CN113992547B - Test method and system for automatically detecting packet loss rate in real-time voice - Google Patents
Test method and system for automatically detecting packet loss rate in real-time voice Download PDFInfo
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- CN113992547B CN113992547B CN202010657781.7A CN202010657781A CN113992547B CN 113992547 B CN113992547 B CN 113992547B CN 202010657781 A CN202010657781 A CN 202010657781A CN 113992547 B CN113992547 B CN 113992547B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L43/00—Arrangements for monitoring or testing data switching networks
- H04L43/08—Monitoring or testing based on specific metrics, e.g. QoS, energy consumption or environmental parameters
- H04L43/0823—Errors, e.g. transmission errors
- H04L43/0829—Packet loss
- H04L43/0835—One way packet loss
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L43/00—Arrangements for monitoring or testing data switching networks
- H04L43/08—Monitoring or testing based on specific metrics, e.g. QoS, energy consumption or environmental parameters
- H04L43/0876—Network utilisation, e.g. volume of load or congestion level
- H04L43/0894—Packet rate
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L43/00—Arrangements for monitoring or testing data switching networks
- H04L43/06—Generation of reports
Abstract
The invention provides a test method for automatically detecting packet loss rate in real-time voice, which comprises the following steps: s1, carrying out network speed calibration on a plurality of terminal devices in real-time communication; s2, starting an audio conference by one terminal device A in the plurality of terminal devices, and adding other terminal devices of the plurality of terminal devices into the same voice channel of the terminal device A; respectively recording voices with specified duration t for a plurality of terminal devices, and then playing voice contents by the terminal device A; s3, collecting and processing the audio files recorded by each terminal device to obtain audio data streams of each terminal device and obtain a single audio packet loss rate value P; and S4, automatically and circularly traversing and detecting, obtaining the average packet loss rate Px through traversal calculation, and outputting a test report. The invention is realized automatically, releases manpower and improves the testing efficiency.
Description
Technical Field
The invention relates to the technical field of computer communication, in particular to a test method and a test system for automatically detecting packet loss rate in real-time voice.
Background
Real-time voice is transmitted and reproduced through a certain technology, and a modern means of high-tech communication and cooperation is provided for people who cannot gather to the same place for conversation communication. Specifically, the device A (a mobile phone and a computer) acquires the sound of a microphone by calling an audio driver, and transmits the voice to another device B, C and D. For example, in the era of "mobile office", people carry out voice conferences through devices such as mobile phones, thereby liberating cumbersome text communication and realizing real-time communication. For example, parents want to chat with children, do not type nor speak Mandarin, and can communicate dialect through real-time voice, which is convenient and intimate.
The prior art has the following defects: and a plurality of people can occasionally have the phenomena of intermittent jamming and the like. Influenced by factors of uncontrollable environments (networks, servers, equipment and the like) at the time, the method is difficult to reproduce and cannot give accurate call quality data. At present, the quality of voice call is tested, and the testing scheme is that network setting is carried out manually, manual voice recording is carried out, and then a conclusion whether the voice call is stuck or not is given. The packet loss rate cannot be perceived, and the repeated simulation operation which requires manpower for the automatic periodic detection cannot be performed wastes time and is inefficient.
Disclosure of Invention
In order to overcome the above problems, the present invention aims to provide a method for automatically detecting packet loss rate in real-time voice, which can liberate manual testing, obtain an accurate packet loss rate value, and improve work efficiency.
The invention is realized by adopting the following scheme: a test method for automatically detecting packet loss rate in real-time voice comprises the following steps:
s1, carrying out network speed calibration on a plurality of terminal devices in real-time communication;
s2, starting an audio conference by one terminal device A in the plurality of terminal devices, and adding other terminal devices of the plurality of terminal devices into the same voice channel of the terminal device A; respectively recording voices with specified duration t for a plurality of terminal devices, and then playing voice contents by the terminal device A;
s3, collecting and processing the audio files recorded by each terminal device to obtain audio data streams of each terminal device and obtain a single audio packet loss rate value P;
and S4, automatically and circularly traversing and detecting, obtaining the average packet loss rate Px through traversal calculation, and outputting a test report.
Further, the step S1 is further specifically: s11, the remote server controls the trigger task and distributes the task to a timer in set time t1, t2 and t3 \8230; \ 8230; (tn);
s12, at time t1, a timer receives a task, acquires the network conditions of a plurality of terminal devices, and uniformly limits the networks of the plurality of terminal devices to the same rate;
step S13, carrying out network speed calibration on the linked terminal equipment, and ensuring that each linked terminal equipment has no network speed error and the network speed is consistent: entering the next step of the terminal equipment A to initiate an audio conference, wherein the network speeds are inconsistent: step S13 is re-executed.
Further, the step S3 is further specifically: step S31, collecting the audio files recorded by each terminal device, analyzing the content of the audio files to obtain corresponding audio data streams, and calculating and recording the single packet loss rate P1 according to the number of ICMP in unit time/the number of all messages in unit time;
and S32, waiting for the timer to execute the set residual time t2, t3 \8230 \ 8230: \ 8230tn, and automatically and circularly traversing to obtain the corresponding single packet loss rates P2, P3 \8230 \ 8230:/\ Pn.
Further, the step S4 is further specifically: and according to the completion of all the cycle execution of the set time, obtaining the average packet loss rate Px by using an average value calculation formula, and outputting a test report, wherein the content of the test report comprises but is not limited to: the method comprises the following steps of execution time, network speed of each terminal device, conference channel number, played audio content file, audio data stream, single packet loss rate and average packet loss rate.
Further, the obtaining of the audio data streams of the respective terminal devices and the obtaining of the single audio packet loss ratio value P specifically include: acquiring files in a wav format stored in terminal equipment, namely an a.wav file of the terminal equipment A and a b.wav file of the terminal equipment B;
analyzing the files in the wav format, and realizing by using a wavread function of matlab in a program code C + +, and acquiring corresponding file headers and data blocks data [ a ] and data [ b ];
storing data blocks data [ a ] and data [ B ] corresponding to the terminal equipment A and the terminal equipment B;
and calculating the packet loss rate P according to a formula of packet loss rate [ (input message-output message)/input message ]. 100%, wherein the packet loss rate P is calculated, namely P = [ (data [ a ] -data [ b ])/data [ a ] ]. 100%.
The invention also provides a test system for automatically detecting the packet loss rate in real-time voice, which comprises a network speed calibration module, a voice recording module, a single packet loss rate acquisition module and an average packet loss rate acquisition module;
the network speed calibration module is used for calibrating the network speed of a plurality of pieces of terminal equipment which are in real-time communication;
the voice recording module is used for starting an audio conference by one terminal device A in the plurality of terminal devices and adding other terminal devices of the plurality of terminal devices into the same voice channel of the terminal device A; respectively recording voices with specified duration t for a plurality of terminal devices, and then playing voice contents by the terminal device A;
the single packet loss rate obtaining module is used for collecting and processing the audio files recorded by each terminal device to obtain the audio data streams of the respective terminal devices and obtain a single audio packet loss rate value P;
the average packet loss rate obtaining module is used for automatically and circularly traversing and detecting, obtaining the average packet loss rate Px through traversal calculation, and outputting a test report.
Further, the network speed calibration module further specifically includes: the remote server controls the trigger tasks and distributes the tasks to the timer at set time t1, t2, t3 \8230, 8230and tn;
at time t1, the timer receives the task, acquires the network conditions of the plurality of terminal devices, and uniformly limits the networks of the plurality of terminal devices to the same rate;
network speed calibration is carried out on the linked terminal equipment, and it is ensured that each linked terminal equipment has no network speed error and the network speed is consistent: entering the next step that the terminal equipment A initiates an audio conference, wherein the network speeds are inconsistent: the wire speed calibration is performed again.
Further, the single packet loss rate obtaining module further specifically includes: collecting audio files recorded by each terminal device, analyzing the content of the audio files to obtain corresponding audio data streams, and calculating and recording the single packet loss rate P1 according to the number of ICMP/the number of all messages in unit time;
waiting for the rest time t2 and t3 of timer execution setting 8230, 8230tn, and automatically and circularly traversing to obtain the corresponding single packet loss rates P2 and P3 8230, 8230pn and Pn.
Further, the average packet loss rate obtaining module is further specifically: and according to the completion of all the cycle execution of the set time, obtaining the average packet loss rate Px by using an average value calculation formula, and outputting a test report, wherein the content of the test report comprises but is not limited to: execution time, network speed of each terminal device, conference channel number, played audio content file, audio data stream, single packet loss rate and average packet loss rate.
Further, the obtaining of the audio data streams of the respective terminal devices and the obtaining of the single audio packet loss ratio value P specifically include: acquiring wav format files stored in terminal equipment, namely an a.wav file of the terminal equipment A and a b.wav file of the terminal equipment B;
analyzing the files in the wav format, and realizing by using wavread functions of matlab in program codes C + +, and acquiring corresponding file headers and data blocks data [ a ] and data [ b ];
storing data blocks data [ a ] and data [ B ] corresponding to the terminal equipment A and the terminal equipment B;
according to the packet loss rate calculation formula, [ (input message-output message)/input message ] + 100%, calculating to obtain the packet loss rate P, namely P = [ (data [ a ] -data [ b ])/data [ a ] ] + 100%.
The invention has the beneficial effects that: 1. according to the testing method, time-consuming manual operation is repeated, and the testing method is realized automatically, so that manpower is released, and the testing efficiency is improved.
2. An innovation is effectively made, the previous test limitation is broken through, the accurate numerical judgment can be made on the packet loss rate of the audio, and the test quality is improved.
3. The test result information is complete, and the method is convenient for optimizing and analyzing the voice call quality.
Drawings
FIG. 1 is a schematic flow diagram of the process of the present invention.
Fig. 2 is a schematic flowchart of a first embodiment of the present invention.
Fig. 3 is a system block diagram of the present invention.
Detailed Description
The invention is further described below with reference to the accompanying drawings.
Referring to fig. 1, a testing method for automatically detecting a packet loss rate in real-time voice according to the present invention includes the following steps:
s1, carrying out network speed calibration on a plurality of terminal devices in real-time communication;
the step S1 is further specifically: s11, the remote server controls the trigger task and distributes the task to a timer at set time t1, t2 and t3 \8230; \8230andtn;
s12, at time t1, a timer receives a task, acquires the network conditions of a plurality of terminal devices, and uniformly limits the networks of the plurality of terminal devices to the same rate;
step S13, network speed calibration is carried out on the linked terminal equipment, and it is ensured that each linked terminal equipment has no network speed error and the network speed is consistent: entering the next step of the terminal equipment A to initiate an audio conference, wherein the network speeds are inconsistent: step S13 is re-executed.
S2, starting an audio conference by one terminal device A in the plurality of terminal devices, and adding other terminal devices of the plurality of terminal devices into the same voice channel of the terminal device A; respectively recording voices with specified duration t for a plurality of terminal devices, and then playing voice contents by the terminal device A;
s3, collecting and processing the audio files recorded by each terminal device to obtain audio data streams of the respective terminal devices and obtain a single audio packet loss rate value P; the step S3 is further specifically: step S31, collecting the audio files recorded by each terminal device, analyzing the content of the audio files to obtain corresponding audio data streams, and calculating and recording the single packet loss rate P1 according to the number of ICMP in unit time/the number of all messages in unit time;
and S32, waiting for the rest time t2, t3, 8230, 823030, tn set by the timer, and automatically and circularly traversing to obtain the corresponding single packet loss rates P2, P3, 8230, pn.
And S4, automatically and circularly traversing and detecting, obtaining the average packet loss rate Px through traversal calculation, and outputting a test report.
The step S4 is further specifically: and according to the completion of all the cycle execution of the set time, obtaining the average packet loss rate Px by using an average value calculation formula, and outputting a test report, wherein the content of the test report comprises but is not limited to: the method comprises the following steps of execution time, network speed of each terminal device, conference channel number, played audio content file, audio data stream, single packet loss rate and average packet loss rate.
Further, the obtaining of the audio data streams of the respective terminal devices and the obtaining of the single audio packet loss ratio value P specifically include: acquiring wav format files stored in terminal equipment, namely an a.wav file of the terminal equipment A and a b.wav file of the terminal equipment B;
analyzing the files in the wav format, and realizing by using wavread functions of matlab in program codes C + +, and acquiring corresponding file headers and data blocks data [ a ] and data [ b ];
storing data blocks data [ a ] and data [ B ] corresponding to the terminal equipment A and the terminal equipment B;
according to the packet loss rate calculation formula, [ (input message-output message)/input message ] + 100%, calculating to obtain the packet loss rate P, namely P = [ (data [ a ] -data [ b ])/data [ a ] ] + 100%.
The invention is further illustrated below with reference to a specific embodiment:
referring to fig. 2, the testing method for automatically detecting the packet loss rate in real-time voice of the present invention,
1. the remote server is used for controlling the triggering task and distributing the task to the timer within the set time t 1;
2. the timer receives the task, acquires the network conditions of the terminal equipment A and the terminal equipment B, and uniformly limits the networks of the terminal equipment A and the terminal equipment B in the same rate;
3. the network speed calibration is carried out on the linked equipment, and the condition that each linked equipment has no network speed error is ensured to be consistent: entering the next step of the terminal device A to initiate an audio conference, wherein the inconsistency is as follows: re-executing the step 3;
4. the terminal device A initiates an audio conference, and the terminal device B joins the audio conference; ensuring that the terminal equipment A and the terminal equipment B are in the same voice channel and consistent: and entering the next step of sending a request for recording the appointed time length t1 to the terminal equipment A and the terminal equipment B, wherein the requests are inconsistent: re-executing the step 4;
5. the system sends a request for recording the specified duration t1 to the terminal equipment A and the terminal equipment B, and the terminal equipment A starts playing the audio content t2 to ensure that the terminal equipment A and the terminal equipment B record normally;
6. acquiring audio files a.wav and b.wav recorded by terminal equipment A and B, checking the duration of the audio files a and B, and ensuring that the recording duration is t1 and is consistent: and entering the next step of analyzing the audio file, wherein the audio file is inconsistent: re-executing the step 5;
7. analyzing the content of the audio file to obtain a corresponding audio data stream, and calculating and recording the single packet loss rate P1 according to the number of ICMP in unit time/the number of all messages in unit time;
8. waiting for the rest time t2, t3 \ 8230; \8230; \ 8230;. Automatically circulating and traversing the system to obtain the corresponding single packet loss rate P2, P3 \ 8230; \8230;
9. and (4) according to the set time, completing all the cycle execution, and obtaining the average packet loss rate Px by using an average value calculation formula. And outputting a test report, wherein the report content comprises system execution time, network speed of each device, conference channel number, played audio content file, audio file, data stream, single packet loss rate and average packet loss rate.
Referring to fig. 3, the present invention further provides a testing system for automatically detecting a packet loss rate in real-time voice, where the testing system includes a network speed calibration module, a voice recording module, a single packet loss rate obtaining module, and an average packet loss rate obtaining module;
the network speed calibration module is used for calibrating the network speed of a plurality of pieces of terminal equipment which are in real-time communication;
the voice recording module is used for starting an audio conference by one terminal device A in the plurality of terminal devices and adding other terminal devices of the plurality of terminal devices into the same voice channel of the terminal device A; respectively recording voices with specified duration t for a plurality of terminal devices, and then playing voice contents by the terminal device A;
the single packet loss rate obtaining module is used for collecting and processing the audio files recorded by each terminal device to obtain the audio data streams of the respective terminal devices and obtain a single audio packet loss rate value P;
the obtaining of the audio data streams of the respective terminal devices and the obtaining of the single audio packet loss ratio value P specifically include: acquiring wav format files stored in terminal equipment, namely an a.wav file of the terminal equipment A and a b.wav file of the terminal equipment B;
analyzing the files in the wav format, and realizing by using wavread functions of matlab in program codes C + +, and acquiring corresponding file headers and data blocks data [ a ] and data [ b ];
storing data blocks data [ a ] and data [ B ] corresponding to the terminal equipment A and the terminal equipment B;
and calculating the packet loss rate P according to a formula of packet loss rate [ (input message-output message)/input message ]. 100%, wherein the packet loss rate P is calculated, namely P = [ (data [ a ] -data [ b ])/data [ a ] ]. 100%.
The average packet loss rate obtaining module is used for automatically and circularly traversing and detecting, obtaining the average packet loss rate Px through traversal calculation, and outputting a test report.
Wherein, the network speed calibration module further specifically comprises: the remote server controls the trigger tasks and distributes the tasks to the timer at set time t1, t2, t3 \8230, 8230and tn;
at time t1, the timer receives the task, acquires the network conditions of the plurality of terminal devices, and uniformly limits the networks of the plurality of terminal devices to the same rate;
network speed calibration is carried out on the linked terminal equipment, and it is ensured that each linked terminal equipment has no network speed error and the network speed is consistent: entering the next step of the terminal equipment A to initiate an audio conference, wherein the network speeds are inconsistent: the wire speed calibration is performed again.
The single packet loss rate obtaining module is further specifically: collecting audio files recorded by each terminal device, analyzing the content of the audio files to obtain corresponding audio data streams, and calculating and recording the single packet loss rate P1 according to the number of ICMP in unit time/the number of all messages in unit time;
waiting for the rest time t2 and t3 of timer execution setting 8230, 8230tn, and automatically and circularly traversing to obtain the corresponding single packet loss rates P2 and P3 8230, 8230pn and Pn.
In addition, the average packet loss rate obtaining module further specifically includes: and according to the completion of all the cycle execution of the set time, obtaining the average packet loss rate Px by using an average value calculation formula, and outputting a test report, wherein the content of the test report comprises but is not limited to: execution time, network speed of each terminal device, conference channel number, played audio content file, audio data stream, single packet loss rate and average packet loss rate.
The above description is only a preferred embodiment of the present invention, and all equivalent changes and modifications made in accordance with the claims of the present invention should be covered by the present invention.
Claims (8)
1. A test method for automatically detecting packet loss rate in real-time voice is characterized in that: the test method comprises the following steps:
s1, carrying out network speed calibration on a plurality of terminal devices in real-time communication; the step S1 is further specifically: s11, the remote server controls the trigger task and distributes the task to a timer in set time t1, t2 and t3 \8230; \ 8230; (tn);
s12, at time t1, a timer receives a task, network conditions of a plurality of terminal devices are obtained, and networks of the plurality of terminal devices are uniformly limited to the same speed;
step S13, network speed calibration is carried out on the linked terminal equipment, and it is ensured that each linked terminal equipment has no network speed error and the network speed is consistent: entering the next step that the terminal equipment A initiates an audio conference, wherein the network speeds are inconsistent: re-executing step S13;
s2, starting an audio conference by one terminal device A in the plurality of terminal devices, and adding other terminal devices of the plurality of terminal devices into the same voice channel of the terminal device A; respectively recording voices with specified duration t for a plurality of terminal devices, and then playing voice contents by the terminal device A;
s3, collecting and processing the audio files recorded by each terminal device to obtain audio data streams of each terminal device and obtain a single audio packet loss rate value P;
and S4, automatically and circularly traversing and detecting to obtain the average packet loss rate Px through traversal calculation and output a test report.
2. The method according to claim 1, wherein the method comprises the following steps: the step S3 is further specifically: step S31, collecting audio files recorded by each terminal device, analyzing the content of the audio files to obtain corresponding audio data streams, and calculating and recording the single packet loss rate P1 according to the number of ICMP in unit time/the number of all messages in unit time;
and S32, waiting for the rest time t2, t3, 8230, 823030, tn set by the timer, and automatically and circularly traversing to obtain the corresponding single packet loss rates P2, P3, 8230, pn.
3. The method according to claim 1, wherein the method comprises the following steps: the step S4 is further specifically: and according to the completion of all the cycle execution of the set time, obtaining the average packet loss rate Px by using an average value calculation formula, and outputting a test report, wherein the content of the test report comprises but is not limited to: execution time, network speed of each terminal device, conference channel number, played audio content file, audio data stream, single packet loss rate and average packet loss rate.
4. The method according to claim 1, wherein the method comprises the following steps: the obtaining of the audio data streams of the respective terminal devices and the obtaining of the single audio packet loss ratio value P specifically includes: acquiring wav format files stored in terminal equipment, namely an a.wav file of the terminal equipment A and a b.wav file of the terminal equipment B;
analyzing the files in the wav format, and realizing by using a wavread function of matlab in a program code C + +, and acquiring corresponding file headers and data blocks data [ a ] and data [ b ];
storing data blocks data [ a ] and data [ B ] corresponding to the terminal equipment A and the terminal equipment B;
according to the packet loss rate calculation formula, [ (input message-output message)/input message ] + 100%, calculating to obtain the packet loss rate P, namely P = [ (data [ a ] -data [ b ])/data [ a ] ] + 100%.
5. A test system for automatically detecting packet loss rate in real-time voice is characterized in that: the test system comprises a network speed calibration module, a voice recording module, a single packet loss rate acquisition module and an average packet loss rate acquisition module;
the network speed calibration module is used for calibrating the network speed of a plurality of pieces of terminal equipment which are in real-time communication;
the network speed calibration module is further specifically: the remote server controls the triggering task and distributes the task to the timer in the set time t1, t2, t3 \8230; \ 8230t n;
at time t1, the timer receives a task, acquires the network conditions of the plurality of terminal devices, and uniformly limits the networks of the plurality of terminal devices to the same rate;
network speed calibration is carried out on the linked terminal equipment, and it is ensured that each linked terminal equipment has no network speed error and the network speed is consistent:
entering the next step of the terminal equipment A to initiate an audio conference, wherein the network speeds are inconsistent: then the network speed calibration is carried out again;
the voice recording module is used for starting an audio conference by one terminal device A in the plurality of terminal devices and adding other terminal devices of the plurality of terminal devices into the same voice channel of the terminal device A; respectively recording voices with specified duration t for a plurality of terminal devices, and then playing voice contents by the terminal device A;
the single packet loss rate obtaining module is used for collecting and processing the audio files recorded by each terminal device to obtain the audio data streams of the respective terminal devices and obtain a single audio packet loss rate value P;
the average packet loss rate obtaining module is used for automatically and circularly traversing and detecting, obtaining the average packet loss rate Px through traversal calculation, and outputting a test report.
6. The system according to claim 5, wherein the system is configured to automatically detect packet loss rate in real-time voice, and comprises: the single packet loss rate obtaining module is further specifically: collecting audio files recorded by each terminal device, analyzing the content of the audio files to obtain corresponding audio data streams, and calculating and recording the single packet loss rate P1 according to the number of ICMP/the number of all messages in unit time;
waiting for the rest time t2, t3 \ 8230, 8230and tn of the timer execution setting, and automatically and circularly traversing to obtain the corresponding single packet loss rates P2, P3 \ 8230, 8230and Pn.
7. The system according to claim 6, wherein the system is configured to automatically detect packet loss rate in real-time voice, and is characterized in that: the average packet loss rate obtaining module is further specifically: and according to the completion of all the cycle execution of the set time, obtaining the average packet loss rate Px by using an average value calculation formula, and outputting a test report, wherein the content of the test report comprises but is not limited to: execution time, network speed of each terminal device, conference channel number, played audio content file, audio data stream, single packet loss rate and average packet loss rate.
8. The system according to claim 5, wherein the system is configured to automatically detect packet loss rate in real-time voice, and comprises: the obtaining of the audio data streams of the respective terminal devices and the obtaining of the single audio packet loss ratio value P specifically include: acquiring files in a wav format stored in terminal equipment, namely an a.wav file of the terminal equipment A and a b.wav file of the terminal equipment B;
analyzing the files in the wav format, and realizing by using wavread functions of matlab in program codes C + +, and acquiring corresponding file headers and data blocks data [ a ] and data [ b ];
storing data blocks data [ a ] and data [ B ] corresponding to the terminal equipment A and the terminal equipment B;
according to the packet loss rate calculation formula, [ (input message-output message)/input message ] + 100%, calculating to obtain the packet loss rate P, namely P = [ (data [ a ] -data [ b ])/data [ a ] ] + 100%.
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| CN106228986A (en) * | 2016-07-26 | 2016-12-14 | 北京奇虎科技有限公司 | The automated testing method of a kind of speech recognition engine, device and system |
| CN109068186A (en) * | 2018-09-21 | 2018-12-21 | 视联动力信息技术股份有限公司 | The treating method and apparatus of packet loss |
| CN110890945A (en) * | 2019-11-20 | 2020-03-17 | 腾讯科技（深圳）有限公司 | Data transmission method, device, terminal and storage medium |
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