CN108650550B - Network transmission quality analysis method and device, computer equipment and storage medium - Google Patents

Network transmission quality analysis method and device, computer equipment and storage medium Download PDF

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Publication number
CN108650550B
CN108650550B CN201810731620.0A CN201810731620A CN108650550B CN 108650550 B CN108650550 B CN 108650550B CN 201810731620 A CN201810731620 A CN 201810731620A CN 108650550 B CN108650550 B CN 108650550B
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streaming media
rtp
video
audio
packet
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CN108650550A (en
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洪凌毅
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Ping An Technology Shenzhen Co Ltd
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Ping An Technology Shenzhen Co Ltd
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/442—Monitoring of processes or resources, e.g. detecting the failure of a recording device, monitoring the downstream bandwidth, the number of times a movie has been viewed, the storage space available from the internal hard disk
    • H04N21/44227—Monitoring of local network, e.g. connection or bandwidth variations; Detecting new devices in the local network
    • 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
    • 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
    • H04L43/50—Testing arrangements
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/75—Media network packet handling
    • H04L65/764—Media network packet handling at the destination 

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Environmental & Geological Engineering (AREA)
  • Databases & Information Systems (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)

Abstract

The present application relates to the field of computer technologies, and in particular, to a method and an apparatus for analyzing network transmission quality, a computer device, and a storage medium. The method comprises the following steps: acquiring an original network data file containing RTP audio and video stream in real-time video application; extracting an RTP data packet in an original network data file; carrying out shunting processing on the RTP data packet according to a preset shunting rule to obtain audio streaming media and video streaming media; and respectively playing the audio streaming media and the video streaming media, and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media. The method has the advantages that the audio and video stream in the real network environment is completely simulated and reproduced, the analysis of the network quality of the user is facilitated, and the problem that the network quality of the user is difficult to analyze in the existing real-time video application is solved.

Description

Network transmission quality analysis method and device, computer equipment and storage medium
Technical Field
The present application relates to the field of computer technologies, and in particular, to a method and an apparatus for analyzing network transmission quality, a computer device, and a storage medium.
Background
Real-time video applications (such as video conferences) generally adopt an RTP common protocol as a media stream transmission protocol, and under the internet environment, the influence of the change of the network environment of an actual user on media transmission is often difficult to be captured.
Content of application
Aiming at the defects of the prior art, the application provides a network transmission quality analysis method, a device, a computer device and a storage medium, which are used for obtaining audio stream media and video stream media through streaming in an RTP (real-time transport protocol) data packet and obtaining network transmission quality according to the playing conditions of the audio stream media and the video stream media.
The technical scheme provided by the application is as follows:
a method of network transmission quality analysis, the method comprising:
acquiring an original network data file containing RTP audio and video stream in real-time video application;
extracting an RTP data packet in the original network data file;
carrying out shunting processing on the RTP data packet according to a preset shunting rule to obtain audio streaming media and video streaming media;
and respectively playing the audio streaming media and the video streaming media, and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media.
Further, in the step of separately playing the audio streaming media and the video streaming media and analyzing and obtaining the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media, the method includes:
analyzing whether the audio streaming media and the video streaming media are stuck, fuzzy or frame loss in the playing process;
and analyzing the network transmission quality by taking the analysis result of the playing condition as a reference index influencing the bandwidth, packet loss, delay and jitter of the network.
Further, in the step of playing the video streaming media, the method includes:
if the video streaming media is H.264 coding, analyzing the video streaming media by adopting an NAL packet format, and decoding and playing the video streaming media.
Further, the step of performing a streaming process on the RTP data packet according to a preset streaming rule to obtain an audio streaming media and a video streaming media includes:
and splitting the RTP data packets out of the audio media stream and the video media stream according to a synchronous source SSRC identifier.
Further, in the step of extracting the RTP packet in the original network data file, the method includes:
analyzing the file format of the original network data file to obtain a file header, a datagram description and a data packet, wherein the file header comprises a MAGIC code, a version number, a time zone, a data length and a link type, the datagram description comprises a timestamp and a datagram length, and the data packet comprises an ETHER packet header, an IP packet header, a UDP packet header, an RTP packet header and RTP data;
and extracting the RTP data packet from the data packet.
Further, after the steps of playing the audio streaming media and the video streaming media respectively and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media, the method includes:
extracting the timestamp from the datagram description;
transmitting and playing back the audio streaming media and the video streaming media according to the time stamps;
acquiring abnormal data generated by transmission;
processing the exception data by analysis software;
and if the analysis software processes the problems existing in the abnormal data, optimizing the network transmission quality according to the problems.
Further, in the step of acquiring an original network data file containing an RTP audio/video stream in the real-time video application, the method includes:
and acquiring an original network data file containing RTP audio and video stream in real-time video application through tcpdump or Wireshark network packet capturing software, wherein the file format of the original network data file is tcpdump format.
The present application also provides a network transmission quality analysis device, the device includes:
the acquisition module is used for acquiring an original network data file containing RTP audio and video stream in real-time video application;
the extraction module is used for extracting the RTP data packet in the original network data file;
the processing module is used for carrying out shunting processing on the RTP data packet according to a preset shunting rule to obtain audio streaming media and video streaming media;
and the analysis module is used for respectively playing the audio streaming media and the video streaming media and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media.
The present application further provides a computer device comprising a memory and a processor, the memory storing a computer program, wherein the processor implements the steps of any of the above methods when executing the computer program.
The present application also provides a computer-readable storage medium, on which a computer program is stored, which, when being executed by a processor, carries out the steps of the method of any of the above.
According to the technical scheme, the method has the advantages that: the method comprises the steps of obtaining an original network data file in real-time video application, extracting an RTP data packet in the original network data file, dividing streams in the RTP data packet to obtain audio stream media and video stream media, analyzing according to the playing conditions of the audio stream media and the video stream media to obtain network transmission quality, completely simulating and reproducing audio and video streams in a real network environment, helping to analyze user network quality, and aiming at solving the problem that the existing real-time video application is difficult to analyze the user network quality.
Drawings
Fig. 1 is a flowchart of a network transmission quality analysis method provided by an embodiment of the present application;
fig. 2 is a functional block diagram of a network transmission quality analysis apparatus according to an embodiment of the present application;
FIG. 3 is a block diagram schematically illustrating a computer device according to an embodiment of the present disclosure;
fig. 4 is a schematic diagram of a protocol at a transport layer in RTP provided by an embodiment of the present application;
fig. 5 is a schematic diagram of an interface between RTP and UDP provided by an embodiment of the present application.
Detailed Description
In order to make the objects, technical solutions and advantages of the present application more apparent, the present application is described in further detail below with reference to the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are merely illustrative of the present application and are not intended to limit the present application.
As shown in fig. 1, an embodiment of the present application provides a method for analyzing network transmission quality, where the method includes the following steps:
step S101, acquiring an original network data file containing RTP audio and video stream in real-time video application.
In this embodiment, an original network data file in a real-time video application is obtained, where the real-time video application uses an RTP common Protocol as a media stream transmission Protocol, the original network data file includes an audio/video stream encapsulated by an RTP, and the RTP, which is a Reliable Transport Protocol, is an english abbreviation of a reusable Transport Protocol and is a Protocol for managing transmission and reception of EIGRP packets. By "reliable delivery" is meant that delivery is guaranteed, which is implemented by relying on the Cisco proprietary algorithm called "reliable multicast", where the delivery router sends update information to the multicast IP address 224.0.0.10, and each neighbor receiving a reliable multicast packet sends a unicast acknowledgement packet, and if EIGRP does not acknowledge from a neighbor, it will use unicast to retransmit the same data, and if it still does not acknowledge after 16 unicast attempts, the neighbor will be declared to disappear.
Although the name includes "reliable," RTP does not only provide reliable transport, it also provides unreliable transport. When reliable transport is used, RTP requires the other party to send back an ACK acknowledgement; RTP does not require an ACK when unreliable transport is used.
In this embodiment, step S101 includes:
and acquiring an original network data file transmitted by real-time video application software through network packet capturing software, wherein the original network data file comprises RTP audio and video streams.
Network packet capturing software is adopted for capturing packets, particularly, data packets transmitted in a network can be completely intercepted through tcpdump to provide analysis, the tcpdump supports filtering aiming at a network layer, a protocol, a host, a network or a port, and logic statements such as and, or, not and the like are provided to help a user to remove useless information; of course, the packet capturing can also be performed by using a Wireshark (hereinafter referred to as ethernet), which is a network packet analysis software, and the function of the network packet analysis software is to capture the network packet and display the most detailed network packet data as far as possible. Wireshark uses WinPCAP as an interface, and directly exchanges data with a network card to obtain an original network data file in real-time video application software.
Specifically, in step S101, the method includes:
and acquiring an original network data file containing RTP audio and video stream in the real-time video application through tcpdump or Wireshark network packet capturing software, wherein the file format of the original network data file is tcpdump format.
And acquiring the original network data file by using tcpdump or Wireshark network package capturing software, wherein the file format of the acquired original network data file is in a tcpdump format, and the original network data file is in a tcpdump format, so that the original network data file can be conveniently analyzed subsequently.
In this embodiment, before step S101, the method includes:
and screening out network data generated by the real-time video application.
The network data generated by the real-time video application is screened out by analyzing all the network data, then the network data generated by the real-time video application is analyzed to obtain an original network data file containing RTP audio and video streams, then an RTP data packet is obtained from the original network data file, the RTP data packet is divided into audio media streams and video media streams according to an SSRC identifier, then the audio media streams are analyzed and processed, the audio media streams are usually in an independent RTP packet structure and can be directly decoded and played, specifically, the audio media streams and the video media streams can be played and analyzed through Adobe Audio application software, so that the playing conditions of the audio media streams and the video media streams are analyzed, whether the playing of the audio media streams and the video media streams is in a stuck, fuzzy or frame loss condition is analyzed, and the network quality of a user is analyzed and obtained. The network data is screened firstly, so that the network data generated by the real-time video application is obtained, the processing amount of the network data is reduced, and the running speed is improved.
And step S102, extracting the RTP data packet in the original network data file.
In this embodiment, by extracting the RTP packet in the original network data file, the application program using the RTP protocol runs on top of the RTP, and the program performing the RTP runs on the upper layer of the UDP, in order to use the port number and checksum of the UDP. As shown in fig. 4, RTP can be viewed as a sub-layer of the transport layer. The audio and television data blocks generated by the multimedia application are encapsulated in RTP packets, each RTP packet being encapsulated in a UDP message segment and then encapsulated in an IP data packet.
From the application developer's perspective, the RTP executive can be considered part of the application because the developer must integrate the RTP into the application. At a sending end, a developer has to write a program for executing an RTP protocol into an application program for creating an RTP packet, and then the application program sends the RTP packet to a socket interface (socket interface) of UDP, as shown in fig. 5; similarly, at the receiving end, since the RTP packet is input to the application program through the UDP socket interface, the developer must write a program for executing the RTP protocol to the application program for extracting the media data from the RTP packet.
In this embodiment, step S102 includes:
analyzing the file format of an original network data file to obtain a file header, datagram description and a data packet, wherein the file header comprises a MAGIC code, a version number, a time zone, data length and a link type, the datagram description comprises a timestamp and datagram length, and the data packet comprises an ETHER packet header, an IP packet header, a UDP packet header, an RTP packet header and RTP data;
the RTP packets are extracted from the data packets.
After the original network data file is obtained, the file format of the original network data file is analyzed, in this embodiment, the tcpdump format of the original network data file is analyzed, and a header, a datagram description and a data packet are obtained through the analysis, where the header includes a MAGIC code, a version number, a time zone, a data length and a link type, the datagram description includes a timestamp and a datagram length, and the data packet includes an etherer header, an IP header, a UDP header, an RTP header and RTP data. And then, an RTP data packet is extracted from the data packet to obtain the RTP data packet.
Extracting RTP packets from the media data of the application program, specifically, the original network data file is in a tcpdump format, the file format comprises a file header, a datagram descriptor and RTP packets, wherein, the file head is composed of MAGIC code + version code + time zone + data length + link type, the datagram descriptor is composed of timestamp + datagram length, the RTP data packet is composed of ETHER packet head + IP packet head + UDP packet head + RTP data, RTP is a real-time transmission protocol providing end-to-end transmission reset for supporting the transmission of real-time data in single target broadcast and multi-target broadcast network service, RTP can be regarded as a sub-layer of a transmission layer, the sound and video data blocks generated by multimedia application program are encapsulated in RTP information packets, each RTP information packet is encapsulated in UDP message segment and then encapsulated in IP data packet, wherein the IP data packet is a packet transmitted on the network defined by the TCP/IP protocol.
Specifically, first, an original network data file containing an RTP audio/video stream is obtained through tcpdump network package capturing software, where the original network data file is in a tcpdump format, and the format of the original network data file is analyzed as follows:
1. file header (MAGIC code + version number + time zone + data length + link type)
2. Datagram descriptor (time stamp + datagram length)
3. Data packet (ETHER packet head + IP packet head + UDP packet head + RTP packet head)
The method comprises the steps of obtaining RTP data packets from an original network data file according to the format, dividing the RTP data packets into audio media streams and video media streams according to SSRC identifiers, analyzing and processing the audio media streams and the video media streams respectively, wherein the audio media streams are generally independent RTP packet structures and can be decoded and played directly, the video media streams are generally H.264 codes and need to be analyzed and decoded and played according to NAL packet formats, Jffmpeg is a Java multimedia framework plug-in which can be used for playing files of audio and video formats of most formats, analyzing whether the playing of the audio media streams or the video media streams is in a pause, fuzzy or frame loss condition or not, and the video media streams or the audio media streams can be used as index references of bandwidth, packet loss, delay, jitter and the like of a user network, so that the quality of the user network is obtained through analysis.
On the other hand, an original network data file containing RTP audio and video streams can be obtained through Wireshark network packet capturing software, the original network data file is in a tcpdump format, and the format of the original network data file is analyzed as follows:
1. file header (MAGIC code + version number + time zone + data length + link type)
2. Datagram descriptor (time stamp + datagram length)
3. Data packet (ETHER packet head + IP packet head + UDP packet head + RTP packet head)
The method comprises the steps of extracting RTP data packets from an original network data file, dividing the RTP data packets into audio media streams and video media streams according to SSRC identifiers, transmitting and playing back the audio media streams and the video media streams according to timestamp identifiers so as to reproduce abnormal media streams possibly generated due to network transmission reasons in an actual production environment, analyzing whether the playing of the audio media streams or the video media streams is in a stuck, fuzzy or frame loss condition or not, analyzing problems possibly existing when software processes abnormal data, and helping to analyze the network quality of a user.
In some embodiments, in step S102, the method includes:
extracting a data packet in an original network data file;
and acquiring the RTP data packet in the data packet.
And removing a file header and a datagram descriptor in the original network data file to finally obtain a data packet in the original network data file, wherein the data packet comprises an ETHER packet header, an IP packet header, a UDP packet header, an RTP packet header and an RTP data packet.
The ETHER packet header, the IP packet header, the UDP packet header and the RTP packet header in the data packet are removed, and the RTP data packet is left, so that irrelevant information is removed, and the later data processing amount is reduced.
Step S103, carrying out shunting processing on the RTP data packet according to a preset shunting rule to obtain audio streaming media and video streaming media.
The RTP data packets are split by a preset splitting rule, specifically, the RTP data packets can be split into an audio media stream and a video media stream according to a synchronization Signal Source (SSRC) identifier, where the synchronization Signal Source (SSRC) identifier: the special source is that after the mixer receives the RTP messages of one or more synchronous sources, a new combined RTP message is generated through mixing processing, the mixer is used as the SSRC of the combined RTP message, and all the original SSRCs are transmitted to the receiver as CSRCs, so that the receiver knows all the SSRCs forming the combined message, and the subsequent video and audio analysis is convenient.
In this embodiment, step S103 includes:
the RTP packets are streamed out of the audio media stream and the video media stream according to the synchronization source SSRC identifier.
In the RTP protocol, a Synchronization Source (SSRC) is defined as the source of the RTP packet stream, identified by a 32-bit SSRC identifier in the RTP header, so that it is not dependent on the network address. In general, changes of a microphone, an audio interface, a camera and a video interface can cause changes of the SSRC; the synchronization source SSRC is used for identifying the synchronization source, the identifier is randomly selected, two synchronization sources participating in the same video conference cannot have the same SSRC, and therefore the audio media stream and the video media stream in the RTP data packet can be accurately streamed.
And step S104, respectively playing the audio streaming media and the video streaming media, and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media.
In the step of playing the video streaming media, the method includes:
if the video streaming media is H.264 coded, the video streaming media is analyzed and decoded for playing by adopting the NAL packet format.
In this embodiment, an audio stream media and a video stream media are respectively played, and the playing of the audio stream media and the video stream media is analyzed, where the audio stream media is in an independent RTP packet structure and can be directly decoded and played; the video media stream is usually H.264 coding, RTP packet context is relevant, analyze and decode and broadcast according to NAL packet format, H.264 is a video high compression technology, call MPEG-4AVC at the same time, or MPEG-4Part10, under the same reconstructed image quality, H.264 has higher compression ratio, better IP and wireless network channel adaptability than other video compression coding, the high compression ratio makes the data bulk of the picture reduce, have brought convenience for storage and transmission; and then, whether the playing of the audio media stream and the video media stream is blocked, fuzzy or frame loss is analyzed to be used as index references of bandwidth, packet loss, delay, jitter and the like of the user network, and the network quality of the user can be analyzed according to the playing conditions of the audio media stream and the video media stream.
In step S104, the method includes:
analyzing whether the audio streaming media and the video streaming media are stuck, fuzzy or frame loss in the playing process;
and analyzing the network transmission quality by taking the analysis result of the playing condition as a reference index influencing the bandwidth, packet loss, delay and jitter of the network.
And analyzing the audio streaming media and the video streaming media respectively, analyzing whether the audio streaming media and the video streaming media are stuck, fuzzy or frame loss in the playing process, and using the analysis result of the playing condition as a reference index influencing the bandwidth, the packet loss, the delay and the jitter of the network for analyzing the transmission quality of the network. That is to say, whether the audio streaming media or the video streaming media are stuck, fuzzy or frame loss in the playing process is determined, and the playing condition of whether the audio streaming media or the video streaming media are stuck, fuzzy or frame loss is used as a reference index influencing the bandwidth, packet loss, delay and jitter of the network, so as to analyze the transmission quality of the network.
After step S104, the method comprises:
extracting a timestamp from the datagram description;
transmitting and playing back the audio streaming media and the video streaming media according to the time stamps;
acquiring abnormal data generated by transmission;
processing the abnormal data through analysis software;
and if the analysis software processes the problems existing in the abnormal data, optimizing the network transmission quality according to the problems.
Extracting a time stamp from the datagram description, transmitting and replaying the audio streaming media and the video streaming media according to the time stamp so as to reproduce and analyze abnormal data possibly generated due to network transmission reasons in an actual production environment, processing the abnormal data through analysis software, optimizing the network transmission quality according to the problem if the analysis software processes the problem of the abnormal data, and not optimizing the network transmission quality if the analysis software processes the problem of the abnormal data.
In some embodiments, after step S104, the method comprises:
and judging whether to optimize the media transmission algorithm of the software application according to the network transmission quality.
In the embodiment, the network transmission quality of the user is analyzed, whether the media transmission algorithm of the software application needs to be optimized is judged according to whether the playing of the audio media stream and the video media stream is stuck, fuzzy or frame loss, if the media transmission algorithm of the software application needs to be optimized, the media transmission algorithm of the software application is optimized according to whether the playing of the audio media stream and the video media stream is stuck, fuzzy or frame loss, the audio and video streams in the real network environment can be reproduced and completely simulated in the software development and debugging process, the influence of the network environment change of the user on the media transmission is captured, the media transmission algorithm of the software is further optimized, the media stream transmission quality of the software is improved, and the user experience is improved.
In some embodiments, after step S104, the method comprises:
and classifying and storing the playing conditions of the audio media stream and the video media stream and the network transmission quality obtained by analysis.
In this embodiment, the playing condition of the audio/video stream and the network transmission quality obtained by analysis are classified and stored, specifically, the playing condition of the audio media stream is classified and stored according to the stuck, murky and frame loss as tags, and the video media stream can also be classified and stored according to the stuck, fuzzy or frame loss as tags, so that the playing condition of the audio/video stream is clearly and directly obtained, and network optimization operation is conveniently performed on the playing condition of the audio/video stream at a later stage; the content of network optimization comprises: the compression ratio in the media transmission algorithm is optimized, so that the data volume of network transmission is reduced, the network transmission rate is improved, and the pause phenomenon is reduced; according to parameter optimization, parameters such as access, switching and the like are optimized, the situations of pause, blur or frame loss in playing of audio media streams and video media streams are reduced, and of course, problematic hardware can be replaced through hardware detection, a base station neighbor set is modified to enable switching to be reasonable, and switching call drop is reduced to optimize a network.
In summary, an original network data file in a real-time video application is obtained, an RTP data packet in the original network data file is extracted, an audio streaming media and a video streaming media are obtained by streaming in the RTP data packet, network transmission quality is obtained by analyzing playing conditions of the audio streaming media and the video streaming media, audio and video streaming in a real network environment is completely simulated and reproduced, and analysis of user network quality is facilitated.
As shown in fig. 2, an apparatus 1 for analyzing network transmission quality according to an embodiment of the present application includes an obtaining module 11, an extracting module 12, a processing module 13, and an analyzing module 14.
The acquiring module 11 is configured to acquire an original network data file containing an RTP audio/video stream in a real-time video application.
In this embodiment, an original network data file in a real-time video application is obtained, where the real-time video application uses an RTP common Protocol as a media stream transmission Protocol, the original network data file includes an audio/video stream encapsulated by an RTP, and the RTP, which is a Reliable Transport Protocol, is an english abbreviation of a reusable Transport Protocol and is a Protocol for managing transmission and reception of EIGRP packets. By "reliable delivery" is meant that delivery is guaranteed, which is implemented by relying on the Cisco proprietary algorithm called "reliable multicast", where the delivery router sends update information to the multicast IP address 224.0.0.10, and each neighbor receiving a reliable multicast packet sends a unicast acknowledgement packet, and if EIGRP does not acknowledge from a neighbor, it will use unicast to retransmit the same data, and if it still does not acknowledge after 16 unicast attempts, the neighbor will be declared to disappear.
Although the name includes "reliable," RTP does not only provide reliable transport, it also provides unreliable transport. When reliable transport is used, RTP requires the other party to send back an ACK acknowledgement; RTP does not require an ACK when unreliable transport is used.
In this embodiment, the obtaining module 11 includes:
and the first sub-acquisition module is used for acquiring an original network data file transmitted by real-time video application software through network packet capturing software, wherein the original network data file comprises RTP audio and video streams.
Network packet capturing software is adopted for capturing packets, particularly, data packets transmitted in a network can be completely intercepted through tcpdump to provide analysis, the tcpdump supports filtering aiming at a network layer, a protocol, a host, a network or a port, and logic statements such as and, or, not and the like are provided to help a user to remove useless information; of course, the packet capturing can also be performed by using a Wireshark (hereinafter referred to as ethernet), which is a network packet analysis software, and the function of the network packet analysis software is to capture the network packet and display the most detailed network packet data as far as possible. Wireshark uses WinPCAP as an interface, and directly exchanges data with a network card to obtain an original network data file in real-time video application software.
Specifically, the acquisition module 11 includes:
and the tcpdump format file acquisition module is used for acquiring an original network data file containing RTP audio and video stream in the real-time video application through tcpdump or Wireshark network packet capturing software, wherein the file format of the original network data file is tcpdump format.
And acquiring the original network data file by using tcpdump or Wireshark network package capturing software, wherein the file format of the acquired original network data file is in a tcpdump format, and the original network data file is in a tcpdump format, so that the original network data file can be conveniently analyzed subsequently.
In the present embodiment, the apparatus 1 comprises:
and the screening module is used for screening out the network data generated by the real-time video application.
The network data generated by the real-time video application is screened out by analyzing all the network data, then the network data generated by the real-time video application is analyzed to obtain an original network data file containing RTP audio and video streams, then an RTP data packet is obtained from the original network data file, the RTP data packet is divided into audio media streams and video media streams according to an SSRC identifier, then the audio media streams are analyzed and processed, the audio media streams are usually in an independent RTP packet structure and can be directly decoded and played, specifically, the audio media streams and the video media streams can be played and analyzed through Adobe Audio application software, so that the playing conditions of the audio media streams and the video media streams are analyzed, whether the playing of the audio media streams and the video media streams is in a stuck, fuzzy or frame loss condition is analyzed, and the network quality of a user is analyzed and obtained. The network data is screened firstly, so that the network data generated by the real-time video application is obtained, the processing amount of the network data is reduced, and the running speed is improved.
And the extracting module 12 is configured to extract an RTP data packet in the original network data file.
In this embodiment, by extracting the RTP packet in the original network data file, the application program using the RTP protocol runs on top of the RTP, and the program performing the RTP runs on the upper layer of the UDP, in order to use the port number and checksum of the UDP. As shown in fig. 4, RTP can be viewed as a sub-layer of the transport layer. The audio and television data blocks generated by the multimedia application are encapsulated in RTP packets, each RTP packet being encapsulated in a UDP message segment and then encapsulated in an IP data packet.
From the application developer's perspective, the RTP executive can be considered part of the application because the developer must integrate the RTP into the application. At a sending end, a developer has to write a program for executing an RTP protocol into an application program for creating an RTP packet, and then the application program sends the RTP packet to a socket interface (socket interface) of UDP, as shown in fig. 5; similarly, at the receiving end, since the RTP packet is input to the application program through the UDP socket interface, the developer must write a program for executing the RTP protocol to the application program for extracting the media data from the RTP packet.
In the present embodiment, the extraction module 12 includes:
the file format analysis module is used for analyzing the file format of an original network data file to obtain a file header, datagram description and a data packet, wherein the file header comprises a MAGIC code, a version number, a time zone, a data length and a link type, the datagram description comprises a timestamp and a datagram length, and the data packet comprises an ETHER packet header, an IP packet header, a UDP packet header, a RTP packet header and RTP data;
and the RTP data packet extraction module is used for extracting the RTP data packet from the data packet.
After the original network data file is obtained, the file format of the original network data file is analyzed, in this embodiment, the tcpdump format of the original network data file is analyzed, and a header, a datagram description and a data packet are obtained through the analysis, where the header includes a MAGIC code, a version number, a time zone, a data length and a link type, the datagram description includes a timestamp and a datagram length, and the data packet includes an etherer header, an IP header, a UDP header, an RTP header and RTP data. And then, an RTP data packet is extracted from the data packet to obtain the RTP data packet.
Extracting RTP packets from the media data of the application program, specifically, the original network data file is in a tcpdump format, the file format comprises a file header, a datagram descriptor and RTP packets, wherein, the file head is composed of MAGIC code + version code + time zone + data length + link type, the datagram descriptor is composed of timestamp + datagram length, the RTP data packet is composed of ETHER packet head + IP packet head + UDP packet head + RTP data, RTP is a real-time transmission protocol providing end-to-end transmission reset for supporting the transmission of real-time data in single target broadcast and multi-target broadcast network service, RTP can be regarded as a sub-layer of a transmission layer, the sound and video data blocks generated by multimedia application program are encapsulated in RTP information packets, each RTP information packet is encapsulated in UDP message segment and then encapsulated in IP data packet, wherein the IP data packet is a packet transmitted on the network defined by the TCP/IP protocol.
Specifically, first, an original network data file containing an RTP audio/video stream is obtained through tcpdump network package capturing software, where the original network data file is in a tcpdump format, and the format of the original network data file is analyzed as follows:
1. file header (MAGIC code + version number + time zone + data length + link type)
2. Datagram descriptor (time stamp + datagram length)
3. Data packet (ETHER packet head + IP packet head + UDP packet head + RTP packet head)
The method comprises the steps of obtaining RTP data packets from an original network data file according to the format, dividing the RTP data packets into audio media streams and video media streams according to SSRC identifiers, analyzing and processing the audio media streams and the video media streams respectively, wherein the audio media streams are generally independent RTP packet structures and can be decoded and played directly, the video media streams are generally H.264 codes and need to be analyzed and decoded and played according to NAL packet formats, Jffmpeg is a Java multimedia framework plug-in which can be used for playing files of audio and video formats of most formats, analyzing whether the playing of the audio media streams or the video media streams is in a pause, fuzzy or frame loss condition or not, and the video media streams or the audio media streams can be used as index references of bandwidth, packet loss, delay, jitter and the like of a user network, so that the quality of the user network is obtained through analysis.
On the other hand, an original network data file containing RTP audio and video streams can be obtained through Wireshark network packet capturing software, the original network data file is in a tcpdump format, and the format of the original network data file is analyzed as follows:
1. file header (MAGIC code + version number + time zone + data length + link type)
2. Datagram descriptor (time stamp + datagram length)
3. Data packet (ETHER packet head + IP packet head + UDP packet head + RTP packet head)
The method comprises the steps of extracting RTP data packets from an original network data file, dividing the RTP data packets into audio media streams and video media streams according to SSRC identifiers, transmitting and playing back the audio media streams and the video media streams according to timestamp identifiers so as to reproduce abnormal media streams possibly generated due to network transmission reasons in an actual production environment, analyzing whether the playing of the audio media streams or the video media streams is in a stuck, fuzzy or frame loss condition or not, analyzing problems possibly existing when software processes abnormal data, and helping to analyze the network quality of a user.
In some embodiments, the extraction module 12 comprises:
the first sub-extraction module is used for extracting a data packet in the original network data file;
and the first sub RTP data acquisition module is used for acquiring the RTP data packet in the data packet.
And removing a file header and a datagram descriptor in the original network data file to finally obtain a data packet in the original network data file, wherein the data packet comprises an ETHER packet header, an IP packet header, a UDP packet header, an RTP packet header and an RTP data packet.
The ETHER packet header, the IP packet header, the UDP packet header and the RTP packet header in the data packet are removed, and the RTP data packet is left, so that irrelevant information is removed, and the later data processing amount is reduced.
And the processing module 13 is configured to perform a streaming processing on the RTP data packet according to a preset streaming rule to obtain an audio streaming media and a video streaming media.
The RTP data packets are split by a preset splitting rule, specifically, the RTP data packets can be split into an audio media stream and a video media stream according to a synchronization Signal Source (SSRC) identifier, where the synchronization Signal Source (SSRC) identifier: the special source is that after the mixer receives the RTP messages of one or more synchronous sources, a new combined RTP message is generated through mixing processing, the mixer is used as the SSRC of the combined RTP message, and all the original SSRCs are transmitted to the receiver as CSRCs, so that the receiver knows all the SSRCs forming the combined message, and the subsequent video and audio analysis is convenient.
In this embodiment, the processing module 13 includes:
and the first sub-processing module is used for splitting the RTP data packet into an audio media stream and a video media stream according to the synchronous source SSRC identifier.
In the RTP protocol, a Synchronization Source (SSRC) is defined as the source of the RTP packet stream, identified by a 32-bit SSRC identifier in the RTP header, so that it is not dependent on the network address. In general, changes of a microphone, an audio interface, a camera and a video interface can cause changes of the SSRC; the synchronization source SSRC is used for identifying the synchronization source, the identifier is randomly selected, two synchronization sources participating in the same video conference cannot have the same SSRC, and therefore the audio media stream and the video media stream in the RTP data packet can be accurately streamed.
And the analysis module 14 is configured to play the audio streaming media and the video streaming media respectively, and analyze the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media.
The analysis module 14 includes:
the first analysis module is used for analyzing, decoding and playing the video streaming media by adopting the NAL packet format if the video streaming media is H.264 coding.
In this embodiment, an audio stream media and a video stream media are respectively played, and the playing of the audio stream media and the video stream media is analyzed, where the audio stream media is in an independent RTP packet structure and can be directly decoded and played; the video media stream is usually H.264 coding, RTP packet context is relevant, analyze and decode and broadcast according to NAL packet format, H.264 is a video high compression technology, call MPEG-4AVC at the same time, or MPEG-4Part10, under the same reconstructed image quality, H.264 has higher compression ratio, better IP and wireless network channel adaptability than other video compression coding, the high compression ratio makes the data bulk of the picture reduce, have brought convenience for storage and transmission; and then, whether the playing of the audio media stream and the video media stream is blocked, fuzzy or frame loss is analyzed to be used as index references of bandwidth, packet loss, delay, jitter and the like of the user network, and the network quality of the user can be analyzed according to the playing conditions of the audio media stream and the video media stream.
The analysis module 14 includes:
the second analysis module is used for analyzing whether the audio streaming media and the video streaming media are stuck, fuzzy or frame loss in the playing process;
and the third analysis module is used for analyzing the network transmission quality by taking the analysis result of the playing condition as a reference index influencing the bandwidth, packet loss, delay and jitter of the network.
And analyzing the audio streaming media and the video streaming media respectively, analyzing whether the audio streaming media and the video streaming media are stuck, fuzzy or frame loss in the playing process, and using the analysis result of the playing condition as a reference index influencing the bandwidth, the packet loss, the delay and the jitter of the network for analyzing the transmission quality of the network. That is to say, whether the audio streaming media or the video streaming media are stuck, fuzzy or frame loss in the playing process is determined, and the playing condition of whether the audio streaming media or the video streaming media are stuck, fuzzy or frame loss is used as a reference index influencing the bandwidth, packet loss, delay and jitter of the network, so as to analyze the transmission quality of the network.
The apparatus 1 comprises:
a first extraction module to extract timestamps from the datagram description;
the first transmission module is used for transmitting and playing back the audio streaming media and the video streaming media according to the time stamps;
the first acquisition module is used for acquiring abnormal data generated by transmission;
the first processing module is used for processing the abnormal data through analysis software;
and the first optimization module is used for optimizing the network transmission quality according to the problem if the analysis software processes the problem of the abnormal data.
Extracting a time stamp from the datagram description, transmitting and replaying the audio streaming media and the video streaming media according to the time stamp so as to reproduce and analyze abnormal data possibly generated due to network transmission reasons in an actual production environment, processing the abnormal data through analysis software, optimizing the network transmission quality according to the problem if the analysis software processes the problem of the abnormal data, and not optimizing the network transmission quality if the analysis software processes the problem of the abnormal data.
In some embodiments, the apparatus 1 comprises:
and the optimization judging module is used for judging whether to optimize the media transmission algorithm of the software application according to the network transmission quality.
In the embodiment, the network transmission quality of the user is analyzed, whether the media transmission algorithm of the software application needs to be optimized is judged according to whether the playing of the audio media stream and the video media stream is stuck, fuzzy or frame loss, if the media transmission algorithm of the software application needs to be optimized, the media transmission algorithm of the software application is optimized according to whether the playing of the audio media stream and the video media stream is stuck, fuzzy or frame loss, the audio and video streams in the real network environment can be reproduced and completely simulated in the software development and debugging process, the influence of the network environment change of the user on the media transmission is captured, the media transmission algorithm of the software is further optimized, the media stream transmission quality of the software is improved, and the user experience is improved.
In some embodiments, the apparatus 1 comprises:
and the storage module is used for classifying and storing the playing conditions of the audio media stream and the video media stream and the network transmission quality obtained by analysis.
In this embodiment, the playing condition of the audio/video stream and the network transmission quality obtained by analysis are classified and stored, specifically, the playing condition of the audio media stream is classified and stored according to the stuck, murky and frame loss as tags, and the video media stream can also be classified and stored according to the stuck, fuzzy or frame loss as tags, so that the playing condition of the audio/video stream is clearly and directly obtained, and network optimization operation is conveniently performed on the playing condition of the audio/video stream at a later stage; the content of network optimization comprises: the compression ratio in the media transmission algorithm is optimized, so that the data volume of network transmission is reduced, the network transmission rate is improved, and the pause phenomenon is reduced; according to parameter optimization, parameters such as access, switching and the like are optimized, the situations of pause, blur or frame loss in playing of audio media streams and video media streams are reduced, and of course, problematic hardware can be replaced through hardware detection, a base station neighbor set is modified to enable switching to be reasonable, and switching call drop is reduced to optimize a network.
In summary, an original network data file in a real-time video application is obtained, an RTP data packet in the original network data file is extracted, an audio streaming media and a video streaming media are obtained by streaming in the RTP data packet, network transmission quality is obtained by analyzing playing conditions of the audio streaming media and the video streaming media, audio and video streaming in a real network environment is completely simulated and reproduced, and analysis of user network quality is facilitated.
As shown in fig. 3, in the embodiment of the present application, a computer device is further provided, where the computer device may be a server, and an internal structure of the computer device may be as shown in fig. 3. The computer device includes a processor, a memory, a network interface, and a database connected by a system bus. Wherein the computer designed processor is used to provide computational and control capabilities. The memory of the computer device comprises a nonvolatile storage medium and an internal memory. The non-volatile storage medium stores an operating system, a computer program, and a database. The memory provides an environment for the operation of the operating system and the computer program in the non-volatile storage medium. The database of the computer device is used for storing data such as a model of a network transmission quality analysis method. The network interface of the computer device is used for communicating with an external terminal through a network connection. The computer program is executed by a processor to implement a network transmission quality analysis method.
The processor executes the steps of the network transmission quality analysis method: acquiring an original network data file containing RTP audio and video stream in real-time video application; extracting an RTP data packet in the original network data file; carrying out shunting processing on the RTP data packet according to a preset shunting rule to obtain audio streaming media and video streaming media; and respectively playing the audio streaming media and the video streaming media, and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media.
In an embodiment, the step of analyzing and obtaining the network transmission quality according to the playing conditions of the audio media stream and the video media stream by playing the audio streaming media and the video streaming media respectively includes:
analyzing whether the audio streaming media and the video streaming media are stuck, fuzzy or frame loss in the playing process;
and analyzing the network transmission quality by taking the analysis result of the playing condition as a reference index influencing the bandwidth, packet loss, delay and jitter of the network.
In an embodiment, the step of playing the video streaming media includes:
if the video streaming media is H.264 coding, analyzing the video streaming media by adopting an NAL packet format, and decoding and playing the video streaming media.
In an embodiment, the step of performing a streaming process on the RTP data packet according to a preset streaming rule to obtain an audio streaming media and a video streaming media includes:
and splitting the RTP data packets out of the audio media stream and the video media stream according to a synchronous source SSRC identifier.
In an embodiment, the step of extracting the RTP packet in the original network data file includes:
analyzing the file format of the original network data file to obtain a file header, a datagram description and a data packet, wherein the file header comprises a MAGIC code, a version number, a time zone, a data length and a link type, the datagram description comprises a timestamp and a datagram length, and the data packet comprises an ETHER packet header, an IP packet header, a UDP packet header, an RTP packet header and RTP data;
and extracting the RTP data packet from the data packet.
In an embodiment, the step of analyzing the network transmission quality according to the playing condition of the audio media stream and the video media stream after the step of playing the audio media stream and the video media stream respectively comprises:
extracting the timestamp from the datagram description;
transmitting and playing back the audio streaming media and the video streaming media according to the time stamps;
acquiring abnormal data generated by transmission;
processing the exception data by analysis software;
and if the analysis software processes the problems existing in the abnormal data, optimizing the network transmission quality according to the problems.
In an embodiment, the step of acquiring an original network data file containing an RTP audio/video stream in a real-time video application includes:
and acquiring an original network data file containing RTP audio and video stream in real-time video application through tcpdump or Wireshark network packet capturing software, wherein the file format of the original network data file is tcpdump format.
Those skilled in the art will appreciate that the architecture shown in fig. 3 is only a block diagram of some of the structures associated with the disclosed aspects and is not intended to limit the computing devices to which the disclosed aspects may be applied.
The computer device of the embodiment of the application acquires an original network data file in a real-time video application, extracts an RTP data packet in the original network data file, obtains audio streaming media and video streaming media through streaming in the RTP data packet, obtains network transmission quality according to analysis of playing conditions of the audio streaming media and the video streaming media, completely simulates and reproduces audio and video streaming in a real network environment, helps to analyze user network quality, and aims to solve the problem that the existing real-time video application is difficult to analyze user network quality.
An embodiment of the present application further provides a computer-readable storage medium, on which a computer program is stored, where the computer program, when executed by a processor, implements a network transmission quality analysis method, and specifically: acquiring an original network data file containing RTP audio and video stream in real-time video application; extracting an RTP data packet in the original network data file; carrying out shunting processing on the RTP data packet according to a preset shunting rule to obtain audio streaming media and video streaming media; and respectively playing the audio streaming media and the video streaming media, and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media.
In an embodiment, the step of analyzing and obtaining the network transmission quality according to the playing conditions of the audio media stream and the video media stream by playing the audio streaming media and the video streaming media respectively includes:
analyzing whether the audio streaming media and the video streaming media are stuck, fuzzy or frame loss in the playing process;
and analyzing the network transmission quality by taking the analysis result of the playing condition as a reference index influencing the bandwidth, packet loss, delay and jitter of the network.
In an embodiment, the step of playing the video streaming media includes:
if the video streaming media is H.264 coding, analyzing the video streaming media by adopting an NAL packet format, and decoding and playing the video streaming media.
In an embodiment, the step of performing a streaming process on the RTP data packet according to a preset streaming rule to obtain an audio streaming media and a video streaming media includes:
and splitting the RTP data packets out of the audio media stream and the video media stream according to a synchronous source SSRC identifier.
In an embodiment, the step of extracting the RTP packet in the original network data file includes:
analyzing the file format of the original network data file to obtain a file header, a datagram description and a data packet, wherein the file header comprises a MAGIC code, a version number, a time zone, a data length and a link type, the datagram description comprises a timestamp and a datagram length, and the data packet comprises an ETHER packet header, an IP packet header, a UDP packet header, an RTP packet header and RTP data;
and extracting the RTP data packet from the data packet.
In an embodiment, the step of analyzing the network transmission quality according to the playing condition of the audio media stream and the video media stream after the step of playing the audio media stream and the video media stream respectively comprises:
extracting the timestamp from the datagram description;
transmitting and playing back the audio streaming media and the video streaming media according to the time stamps;
acquiring abnormal data generated by transmission;
processing the exception data by analysis software;
and if the analysis software processes the problems existing in the abnormal data, optimizing the network transmission quality according to the problems.
In an embodiment, the step of acquiring an original network data file containing an RTP audio/video stream in a real-time video application includes:
and acquiring an original network data file containing RTP audio and video stream in real-time video application through tcpdump or Wireshark network packet capturing software, wherein the file format of the original network data file is tcpdump format.
The storage medium of the embodiment of the application acquires an original network data file in real-time video application, extracts an RTP data packet in the original network data file, obtains audio streaming media and video streaming media by streaming in the RTP data packet, obtains network transmission quality according to analysis of playing conditions of the audio streaming media and the video streaming media, completely simulates and reproduces audio and video streaming in a real network environment, helps to analyze user network quality, and aims to solve the problem that the existing real-time video application is difficult to analyze user network quality.
It will be understood by those skilled in the art that all or part of the processes of the methods of the embodiments described above can be implemented by hardware instructions of a computer program, which can be stored in a non-volatile computer-readable storage medium, and when executed, can include the processes of the embodiments of the methods described above. Any reference to memory, storage, database, or other medium provided herein and used in the examples may include non-volatile and/or volatile memory. Non-volatile memory can include read-only memory (ROM), Programmable ROM (PROM), Electrically Programmable ROM (EPROM), Electrically Erasable Programmable ROM (EEPROM), or flash memory. Volatile memory can include Random Access Memory (RAM) or external cache memory. By way of illustration and not limitation, RAM is available in a variety of forms such as Static RAM (SRAM), Dynamic RAM (DRAM), Synchronous DRAM (SDRAM), double-rate SDRAM (SSRSDRAM), Enhanced SDRAM (ESDRAM), synchronous link (Synchlink) DRAM (SLDRAM), Rambus Direct RAM (RDRAM), direct bus dynamic RAM (DRDRAM), and bus dynamic RAM (RDRAM).
The above description is only exemplary of the present application and should not be taken as limiting the present application, as any modification, equivalent replacement, or improvement made within the spirit and principle of the present application should be included in the protection scope of the present application.

Claims (8)

1. A method for analyzing network transmission quality, the method comprising:
acquiring an original network data file containing RTP audio and video stream in real-time video application;
extracting an RTP data packet in the original network data file;
carrying out shunting processing on the RTP data packet according to a preset shunting rule to obtain audio streaming media and video streaming media;
respectively playing the audio streaming media and the video streaming media, and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media;
in the step of extracting the RTP packet in the original network data file, the method includes:
analyzing the file format of the original network data file to obtain a file header, a datagram description and a data packet, wherein the file header comprises a MAGIC code, a version number, a time zone, a data length and a link type, the datagram description comprises a timestamp and a datagram length, and the data packet comprises an ETHER packet header, an IP packet header, a UDP packet header, an RTP packet header and RTP data;
extracting the RTP data packet from the data packet;
after the step of respectively playing the audio streaming media and the video streaming media and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media, the method comprises the following steps:
extracting the timestamp from the datagram description;
transmitting and playing back the audio streaming media and the video streaming media according to the time stamps;
acquiring abnormal data generated by transmission;
processing the exception data by analysis software;
and if the analysis software processes the problems existing in the abnormal data, optimizing the network transmission quality according to the problems.
2. The method of claim 1, wherein in the step of separately playing the audio streaming media and the video streaming media and analyzing the network transmission quality according to the playing condition of the audio streaming media and the video streaming media, the method comprises:
analyzing whether the audio streaming media and the video streaming media are stuck, fuzzy or frame loss in the playing process;
and analyzing the network transmission quality by taking the analysis result of the playing condition as a reference index influencing the bandwidth, packet loss, delay and jitter of the network.
3. The method of claim 1, wherein the step of playing the video streaming media comprises:
if the video streaming media is H.264 coding, analyzing the video streaming media by adopting an NAL packet format, and decoding and playing the video streaming media.
4. The method for analyzing network transmission quality according to claim 1, wherein the step of performing a streaming process on the RTP packet according to a preset streaming rule to obtain an audio streaming media and a video streaming media comprises:
and splitting the RTP data packets out of the audio media stream and the video media stream according to a synchronous source SSRC identifier.
5. The method for analyzing network transmission quality according to claim 1, wherein in the step of obtaining an original network data file containing RTP audio/video stream in the real-time video application, the method comprises:
and acquiring an original network data file containing RTP audio and video stream in real-time video application through tcpdump or Wireshark network packet capturing software, wherein the file format of the original network data file is tcpdump format.
6. An apparatus for analyzing network transmission quality, the apparatus comprising:
the acquisition module is used for acquiring an original network data file containing RTP audio and video stream in real-time video application;
the extraction module is used for extracting the RTP data packet in the original network data file;
the processing module is used for carrying out shunting processing on the RTP data packet according to a preset shunting rule to obtain audio streaming media and video streaming media;
the analysis module is used for respectively playing the audio streaming media and the video streaming media and analyzing the network transmission quality according to the playing conditions of the audio streaming media and the video streaming media;
the extraction module comprises:
the file format analysis module is used for analyzing the file format of an original network data file to obtain a file header, datagram description and a data packet, wherein the file header comprises a MAGIC code, a version number, a time zone, a data length and a link type, the datagram description comprises a timestamp and a datagram length, and the data packet comprises an ETHER packet header, an IP packet header, a UDP packet header, a RTP packet header and RTP data;
the RTP data packet extraction module is used for extracting RTP data packets from the data packets;
the network transmission quality analysis device includes:
a first extraction module to extract timestamps from the datagram description;
the first transmission module is used for transmitting and playing back the audio streaming media and the video streaming media according to the time stamps;
the first acquisition module is used for acquiring abnormal data generated by transmission;
the first processing module is used for processing the abnormal data through analysis software;
and the first optimization module is used for optimizing the network transmission quality according to the problem if the analysis software processes the problem of the abnormal data.
7. A computer device comprising a memory and a processor, the memory storing a computer program, characterized in that the processor, when executing the computer program, implements the steps of the method of any of claims 1 to 5.
8. A computer-readable storage medium, on which a computer program is stored, which, when being executed by a processor, carries out the steps of the method of any one of claims 1 to 5.
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