CN1625086A - Method and system for detecting voice quality - Google Patents
Method and system for detecting voice quality Download PDFInfo
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- CN1625086A CN1625086A CNA2003101169292A CN200310116929A CN1625086A CN 1625086 A CN1625086 A CN 1625086A CN A2003101169292 A CNA2003101169292 A CN A2003101169292A CN 200310116929 A CN200310116929 A CN 200310116929A CN 1625086 A CN1625086 A CN 1625086A
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Abstract
A testing method of the phonetic quality and its system. The method includes the following processes: the tested terminal communicates with the testing terminal through IP network; at the same time intercept the phonetic message at certain node of the communication line between the testing and the tested terminals; the first tester is the speaker at the testing or the tested terminal; the second tester is the listener at the testing or the tested terminal to listen the remarks of the first tester and to give average opinion score and stop intercepting the phonetic message; transform the phonetic message intercepted into the phonetic documents to broadcast; the second tester gives the second average opinion score according to the phonetic documents broadcasted; comparatively analyze the first and the second average opinion scores to get the influence on the phonetic quality from the node to the tested terminal. Correspondingly this invention also discloses the phonetic quality testing system.
Description
Technical field
The present invention relates to the measuring technology in the communications field, refer to a kind of voice quality method of testing and system especially.
Background technology
(voice-bearer has obtained application more and more widely on IP (Internet Protocol) network for Next Generation Network, NGN) the continuous maturation of technology along with next generation network.Fig. 1 is the simple single domain networking of of NGN, promptly has only a Softswitch (SoftSwitch) 1, and actual networking has a plurality of.Softswitch (SoftSwitch) 1 sets up signal by IP network 2 with the media component with encoding and decoding speech function and is connected.This media component can directly be connected on IP network, comprise: media gateway (Media Gateway, MG) 30 and terminal, terminal comprises session initiation protocol (Session InitiationProtocol, SIP) terminal 320, H.323 terminal 322 and H.248 terminal 323 etc. all be the effect of media stream transition, G.711 media component becomes the user's voice signal encoding, G.729 or the speech frame that G.723.1 waits form on IP, to transmit, simultaneously also on the IP network G.711, G.729 or speech frame G.723.1 be decoded into voice signal.Media gateway 30 is connected with some analog stations 300.
The objective examination method that voice quality is provided at present is the test of carrying out between analog station 300, and method for subjective testing can be between the analog station 300 or sip terminal 320, terminal 322 or H.248 test between the terminal 323 H.323; But all be to test end to end, but for single MG 30 or sip terminal 320, H.323 terminal 322, H.248 the codec of terminal 323 does not have reliable method of testing to the influence of voice.
Above-mentioned method for subjective testing, at (the InternationalTelecommunication Union-Telecommunication Standardization Sector of ITU Telecommunication Standardization Sector, ITU-T) P.800 and description arranged P.830, content roughly is: select 30-40 people and form speech quality evaluation group, test after method for subjective testing giveed training, that trains to the effect that plays to the tester to the different phonetic of prerecording, allowing them that what defines is voice well and bad voice, be exactly some steps when introducing test in addition, such as listening voice earlier, on test paper, fill in suggestion etc. according to different grades then.To the requirement of test environment, the dimension in room preferably satisfies the requirement of 5: 4: 3 (length and width height), and room volume can not be less than 20 cubic metres, reflection interval is less than 500 milliseconds, 200-300 millisecond preferably, background noise should have a needed value, generally chooses 50dBA.The method of subjective testing also has several, the test of talk is arranged, answer evaluation test, also have interview and investigation test, practical method is to answer evaluation test, and promptly the tester listens one section voice that record in advance, provide mean opinion score (MeanOpinion Score then, MOS) value, the test organization person averages this 30-40 people's test result again, draws last MOS value.
As shown in Figure 2, traditional method for subjective testing networking, wherein terminal 32a, b refer to sip terminal 320, H.248 terminal 322 and H.323 in the terminal 323 any one, MG 30a, b can be Trunk Media Gateway (Trunk Media Gateway, TMG), integrated access equipment (Integrated Access Device, IAD) or access media gateway (Access Media Gateway, AMG).
According to actual networking, tester's the first and second the third fourths are carried out combined test in twos in test, and need to prove: one group of tester that tester's first is meant in this position of analog station 300a, tester's second third fourth too.Testing procedure is as follows:
Tester's first is put through the analog station 300b of tester's second do caller;
Behind tester's second off-hook, tester's first is play one section preprepared voice;
Behind the end of conversation, tester's second provides the MOS value;
Repeat above step, draw more MOS value;
The a series of MOS values that draw are carried out statistical analysis;
If test between the different MG or the situation between the terminal as long as replace tester position among Fig. 2.
The shortcoming of traditional method for subjective testing is as follows:
Cost is too high; The member who mainly is assessment panel is too many, and the test room requires too high;
Cycle is oversize; With objective examination method relatively, need select training etc. to the tester, the testing time is oversize, the short-term that does not possess the objective examination is advantage repeatably fast;
A kind of test end to end can not be assessed the independent influence of voice quality the codec of calling and called media gateway.
Please refer to shown in Figure 3ly, Fig. 3 is traditional objective examination method networking diagram.(Voice Quality Tester VQT) 4 generally only supports the analog station interface, so for SIP, H.248, the equipment that H.323 waits terminal directly to be connected on the IP network can not be tested owing to the voice quality tester.MG 30a among Fig. 3 and MG 30b can be TMG, IAD or AMG, and testing procedure is as follows:
(1) connect voice quality tester 4 after, put through called (analog station 300a that links to each other with MG 30a and the analog station 300b that links to each other with MG 30b, one as caller another then as called);
(2) behind the called off-hook, VQT 4 plays an end preprepared voice document automatically;
(3) after broadcast finished, VQT 4 carried out perception assessment (Perceptual evaluationof speech quality, calculating PESQ) of voice quality.
PESQ is one of method of objective appraisal voice quality, and the method for estimating voice quality also comprises perceptual speech quality test (Perceptual Speech Quality Measurement, PSQM) and perception analysis measuring system (Perceptual Analysis Measurement System, PAMS), only need that the change objective examination method get final product on VQT if carry out the calculating of PAMS or PSQM.
The voice quality tester can be supported PESQ, a kind of and several method among PAMS and the PSQM.MG 30a, 30b among Fig. 3 can be TMG, IAD or AMG.It is a kind of measuring technology end to end from the analog station mouth to the analog station mouth, test result is the summation of all component of voice document process to voice quality impacts, can not estimate the influence of voice quality the codec of calling and called media gateway;
Adopt the major defect of traditional objective examination method to be:
1, can only be to equipment such as the TMG that the analog station interface is provided, IAD, AMG etc. test, and other terminal equipment can not be tested;
2, needing special hardware device is the transmission that VQT realizes voice document, receives sample code etc.;
3, cost is very high, if carry out the strange land test, then needs two equipment, and cost is higher;
4, can only test end to end, can not assess the independent influence of voice quality the codec of MG.
Summary of the invention
The problem that the present invention solves provides a kind of voice quality method of testing, realizes the influence to voice quality of test single network assembly or local communication circuit.
For addressing the above problem, a kind of voice quality method of testing of the present invention and system, wherein this method may further comprise the steps: tested end is set up conversation by IP network and test lead; A node place of communication line begins to intercept voice message between tested end and test lead simultaneously; First tester at tested end or test lead as the speaker; Second tester hears out first tester as those who answer is corresponding at test lead or tested end, provides first mean opinion score and stops voice message being intercepted; Convert the voice message of intercepting to the voice document broadcast; Second tester provides second mean opinion score according to the voice document of playing; Comparative analysis first mean opinion score and second mean opinion score draw from this node to tested end the influence of voice quality.
Correspondingly, a kind of phonetic quality testing system of the present invention comprises tested end, IP network, test lead, as speaker's first tester and as those who answer's second tester, and IP network is set up signal with tested end and test lead and is connected; First tester is at tested end or test lead place, then second tester is correspondingly positioned at test lead or tested end place, second tester provides first mean opinion score according to having answered voice at tested end or test lead place, this system also comprises: the packet capturing module, be arranged on the communication line one node place between tested end and test lead, when tested end and test lead are set up conversation, begin to intercept voice message, and when tested end or test lead place have answered voice, stop voice message second tester and intercept; The file conversion module converts the voice message of packet capturing module intercepts to can play voice document; Second tester provides second mean opinion score according to the voice document of playing, and first and second mean opinion score of comparative analysis draws from this node to tested end the influence of voice quality.
Compared with prior art, the present invention has the following advantages:
The present invention can comprise TMG to any gateway, IAD, AMG, sip terminal, H.323 terminal H.248 the codec of terminal etc. carry out the assessment of voice quality impacts degree;
Remedied the shortcoming that traditional subjective and objective method of testing can not be carried out the assessment of voice quality impacts degree to single network assembly or local communication circuit.
Description of drawings
Fig. 1 is a next generation network single domain networking schematic diagram.
Fig. 2 is the networking diagram of voice quality subjective testing in the prior art.
Fig. 3 is the networking diagram of voice quality objective examination in the prior art.
Fig. 4 is the networking diagram of voice quality method of testing of the present invention and system.
Fig. 5 is the block diagram of packet capturing module and file conversion module in the phonetic quality testing system of the present invention.
Embodiment
Please refer to Fig. 4,5, phonetic quality testing system of the present invention comprises that (present embodiment adopts media component 3a to tested end, include media gateway and terminal), (present embodiment adopts media component 3b for IP network 2, test lead, include media gateway and terminal), as speaker's first tester and as those who answer's second tester, IP network is set up signal with tested end and test lead and is connected; First tester is at tested end or test lead place, then second tester is correspondingly positioned at test lead or tested end place, second tester provides first mean opinion score (prior art repeats no more) according to having answered voice at tested end or test lead place, and this system also comprises:
Packet capturing module 5 is arranged on the communication line one node place between tested end and test lead, begins to intercept voice message when tested end and test lead are set up conversation, and stops voice message second tester when tested end or test lead place have answered voice and intercept;
File conversion module 6, the voice message that packet capturing module 5 is intercepted converts the voice document that can play to; Second tester provides second mean opinion score according to the voice document of playing, and first and second mean opinion score of comparative analysis draws from this node to tested end the influence of voice quality.
The node setting can be at tested end any place in the communication line of test lead, can test like this from this node to tested this section of end local line the influence of voice quality, even can finish the influence of test single component to voice quality.
First tester is arranged in tested end (be the media component 3a of present embodiment, have codec), and this node (as position 71) between tested end and IP network, to test the influence of tested end codec encodes function to voice quality.
First tester is arranged in test lead (being the media component 3a of present embodiment), and it (is the media component 3b of present embodiment that this node is arranged in tested end, have codec) and IP network between (as position 72), to test of the influence of tested end codec decoding function to voice quality.
Packet capturing module 5 comprises:
Interface module 51 is used to be provided with source address and the destination address that intercepts voice message
Message module 52, the packet of reception IP network;
Receiver module 53, receive message module 52 data packets transmitted and packet is filtered according to source address and destination address, the voice message that acquisition and source address conform to destination address, described source address is IP (Internet agreement) or MAC (medium access control) source address, and the respective objects address is IP or MAC destination address;
Preserve module 54, receive voice message and storage that receiver module 53 transmits, preserve that module 54 saves as the CAP file format with voice message and in the file conversion module 6 voice document be the wav file form.
In addition, packet capturing module 5 can be that already present packet catcher in the prior art (generally all has the Ethernet message of catching IP address, MAC Address, direction or the agreement etc. of appointment on the network by user's needs, and information such as time of this message and length caught in record, preserve into a CAP file), as netmonitor, sniffer, ethereal or netxray.
The specific descriptions of each module are as follows:
(1) interface module 51 is carried out alternately with the user, receives the source IP address that the user imports, target ip address, or source MAC, and destination-mac address sends receiver module 53 to so that the packet of message module 52 is filtered;
Interface module 51 in addition, can also be provided with as the interface: preserve the voice reporting literary fame (CAP filename in the present embodiment) that module 54 will be preserved; G.711, the voice reporting literary fame that will transform in the file conversion module 6, the voice document name of preservation (wav file name in the present embodiment), code/decode type (are the standards that ITU-T formulates, divide G.711A and two kinds of μ G.711, voice signal is carried out pulse code modulation, and speed is 64kbps), path, source address, destination address or the like.
(2) receiver module 53 receives the packet that 52 transmission of message module come up, source IP address and target ip address or source MAC and destination-mac address by interface module 51, packet is filtered, and satisfactory voice message passed to preserve module 54 and save as the CAP file.
(3) preserve module, function is fairly simple, receives the voice message that receiver module 53 transmits exactly, and preserves according to the CAP filename of interface module 51 inputs; The form of preserving is file header (8 byte file sign, the time of 2 bytes, the month of 2 bytes of 128 bytes, the week of 2 bytes, 2 byte dates, 2 bytes the time, the branch of 2 bytes, the second of 2 bytes, the millisecond of 2 bytes, the file size of 4 bytes, the filler code 0 of 100 bytes), be the voice message (byte number is to catch message length) of catching afterwards), the voice message of still catching then, ..., the last CAP end of file.
(4) message module 52 receives the packet (directly carrying out alternately with the PC network interface card in the present embodiment) of IP network, and data packet delivery to receiver module 53.
(5) file conversion module 6, the voice message that packet capturing module 5 is intercepted converts the voice document that can play to, in the present embodiment CAP file conversion is become wav file.Be provided with by interface module 51: the path and the filename of the path of the CAP file that change and filename and the wav file that will preserve, source IP address, target ip address, source port number, destination port number (as option), code/decode type; To meeting source IP address, target ip address, source port number, the CAP file of destination port number and code/decode type is decoded, and decoded data are preserved into the wav file of 16bit linear PCM coding.
Need to prove that the CAP file can be the CAP file that packet catcher is preserved in the prior art, the flow process of handling these CAP files is the same basically, difference is that the file header and the CAP message format of the CAP file that different packet catchers generates is slightly different: the file header length of the CAP file that ethereal generates is 24 bytes, the form of CAP message is the reserve bytes of 4 bytes, the time of reception of 4 bytes, 4 bytes catch long the exhaling of Ethernet message, 4 bytes catch the Ethernet message length, the Ethernet message of actual acquisition (byte number is to catch the Ethernet message length), ..., the end of file; The file header length of the CAP file that netmonitor generates is 128 bytes, the form of CAP message is the time of reception of 4 bytes, 2 bytes catch the Ethernet message length, 2 bytes catch the Ethernet message length, the Ethernet message of actual acquisition (byte number is to catch the Ethernet message length), ..., the end of file; The file header length of the CAP file that netxray generates is 128 bytes, the form of CAP message is the time of reception of 4 bytes, the reserve bytes of 4 bytes, 2 bytes catch the Ethernet message length, 2 bytes catch Ethernet message length, the reserve bytes of 16 bytes, the Ethernet message of actual acquisition (byte number is to catch the Ethernet message length), ..., the end of file; The file header length of the CAP file that sniffer generates is 128 bytes, the form of CAP message is the time of reception of 4 bytes, the reserve bytes of 4 bytes, 2 bytes catch the Ethernet message length, 2 bytes catch Ethernet message length, the reserve bytes of 28 bytes, the Ethernet message of actual acquisition (byte number is to catch the Ethernet message length), ..., the end of file; File conversion module 6 is handled at different file headers and CAP message.
Correspondingly, voice quality method of testing of the present invention may further comprise the steps:
(1) tested end is set up conversation by IP network and test lead;
(2) simultaneously between tested end and test lead a node place of communication line begin to intercept voice message;
(3) first testers at tested end or test lead as the speaker;
(4) second testers hear out first tester as those who answer is corresponding at test lead or tested end, provide first mean opinion score and stop voice message being intercepted;
(5) voice message with intercepting converts the voice document broadcast to;
(6) second testers provide second mean opinion score according to the voice document of playing;
(7) repeating step (1)-(6) draw a series of first and second mean opinion score;
(8) comparative analysis first mean opinion score and second mean opinion score draw from this node to tested end the influence of voice quality.
Please refer to shown in Figure 4ly, test lead is media component 3a, and tested end has codec for media component 3b, directly is connected with IP network.Packet capturing module 5 can be carried out packet capturing in any position of IP network, if the decoding function of media component 3b codec is tested the influence of voice quality, first tester is positioned at test lead for the speaker, and second tester for those who answer is positioned at tested end, the recommended location of packet capturing module 5 is 72.
The function that the codec realization of tested end media component 3b is decoded to first tester's voice is so packet capturing module 5 operates in position 72 (second tester and position 72 are at same physical location).Testing procedure is as follows:
(1) after conversation was set up at two ends, 72 startup packet capturing modules 5 were carried out packet capturing (voice message intercepting) in the position;
(2) first testers say one section preprepared;
After (3) second testers hear out first tester's words, provide first mean opinion score, stop 5 packet capturings of packet capturing module simultaneously;
(4) utilize file conversion module 6 that the voice message of catching (CAP file) is converted to wav file;
(5) second testers listen wav file on computers, provide second mean opinion score again;
(6) repeat the 2-6 step, obtain a series of first and second mean opinion score and carry out statistical analysis;
(7) statistical analysis and the comparison by first and second mean opinion score, the decoding function of codec that can draw tested end media component 3b is to the situation that influences of voice quality.
If packet capturing module 5 is a packet catcher of the prior art, its step is identical, repeats no more.
Please continue with reference to Fig. 4, if tested end is media component 3a, and the encoding function of codec tested the influence of voice quality, tested end second tester is positioned at test lead media component 3b for those who answer to first tester for the speaker is positioned at, and the recommended location of packet capturing module 5 is 71.
The function that the codec realization of tested end media component 3a is encoded to first tester's voice is so packet capturing module 5 operates in position 71.Testing procedure is as follows:
(1) after conversation was set up at two ends, 71 startup packet capturing modules 5 were carried out packet capturing (voice message intercepting) in the position;
(2) first testers say one section preprepared;
After (3) second testers hear out first tester's words, provide first mean opinion score, stop 5 packet capturings of packet capturing module simultaneously;
(4) utilize file conversion module 6 that the voice message of catching (CAP file) is converted to wav file;
(5) second testers listen wav file on computers, provide second mean opinion score again;
(6) repeat the 2-6 step, obtain a series of first and second mean opinion score and carry out statistical analysis;
(7) statistical analysis and the comparison by first and second mean opinion score, the encoding function of codec that can draw tested end media component 3a is to the situation that influences of voice quality.
If the packet capturing module is operated in any position between position 71 and the position 72, be on the IP network, can test out the subjective assessment of each networking component/local communication circuit to voice quality impacts equally, Test Networking figure as shown in Figure 6, its step is identical, repeats no more.
Claims (14)
1. a voice quality method of testing is characterized in that, this method may further comprise the steps:
(1) tested end is set up conversation by IP network and test lead;
(2) simultaneously between tested end and test lead a node place of communication line begin to intercept voice message;
(3) first testers at tested end or test lead as the speaker;
(4) second testers hear out first tester as those who answer is corresponding at test lead or tested end, provide first mean opinion score and stop voice message being intercepted;
(5) voice message with intercepting converts the voice document broadcast to;
(6) second testers provide second mean opinion score according to the voice document of playing;
(7) comparative analysis first mean opinion score and second mean opinion score draw from this node to tested end the influence of voice quality.
2. voice quality method of testing as claimed in claim 1 is characterized in that, in execution in step (7) before, repeating step (1)-(6) draw a series of first and second mean opinion score.
3. voice quality method of testing as claimed in claim 2 is characterized in that tested end has codec, directly is connected with IP network.
4. voice quality method of testing as claimed in claim 3 is characterized in that first tester is positioned at tested end, and this node is between tested end and IP network, to test the influence of tested end codec encodes function to voice quality.
5. voice quality method of testing as claimed in claim 3 is characterized in that first tester is positioned at test lead, and this node is between tested end and IP network, to test the influence of tested end codec decoding function to voice quality.
6. voice quality method of testing as claimed in claim 3 is characterized in that tested end comprises media gateway and terminal, and this terminal comprises conversation beginning protocol terminal, H.323 terminal and terminal H.248.
A phonetic quality testing system comprise tested end, IP network, test lead, as speaker's first tester and as those who answer's second tester, IP network is set up signal with tested end and test lead and is connected; First tester is at tested end or test lead place, and then second tester is correspondingly positioned at test lead or tested end place, and second tester provides first mean opinion score according to having answered voice at tested end or test lead place, it is characterized in that this system also comprises:
The packet capturing module is arranged on the communication line one node place between tested end and test lead, begins to intercept voice message when tested end and test lead are set up conversation, and stops voice message second tester when tested end or test lead place have answered voice and intercept;
The file conversion module converts the voice message of packet capturing module intercepts to can play voice document; Second tester provides second mean opinion score according to the voice document of playing, and first and second mean opinion score of comparative analysis draws from this node to tested end the influence of voice quality.
8. phonetic quality testing system as claimed in claim 7 is characterized in that, the packet capturing module comprises:
Interface module is used to be provided with source address and the destination address that intercepts voice message
The message module, the packet of reception IP network;
Receiver module receives message module data packets transmitted and according to source address and destination address packet is filtered, the voice message that acquisition and source address conform to destination address;
Preserve module, receive voice message and storage that receiver module transmits.
9. phonetic quality testing system as claimed in claim 8 is characterized in that, described source address is IP or mac source address, and the respective objects address is IP or MAC destination address.
10. phonetic quality testing system as claimed in claim 9 is characterized in that, preserve that module saves as the CAP file format with voice message and in the file conversion module voice document be the wav file form.
11. phonetic quality testing system as claimed in claim 7 is characterized in that, the packet capturing module is netmonitor, sniffer, ethereal or netxray.
12. as any described phonetic quality testing system of claim 7-10, it is characterized in that tested end has codec, directly be connected with IP network.
13. voice quality method of testing as claimed in claim 12 is characterized in that first tester is positioned at tested end, and this node is between tested end and IP network, to test the influence of tested end codec encodes function to voice quality.
14. voice quality method of testing as claimed in claim 13 is characterized in that first tester is positioned at test lead, and this node is between tested end and IP network, to test the influence of tested end codec decoding function to voice quality.
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Cited By (6)
Publication number | Priority date | Publication date | Assignee | Title |
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CN101141768B (en) * | 2007-10-12 | 2010-06-16 | 中兴通讯股份有限公司 | Test method and device for simulating voice quality variation aroused by network characteristic |
CN102208990A (en) * | 2010-03-29 | 2011-10-05 | 华为技术有限公司 | Voice quality analysis method and device |
CN101534353B (en) * | 2009-03-31 | 2012-01-04 | 华为技术有限公司 | Wireless network vocal quality measuring method and terminal thereof |
CN102595467A (en) * | 2011-01-04 | 2012-07-18 | 中国移动通信集团公司 | Data acquisition method and data acquisition device |
WO2014029291A1 (en) * | 2012-08-24 | 2014-02-27 | 华为技术有限公司 | Method, network element and system for evaluating voice quality |
CN106714226A (en) * | 2015-11-13 | 2017-05-24 | 中国移动通信集团公司 | Voice quality evaluation method, device and system |
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Publication number | Priority date | Publication date | Assignee | Title |
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CN101141768B (en) * | 2007-10-12 | 2010-06-16 | 中兴通讯股份有限公司 | Test method and device for simulating voice quality variation aroused by network characteristic |
CN101534353B (en) * | 2009-03-31 | 2012-01-04 | 华为技术有限公司 | Wireless network vocal quality measuring method and terminal thereof |
CN102208990A (en) * | 2010-03-29 | 2011-10-05 | 华为技术有限公司 | Voice quality analysis method and device |
WO2011120367A1 (en) * | 2010-03-29 | 2011-10-06 | 华为技术有限公司 | Method and device for analyzing voice quality |
CN102208990B (en) * | 2010-03-29 | 2013-10-09 | 华为技术有限公司 | Voice quality analysis method and device |
CN102595467A (en) * | 2011-01-04 | 2012-07-18 | 中国移动通信集团公司 | Data acquisition method and data acquisition device |
CN102595467B (en) * | 2011-01-04 | 2014-09-10 | 中国移动通信集团公司 | Data acquisition method and data acquisition device |
WO2014029291A1 (en) * | 2012-08-24 | 2014-02-27 | 华为技术有限公司 | Method, network element and system for evaluating voice quality |
CN103632680A (en) * | 2012-08-24 | 2014-03-12 | 华为技术有限公司 | Speech quality assessment method, network element and system |
CN103632680B (en) * | 2012-08-24 | 2016-08-10 | 华为技术有限公司 | A kind of speech quality assessment method, network element and system |
US9641673B2 (en) | 2012-08-24 | 2017-05-02 | Huawei Technologies Co., Ltd. | Method, network element, and system for assessing voice quality |
CN106714226A (en) * | 2015-11-13 | 2017-05-24 | 中国移动通信集团公司 | Voice quality evaluation method, device and system |
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