CN103517287A - An end-to-end bidirectional voice quality test method, a system and a voice test server - Google Patents
An end-to-end bidirectional voice quality test method, a system and a voice test server Download PDFInfo
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- CN103517287A CN103517287A CN201210201755.9A CN201210201755A CN103517287A CN 103517287 A CN103517287 A CN 103517287A CN 201210201755 A CN201210201755 A CN 201210201755A CN 103517287 A CN103517287 A CN 103517287A
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Abstract
The invention provides an end-to-end bidirectional voice quality test method, a system and a voice test server. The test method comprises the following steps: when a calling subscriber terminal transmits media stream files to a called subscriber terminal, the voice quality test server obtains first network status information transmitted by the calling subscriber terminal. The first network status information expresses a network status of a voice channel from the calling subscriber terminal to the called subscriber terminal. When the called subscriber terminal makes the received preset media stream files go back to the calling subscriber terminal in a loop mode, the voice quality test server obtains second network status information transmitted by the called subscriber terminal. The second network status information expresses a network status of a voice channel from the called subscriber terminal to the calling subscriber terminal, so that end-to-end bidirectional network parameters are obtained.
Description
Technical field
The present invention relates to communication technical field, relate in particular to a kind of IMS(IP Multimedia Subsystem, IP Multimedia System) method of testing, system and the tone testing server of end-to-end double-directional speech quality in network.
Background technology
IMS architectural framework has strengthened the controlled and manageability based on IP stream, has represented the development trend of network.Each aspect of the networks such as the introducing of IMS and access, carrying, control, business, support is relevant, also terminal transformation, the development tactics in broadband, operation mode are brought challenges, so each operator need suit measures to local conditions to be selected according to demand simultaneously.Realizing fusion is a long-term target, and the factor relating to is a lot.IMS just provides the possibility of integration technology at network and service layer, but also needs conscientiously to study its feasibility, economy, complexity and manageability.
Because IMS bearer network is based on IP group technology, but the stability of IP metropolitan area network can not effectively guarantee VoIP voice quality aspect time delay, packet loss and shake.Although IETF(Internet Engineering Task Force, Internet Engineering task groups) formulated the RTCP(RTP Control Protocol that affects several key indexs of IP network voice quality (time delay, packet loss and shake), RTP Control Protocol) standard.
But, according to RTCP standard, can not accurately judge call double-directional speech quality, this is because initial RTCP standard code call folk prescription only can gather opposite end to the network parameter of local terminal.
Summary of the invention
In order to solve the problems of the technologies described above, to the invention provides a kind of method of testing, system and tone testing server of end-to-end double-directional speech quality, thereby can obtain end-to-end bilateral network parameter.
In order to achieve the above object, the invention provides a kind of method of testing of end-to-end double-directional speech quality, when carrying out voice quality test, described method of testing comprises:
When call subscriber terminal is preset media stream file to called user terminal transmission, voice quality testing server obtains the first network condition information that described call subscriber terminal sends, and described first network condition information represents that described call subscriber terminal is to the network condition of the voice channel of described called user terminal;
Described called user terminal by the described default media stream file loopback receiving when the described call subscriber terminal, described voice quality testing server obtains the second network condition information that described called user terminal sends, and described second network condition information represents that described called user terminal is to the network condition of the voice channel of described call subscriber terminal.
Preferably, described method of testing also comprises:
Described voice quality testing server, according to described first network condition information and second network condition information, obtains the end-to-end two-way network condition information of this call between described call subscriber terminal and described called user terminal.
Preferably, at described call subscriber terminal, receive after whole loopback media stream file, described method of testing also comprises:
Described voice quality testing server receives the loopback media stream file that described call subscriber terminal is uploaded;
Described voice quality testing server, according to the loopback media stream file and the described default media stream file that receive, obtains the voice quality MOS mean opinion score value of this call between described call subscriber terminal and called user terminal.
Preferably, after obtaining the voice quality MOS value of this call, described method of testing also comprises:
Described voice quality testing server, according to the voice quality MOS value of described end-to-end two-way network condition information and described this call, is determined the fault point of the voice quality reduction that causes this call.
Preferably, between described voice quality testing server and described call subscriber terminal and described called user terminal, maintain default heartbeat mechanism.
Preferably, when voice quality test starts, described method of testing also comprises:
Described voice quality testing server sends tone testing order to described call subscriber terminal, and described call subscriber terminal establishes call connection according to described tone testing order and described called user terminal;
After call connection is set up, the voice channel that described voice quality testing server connects bearing call carries out voice quality test.
In order to achieve the above object, the present invention also provides a kind of voice quality testing server, comprising:
The first acquisition module, for when call subscriber terminal is preset media stream file to called user terminal transmission, obtain the first network condition information that described call subscriber terminal sends, described first network condition information represents that described call subscriber terminal is to the network condition of the voice channel of the bearing call connection of described called user terminal;
The second acquisition module, for described called user terminal by the described default media stream file loopback receiving when the described call subscriber terminal, obtain the second network condition information that described called user terminal sends, described second network condition information represents that described called user terminal is to the network condition of the voice channel of the bearing call connection of described call subscriber terminal.
Preferably, described voice quality testing server also comprises:
The first processing module, for according to described first network condition information and second network condition information, obtains the end-to-end two-way network condition information of this call of described call subscriber terminal and described called user terminal.
Preferably, described voice quality testing server also comprises:
Receiver module, the loopback media stream file of uploading for receiving described call subscriber terminal;
The second processing module, for according to the described loopback media stream file and the described default media stream file that receive, obtains the voice quality MOS mean opinion score value of this call between call subscriber terminal and called user terminal.
Preferably, described voice quality testing server also comprises:
The 3rd processing module, for according to the voice quality MOS value of described end-to-end two-way network condition information and described this call, determines the fault point of the voice quality reduction that causes this call.
Preferably, between described voice quality testing server and described call subscriber terminal and described called user terminal, maintain default heartbeat mechanism.
Preferably, described voice quality testing server also comprises:
Tone testing module, for sending tone testing order to described call subscriber terminal, described call subscriber terminal establishes call connection according to described tone testing order and described called user terminal; After call connection is set up, the voice channel that bearing call is connected carries out voice quality test.
In order to achieve the above object, the present invention also provides a kind of test macro of end-to-end double-directional speech quality, comprising:
Tone testing server as described above and at least two user terminals, wherein said tone testing server is connected with described user terminal respectively.
As shown from the above technical solution, embodiments of the invention have following beneficial effect: first, when call subscriber terminal is preset media stream file to called user terminal transmission, voice quality testing server obtains the first network condition information that call subscriber terminal sends, and wherein first network condition information represents that call subscriber terminal is to the network condition of the voice channel of the bearing call connection of called user terminal; At called user terminal by the default media stream file loopback receiving during to call subscriber terminal, voice quality testing server obtains the second network condition information that called user terminal sends, second network condition information represents that called user terminal arrives the network condition of the voice channel that the bearing call of described call subscriber terminal connects, thereby can obtain the situation of end-to-end bilateral network quality, overcome conventional art and only can gather opposite end to the deficiency of the network quality situation of local terminal;
Secondly, the loopback media stream file that call subscriber terminal sends the called user terminal receiving is uploaded to voice quality testing server, by voice quality testing server, according to loopback media stream file and default media stream file, calculated the voice quality MOS(Mean Opinion Score of this call, mean opinion score) value, voice quality testing server can be according to the comparing result of the voice quality MOS value of this call and end-to-end bilateral network quality condition, and whether tentative diagnosis called end equipment exists the faults such as user's plate noise.
Again, voice quality testing server can obtain the network condition of institute's supervising the network, when network variation, can notify in time O&M personnel by modes such as alarms.
Accompanying drawing explanation
Fig. 1 represents the method for testing flow chart of end-to-end double-directional speech quality in embodiments of the invention;
Fig. 2 represents the sequential chart of the method for testing of end-to-end double-directional speech quality in embodiments of the invention;
Fig. 3 represents the structured flowchart of voice quality testing server in embodiments of the invention;
Fig. 4 represents the structure chart of the test macro of end-to-end double-directional speech quality in embodiments of the invention.
Embodiment
In an embodiment of the present invention, when call subscriber terminal is preset media stream file to called user terminal transmission, voice quality testing server obtains the first network condition information that call subscriber terminal sends, and wherein first network condition information represents that call subscriber terminal is to the network condition of the voice channel of the bearing call connection of called user terminal; At called user terminal by the default media stream file loopback receiving during to call subscriber terminal, voice quality testing server obtains the second network condition information that called user terminal sends, and second network condition information represents that called user terminal arrives the network condition of the voice channel that the bearing call of described call subscriber terminal connects.
For making the object of the embodiment of the present invention, technical scheme and advantage are clearer, below in conjunction with the accompanying drawing in the embodiment of the present invention, technical scheme in embodiments of the invention is clearly and completely described, obviously, described embodiment is a part of the present invention, rather than whole embodiment.Embodiment based in the present invention, those of ordinary skills, not making the every other embodiment obtaining under creative work prerequisite, belong to the scope of protection of the invention.
Referring to Fig. 1, be the method for testing flow chart of end-to-end double-directional speech quality in embodiments of the invention, concrete steps are as follows:
In an embodiment of the present invention, call subscriber terminal receives after tone testing order, and call subscriber terminal establishes call connection according to tone testing order and called user terminal, wherein in this tone testing order, carries the address information of called user terminal.
In an embodiment of the present invention, call subscriber terminal and called user terminal, except having voice call capability, also possess the ability of voice quality test simultaneously.This voice quality testing server is initiated voice quality test, the analysis of voice channel to cause the voice that voice quality reduces for controlling user terminal, and is determined concrete node or the path that causes that voice quality reduces.
In an embodiment of the present invention, for example, due to voice quality testing server (IADMS, integrated Access Device Management System) need to initiatively initiate voice quality test, between this caller/called user terminal and voice quality testing server, must guarantee that at least one passage can use.When caller/called user terminal is placed on fire compartment wall/NAT(Network Address Translation; network address translation) below time; between caller/called user terminal and voice quality testing server, need to arrange that fire compartment wall/NAT penetrates and keepalive mechanism, for example, between voice quality testing server and call subscriber terminal and called user terminal, maintain default heartbeat mechanism.
Voice quality testing server is responsible for carrying out alternately with user, and manually originating end opposite end double-directional speech quality test of user also can customized task originating end opposite end double-directional speech quality test in batches.
For example: after call subscriber terminal and called user terminal call are set up, call subscriber terminal is play a default recording file to called user terminal, in communication process, call subscriber terminal regularly sends to voice quality testing server to the network condition of called user terminal with RTCP message format by the call subscriber terminal of collection, and wherein the content of RTCP message comprises: time delay, packet loss and flap-statistics.
In an embodiment of the present invention, this first network condition information represents that call subscriber terminal is to the network condition of the voice channel of the bearing call connection of called user terminal, this first network condition information can adopt the form of RTCP message, comprising: time delay, packet loss and flap-statistics etc.
In an embodiment of the present invention, call subscriber terminal receives after the tone testing order of voice quality testing server, from trend called user terminal, make a call, the privately owned parameter of a SIP from field is carried in this calling, voice quality test call while being used for indicating this calling to called user terminal.
In an embodiment of the present invention, this second network condition information represents that called user terminal is to the network condition of the voice channel of the bearing call connection of call subscriber terminal, this second network condition information can adopt the form of RTCP message, comprising: time delay, packet loss and flap-statistics etc.; After call connection is set up, when called user terminal is judged this call connection and is voice quality test call, called user terminal can be looped back to call subscriber terminal by loopback passage by the default media stream file receiving, for example, by the E1 loopback of TG equipment (trunking) or the SLIC interface loopback of IAD equipment (integrated access equipment).
Loopback passage is the voice channel between called user terminal and call subscriber terminal, presentation medium stream file is from call subscriber terminal transfers to called user terminal, from called user terminal, be back to again the passage of ring certainly of call subscriber terminal, in an embodiment of the present invention, called user terminal can be looped back to sendaisle again by the default media stream file receiving from sendaisle, utilize sendaisle by loopback media stream file loopback to call subscriber terminal, by adopting identical passage, can be convenient to the follow-up comparative analysis of carrying out voice quality.
In an embodiment of the present invention, voice quality testing server gathers according to first network condition information and the second network condition information information of carrying out received, for example, gather network delay, packet loss and wobble information, with critic network quality.
In an embodiment of the present invention, call subscriber terminal is preserved into the loopback media stream file receiving the audio file (for example WAV form) of predetermined format.After call stops, call subscriber terminal is uploaded this audio file to voice quality testing server.
In an embodiment of the present invention, can adopt objective speech quality assessment (Perceptual Evaluation of Speech Quality, be called for short PESQ) method, it can pass through existing tester or software to the test of voice, has the voice (loopback media stream file) of decline to contrast and be achieved respectively to raw tone (default media stream file) with after system is processed.
In another embodiment of the present invention, for guaranteeing voice quality MOS value, calculate accurately, before call subscriber terminal media stream file, insert one section of DTMF(Dual-Tone Multifrequency, dual-tone multifrequency) sound, after broadcasting finishes, insert again one section of dtmf tone, and do not comprise above-mentioned dtmf tone in the media stream file that guarantees to play.Dtmf tone when call subscriber terminal is received excision call starts after loopback media stream file dtmf tone and end of conversation, thus make the loopback media stream file of preserving consistent with the default media stream file content assurance of broadcasting.
By call subscriber terminal, the loopback media stream file of receiving is uploaded to voice quality testing server, by voice quality testing server, is calculated the voice quality MOS value of this call, can effectively save the operation resource of call subscriber terminal.
Voice quality testing server is received the voice quality MOS value that starts to calculate this call after loopback media stream file; analyze this secondary half-session opposite end bilateral network situation simultaneously; if discovering network is abnormal or unit exception, result is pushed to user with forms such as alarms.
Voice quality testing server is for computing network time delay, shake and the packet loss situation that affects on voice quality, also need to know call subscriber terminal and the compensation deals ability of called user terminal to above-mentioned network parameter fluctuation, these information are carried in voice quality test registration packet at user terminal.
ITUT has requirement to VoIP business to network QoS, and for example time delay is less than 150ms end to end, and delay jitter is less than 30ms, and packet loss is less than in 1% situation, can guarantee the service quality of speech business, otherwise voice quality will decline.
If the decline numerical value of the decline numerical value of the voice quality that network quality causes and speech quality MOS value is much smaller, can tentatively judge that the subscriber's line of called user terminal partly has problems.If when network quality index does not have the situation of obviously severe variation, voice quality obviously reduces, through repeatedly adding up and can draw above-mentioned conclusion.
Although above-described process flow comprises a plurality of operations that occur with particular order, but should have a clear understanding of, these processes can comprise more or less operation, and these operations can sequentially carry out or executed in parallel (for example using parallel processor or multi-thread environment).
As shown from the above technical solution, embodiments of the invention have following beneficial effect: first, when call subscriber terminal is preset media stream file to called user terminal transmission, voice quality testing server obtains the first network condition information that call subscriber terminal sends, and wherein first network condition information represents that call subscriber terminal is to the network condition of the voice channel of the bearing call connection of called user terminal; At called user terminal by the default media stream file loopback receiving during to call subscriber terminal, voice quality testing server obtains the second network condition information that called user terminal sends, second network condition information represents that called user terminal arrives the network condition of the voice channel that the bearing call of described call subscriber terminal connects, thereby can obtain the situation of end-to-end bilateral network quality, overcome conventional art and only can gather opposite end to the deficiency of the network quality situation of local terminal;
Secondly, the loopback media stream file that call subscriber terminal sends the called user terminal receiving is uploaded to voice quality testing server, by voice quality testing server, according to loopback media stream file and default media stream file, calculate end-to-end voice quality MOS(Mean Opinion Score, mean opinion score), voice quality testing server can be according to the comparing result of end-to-end voice quality MOS and end-to-end bilateral network quality condition, and whether tentative diagnosis called end equipment exists the faults such as user's plate noise.
Again, voice quality testing server can obtain the network condition of institute's supervising the network, when network variation, can notify in time O&M personnel by modes such as alarms.
Referring to Fig. 2, sequential chart for the method for testing of end-to-end double-directional speech quality in embodiments of the invention, the present embodiment makes a call to called user terminal with voice quality testing server indication call subscriber terminal, it is example that call subscriber terminal and called user terminal carry out voice quality test to the voice channel of setting up respectively, voice quality testing server is controlled to the detailed process that call subscriber terminal makes a call and voice quality is tested and be described.As shown in Figure 2, the method for the present embodiment specifically comprises the steps:
Step 1, voice quality testing server send tone testing order to call subscriber terminal;
In the present embodiment, between call subscriber terminal and called user terminal, initiate and establish call connection, voice channel is carried out in the test process of voice quality, all under the control of voice quality testing server, carry out.This voice quality testing server can corresponding communication system in a plurality of user terminals, and simultaneously to a plurality of user terminals tone testing to each other carry out centralized monitor.
Particularly, in voice quality testing server, can be according to the actual conditions of a plurality of user terminals of correspondence and tone testing demand, make a call mutually to each other order, time and the concrete mode of tone testing of each user terminal carried out to configuration in advance.Based on this pre-configured information, the time that voice quality testing server can appointment in configuration information, to the voice terminal of appointment, send default tone testing order, thereby indicate the beginning of this test call.
Further, when the voice channel of needs tests is more, tone testing server can also, by reasonably dividing pairing time, cycle that each road voice are tested in configuration information, carry out the voice quality test to each voice terminal voice channel to each other in an orderly manner.
In the present embodiment, with tone testing server, according to configuration information, to call subscriber terminal, send tone testing order, indication call subscriber terminal makes a call to called user terminal, and the voice channel of setting up is carried out to voice quality test for example, the voice quality method of testing of the embodiment of the present invention is described.Particularly, in the tone testing order that tone testing server sends, can carry the relevant indication information that is used to indicate that call subscriber terminal is called out and carries out voice quality test operation, such as the identification information of called user terminal etc.According to the identification information of this called user terminal, call subscriber terminal can be learnt the opposite end that this time makes a call and carry out tone testing, thereby under indication, can operate accordingly in tone testing order.
What certainly can understand is, in an embodiment of the present invention, according to actual demand, the tone testing order that is handed down to call subscriber terminal can also comprise more information, the operation of being correlated with for indication call subscriber terminal, thereby the calling procedure of call subscriber terminal and voice quality test process are better controlled, and embodiments of the invention specifically do not limit this.
Step 2, call subscriber terminal make a call to called user terminal, and connect;
Namely, after the tone testing order receiving of call subscriber terminal basis, according to the identification information of the called user terminal carrying in this tone testing order, to the called user terminal of appointment, make a call, and after making a call, set up the call connection with called user terminal.
Step 3, call subscriber terminal send default media stream file to called user terminal;
Namely, set up after the call connection between call subscriber terminal and called user terminal, call subscriber terminal and called user terminal will be respectively carry out the assessment test of voice quality to the voice channel of setting up, and respectively the voice quality class information of each self-test be returned to voice quality testing server and carry out test analysis.
Step 4 ~ step 5, called user terminal by the default media stream file loopback receiving to call subscriber terminal;
As shown in Figure 4, can be arranged to TG(trunking) E1 interface loopback or the IAD(summation access device of equipment) the SLIC interface loopback of equipment.
Step 6, call subscriber terminal send call subscriber terminal to the first network condition information of the voice channel of called user terminal to tone testing server;
Step 7, called user terminal send called user terminal to the second network condition information of the voice channel of call subscriber terminal to tone testing server;
After step 8, default media stream file finish playing, call subscriber terminal sends the message that stops call to called user terminal;
Step 9, call subscriber terminal are uploaded to tone testing server by the loopback media stream file receiving;
Step 10, voice quality testing server calculate the voice quality MOS mean opinion score value of this call;
Voice quality testing server, according to the loopback media stream file and the described default media stream file that receive, obtains the voice quality MOS mean opinion score value of this call between described call subscriber terminal and called user terminal.
Referring to Fig. 3, be the structured flowchart of voice quality testing server in embodiments of the invention, this voice quality testing server comprises:
The first acquisition module 31, for when call subscriber terminal is preset media stream file to called user terminal transmission, obtain the first network condition information that described call subscriber terminal sends, described first network condition information represents that described call subscriber terminal is to the network condition of the voice channel of the bearing call connection of described called user terminal;
The second acquisition module 32, for described called user terminal by the described default media stream file loopback receiving when the described call subscriber terminal, obtain the second network condition information that described called user terminal sends, described second network condition information represents that described called user terminal is to the network condition of the voice channel of the bearing call connection of described call subscriber terminal.
In another embodiment of the present invention, voice quality testing server also comprises:
The first processing module 33, for according to described first network condition information and second network condition information, obtains the end-to-end two-way network condition information of this call of described call subscriber terminal and described called user terminal.
In another embodiment of the present invention, voice quality testing server also comprises:
The second processing module 35, for according to the loopback media stream file and the described default media stream file that receive, obtains the voice quality MOS value of this call between call subscriber terminal and called user terminal.
In another embodiment of the present invention, voice quality testing server also comprises:
The 3rd processing module, for according to the voice quality MOS value of described end-to-end two-way network condition information and described this call, determines the fault point of the voice quality reduction that causes this call.
In another embodiment of the present invention, between voice quality testing server and described call subscriber terminal and described called user terminal, maintain default heartbeat mechanism.
In another embodiment of the present invention, described voice quality testing server also comprises:
Tone testing module, for sending tone testing order to described call subscriber terminal, described call subscriber terminal establishes call connection according to described tone testing order and described called user terminal; After call connection is set up, the voice channel that bearing call is connected carries out voice quality test.
Referring to Fig. 4, structural representation for the test macro of end-to-end double-directional speech quality in embodiments of the invention, this test macro comprises: tone testing server 41 as above and at least two user terminals 42,43, wherein tone testing server 41 is connected with user terminal 42,43 respectively.
The above is only the preferred embodiment of the present invention; it should be pointed out that for those skilled in the art, under the premise without departing from the principles of the invention; can also make some improvements and modifications, these improvements and modifications also should be considered as protection scope of the present invention.
Claims (13)
1. a method of testing for end-to-end double-directional speech quality, when carrying out voice quality test, is characterized in that, described method of testing comprises:
When call subscriber terminal is preset media stream file to called user terminal transmission, voice quality testing server obtains the first network condition information that described call subscriber terminal sends, and described first network condition information represents that described call subscriber terminal is to the network condition of the voice channel of described called user terminal;
Described called user terminal by the described default media stream file loopback receiving when the described call subscriber terminal, described voice quality testing server obtains the second network condition information that described called user terminal sends, and described second network condition information represents that described called user terminal is to the network condition of the voice channel of described call subscriber terminal.
2. the method for testing of end-to-end double-directional speech quality according to claim 1, is characterized in that, described method of testing also comprises:
Described voice quality testing server, according to described first network condition information and second network condition information, obtains the end-to-end two-way network condition information of this call between described call subscriber terminal and described called user terminal.
3. the method for testing of end-to-end double-directional speech quality according to claim 2, is characterized in that, at described call subscriber terminal, receives after whole loopback media stream file, and described method of testing also comprises:
Described voice quality testing server receives the loopback media stream file that described call subscriber terminal is uploaded;
Described voice quality testing server, according to the loopback media stream file and the described default media stream file that receive, obtains the voice quality MOS mean opinion score value of this call between described call subscriber terminal and called user terminal.
4. the method for testing of end-to-end double-directional speech quality according to claim 3, is characterized in that, after obtaining the voice quality MOS value of this call, described method of testing also comprises:
Described voice quality testing server, according to the voice quality MOS value of described end-to-end two-way network condition information and described this call, is determined the fault point of the voice quality reduction that causes this call.
5. the method for testing of end-to-end double-directional speech quality according to claim 1, is characterized in that, maintains default heartbeat mechanism between described voice quality testing server and described call subscriber terminal and described called user terminal.
6. the method for testing of end-to-end double-directional speech quality according to claim 1, is characterized in that, when voice quality test starts, described method of testing also comprises:
Described voice quality testing server sends tone testing order to described call subscriber terminal, and described call subscriber terminal establishes call connection according to described tone testing order and described called user terminal;
After call connection is set up, the voice channel that described voice quality testing server connects bearing call carries out voice quality test.
7. a voice quality testing server, is characterized in that, comprising:
The first acquisition module, for when call subscriber terminal is preset media stream file to called user terminal transmission, obtain the first network condition information that described call subscriber terminal sends, described first network condition information represents that described call subscriber terminal is to the network condition of the voice channel of the bearing call connection of described called user terminal;
The second acquisition module, for described called user terminal by the described default media stream file loopback receiving when the described call subscriber terminal, obtain the second network condition information that described called user terminal sends, described second network condition information represents that described called user terminal is to the network condition of the voice channel of the bearing call connection of described call subscriber terminal.
8. voice quality testing server according to claim 7, is characterized in that, described voice quality testing server also comprises:
The first processing module, for according to described first network condition information and second network condition information, obtains the end-to-end two-way network condition information of this call of described call subscriber terminal and described called user terminal.
9. voice quality testing server according to claim 8, is characterized in that, described voice quality testing server also comprises:
Receiver module, the loopback media stream file of uploading for receiving described call subscriber terminal;
The second processing module, for according to the described loopback media stream file and the described default media stream file that receive, obtains the voice quality MOS mean opinion score value of this call between call subscriber terminal and called user terminal.
10. voice quality testing server according to claim 9, is characterized in that, described voice quality testing server also comprises:
The 3rd processing module, for according to the voice quality MOS value of described end-to-end two-way network condition information and described this call, determines the fault point of the voice quality reduction that causes this call.
11. voice quality testing servers according to claim 7, is characterized in that, maintain default heartbeat mechanism between described voice quality testing server and described call subscriber terminal and described called user terminal.
12. voice quality testing servers according to claim 7, is characterized in that, described voice quality testing server also comprises:
Tone testing module, for sending tone testing order to described call subscriber terminal, described call subscriber terminal establishes call connection according to described tone testing order and described called user terminal; After call connection is set up, the voice channel that bearing call is connected carries out voice quality test.
The test macro of 13. 1 kinds of end-to-end double-directional speech quality, is characterized in that, comprising:
Tone testing server as described in as arbitrary in claim 7 ~ 12 and at least two user terminals, wherein said tone testing server is connected with described user terminal respectively.
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| CN106550375A (en) * | 2015-09-22 | 2017-03-29 | 深圳市赛格导航科技股份有限公司 | A kind of automatic navigator call automated testing method |
| CN106550375B (en) * | 2015-09-22 | 2021-01-29 | 深圳市赛格导航科技股份有限公司 | Automatic test method for vehicle-mounted navigator call |
| CN110351155A (en) * | 2018-04-03 | 2019-10-18 | 苏州景昱医疗器械有限公司 | The program-controlled network performance test methods and system of implantable medical device |
| CN110351155B (en) * | 2018-04-03 | 2020-12-22 | 苏州景昱医疗器械有限公司 | Program-controlled network performance testing method and system of implantable medical equipment |
| CN110365548A (en) * | 2018-04-10 | 2019-10-22 | 华为技术有限公司 | A kind of network for formance measuring method and its device |
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