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2. The
Filter
interface
is
displayed.
Select
UL
(Tx)
vocoder
packet
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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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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.
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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.
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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.
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
of MS B.
voice.
MS B.
voice.
BM Ater Remote
Loopback
of MS B.
of MS B.
TC Ater Remote
Loopback
MS B.
voice.
When the interface board port timeslot loopback is performed on the Abis or Ater interface board, the
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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.
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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.
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Query Result box and then choosing Save As from the short-cut menu.
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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
Abis
BTS
TRX
Ater
GTNU
Ater
GTNU
TC
TC
k Type
Local
Speec
Speec
Local
Local
Remote
Remote
Remote
Remote
Interface
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Loopback
h Loop
h Loop
Test
Test
Loopback
Loopback
Loopback
Confidentiality
Loopback
Loopback
Loopback
(Next To
(Next To
(Next To
the Abis
the Abis
the A
Interface)
Interface)
Interface)
One-way
audio
Crosstalk
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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R008 Series
R009 Series
R008C012SPC26
BTS3900
R008C11SPC031
BSC6000
N/A
versions
BSC6900
N/A
V900R011C00SPC720 and
later versions
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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.
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
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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.
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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.
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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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Product Type
R008 Series
R009 Series
R008C012SPC26
BTS3900
R008C11SPC031
BSC6000
V900R008C12SPC019 and
N/A
new DTRU)
later versions
BSC6900
N/A
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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.
The
one-way
audio
logs
are
saved
in
the
\\OMU
Information
Result of one-way audio detection. 0: one-way audio does not exist; 1: one-way audio
exists; 2: crosstalk exists
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
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No.
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.
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
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
14
15
16
Service type
20090504_000407_0000.rar
This file can be opened directly by the Excel without the assistance of other tools.
V.
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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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)
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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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