本文選題：WebRTC + 實(shí)時(shí)通信��； 參考：《北京郵電大學(xué)》2014年碩士論文

【摘要】：WebRTC (Web Real-Time Communications)是一種不依賴于第三方插件的由瀏覽器原生支持的實(shí)時(shí)音視頻通信技術(shù)。作為一種新興的web實(shí)時(shí)通信方式,WebRTC以其較低的開發(fā)維護(hù)成本,較高的用戶體驗(yàn),基于瀏覽器的跨平臺(tái)特性以及標(biāo)準(zhǔn)化等優(yōu)勢(shì)受到業(yè)界的廣泛關(guān)注。WebRTC技術(shù)的出現(xiàn)在對(duì)互聯(lián)網(wǎng)實(shí)時(shí)通信應(yīng)用產(chǎn)生變革的同時(shí),對(duì)電信網(wǎng)實(shí)時(shí)通信業(yè)務(wù)也帶來了機(jī)遇。實(shí)現(xiàn)WebRTC技術(shù)與IMS的結(jié)合,運(yùn)營(yíng)商可以將WebRTC的龐大潛在用戶群、低開發(fā)維護(hù)成本、快速部署等優(yōu)點(diǎn)應(yīng)用在IMS的業(yè)務(wù)中,為用戶提供更豐富的體驗(yàn),促使傳統(tǒng)用戶向IMS網(wǎng)絡(luò)遷移,創(chuàng)造巨大的商業(yè)價(jià)值。 本文針對(duì)WebRTC系統(tǒng)中的媒體子系統(tǒng)進(jìn)行設(shè)計(jì)和實(shí)現(xiàn),意在基于WebRTC技術(shù)實(shí)現(xiàn)實(shí)時(shí)的音視頻通信。媒體子系統(tǒng)具有兩種應(yīng)用場(chǎng)景,一種是在互聯(lián)網(wǎng)架構(gòu)中實(shí)現(xiàn)瀏覽器間的實(shí)時(shí)通信,另一種是實(shí)現(xiàn)瀏覽器WebRTC客戶端與IMS網(wǎng)絡(luò)終端間的媒體互通,嘗試將WebRTC技術(shù)同IMS業(yè)務(wù)相結(jié)合。本文在對(duì)WebRTC標(biāo)準(zhǔn)化過程中的相關(guān)協(xié)議規(guī)范以及實(shí)現(xiàn)WebRTC系統(tǒng)涉及到的關(guān)鍵技術(shù)進(jìn)行研究的基礎(chǔ)上,從信令和媒體兩個(gè)層面對(duì)媒體子系統(tǒng)在兩種應(yīng)用場(chǎng)景中的功能需求做出分析。在此基礎(chǔ)上,本文給出了瀏覽器間媒體通信模塊以及WebRTC客戶端同IMS網(wǎng)絡(luò)之間媒體互通模塊的詳細(xì)設(shè)計(jì),并且介紹了通信網(wǎng)絡(luò)的架構(gòu)和通信業(yè)務(wù)的流程等。在互通場(chǎng)景中,本文著重介紹了媒體互通架構(gòu)中的關(guān)鍵模塊——媒體網(wǎng)關(guān),該網(wǎng)關(guān)遵從ICE協(xié)議分別同WebRTC客戶端和IMS網(wǎng)絡(luò)終端建立P2P通信,并進(jìn)行異構(gòu)網(wǎng)絡(luò)間媒體傳輸協(xié)議的轉(zhuǎn)換,以中繼的方式實(shí)現(xiàn)媒體流的互通。測(cè)試結(jié)果驗(yàn)證了媒體子系統(tǒng)實(shí)現(xiàn)的正確性和穩(wěn)定性。
[Abstract]:WebRTC Real-Time Communications is a real-time audio and video communication technology that is not dependent on third-party plug-ins and is native to browsers. As a new way of web real-time communication, WebRTC has low cost of development and maintenance and high user experience. The cross-platform characteristics and standardization of browsers have attracted wide attention of the industry. The emergence of .WebRTC technology has brought about a revolution in the application of real-time communication over the Internet and an opportunity for real-time communication services in telecommunications networks. With the combination of WebRTC technology and IMS, operators can apply the advantages of WebRTC, such as large potential user group, low development and maintenance cost and rapid deployment, to IMS business, providing users with a richer experience and promoting traditional users to migrate to IMS network. Create great commercial value. In this paper, the media subsystem in WebRTC system is designed and implemented to realize real-time audio and video communication based on WebRTC technology. The media subsystem has two application scenarios, one is to realize the real-time communication between browsers in the Internet architecture, the other is to realize the media interworking between the browser WebRTC client and the IMS network terminal, and try to combine the WebRTC technology with the IMS service. Based on the research of the related protocol specification in the process of WebRTC standardization and the key technologies involved in the implementation of WebRTC system, this paper analyzes the functional requirements of the media subsystem in the two application scenarios from the two aspects of signaling and media. On this basis, this paper gives the detailed design of the media communication module between browsers and the media interworking module between WebRTC client and IMS network, and introduces the architecture of communication network and the flow of communication service. In the interworking scenario, this paper mainly introduces the key module of media interworking architecture, media gateway. The gateway establishes P2P communication with WebRTC client and IMS network terminal according to ICE protocol, respectively. The interworking of media flows is realized by relaying the media transport protocols between heterogeneous networks. The test results verify the correctness and stability of the media subsystem.
【學(xué)位授予單位】：北京郵電大學(xué)
【學(xué)位級(jí)別】：碩士
【學(xué)位授予年份】：2014
【分類號(hào)】：TP393.092

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