本文選題：WebRTC + WEB前端子系統(tǒng)��； 參考：《北京郵電大學(xué)》2014年碩士論文

【摘要】：隨著互聯(lián)網(wǎng)技術(shù)的飛速發(fā)展和瀏覽器功能的日益增強(qiáng),各種音／視頻通信應(yīng)用不斷涌現(xiàn)。傳統(tǒng)的音／視頻通信應(yīng)用需要安裝插件或者把應(yīng)用做成客戶端的形式才能提供實(shí)時(shí)音／視頻通信能力,因?yàn)榛ヂ?lián)網(wǎng)的音頻、視頻通信服務(wù)技術(shù)一般都是私有技術(shù)。WebRTC (Web Real-Time Communication)是一種開源的軟件架構(gòu),直接在瀏覽器中集成了媒體采集、媒體編解碼、媒體信號處理、碼率控制、傳輸控制、差錯(cuò)控制等功能,為瀏覽器內(nèi)部實(shí)時(shí)音／視頻通信奠定了基礎(chǔ)。同時(shí),HTML5增加了新標(biāo)簽(audio/video/canvas)支持網(wǎng)頁中直接播放多媒體和繪制圖像�；谝陨霞夹g(shù),Web開發(fā)者可以直接在瀏覽器中創(chuàng)建視頻或語音聊天應(yīng)用,音／視頻通信業(yè)務(wù)將有龐大的市場需求。 本文旨在利用Web應(yīng)用的一致性屏蔽通信終端的差異性,開發(fā)一個(gè)可以在任何集成了WebRTC技術(shù)的瀏覽器上運(yùn)行的Web應(yīng)用,用戶只需通過網(wǎng)絡(luò)就可以使用相同的界面進(jìn)行實(shí)時(shí)通信,提高了用戶體驗(yàn),降低了軟件開發(fā)、運(yùn)行和維護(hù)的成本。本文首先圍繞背景展開介紹了WebRTC的架構(gòu)和特征,研究了RTCWeb的關(guān)鍵技術(shù)；然后,分析了系統(tǒng)的功能需求和非功能需求�；跒g覽器提供的WebRTC JavaScript API,利用RTCWeb關(guān)鍵技術(shù)和主流的Web開發(fā)工具,本文設(shè)計(jì)并實(shí)現(xiàn)了WebRTC系統(tǒng)的WEB前端子系統(tǒng)。本系統(tǒng)使用HTML5WebSocket協(xié)議定義了一個(gè)全雙工的通信信道,通過調(diào)用Web Socket API完成通道的建立、消息的發(fā)送和接收；使用XMPP協(xié)議棧完成與即時(shí)消息相關(guān)的功能；基于ROAP協(xié)議棧完成與多媒體通信相關(guān)的交互功能。WEB前端UI界面和業(yè)務(wù)邏輯使用主流的JavaWeb開發(fā)技術(shù)(Java/JSP/JavaScript/Struts2/JPA)實(shí)現(xiàn),完成了好友列表和通訊錄功能；通過HTML5標(biāo)簽和WebRTC JavaScript API實(shí)現(xiàn)Web實(shí)時(shí)音頻、視頻通信功能。本系統(tǒng)支持好友列表和通訊錄,支持即時(shí)消息和狀態(tài)呈現(xiàn),支持在集成了WebRTC的瀏覽器內(nèi)部實(shí)時(shí)音頻／視頻會話。最后,本系統(tǒng)經(jīng)過嚴(yán)格測試,達(dá)到了預(yù)期的效果。
[Abstract]:With the rapid development of Internet technology and the increasing function of browser, various audio / video communication applications are emerging. Traditional audio / video communication applications require plugins or client-side applications to provide real-time audio / video communication capabilities because of the audio content of the Internet. Video communication service technology is usually a private technology. WebRTC / Web Real-Time Communication is an open source software architecture, which directly integrates the functions of media collection, media coding and decoding, media signal processing, rate control, transmission control, error control and so on in the browser. It lays the foundation for real-time audio / video communication in browser. At the same time, HTML5 has added a new tag, audio / video- / canvas, which supports direct multimedia playback and rendering of images in web pages. Based on the above technology, Web developers can directly create video or voice chat applications in the browser, audio / video communication business will have a huge market demand. The purpose of this paper is to use the consistency of Web applications to shield the differences of communication terminals, and to develop a Web application that can be run on any browser that integrates WebRTC technology. Users can use the same interface for real-time communication only through the network. Improved user experience and reduced the cost of software development, operation and maintenance. This paper firstly introduces the architecture and characteristics of WebRTC, studies the key technologies of RTCWeb, and then analyzes the functional and non-functional requirements of the system. This paper designs and implements the WEB front-end subsystem of WebRTC system based on the WebRTC JavaScript API, provided by browser and the key technology of RTCWeb and the mainstream Web development tools. This system uses HTML5WebSocket protocol to define a full-duplex communication channel, through calling Web Socket API to complete the establishment of channels, message sending and receiving, using XMPP protocol stack to complete the functions related to instant messaging. Based on the ROAP protocol stack, the interactive function related to multimedia communication. Web front-end UI interface and business logic are implemented using the mainstream JavaWeb development technology. Java / JSP / JavaScript / Struts2 / JPA) is implemented, and the functions of friend list and address book are completed, and Web real-time audio is realized through HTML5 tags and WebRTC JavaScript API. Video communication function. The system supports friend list and address book, instant message and status rendering, and real-time audio / video session in the browser integrated with WebRTC. Finally, the system is tested strictly and the expected effect is achieved.
【學(xué)位授予單位】：北京郵電大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2014
【分類號】：TP311.52;TP393.09

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