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Page 5
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.
Page 6
Contents
Voice
Voicequality
quality
The speech quality is mainly related
to three factors: code, bit error, and
handover (HO). The coding factor
Directly-related
Directly-related Bit
Biterror
error(frame
(frame benefits the speech quality. The bit
Code
Code HO
HO
factors
factors erasure)
erasure) error and handover factors,
however, damage the speech
quality.
Indirectly-
optimization
Engineering
coding
Interference
Algorithms
Indirectly- Frequent
FrequentHOs,
optimization
Engineering
codingrate
Call
Interference
Algorithms
High/low
HOs,
network
Full/half
Speech
version
Calldrop
High/low
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Full/half
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related
relatedfactors To optimize the speech quality, you
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factors HOs,
rate
and
andunreasonable need to select reasonable codes
drop
unreasonable
rate
HOs
HOs and reduce the effect of the bit error
rate (BER) and handovers on the
speech quality.
HO Optimization Packet
Frame Theft of Physical
Parameters,
3.5-Generation Power
Threshold Self-Adaptive
HO Optimization Packet
Long Call Drop Timer
is normal.
Air Interface Quality
3.5-Generation Power
algorithms,
algorithms,and
Long Call Drop Timer
F2H HO Threshold
Discarded Packets
CoBCCH Resident
Interference Quick
TOP Optimization
Air Interface Quality
Neighboring Cell
F2H HO Threshold
Discarded Packets
CoBCCH Resident
Interference Quick
Intermodulation
TOP Optimization
Troubleshooter
Compensation
optimizing DD VV
Compensation
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Messages
optimizing
Strategy
Messages
The methods for improving the call
Control
Traffic
Strategy
strategies
Control
TT AA
Traffic
strategies
drop rate and handover success
XX DD
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
The handover success rate.
Thechannel
channelisisnormal.
normal.
For more detailed feedback information, see the attachment "Checklist for Data Provided for
Speech Quality Problems".
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%.
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.
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).
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.
After the adjustment of optimization measures for uplink low CIR, the speech problem on this site
disappears after several times of verification.
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.
Shijiazhuang 68 32 16
Thailand Chengdu Shantou Shijiazhuang 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. 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.
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.
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
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.
VQE: mainly includes four sub-features, such as AEC, ALC, ANR, and ANC.
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.
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
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
Specifies whether to enable the RATSCCH procedure during a call setup. In the RATSCCH procedure, the rate
RATSCCHENABLED OFF
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.
Reduce the impact of inaccurate estimation of Signal-to-Noise Ratio (SNR) or the changes in channel conditions
AMR Uplink Adaptive threshold allowed YES
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.
Specifies the threshold for reporting the speech channel alarm. If the number of one-way audio that occurs in an
Speech Channel Alarm Threshold 10
hour on the BSC exceeds this threshold, the speech channel alarm is reported.
Enhancement of speech quality
Specifies the threshold for reporting the speech channel resume alarm. If the number of one-way audio that
Speech Channel Resume Alarm Threshold 6
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.
Specifies the class-1 one-way audio detection period. If the FER within the period specified by this parameter
MUTECHECKCLASS1PERIOD 5
exceeds the value of Exceptional Frame Threshold(%), you can infer that one-way audio occurs.
Specifies the threshold for the proportion of the number of bad frames to the total number of TRAU frames. If the
EXCEPFRAMETHRES 25
FER exceeds this threshold within the value of Period of Mute Detect Class1(s), one-way audio may occur.
Specifies whether to enable the one-way audio and no audio detection function to improve the accuracy of one-
MUTECHECKCLASS2SWITCH ON
way report.
Specifies the period for sending the TRAU test frame after the class-2 one-way audio detection function is enabled.
DETECTFRAMEPERIOD 2
One TRAU test frame is sent in each period until the response from the peer end is received or the timer expires.
(1) In the existing network of non-AMR speech version, after the AFRSAMULFRM The value is identical to that of the SAMULFRM.
are different, which may also lead to the deterioration of the quality The value is identical to that of the
ULQUALIMITAMRHR
ULQUALIMIT.
statistics after the AMR is enabled. INTRACELLFHHOEN N/A
This can be avoided through configuring the AMR parameters and PC Parameter
The value is identical to that of the
non-AMR parameters to be consistent with each other in the DLAFSREXQUALHIGHTHRED
DLFSREXQUALHIGHTHRED.
The value is identical to that of the
algorithm. However, this may lose some gain brought about the AMR DLAFSREXQUALLOWTHRED
DLFSREXQUALLOWTHRED.
The value is identical to that of the
feature. Considering from the overall performance, do not use the DLAHSREXQUALHIGHTHRED
DLHSREXQUALHIGHTHRED.
The value is identical to that of the
parameter mapping on the right side, unless otherwise to solve the DLAHSREXQUALLOWTHRED
DLHSREXQUALLOWTHRED.
The value is identical to that of the
problem of the deterioration of uplink receiving quality at levels 6 and ULFSREXQUALHIGHTHRED
ULFSREXQUALHIGHTHRED.
The value is identical to that of the
7 after the AMR is enabled. ULFSREXQUALLOWTHRED
ULFSREXQUALLOWTHRED.
The value is identical to that of the
ULHSREXQUALHIGHTHRED
ULHSREXQUALHIGHTHRED.
The value is identical to that of the
ULHSREXQUALLOWTHRED
ULHSREXQUALLOWTHRED.
Channel Parameter
The value is lower than that of the
AMRTCHHPRIORLOAD
TCHBUSYTHRES.
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%.
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.
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
Handover Enables the half rate to be handed over the full rate,
INHOH2FTH 16
parameters improving the user perception.
Sets the triggering time of handover to a proper value to
INFHHOSTAT(s) 5
reduce the impact of handover on the speech quality.
Sets the triggering time of handover to a proper value to
INFHHOLAST(s) 4
reduce the impact of handover on the speech quality.
Sets the concentric cell-related parameters to proper
OPTILAYER SysOpt
values to reduce the number of handovers.
Sets the concentric cell-related parameters to proper
HOALGOPERMLAY SysOpt
values to reduce the number of handovers.
Sets the concentric cell-related parameters to proper
ACCESSOPTILAY USubcell
values to reduce the number of handovers.
BTS
BSC China Areas Outside China SingleRAN
BTS3900 BTS3012 BTS3900 BTS3012 MBTS
BTS3900
BSC6900 BTS3000 BTS3000
GBSS12 V900R012C01SPH51 Not recommended. Not recommended. V100R012C00SPC04 V200R009C00SPC00
V100R003C00SPC
.0 350 and later
2 and later versions 2 and later versions 2 and later versions
versions
IPCLK Server
IPCLK1000 V100R002C01SPC200 and later versions
(Huawei)
For detailed parameter configurations, see the Version Policies and Configuration Requirements
for IP-Based GSM V1.26 in the attachments on page 72.