基于WebRTC和Jingle的融合通信研究與實現(xiàn)
發(fā)布時間：2018-11-05 17:36
【摘要】：目前，隨著互聯(lián)網(wǎng)越來越廣泛，各種應(yīng)用增長迅速，而視頻通話的需求一直在不斷增長。如今，已經(jīng)不是一種終端統(tǒng)一整個通信，市面上各種終端的出現(xiàn)，對于人們來說融合通信的需求越來越緊迫。而瀏覽器的便捷性，使基于瀏覽器的應(yīng)用也越來越普遍，隨著WebRTC技術(shù)的出現(xiàn)，無須插件支持的基于瀏覽器的視頻通話成為了現(xiàn)實，而目前在全球瀏覽器廠商中越來越多的廠商加入到WebRTC技術(shù)的大潮中。 另一方面，隨著開源的普及，支持XMPP協(xié)議的即時通信也成為眾多廠商的選擇。對于Jingle協(xié)議，作為XMPP協(xié)議的擴展協(xié)議，由于支持P2P，，以及語音視頻也逐漸展現(xiàn)出潛力和未來發(fā)展的趨勢，因此本文基于標(biāo)準(zhǔn)在研究WebRTC的視頻通話技術(shù)以及Jingle對語音視頻的支持基礎(chǔ)上，繼續(xù)研究了兩種異構(gòu)網(wǎng)絡(luò)的實現(xiàn)的可能性，并通過采取融合網(wǎng)關(guān)的形式來將兩種異構(gòu)網(wǎng)絡(luò)聯(lián)系起來。 本文通過研究實現(xiàn)的可能性，提出來兩種融合方案，并設(shè)計了兩者融合的整體架構(gòu)，對信令網(wǎng)關(guān)和媒體網(wǎng)關(guān)進行了設(shè)計，以及信令交互流程，信息流等進行了詳細設(shè)計。之后對信令網(wǎng)關(guān)和媒體網(wǎng)關(guān)進行了實現(xiàn)。信令網(wǎng)關(guān)主要實現(xiàn)協(xié)議轉(zhuǎn)換，媒體網(wǎng)關(guān)主要實現(xiàn)VP8和H.264的RTP打包和解包，以及VP8和H.264的編解碼轉(zhuǎn)換。 最后，通過對融合網(wǎng)關(guān)的性能和功能進行了測試分析。功能性測試上，設(shè)置測試點進行功能性測試，測試均通過，并對畫面采用主觀性評價測試方法進行了功能測試，畫面表現(xiàn)正常。性能測試分為丟包率測試和延遲測試。丟包率測試方面，客戶端發(fā)送到融合網(wǎng)關(guān)的丟包率在1%以下。對信令包進行分析，測試丟包率為0%，編解碼延遲測試上，從H.264到VP8編解碼耗時33.02ms，從VP8到H.264的編解碼延遲在25.04ms。測試結(jié)果滿足了實時的要求。說明了融合網(wǎng)關(guān)設(shè)計的合理性和可行性。從標(biāo)準(zhǔn)上實現(xiàn)了WebRTC和Jingle之間的互通。
[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.
【學(xué)位授予單位】：華南理工大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2014
【分類號】：TN919.8
本文編號：2312807
[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.
【學(xué)位授予單位】：華南理工大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2014
【分類號】：TN919.8
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