【摘要】：RTCWeb(Real-Time Communications in WEB-browsers)是指Web應(yīng)用通過調(diào)用瀏覽器提供的API,在不需要插件的情況下實(shí)現(xiàn)瀏覽器之間的實(shí)時(shí)音視頻通信。WebRTC系統(tǒng)是基于瀏覽器的集音視頻通信、及時(shí)消息、通訊錄、好友分組于一體,且能夠與使用SIP協(xié)議的VoIP系統(tǒng)進(jìn)行互通的系統(tǒng)。 本文關(guān)注WebRTC系統(tǒng)中信令子系統(tǒng)的設(shè)計(jì)與實(shí)現(xiàn),信令子系統(tǒng)是WebRTC系統(tǒng)中至關(guān)重要的組成部分,其主要功能是負(fù)責(zé)音視頻通信的媒體協(xié)商和會話控制。瀏覽器之間的實(shí)時(shí)通信采用的信令協(xié)議為WebSocket承載的ROAP(RTCWeb Offer/Answer Protocol)協(xié)議,WebRTC應(yīng)用與WebRTC服務(wù)器之間建立WebSocket連接,然后由服務(wù)器負(fù)責(zé)ROAP消息的轉(zhuǎn)發(fā)和會話控制。WebRTC系統(tǒng)與其他VoIP系統(tǒng)的互通由WebRTC網(wǎng)關(guān)實(shí)現(xiàn),WebRTC網(wǎng)關(guān)能夠與任何支持SIP協(xié)議的VoIP系統(tǒng)互通,其主要功能是ROAP協(xié)議與SIP協(xié)議的相互轉(zhuǎn)換以及會話控制。本文詳細(xì)設(shè)計(jì)了WebRTC系統(tǒng)中信令子系統(tǒng)的網(wǎng)絡(luò)架構(gòu)、業(yè)務(wù)流程以及具體模塊,根據(jù)設(shè)計(jì)實(shí)現(xiàn)了信令子系統(tǒng),最后對信令子系統(tǒng)進(jìn)行了功能測試,驗(yàn)證了可行性。
[Abstract]:RTCWeb (Real-Time Communications in WEB-browsers) means that Web application realizes real-time audio and video communication between browsers by calling API, provided by browser without the need of plug-in. WebRTC system is a browser-based collection of audio and video communication, timely message. Address book, a system in which friends are grouped together and interoperable with VoIP systems using the SIP protocol. This paper focuses on the design and implementation of signaling subsystem in WebRTC system. Signaling subsystem is an important part of WebRTC system. Its main function is responsible for media negotiation and session control of audio and video communication. The signaling protocol used in real-time communication between browsers is the ROAP (RTCWeb Offer/Answer Protocol protocol hosted by WebSocket. The WebSocket connection between the WebRTC application and the WebRTC server is established. Then the server is responsible for ROAP message forwarding and session control. The interworking between WebRTC system and other VoIP systems is realized by WebRTC gateway. WebRTC gateway can interoperate with any VoIP system supporting SIP protocol. Its main function is the conversion between ROAP protocol and SIP protocol and session control. In this paper, the network architecture, business process and specific modules of signaling subsystem in WebRTC system are designed in detail. According to the design, the signaling subsystem is implemented. Finally, the function of signaling subsystem is tested and the feasibility is verified.
【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】：TP393.092

【參考文獻(xiàn)】

相關(guān)期刊論文 前1條

1 樂利鋒;彭晉;段曉東;;RTCWeb及其與IMS的融合研究[J];電信科學(xué);2013年01期

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本文編號：2361816