CN101111046A - Method and system for locating noise problems - Google Patents

Abstract

A noise problem location method and system. The method includes: when the voice data flows through multiple interfaces in the communication system, respectively acquiring the voice data flowing through the multiple interfaces; converting the respectively acquired voice data into voices of the same format Data; comparing each speech data in the same format, and determining the speech frame causing the noise according to the comparison result; and locating the problem according to the determined speech frame causing the noise. The embodiment of the present invention can timely and accurately detect the abnormal frame that causes the noise, and can timely and accurately locate the problem according to the abnormal frame, thereby improving the accuracy of the noise problem location, improving the noise problem location efficiency, and improving the reliability of the equipment. maintainability.

Description

Method and system for locating noise problems

technical field

The invention relates to the technical field of network communication, in particular to a method and system for locating noise problems.

Background technique

With the continuous development of mobile communication technology, mobile users have higher and higher requirements on communication performance such as voice quality. Therefore, when a communication performance problem such as voice quality occurs, it is necessary to quickly locate the problem according to the noise, so as to help solve the noise problem.

At present, the noise problem location methods mainly include the following two types:

Method 1. Use the measurement report.

Obtain the bit error rate in the measurement report. According to the bit error rate, it can be determined whether the noise is caused by the bad wireless environment, so that it can be inferred whether there is a problem with the air interface quality.

Method 2: Use the voice monitoring tool to monitor the voice.

The maintenance personnel use the voice monitoring tool to monitor the call process at the interface between the devices. The maintenance personnel judge whether there is any noise in the monitored voice, and determine whether there is a problem above or below the interface according to the judgment result.

During the research and practice of the prior art, the inventor found that the above prior art has at least the following problems:

The only method is to guess whether there is a problem with the air quality. And because the measurement report is a post-statistical index, therefore, it is impossible to determine which TDMA (TimeDivision Multiple Access, Time Division Multiple Access) frame the noise is caused by using the measurement report, and then it is not possible to perform an operation on the TDMA frame causing the noise. Subsequent problem positioning.

Method 2 needs to manually judge whether there is noise. Different people have different feelings about noise, which will lead to different judgment results, so that the positioning of noise problems is not accurate. Moreover, manual participation will lead to low noise problem location efficiency, thus making the maintainability of the equipment poor. In addition, the implementation principle of the voice monitoring tool is not exactly the same as that of the network equipment in the actual communication system, which further makes the positioning of the noise problem inaccurate. For example, currently commonly used voice monitoring tools can convert the voice frames of the A interface and the Abis interface into voice in real time. The existing voice monitoring tools will not discard voice frames marked as IDLE during the conversion process. However, in actual communication systems The BTS (Base Transceiver Station, Base Transceiver Station) will discard the voice frame marked as IDLE. Perhaps the voice frames dropped by the BTS are causing the noise, but the noise cannot be detected with voice monitoring tools.

Contents of the invention

Embodiments of the present invention provide a method and system for locating noise problems, which can timely and accurately detect abnormal frames that cause noise, and can promptly and accurately locate problems based on the abnormal frames, thereby improving the accuracy of noise problem location and improving This improves the efficiency of locating noise problems and improves the maintainability of equipment.

A method for locating noise problems provided by an embodiment of the present invention includes:

In the case that the voice data flows through multiple interfaces in the communication system, respectively acquiring the voice data flowing through the multiple interfaces;

Converting the respectively acquired voice data into voice data of the same format;

Comparing each voice data in the same format, determining the voice frame that causes noise according to the comparison result;

Problem location is performed according to the determined speech frames that cause noise.

The embodiment of the present invention also provides a noise positioning system, which is applied to a communication system in which voice data flows through multiple interfaces, and the system includes:

Multiple acquisition modules, one acquisition module acquires voice data flowing through an interface;

One or more conversion modules, used to convert the voice data respectively acquired by the acquisition module into voice data of the same format, and output them;

A comparison module, used to compare the voice data in the same format respectively, and output the comparison result;

The locating module is configured to determine the speech frame causing the noise according to the comparison result output by the comparing module, and locate the problem according to the determined speech frame causing the noise.

From the description of the above technical solution, it can be seen that by comparing the voice data obtained by different interfaces, such as bit-level comparison, the abnormal frame that causes noise can be detected in time and accurately; by positioning the problem according to the abnormal frame, it can be timely, Accurately determine the cause of the noise, thereby improving the accuracy of noise problem location, improving the noise problem location efficiency, and finally improving the maintainability of the equipment.

Description of drawings

Fig. 1 is a schematic diagram of obtaining voice data flowing through each interface in an embodiment of the present invention;

Fig. 2 is a schematic diagram of voice data that compares flows through each interface according to an embodiment of the present invention;

Fig. 3 is the schematic diagram of the bit-level comparison result of voice data in the embodiment of the present invention;

FIG. 4 is a schematic diagram of waveform comparison of restored speech frames according to an embodiment of the present invention.

Detailed ways

During a terminal call in a communication system, voice data often flows through multiple interfaces in the communication system, and the smallest unit of voice data is generally a voice frame. If the voice data in multiple interfaces through which the voice data flows is obtained, and the voice data flowing through each interface is compared, for example, a bit-level comparison is performed, the changed voice in the voice data of different interfaces can be determined frame, so that the speech frame that causes the noise can be accurately located; the cause of the speech frame that causes the noise can be accurately located by using the speech frame that causes the noise, so that the noise problem can be accurately located.

The noise problem locating method according to the embodiment of the present invention will be described below.

First, it is necessary to obtain voice data flowing through multiple interfaces of the communication system. The interfaces here can be two, three, four or more interfaces. The interface may be an interface between the terminal and the base station, an interface between the base station and the network-side device through which the voice data flows, or an interface between the network-side devices through which the voice data flows, etc. For example, the interface may be Um interface, Abis interface, Asub interface, A interface, etc.; when applied to a WCDMA system, the interface may also be a Uu interface, an Iu interface, an Iur interface, an Iub interface, etc. in a WCDMA system. The interfaces here are different in different communication systems, and the embodiment of the present invention does not limit the specific expression form of the interfaces, nor does it limit the number of multiple interfaces.

There are many ways to obtain the voice data flowing through the interface. The implementation of the present invention can use various existing voice data capture means, including professional voice data capture tools on the market to obtain the voice data flowing through the interface. Voice data capture can be customized. The embodiment of the present invention does not limit the specific implementation process of acquiring the voice data flowing through the interface.

After acquiring the voice data flowing through multiple interfaces, it is necessary to convert the acquired voice data into voice data in a unified format, so as to facilitate the comparison of the voice data, which is usually at the bit level. The unified format here can be determined according to the specific requirements of voice data comparison. For example, the unified format can be a format supported by the voice frame automatic comparison tool for voice comparison. If the speech frame automatic comparison tool has a speech data format conversion function, the speech data format conversion process in the embodiment of the present invention can be realized by the speech frame automatic comparison tool. Of course, the voice data format conversion process can also be implemented by other devices independent of the voice frame automatic comparison tool. The embodiment of the present invention can adopt the various tools of existing support voice data format conversion to realize voice data format conversion, the embodiment of the present invention does not limit the specific implementation process of voice data format conversion, does not limit the voice data after the format conversion specific format.

The voice data converted from the above format can be stored in different voice files, that is, one interface corresponds to one voice file. Of course, multiple interfaces may also correspond to one voice file, for example, the format-converted voice data corresponding to multiple interfaces are stored in segments in the same voice file.

After voice data format conversion is performed, it is necessary to perform bit-level comparison on voice data corresponding to different interfaces with the same format. The comparison process can be a pairwise comparison. For example, interface 1, interface 2, and interface 3 respectively correspond to voice data 1, voice data 2, and voice data 3 after format conversion, and the pairwise comparison can be voice data 1 and voice data 2, and The comparison between the voice data 2 and the voice data 3 may also be the comparison between the voice data 1 and the voice data 2, and the voice data 1 and the voice data 3. The bit-level comparison here can use various existing automatic speech frame comparison tools, for example, use a sliding window method to compare each speech data in the same format bit by bit. The embodiment of the present invention does not limit the specific implementation process of the voice data bit-level comparison.

The principle of bit-level comparison of voice data in the embodiments of the present invention may include: in the uplink direction, the format-converted voice data corresponding to the interface close to the terminal can be used as a benchmark for comparison; in the downlink direction, the interface close to the core network side can be used The corresponding format-converted speech data is used as a benchmark for comparison.

The difference of voice data can be determined by bit-level comparison. For example, it can be determined whether a voice frame is lost, whether the bits in the voice frame have changed, and whether abnormal phenomena such as splicing frames have occurred, so that the results can be compared according to the bit level. The resulting anomalies directly determine the speech frame causing the noise. Certainly, it is also possible to restore the speech frame in which the abnormal phenomenon occurs to speech, and then determine whether the speech frame in which the abnormal phenomenon occurs is a speech frame causing noise according to the waveform restored to speech.

After the noise-causing voice frame is determined, the network element of the noise-causing voice frame can be determined according to the noise-causing voice frame, so as to accurately locate the noise problem. For example, in the downlink direction, the format-converted speech frames corresponding to the A interface, the format-converted speech frames corresponding to the Asub interface, and the format-converted speech frames corresponding to the Abis interface are the same, but the format-converted speech frames corresponding to the Um interface If the converted speech frame is different from the format-converted speech frame corresponding to other interfaces, it can be basically determined that the speech frame causing the noise is generated by the BTS, and a problem has occurred in the BTS.

The method for locating the noise problem provided by the embodiments of the present invention will be described in detail below by taking two specific noise problem locating as examples and referring to the accompanying drawings 1 , 2 , 3 , and 4 .

Noise problem positioning example 1, frame missing noise problem.

The communication system in Figure 1 includes: MS (Mobile Station, mobile station), BTS, BSC (BaseStation Controller, base station controller) and MSC/VLR (Mobile services Switching Center or Mobile Switching Centre, Mobile Switching Center/Visitor Location Register, Visiting Subscriber Location Register). The interface between MS and base station is UM interface, the interface between BTS and BSC is Abis interface, the interface between BSC and TC (Transcoder, code converter) board is Asub interface, and the interface between TC and MSC/VLR It is the A interface, and the Asub interface is the internal interface. During MS communication, in the uplink direction, the voice data sent by the MS is transmitted to the base station through the Um interface, and the base station transmits the received voice data to the BTS, and the BTS transmits the voice data to the BSC through the Abis interface, and the BSC communicates with the A through the Asub interface. The interface transmits voice data to MSC/VLR. That is to say, voice data in the uplink direction flows through the Um interface, the Abis interface, the Asub interface and the A interface. The flow direction of the voice data in the downlink direction is just opposite to the flow direction of the voice data in the uplink direction.

The embodiment of the present invention can capture voice data at the UM interface, the Abis interface, the Asub interface and the A interface respectively, which can be realized by setting a voice data capture tool. The voice data capture tools here are Huawei T300, NetHawk, K1205, K1297, Huawei MTB and so on. For example, use the capture tool Huawei T300 to capture voice frames flowing through the Um interface, that is, voice data; use NetHawk or K1205 or K1297 or Huawei MTB to capture voice frames flowing through the Abis interface, Asub interface, and A interface respectively.

After the voice data flowing through the UM interface, the Abis interface, the Asub interface and the A interface are obtained respectively by using the voice data capture tool, format conversion is performed on the voice data flowing through each interface. For example, according to the GSM08.60 or 08.61 protocol, the voice data flowing through the Um interface, the Abis interface, the Asub interface, and the A interface are converted into voice data in a unified TRAU frame format or PCM code stream format. The converted voice data can be stored in the voice files corresponding to each interface in the manner shown in Table 1.

Voice data stored in the voice file in Table 1

……#@Time @TS @Subch @UpORDown @SequenceNo @TRAU specific voice frame data#@Time: XXXXXX @TS: YYY Subch: XX @X @YYYY @TRAU XXXXXXXX...@Time: 2007-03 -19 18:22:30.904954 @TS:02 ubch:01 @U @FFFF @TRAU00008808B25FF09BBB96FB10BF4DCB4E9A3E8B02C40D95A7AB3D954590C2F0109629FD65B91AE2FF…

In Table 1, the character "#" indicates the comment line of the behavior, and the character "@" indicates the explanation field, which is to explain the fields in the voice data. If an extension field needs to be added in the speech frame, it is only necessary to add the "@" header to the speech file shown in Table 1 above.

The meaning of each field in Table 1 is shown in Table 2:

Table 2

Field Name meaning length Ranges @Time: XXXXXX date+time 26 @TS: YYY slot number 2 0-999 @Subch: XX sub-slot number 2 0-3 @X up/down 1 U-uplink, D-downlink @YYYY serial number 4 0-FFFF @TRAUXXXXXXXX … Specific voice frame data 80

After the voice data flowing through each interface is converted into a unified format and stored in a voice file, the existing voice frame automatic comparison tool can be used to perform automatic bit-level comparison of the voice data in the voice file and in a unified format, or Bit-level comparisons may also be performed on received user commands. As shown in Fig. 2, the voice frame after the format conversion corresponding to the Um interface, the voice frame after the format conversion corresponding to the Abis interface, the voice frame after the format conversion corresponding to the Asub interface, and the voice frame after the format conversion corresponding to the A interface are respectively Input into speech frame automatic comparison tool. For voice frames in the uplink direction, the voice frame automatic comparison tool can synchronize the format-converted voice frames corresponding to the Abis interface, Asub interface, and A interface with the format-converted voice frames corresponding to the Um interface, that is, to automatically search and locate deal with. Then, the voice frame automatic comparison tool compares the format-converted voice frames corresponding to the Abis interface, the Asub interface, and the A interface with the format-converted voice frames corresponding to the Um interface, bit by bit, using the sliding window technology. The speech frame automatic comparison tool outputs the position of the inconsistent speech frame and the content of the inconsistent speech frame.

For example, the comparison result output by the speech frame automatic comparison tool is shown in FIG. 3 .

The left side of FIG. 3 is the format-converted speech frame corresponding to the Um interface, and the right side of FIG. 3 is the format-converted speech frame corresponding to the Abis interface. The IDLE frame on the right side in Figure 3 is a voice frame that is not sent to the MS, and the SPEECH frame on the right side in Figure 3 is a voice frame that should be delivered to the MS. However, due to the 000017f0 (downlink EFR voice frame The frame header is 000017f0) is incorrectly marked as 00001df0, and the frame header of the IDLE frame is 00001df0, so that the base station recognizes the SPEECH frame as an IDLE frame, causing the base station to discard the SPEECH frame. In this way, one frame is missing in the voice data flowing through the Um interface.

The embodiments of the present invention can convert the inconsistent speech frames obtained by comparison into PCM format code streams to confirm whether there is noise. For example, convert the voice frame of the Abis interface and the voice frame of the Um interface in Figure 3 into a PCM format code stream, and obtain the waveforms of the two PCM format code streams. The waveforms of the two PCM format streams output using the Cool Edit tool are shown in Figure 4. It can be clearly seen from Fig. 4 that the waveform corresponding to the Um interface is a noise waveform. From the sound waveform converted from the inconsistent speech frames, it can be clearly known that there is noise on the MS side.

After it is determined that 000017f0 in the frame header of the SPEECH frame is incorrectly marked as 00001df0, the embodiment of the present invention may determine that a problem occurs with the TC board. In this way, the positioning of the noise problem can be realized in a timely and accurate manner. Furthermore, the embodiment of the present invention can further analyze the reason why the SPEECH frame header is wrongly marked in combination with the GSM0860, 0861 protocol, TRX voice processing module and TC board processing principle.

For voice frames in the downlink direction, the voice frame automatic comparison tool can synchronize the format-converted voice frames corresponding to the Um interface, Abis interface, and Asub interface with the format-converted voice frames corresponding to the A interface, that is, to automatically search and locate deal with. Then, the speech frame automatic comparison tool compares the format-converted speech frames corresponding to the Um interface, Abis interface, and Asub interface with the format-converted speech frames corresponding to the A interface by using the sliding window technology bit by bit. The speech frame automatic comparison tool outputs the position of the inconsistent speech frame and the content of the inconsistent speech frame. The specific implementation process of determining the noise frame and determining the network element generating the noise frame according to the information output by the voice frame field comparison tool is basically the same as the description of the above embodiment, and will not be described in detail here.

Noise problem positioning example 2, switching noise problem.

In the embodiment of the present invention, voice data grabbing tools can be respectively set on the Abis interface of the switched source channel, the Abis interface of the switched target channel, and the Asub interface. The voice data capture tools here are NetHawk, K1205, K1297, Huawei MTB, etc.

After the voice data flowing through the switched source channel Abis interface, the switched target channel Abis interface, and the Asub interface are respectively obtained by using the voice data capture tool, format conversion is performed on the voice data flowing through each interface. For example, according to the GSM08.60 or 08.61 protocol, the voice data flowing through the Abis interface and the Asub interface are converted into voice data in a unified TRAU frame format or PCM code stream format. The converted voice data can be stored in the voice files corresponding to each interface in the manner shown in Table 1.

After the voice data flowing through each interface is converted into a unified format, an automatic bit-level comparison is performed on the voice data in the unified format by using an existing voice frame automatic comparison tool. That is, the format-converted speech frames corresponding to the switched source channel Abis interface and the format-converted speech frames corresponding to the switched target channel Abis interface are respectively synchronized with the format-converted speech frames corresponding to the Asub interface, and use the sliding The window technique performs a bit-by-bit comparison. Set the voice frame automatic comparison tool to output the comparison result as follows:

In the uplink direction, the last voice frame of the switched source channel Abis interface is shown in Table 3;

table 3

Conn: 7 Line: 1 TS: 1 Subch: 3 24124 14:45:10.43674700 00 88 00 A2 29 A9 EE 95 36 F0 23 C6 1A A7 6C B9 62 81 D8 F6 79 B6 75 EC B 0 98A 9 A 2E A D7 65 A4 A9 B6 FF

In the uplink direction, the first voice frame of the switched target channel Abis interface is shown in Table 4;

Table 4

Conn: 16 Line: 1 TS: 4 Subch: 0 24135 14:45:10.43974700 00 88 08 D1 A0 83 9B FC 09 EA 45 CB FB C7 2B A5 E2 D5 22 BA 3E D7 5B D5 1 D4 B 4 E1 36 AD 81 8B 41 80 24 9A FF

In the uplink direction, the voice frame of the Asub interface is shown in Table 5;

table 5

Asub Conn: 4 Line: 1 TS: 1 Subch: 3 4140595 14:45:10.00806300 00 88 00 A2 29 A9 EE 95 36 F0 23 C3 9B FC 09 EA 45 CB FB C7 2B A5 E2 D5 22 B 5 3E D7 E136 81 B6 AD 01 E8 44 8B 41 80 24 9A FF

It should be noted that: 00 00 88 00 is the sync header and C bit of the TRAU frame, indicating that the TRAU frame is an SP frame, that is, a voice frame. 00 00 88 08 is the sync header and C bit of the TRAU frame, indicating that the TRAU frame is a BFI frame, that is, a bad frame.

It can be seen from the above Table 3, Table 4, and Table 5 that the bold part in the voice frame of the Asub interface is "00 0088 00 A2 29 A9 EE 95 36 F0 23 C" and the last voice of the switched source channel Abis interface The bold part in the frame is the same; the italic part in the Asub interface voice frame is "9B FC 09 EA 45 CBFB C7 2B A5 E2 D5 22 BA 3E D7 5B D5 D4 E1 36 81 B6 AD 01 E8 44 8B 41 80 249A FF" It is the same as the italic part in the first voice frame of the Abis interface indicating the data on the target channel and the Asub interface of the switched target channel. Therefore, it can be concluded that the speech frame of the Asub interface in Table 5 is formed by "splicing" the last speech frame of the Abis interface of the source channel and the first speech frame of the Abis interface of the target channel. It is this spliced voice frame that causes noise on the MS side.

After determining that the speech frames flowing through the Asub interface are spliced speech frames, the embodiment of the present invention can determine that a problem has occurred in the BSC. In this way, the positioning of the noise problem can be realized in a timely and accurate manner.

The above example 2 is described for the uplink direction. For the downlink direction, the voice frame automatic comparison tool converts the format-converted voice frames corresponding to the switched source channel Abis interface and the format-converted voice frames corresponding to the switched target channel Abis interface. Synchronize with the format-converted speech frame corresponding to the Asub interface, and use the sliding window technology to perform bit-by-bit comparison. If the format-converted voice frame corresponding to the Asub interface does not appear in the format-converted voice frame corresponding to the switched source channel Abis interface, or the format-converted voice frame corresponding to the switched target channel Abis interface, then the If the frame is lost, it can be determined that there is a problem with the BSC. The specific implementation process is basically the same as the implementation process in the uplink direction, and will not be described in detail here.

The noise problem locating system provided by the embodiments of the present invention will be described below.

The noise problem locating system provided by the embodiments of the present invention is applied to a communication system in which voice data flows through multiple interfaces, and the system includes: multiple acquisition modules, one or more conversion modules, comparison modules and location modules.

An acquisition module acquires voice data flowing through an interface. The interface here may be a UM interface, an Abis interface, an Asub interface, an A interface, and so on. The acquisition modules here can be Huawei T300, NetHawk, K1205, K1297, Huawei MTB, etc.

The conversion module converts the voice data acquired by the acquisition module into voice data of the same format, and outputs them. There can be one or more conversion modules. One conversion module can convert the voice data acquired by one acquisition module, and one conversion module can also convert the voice data acquired by multiple acquisition modules. If there is only one conversion module, the conversion module needs to perform format conversion on the voice data obtained by all the acquisition modules. The format of the converted voice data may be the format of the voice data acquired by one of the acquisition modules, or may be different from the format of the voice data acquired by any one of the acquisition modules. The conversion module and the acquisition module can be set together in the same device, or can be set independently.

The comparison module receives the format-converted voice data output by the conversion module, performs bit-level comparison on the format-converted voice data, and outputs the comparison result. The comparison here may be a pairwise comparison, and there are many specific implementation manners of the pairwise comparison, which are specifically as described in the foregoing method implementation manners. The bit module here can be an automatic comparison tool for existing speech frames. The principles for the comparison module to perform bit-level comparison of voice data may include: in the uplink direction, the converted voice data in the format corresponding to the interface close to the terminal can be used as a benchmark for comparison; in the downlink direction, the voice data corresponding to the interface close to the core network side can be used The format-converted speech data is used as a benchmark for comparison. The comparison result output by the comparison module may be the position of the inconsistent speech frame and the content of the inconsistent speech frame.

The location module receives the comparison result output by the comparison module, and the location module can determine the abnormal frame that causes the noise according to the comparison result output by the comparison module, and locates the problem according to the determined abnormal frame. The positioning module can determine whether the voice frame is lost, whether the bits in the voice frame have changed, whether there are abnormal phenomena such as splicing frames according to the comparison result output by the comparing module, so that the positioning module can directly determine the cause of the abnormal phenomenon according to the abnormal phenomenon. Noisy speech frames. The specific implementation process of the problem location by the location module is as described in the above method implementation.

The positioning module can also restore the speech frame with abnormal phenomenon to speech, and then determine whether the speech frame with abnormal phenomenon is a speech frame that causes noise according to the waveform restored to speech. At this time, the positioning module includes: a reduction sub-module.

The restoration sub-module can restore the different speech frames output by the comparison module to speech, and output the restored speech waveform. The voice waveform output by the restoration sub-module can be in the form of a data file, and at the same time, the voice waveform output by the restoration sub-module can be displayed on the screen.

The positioning module may further include: a positioning sub-module.

The positioning sub-module determines the voice frame that causes noise according to the voice waveform output by the reduction sub-module, for example, the positioning sub-module compares the amplitude of the voice frame waveform in Figure 4, and when it is determined that the waveform amplitude of the abnormal voice frame exceeds the waveform amplitude of another voice frame When the value is predetermined, it is determined that the speech frame in the right diagram of FIG. 4 is the speech frame that causes noise. The locating sub-module performs problem locating according to the determined speech frames that cause noise. The specific implementation process of the problem location by the location module is as described in the above method implementation.

The function of the positioning sub-module can also be accomplished by comparing the waveform of the abnormal speech frame with the predetermined value of the waveform amplitude of another speech frame by manual means such as direct visual inspection. At this time, the positioning sub-module may not be specially set in the system.

Although the present invention has been described by way of example, those of ordinary skill in the art know that there are many variations and changes in the present invention without departing from the spirit of the invention, and the claims of the application document of the present invention include these variations and changes.

Claims (10)

1. A noise problem localization method is characterized in that, comprising: In the case that the voice data flows through multiple interfaces in the communication system, respectively acquiring the voice data flowing through the multiple interfaces; Converting the respectively acquired voice data into voice data of the same format; Comparing each voice data in the same format, determining the voice frame that causes noise according to the comparison result; Problem location is performed according to the determined speech frames that cause noise. 2. The method according to claim 1, wherein said acquiring the voice data flowing through said plurality of interfaces respectively comprises: The voice data flowing through at least two interfaces of the Um interface, the Abis interface, the Asub interface, and the A interface are acquired respectively. 3. The method according to claim 2, wherein said comparing each voice data of said same format comprises: In the uplink direction, the voice data of the same format corresponding to the voice data flowing through at least one of the Abis interface, the Asub interface, and the A interface and the voice data of the same format corresponding to the voice data flowing through the Um interface are bit-coded level comparison; or In the downlink direction, the voice data of the same format corresponding to the voice data flowing through at least one of the Um interface, the Abis interface, and the Asub interface and the voice data of the same format corresponding to the voice data flowing through the A interface are bit-coded level comparison. 4. The method according to claim 1, wherein said acquiring the voice data flowing through said plurality of interfaces respectively comprises: The voice data flowing through the switched source channel Abis interface, target channel Abis interface, and Asub interface are acquired respectively. 5. The method according to claim 4, wherein said comparing each voice data of said same format comprises: Bit-level comparison is performed between the voice data of the same format corresponding to the voice data flowing through the Abis interface of the switched source channel and the Abis interface of the target channel and the voice data of the same format corresponding to the voice data flowing through the Asub interface. 6. method as claimed in claim 1, is characterized in that, the voice data that described described separately acquisition is converted into the voice data of same format respectively comprises: Converting the separately acquired voice data into TRAU or PCM format voice data. 7. The method according to claim 1, wherein said comparing the voice data in the same format comprises: A bit-by-bit comparison is performed on the voice data in the same format by using a sliding window method. 8. the method as claimed in claim 1 or 2 or 3 or 4 or 5 or 6 or 7, is characterized in that, The described speech frame that causes noise is determined according to the comparison result includes: All the compared speech frames that are different are restored to speech, and the speech frame that causes the noise is determined according to the restored speech waveform. 9. A noise location system, applied to a communication system in which voice data flows through a plurality of interfaces, characterized in that the system comprises: Multiple acquisition modules, one acquisition module acquires voice data flowing through an interface; One or more conversion modules, used to convert the voice data respectively acquired by the acquisition module into voice data of the same format, and output them; A comparison module, used to compare the voice data in the same format respectively, and output the comparison result; The locating module is configured to determine the speech frame causing the noise according to the comparison result output by the comparing module, and locate the problem according to the determined speech frame causing the noise. 10. The system according to claim 9, wherein the positioning module comprises: The reduction sub-module is used to restore the different speech frames output by the comparison module to speech, and output the restored speech waveform; Or the positioning module includes: The reduction sub-module is used to restore the different speech frames output by the comparison module to speech, and output the restored speech waveform; The positioning sub-module is used to determine the speech frame causing the noise according to the speech waveform output by the restoring sub-module, and locate the problem according to the determined speech frame causing the noise.

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