by the

remote speech channel loopback.

The following table lists the test results of the interface board port timeslot loopback in

normal cases:

Loopback Mode/Test MS MS A (Under the Local BSC) MS B (Under Another BSC)

A Interface Local

Loopback

MS A can hear the voice of

MS B and its own voice.

MS B cannot hear its own

voice or the voice of MS A.

A Interface Remote

Loopback

MS A cannot hear its own

voice but can hear the voice

of MS B.

MS B cannot hear the voice

of MS A but can hear its own

voice.

Abis Local LoopbackMS A cannot hear its own

voice or the voice of MS B.

MS B can hear the voice of

MS A and its own voice.

Abis Remote Loopback

MS A can hear its own voice

but cannot hear the voice of

MS B.


MS A but cannot hear its own

voice.

BM Ater Local LoopbackMS A can hear its own voice

and the voice of MS B.

MS B cannot hear the voice

of MS A or its own voice.

BM Ater Remote

Loopback

MS A cannot hear its own


of MS B.

MS B cannot hear its own


of MS B.

TC Ater Local LoopbackMS A cannot hear the voice of

MS B or its own voice.

MS B can hear its own voice

and the voice of MS A.

TC Ater Remote

Loopback

MS A can hear its own voice

but cannot hear the voice of

MS B.


MS A but cannot hear it own

voice.

When the interface board port timeslot loopback is performed on the Abis or Ater interface board, the



timeslot to be looped back needs to be set correctly. Assume the E1 timeslot is X, you can calculate X

by using the timeslot (Y) that is queried through the single user resource query function. The formula is

X = Y/8.

The timeslot loopback may lead to the loopback of other calls carried on the timeslot. Therefore, perform

the timeslot loopback when no other calls are initiated on the timeslot. On the LMT, choose BSC

Maintenance > Maintain User Resource > Query Abis Interface Timeslot State to query whether the

Abis interface is occupied. On the LMT, choose BSC Maintenance > Maintain User Resource >

Maintain Ater Resource to query whether the Ater interface is occupied.

3.1.2 TRX/BTS Speech Loop Test

A. BTS Speech Loop Test

The BTS Speech Loop Test tests the connection of the TRX speech channel between the

Um interface and the BTS DBUS. The TMU loops back the DBUS timeslot of the speech

channel to be tested. If the user who occupies a channel can hear his own voice, it

indicates that the channel is connected through; otherwise, the channel is not connected

through.

If speech problems such as noises occur, perform the BTS speech loop test to check

whether you can hear noises. If you can hear noises, it indicates that the speech

problems are caused by boards or NEs above the Abis interface.



B. TRX Speech Loop Test

The TRX Speech Loop Test tests connection of the TRX speech channel between the

Um interface and the DSP. If the user who occupies a channel can hear his/her own

voice, it indicates that the channel is connected through and the loopback is performed

properly.

If speech problems such as noises occur, perform the TRX speech loop test to check

whether you can hear noises. If you can hear noises, it indicates that the speech

problems occur on the direction from the TRX to the BTS. If you cannot hear noises, it

indicates that the speech problems occur on the direction from the TRX to the MSC.



The BTS Speech Loop Test indicates the loopback test of DEBUS timeslot of the speech channel on the

TMU on the BTS. The TRX Speech Loop Test indicates the loopback test of the speech channel on the

TRX board. The test result indicates whether the problem occurs on the BTS side or the BSC side.

Appendix: Methods for Determining the Relevant Resources Obtained and

Occupied by the MS

On the LMT, choose BSC Maintenance > Maintain User Resource > Query Single

User Resources. The Query Single User Resources dialog box is displayed. In the

dialog box, enter the IMSI of the calling MS (or the MSISDN of the called MS) to query

the resources occupied by the call. Then, save the query result by right-clicking the

Query Result box and then choosing Save As from the short-cut menu.



3.2 Loopback on the BSC6900

In the case of speech problems such as one-way audio, noises, or discontinuity, perform

the loopback test.

The STR CALLRESLOP command is executed to set the loopback test.

The procedure is as follows:

Step 1. Run the DSP CALLRES command to query the channel No. of the DSP occupied

by the call.

Step 2. Enter the STR CALLRESLOP command in the Command Input(F5) box and

then enter the IMSI of the calling MS or the MSISDN of the called MS to enable the

loopback test, as shown in the following figure.



Enter the STP CALLRESLOP command in the Command Input(F5) box and then enter

the IMSI of the calling MS or the MSISDN of the called MS to disable the loopback test,

as shown in the following figure.

Loop Position: The loopback test can be performed on almost all interface boards and

GTNUs.

It is recommended that the loopback test be performed from the Abis interface board and

the A interface board. That is, the NSS interface unit and the BSS interface unit.

(1) Perform the loopback test over the Abis interface on the BSS side. Run the

LOPLOC=BSSABISINTF, LOPDIR=MS; command to loop back the speech call on

the Abis interface to the MS.

Test result:



If the user cannot hear his/her own voice, it indicates that speech problems occur on the

direction from the MS to the BTS or from the BTS to the MS.

In this case, perform the BTS speech loop test. If the user still cannot hear his/her own

voice, perform the TRX speech loop test.

The procedures for performing the BTS speech loop test and the TRX speech loop test

are as follows:

Choose BTS Maintenance > Maintain TRX > Test Channel Loopback. On the Test

Channel Loopback interface, choose the site, cell, TRX No., channel No., and test

project of the MS to be looped back. In the Test Project box, choose BTS Voice Loop

Test and TRX Voice Loop Test successively. Read carefully the three items listed in the

Description box and then set the test duration. Then, click Start to perform the test.

If the user can hear his/her own voice, it indicates that speech problems are caused by

boards or NEs above the Abis interface.

(2) Perform the loopback test over the Ater interface on the BSS side. Run the

LOPLOC=BSSATERINTF, LOPDIR=MS; command to loop back the speech call on

the Ater interface board.

Test result:



If the user can hear his/her own voice, it indicates that speech problems occur on the TC

subrack.


BM subrack.

(3) If the speech problems are determined to occur on the BM subrack, perform the

loopback test on the GTNU (next to the Abis interface) on the BSS side by running

the LOPLOC=BSSTNUTOABIS, LOPDIR=MS; command.

Test result:

If the user can hear his/her own voice, it indicates that speech problems occur between

the GTNU in the BM subrack and the Ater interface board.

If the user cannot hear his/her own voice, it indicates that speech problems occur

between the GTNU in the BM subrack and the Abis interface board.

In this case, further actions need to be taken to locate the problem on the GTNU.

(4) If the speech problems are determined to occur on the TC subrack, do as follows:

(a) Perform the loopback test over the Ater interface on the NSS side by running the

LOPLOC=NSSATERINTF, LOPDIR=MS; command.

Test result:

If the user can hear his/her own voice, it indicates that speech problems occur on the

NEs or boards in the TC subrack.


between two Ater interface boards. In this case, check the lines on the Ater interface and

the status of the Ater interface board in the TC subrack.

(b) Perform the loopback test on the GTNU (next to the Abis interface) on the NSS side

by running the LOPLOC= NSSTNUTOABIS, LOPDIR=MS; command.

Test result:


NEs or boards above the GTNU in the TC subrack.


between the GTNU in the TC subrack and the Ater interface board. In this case, further

actions need to be taken to locate the problem on the GTNU.



(c) Perform the loopback test on the TC subrack (next to the Abis interface) on the NSS

side by running the LOPLOC= NSSTCTOABIS, LOPDIR=MS; command.

Test result:

If the user can hear his/her voice, it indicates that speech problems occur on the DSP or

boards and NEs above the DSP in the TC subrack.


between the GTNU in the TC subrack and the DSP. In this case, further actions need to

be taken to locate the problem on the GTNU.

(d) Perform the loopback test on the TC subrack (next to the A interface) on the NSS

side by running the LOPLOC= NSSTCTOA, LOPDIR=MS; command.

Test result:


NEs or boards above the DSP in the TC subrack.


DSP in the TC subrack. In this case, perform the further analysis according to the

recording file.

(5) Perform the loopback test over the A interface. Run the LOPLOC=NSSAINTF,

LOPDIR=MS; command to loop back the speech call on the A interface board to the

MS.

Test result:


MSC side.


BSC.

Step 3. If crosstalk occurs, perform the loopback tests as described in Step 2 again.

The following table needs to be filled in:

Loopbac

k Type

Abis

Local

BTS

Speec

TRX

Speec

Ater

Local

GTNU

Local

Ater

Remote

GTNU

Remote

TC

Remote

TC

Remote

A

Interface



Loopback h Loop

Test

h Loop

Test

Loopback Loopback Loopback Loopback

(Next To

the Abis

Interface)

Loopback

(Next To

the Abis

Interface)

Loopback

(Next To

the A

Interface)

Loopback

One-way

audio

Crosstalk

IV. Operations for One-Way Audio Detection

In this function, the one-way audio detection of two classes is provided. In class-1

detection, the detection threshold is adjusted flexibly to find the one-way audio as many

as possible. In class-2 detection, the one-way audio problems found in class-1 are

checked to ensure the accuracy. In addition, the information about fault location is added

to facilitate troubleshooting.

This function can detect only the one-way audio caused by BSS faulty and occurred on

the TC between the BTS (TRX) and the BSC. In BM/TC separated mode, the detection

range of this function is as follows:

Currently, the Abis interface and the A interface support the class-2 one-way audio

detection only when these interfaces work in TDM mode. In addition, the function cannot

be implemented without the support of the BTS (the TRX should support the class-2 one-

way audio detection). If the BTS fails to support this function, the one-way audio

detection and location cannot be performed.



4.1 One-Way Audio Detection on the BSC6000

The following table lists the earliest BSC and BTS versions that support the one-way

audio detection function. It is recommended that the one-way audio detection function be

enabled.

Product Type R008 Series R009 Series

BTS3012 (old and new DTRU) R008C012SPC26 All R009 versions

BTS3900 R008C11SPC031R009C00SPC072 and later

versions

BSC6000R008C12SPC019 and later

versionsN/A

BSC6900 N/AV900R011C00SPC720 and

later versions

4.1.1 Configuring BSC Attributes

Step 1: Start the LMT.

Step 2: On the Management Tree tab page, click BSC6000, and then press

Ctrl+Shift+F12 to enter the super user mode. Right-click BSC6000 and then choose

Configure BSC Attributes from the shortcut menu, as shown in the following figure.



Step 3: In the displayed Configure BSC Attributes dialog box, click the Inner Software

Parameters tab, and then set the following parameters: Speech Channel Alarm

Threshold, Speech Channel Resume Alarm Threshold, and TC Mute Detect Class 1

Detect Threshold(%), as shown in the following figure.

Speech Channel Alarm Threshold: This parameter 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. The smaller the value

is, the easier the speech channel alarm is reported. This parameter is valid only when

Mute Detect Class 2 Switch in cell attributes is set to Yes.

Speech Channel Alarm Resume Threshold: This parameter 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. The greater the value is, the easier the speech channel resume alarm is

reported. This parameter is valid only when Mute Detect Class 2 Switch in cell

attributes is set to Yes.



TC Mute Detect Class1 Threshold(%): This parameter specifies the threshold for the

proportion of the number of bad frames to the total number of TRAU frames. If the

proportion of bad frames exceeds this threshold within the value of Mute Timer, one-way

audio may occur. You can determine whether to detect the class-2 one-way audio

according to the setting of Mute Detect Class2 Switch.

4.1.2 Configuring Cell Attributes

Step 1: On the Management Tree tab page, click BSC6000, and then press

Ctrl+Shift+F12 to enter the super user mode. Right-click the cell to be configured, and

then choose Configure Cell Attributes from the shortcut menu, as shown in the

following figure.

Step 2: In the displayed Configure Cell Attributes dialog box, choose the cell to be

configured. Then, choose Other Attribute > Advanced > Inner Cell Soft Parameters.

On the displayed Inner Cell Soft Parameters tab page, set the following parameters: TC



Mute Detect Flag, Mute Detect Class2 Switch, Period of Mute Detect Class1(s),

Exceptional Frame Threshold(%), Period of Detect Frames(s), and Mute Detect

Time(s), as shown in the following figure.

TC Mute Detect Flag: This parameter specifies whether the one-way audio detection is

allowed for the TC. If this parameter is set to No, the class-2 one-way audio detection is

not triggered.



Mute Detect Class 2 Switch: This parameter specifies whether to enable the class-2

one-way audio detection function.

Period of Mute Detect Class1(s): This parameter specifies the one-way audio detection

period. If the proportion of bad frames within the period specified by this parameter

exceeds the value of Exceptional Frame Threshold(%), you can infer that the one-way

audio occurs. Class-2 one-way audio detection is performed if Mute Detect Class 2

Switch is set to Yes.

Exceptional Frame Threshold(%): This parameter specifies the threshold for the

proportion of the number of bad frames to the total number of TRAU frames. If the

proportion of bad frames exceeds this threshold within the value of Period of Mute

Detect Class1(s), one-way audio may occur. You can determine whether to detect the

level-2 one-way audio according to the setting of Mute Detect Class2 Switch.

Period of Detect Frame(s): This parameter 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.

Mute Detect time(s): This parameter specifies the class-2 one-way audio detection

period. After the BTS or TC sends the class-2 one-way audio detection request, if no

response or TRAU test frame is received within the period specified by this parameter,

you can infer that the class-2 one-way audio detection fails; if the detection response is

received but no TRAU test frame is received within the period specified by this

parameter, you can infer that one-way audio occurs; if both the detection response and

the correct TRAU test frame are received within the period specified by this parameter,

the speech channel is inferred to be normal.

4.1.3 Configuring DPU Board Attributes

Step 1: On the BSC Device Panel tab page, right-click the DPU (GDPUC or GDPUX),

and then choose Configure Board Attributes from the shortcut menu, as shown in the

following figure.



Step 2: The Configure Board Attributes dialog box is displayed. Click Next. Set the

parameters Mute Detect and Mute Timer, as shown in the following figure.

Mute Detect: This parameter specifies whether to enable the one-way audio and no

audio detection function.

Mute Timer: This parameter specifies the duration of the one-way audio and no audio

that are detected by the BSC. If Mute Detect is set to On, the BSC instructs the

GDPUC/GDPUX to check whether one-way audio or no audio occurs after a call is set

up. Mute Timer indicates the duration of consecutive one-way audio after an alarm

related to the one-way audio is reported. These two attributes are available only for the

DPU software.

4.2 One-Way Audio Detection on the BSC6900

The following table lists the earliest BSC and BTS versions that support the one-way

audio detection function. It is recommended that the one-way audio detection function be

enabled.



Product Type R008 Series R009 Series

BTS3012 (old DTRU and

new DTRU)R008C012SPC26 All R009 versions

BTS3900 R008C11SPC031 R009C00SPC072 and later versions

BSC6000V900R008C12SPC019 and

later versionsN/A

BSC6900 N/AV900R011C00SPC720 and later

versions

4.2.1 Configuring BSC Attributes

Step 1: Start the Web LMT.

Step 2: On the Web LMT, run the SET BSCINNSOFT command to set the following

parameters: Speech Channel Alarm Threshold and Speech Channel Alarm Resume

Threshold.

Speech Channel Alarm Threshold: This parameter 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. The smaller the value

is, the easier the speech channel alarm is reported.

Speech Channel Alarm Resume Threshold: This parameter 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. The greater the value is, the easier the speech channel resume alarm is

reported. The value of Speech Channel Alarm Resume Threshold should be smaller

than that of Speech Channel Alarm Threshold. Otherwise, the alarm is intermittent.

4.2.2 Configuring Cell Attributes

Step 1: Start the Web LMT.

Step 2: On the Web LMT, run the SET GCELLSOFT command to set the following

parameters: Mute Detect Class1 Switch, Period of Mute Detect Class1, Exceptional

Frame Threshold of Mute Detect Class1, Mute Detect Class2 Switch, Period of Mute

Detect Class2 TRAU Frame, and Period of Mute Detect Class2, as shown in the

following figure.



Mute Detect Class1 Switch: This parameter specifies whether to enable the class-1

one-way audio detection function.

Period of Mute Detect Class1: This parameter specifies the class-1 one-way audio

detection period. If the proportion of bad frames within the period specified by this

parameter exceeds the value of Exceptional Frame Threshold of Mute Detect Class1,

you can infer that one-way audio occurs. Class-2 one-way audio detection is performed if

Mute Detect Class2 Switch is set to ENABLE(ENABLE).

Exceptional Frame Threshold of Mute Detect Class1: This parameter specifies the

threshold for the proportion of the number of bad frames to the total number of TRAU

frames. If the proportion of bad frames exceeds this threshold within the value of Period

of Mute Detect Class1, one-way audio may occur. Class-2 one-way audio detection is

performed if Mute Detect Class2 Switch is set to ENABLE(ENABLE).

Mute Detect Class2 Switch: This parameter specifies whether to enable the class-2

one-way audio detection function. This function is used to check the one-way audio

problem detected by the class-1 one-way audio detection. In this way, the accuracy of

one-way audio detection is improved.

Period of Mute Detect Class2 TRAU Frame: This parameter specifies the period for

sending the TRAU test frames 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.

Period of Mute Detect Class2: This parameter specifies the class-2 one-way audio

detection period. After the BTS or TC sends the class-2 one-way audio detection request,

if no detection response or correct TRAU test frame is received within the period

specified by this parameter, you can infer that the class-2 one-way audio detection fails; if

the detection response is received but no correct TRAU test frame is received within the

period specified by this parameter, you can infer that one-way audio occurs; if both the

detection response and the correct TRAU test frame are received within the period

specified by this parameter, the speech channel is inferred to be normal.

4.3 Application of Alarms and Logs

After a class-2 one-way audio is detected, you can set two alarm parameters to obtain

the alarm information. The two parameters are Speech Channel Alarm Threshold and

Speech Channel Resume Alarm Threshold. The alarm reporting period is one hour. If

the number of one-way audio that occurs in an hour exceeds the value of Speech

Channel Alarm Threshold, the one-way audio alarm is reported. If the number of one-

way audio that occurs in an hour is smaller than the value of Speech Channel Resume

Alarm Threshold, the one-way audio alarm is cleared.

The BSC records a piece of one-way audio log each time a class-2 one-way audio is

detected. The one-way audio logs are saved in the \\OMU IP\bsc6000\data\mtndata\log\

txt\mclog directory with the file name extension of .mclog. The format of the logs is as

follows:

No. Information

1 Time that one-way audio occurs (format: YYYYMMDDHHMMSS)

2 Link where one-way audio detection is performed. 0: uplink; 1: downlink

3 Result of one-way audio detection. 0: one-way audio does not exist; 1: one-way audio

exists; 2: crosstalk exists

4 Number of frames that are checked in class-1 one-way audio detection

5 Ratio of the number of bad frames to the total number of frames being checked

6 Site No., cell No., TRX No., and channel No. of the call



No. Information

7 Telephone number (or TMSI/IMSI) of the calling and called parties. For the outgoing call, the

TMSI/IMSI of the calling party and the called number are recorded; for the incoming call, the

TMSI/IMSI of the called party and the calling number are recorded. If both the telephone

number and TMSI/IMSI exist, they are all recorded.

8 Transmission mode used over the Abis interface. 0: TDM; 1: IP; 2: HDLC

9 Subrack No. and slot No. of the Abis interface board where the call is carried, port No.,

timeslot No., sub-timeslot No. of the call. The following information is also provided: IP

address and port No. of the Abis interface board (available only in IP transmission mode), or

HDLC port No. and sub-port No. of the Abis interface board (available only in HDLC

transmission mode).

10 Subrack No., slot No., port No., timeslot No., and sub-timeslot No. of the GEIUT in the BM

subrack. This information is unavailable in BM/TC combined mode.

11 Subrack No., slot No., port No., timeslot No., and sub-timeslot No. of the GEIUT in the TC

subrack. This information is unavailable in BM/TC combined mode.

12 Subrack No. and slot No. of the DPU, DSP No., and DSP timeslot No.

13 CIC No. over the A interface

14 DPC group number

15 Uplink and downlink DTX switch

16 Service type

The following file is an example of the one-way audio log:

This file can be opened directly by the Excel without the assistance of other tools.

V. Dialing Test Procedure and Precautions for Subjective

Speech Problems

When processing the preceding speech problems, such as one-way audio, mute, and

noises, field engineers should perform two tests: one is the dialing test at the site and the

other is the loopback test in the equipment room.

1. Select a site.

Stay in the serving cell of the problem site during the test to reduce the handover

probability. In addition, disable the function of measurement report (MR) preprocessing if

it is enabled at the site.

2. Adjust the time on the laptop and provide the feedback on the time difference between



the laptop and the GBAM.

Synchronize the time on the laptop with that on the GBAM to one second.

3. Start the TC recording on the BSC to trace both the calling MS and the called MS.

4. Trace the RSL signaling in the cell and the single user signaling of both the calling

party and the called party.

The tracing signaling file of a single user is named after the phone number of the MS.

Click … to modify the file name. Then, report the relations between the phone number

and the MS. For example, MS 1 used in the drive test uses the phone number of xxx, as

shown in the following figure.

5. Start the dialing test by initiating a call.

After the preceding preparations are complete, use TEMS or Probe to perform the dialing

test. Use an MS to call another MS (fixed-line phone is better).

When one-way audio, mute, or noises occur, the party should dial # twice on the TEMS



and cries, "One-way audio" ("Mute" or "Noises"), which facilitates the fast locating of the

problem point during data analysis. Then, record the time, information about the calling

party and the called party, and the duration of one-way audio, no audio, or noises to

facilitate the subsequent problem location.

6. The loop test procedure and precautions are as follows:

Remarks: Get the test record table ready before the test. For the record table template,

see III Operations for Speech Loopback Test. To ensure the accuracy of the loop test

result, it is recommended that each kind of loop test be performed for a minimum of four

times.

After speech problem occurs, filed engineers should perform the following tests in

sequence: remote speech channel loopback, timeslot loopback on the A interface board,

timeslot loopback on the Abis interface board, BTS speech loop test, and TRX speech

loop test.

If the speech problem still exists after each loop test is complete, the loop test result is

proved to be reliable. If the speech problem disappears after a loop test is complete, no

subsequent loop test is required.

The record table that is generated after the loop tests is as follows:

Loopback Mode (Calling Party) Effect on the Calling Party Effect on the Called Party

Remote speech channel loopback

(MSC)

Remote speech channel loopback

(BTS)

DSP channel loopback (A)

DSP channel loopback (Abis)

A interface local loopback

A interface remote loopback

Abis local loopback

Abis remote loopback

BM Ater local loopback

BM Ater remote loopback

TC Ater local loopback

TC Ater remote loopback

BTS speech loop test

TRX speech loop test



To ensure the accuracy of the loop test result, perform each type of loop tests for multiple times.


Operation Guide for Speech Tests

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Transcript of Operation Guide for Speech Tests