110857247 3 gsm speech quality influence factors troubleshooting methods and tools deliverables...
DESCRIPTION
GSMTRANSCRIPT
HUAWEI TECHNOLOGIES CO., LTD.
www.huawei.com
Huawei Confidential
Security Level:
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23/4/13
Prepared by: GUL Network I&V and Maintenance
Department
Reviewed by: Qi Haofeng
GSM Speech Quality:Influence Factors + Troubleshooting Methods and
Tools + Deliverables
July 30, 2011
Abstract:
This document mainly discusses the main factors that affect the speech quality
of a GSM network, principles of improving the functions related to speech
quality, and suggested values of some key parameters. In addition, this
document lists the deliverables (see the attachments) that field engineers
should submit when reporting speech quality problems or evaluating the speech
quality, including drive test information, counter information, and guides to
related tools (see the operation guide). This document aims to quickly locate
and solve speech quality problems and to evaluate the speech quality and
prevent speech quality problems based on the collected information about the
existing network.
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 3
R&D Support
For speech quality problems, we can provide trainings and 7x24 hour technical support.
List of R&D support engineers
Name Employee ID Phone
Yang Zhengjie (Wireless
Network)
00127669 See the phone book.
Yang Chunjie 00119951 See the phone book.
Feng Lei (Core Network) 00151560 See the phone book.
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Contents
• Evaluation Standards and Principles of Speech
Quality (MOS) • Statistics and Analysis of Factors Affecting the MOS
• Subjective Speech Problem Handling
• Voice-Related Key Parameters:
Quality Parameters
Codec Parameters
Handover Parameters
AoIP Parameters
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FrutigerNext LT Medium
外部使用字体 : Arial
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Page 5
Evaluation Standards and Principles of Speech Quality (MOS)
Subjective EvaluationThis method indicates that many people compare the original voice sample with the degraded file
processed by the system by their own subjective perceptions, then mark the mean opinion score
(MOS) (ITU-T 800) value (full score: five points), and finally obtain the average value.
Objective EvaluationThis method indicates that the score is obtained through comparing the degraded voice file after
transmission with the original voice sample file by using a certain algorithm, such as PAMS (ITU-T
P 861) and PESQ (ITU-T P 862.1).
Parameter EvaluationThis method indicates that the voice after transmission is not evaluated, and the original voice is
not obtained. Instead, the voice after transmission is evaluated through some parameters of
wireless transmission network, which has a promising prospect in wireless network, such as
RXQUAL, VQI of Huawei, and SQI of E///.
Currently, carriers all over the world treat the speech quality as the key indicator for network
acceptance. Among them, PESQ algorithm is the widely-used scoring standard. In this algorithm,
the PESQ calculation result is mapped into the MOS value, ranging from 1.0 to 4.5.
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英文目录正文 :28-30pt
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Evaluation Standards and Principles of Speech Quality (MOS)
PESQ: Perceptual evaluation of speech quality
This method is used for E2E network speech quality test. It is to compare the
original voice sample on the transmitting end in the network (narrowband) with
the distorted degraded voice file received on the receiving end, evaluate the
difference between the two signals through complex signal processing, and
finally obtain the speech quality value using the PESQ algorithm.
After the PESQ algorithm is processed, the following four metrics are obtained:
• PESQ RAW SCORE (the raw score)
• P.862.1 (the score is obtained through the P.862.1 mapping mode based on the raw score)• PESQ-LQ (the score is obtained through the Psytechnics mapping mode) • PESQ-Ie (The score is obtained through the mutilation factor of instrumental models defined by
P.834)
Among them, the value of P.862.1 is widely regarded as the reference value in voice
evaluation.
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 7
Contents
Evaluation Standards and Principles of Speech
Quality (MOS)
Statistics and Analysis of Factors Affecting the
MOS
Subjective Speech Problem Handling
Voice-Related Key Parameters: Quality Parameters Codec Parameters Handover Parameters AoIP Parameters
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential
Statistics and Analysis of Factors Affecting the MOS
Voice qualityVoice quality
CodeCode Bit error (frame erasure)
Bit error (frame erasure) HOHODirectly-related
factors
Directly-related factors
Indirectly-related factors
Indirectly-related factors
Traffic
Traffic
Full/half
rate
Full/half
rate
High/low
coding rate
High/low
coding rate
Air Interface Quality
Air Interface Quality
Traffic Busy Threshold
Traffic Busy Threshold
Rate adjust Threshold
Rate adjust Threshold
F2H HO Threshold
F2H HO Threshold
Threshold Self-Adaptive
Threshold Self-Adaptive
Speech
version
Speech
version
Frequent HOs, PingPong HOs,
and unreasonable HOs
Frequent HOs, PingPong HOs,
and unreasonable HOs
Too low PN Rule
Too low PN Rule
Inappropriate
Neighboring Cell
Inappropriate
Neighboring Cell
Too Small HO Hysteresis
Too Small HO Hysteresis
Frame Theft of Physical
Messages
Frame Theft of Physical
Messages
Parameters, algorithms, and
optimizing strategies
Parameters, algorithms, and
optimizing strategies
The channel is normal.The channel is normal.
Interference
Interference
Algorithms
Algorithms
Engineering
network
optimization
Engineering
network
optimization
3.5-Generation Power
Control
3.5-Generation Power
Control
DTX
DTX
VAD
VAD
Anti-Interference solution
Anti-Interference solution
Intermodulation
Interference Quick
Troubleshooter
Intermodulation
Interference Quick
Troubleshooter
TOP Optimization
TOP Optimization
HO Optimization Packet
HO Optimization Packet
Call drop
Call drop
Long Call Drop Timer
Long Call Drop Timer
CoBCCH Resident
Strategy
CoBCCH Resident
Strategy
Discarded Packets
Compensation
Discarded Packets
Compensation
The speech quality is mainly related to three factors: code, bit error, and handover (HO). The coding factor benefits the speech quality. The bit error and handover factors, however, damage the speech quality.
To optimize the speech quality, you need to select reasonable codes and reduce the effect of the bit error rate (BER) and handovers on the speech quality.
The prerequisite is that the channel is normal.
The methods for improving the call drop rate and handover success rate usually damage the speech quality and the experience of subscribers. Therefore, strategies that optimize the speech quality may affect the call drop rate and handover success rate.
Page 8
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Statistics and Analysis of Factors Affecting the MOS
The MOS handling process is as follows:
MOS fails to reach the standard.
Make clear the actual situation of the problem (including
average MOS or deterioration conditions of the percentage of high scores, testing methods,
and instruments).
001
After the optimization and adjustment, verify whether the
problem is solved through drive test.
001
End
Yes
No
Trouble-shoot the occupation of voice
version and encoding rate.
002
Trouble-shoot the percentage of the number of handover times and that
of MOS dotting.
003
Trouble-shoot the data configuration and
transmission quality.
004
Trouble-shoot the TC recording and air
interface frame data.
005
Trouble-shoot the air interface quality.
006
Analyze the preceding factors and specify the reasons that lead to the
MOS problems.
007
After the preceding troubleshooting, optimization & adjustment, and
verification, if the MOS problems are still not solved, perform the
escalation processing.
008
During the processing, perform the troubleshooting from the easier to the more advanced following the dashed in red.
In existing networks, two methods
are available for the MOS acceptance
standard: One is that the average
MOS for the drive test of the entire
network shall reach a value. The
other is that the proportion of high
scores in the MOS shall be larger
than the required value, or the
proportion of low scores shall be
lower than a certain value, and
comparison between the two drive
test data (such as migration and
version upgrade) shall be performed.
No matter which method is used,
when the MOS does not reach the
standard, troubleshooting is carried
out based on factors affecting the
MOS in the process.
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 10
1. Problem specifying Specify the details about the MOS problem, including the current test value, target value, and the gap
between the two. Specify the standard of the MOS appraisal. Specify whether it is MS-MS, or MS-PSTN, whether the MOS
appraisal value is the overall average score or percentage of high scores, and whether the up-link and down-
link are appraised separately. Contrast the test MOS values by using instruments or terminals, it is found that there is no change. Contrast the test time frame (start from what time point and to what time point the test ends), test route, and
test period (to ensure the comparability of MOS tests, the tests shall be performed on the same day in
different weeks).
2. Speech version and coding proportion analysis There is a large difference for the MOS in different speech versions. In normal conditions, the sequence for
the MOS baseline performance is: FAMR > EFR > HAMR > FR > HR. For example, the MOS for the EFR in
the MS-MS test can be 4.0, whereas that for the HR will be 3.0. Therefore, for MOS problems incurred before
and after migration, contrast the occupation proportion of each speech version in the drive test, and check
whether the proportion of half rate is increased.
Statistics and Analysis of Factors Affecting the MOS
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3. Handover times and proportion analysis If the MOS problem occurs before or after migration (upgrade), check whether the ratio between the number
of MOS dotting and that of handover times is changed. The larger this ratio is, the smaller the effect made by
the handover on the overall MOS is. Analyze whether ping-pong handover exists in the drive test data, or whether the handover times in counters
are too much (usually, the number of handover times in each call is within 1 in existing networks). If such
case exists, modify the corresponding parameter configuration to reduce the effect of handover on the MOS.
If the PN for the PBGT (better cell) handover is added, and the PBGTSTAT (s) parameter is set to 5s and
the PBGTLAST (s) parameter is set to 4s, the judgment time for handover is delayed, and the handover is
reduced.
4. Comparative analysis of data configuration and transmission If the MOS problem occurs before or after migration (upgrade), check whether the data configuration and
transmission mode are changed, for example, whether parameters including cell handover and power control
are changed, and whether cells or frequency band is added or reduced. In addition, trouble-shoot the radio
frequency channels in the area where problems occur, and check whether KPIs in the traffic transmission are
incorrect, which affect the MOS test result. For new-built network, check whether alarm information is displayed on the NEs such as the BTS, BSC, and
transmission in the test, which affects the test result. Check whether the network KPIs are abnormal. Usually,
KPIs include TCH call drop rate, success rate of wireless handover, and TCH congestion rate.
Statistics and Analysis of Factors Affecting the MOS
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5. Analysis on TC recording and Um interface frame capture
Mainly check whether there is any problem on the TC and Probe terminals and the area where the
problem occurs.
6. Um interface quality analysis
For comparison before and after migration (upgrade), analyze based on the quality before and
after migration (upgrade), and check whether quality deterioration exists in all areas or part of
areas. If quality deterioration exists in some areas (including cases that call drop or handover
failure occurs due to poor quality of Um interfaces), perform analysis based on the drive test data,
and make clear whether the poor quality is caused by cases such as interference, poor coverage,
and missing cross and neighboring cells.
Statistics and Analysis of Factors Affecting the MOS
Data Wave Mode Comparison FilesWhether Difference
Exists in Wave Mode Comparision
MOS Compared with the Original Sample
TC recording file
Probe file
UpIn vs. UpOut
DnIn vs. DnOut
Original sample vs. MS uplink voice dataMS downlink voice data vs. Recording (degraded) file
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Voice services are key services in a GSM network. The quality of voice services is determined by many factors. To report MOS problems or speech problems, you need to report all factors related to voice during the drive test.
For details about the distributed troubleshooting, see the Guide to Locating and Isolating GSM Speech Problems.
Statistics and Analysis of Factors Affecting the MOS
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Good speech quality first depends on good network quality. The quality of the entire network,
however, should be evaluated by counters. For details, see the following table.
For more detailed feedback information, see the attachment "Checklist for Data Provided for
Speech Quality Problems".
Feedback Information About Counter Data Related to Voice Problems
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Typical Case 1:
In a site, it is required that the average value of MS-MS MOS should be
greater than 3.50 according to the drive test result of the entire network after the
migration. The value, however, is only 3.35 according to the drive test result.
Therefore, the value of MOS fails to reach the standard.
According to the statistics and analysis of the drive test information, the ratio of half rate channels
reaches 75%. This is the main factor that affects the overall MOS. The details are as follows:
Through parameter configuration and counter
analysis, it is discovered that the problem is
caused by that the values of TCH Traffic Busy
Threshold of many cells are set to be too small
(30%). After adjusting the values of this
parameter and performing another drive test, the
ratio of occupied half rate channels is reduced to
47% and the value of MOS reaches 3.52, which
exceeds the acceptance standard.
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Typical Case 2:
In a site, it is required that more than 95% MS-PSTN MOS values should be greater than 2.7 in the
drive test of the entire network after the migration. According to the drive test result, only 90% MS-PSTN
MOS values are greater than 2.7. Therefore, the MOS value fails to reach the standard.
A: According to the analysis, no known speech problem exists in the BSS after the migration. The reason that the
MOS value of the existing network fails to reach the standard is that the quality over the Um interface is low,
many handovers occur, and the ratio of half rate channels is high.
B. After optimizing the concentric handover parameters, adjusting the Assign Optimum Layer and the Pref.
Subcell in HO of Intra-BSC parameters and their thresholds, and optimizing the number of handovers, the ratio
between traffic and number of successful handovers rises from 68.4 to 71.9. In addition, the ratio between
number of MOS values and number of handovers in the drive test rises from 1.86 to 2.76. Therefore, the ratio of
MOS values that are grater than 2.7 of the entire network rises about 4%.
C. After optimizing cells one by one and expanding the capacity of busy cells, the ratio of half rate channels in the
test is reduced from 46% to about 35%. This improves the overall MOS.
D. After optimizing problem sites one by one and take optimization measures at a low carrier-to-interference ratio
(CIR), the ratio of Um interfaces whose quality is at level 0 to level 4 rises 2%.
After taking a series of optimization measures, the MOS value in the drive test is improved obviously. The ratio of
MOS values that are greater than 2.7 rises about 10% and reaches over 95%.
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential
The MOS value of a migrated site is 0.1 less than that of
the original network.
According to the analysis, the smaller MOS value is
mainly caused by abnormal low MOS values. Based on
the problem location and isolation process and the
analysis of the TC recording over the A interface on the
BSC and on the core network, engineers find that the
TC recording is normal on both areas. However, packet
loss occurs on the downlink recording data before the
data enters the A interface and the core network, as
shown in the figure. It is concluded that the PSTN
causes the low MOS value.
The DT data after processing on the PSTN shows that
the MOS values are better than those of the original
network.
Typical Case 3:
The upper figure: Downlink voice sample on the A interface The lower figure: Original voice sample on the PSTN
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential
The TFO fails to be established sometimes when the TFO of a Huawei BSC interconnects with
that of an Ericsson or NSN BSC.
Cause:
On a Huawei BSC earlier than BSC6900 V900R011C00SPC756, the TFO-related protocol content
has not been updated in the implementations and therefore the TFO cannot interconnect with the
TFO of BSCs from other vendors.
Problem Description:
The TFO fails to be established when a Huawei HAMR channel interconnects with an Ericsson
FAMR channel.
Upon receiving the speech version of the FAMR from the Ericsson BSC during a TFO negotiation,
the Huawei BSC decides that the TFO frame type from the Ericsson BSC is AMR_TFO_16k and
enters the TFO establishment process normally. However, the Huawei BSC keeps receiving TFO
frames of AMR_TFO_8+8k from the Ericsson BSC, resulting in failures to establish the TFO.
As defined by the GSM protocols, when the HAMR speech version (excluding 7.95 kbit/s) is used
during the TFO negotiation, the TFO frame type must be AMR_TFO_8+8k. Therefore, Huawei
needs to change the frame search mode to resolve the problem.
Typical Case 4:
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 19
Contents
Evaluation Standards and Principles of Speech
Quality (MOS)
Statistics and Analysis of Factors Affecting the
MOS
Subjective Speech Problem Handling
Voice-Related Key Parameters: Quality Parameters Codec Parameters Handover Parameters AoIP Parameters
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 20
Subjective Speech Problem Handling
Currently, for problems such as one-way audio and noise location, the
main application methods on the BSS side are speech loopback test
and TC recording. The speech loopback function can define the NE
where the problem occurs, while the TC recording function can
determine whether the problem is from the TC and the specific
changes. For the analysis methods of loopback tests and the TC
recording file analysis, see the attachment Operation Guide for Speech
Tests.
The following mainly introduces the handling methods of subjective
speech problems such as one-way audio, noise, echo, crosstalk, and
voice make-and-break.
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1. One-way audio One party of the two call parties cannot hear the voice from the peer end, or both the two parties cannot hear the voice on the
peer end, which is presented as one-way audio or no audio. When one-way audio occurs, mute also occurs.
Handling process: Specify the area scope where the problem occurs and the specific situations. Confirm it is uplink one-way audio (the party
who holds the MS cannot hear any voice, but the one who is on the PSTN side can hear the voice) or downlink one-way
audio (the party holds the MS can hear voices, but the one who is on the PSTN side cannot). Enable the one-way audio detection function (confirm whether the current version supports it or not first), analyze the one-
way detection logs of the whole day, and find out the suspicious resources for dialing test. Perform the dialing test on the
site where the problem occurs. For detailed dialing test procedures, see the attachment Operation Guide for Speech
Tests. During the dialing test, perform TC recording and single user tracking. Perform loopback when the problem
reoccurs, and confirm the NE where the problem occurs. Analyze the trunk performance measurement of the A interface, and fond out abnormal occupation timeslots (Rules: The
A interface has 31 timeslots in total, while the average busy hour of the 31 timeslots is less than 30s, and the number of
timeslots whose average busy hour is less than 30s is at least 28. However, networks charged by second are excluded,
which needs special treatment). Combined with specified CIC dialing test, hardware connection of interfaces, and data
configuration, check whether there are problems such as crossed pair on the A interface or incorrect connection of lines
(In TDM transmission mode, if the E1 line on the A interface is not configured with the SS7 signaling link, or the E1 line on
the Abis interface is not configured with the RSL and OML links, the E1 line on the corresponding port is incorrectly
connected, or no alarm is generated even if crossed pair are made (as long as it is not suspended). However, when users
occupy this port, one-way audio or no audio occurs).
Subjective Speech Problem Handling
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2. NoiseDuring the call, abnormal voices such as bubbles, clicks, and metallic sounds occur. When it is at its worst, only
noise can be heard and the normal speech cannot be heard completely. Usually, noises can be divided into two
types: noise in normal conversation and handover noise. However, noises are mainly caused by bit errors,
including
bit errors caused by frequency interference, voice processing software, and equipment hardware.
Handling process: Specify the area scope where the problem occurs and the specific situations. Select the site where the problem occurs for dialing test. For detailed dialing test procedures, see the
attachment Operation Guide for Speech Tests. During the dialing test, perform TC recording and single user
tracking. Perform loopback when the problem reoccurs, and confirm the NE where the problem occurs. Check the alarm and transmission connection line of the site where the problem occurs to see whether there
is any looseness or damage. Check transmission indexes in the traffic statistics, for example, whether
problems including packet loss, jitter, and too-long delay exist.
3. Speech make-and-breakSpeech make-and-break mainly presents like this: there is a sense of pause in the call, and listeners
may miss half a word or several words. When the make-and-break is obvious, it may affect the normal
conversation.
For the troubleshooting procedures, see the noise handling process.
Subjective Speech Problem Handling
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4. EchoEchoes are mainly divided into two categories: acoustics echoes and electrical echoes. Echoes caused by MS
calling MS are called acoustics echoes, whereas echoes caused by MS calling PSTN are called electrical
echoes.
Handling process: Check whether the handsfree function is enabled or the headset mode is used. Then, check whether the echo
is disappeared or lowered after the handsfree function is disabled or the volume is lowered. Acoustics echoes are usually caused by the noncompliance of isolation of terminals to the protocol
requirements. During the test, adjust the volume of the MS on the peer end. If the echo volume heard on the
local end is obviously changed, it indicates that the echo is produced by the MS on the peer end. You can
change another MS for re-test.
Usually, acoustics echoes are strongly relevant to MSs. The solution to acoustics echoes: Enable the AEC
function on the BSC side to help MSs to further eliminate echoes. Electrical echoes usually caused by configuration or engineering problems. For example, the call routing data
configuration is incorrect, hybrid coils on the fixed network side do not meet the relevant telecom standards,
and the produced echo volume exceeds the processing capability of the echo canceler.
Subjective Speech Problem Handling
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5. CrosstalkIn the call, not only the speech of the called party can be heard, but also a third-party speech can be heard, or the
speech of the called party cannot be heard, instead, a third-party speech is heard. The common reasons for
crosstalk are Um interface crosstalk, core network crosstalk, incorrect data configuration, and abnormal
connection.
Handling process: Encrypt the Um interface: Enabling the Um interface encryption is the root solution to Um interface crosstalk. In BSS data configuration, the configured value of the T3109 timer must be larger than the value configured
in RLT. When the MSC equipment is not Huawei equipment, enable the Call Re-establishment switch on the BSC
side, and set the call re-establishment timer to 45s. Record information such as routing, equipment resources, and transmission about each crosstalk and
analyze them one by one. If it is found that all crosstalks occur in long distance or cross-network (a China
Mobile subscriber calls a China Unicom subscriber) calls, basically it can be concluded that the crosstalk has
something to do with the core network, and the core network engineers need to participate in the fault
location. According to the customers' complaint information, draw the CDR from the MSC and find out the
corresponding CIC to perform the designated dialing test to check whether the CIC timeslot appears
regularly. If it is regularly appears, trouble shoot the hardware connection or data configuration. Meanwhile,
check the data configuration and E1 connection of the problem points.
Subjective Speech Problem Handling
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The table on the
right side lists the
information on
handling subjective
speech problems. It
should be collected
and submitted after
the completion of
the test.
For details, see the
attachment
Checklist for Data
Provided for Speech
Quality Problems
(deliverables).
Subjective Speech Problem HandlingNo. Feedback on Speech Quality Problems Output Description
1Detailed descriptions of the problem, including the scenario in which the problem occurs and the probability that the problem occurs
For example, record whether noises periodically occur (namely whether the noise occurs once every x second (s) or every x minute (s)), and whether noises persist during the calls.
2Tracing signaling of a single user and descriptions of the calling and called MSs
For example, the information about the TEMS of the calling MS is as follows: The MSISDN is 13913140397 and the IMSI is 460512300000397.
3 Log data about the TEMS test Record the test log data when the problem occurs.
4TC recording files and Um interface frame data captured by the Probe
The files and data need to be configured before the test. Probe is a test tool of Huawei and is used to capture the information about the Um interface frames. The TC recording files are in the format of *.dat.
5 Loopback test results about the problemList of loopback test results of interfaces (see the following table))
6BSS version information and information about core network vendors, configuration data and information about test sites
For BSC6900V900R011C00SP720, it is a CME configuration file and a MML configuration file in the format of *.txt. For versions earlier than BSC6900V900R011C00SP720, it is a *.dat file. For the BTS3012, you need to specify whether the new DTRUs or old DTRUs are used. For the BTS3900, you need to specify the types of TRXs.
7Information about the engineering parameter table within the test area
The information is in the format of *.cel that is supported by the Nastar and TEMS.
8 Transmission mode on the entire networkThe transmission mode is all-TDM, all-IP, or hybrid.
9BTS log, alarm log, one-way audio log, and related alarm information about the problem site
Logs and alarm information
10 DSP, DEBUGg, GCSR, and CHR logs of the BSC Log files at the test segments
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Typical Case 1:A user complains that the speech quality is bad under a BTS at a site. The case is that everything is normal
in the outdoor coverage area of this cell, but uplink quality make-and-break and noise occur in the indoor
coverage areas of this BTS.
A. Perform TC recording and make analysis at the problem site, and find that the voice make-and-break exists
before the voice enters into the TC. For
the speech frame structure at the break-and-make
points, the corresponding frames are all No-Data
frames, namely, one or more No-Data frames
appear before two continuous speech frames without
the transition of
SID frame. This usually is caused by the failure in BTS decoding.
B. During the test at the problem site, the uplink noise is extremely serious and continuous. Through the TRX
loopback, the calling party can hear its own speech with make-and-break, it demonstrated that the voice make-and-
break problem exists between the Um interface and the BTS DSP.
C. According to the test log analysis, it is concluded that the quality of the Um interface in the area where the
problem occurs is poor, and the proportion of uplink quality 5, 6, and 7 is 64%, which is the major cause for voice
break-and-make.
After the adjustment of optimization measures for uplink low CIR, the speech problem on this site
disappears after several times of verification.
RxQual 0~4 5~7 RxLev AVGWhole 90.82% 9.18% -79.7Indoor 36.05% 63.95% -89.2
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential
Typical Case 2: A leader of a customer of a office complains that one-way audio exists in a certain probability
during the call, and asks Huawei to solve the problem as soon as possible.
A. Perform the dialing test at the problem site, and perform speech loopback when the problem reoccurs. After the calling party A (external network) has a conversation with the called party B (under the problem site), A cannot hear B, which is
uplink one-way audio. Enable the remote loopback of the A interface on the B side, A can hear his/her own voice, indicating that the problem does not exist on
the MSC side or on routing nodes after the A interface. Enable the local loopback of the A interface on the B side, B cannot hear his/her
own voice, indicating that the problem exists on routing nodes before the A interface on the B side. Enable the BTS speech loopback, B can hear his/her own voice, indicating that the problem exists between the Abis interface and the A
interface, namely the BSC.
B. Analyze the TC recording file, and find that when the one-way
audio occurs, the call works properly when the uplink voice data
enters into the TC. However, when the uplink voice data goes
out of the TC, no voice data is available.
Upon analysis, it is concluded that the BTS sends abnormal
frames, which leads to the TC scheduling
memory error, leading to one-way audio. This problem is
solved after the code optimization.
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Typical Case 2: A user of an overseas office complains that one-way audio and no audio problems occur with a high reoccurrence probability.
Location process:
Because the Abis interface of this office adopts the HDLC transmission, and does not support the one-way audio detection function, the one-way
audio detection cannot be enabled. Upon the analysis on the trunk performance measurement of the A interface, the average occupied duration of
the three ports under the three BSCs is less than 30s, which is far below the average occupied duration of all BSCs. The distribution of occupied
duration has obvious time intervals. In addition, there are more than 28 timeslots that the average occupied duration is less than 30s for each port.
Perform CIC dialing test for specified A interface, the one-way audio occurs. Check the transmission, and find that the E1 lines of the A interface on
the three ports are incorrectly connected. After the transmission is adjusted, the one-way audio does not reoccur.
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OfficeAverage Call Duration
Shortest Call Duration in Normal Cases
Number of BSCs
Thailand 167 62 3
Chengdu 68 36 49
Shantou 80 44 11
Shijiazhuang 68 32 16
Hangzhou 87 30 63
1. In normal cases, the average call duration of all evaluation offices is more than 60s (Upon the analysis on 136 BSCs, there is no
super short call caused by crossed pair).
2. In abnormal cases, the average call duration is less than 29s (Thailand).
3. Upon the analysis on the A interface occupation measurement for 10 BSCs in Nigeria, a large number of trunk occupied durations
of the A interface on the port are less than 30s, with the shortest one is 12.69s. This may be relevant to the strategy of charging by
second in Africa. Therefore, in Africa, the case that checking the A interface connection based on the situation that the A interface
trunk occupies the super short call may be altered according to the actual situation.
Typical Case 2 (continued)
Thailand Chengdu Shantou Shijiazhuang Hangzhou
Average Call Duration Shortest Call Duration in Normal Cases
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Typical Case 3: An AoIP office reports that metallic sounds occur in the existing network, consequently, some users ask
to cancel the network service, provide cause analysis, and immediately solve the problem.
Analyze the TC recording and Probe file, there is no noise in the uplink UM interface voice on the calling
side and in the uplink voice before entering into the core network on the BSC side. However, after the
voice passes through the core network, and when enters into the BSC downlink (the called side uses the
EFR speech version), the noise appear. This, as a result, can be determined that the noise is caused
during the processing of core network.
The core network confirmed that in some cases, the DSP cannot
complete the call processing with 20 ms, and need to re-process it
120 ms later. In the 120 ms, the DSP will send the previous data
again and again, causing the metallic sounds (which is complained
by users) acoustically. After the core network engineers optimize the
scheduling algorithm of the internal DSP, re-test the problem
message on site.
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Contents
Evaluation Standards and Principles of Speech
Quality (MOS) Statistics and Analysis of Factors Affecting the
MOS Subjective Speech Problem Handling Voice-Related Key Parameters:
Quality Parameters Coding Parameters Handover Parameters AoIP Parameters
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Voice-Related Key Parameters:
Quality Parameters Parameters related to improving the speech quality at a low CIR Parameters related to user experience Parameters related to power control Other quality-related parameters
Coding Parameters Speech versions Parameters related to VQE Parameters related to channel allocation
Handover Parameters Handover-related parameters
AoIP Parameters Mapping versions related to AoIP AoIP-related parameters
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Voice-Related Key Parameters – Summary of Quality Parameters
For details about the mapping versions that support voice-related features, see the Reference List of Core Parameters Related to Speech Quality in the attachments.
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Description of the Enhanced Interference Cancellation Combining (EICC) Function:
Among signals received by dual antennas, the interference is related to both the
space (between antennas) and the time. The EICC function considers both the
relationship between interference and space and the relationship between interference
and time. In this way, it suppresses interference more effectively and improves the
voice quality.Suggested Parameter Settings:
STIRC Allowed: Yes when serious interference exists.
Quality Parameters: Parameters Related to Improving the Speech Quality at a Low C/I
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Quality Parameters: User Experience-Related Parameters
VQE: mainly includes four sub-features, such as AEC, ALC, ANR, and ANC.
Description of the Acoustic Echo Cancellation (AEC) Function
This function cancels the acoustic echo generated during the call. It determines
whether the signals input by the local and remote ends are echo to the local end. If
the signals received by the local end are echo from the remote end, this function
attenuates the echo and replaces the echo with comfortable noise. If the signals
received by the local end are voice of the speaker, this function keeps the voice
unchanged.
Suggested Parameter Settings:
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Description of the Adaptive Level Control (ALC) Function:This function controls the level automatically. It evaluates the voice level of the input signals and
controls
the gain of the input signals. Specifically, it adjusts the output voice signals to the target level and
ensures that the level of the signals is stable and the signals can be understood. Therefore, the hearer
thinks that the volume is proper and has a good experience to the voice.
Suggested Parameter Settings:
Description of the Adaptive Noise Reduction (ANR) Function:This function is mainly used to reduce the background noise in the voice without damaging the voice. In
this way, it makes the voice acceptable to the hearer.
Suggested Parameter Settings:
Quality Parameters: User Experience-Related Parameters
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Description of the Automatic Noise Compensation (ANC) Function:
This function compensates the noise automatically. It evaluates the level of
the background noise at the local end and the voice level at the remote end.
When the background nose at the local end is great, this function turns up
the volume of the voice input by the remote end. This improves the signal-to-
noise ratio between the voice at the remote end and the background noise at
the local end. Therefore, the hearer at the local end can hear the voice of the
speaker at the remote end clearly.
Suggested Parameter Settings:
Quality Parameters: User Experience-Related Parameters
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Quality Parameters: Parameters Related to Power Control
Parameters Related to Power Control: III Power Control Algorithm Switch III Power Control Optimization Algorithm Switch
Basic Principle: Power control: When the uplink and downlink signals are strong, reduce the
uplink and downlink transmitting power to reduce the interference of the entire
network. Remarks: The principle of 3.5-generation power control algorithm is advanced in
the industry. This algorithm implements power control based on the quality.
Suggested Parameter Settings: Currently, 3.5-generation power control algorithm is widely promoted globally. It
is required to enable both III power control algorithm switch and III power control
optimization algorithm switch. The 3rd-generation power control algorithm,
however, is not recommended.
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Description of the Counter Function: Radio Link Timeout: This counter defines the time of radio link connection failure for downlink links. The criterion is that whether the
SACCH measurement report can be correctly decoded.
SACCH Multi-Frames: This counter defines the time of radio link connection failure for uplink links. The criterion is that whether the
SACCH measurement report can be correctly decoded.
Suggested Parameter Settings:
Note: The suggested parameter values are to end the call in the case that the UM interface quality is bad,
avoiding continuous impact of continuous bad quality on the speech MOS values. These suggested values
may affect the call drop rate. Therefore, you are advised to use them only when you handle speech
problems. For other KPI handling, see the parameter baselines.
TC CRC Check: According to GSM specifications, the BSC performs CRC check for each uplink data (TRAU frame) from the BTS. If the TRAU frame fails
to pass the CRC check, the BSC regards it as an invalid frame and smoothens it. This avoids the noise caused by parameter
transmission errors and improves the speech quality.
Suggested Parameter Settings: TC CRC Allowed: ON
Quality Parameters: Other Related Parameters
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Voice-Related Key Parameters:
Quality Parameters Parameters related to improving the speech quality at a low CIR Parameters related to user experience Parameters related to power control Other quality-related parameters
Codec Parameters Speech versions Parameters related to VQE Parameters related to channel allocation
Handover Parameters Handover-related parameters
AoIP Parameters Mapping versions related to AoIP AoIP-related parameters
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Voice-Related Key Parameters – Summary of Codec Parameters
Codec Parameters Parameter Name Commend Value Parameter Description
Speech versions FAMR/EFR/HAMR/FR/HR AMRAdjusts the speech coding mode on the uplink and downlink according to changes in Um interface quality, thereby improving the speech quality.
Enhancement of speech quality
TFO Switch ON Reduces the impact of speech transcoding on the speech quality, improving the speech quality.
RTPSWITCHOFF
Specifies whether to enable the delay function to be implemented between the BTS where the calling MS is located and the BTS where the called MS is located.
TFOOptSwitch OFF When the speech versions on the two sides are inconsistent, establish the TFO after optimization.
RATSCCHENABLED OFFSpecifies whether to enable the RATSCCH procedure during a call setup. In the RATSCCH procedure, the rate set of AMR calls can be dynamically adjusted during a call to improve speech quality.
EPLC Switch OFF Compensates the packets that are lost during the transmission.
AMR Uplink Adaptive threshold allowed YESReduce the impact of inaccurate estimation of Signal-to-Noise Ratio (SNR) or the changes in channel conditions following the time on the Adaptive Multi Rate (AMR).
Voice Quality report switch YES Uses the voice quality index (VQI) to monitor the speech quality on the network in real time.
TrFO Switch YES Reduces the impact of TC coding and encoding on the speech quality, improving the speech quality.
Speech Channel Alarm Threshold 10Specifies the threshold for reporting the speech channel alarm. If the number of one-way audio that occurs in an hour on the BSC exceeds this threshold, the speech channel alarm is reported.
Speech Channel Resume Alarm Threshold 6Specifies the threshold for reporting the speech channel resume alarm. If the number of one-way audio that occurs in an hour on the BSC is smaller than this threshold, the speech channel resume alarm is reported.
TCMUTEDETECTFLAG ON Specifies whether to enable the class-1 one-way audio detection function.
MUTECHECKCLASS1PERIOD 5Specifies the class-1 one-way audio detection period. If the FER within the period specified by this parameter exceeds the value of Exceptional Frame Threshold(%), you can infer that one-way audio occurs.
EXCEPFRAMETHRES 25Specifies the threshold for the proportion of the number of bad frames to the total number of TRAU frames. If the FER exceeds this threshold within the value of Period of Mute Detect Class1(s), one-way audio may occur.
MUTECHECKCLASS2SWITCH ONSpecifies whether to enable the one-way audio and no audio detection function to improve the accuracy of one-way report.
DETECTFRAMEPERIOD 2Specifies the period for sending the TRAU test frame after the class-2 one-way audio detection function is enabled. One TRAU test frame is sent in each period until the response from the peer end is received or the timer expires.
MUTECHECKPEIROD 4 Specifies the class-2 one-way audio detection period.
Channel allocation
CHALLOCSTRATEGY CAPABILITY Allocates the channel with good quality, improving the speech quality.
TCHBUSYTHRES 60 Improves properly the proportion of full rate occupation, improving the speech quality.
TCHTRICBUSYOVERLAYTHR 70 Improves properly the proportion of full rate occupation, improving the speech quality.
TCHTRIBUSYUNDERLAYTHR 60 Improves properly the proportion of full rate occupation, improving the speech quality.
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Codec Parameters: Speech Versions
Due to coding differences, different speech versions have different speech quality MOS values. The
following table lists the MOS values at different encoding rates and at different C/Is. Overall, the
MOS in FAMR is greater than that in EFR, the MOS in EFR is greater than that in HAMR, the MOS
in EFR is greater than that in FR, and the MOS in FR is greater than that in HR.
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The key idea of speech version 3 (AMR) is
achieving the best balance between the
speech quality and system capacity by
continuously adjusting the uplink and downlink
voice coding schemes according to the quality
changes of the uplink and downlink signals on
the GSM Um interface.
For the recommended settings of AMR
parameters, see the table on the right.
Note: A bad quality cell indicates a cell where
over 5% of the uplink and downlink receiving
qualities are at levels 5, 6, and 7 according to
the counters or a cell where over 5%
interference bands are interference band 4 and
interference band 5 according to the
interference band statistics.
Codec Parameters: Speech Versions
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Codec Parameters: Speech Versions
Description of AMR gain reflected in MOS and enabling scenarios
Scenario 1:
Compared with the HR speech version, the gain of HAMR speech version reflected in the MOS exists all the time and is obvious.
If the FR and HR speech versions are used on site, the AMR function can be enabled to obtain better speech quality and MOS
values.
Scenario 2:
Compared with the EFR, the MOS gain is mainly reflected in low CIR scenario. When the UM interface quality is good, the gain is not
obvious.
Compared with common FR, in the same scenario, the gain of FAMR on the MOS exists all the time.
If the on-site speech versions include the EFR and HR versions, see the following items for reference to determine whether it needs
to enable the AMR function to obtain better speech quality:
(1) Judge according to the half rate traffic.
If half rate occupation exists in busy hours in the traffic statistics, it is recommended to enable the half rate HAMR (half rate version
3).
(2) Judge according to the receiving quality:
If the Um interface quality is good, usually it is regarded that the proportion of uplink and downlink receiving quality at levels 0-5
exceeds 98%, the MOS gain of the EFR is not obvious after the FAMR is enabled. In this scenario, the FAMR function does not need
to be enabled.
If the receiving quality of the cell is bad, usually it is regarded that the proportion of uplink and downlink receiving quality at levels 5-7
exceeds 5%, the proportion of interference band at levels 4-5 exceeds 5%, or it is found that the quality is bad in the drive test areas,
or the receiving quality is bad at some cells (the proportion of receiving quality at levels 5-7 exceeds 5%), it is recommended to
enable the FAMR function. The MOS gain in this scenario is reflected in the case that the AMR improves the speech quality through
changing the encoding mode of the AMR rate in low CIR.
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Codec Parameters: Speech VersionsDescription of gain to capacity and coverage produced by the AMR and enabling scenarios
Capacity:
Because both the uplink and downlink of the AMR have good immunity from interference, the AMR can tolerate larger identical and
adjacent channel interference, achieving tighter multiplexing of frequency and improving utilization of frequency spectrum resources. In
addition, the speech performance of the HR AMR equals that of the FR AMR. However, the occupied wireless interface bandwidth of it is
only the half of that of the FR AMR. In this case, the AMR can effectively improve the frequency spectrum utilization, increase the system
capacity, and reduce the cost through methods such as tight multiplexing and half rate AMR. When the AMR penetration is 100% through
simulation, the system capacity can increase 140% compared with the EFR speech version.
Coverage:
The AMR code has good immunity from interference. Therefore, the AMR code can support lower CIR compared with non-AMR codes in
the same frame error ratio (FER). In other words, the AMR can make a call in areas where non-AMR codes cannot make a call in the past.
Therefore, the AMR code has better coverage performance, and has large gain for breadth coverage (spacious areas) and depth coverage
(fade areas, shadow areas, and floor and indoor areas).
However, because the robustness of the AMR speech frame and that of the SACCH frame, the actual covering power is decided by that of
the SACCH channel. Therefore, in actual applications, the value of the RLT and the number of SACCH frames shall be set to a larger value
for the AMR channel. In this way, the robustness of the SACCG channel is increased, improving the AMR network coverage performance
and reducing the call drop rate.
To improve the network capacity and coverage, you are advised to enable the AMR function.
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential
Impact of the AMR function on KPIs:
(1) In the existing network of non-AMR speech version, after the
AMR function is enabled, the ratio of uplink receiving quality at
level 6 and level 7 deteriorates.
The causes are as follows:
Different algorithms specified in the protocol lead to the different
results of the AMR and non-AMRs.
The suggested values of AMR parameters and non-AMR parameters
in algorithms such as the handover, power control, and call control
are different, which may also lead to the deterioration of the quality
statistics after the AMR is enabled.
This can be avoided through configuring the AMR parameters and
non-AMR parameters to be consistent with each other in the
algorithm. However, this may lose some gain brought about the AMR
feature. Considering from the overall performance, do not use the
parameter mapping on the right side, unless otherwise to solve the
problem of the deterioration of uplink receiving quality at levels 6 and
7 after the AMR is enabled.
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Codec Parameters: Speech VersionsParameter Mapping
Call Parameter
AFRSAMULFRM The value is identical to that of the SAMULFRM.
AFRDSBLCNT The value is identical to that of the RLT.
AHRSAMULFRM The value is identical to that of the SAMULFRM.
AHRDSBLCNT The value is identical to that of the RLT.
HO Parameter
RXLEVOFF 0
DLQUALIMITAMRFRThe value is identical to that of the DLQUALIMIT.
ULQUALIMITAMRFRThe value is identical to that of the ULQUALIMIT.
DLQUALIMITAMRHRThe value is identical to that of the DLQUALIMIT.
ULQUALIMITAMRHRThe value is identical to that of the ULQUALIMIT.
INTRACELLFHHOEN N/A
PC Parameter
DLAFSREXQUALHIGHTHREDThe value is identical to that of the DLFSREXQUALHIGHTHRED.
DLAFSREXQUALLOWTHREDThe value is identical to that of the DLFSREXQUALLOWTHRED.
DLAHSREXQUALHIGHTHREDThe value is identical to that of the DLHSREXQUALHIGHTHRED.
DLAHSREXQUALLOWTHREDThe value is identical to that of the DLHSREXQUALLOWTHRED.
ULFSREXQUALHIGHTHREDThe value is identical to that of the ULFSREXQUALHIGHTHRED.
ULFSREXQUALLOWTHREDThe value is identical to that of the ULFSREXQUALLOWTHRED.
ULHSREXQUALHIGHTHREDThe value is identical to that of the ULHSREXQUALHIGHTHRED.
ULHSREXQUALLOWTHREDThe value is identical to that of the ULHSREXQUALLOWTHRED.
Channel Parameter
AMRTCHHPRIORLOADThe value is lower than that of the TCHBUSYTHRES.
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential
Because when the statistics of BER of AMR and that of non-BMR are collected, the algorithm is different, which will lead to the
case that the proportion of uplink receiving quality at level 6 and 7 for AMR is larger than that for non-AMRs.
Currently, the calculation accuracy of receiving quality for products is improved by adjusting the corresponding software parameters through
version optimization in BSS9.0. The specific parameters are as follows:
Cell software parameter command: SET GCELLBTSSOFT: IDTYPE=BYID, CELLID=0, ITEMINDEX=57, ITEMVALUE=219;
Note: After the AMR is enabled, the proportion of uplink receiving quality at level 6 and 7 deteriorates, which is only the change in
statistics, and does not affect the actual user perception and UM interface quality.
Other vendors (including Ericsson and Nokia Siemens Networks) also have the problem that the proportion of uplink receiving
quality at level 6 and 7 deteriorates after the AMR is enabled. The figure on the right side lists the comparison data of receiving
quality of Ericsson in existing networks before and after the AMR
is enabled. It can be seen that the proportion of uplink receiving
quality at level 6 and 7 deteriorates, which reduces about 0.5%.
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Codec Parameters: Speech Versions
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential
(2) Due to the difference of statistics formula for the threshold to define that the AMR and non-AMR traffic is
busy, the half rate traffic is reduced after the AMR is enabled.
The problem can be solved by modifying LOADSTATYPE (cell load calculation type) in GBSS9.0. The specific
adjustment is as follows:
SET GCELLCHMGAD: LOADSTATYPE=NODYNPDCH;
However, the problem can be avoided only through lowering the AMRTCHHPRIORLOAD parameter in GBSS8.1.
(3) When the Abis interface is in HDLC transmission mode, due to the allocation of half rate channel bandwidth
algorithm in HDLC mode (The HAMR channel bandwidth is allocated according to the highest rate of HAMR rates
in HDLC mechanism. In this way, when the centralized rate is larger than or equal to 5.9 kbit/s, the bandwidth
exceeds that of the common HR version), the transmission resources occupied by the HAMR after the AMR
function increase. In case that there are much half rate traffic, transmission resource congestion may occur.
(4) The drive test HQI decreases due to statistical differences of the TEMS when the AMR function is enabled. When
the DTX is enabled, the TEMS measures the values of RXQUAL_FULL and RXQUAL_SUB simultaneously. Manual
collection of the HQI only involves the RXQUAL_SUB values. For the non-AMR traffic, the TEMS collects the
RXQUAL_SUB values based on eight voice frames and four SACCH frames. However, the TEMS collects the
RXQUAL_SUB values only based on SACCH frames in the AMR traffic. As a result, the HQI results deteriorate
when the AMR is enabled. On Dingli devices, the same problem occurs because the TEMS collects the
RXQUAL_SUB values based on the SID_UpDATA and SACCH frames when the AMR is enabled. This problem is
caused by a defect in the TEMS and no solution is available.
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Codec Parameters: Speech Versions
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(5) The MOS values have not achieved any gains during the DT when the AMR function is enabled.
The major cause lies in the selection of the test points. This problem mainly occurs along the main streets or roads
where the proportion of the half rate channels is low. In these scenarios, the voice quality almost reaches the
maximum and therefore the AMR function cannot implement the signal gains. The gains of the MOS values can be
achieved during the drive test only when the following requirements are met:
a. Perform a drive test along a route that consists of main roads and side roads.
b. Ensure that the proportion of half rate channel is not less than 15% during the drive test.
(6) The call drop rate increases a little.
The AMR function helps improve the capability of voice frames to resist interference. With the AMR function, call
duration can be prolonged within an area where the signal quality is poor. However, the capabilities of SACCH
frames have not been improved with the AMR function. The determination of call drops is based on the fact
whether the SACCH frames can be correctly demodulated. With the AMR function enabled, the call duration can
be prolonged within an area where the signal quality is poor because users seldom terminate calls. As a result,
there will be a high probability that call drops occur.
To resolve the problem, modify the values of radio link counter AFRDSBLCNT/AHRDSBLCNT and
AFRSAMULFRM/AHRSAMULFRM for AMR calls to ensure that their values are the same as those for non-AMR
calls. In this case, however, the AMR function cannot be fully achieved.
Note: For detailed description, see the 04 GSM BSS AMR Performance Technical Disclosure.
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Codec Parameters: Speech Versions
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Description of the Tandem Free Operation (TFO) Function:
During an MS-MS call, this function reduces one voice coding/decoding session.
This reduces the effect of voice coding on the speech quality and improves the
speech quality.
Suggested Parameter Settings: TFO Switch: Enable Measure Link Delay Switch: Disable Support TFO Codec Optimize: No Is RATSCCH Function Enabled: Disable
Codec Parameters: Parameters Related to VQE
Note:•The GBSS 8.1 and GBSS 9.0 use different TFO mechanisms. You are advised to use a BSC and BTSs of correct mapping versions when the TFO function is enabled. Incorrect version mapping leads to a decrease in TFO gains. •When the TFO function is enabled on the BSC6900 V900R011C00SPC750 or earlier versions, the proportion of half rate channels increases and the proportion of the MOS values greater than 3.0 may decrease. •When the TFO of a Huawei BSC interconnects with that of a BSC from another vendor, ensure that the onsite BSC version is BSC6900 V900R011C00SPC756 or a later version.
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Description of the Enhance Packet Loss Concealment (EPLC) Function: If packet loss occurs during the transmission of voice packets in the network, the speech quality may drop
sharply. This function ensures that the sound effect after the decoding is similar to that before the decoding by
hiding and reducing the effect of packet loss on the speech quality when decoding the received voice packets.
Suggested Parameter Settings: EPLC Switch: Off
Description of the AMR Adaptive Threshold Function: This function reduces the effect of incorrect estimation of the C/I or changes of the channel conditions with
time on the AMR performance. You do not need to estimate the value of this parameter. The value of this
parameter at various rates are adjusted automatically according to the fluctuation of the C/I. In this way, the
AMR performance will not be affected by incorrect estimation of the C/I or by changes of the channel
conditions with time.
Suggested Parameter Settings: AMR Uplink Adaptive Threshold Allowed: Yes
AMR Downlink Adaptive Threshold Allowed: No
Codec Parameters: Parameters Related to Enhancement of Speech Quality
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Description of the Mute Detection Function:
In the TDM scenario, you can set the Mute Detect Class1 Switch parameter to flexibly adjust the detection
conditions and to increase the chance of detecting mute. Then, you can configure the Detect Class2 Switch
parameter to confirm the mutes detected by Mute Detect Class1 Switch to improve the correctness of the
mute report. This function provides additional information for locating problems, preliminarily determines the
problem devices, and narrows the scope of problem devices.
Suggested Parameter Settings:
Codec Parameters: Parameters Related to VQE
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Description of the Voice Quality Index (VQI) Function This algorithm evaluates the speech quality through parameters that calculate the radio transmission quality
over the Um interface. The value of VQI is calculated after the Um interface link parameters are measured.
The value of VQI can correctly reflect the end-to-end quality of voice calls. Therefore, the network operator
can monitor the speech quality of the network in real time without knowing the experience of end users.
Suggested Parameter Settings: Report Speech Quality: Report Report Downlink VQI Allowed: Disable
Description of the Transcoder Free Operation (TrFO) Function: This function prevents the TC from processing voice signals when a voice or multimedia call is established in
the A over IP scenario. As the TC does not exist in physical links for transmitting voice signals when the TrFO
function is enabled, this function reduces the damages to the speech quality by the TC and improves the
speech quality.
Suggested Parameter Settings: This function is defined in 3GPP R4 and needs the support of core network equipment.
Codec Parameters: Parameters Related to VQE
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Codec Parameters: Parameters Related to Channel Allocation
Parameters Related to Channel Allocation: Channel Allocate Strategy Traffic Busy Threshold Tch Traffic Busy Overlay/Underlay Threshold
Basic Principles: The channel allocation strategy helps to allocate high-quality channels and to improve the speech
quality. A full-rate channel is better than a half-rate channel in improving the speech quality. Therefore,
increasing the ratio of full-rate channels helps to improve the overall MOS. Note: When allocating channels, be sure to consider whether congestion exists in the cell. If no
congestion exists, you can set the traffic busy threshold to a larger value.
Suggested Parameter Settings: Channel Allocate Strategy: Quality preferred Traffic Busy Threshold: 60 Tch Traffic Busy Overlay Threshold: 70 Tch Traffic Busy Underlay Threshold: 60
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 55
Voice-Related Key Parameters:
Quality Parameters Parameters related to improving the speech quality at a low CIR Parameters related to user experience Parameters related to power control Other quality-related parameters
Codec Parameters Speech versions Parameters related to VQE Parameters related to channel allocation
Handover Parameters Handover-related parameters
AoIP Parameters Mapping versions related to AoIP AoIP-related parameters
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 56
Voice-Related Key Parameters – Summary of Handover Parameters
Handover Parameters
Parameter Name Commend Value Parameter Description
Handover parameters
HOCTRLSWITCH HOALGORITHM1 Indicates the handover algorithm.
PBGTSTAT(s) 5Sets the triggering time of handover to a proper value to reduce the impact of handover on the speech quality.
PBGTLAST(s) 4Sets the triggering time of handover to a proper value to reduce the impact of handover on the speech quality.
ULDATAFWDTMR 180Delays the release time of the old channel, reducing the number of frames that are lost during the handover.
INTRACELLFHHOEN YES Triggers the AMR TCHF/TCHH handover.
INHOH2FTH 16Enables the half rate to be handed over the full rate, improving the user perception.
INFHHOSTAT(s) 5Sets the triggering time of handover to a proper value to reduce the impact of handover on the speech quality.
INFHHOLAST(s) 4Sets the triggering time of handover to a proper value to reduce the impact of handover on the speech quality.
OPTILAYER SysOptSets the concentric cell-related parameters to proper values to reduce the number of handovers.
HOALGOPERMLAY SysOptSets the concentric cell-related parameters to proper values to reduce the number of handovers.
ACCESSOPTILAY USubcellSets the concentric cell-related parameters to proper values to reduce the number of handovers.
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Handover Parameters: Handover-Related Parameters
Handover-Related Parameters: HO Algorithm Selection Switch PN Rule of PBGT Timer for UL Data Forward
Basic Principle When other KPIs are not affected, it is recommended to increase the PN of PBGT
handover and to delay triggering a handover. This improves the speech quality.
Uplink data smooth timer: This timer delays releasing old channels and
reduce the number of lost frames during the handover. Suggested Parameter Settings:
HO Algorithm Selection Switch: HO I Algorithm PBGTSTAT (s): 5; PBGTLAST (s): 4
Note: The suggested parameter values are to reduce the PBGT handover times, reducing
the impact of handover on voice MOS values. These suggested values will impact on the
success rate of handovers, you are advised to use them only when you handle speech
problems. If you handle other KPIs, see the parameter baselines. Timer for UL Data Forward (ms): 180
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 58
Description of the Intracell F-H HO Function:
For AMR calls, it is allowed to perform AMR full/half rate handover according to the C/I value
of the current call. When the C/I value is great, you can switch full rate to half rate to increase
the traffic capacity. When the C/I value is small, you can switch half rate to full rate to improve
the experience of subscribers.
Suggested Parameter Settings: Intracell F-H HO Allowed: Yes TCH F-H Threshold: 16 Intracell F-H HO Stat. Time (s): 5 Intracell F-H HO Last Time (s): 4
Description of the CoBCCH Access Strategy Function:
To reduce unnecessary handovers between the UL and OL in a concentric cell and to avoid
the damage to voice caused by unnecessary handovers, you need to set the Assign Optimum
Layer and Pref. Subcell in HO of Intra-BSC parameters properly.
Suggested Parameter Settings: Assign Optimum Layer: System optimization Subcell in HO of Intra-BSC : System optimization Incoming-to-BSC HO Optimum Layer: Underlaid subcell
Handover Parameters: Handover-Related Parameters
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 59
Voice-Related Key Parameters Quality Parameters
Parameters related to improving the speech quality at a low CIR Parameters related to user experience Parameters related to power control Other quality-related parameters
Codec Parameters Speech versions Parameters related to VQE Parameters related to channel allocation
Handover Parameters Handover-related parameters
AoIP Parameters Mapping versions related to AoIP AoIP-related parameters
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NE versions on the core network
Mapping Versions Related to AoIP
BSC
BTS
China Areas Outside China SingleRAN MBTSBTS3900 BTS3012 BTS3900 BTS3012
GBSS9.0
BSC6900 V900R011C00SPH732 and later versions
BTS3000 V300R009C00SPC
030 and later versions
BTS3000 V300R009C00SPC
030 and later versions
BTS3000 V100R009C00SPC089 and later versions
BTS3000 V200R009C00SPC002 and later versions
BTS3900 V100R002C00SPC
250 and later versions
GBSS12.0
BSC6900 V900R012C01SPH512 and later versions
Not recommended. Not recommended.BTS3000
V100R012C00SPC042 and later versions
BTS3000 V200R009C00SPC002 and later versions
BTS3900 V100R003C00SPC
350 and later versions
MSC (Huawei)CPCI V100R008C03SPH209 (for areas outside China), CPCI V100R007C05SPH209 (for China), or ATCA V200R008C03SPH209 and later versions
UMG (Huawei) V200R008C03SPH117 and later versions
IPCLK Server (Huawei)
IPCLK1000 V100R002C01SPC200 and later versions
NE Versions on the Wireless Network
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 61
Setting of the wireless network
(1) All MSCs and BSCs are configured with AMR full-rate (FR) codec set 1 (12.20 kbit/s, 7.40 kbit/s, 5.90 kbit/s,
and 4.75 kbit/s) and AMR half-rate (HR) codec set 1 (7.40 kbit/s, 5.90 kbit/s, and 4.75 kbit/s).
(2) All cells are configured with the same speech version and rate set.
(3) Ensure that bit 13 in reserved parameter 3 is set to 0 in versions earlier than BSC6900
V900R011C00SPH728 and set to 1 in BSC6900 V900R011C00SPH728 and later versions.
(4) Ensure that bits 14 and 15 in reserved parameter 3 are set to 00 in GBSS9.0.
In the case of upgrade, however, set this parameter the same as the value before the upgrade. To query the
parameter, run the following command:
LST OTHSOFTPARA: LstFormat=VERTICAL;
Bits 14 and 15 indicate the strategy by which the BSC selects the speech version during intra-BSC handover in
the A over IP mode. The values are as follows:
00: MSC strategy
01: BSC strategy
10: Speech version in originating cell preferred
11: MSC strategy
(5) In GBSS12.0, set the speech version selection policy during intra-BSC handover to MSC Strategy by
running the following command:
SET AITFOTHPARA: CNNODEIDX=XX, SpeechVerStrategyInAss=MSC_STRATEGY ;
(6) Set the timer T25 (INTRABSCCODECHOCMDTIMER) to 5000 ms by running the SET GCELLTMR
command.
AoIP-Related Parameters on the Wireless Network
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Setting of the Huawei core network (for reference)
(1) The A over IP interface in BSC6900V900R011 is
specified in a standard 3GPP protocol. In the
MSC, configure BSC Bearer Type as IPSTD (IP
Type of standard).
AoIP-Related Parameters on the Core Network
(2) On the core network, configure HR AMR and FR AMR codec set as S1. The network-supported HR AMR list includes 12.20 kbit/s, 7.40 kbit/s, 5.90 kbit/s, and 4.75 kbit/s; and the network-supported FR AMR list includes 7.40 kbit/s, 5.90 kbit/s, and 4.75 kbit/s.(3) Set bit 8 in soft parameter P13 to 0 to enable the TrFO compatible with AMR FR/HR. Bit 8 in soft parameter P13 controls the TrFO compatible with AMR FR/HR (UMG). 0: Enables TrFO compatible with AMR FR/HR.1: Disables TrFO compatible with AMR FR/HR. This is the default.(4) Run the following command to set bit 9 in soft parameter P13 to 0:MOD SFP: ID=P13, MODTYPE=BIT, BIT=9, BITVAL=0; Bit 8 in soft parameter P13 controls the optimization for TrFO compatible with AMR FR/HR (UMG).(5) Run the following command to set related parameters: SET UPPARA: RC2CMR=OPEN, CMR2RC=OPEN; For detailed parameter configurations, see the Version Policies and Configuration Requirements for IP-Based GSM V1.26 in the attachments on page 72.
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential
Troubleshooting Procedure for Speech Problems in A over IP
The procedure for troubleshooting speech problems in A over IP
transmission mode is the same as that for other transmission modes
except for the speech version and parameters for the A over IP. The
suggested troubleshooting procedure is as follows:
(1) Check that the version mapping of BSC, BTS, and NEs on the core
network meets the version requirements.
(2) Check that the AoIP-related parameters are correctly configured.
(3) Check that the IP transmission quality meets the QoS.
(4) Based on the preceding check, troubleshoot the problem in the aspect
of the MOS and subjective user experience.
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Typical Case 1: After the A over IP reconstruction is complete at a site, no audio occurs for a short or a long period of
time.
Cause:
The AoIP-related parameters in the wireless network and the core network are not correctly configured
according to the requirements in the Version Policies and Configuration Requirements for IP-Based
GSM V1.26. No audio may occur sometimes because the Um interface quality is not satisfactory.
To use the TrFo function during AMR handover between FR and HR in the A over IP mode, ensure the
following settings:
(1) If the Huawei core network does not support the TrFO during AMR handover between FR and HR,
bit 13 in reserved parameter 3 must be set to 0 in the wireless network. Otherwise, users of MSs on the
half rate channel cannot hear any audio during calls.
(2) If the Huawei core network supports the TrFO during AMR handover between FR and HR, bit 13 in
reserved parameter 3 must be set to 1 by default in the wireless network and bits 8 and 9 in soft
parameter P13 must both be set to 0. Otherwise, no audio may occur for a short period of time.
In both scenarios, the SET UPPARA command must be executed to set RC2CMR and CMR2RC to
OPEN in the core network.
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Typical Case 2: After the A over IP reconstruction is complete and the AMR codec set is modified at a site, the call drop rate
deteriorates.
The AMR codec set before the IP reconstruction consists of the ACS: {12.2 , 10.2 , 7.95 , 5.90} and the
HACS: {7.40 , 6.70 , 5.90}. After the A over IP reconstruction, the AMR full-rate (FR) codec set 1 (12.20 kbit/s,
7.40 kbit/s, 5.90 kbit/s, and 4.75 kbit/s) and AMR half-rate (HR) codec set 1 (7.40 kbit/s, 5.90 kbit/s, and 4.75
kbit/s) are used as defined by the relevant GSM protocol.
The BSC assigns and sends AMR codec set configuration to BTSs in the following ways before and after the
changes in the AMR codec set.
When a BSC in A over IP mode is configured with the standard AMR codec set 1, the MSC compares the
capabilities of the BSC and MSs and then sends the standard AMR codec set 1 (including 12.20 kbit/s, 7.40
kbit/s, 5.90 kbit/s, and 4.75 kbit/s) to BTSs.
When a BSC in A over IP mode is configured with a non-standard AMR codec set, the MSC detects the non-
standard AMR codec set and sends the AMR single codec set to BTSs under the BSC.
Compared with the AMR multiple codec set (including 12.20 kbit/s, 7.40 kbit/s, 5.90 kbit/s, and 4.75 kbit/s), the
AMR single codec set (including only 12.20 kbit/s) cannot enable the BSC to select adaptable codec rates
based on the Um interface quality. As to user experience, when the AMR multiple codec set is configured,
users can maintain calls and call drop rate may increase if the Um interface quality is poor. When the AMR
single codec set is configured, the voice quality will be the worst and users may have to hang up. Therefore,
the call drop rate is different for two different AMR codec sets.
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 66
Typical Case 3: When a Huawei BSS in A over IP mode interconnects with the MSC from other vendors, such
as NSN or ZTE, no audio can be heard sometimes by both parties when the speech version of
any party is half rate.
Onsite BSC version: BSC6900 V900R011SPH722
Cause:
A Huawei BSC processes HR voice frames in a different way from some BSCs supplied by
other vendors. In the A over IP mode, the Huawei BSC encodes HR voice frames into packets
of 14 bytes. However, some BSCs supplied by other vendors have used the latest codec
mechanism in related GSM protocols by adding a TOC byte to the front of the 14 bytes of HR
voice frames. The TOC byte indicates a voice frame, non-voice frame, or null frame. When a
Huawei BSC interconnects with an MSC from another vendor and the HR voice frames
processed by the Huawei BSC are sent to the MSC, the MSC fails the frame format check and
dumps the frames directly. Therefore, the problem occurs.
Solution: Upgrade to the BSC6900 V900R011SPH726
Occurrence scenario: A BSC6900 V900R011SPH726 in A over IP mode interconnects with
the MSC from NSN or ZTE and the HR speech version calls are initiated.
HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 67
Attachments
Checklist for Data Provided for Speech Q
Operation Guide for Speech Tests
Reference List of Core Parameters Relate
Version Policies and Configuration Requir
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