基于WebRTC和Jingle的融合通信研究与实现

发布时间：2018-11-05 17:36

【摘要】：目前，随着互联网越来越广泛，各种应用增长迅速，而视频通话的需求一直在不断增长。如今，已经不是一种终端统一整个通信，市面上各种终端的出现，对于人们来说融合通信的需求越来越紧迫。而浏览器的便捷性，使基于浏览器的应用也越来越普遍，随着WebRTC技术的出现，无须插件支持的基于浏览器的视频通话成为了现实，而目前在全球浏览器厂商中越来越多的厂商加入到WebRTC技术的大潮中。另一方面，随着开源的普及，支持XMPP协议的即时通信也成为众多厂商的选择。对于Jingle协议，作为XMPP协议的扩展协议，由于支持P2P，，以及语音视频也逐渐展现出潜力和未来发展的趋势，因此本文基于标准在研究WebRTC的视频通话技术以及Jingle对语音视频的支持基础上，继续研究了两种异构网络的实现的可能性，并通过采取融合网关的形式来将两种异构网络联系起来。本文通过研究实现的可能性，提出来两种融合方案，并设计了两者融合的整体架构，对信令网关和媒体网关进行了设计，以及信令交互流程，信息流等进行了详细设计。之后对信令网关和媒体网关进行了实现。信令网关主要实现协议转换，媒体网关主要实现VP8和H.264的RTP打包和解包，以及VP8和H.264的编解码转换。最后，通过对融合网关的性能和功能进行了测试分析。功能性测试上，设置测试点进行功能性测试，测试均通过，并对画面采用主观性评价测试方法进行了功能测试，画面表现正常。性能测试分为丢包率测试和延迟测试。丢包率测试方面，客户端发送到融合网关的丢包率在1%以下。对信令包进行分析，测试丢包率为0%，编解码延迟测试上，从H.264到VP8编解码耗时33.02ms，从VP8到H.264的编解码延迟在25.04ms。测试结果满足了实时的要求。说明了融合网关设计的合理性和可行性。从标准上实现了WebRTC和Jingle之间的互通。
[Abstract]:At present, with the increasing popularity of the Internet, applications are growing rapidly, and the demand for video calls has been growing. Nowadays, it is not a kind of terminal to unify the whole communication. The appearance of all kinds of terminals in the market makes it more and more urgent for people to integrate communication. The convenience of browser makes browser-based applications more and more common. With the emergence of WebRTC technology, browser-based video calls without plug-in support become a reality. At present in the global browser manufacturers more and more manufacturers join the tide of WebRTC technology. On the other hand, with the popularity of open source, instant messaging supporting XMPP protocol has become the choice of many vendors. For the Jingle protocol, as an extension of the XMPP protocol, because of its support for P2P, as well as the voice and video, it gradually shows the potential and the future development trend. Therefore, based on the standard, this paper continues to study the possibility of implementing two heterogeneous networks on the basis of studying the video calling technology of WebRTC and the support of voice and video by Jingle. The two heterogeneous networks are connected by adopting the form of fusion gateway. By studying the possibility of implementation, this paper proposes two fusion schemes, designs the whole architecture of the fusion, designs the signaling gateway and the media gateway, and designs the signaling interaction flow and information flow in detail. Then the signaling gateway and the media gateway are implemented. The signaling gateway mainly implements the protocol conversion, the media gateway mainly implements the RTP packaging and unpacking of VP8 and H.264, and the codec conversion between VP8 and H.264. Finally, the performance and function of the fusion gateway are tested and analyzed. In the functional test, the test points were set up for functional testing, and the tests passed, and the subjective evaluation test method was used to test the function of the screen, and the performance of the picture was normal. Performance testing is divided into packet loss rate test and delay test. In terms of packet loss rate test, the rate of packet loss sent by client to fusion gateway is less than 1%. The analysis of signaling packets shows that the packet loss rate is 0. In the coding and decoding delay test, the coding and decoding time from H. 264 to VP8 is 33.02 Ms, and the coding and decoding delay from VP8 to H. 264 is 25.04 ms. The test results meet the real-time requirements. The rationality and feasibility of the design of fusion gateway are explained. The interworking between WebRTC and Jingle is realized by standard.
【学位授予单位】：华南理工大学
【学位级别】：硕士
【学位授予年份】：2014
【分类号】：TN919.8

【参考文献】