WO2020006912A1 - 网络传输质量分析方法、装置、计算机设备和存储介质 - Google Patents

网络传输质量分析方法、装置、计算机设备和存储介质 Download PDF

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Publication number
WO2020006912A1
WO2020006912A1 PCT/CN2018/108849 CN2018108849W WO2020006912A1 WO 2020006912 A1 WO2020006912 A1 WO 2020006912A1 CN 2018108849 W CN2018108849 W CN 2018108849W WO 2020006912 A1 WO2020006912 A1 WO 2020006912A1
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Prior art keywords
video
audio
media
rtp
network
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English (en)
French (fr)
Inventor
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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Classifications

    • 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 

Definitions

  • the present application relates to the field of computer technology, and in particular, to a method, a device, a computer device, and a storage medium for analyzing network transmission quality.
  • Real-time video applications (such as video conferencing) generally use the RTP public protocol as the media stream transmission protocol.
  • RTP public protocol As the media stream transmission protocol.
  • the impact of changes in the actual user's network environment on media transmission is often elusive.
  • this application proposes a network transmission quality analysis method, device, computer equipment, and storage medium.
  • Audio streaming media and video streaming media are obtained through streaming in RTP packets.
  • the situation analysis obtains the network transmission quality, which aims to solve the problem of existing real-time video applications and difficult to analyze the user's network quality.
  • a network transmission quality analysis method includes:
  • the step of playing the audio streaming media and the video streaming media separately and analyzing and obtaining the network transmission quality according to the playback conditions of the audio media streaming and the video media streaming includes:
  • the analysis result of the playback situation is used as a reference index affecting the bandwidth, packet loss, delay, and jitter of the network, and the network transmission quality is analyzed.
  • step of playing the video streaming media includes:
  • the video streaming media is H.264 encoding
  • the video streaming media is parsed and decoded by using a NAL packet format.
  • the step of performing streaming processing on the RTP data packet to obtain audio streaming media and video streaming media according to a preset streaming rule includes:
  • the RTP data packet is divided out of the audio media stream and the video media stream according to a synchronization source SSRC identifier.
  • step of extracting the RTP data packet in the original network data file includes:
  • the file header includes a MAGIC code, a version number, a time zone, a data length, and a link type.
  • the datagram description includes a time stamp.
  • datagram length the data packet includes an ETHER header, an IP header, a UDP header, an RTP header, and RTP data;
  • the method includes:
  • the analysis software processes a problem with the abnormal data
  • the network transmission quality is optimized according to the problem.
  • the step of obtaining the original network data file including the RTP audio and video stream in the real-time video application includes:
  • the original network data file containing the RTP audio and video stream in the real-time video application is obtained through tcpdump or Wireshark network capture software, and the file format of the original network data file is tcpdump format.
  • the present application also provides a network transmission quality analysis device, where the device includes:
  • An acquisition module which is used to acquire the original network data file containing the RTP audio and video stream in the real-time video application;
  • An extraction module configured to extract an RTP data packet in the original network data file
  • a processing module configured to perform streaming processing on the RTP data packet to obtain audio streaming media and video streaming media according to a preset streaming rule
  • An analysis module is configured to respectively play the audio streaming media and the video streaming media, and analyze the network transmission quality according to the playback conditions of the audio media streaming and the video media streaming.
  • the present application further provides a computer device including a memory and a processor, where the memory stores computer-readable instructions, and when the processor executes the computer-readable instructions, implements the steps of the method according to any one of the foregoing.
  • the present application also provides a computer non-volatile readable storage medium having computer-readable instructions stored thereon, which are executed by a processor to implement the steps of the method according to any one of the foregoing.
  • the present application has the beneficial effects of obtaining the original network data file in the real-time video application, extracting the RTP data packet in the original network data file, and obtaining the audio streaming media and the video streaming media through the RTP data packet distribution.
  • Stream and video media stream playback situation analysis to obtain network transmission quality, fully simulate and reproduce the audio and video streams in the real network environment, help analyze user network quality, aim to solve existing real-time video applications, difficult to analyze user network quality The problem.
  • FIG. 1 is a flowchart of applying 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 device provided by an embodiment of the present application.
  • FIG. 3 is a schematic block diagram of a structure of a computer device provided by an embodiment of the present application.
  • FIG. 4 is a schematic diagram of an RTP protocol on a transport layer applied according to 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.
  • an embodiment of the present application provides a method for analyzing network transmission quality.
  • the method includes the following steps:
  • Step S101 Obtain an original network data file containing a RTP audio and video stream in a real-time video application.
  • the original network data file in the real-time video application is obtained.
  • the real-time video application uses the RTP public protocol as the media stream transmission protocol.
  • the original network data file contains the RTP-encapsulated audio and video stream.
  • the reliable transmission protocol RTP is Reliable Transport.
  • the English abbreviation of Protocol is a protocol used to manage the sending and receiving of EIGRP packets.
  • "Reliable sending” means that sending is guaranteed. This guarantee relies on Cisco's proprietary algorithm. This proprietary algorithm is called “reliable multicast.”
  • the sending router sends the update information to the group. Multicast IP address 224.0.0.10, every neighbor that receives reliable multicast packets will send a unicast acknowledgement packet. If EIGRP does not get a response from a neighbor, it will use unicast to resend the same data. If it still does not answer after 16 unicast attempts, this neighbor will be declared disappeared.
  • RTP does not only provide reliable transmission, it also provides unreliable transmission.
  • RTP requires the other party to send back an ACK acknowledgement; when using unreliable transmission, RTP does not require ACK.
  • step S101 includes:
  • the original network data file transmitted by the real-time video application software is obtained through the network packet capture software, where the original network data file includes RTP audio and video streams.
  • Network capture software is used to capture packets. Specifically, tcpdump can completely intercept data packets transmitted on the network for analysis. It supports filtering at the network layer, protocol, host, network or port, and provides and, or, Not and other logical statements to help you remove useless information; of course, you can also use Wireshark to capture packets.
  • Wireshark (formerly known as Ethereal) is a network packet analysis software. The function of network packet analysis software is to capture network packets and perform Probably the most detailed network packet information. Wireshark uses WinPCAP as an interface to directly exchange data packets with the network card to obtain the original network data files in the real-time video application software.
  • step S101 the method includes:
  • tcpdump or Wireshark network capture software to obtain the original network data files containing RTP audio and video streams in real-time video applications.
  • the file format of the original network data files is tcpdump.
  • the file format of the obtained original network data file is tcpdump.
  • the original network data file is in tcpdump format, which is convenient for subsequent analysis of the original network data file.
  • the method before step S101, the method includes:
  • the network data generated by the real-time video application is filtered, and then the network data generated by the real-time video application is analyzed to obtain the original network data file containing the RTP audio and video streams, and then the original network data is obtained from the original network data.
  • Obtain RTP data packets in the file divide the RTP data packets out of the audio media stream and video media stream according to the SSRC identifier, and then analyze and process the audio media stream.
  • the audio media stream is usually an independent RTP packet structure and can be directly decoded and played.
  • the Adobe Audition application software can be used to analyze and analyze the audio media stream and the video media stream, thereby analyzing the playback of the audio media stream and the video media stream, and analyzing whether the playback of the audio media stream and the video media stream is stuttered, blurred, or lost. Frame conditions, so as to obtain the user network quality.
  • the network data is filtered to obtain the network data generated by the real-time video application, which reduces the processing amount of the network data and improves the operating speed.
  • Step S102 Extract the RTP data packet in the original network data file.
  • RTP can be seen as a sublayer of the transport layer.
  • the audio and TV data blocks generated by the multimedia application are encapsulated in RTP packets, each RTP packet is encapsulated in a UDP message segment, and then encapsulated in an IP data packet.
  • the RTP executor can be considered as part of the application because the developer must integrate RTP into the application.
  • the developer must write the program that implements the RTP protocol into the application that creates the RTP packet, and then the application sends the RTP packet to the UDP socket interface, as shown in Figure 5;
  • the RTP packet is input to the application through the UDP socket interface, so the developer must write the program that implements the RTP protocol to the application that extracts the media data from the RTP packet.
  • step S102 includes:
  • the file header includes the MAGIC code, version number, time zone, data length, and link type.
  • the datagram description includes time stamp and datagram length.
  • the packet includes ETHER header, IP header, UDP header, RTP header, and RTP data;
  • the file format of the original network data file After obtaining the original network data file, analyze the file format of the original network data file. In this embodiment, analyze the tcpdump format of the original network data file.
  • the file header includes the MAGIC code, version number, time zone, data length, and link type.
  • the datagram description includes the timestamp and datagram length.
  • the data packet includes the ETHER header, IP header, UDP header, RTP header, and RTP data. Then extract the RTP data packet from the data packet to obtain the RTP data packet.
  • the file format includes a file header, a datagram descriptor, and an RTP data packet.
  • the file header consists of the MAGIC code + version code + Time zone + data length + link type
  • datagram descriptor consists of timestamp + datagram length
  • RTP data packet consists of ETHER header + IP header + UDP header + RTP header + RTP data
  • RTP is a kind of end-to-end End-to-end reset real-time transmission protocol, used to support the transmission of real-time data in single-target broadcast and multi-target broadcast network services.
  • RTP can be seen as a sub-layer of the transport layer.
  • each RTP information packet is encapsulated in a UDP message segment, and then encapsulated in an IP data packet, where the IP data packet is a packet transmitted on the network as defined by the TCP / IP protocol.
  • the original network data file containing the RTP audio and video stream is obtained through the tcpdump network packet capture software.
  • the original network data file is in the tcpdump format.
  • the format of the original network data file is analyzed as follows:
  • the audio media stream Usually it is an independent RTP packet structure, which can be directly decoded and played.
  • the video media stream is usually H.264 encoded, which needs to be parsed and decoded according to the NAL packet format.
  • Jffmpeg is a Java
  • the multimedia framework plug-in can be used to play most formats of audio and video files, and analyze whether the playback of these audio media streams or video media streams is stuttered, blurred, or dropped. It can be used as the bandwidth and loss of the user's network. Packets, delays, and jitters are referenced to analyze and obtain user network quality.
  • the Original network data file is in the tcpdump format.
  • the format of the original network data file is analyzed as follows:
  • the RTP data packet is extracted from the original network data file, and then the RTP data packet is divided out of the audio media stream and the video media stream according to the SSRC identifier.
  • the audio media stream and the video media stream can be transmitted and played back according to the timestamp identifier to Reproduce abnormal media streams that may occur due to network transmission in the actual production environment, analyze whether the playback of these audio media streams or video media streams are stuttered, blurred, or dropped frames, and analyze problems that may exist when the software processes abnormal data. Help analyze user network quality.
  • step S102 the method includes:
  • the file header and the datagram descriptor in the original network data file are removed, and finally a data packet in the original network data file is obtained.
  • the data packet includes an ETHER header, an IP header, a UDP header, an RTP header, and an RTP data packet.
  • the ETHER header, IP header, UDP header, and RTP header are removed from the data packet, and the rest is the RTP data packet, which removes irrelevant information and reduces the amount of later data processing.
  • Step S103 Perform streaming processing on the RTP data packet according to a preset streaming rule to obtain audio streaming media and video streaming media.
  • RTP data packets are shunted by preset shunting rules. Specifically, the RTP data packets can be streamed out of the audio media stream and video media stream according to the SSRC identifier.
  • the SSRC identifier 32 bits, used to identify the synchronization source. This identifier is randomly selected. Two synchronization sources participating in the same video conference cannot have the same SSRC. Here, the synchronization source refers to the source that generates the media stream.
  • a 32-bit SSRC identifier in the RTP header is used to identify, instead of relying on the network address, the receiver will distinguish different sources according to the SSRC identifier and group RTP messages.
  • the special source refers to the mixer After receiving RTP messages from one or more synchronous sources, a new combined RTP message is generated after the mixing process, and the mixer is used as the SSRC of the combined RTP message, and all the original SSRCs are transmitted as CSRC to The receiver allows the receiver to know the SSRCs that make up the combined message, which facilitates subsequent analysis of video and audio separately.
  • step S103 includes:
  • the RTP data packet is divided out of the audio media stream and the video media stream according to the synchronization source SSRC identifier.
  • a synchronization source In the RTP protocol, a synchronization source (SSRC, Synchronization source) is defined as the source of the RTP packet stream, and is identified by a 32-bit SSRC identifier in the RTP header, so that it does not depend on the network address. Usually changes in the microphone, audio interface, camera, and video interface will cause changes in the SSRC; the synchronization source SSRC is used to identify the synchronization source. This identifier is randomly selected. Two synchronization sources participating in the same video conference cannot have The same SSRC, so that the audio media stream and video media stream in the RTP packet can be accurately split out.
  • SSRC Synchronization source
  • step S104 the audio streaming media and the video streaming media are played separately, and the network transmission quality is analyzed according to the playback conditions of the audio media streaming and the video media streaming.
  • the step of playing the video streaming media includes:
  • the video streaming media is H.264 encoding
  • the video streaming media is parsed and decoded by using the NAL packet format.
  • the audio streaming media and the video streaming media are played separately, and the playback of the audio media streaming and the video media streaming is analyzed.
  • the audio media streaming is an independent RTP packet structure and can be directly decoded and played.
  • the video media streaming is usually H .264 encoding, RTP packets are context-dependent, and they are parsed and decoded according to the NAL packet format.
  • H.264 is a video high compression technology, also known as MPEG-4 AVC, or MPEG-4 Part10: Under the same reconstructed image quality, H.264 has a higher compression ratio, better IP and wireless network channel adaptability than other video compression coding. The high compression rate reduces the amount of data in the image, which is used for storage and transmission.
  • step S104 the method includes:
  • Analyze audio streaming media and video streaming media separately analyze whether audio streaming media and video streaming media are stuck, blurry, or drop frames during playback, and use the analysis results of playback conditions as affecting network bandwidth, packet loss, delay, and A reference index for jitter, used to analyze network transmission quality. That is to say, whether the audio streaming or video streaming is stuttered, blurred, or dropped during playback, and whether the stuttered, blurred, or dropped frames are affected as the network bandwidth, packet loss, delay, and jitter. Reference indicators to analyze network transmission quality.
  • step S104 the method includes:
  • the network transmission quality is optimized based on the problems.
  • step S104 the method includes:
  • the user's network transmission quality is analyzed, and it is determined whether the media transmission algorithm of the software application needs to be optimized according to whether the playback of the audio media stream and the video media stream is stuttered, blurred, or dropped.
  • the applied media transmission algorithm optimizes the software application's media transmission algorithm based on whether the audio media stream and video media stream play are stuttered, blurred, or dropped frames. It can be reproduced and completely during the software development and debugging process. Simulate the audio and video streams in a real network environment, capture the impact of changes in the user's network environment on media transmission, and then optimize the software's media transmission algorithm to improve the software's media stream transmission quality and user experience.
  • step S104 the method includes:
  • the playback conditions of the audio and video streams and the network transmission quality obtained by analysis are classified and stored.
  • the playback conditions of the audio media streams are classified and stored according to stutter, noise, and dropped frames as tags.
  • the optimization content includes: optimizing the compression rate in the media transmission algorithm, thereby reducing the amount of data transmitted on the network, increasing the network transmission rate, and reducing the stutter phenomenon; according to the parameter optimization, optimizing the parameters such as access and switching to reduce the audio media stream
  • stuttering, blurring, or dropped frames when playing with video media streams of course, you can also use hardware detection to replace the problematic hardware, modify the base station neighbor set to make the handover reasonable, and reduce the number of dropped calls to optimize the network.
  • the original network data file in the real-time video application is obtained, the RTP data packet in the original network data file is extracted, and the audio streaming media and video streaming media are obtained through the streaming in the RTP data packet.
  • the audio media streaming and video media streaming The analysis of the playback situation can obtain the network transmission quality, completely simulate and reproduce the audio and video streams in the real network environment, help analyze the user's network quality, and aim to solve the existing real-time video applications, which is difficult to analyze the user's network quality.
  • an embodiment of the present application provides a network transmission quality analysis device.
  • the device 1 includes an acquisition module 11, an extraction module 12, a processing module 13, and an analysis module 14.
  • the obtaining module 11 is configured to obtain an original network data file including an RTP audio and video stream in a real-time video application.
  • the original network data file in the real-time video application is obtained.
  • the real-time video application uses the RTP public protocol as the media stream transmission protocol.
  • the original network data file contains the RTP-encapsulated audio and video stream.
  • the reliable transmission protocol RTP is Reliable Transport.
  • the English abbreviation of Protocol is a protocol used to manage the sending and receiving of EIGRP packets.
  • "Reliable sending” means that sending is guaranteed. This guarantee relies on Cisco's proprietary algorithm. This proprietary algorithm is called “reliable multicast.”
  • the sending router sends the update information to the group. Multicast IP address 224.0.0.10, every neighbor that receives reliable multicast packets will send a unicast acknowledgement packet. If EIGRP does not get a response from a neighbor, it will use unicast to resend the same data. If it still does not answer after 16 unicast attempts, this neighbor will be declared disappeared.
  • RTP does not only provide reliable transmission, it also provides unreliable transmission.
  • RTP requires the other party to send back an ACK acknowledgement; when using unreliable transmission, RTP does not require ACK.
  • the obtaining module 11 includes:
  • the first sub-obtaining module is configured to obtain the original network data file transmitted by the real-time video application software through a network capture software, where the original network data file includes an RTP audio and video stream.
  • Network capture software is used to capture packets. Specifically, tcpdump can completely intercept data packets transmitted on the network for analysis. It supports filtering at the network layer, protocol, host, network or port, and provides and, or, Not and other logical statements to help you remove useless information; of course, you can also use Wireshark to capture packets.
  • Wireshark (formerly known as Ethereal) is a network packet analysis software. The function of network packet analysis software is to capture network packets and perform Probably the most detailed network packet information. Wireshark uses WinPCAP as an interface to directly exchange data packets with the network card to obtain the original network data files in the real-time video application software.
  • the obtaining module 11 includes:
  • the tcpdump format file acquisition module is used to obtain the original network data file containing the RTP audio and video stream in the real-time video application through tcpdump or Wireshark network capture software.
  • the file format of the original network data file is tcpdump format.
  • the file format of the obtained original network data file is tcpdump.
  • the original network data file is in tcpdump format, which is convenient for subsequent analysis of the original network data file.
  • the apparatus 1 includes:
  • the filtering module is used for filtering out network data generated by a real-time video application.
  • the network data generated by the real-time video application is filtered, and then the network data generated by the real-time video application is analyzed to obtain the original network data file containing the RTP audio and video streams, and then the original network data is obtained from the original network data.
  • Obtain RTP data packets in the file divide the RTP data packets out of the audio media stream and video media stream according to the SSRC identifier, and then analyze and process the audio media stream.
  • the audio media stream is usually an independent RTP packet structure and can be directly decoded and played.
  • the Adobe Audition application software can be used to analyze and analyze the audio media stream and the video media stream, thereby analyzing the playback of the audio media stream and the video media stream, and analyzing whether the playback of the audio media stream and the video media stream is stuttered, blurred, or lost. Frame conditions, so as to obtain the user network quality.
  • the network data is filtered to obtain the network data generated by the real-time video application, which reduces the processing amount of the network data and improves the operating speed.
  • the extraction module 12 is configured to extract an RTP data packet in the original network data file.
  • RTP can be seen as a sublayer of the transport layer.
  • the audio and TV data blocks generated by the multimedia application are encapsulated in RTP packets, each RTP packet is encapsulated in a UDP message segment, and then encapsulated in an IP data packet.
  • the RTP executor can be considered as part of the application because the developer must integrate RTP into the application.
  • the developer must write the program that implements the RTP protocol into the application that creates the RTP packet, and then the application sends the RTP packet to the UDP socket interface, as shown in Figure 5;
  • the RTP packet is input to the application through the UDP socket interface, so the developer must write the program that implements the RTP protocol to the application that extracts the media data from the RTP packet.
  • the extraction module 12 includes:
  • File format analysis module used to analyze the file format of the original network data file, to obtain the file header, datagram description and data packet.
  • the file header includes the MAGIC code, version number, time zone, data length, and link type. Stamp and datagram length, the data packet includes ETHER header, IP header, UDP header, RTP header and RTP data;
  • RTP data packet extraction module used to extract RTP data packets from the data packets.
  • the file format of the original network data file After obtaining the original network data file, analyze the file format of the original network data file. In this embodiment, analyze the tcpdump format of the original network data file.
  • the file header includes the MAGIC code, version number, time zone, data length, and link type.
  • the datagram description includes the timestamp and datagram length.
  • the data packet includes the ETHER header, IP header, UDP header, RTP header, and RTP data. Then extract the RTP data packet from the data packet to obtain the RTP data packet.
  • the file format includes a file header, a datagram descriptor, and an RTP data packet.
  • the file header consists of the MAGIC code + version code + Time zone + data length + link type
  • datagram descriptor consists of timestamp + datagram length
  • RTP data packet consists of ETHER header + IP header + UDP header + RTP header + RTP data
  • RTP is a kind of end-to-end End-to-end reset real-time transmission protocol, used to support the transmission of real-time data in single-target broadcast and multi-target broadcast network services.
  • RTP can be seen as a sub-layer of the transport layer.
  • each RTP information packet is encapsulated in a UDP message segment, and then encapsulated in an IP data packet, where the IP data packet is a packet transmitted on the network as defined by the TCP / IP protocol.
  • the original network data file containing the RTP audio and video stream is obtained through the tcpdump network packet capture software.
  • the original network data file is in the tcpdump format.
  • the format of the original network data file is analyzed as follows:
  • the audio media stream Usually it is an independent RTP packet structure, which can be directly decoded and played.
  • the video media stream is usually H.264 encoded, which needs to be parsed and decoded according to the NAL packet format.
  • Jffmpeg is a Java
  • the multimedia framework plug-in can be used to play most formats of audio and video files, and analyze whether the playback of these audio media streams or video media streams is stuttered, blurred, or dropped. It can be used as the bandwidth and loss of the user's network. Packets, delays, and jitters are referenced to analyze and obtain user network quality.
  • the Original network data file is in the tcpdump format.
  • the format of the original network data file is analyzed as follows:
  • the RTP data packet is extracted from the original network data file, and then the RTP data packet is divided out of the audio media stream and the video media stream according to the SSRC identifier.
  • the audio media stream and the video media stream can be transmitted and played back according to the timestamp identifier to Reproduce abnormal media streams that may occur due to network transmission in the actual production environment, analyze whether the playback of these audio media streams or video media streams are stuttered, blurred, or dropped frames, and analyze problems that may exist when the software processes abnormal data. Help analyze user network quality.
  • the extraction module 12 includes:
  • a first sub-extraction module for extracting data packets in the original network data file
  • the first sub-RTP data acquisition module is configured to acquire an RTP data packet in the data packet.
  • the file header and the datagram descriptor in the original network data file are removed, and finally a data packet in the original network data file is obtained.
  • the data packet includes an ETHER header, an IP header, a UDP header, an RTP header, and an RTP data packet.
  • the ETHER header, IP header, UDP header, and RTP header are removed from the data packet, and the rest is the RTP data packet, which removes irrelevant information and reduces the amount of later data processing.
  • the processing module 13 is configured to perform streaming processing on the RTP data packet according to a preset streaming rule to obtain audio streaming media and video streaming media.
  • RTP data packets are shunted by preset shunting rules. Specifically, the RTP data packets can be streamed out of the audio media stream and video media stream according to the SSRC identifier.
  • the SSRC identifier 32 bits, used to identify the synchronization source. This identifier is randomly selected. Two synchronization sources participating in the same video conference cannot have the same SSRC. Here, the synchronization source refers to the source that generates the media stream.
  • a 32-bit SSRC identifier in the RTP header is used to identify, instead of relying on the network address, the receiver will distinguish different sources according to the SSRC identifier and group RTP messages.
  • the special source refers to the mixer After receiving RTP messages from one or more synchronous sources, a new combined RTP message is generated after the mixing process, and the mixer is used as the SSRC of the combined RTP message, and all the original SSRCs are transmitted as CSRC to The receiver allows the receiver to know the SSRCs that make up the combined message, which facilitates subsequent analysis of video and audio separately.
  • the processing module 13 includes:
  • the first sub-processing module is configured to divide the RTP data packet out of the audio media stream and the video media stream according to the synchronization source SSRC identifier.
  • a synchronization source In the RTP protocol, a synchronization source (SSRC, Synchronization source) is defined as the source of the RTP packet stream, and is identified by a 32-bit SSRC identifier in the RTP header, so that it does not depend on the network address. Usually changes in the microphone, audio interface, camera, and video interface will cause changes in the SSRC; the synchronization source SSRC is used to identify the synchronization source. This identifier is randomly selected. Two synchronization sources participating in the same video conference cannot have The same SSRC, so that the audio media stream and video media stream in the RTP packet can be accurately split out.
  • SSRC Synchronization source
  • the analysis module 14 is configured to play audio streaming media and video streaming media respectively, and analyze network transmission quality according to the playback conditions of the audio media streaming and the video media streaming.
  • the analysis module 14 includes:
  • the first analysis module is configured to parse and decode the video streaming media by using the NAL packet format if the video streaming media is H.264 encoding.
  • the audio streaming media and the video streaming media are played separately, and the playback of the audio media streaming and the video media streaming is analyzed.
  • the audio media streaming is an independent RTP packet structure and can be directly decoded and played.
  • the video media streaming is usually H .264 encoding, RTP packets are context-dependent, and they are parsed and decoded according to the NAL packet format.
  • H.264 is a video high compression technology, also known as MPEG-4 AVC, or MPEG-4 Part10: Under the same reconstructed image quality, H.264 has a higher compression ratio, better IP and wireless network channel adaptability than other video compression coding. The high compression rate reduces the amount of data in the image, which is used for storage and transmission.
  • the analysis module 14 includes:
  • a second analysis module which is used to analyze whether audio streaming media and video streaming media are stuck, blurred, or dropped during playback
  • the third analysis module is used to analyze the network transmission quality by using the analysis result of the playback situation as a reference index affecting the bandwidth, packet loss, delay, and jitter of the network.
  • Analyze audio streaming media and video streaming media separately analyze whether audio streaming media and video streaming media are stuck, blurry, or drop frames during playback, and use the analysis results of playback conditions as affecting network bandwidth, packet loss, delay, and A reference index for jitter, used to analyze network transmission quality. That is to say, whether the audio streaming or video streaming is stuttered, blurred, or dropped during playback, and whether the stuttered, blurred, or dropped frames are affected as the network bandwidth, packet loss, delay, and jitter. Reference indicators to analyze network transmission quality.
  • Device 1 includes:
  • a first extraction module configured to extract a timestamp from a datagram description
  • a first transmission module configured to transmit and play back audio streaming media and video streaming media according to a timestamp
  • a first acquisition module configured to acquire abnormal data generated by transmission
  • a first processing module configured to process abnormal data through analysis software
  • the first optimization module is configured to optimize the network transmission quality according to the problem if the analysis software processes the problem with the abnormal data.
  • the apparatus 1 includes:
  • the optimization judging module is used to judge whether to optimize the media transmission algorithm of the software application according to the network transmission quality.
  • the user's network transmission quality is analyzed, and it is determined whether the media transmission algorithm of the software application needs to be optimized according to whether the playback of the audio media stream and the video media stream is stuttered, blurred, or dropped.
  • the applied media transmission algorithm optimizes the software application's media transmission algorithm based on whether the audio media stream and video media stream play are stuttered, blurred, or dropped frames. It can be reproduced and completely during the software development and debugging process. Simulate the audio and video streams in a real network environment, capture the impact of changes in the user's network environment on media transmission, and then optimize the software's media transmission algorithm to improve the software's media stream transmission quality and user experience.
  • the apparatus 1 includes:
  • the storage module is configured to classify and store the playback conditions of the audio media stream and the video media stream and the network transmission quality obtained through analysis.
  • the playback conditions of the audio and video streams and the network transmission quality obtained by analysis are classified and stored.
  • the playback conditions of the audio media streams are classified and stored according to stutter, noise, and dropped frames as tags.
  • the optimization content includes: optimizing the compression rate in the media transmission algorithm, thereby reducing the amount of data transmitted on the network, increasing the network transmission rate, and reducing the stutter phenomenon; according to the parameter optimization, optimizing the parameters such as access and switching to reduce the audio media stream
  • stuttering, blurring, or dropped frames when playing with video media streams of course, you can also use hardware detection to replace the problematic hardware, modify the base station neighbor set to make the handover reasonable, and reduce the number of dropped calls to optimize the network.
  • the original network data file in the real-time video application is obtained, the RTP data packet in the original network data file is extracted, and the audio streaming media and video streaming media are obtained through the streaming in the RTP data packet.
  • the audio media streaming and video media streaming The analysis of the playback situation can obtain the network transmission quality, completely simulate and reproduce the audio and video streams in the real network environment, help analyze the user's network quality, and aim to solve the existing real-time video applications, which is difficult to analyze the user's network quality.
  • a computer device is also provided in the embodiment of the present application.
  • the computer device may be a server, and its internal structure may be as shown in FIG. 3.
  • the computer device includes a processor, a memory, a network interface, and a database connected through a system bus.
  • the computer design processor is used to provide computing and control capabilities.
  • the memory of the computer device includes a non-volatile storage medium and an internal memory.
  • the non-volatile storage medium stores an operating system, computer-readable instructions, and a database.
  • the memory provides an environment for operating systems and computer-readable instructions in a non-volatile storage medium.
  • the database of the computer equipment is used to store data such as models of network transmission quality analysis methods.
  • the network interface of the computer device is used to communicate with an external terminal through a network connection.
  • the computer-readable instructions are executed by a processor to implement a network transmission quality analysis method.
  • the processor executes the steps of the network transmission quality analysis method: obtaining an original network data file containing an RTP audio and video stream in a real-time video application; extracting an RTP data packet in the original network data file; RTP data packets are streamed to obtain audio streaming media and video streaming media; the audio streaming media and the video streaming media are played separately, and the network transmission quality is analyzed according to the playback conditions of the audio media streaming and the video media streaming.
  • the foregoing steps of playing the audio streaming media and the video streaming media separately, and analyzing and obtaining the network transmission quality according to the playback conditions of the audio media streaming and the video media streaming include:
  • the analysis result of the playback situation is used as a reference index affecting the bandwidth, packet loss, delay, and jitter of the network, and the network transmission quality is analyzed.
  • the step of playing the video streaming media includes:
  • the video streaming media is H.264 encoding
  • the video streaming media is parsed and decoded by using a NAL packet format.
  • the foregoing step of performing stream processing on the RTP data packet to obtain audio streaming media and video streaming media according to a preset streaming rule includes:
  • the RTP data packet is divided out of the audio media stream and the video media stream according to a synchronization source SSRC identifier.
  • the step of extracting the RTP data packet in the original network data file includes:
  • the file header includes a MAGIC code, a version number, a time zone, a data length, and a link type.
  • the datagram description includes a time stamp.
  • datagram length the data packet includes an ETHER header, an IP header, a UDP header, an RTP header, and RTP data;
  • the step of playing the audio streaming media and the video streaming media separately according to the playback conditions of the audio media streaming and the video media streaming, after analyzing the network transmission quality includes:
  • the analysis software processes a problem with the abnormal data
  • the network transmission quality is optimized according to the problem.
  • the step of obtaining the original network data file including the RTP audio and video stream in the real-time video application includes:
  • the original network data file containing the RTP audio and video stream in the real-time video application is obtained through tcpdump or Wireshark network capture software, and the file format of the original network data file is tcpdump format.
  • FIG. 3 is only a block diagram of a part of the structure related to the solution of the present application, and does not constitute a limitation on the computer equipment to which the solution of the present application is applied.
  • the computer device in the embodiment of the present application obtains the original network data file in the real-time video application, extracts the RTP data packet in the original network data file, and obtains the audio streaming media and the video streaming media through the RTP data packet distribution.
  • the analysis of media stream playback can obtain the network transmission quality, completely simulate and reproduce the audio and video streams in the real network environment, help analyze the user's network quality, and aim to solve the existing real-time video applications, which is difficult to analyze the network quality of users.
  • An embodiment of the present application further provides a computer non-volatile readable storage medium, which stores computer-readable instructions.
  • a computer non-volatile readable storage medium which stores computer-readable instructions.

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Abstract

本申请涉及一种网络传输质量分析方法、装置、计算机设备和存储介质。获取包含RTP音视频流的原始网络数据文件；提取原始网络数据文件中的RTP数据包；根据预设分流规则将RTP数据包进行分流处理得到音频流媒体和视频流媒体；分别播放所述音频流媒体和所述视频流媒体，根据音频媒体流和视频媒体流的播放情况，分析网络传输质量。

Description

网络传输质量分析方法、装置、计算机设备和存储介质
本申请要求于2018年7月5日提交中国专利局、申请号为2018107316200，申请名称为“网络传输质量分析方法、装置、计算机设备和存储介质”的中国专利申请的优先权，其全部内容通过引用结合在本申请中。
技术领域
本申请涉及计算机技术领域，特别涉及一种网络传输质量分析方法、装置、计算机设备和存储介质。
 背景技术
实时视频应用（如视频会议）一般采用RTP公共协议作为媒体流传输协议，互联网环境下，实际用户的网络环境变化对媒体传输的影响经常难以捉摸，在现有的技术中，难于分析用户的网络质量。
 技术问题
针对现有技术不足，本申请提出一种网络传输质量分析方法、装置、计算机设备和存储介质，通过RTP数据包中分流得到音频流媒体和视频流媒体，根据音频媒体流和视频媒体流的播放情况分析得到网络传输质量，旨在解决现有的实时视频应用，难于分析用户的网络质量的问题。
 技术解决方案
本申请提出的技术方案是：
一种网络传输质量分析方法，所述方法包括：
获取实时视频应用中包含RTP音视频流的原始网络数据文件；
提取所述原始网络数据文件中的RTP数据包；
根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体；
分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量。
进一步地，在所述分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析得到网络传输质量的步骤中，包括：
分析所述音频流媒体和所述视频流媒体在播放过程中是否卡顿、模糊或丢帧；
将所述播放情况的分析结果作为影响所述网络的带宽、丢包、延迟和抖动的参考指标，分析所述网络传输质量。
进一步地，在所述播放所述视频流媒体的步骤中，包括：
若所述视频流媒体为H.264编码，则通过采用NAL封包格式对所述视频流媒体进行解析并解码播放。
进一步地，在所述根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体的步骤中，包括：
按照同步信源SSRC标识符将所述RTP数据包分流出所述音频媒体流和所述视频媒体流。
进一步地，在所述提取所述原始网络数据文件中的RTP数据包的步骤中，包括：
分析所述原始网络数据文件的文件格式，获得文件头、数据报描述和数据包，所述文件头包括MAGIC码、版本号、时区、数据长度和链路类型，所述数据报描述包括时间戳和数据报长度，所述数据包包括ETHER包头、IP包头、UDP包头、RTP包头和RTP数据；
从所述数据包中提取所述RTP数据包。
进一步地，在所述分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量的步骤之后，所述方法包括：
从所述数据报描述提取所述时间戳；
按照所述时间戳将所述音频流媒体和所述视频流媒体进行传输回放；
获取因传输所产生的异常数据；
通过分析软件处理所述异常数据；
若所述分析软件处理所述异常数据存在的问题，则根据所述问题优化所述网络传输质量。
进一步地，在所述获取实时视频应用中包含RTP音视频流的原始网络数据文件的步骤中，包括：
通过tcpdump或者Wireshark网络抓包软件获取实时视频应用中包含RTP音视频流的原始网络数据文件，所述原始网络数据文件的文件格式为tcpdump格式。
本申请还提供一种网络传输质量分析装置，所述装置包括：
获取模块，用于获取实时视频应用中包含RTP音视频流的原始网络数据文件；
提取模块，用于提取所述原始网络数据文件中的RTP数据包；
处理模块，用于根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体；
分析模块，用于分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量。
本申请还提供一种计算机设备，包括存储器和处理器，所述存储器存储有计算机可读指令，所述处理器执行所述计算机可读指令时实现上述任一项所述的方法的步骤。
本申请还提供一种计算机非易失性可读存储介质，其上存储有计算机可读指令，所述计算机可读指令被处理器执行时实现上述任一项所述的方法的步骤。
 有益效果
根据上述的技术方案，本申请有益效果：获取实时视频应用中原始网络数据文件，提取原始网络数据文件中的RTP数据包，通过RTP数据包中分流得到音频流媒体和视频流媒体，根据音频媒体流和视频媒体流的播放情况分析得到网络传输质量，完全模拟和重现真实网络环境中的音视频流，帮助分析用户网络质量，旨在解决现有的实时视频应用，难于分析用户的网络质量的问题。
 附图说明
图1是应用本申请实施例提供的网络传输质量分析方法的流程图；
图2是应用本申请实施例提供的网络传输质量分析装置的功能模块图；
图3是应用本申请实施例提供的计算机设备的结构示意框图；
图4是应用本申请实施例提供的RTP是传输层上的协议示意图；
图5是应用本申请实施例提供的RTP和UDP之间的接口示意图。
 本发明的最佳实施方式
为了使本申请的目的、技术方案及优点更加清楚明白，以下结合附图及实施例，对本申请进行进一步详细说明。应当理解，此处所描述的具体实施例仅仅用以解释本申请，并不用于限定本申请。
如图1所示，本申请实施例提出一种网络传输质量分析方法，所述方法包括以下步骤：
步骤S101、获取实时视频应用中包含RTP音视频流的原始网络数据文件。
在本实施例中，获取实时视频应用中原始网络数据文件，其中实时视频应用采用RTP公共协议作为媒体流传输协议，原始网络数据文件中包含RTP封装的音视频流，可靠传输协议RTP是Reliable Transport Protocol的英文缩写，是用来管理EIGRP数据包的发送和接收的协议。“可靠的发送”是指发送是有保障的，这种保障依赖Cisco公司私有的算法来实现，这个私有的算法被称为“可靠组播（reliable multicast）”，发送路由器将更新信息发送到组播IP地址224.0.0.10，每一个接收可靠组播数据包的邻居都会发送一个单播的确认数据包，如果EIGRP没有从某个邻居那里得到应答，它将使用单播来重发同样的数据，如果在16次单播尝试后，它仍然没有应答，则此邻居将被宣告消失。
虽然名称中包含“可靠”，但RTP并不仅仅提供可靠传输，它也提供不可靠传输。当使用可靠传输时，RTP要求对方发回ACK确认；当使用不可靠传输时，RTP不要求ACK。
在本实施例中，在步骤S101中，包括：
通过网络抓包软件获取实时视频应用软件传输的原始网络数据文件，其中原始网络数据文件中包括RTP音视频流。
采用网络抓包软件进行抓包，具体地，通过tcpdump可以将网络中传送的数据包完全截获下来提供分析，它支持针对网络层、协议、主机、网络或端口的过滤，并提供and、or、not等逻辑语句来帮助你去掉无用的信息；当然，还可以通过Wireshark来进行抓包，Wireshark（前称Ethereal）是一个网络封包分析软件，网络封包分析软件的功能是撷取网络封包，并尽可能显示出最为详细的网络封包资料。Wireshark使用WinPCAP作为接口，直接与网卡进行数据报文交换，从而获取实时视频应用软件中的原始网络数据文件。
具体地，在步骤S101中，包括：
通过tcpdump或者Wireshark网络抓包软件获取实时视频应用中包含RTP音视频流的原始网络数据文件，原始网络数据文件的文件格式为tcpdump格式。
使用tcpdump或者Wireshark网络抓包软件获取原始网络数据文件，得到的原始网络数据文件的文件格式为tcpdump格式，原始网络数据文件为tcpdump格式，便于后续对原始网络数据文件进行分析。
在本实施例中，在步骤S101之前，所述方法包括：
筛选出实时视频应用产生的网络数据。
通过先对所有网络数据进行分析，从而筛选出实时视频应用产生的网络数据，再对实时视频应用产生的网络数据进行分析，以获取包含RTP音视频流的原始网络数据文件，然后从原始网络数据文件中获取RTP数据包，将RTP数据包按照SSRC标识符分流出音频媒体流和视频媒体流，再对音频媒体流进行分析处理，音频媒体流通常为独立的RTP包结构，可以直接解码播放，具体地，可以通过Adobe Audition应用软件播放分析音频媒体流和视频媒体流，从而分析音频媒体流和视频媒体流的播放情况，分析音频媒体流和视频媒体流的播放是否有卡顿、模糊或丢帧情况，从而分析得到用户网络质量。先对网络数据进行筛选，从而获取实时视频应用产生的网络数据，减少网络数据的处理量，提高运行速度。
步骤S102、提取原始网络数据文件中的RTP数据包。
在本实施例中，通过提取原始网络数据文件中的RTP数据包，使用RTP协议的应用程序运行在RTP之上，而执行RTP的程序运行在UDP的上层，目的是为了使用UDP的端口号和检查和。如图4所示，RTP可以看成是传输层的子层。由多媒体应用程序生成的声音和电视数据块被封装在RTP信息包中，每个RTP信息包被封装在UDP消息段中，然后再封装在IP数据包中。
从应用开发人员的角度来看，可把 RTP 执行程序看成是应用程序的一部分，因为开发人员必需把 RTP 集成到应用程序中。在发送端，开发人员必需把执行 RTP 协议的程序写入到创建 RTP 信息包的应用程序中，然后应用程序把 RTP 信息包发送到 UDP 的套接接口（socket interface），如图5所示；同样，在接收端，RTP 信息包通过 UDP 套接接口输入到应用程序，因此开发人员必需把执行 RTP 协议的程序写入到从 RTP 信息包中抽出媒体数据的应用程序。
在本实施例中，在步骤S102中，包括：
分析原始网络数据文件的文件格式，获得文件头、数据报描述和数据包，文件头包括MAGIC码、版本号、时区、数据长度和链路类型，数据报描述包括时间戳和数据报长度，数据包包括ETHER包头、IP包头、UDP包头、RTP包头和RTP数据；
从数据包中提取RTP数据包。
在获取原始网络数据文件之后，对原始网络数据文件的文件格式进行分析，在本实施例中，对原始网络数据文件的tcpdump格式进行分析，经分析可获得文件头、数据报描述和数据包，其中文件头包括MAGIC码、版本号、时区、数据长度和链路类型，数据报描述包括时间戳和数据报长度，数据包包括ETHER包头、IP包头、UDP包头、RTP包头和RTP数据。再从数据包中提取RTP数据包，获得RTP数据包。
从应用程序的媒体数据中提取RTP信息包，具体地，原始网络数据文件为tcpdump格式，该文件格式包括文件头、数据报描述符以及RTP数据包，其中，文件头由MAGIC码+版本码+时区+数据长度+链路类型组成，数据报描述符由时间戳+数据报长度组成，RTP数据包由ETHER包头+IP包头+UDP包头+RTP包头+RTP数据组成，RTP是一种提供端对端传输复位的实时传输协议，用来支持在单目标广播和多目标广播网络服务中传输实时数据，RTP可以看成是传输层的子层，由多媒体应用程序生成的声音和视频数据块被封装在RTP信息包中，每个RTP信息包被封装在UDP消息段中，然后再封装在IP数据包中，其中IP数据包为TCP/IP协议定义的一个在网络上传输的包。
具体地，首先通过tcpdump网络抓包软件，获取包含RTP音视频流的原始网络数据文件，该原始网络数据文件是tcpdump格式的，通过分析该原始网络数据文件的格式如下：
1.文件头（MAGIC码+版本号+时区+数据长度+链路类型）
2.数据报描述符（时间戳+数据报长度）
3.数据包（ETHER包头+IP包头+UDP包头+RTP包头+RTP数据包）
根据上述格式从原始网络数据文件中获取RTP数据包，然后将RTP数据包按照SSRC标识符分流出音频媒体流和视频媒体流，再对音频媒体流和视频媒体流分别进行分析处理，音频媒体流通常为独立的RTP包结构，可以直接解码播放，而视频媒体流通常为H.264编码，需要按照NAL封包格式来解析并解码播放，Jffmpeg 是一个 Java 的多媒体框架插件，可以用来播放大部分格式的音频和视频格式的文件，分析这些音频媒体流或者视频媒体流的播放是否有卡顿、模糊或丢帧情况，可以作为用户网络的带宽、丢包、延迟和抖动等指标参考，从而分析得到用户网络质量。
另一方面，还可以通过Wireshark网络抓包软件获取包含RTP音视频流的原始网络数据文件，该原始网络数据文件是tcpdump格式的，通过分析该原始网络数据文件的格式如下：
1．文件头（MAGIC码+版本号+时区+数据长度+链路类型）
2.数据报描述符（时间戳+数据报长度）
3. 数据包（ETHER包头+IP包头+UDP包头+RTP包头+RTP数据包）
从原始网络数据文件中提取出RTP数据包，然后将RTP数据包按照SSRC标识符分流出音频媒体流和视频媒体流，可以按照时间戳标识符将音频媒体流和视频媒体流进行传输回放，以重现实际生产环境中可能由于网络传输原因产生的异常媒体流，分析这些音频媒体流或者视频媒体流的播放是否有卡顿、模糊或丢帧情况，分析软件处理异常数据时可能存在的问题，帮助分析用户网络质量。
在一些实施例中，在步骤S102中，包括：
提取原始网络数据文件中的数据包；
获取数据包中的RTP数据包。
去除原始网络数据文件中的文件头和数据报描述符，最后得到原始网络数据文件中的数据包，该数据包中包含ETHER包头、IP包头、UDP包头、RTP包头以及RTP数据包。
去掉数据包中的ETHER包头、IP包头、UDP包头以及RTP包头，剩下的就是RTP数据包了，去除了无关信息，减少后期的数据处理量。
步骤S103、根据预设分流规则将RTP数据包进行分流处理得到音频流媒体和视频流媒体。
通过预设分流规则将RTP数据包进行分流处理，具体地，可以将RTP数据包按照同步信源（SSRC）标识符分流出音频媒体流和视频媒体流，同步信源(SSRC)标识符：占32位，用于标识同步信源，该标识符是随机选择的，参加同一视频会议的两个同步信源不能有相同的SSRC，这里的同步信源是指产生媒体流的信源，它通过RTP报头中的一个32位数字SSRC标识符来标识，而不依赖于网络地址，接收者将根据SSRC标识符来区分不同的信源，进行RTP报文的分组，特约信源是指当混合器接收到一个或多个同步信源的RTP报文后，经过混合处理产生一个新的组合RTP报文，并把混合器作为组合RTP报文的SSRC，而将原来所有的SSRC都作为CSRC传送给接收者，使接收者知道组成组合报文的各个SSRC，方便后续对视频和音频分别进行分析。
在本实施例中，在步骤S103中，包括：
按照同步信源SSRC标识符将RTP数据包分流出音频媒体流和视频媒体流。
在RTP协议中， 定义同步信源(SSRC，Synchronization source)为RTP包流的源，用RTP报头中32位数值的SSRC标识符进行标识，使其不依赖于网络地址。通常麦克风，音频接口，摄像头，视频接口的变化，都会导致SSRC的变化；同步信源SSRC用于标识同步信源，该标识符是随机选择的，参加同一视频会议的两个同步信源不能有相同的SSRC，从而能准确将RTP数据包中的音频媒体流和视频媒体流分流出来。
步骤S104、分别播放音频流媒体和视频流媒体，根据音频媒体流和视频媒体流的播放情况，分析网络传输质量。
在所述播放视频流媒体的步骤中，包括：
若视频流媒体为H.264编码，则通过采用NAL封包格式对视频流媒体进行解析并解码播放。
在本实施例中，分别播放音频流媒体和视频流媒体，对音频媒体流和视频媒体流的播放进行分析，音频媒体流为独立RTP包结构，可以直接进行解码播放；视频媒体流通常为H.264编码，RTP包上下文相关，通过按照NAL封包格式来解析并解码播放，H.264是一种视频高压缩技术，同时称为MPEG-4 AVC，或MPEG-4 Part10，在相同的重建图像质量下，H.264比其他视频压缩编码具有更高的压缩比、更好的IP和无线网络信道适应性，高压缩率使图像的数据量减少，给存储和传输带来了方便；然后通过分析这些音频媒体流和视频媒体流的播放是否有卡顿、模糊或者丢帧情况，可以作为用户网络的带宽、丢包、延迟以及抖动等指标参考，根据音频媒体流和视频媒体流的播放情况即可分析得到用户的网络质量。
在步骤S104中，包括：
分析音频流媒体和视频流媒体在播放过程中是否卡顿、模糊或丢帧；
将播放情况的分析结果作为影响网络的带宽、丢包、延迟和抖动的参考指标，分析网络传输质量。
分别对音频流媒体和视频流媒体进行分析，分析音频流媒体和视频流媒体在播放过程中是否卡顿、模糊或丢帧，将播放情况的分析结果作为影响网络的带宽、丢包、延迟和抖动的参考指标，用于分析网络传输质量。也就是说，音频流媒体或视频流媒体在播放过程中是否卡顿、模糊或丢帧，将是否卡顿、模糊或丢帧这种播放情况作为影响网络的带宽、丢包、延迟和抖动的参考指标，进而分析网络传输质量。
在步骤S104之后，所述方法包括：
从数据报描述提取时间戳；
按照时间戳将音频流媒体和视频流媒体进行传输回放；
获取因传输所产生的异常数据；
通过分析软件处理异常数据；
若分析软件处理异常数据存在的问题，则根据问题优化网络传输质量。
从数据报描述提取时间戳，按照时间戳将音频流媒体和视频流媒体进行传输回放，以重现和分析实际生产环境中的可能由于网络传输原因产生的异常数据，通过分析软件处理异常数据，若分析软件处理异常数据存在的问题，则根据问题优化网络传输质量，若分析软件处理异常数据不存在的问题，则不需要优化网络传输质量。
在一些实施例中，在步骤S104之后，所述方法包括：
根据网络传输质量判断是否优化软件应用的媒体传输算法。
在本实施例中，分析用户的网络传输质量，针对音频媒体流和视频媒体流的播放是否存在卡顿、模糊或者丢帧的情况来判断是否需要优化软件应用的媒体传输算法，若需要优化软件应用的媒体传输算法，则针对音频媒体流和视频媒体流的播放是否存在卡顿、模糊或者丢帧的情况来优化软件应用的媒体传输算法，能在软件开发和调试过程中，重现和完全模拟真实网络环境中的音视频流，捕获用户的网络环境变化对媒体传输的影响，进而对软件的媒体传输算法进行优化，提高软件的媒体流传输质量，提高用户的体验。
在一些实施例中，在步骤S104之后，所述方法包括：
将音频媒体流和视频媒体流的播放情况以及分析得到的网络传输质量进行分类和存储。
在本实施例中，对音视频流的播放情况以及分析得到的网络传输质量进行分类和存储，具体地，对音频媒体流的播放情况根据卡顿、杂音以及丢帧作为标签进行分类以及存储，也可以通过根据卡顿、模糊或者丢帧作为标签对视频媒体流进行分类和存储，从而清楚、直接地获得音视频流的播放情况，方便后期针对音视频流的播放情况进行网络优化操作；网络优化的内容包括：优化媒体传输算法中的压缩率，从而减少网络传输的数据量，提高网络传输速率，减少卡顿现象；根据参数优化，让接入、切换等参数最优化，减少音频媒体流和视频媒体流播放的卡顿、模糊或者丢帧的情况，当然，还可以通过硬件检测以更换有问题的硬件、修改基站邻集以使切换合理，减少切换掉话来优化网络。
综上所述，获取实时视频应用中原始网络数据文件，提取原始网络数据文件中的RTP数据包，通过RTP数据包中分流得到音频流媒体和视频流媒体，根据音频媒体流和视频媒体流的播放情况分析得到网络传输质量，完全模拟和重现真实网络环境中的音视频流，帮助分析用户网络质量，旨在解决现有的实时视频应用，难于分析用户的网络质量的问题。
如图2所示，本申请实施例提出一种网络传输质量分析装置，装置1包括获取模块11、提取模块12、处理模块13和分析模块14。
获取模块11，用于获取实时视频应用中包含RTP音视频流的原始网络数据文件。
在本实施例中，获取实时视频应用中原始网络数据文件，其中实时视频应用采用RTP公共协议作为媒体流传输协议，原始网络数据文件中包含RTP封装的音视频流，可靠传输协议RTP是Reliable Transport Protocol的英文缩写，是用来管理EIGRP数据包的发送和接收的协议。“可靠的发送”是指发送是有保障的，这种保障依赖Cisco公司私有的算法来实现，这个私有的算法被称为“可靠组播（reliable multicast）”，发送路由器将更新信息发送到组播IP地址224.0.0.10，每一个接收可靠组播数据包的邻居都会发送一个单播的确认数据包，如果EIGRP没有从某个邻居那里得到应答，它将使用单播来重发同样的数据，如果在16次单播尝试后，它仍然没有应答，则此邻居将被宣告消失。
虽然名称中包含“可靠”，但RTP并不仅仅提供可靠传输，它也提供不可靠传输。当使用可靠传输时，RTP要求对方发回ACK确认；当使用不可靠传输时，RTP不要求ACK。
在本实施例中，获取模块11包括：
第一子获取模块，用于通过网络抓包软件获取实时视频应用软件传输的原始网络数据文件，其中原始网络数据文件中包括RTP音视频流。
采用网络抓包软件进行抓包，具体地，通过tcpdump可以将网络中传送的数据包完全截获下来提供分析，它支持针对网络层、协议、主机、网络或端口的过滤，并提供and、or、not等逻辑语句来帮助你去掉无用的信息；当然，还可以通过Wireshark来进行抓包，Wireshark（前称Ethereal）是一个网络封包分析软件，网络封包分析软件的功能是撷取网络封包，并尽可能显示出最为详细的网络封包资料。Wireshark使用WinPCAP作为接口，直接与网卡进行数据报文交换，从而获取实时视频应用软件中的原始网络数据文件。
具体地，获取模块11包括：
tcpdump格式文件获取模块，用于通过tcpdump或者Wireshark网络抓包软件获取实时视频应用中包含RTP音视频流的原始网络数据文件，原始网络数据文件的文件格式为tcpdump格式。
使用tcpdump或者Wireshark网络抓包软件获取原始网络数据文件，得到的原始网络数据文件的文件格式为tcpdump格式，原始网络数据文件为tcpdump格式，便于后续对原始网络数据文件进行分析。
在本实施例中，装置1包括：
筛选模块，用于筛选出实时视频应用产生的网络数据。
通过先对所有网络数据进行分析，从而筛选出实时视频应用产生的网络数据，再对实时视频应用产生的网络数据进行分析，以获取包含RTP音视频流的原始网络数据文件，然后从原始网络数据文件中获取RTP数据包，将RTP数据包按照SSRC标识符分流出音频媒体流和视频媒体流，再对音频媒体流进行分析处理，音频媒体流通常为独立的RTP包结构，可以直接解码播放，具体地，可以通过Adobe Audition应用软件播放分析音频媒体流和视频媒体流，从而分析音频媒体流和视频媒体流的播放情况，分析音频媒体流和视频媒体流的播放是否有卡顿、模糊或丢帧情况，从而分析得到用户网络质量。先对网络数据进行筛选，从而获取实时视频应用产生的网络数据，减少网络数据的处理量，提高运行速度。
提取模块12，用于提取原始网络数据文件中的RTP数据包。
在本实施例中，通过提取原始网络数据文件中的RTP数据包，使用RTP协议的应用程序运行在RTP之上，而执行RTP的程序运行在UDP的上层，目的是为了使用UDP的端口号和检查和。如图4所示，RTP可以看成是传输层的子层。由多媒体应用程序生成的声音和电视数据块被封装在RTP信息包中，每个RTP信息包被封装在UDP消息段中，然后再封装在IP数据包中。
从应用开发人员的角度来看，可把 RTP 执行程序看成是应用程序的一部分，因为开发人员必需把 RTP 集成到应用程序中。在发送端，开发人员必需把执行 RTP 协议的程序写入到创建 RTP 信息包的应用程序中，然后应用程序把 RTP 信息包发送到 UDP 的套接接口（socket interface），如图5所示；同样，在接收端，RTP 信息包通过 UDP 套接接口输入到应用程序，因此开发人员必需把执行 RTP 协议的程序写入到从 RTP 信息包中抽出媒体数据的应用程序。
在本实施例中，提取模块12包括：
文件格式分析模块，用于分析原始网络数据文件的文件格式，获得文件头、数据报描述和数据包，文件头包括MAGIC码、版本号、时区、数据长度和链路类型，数据报描述包括时间戳和数据报长度，数据包包括ETHER包头、IP包头、UDP包头、RTP包头和RTP数据；
RTP数据包提取模块，用于从数据包中提取RTP数据包。
在获取原始网络数据文件之后，对原始网络数据文件的文件格式进行分析，在本实施例中，对原始网络数据文件的tcpdump格式进行分析，经分析可获得文件头、数据报描述和数据包，其中文件头包括MAGIC码、版本号、时区、数据长度和链路类型，数据报描述包括时间戳和数据报长度，数据包包括ETHER包头、IP包头、UDP包头、RTP包头和RTP数据。再从数据包中提取RTP数据包，获得RTP数据包。
从应用程序的媒体数据中提取RTP信息包，具体地，原始网络数据文件为tcpdump格式，该文件格式包括文件头、数据报描述符以及RTP数据包，其中，文件头由MAGIC码+版本码+时区+数据长度+链路类型组成，数据报描述符由时间戳+数据报长度组成，RTP数据包由ETHER包头+IP包头+UDP包头+RTP包头+RTP数据组成，RTP是一种提供端对端传输复位的实时传输协议，用来支持在单目标广播和多目标广播网络服务中传输实时数据，RTP可以看成是传输层的子层，由多媒体应用程序生成的声音和视频数据块被封装在RTP信息包中，每个RTP信息包被封装在UDP消息段中，然后再封装在IP数据包中，其中IP数据包为TCP/IP协议定义的一个在网络上传输的包。
具体地，首先通过tcpdump网络抓包软件，获取包含RTP音视频流的原始网络数据文件，该原始网络数据文件是tcpdump格式的，通过分析该原始网络数据文件的格式如下：
1.文件头（MAGIC码+版本号+时区+数据长度+链路类型）
2.数据报描述符（时间戳+数据报长度）
3.数据包（ETHER包头+IP包头+UDP包头+RTP包头+RTP数据包）
根据上述格式从原始网络数据文件中获取RTP数据包，然后将RTP数据包按照SSRC标识符分流出音频媒体流和视频媒体流，再对音频媒体流和视频媒体流分别进行分析处理，音频媒体流通常为独立的RTP包结构，可以直接解码播放，而视频媒体流通常为H.264编码，需要按照NAL封包格式来解析并解码播放，Jffmpeg 是一个 Java 的多媒体框架插件，可以用来播放大部分格式的音频和视频格式的文件，分析这些音频媒体流或者视频媒体流的播放是否有卡顿、模糊或丢帧情况，可以作为用户网络的带宽、丢包、延迟和抖动等指标参考，从而分析得到用户网络质量。
另一方面，还可以通过Wireshark网络抓包软件获取包含RTP音视频流的原始网络数据文件，该原始网络数据文件是tcpdump格式的，通过分析该原始网络数据文件的格式如下：
1．文件头（MAGIC码+版本号+时区+数据长度+链路类型）
2.数据报描述符（时间戳+数据报长度）
3. 数据包（ETHER包头+IP包头+UDP包头+RTP包头+RTP数据包）
从原始网络数据文件中提取出RTP数据包，然后将RTP数据包按照SSRC标识符分流出音频媒体流和视频媒体流，可以按照时间戳标识符将音频媒体流和视频媒体流进行传输回放，以重现实际生产环境中可能由于网络传输原因产生的异常媒体流，分析这些音频媒体流或者视频媒体流的播放是否有卡顿、模糊或丢帧情况，分析软件处理异常数据时可能存在的问题，帮助分析用户网络质量。
在一些实施例中，提取模块12包括：
第一子提取模块，用于提取原始网络数据文件中的数据包；
第一子RTP数据获取模块，用于获取数据包中的RTP数据包。
去除原始网络数据文件中的文件头和数据报描述符，最后得到原始网络数据文件中的数据包，该数据包中包含ETHER包头、IP包头、UDP包头、RTP包头以及RTP数据包。
去掉数据包中的ETHER包头、IP包头、UDP包头以及RTP包头，剩下的就是RTP数据包了，去除了无关信息，减少后期的数据处理量。
处理模块13，用于根据预设分流规则将RTP数据包进行分流处理得到音频流媒体和视频流媒体。
通过预设分流规则将RTP数据包进行分流处理，具体地，可以将RTP数据包按照同步信源（SSRC）标识符分流出音频媒体流和视频媒体流，同步信源(SSRC)标识符：占32位，用于标识同步信源，该标识符是随机选择的，参加同一视频会议的两个同步信源不能有相同的SSRC，这里的同步信源是指产生媒体流的信源，它通过RTP报头中的一个32位数字SSRC标识符来标识，而不依赖于网络地址，接收者将根据SSRC标识符来区分不同的信源，进行RTP报文的分组，特约信源是指当混合器接收到一个或多个同步信源的RTP报文后，经过混合处理产生一个新的组合RTP报文，并把混合器作为组合RTP报文的SSRC，而将原来所有的SSRC都作为CSRC传送给接收者，使接收者知道组成组合报文的各个SSRC，方便后续对视频和音频分别进行分析。
在本实施例中，处理模块13包括：
第一子处理模块，用于按照同步信源SSRC标识符将RTP数据包分流出音频媒体流和视频媒体流。
在RTP协议中， 定义同步信源(SSRC，Synchronization source)为RTP包流的源，用RTP报头中32位数值的SSRC标识符进行标识，使其不依赖于网络地址。通常麦克风，音频接口，摄像头，视频接口的变化，都会导致SSRC的变化；同步信源SSRC用于标识同步信源，该标识符是随机选择的，参加同一视频会议的两个同步信源不能有相同的SSRC，从而能准确将RTP数据包中的音频媒体流和视频媒体流分流出来。
分析模块14，用于分别播放音频流媒体和视频流媒体，根据音频媒体流和视频媒体流的播放情况，分析网络传输质量。
分析模块14包括：
第一分析模块，用于若视频流媒体为H.264编码，则通过采用NAL封包格式对视频流媒体进行解析并解码播放。
在本实施例中，分别播放音频流媒体和视频流媒体，对音频媒体流和视频媒体流的播放进行分析，音频媒体流为独立RTP包结构，可以直接进行解码播放；视频媒体流通常为H.264编码，RTP包上下文相关，通过按照NAL封包格式来解析并解码播放，H.264是一种视频高压缩技术，同时称为MPEG-4 AVC，或MPEG-4 Part10，在相同的重建图像质量下，H.264比其他视频压缩编码具有更高的压缩比、更好的IP和无线网络信道适应性，高压缩率使图像的数据量减少，给存储和传输带来了方便；然后通过分析这些音频媒体流和视频媒体流的播放是否有卡顿、模糊或者丢帧情况，可以作为用户网络的带宽、丢包、延迟以及抖动等指标参考，根据音频媒体流和视频媒体流的播放情况即可分析得到用户的网络质量。
分析模块14包括：
第二分析模块，用于分析音频流媒体和视频流媒体在播放过程中是否卡顿、模糊或丢帧；
第三分析模块，用于将播放情况的分析结果作为影响网络的带宽、丢包、延迟和抖动的参考指标，分析网络传输质量。
分别对音频流媒体和视频流媒体进行分析，分析音频流媒体和视频流媒体在播放过程中是否卡顿、模糊或丢帧，将播放情况的分析结果作为影响网络的带宽、丢包、延迟和抖动的参考指标，用于分析网络传输质量。也就是说，音频流媒体或视频流媒体在播放过程中是否卡顿、模糊或丢帧，将是否卡顿、模糊或丢帧这种播放情况作为影响网络的带宽、丢包、延迟和抖动的参考指标，进而分析网络传输质量。
装置1包括：
第一提取模块，用于从数据报描述提取时间戳；
第一传输模块，用于按照时间戳将音频流媒体和视频流媒体进行传输回放；
第一获取模块，用于获取因传输所产生的异常数据；
第一处理模块，用于通过分析软件处理异常数据；
第一优化模块，用于若分析软件处理异常数据存在的问题，则根据问题优化网络传输质量。
从数据报描述提取时间戳，按照时间戳将音频流媒体和视频流媒体进行传输回放，以重现和分析实际生产环境中的可能由于网络传输原因产生的异常数据，通过分析软件处理异常数据，若分析软件处理异常数据存在的问题，则根据问题优化网络传输质量，若分析软件处理异常数据不存在的问题，则不需要优化网络传输质量。
在一些实施例中，装置1包括：
优化判断模块，用于根据网络传输质量判断是否优化软件应用的媒体传输算法。
在本实施例中，分析用户的网络传输质量，针对音频媒体流和视频媒体流的播放是否存在卡顿、模糊或者丢帧的情况来判断是否需要优化软件应用的媒体传输算法，若需要优化软件应用的媒体传输算法，则针对音频媒体流和视频媒体流的播放是否存在卡顿、模糊或者丢帧的情况来优化软件应用的媒体传输算法，能在软件开发和调试过程中，重现和完全模拟真实网络环境中的音视频流，捕获用户的网络环境变化对媒体传输的影响，进而对软件的媒体传输算法进行优化，提高软件的媒体流传输质量，提高用户的体验。
在一些实施例中，装置1包括：
存储模块，用于将音频媒体流和视频媒体流的播放情况以及分析得到的网络传输质量进行分类和存储。
在本实施例中，对音视频流的播放情况以及分析得到的网络传输质量进行分类和存储，具体地，对音频媒体流的播放情况根据卡顿、杂音以及丢帧作为标签进行分类以及存储，也可以通过根据卡顿、模糊或者丢帧作为标签对视频媒体流进行分类和存储，从而清楚、直接地获得音视频流的播放情况，方便后期针对音视频流的播放情况进行网络优化操作；网络优化的内容包括：优化媒体传输算法中的压缩率，从而减少网络传输的数据量，提高网络传输速率，减少卡顿现象；根据参数优化，让接入、切换等参数最优化，减少音频媒体流和视频媒体流播放的卡顿、模糊或者丢帧的情况，当然，还可以通过硬件检测以更换有问题的硬件、修改基站邻集以使切换合理，减少切换掉话来优化网络。
综上所述，获取实时视频应用中原始网络数据文件，提取原始网络数据文件中的RTP数据包，通过RTP数据包中分流得到音频流媒体和视频流媒体，根据音频媒体流和视频媒体流的播放情况分析得到网络传输质量，完全模拟和重现真实网络环境中的音视频流，帮助分析用户网络质量，旨在解决现有的实时视频应用，难于分析用户的网络质量的问题。
如图3所示，本申请实施例中还提供一种计算机设备，该计算机设备可以是服务器，其内部结构可以如图3所示。该计算机设备包括通过系统总线连接的处理器、存储器、网络接口和数据库。其中，该计算机设计的处理器用于提供计算和控制能力。该计算机设备的存储器包括非易失性存储介质、内存储器。该非易失性存储介质存储有操作系统、计算机可读指令和数据库。该内存器为非易失性存储介质中的操作系统和计算机可读指令的运行提供环境。该计算机设备的数据库用于存储网络传输质量分析方法的模型等数据。该计算机设备的网络接口用于与外部的终端通过网络连接通信。该计算机可读指令被处理器执行时以实现一种网络传输质量分析方法。
上述处理器执行上述网络传输质量分析方法的步骤：获取实时视频应用中包含RTP音视频流的原始网络数据文件；提取所述原始网络数据文件中的RTP数据包；根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体；分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量。
在一个实施例中，上述分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析得到网络传输质量的步骤中，包括：
分析所述音频流媒体和所述视频流媒体在播放过程中是否卡顿、模糊或丢帧；
将所述播放情况的分析结果作为影响所述网络的带宽、丢包、延迟和抖动的参考指标，分析所述网络传输质量。
在一个实施例中，上述播放所述视频流媒体的步骤中，包括：
若所述视频流媒体为H.264编码，则通过采用NAL封包格式对所述视频流媒体进行解析并解码播放。
在一个实施例中，上述根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体的步骤中，包括：
按照同步信源SSRC标识符将所述RTP数据包分流出所述音频媒体流和所述视频媒体流。
在一个实施例中，上述提取所述原始网络数据文件中的RTP数据包的步骤中，包括：
分析所述原始网络数据文件的文件格式，获得文件头、数据报描述和数据包，所述文件头包括MAGIC码、版本号、时区、数据长度和链路类型，所述数据报描述包括时间戳和数据报长度，所述数据包包括ETHER包头、IP包头、UDP包头、RTP包头和RTP数据；
从所述数据包中提取所述RTP数据包。
在一个实施例中，上述分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量的步骤之后，包括：
从所述数据报描述提取所述时间戳；
按照所述时间戳将所述音频流媒体和所述视频流媒体进行传输回放；
获取因传输所产生的异常数据；
通过分析软件处理所述异常数据；
若所述分析软件处理所述异常数据存在的问题，则根据所述问题优化所述网络传输质量。
在一个实施例中，上述获取实时视频应用中包含RTP音视频流的原始网络数据文件的步骤中，包括：
通过tcpdump或者Wireshark网络抓包软件获取实时视频应用中包含RTP音视频流的原始网络数据文件，所述原始网络数据文件的文件格式为tcpdump格式。
本领域技术人员可以理解，图3中示出的结构，仅仅是与本申请方案相关的部分结构的框图，并不构成对本申请方案所应用于其上的计算机设备的限定。
本申请实施例的计算机设备，获取实时视频应用中原始网络数据文件，提取原始网络数据文件中的RTP数据包，通过RTP数据包中分流得到音频流媒体和视频流媒体，根据音频媒体流和视频媒体流的播放情况分析得到网络传输质量，完全模拟和重现真实网络环境中的音视频流，帮助分析用户网络质量，旨在解决现有的实时视频应用，难于分析用户的网络质量的问题。
本申请一实施例还提供一种计算机非易失性可读存储介质，其上存储有计算机可读指令，计算机可读指令被处理器执行时实现如上述各方法的实施例的流程。
以上所述仅为本申请的较佳实施例而已，并不用以限制本申请，凡在本申请的精神和原则之内所作的任何修改、等同替换和改进等，均应包含在本申请的保护范围之内。

Claims (20)

  1. 一种网络传输质量分析方法，其特征在于，所述方法包括：
    获取实时视频应用中包含RTP音视频流的原始网络数据文件；
    提取所述原始网络数据文件中的RTP数据包；
    根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体；
    分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量。
  2. 根据权利要求1所述的网络传输质量分析方法，其特征在于，在所述分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析得到网络传输质量的步骤中，包括：
    分析所述音频流媒体和所述视频流媒体在播放过程中是否卡顿、模糊或丢帧；
    将所述播放情况的分析结果作为影响所述网络的带宽、丢包、延迟和抖动的参考指标，分析所述网络传输质量。
  3. 根据权利要求1所述的网络传输质量分析方法，其特征在于，在所述播放所述视频流媒体的步骤中，包括：
    若所述视频流媒体为H.264编码，则通过采用NAL封包格式对所述视频流媒体进行解析并解码播放。
  4. 根据权利要求1所述的网络传输质量分析方法，其特征在于，在所述根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体的步骤中，包括：
    按照同步信源SSRC标识符将所述RTP数据包分流出所述音频媒体流和所述视频媒体流。
  5. 根据权利要求1所述的网络传输质量分析方法，其特征在于，在所述提取所述原始网络数据文件中的RTP数据包的步骤中，包括：
    分析所述原始网络数据文件的文件格式，获得文件头、数据报描述和数据包，所述文件头包括MAGIC码、版本号、时区、数据长度和链路类型，所述数据报描述包括时间戳和数据报长度，所述数据包包括ETHER包头、IP包头、UDP包头、RTP包头和RTP数据；
    从所述数据包中提取所述RTP数据包。
  6. 根据权利要求5所述的网络传输质量分析方法，其特征在于，在所述分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量的步骤之后，所述方法包括：
    从所述数据报描述提取所述时间戳；
    按照所述时间戳将所述音频流媒体和所述视频流媒体进行传输回放；
    获取因传输所产生的异常数据；
    通过分析软件处理所述异常数据；
    若所述分析软件处理所述异常数据存在的问题，则根据所述问题优化所述网络传输质量。
  7. 根据权利要求1所述的网络传输质量分析方法，其特征在于，在所述获取实时视频应用中包含RTP音视频流的原始网络数据文件的步骤中，包括：
    通过tcpdump或者Wireshark网络抓包软件获取实时视频应用中包含RTP音视频流的原始网络数据文件，所述原始网络数据文件的文件格式为tcpdump格式。
  8. 一种网络传输质量分析装置，其特征在于，所述装置包括：
    获取模块，用于获取实时视频应用中包含RTP音视频流的原始网络数据文件；
    提取模块，用于提取所述原始网络数据文件中的RTP数据包；
    处理模块，用于根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体；
    分析模块，用于分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量。
  9. 根据权利要求8所述的网络传输质量分析装置，其特征在于，所述分析模块包括：
    第二分析模块，用于分析所述音频流媒体和所述视频流媒体在播放过程中是否卡顿、模糊或丢帧；
    第三分析模块，用于将所述播放情况的分析结果作为影响所述网络的带宽、丢包、延迟和抖动的参考指标，分析所述网络传输质量。
  10. 根据权利要求8所述的网络传输质量分析装置，其特征在于，所述分析模块包括：
    第一分析模块，用于若所述视频流媒体为H.264编码，则通过采用NAL封包格式对所述视频流媒体进行解析并解码播放。
  11. 根据权利要求8所述的网络传输质量分析装置，其特征在于，所述处理模块包括：
    第一子处理模块，用于按照同步信源SSRC标识符将所述RTP数据包分流出所述音频媒体流和所述视频媒体流。
  12. 根据权利要求8所述的网络传输质量分析装置，其特征在于，所述提取模块包括：
    文件格式分析模块，用于分析所述原始网络数据文件的文件格式，获得文件头、数据报描述和数据包，所述文件头包括MAGIC码、版本号、时区、数据长度和链路类型，所述数据报描述包括时间戳和数据报长度，所述数据包包括ETHER包头、IP包头、UDP包头、RTP包头和RTP数据；
    RTP数据包提取模块，用于从所述数据包中提取所述RTP数据包。
  13. 根据权利要求12所述的网络传输质量分析装置，其特征在于，所述装置包括：
    第一提取模块，用于从所述数据报描述提取所述时间戳；
    第一传输模块，用于按照所述时间戳将所述音频流媒体和所述视频流媒体进行传输回放；
    第一获取模块，用于获取因传输所产生的异常数据；
    第一处理模块，用于通过分析软件处理所述异常数据；
    第一优化模块，用于若所述分析软件处理所述异常数据存在的问题，则根据所述问题优化所述网络传输质量。
  14. 根据权利要求8所述的网络传输质量分析装置，其特征在于，所述获取模块包括：
    tcpdump格式文件获取模块，用于通过tcpdump或者Wireshark网络抓包软件获取实时视频应用中包含RTP音视频流的原始网络数据文件，所述原始网络数据文件的文件格式为tcpdump格式。
  15. 一种计算机设备，包括存储器和处理器，所述存储器存储有计算机可读指令，其特征在于，所述处理器执行所述计算机可读指令时实现网络传输质量分析方法，所述网络传输质量分析方法包括：
    获取实时视频应用中包含RTP音视频流的原始网络数据文件；
    提取所述原始网络数据文件中的RTP数据包；
    根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体；
    分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量。
  16. 根据权利要求15所述的计算机设备，其特征在于，在所述分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析得到网络传输质量的步骤中，包括：
    分析所述音频流媒体和所述视频流媒体在播放过程中是否卡顿、模糊或丢帧；
    将所述播放情况的分析结果作为影响所述网络的带宽、丢包、延迟和抖动的参考指标，分析所述网络传输质量。
  17. 根据权利要求15所述的计算机设备，其特征在于，在所述播放所述视频流媒体的步骤中，包括：
    若所述视频流媒体为H.264编码，则通过采用NAL封包格式对所述视频流媒体进行解析并解码播放。
  18. 根据权利要求15所述的计算机设备，其特征在于，在所述根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体的步骤中，包括：
    按照同步信源SSRC标识符将所述RTP数据包分流出所述音频媒体流和所述视频媒体流。
  19. 一种计算机非易失性可读存储介质，其上存储有计算机可读指令，其特征在于，所述计算机可读指令被处理器执行时实现网络传输质量分析方法，所述网络传输质量分析方法包括：
    获取实时视频应用中包含RTP音视频流的原始网络数据文件；
    提取所述原始网络数据文件中的RTP数据包；
    根据预设分流规则将所述RTP数据包进行分流处理得到音频流媒体和视频流媒体；
    分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析网络传输质量。
  20. 根据权利要求19所述的计算机非易失性可读存储介质，其特征在于，在所述分别播放所述音频流媒体和所述视频流媒体，根据所述音频媒体流和所述视频媒体流的播放情况，分析得到网络传输质量的步骤中，包括：
    分析所述音频流媒体和所述视频流媒体在播放过程中是否卡顿、模糊或丢帧；
    将所述播放情况的分析结果作为影响所述网络的带宽、丢包、延迟和抖动的参考指标，分析所述网络传输质量。
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