Operation Guide For Speech Tests

Operation Guide for Speech Tests
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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.
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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.
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1.2 Generation 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.
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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.
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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,
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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
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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.
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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.
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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
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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
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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
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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 A Interface Local Loopback A Interface Remote Loopback Abis Local Loopback Abis Remote Loopback MS A (Under the Local BSC) MS A can hear the voice of MS B and its own voice. MS A cannot hear its own voice but can hear the voice of MS B. MS A cannot hear its own voice or the voice of MS B. MS A can hear its own voice but cannot hear the voice of MS B. BM Ater Local Loopback BM Ater Remote Loopback TC Ater Local Loopback TC Ater Remote Loopback MS A can hear its own voice and the voice of MS B. MS A cannot hear its own voice but can hear the voice of MS B. MS A cannot hear the voice of MS B or its own voice. MS A can hear its own voice but cannot hear the voice of MS B. MS B (Under Another BSC) MS B cannot hear its own voice or the voice of MS A. MS B cannot hear the voice of MS A but can hear its own voice. MS B can hear the voice of MS A and its own voice. MS B can hear the voice of MS A but cannot hear its own voice. MS B cannot hear the voice of MS A or its own voice. MS B cannot hear its own voice but can hear the voice of MS B. MS B can hear its own voice and the voice of MS A. MS B can hear the voice of MS A but cannot hear it own voice.
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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.
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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.
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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
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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.
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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:
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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:
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If the user can hear his/her own voice, it indicates that speech problems occur on the TC subrack. If the user cannot hear his/her own voice, it indicates that speech problems occur on the 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. If the user cannot hear his/her own voice, it indicates that speech problems occur 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: If the user can hear his/her own voice, it indicates that speech problems occur on the NEs or boards above the GTNU in the TC subrack. If the user cannot hear his/her own voice, it indicates that speech problems occur 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.
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(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. If the user cannot hear his/her own voice, it indicates that speech problems occur 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: If the user can hear his/her own voice, it indicates that speech problems occur on the NEs or boards above the DSP in the TC subrack. If the user cannot hear his/her own voice, it indicates that speech problems occur on the 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: If the user can hear his/her own voice, it indicates that speech problems occur on the MSC side. If the user cannot hear his/her own voice, it indicates that speech problems occur on the 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 2013-8-8
BTS Speec
TRX Speec
Ater Local
GTNU Local
Ater Remote
GTNU Remote
TC Remote
TC Remote
A Interface
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Loopbac k
h Loop Test
h Loop Test
Loopbac k
Loopbac k
Loopbac k
Loopbac k (Next To the Abis Interface)
Loopbac k (Next To the Abis Interface)
Loopbac k (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 oneway audio detection). If the BTS fails to support this function, the one-way audio detection and location cannot be performed.
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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 BTS3012 (old and new DTRU) BTS3900 BSC6000 BSC6900 R008 Series R008C012SPC26 R008C11SPC031 R008C12SPC019 and later versions N/A V900R011C00SPC720 and later versions R009 Series All R009 versions R009C00SPC072 and later versions N/A
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.
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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.
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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.
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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.
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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),
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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.
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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.
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Product Type BTS3012 (old DTRU and new DTRU) BTS3900 BSC6000 BSC6900
R008 Series R008C012SPC26 R008C11SPC031 V900R008C12SPC019 and later versions N/A
R009 Series All R009 versions R009C00SPC072 and later versions N/A V900R011C00SPC720 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.
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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
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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 oneway 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. 1 2 3 4 5 6 Information Time that one-way audio occurs (format: YYYYMMDDHHMMSS) Link where one-way audio detection is performed. 0: uplink; 1: downlink Result of one-way audio detection. 0: one-way audio does not exist; 1: one-way audio exists; 2: crosstalk exists Number of frames that are checked in class-1 one-way audio detection Ratio of the number of bad frames to the total number of frames being checked Site No., cell No., TRX No., and channel No. of the call Huawei Confidential Page 31 of 35
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Information 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 9
Transmission mode used over the Abis interface. 0: TDM; 1: IP; 2: HDLC 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 11 12 13 14 15 16
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. 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. Subrack No. and slot No. of the DPU, DSP No., and DSP timeslot No. CIC No. over the A interface DPC group number Uplink and downlink DTX switch Service type
The following file is an example of the one-way audio log:
20090504_000407_0000.rar
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
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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
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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) 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 2013-8-8 Huawei Confidential Page 34 of 35 Effect on the Calling Party Effect on the Called Party
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To ensure the accuracy of the loop test result, perform each type of loop tests for multiple times.
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Operation Guide For Speech Tests

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Operation Guide for Speech Tests