基于AVS标准的音视频通信系统的设计与实现

发布时间:2018-04-23 02:10

  本文选题:AVS音视频通信 + RTP/RTCP协议 ; 参考:《电子科技大学》2013年硕士论文


【摘要】:随着网络和多媒体技术的发展,IP(Internet Protocol)网络上的音视频通信技术(简称VoIP:Voice over Internet Protocol)被广泛应用到即时通信、视频会议等产品中。音视频通信技术涵盖了音视频编码、流媒体传输、通信控制等诸多方面的内容。AVS(Audio Video Standard)是我国颁布具备独立自主知识产权的第二代信源编码标准。本文主要讨论基于AVS视频编码标准的音视频通信系统的框架设计及客户端、服务器端的功能设计和通讯流程的实现。首先,论文介绍了流媒体传输和同步控制的相关理论,重点讨论了实时传输协议(RTP:Real-time Tranport Protocol)和实时流传输控制协议(RTCP:Real-time Transport Control Protocol)、实时流传输协议(RTSP:Real-time Transport Steaming Protocol)的实现。通过利用RTCP协议提供的控制信息,设计了音频媒体和视频媒体之间的同步控制算法。在实现RTP/RTCP协议的基础上,不但很好地控制了音视频数据的实时传输,而且达到了音视频媒体的流内同步和流间同步的控制效果。其次,论文针对视频会议等方面的需求研究了AVSIM(Audio Video Standard Intant Messager)视频通信功能的架构,实现了视频通信控制协议的定义,并通过统一建模语言(UML:Unified Modeling Language)剖析了会议模块的功能设计。最后,论文讨论了NAT(Network Address Translation)穿越的分类以及NAT技术的原理以及实现,比较了STUN(Simple Traversal of UDP over NATs)穿越、TURN(Traversal Using Relays around NAT)穿越、UPnP(Universal Plug and Play)穿越、TCP/UDP (Transmission Control Protocol/User Datagram Protocol)打洞穿越、HTTP(Hypertext transfer protocol) Tunnel穿越以及客户端NAT模块的工作流程。同时阐述了整体系统的软件实现和测试分析,并对当前的工作进行了总结和展望。
[Abstract]:With the development of network and multimedia technology, audio and video communication technology (VoIP:Voice over Internet Protocol) on IP (Internet Protocol) network is widely used in instant communications, video conferencing and other products. Audio and video communication technology covers audio video coding, streaming media transmission, communication control and many other aspects of.AVS (Audio). Video Standard) is the second generation of source code standard which has independent intellectual property rights in China. This paper mainly discusses the frame design of audio and video communication system based on AVS video coding standard, the client, the function design of the server and the realization of the communication process. First, the phase of streaming media transmission and synchronous control is introduced in this paper. According to the theory, the real time transmission protocol (RTP:Real-time Tranport Protocol) and the real-time flow transmission control protocol (RTCP:Real-time Transport Control Protocol), the realization of the real time flow transmission protocol (RTSP:Real-time Transport Steaming Protocol) are implemented. The audio media and video are designed by using the control information provided by the RTCP protocol. The synchronization control algorithm between the media, based on the implementation of the RTP/RTCP protocol, not only controls the real-time transmission of audio and video data, but also achieves the control effect of intra streaming synchronization and inter flow synchronization in audio and video media. Secondly, the paper studies AVSIM (Audio Video Standard Intant Messag) for video conferencing and other aspects. Er) the architecture of video communication function, the definition of the video communication control protocol is realized, and the function design of the conference module is analyzed through the unified modeling language (UML:Unified Modeling Language). Finally, the paper discusses the classification of the NAT (Network Address Translation) crossing and the principle and implementation of NAT technology, and compares STUN (Simple Tra). Versal of UDP over NATs) traversing, TURN (Traversal Using Relays around NAT) crossing. The software implementation and test analysis are summarized, and the current work is summarized and prospected.

【学位授予单位】:电子科技大学
【学位级别】:硕士
【学位授予年份】:2013
【分类号】:TN912.3

【参考文献】

相关博士学位论文 前2条

1 孙勇;下一代网络视频传输关键技术研究[D];北京邮电大学;2008年

2 邓钱华;无线网络视频传输的质量控制[D];北京邮电大学;2009年

相关硕士学位论文 前3条

1 朱卫锋;H.323协议栈中语音编码和媒体同步的设计与实现[D];华中科技大学;2004年

2 孙小平;嵌入式IPv6实时通信技术的研究[D];西安电子科技大学;2007年

3 孙君文;基于STUNT协议的内网穿越解决方案的设计和实现[D];华东师范大学;2009年



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