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GSM Speech Quality: Influence Factors + Troubleshooting Methods and Tools + Deliverables
Prepared by: GUL Network I&V and Maintenance Department Reviewed by: Qi Haofeng
www.huawei.com
Huawei Confidential
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.
R&D Support
For speech quality problems, we can provide trainings and 7x24 hour technical support.
Name
Yang Zhengjie (Wireless
Network) Yang Chunjie Feng Lei (Core Network)
Employee ID
00127669
Phone
See the phone book.
00119951 00151560
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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
Objective Evaluation
This method indicates that the score is obtained through comparing the degraded voice file after transmission with the original voice sample file by using a certain algorithm, such as PAMS (ITU-T P 861) and PESQ (ITU-T P 862.1).
Parameter Evaluation
This method indicates that the voice after transmission is not evaluated, and the original voice is not obtained. Instead, the voice after transmission is evaluated through some parameters of wireless transmission network, which has a promising prospect in wireless network, such as RXQUAL, VQI of Huawei, and SQI of E///. Currently, carriers all over the world treat the speech quality as the key indicator for network acceptance. Among them, PESQ algorithm is the widely-used scoring standard. In this algorithm, the PESQ calculation result is mapped into the MOS value, ranging from 1.0 to 4.5.
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PESQ RAW SCORE (the raw score) P.862.1 (the score is obtained through the P.862.1 mapping mode based on the raw score) PESQ-LQ (the score is obtained through the Psytechnics mapping mode) PESQ-Ie (The score is obtained through the mutilation factor of instrumental models defined by P.834) Among them, the value of P.862.1 is widely regarded as the reference value in voice evaluation.
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Contents
Quality (MOS)
Quality Parameters
Codec Parameters
Handover Parameters AoIP Parameters
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Code Code
HO HO
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.
High/low High/low coding rate coding rate Full/half Full/half rate rate
Algorithms Algorithms
Frequent HOs, Frequent HOs, PingPong HOs, PingPong HOs, and unreasonable and unreasonable HOs HOs
Parameters, Parameters, algorithms, and algorithms, and optimizing optimizing strategies strategies
CoBCCH Resident CoBCCH Resident Strategy Strategy HO Optimization Packet HO Optimization Packet Frame Theft of Physical Frame Theft of Physical Messages Messages Too Small HO Hysteresis Too Small HO Hysteresis Inappropriate Inappropriate Neighboring Cell Neighboring Cell Too low PN Rule Too low PN Rule Discarded Packets Discarded Packets Compensation Compensation Long Call Drop Timer Long Call Drop Timer TOP Optimization TOP Optimization
Anti-Interference solution Anti-Interference solution Intermodulation Intermodulation Interference Quick Interference Quick Troubleshooter Troubleshooter
Rate adjust Threshold Rate adjust Threshold Air Interface Quality Air Interface Quality Traffic Traffic Traffic Busy Threshold Traffic Busy Threshold
D V D V T A T A X D X D
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001
Make clear the actual situation of the problem (including average MOS or deterioration conditions of the percentage of high scores, testing methods, and instruments).
During the processing, perform the troubleshooting from the easier to the more advanced following the dashed in red.
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.
008
After the preceding troubleshooting, optimization & adjustment, and verification, if the MOS problems are still not solved, perform the escalation processing.
Analyze the preceding factors and specify the reasons that lead to the MOS problems.
001 No
After the optimization and adjustment, verify whether the problem is solved through drive test.
Yes
End
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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 downlink 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).
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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. If the MOS problem occurs before or after migration (upgrade), check whether the data configuration and transmission mode are changed, for example, whether parameters including cell handover and power control are changed, and whether cells or frequency band is added or reduced. In addition, trouble-shoot the radio frequency channels in the area where problems occur, and check whether KPIs in the traffic transmission are incorrect, which affect the MOS test result. For new-built network, check whether alarm information is displayed on the NEs such as the BTS, BSC, and transmission in the test, which affects the test result. Check whether the network KPIs are abnormal. Usually, KPIs include TCH call drop rate, success rate of wireless handover, and TCH congestion rate.
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TC recording file
UpIn vs. UpOut DnIn vs. DnOut Original sample vs. MS uplink voice data MS downlink voice data vs. Recording (degraded) file
Probe file
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.
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For more detailed feedback information, see the attachment "Checklist for Data Provided for Speech Quality 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%.
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Typical Case 3:
The MOS value of a migrated site is 0.1 less than that of the original network.
The DT data after processing on the PSTN shows that the MOS values are better than those of the original network.
The upper figure: Downlink voice sample on the A interface The lower figure: Original voice sample on the PSTN
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Typical Case 4:
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
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.
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Contents
Quality (MOS)
Quality Parameters
Codec Parameters
Handover Parameters AoIP Parameters
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3. Speech make-and-break
Speech make-and-break mainly presents like this: there is a sense of pause in the call, and listeners may miss half a word or several words. When the make-and-break is obvious, it may affect the normal conversation. For the troubleshooting procedures, see the noise handling process.
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Handling process:
Check whether the handsfree function is enabled or the headset mode is used. Then, check whether the echo is disappeared or lowered after the handsfree function is disabled or the volume is lowered. Acoustics echoes are usually caused by the noncompliance of isolation of terminals to the protocol requirements. During the test, adjust the volume of the MS on the peer end. If the echo volume heard on the local end is obviously changed, it indicates that the echo is produced by the MS on the peer end. You can change another MS for re-test. Usually, acoustics echoes are strongly relevant to MSs. The solution to acoustics echoes: Enable the AEC function on the BSC side to help MSs to further eliminate echoes. Electrical echoes usually caused by configuration or engineering problems. For example, the call routing data configuration is incorrect, hybrid coils on the fixed network side do not meet the relevant telecom standards, and the produced echo volume exceeds the processing capability of the echo canceler.
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Handling process:
Encrypt the Um interface: Enabling the Um interface encryption is the root solution to Um interface crosstalk. In BSS data configuration, the configured value of the T3109 timer must be larger than the value configured in RLT. When the MSC equipment is not Huawei equipment, enable the Call Re-establishment switch on the BSC side, and set the call re-establishment timer to 45s. Record information such as routing, equipment resources, and transmission about each crosstalk and analyze them one by one. If it is found that all crosstalks occur in long distance or cross-network (a China Mobile subscriber calls a China Unicom subscriber) calls, basically it can be concluded that the crosstalk has something to do with the core network, and the core network engineers need to participate in the fault location. According to the customers' complaint information, draw the CDR from the MSC and find out the corresponding CIC to perform the designated dialing test to check whether the CIC timeslot appears regularly. If it is regularly appears, trouble shoot the hardware connection or data configuration. Meanwhile, check the data configuration and E1 connection of the problem points.
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Tracing signaling of a single user and descriptions of the calling and called MSs
Log data about the TEMS test
For example, the information about the TEMS of the calling MS is as follows: The MSISDN is 13913140397 and the IMSI is 460512300000397. Record the test log data when the problem occurs. 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. List of loopback test results of interfaces (see the following table)) 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. The information is in the format of *.cel that is supported by the Nastar and TEMS. The transmission mode is all-TDM, all-IP, or hybrid. Logs and alarm information Log files at the test segments
BSS version information and information about core network vendors, configuration data and information about test sites
7 8 9 10
Information about the engineering parameter table within the test area Transmission mode on the entire network BTS log, alarm log, one-way audio log, and related alarm information about the problem site DSP, DEBUGg, GCSR, and CHR logs of the BSC
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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-andbreak 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. RxQual 0~4 5~7 RxLev AVG
Whole Indoor 90.82% 36.05% 9.18% 63.95% -79.7 -89.2
After the adjustment of optimization measures for uplink low CIR, the speech problem on this site disappears after several times of verification.
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Typical Case 2:
A leader of a customer of a office complains that one-way audio exists in a certain probability during the call, and asks Huawei to solve the problem as soon as possible.
A. Perform the dialing test at the problem site, and perform speech loopback when the problem reoccurs. After the calling party A (external network) has a conversation with the called party B (under the problem site), A cannot hear B, which is uplink one-way audio. Enable the remote loopback of the A interface on the B side, A can hear his/her own voice, indicating that the problem does not exist on the MSC side or on routing nodes after the A interface. Enable the local loopback of the A interface on the B side, B cannot hear his/her own voice, indicating that the problem exists on routing nodes before the A interface on the B side. Enable the BTS speech loopback, B can hear his/her own voice, indicating that the problem exists between the Abis interface and the A interface, namely the BSC. B. Analyze the TC recording file, and find that when the one-way audio occurs, the call works properly when the uplink voice data enters into the TC. However, when the uplink voice data goes out of the TC, no voice data is available. Upon analysis, it is concluded that the BTS sends abnormal frames, which leads to the TC scheduling memory error, leading to one-way audio. This problem is solved after the code optimization.
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Typical Case 2:
A user of an overseas office complains that one-way audio and no audio problems occur with a high reoccurrence probability.
Location process:
Because the Abis interface of this office adopts the HDLC transmission, and does not support the one-way audio detection function, the one-way audio detection cannot be enabled. Upon the analysis on the trunk performance measurement of the A interface, the average occupied duration of the three ports under the three BSCs is less than 30s, which is far below the average occupied duration of all BSCs. The distribution of occupied duration has obvious time intervals. In addition, there are more than 28 timeslots that the average occupied duration is less than 30s for each port. Perform CIC dialing test for specified A interface, the one-way audio occurs. Check the transmission, and find that the E1 lines of the A interface on the three ports are incorrectly connected. After the transmission is adjusted, the one-way audio does not reoccur.
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Average Shortest Call Number of Call Duration in BSCs Duration Normal Cases 167 68 80 68 87 62 36 44 32 30 3 49 11 16 63
Shantou
Shijiazhuang
Hangzhou
Hangzhou
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.
3.
In abnormal cases, the average call duration is less than 29s (Thailand).
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. HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 29
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
Quality (MOS)
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 Speech versions Parameters related to VQE Parameters related to channel allocation Handover-related parameters Mapping versions related to AoIP AoIP-related parameters
Coding Parameters
Handover Parameters
AoIP Parameters
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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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Quality Parameters: Parameters Related to Improving the Speech Quality at a Low C/I
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:
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received by the local end are voice of the speaker, this function keeps the voice
unchanged. Suggested Parameter Settings:
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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.
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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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.
TC CRC Allowed: ON
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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
Handover Parameters
AoIP Parameters
EXCEPFRAMETHRES
25
MUTECHECKCLASS2SWITCH
ON
2 4 CAPABILITY 60 70 60
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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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Impact of the AMR function on KPIs: (1) In the existing network of non-AMR speech version, after the
Call Parameter AFRSAMULFRM AFRDSBLCNT AHRSAMULFRM AHRDSBLCNT HO Parameter RXLEVOFF DLQUALIMITAMRFR ULQUALIMITAMRFR DLQUALIMITAMRHR ULQUALIMITAMRHR INTRACELLFHHOEN PC Parameter DLAFSREXQUALHIGHTHRED DLAFSREXQUALLOWTHRED DLAHSREXQUALHIGHTHRED DLAHSREXQUALLOWTHRED ULFSREXQUALHIGHTHRED ULFSREXQUALLOWTHRED ULHSREXQUALHIGHTHRED ULHSREXQUALLOWTHRED Channel Parameter AMRTCHHPRIORLOAD The value is lower than that of the TCHBUSYTHRES. The value is identical to that of the DLFSREXQUALHIGHTHRED. The value is identical to that of the DLFSREXQUALLOWTHRED. The value is identical to that of the DLHSREXQUALHIGHTHRED. The value is identical to that of the DLHSREXQUALLOWTHRED. The value is identical to that of the ULFSREXQUALHIGHTHRED. The value is identical to that of the ULFSREXQUALLOWTHRED. The value is identical to that of the ULHSREXQUALHIGHTHRED. The value is identical to that of the ULHSREXQUALLOWTHRED. 0 The value is identical to that of the DLQUALIMIT. The value is identical to that of the ULQUALIMIT. The value is identical to that of the DLQUALIMIT. The value is identical to that of the ULQUALIMIT. N/A
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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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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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. HUAWEI TECHNOLOGIES CO., LTD. Huawei Confidential Page 48
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.
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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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
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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hiding and reducing the effect of packet loss on the speech quality when decoding the received voice packets.
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.
AMR Uplink Adaptive Threshold Allowed: Yes AMR Downlink Adaptive Threshold Allowed: No
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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:
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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.
Channel Allocate Strategy: Quality preferred Traffic Busy Threshold: 60 Tch Traffic Busy Overlay Threshold: 70 Tch Traffic Busy Underlay Threshold: 60
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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
Handover Parameters
AoIP Parameters
HOCTRLSWITCH
PBGTSTAT(s) PBGTLAST(s) ULDATAFWDTMR INTRACELLFHHOEN Handover parameters INHOH2FTH INFHHOSTAT(s) INFHHOLAST(s) OPTILAYER HOALGOPERMLAY ACCESSOPTILAY
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Handover-Related Parameters:
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.
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.
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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
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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 Speech versions Parameters related to VQE Parameters related to channel allocation Handover-related parameters
Codec Parameters
Handover Parameters
AoIP Parameters
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GBSS9. 0
BTS3000 BTS3000 BTS3900 BSC6900 BTS3000 BTS3000 V300R009C00SPC V300R009C00SPC V100R002C00SPC V900R011C00SPH73 V100R009C00SPC08 V200R009C00SPC00 030 and later 030 and later 250 and later 2 and later versions 9 and later versions 2 and later versions versions versions versions
BSC6900 BTS3000 GBSS12 V900R012C01SPH51 Not recommended. Not recommended. V100R012C00SPC04 .0 2 and later versions 2 and later versions
BTS3900 BTS3000 V100R003C00SPC V200R009C00SPC00 350 and later 2 and later versions versions
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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:
(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. Huawei Confidential Page 61
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 nonstandard 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.
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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.
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Attachments
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Thank you
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