【摘要】：自從2011年 Google 將 WebRTC( Web Real-Time Communication)開源以后,各種基于WebRTC的Web應(yīng)用增長迅速,也有越來越多的瀏覽器支持WebRTC。人們只需打開瀏覽器就可以進(jìn)行音視頻聊天,而無需安裝具體的PC應(yīng)用。但是在移動(dòng)互聯(lián)網(wǎng)的背景下,WebRTC在移動(dòng)端的發(fā)展則相對緩慢,當(dāng)今市場更是沒有一款基于WebRTC的應(yīng)用能夠支持Web和移動(dòng)端的聯(lián)合通信。本文重點(diǎn)關(guān)注基于WebRTC的異構(gòu)終端之間的通信問題,異構(gòu)終端指Web端和Android端。我們模擬視頻會議的場景,不同終端的成員都可以加入到同一個(gè)會議室同其他終端音視頻聊天。本文結(jié)合WebRTC端到端傳輸特點(diǎn)將其應(yīng)用到視頻會議系統(tǒng)環(huán)境中,并設(shè)計(jì)一套方案實(shí)現(xiàn)視頻會議中成員的加入和離開。我們選擇XMPP作為信令的承載協(xié)議,以及XMPP的擴(kuò)展協(xié)議Jingle作為會話控制協(xié)議,利用WebRTC提供的API實(shí)現(xiàn)本地音視頻流的采集、傳輸與播放。每個(gè)客戶端需要先登錄到XMPP服務(wù)器,然后輸入房間號以加入視頻會議房間,當(dāng)有其他成員加入時(shí),媒體流服務(wù)器便會轉(zhuǎn)發(fā)媒體流給房間內(nèi)的其他成員。信令服務(wù)器還會維持每個(gè)客戶端的狀態(tài),客戶端通過心跳機(jī)制一直向服務(wù)器發(fā)送消息來保證自己在線狀態(tài),當(dāng)服務(wù)器一定時(shí)間收不到該客戶端發(fā)送消息便會認(rèn)為該客戶端離線�？蛻舳撕托帕罘⻊�(wù)器之間的信道區(qū)別于傳輸媒體流所用的信道。根據(jù)本文設(shè)計(jì),我們搭建服務(wù)器端,并開發(fā)Android客戶端應(yīng)用。最終成功完成此視頻會議系統(tǒng),實(shí)現(xiàn)各異構(gòu)終端的兩兩連接以及跨終端連接。本文將給出Android客戶端詳細(xì)設(shè)計(jì)與實(shí)現(xiàn)。最后,本文對該系統(tǒng)丟包率、時(shí)延和幀率進(jìn)行了測試,經(jīng)過對數(shù)據(jù)的分析對比,發(fā)現(xiàn)WebRTC技術(shù)可以應(yīng)對網(wǎng)絡(luò)波動(dòng)情況,有較低的時(shí)延,能滿足視頻會議系統(tǒng)的要求,從而驗(yàn)證了此方案的可行性。
[Abstract]:Since Google was open to WebRTC (Web Real-Time Communication) in 2011, a variety of WebRTC based Web applications have grown rapidly, and more and more browsers support WebRTC. people for audio and video chatting without the need to install specific PC. But in the context of mobile Internet, WebRTC is in the context of the mobile Internet. The development of mobile terminal is relatively slow, and there is no WebRTC based application that can support the joint communication between Web and mobile terminal. This paper focuses on the communication problem between the heterogeneous terminals based on WebRTC, the heterogeneous terminal refers to the Web end and the Android side. In the same conference room, we chat with other terminal audio and video. This paper applies WebRTC end-to-end transmission characteristics to the video conference system environment, and designs a set of solutions to join and leave members in video conferencing. We choose XMPP as a signalling protocol, and XMPP extension protocol Jingle as a session control. The protocol uses the API provided by WebRTC to collect, transmit and play the local audio and video stream. Each client needs to log in to the XMPP server first, then enter the room number to join the video conference room. When other members join, the media server will forward the media stream to the other members in the room. The signaling server will be maintained. Each client's state, the client sends messages to the server through the heartbeat mechanism to ensure its own online status. When the server is not able to receive the client for a certain time, the client will think the client is offline. The channel between the client and the signaling server is different from the channel used by the transmission media stream. We build the server side and develop the Android client application. Finally, we successfully complete the video conference system to realize the 22 connection and the cross terminal connection of the heterogeneous terminals. This paper will give the detailed design and implementation of the Android client. Finally, the packet loss rate, the time delay and the frame rate are tested, and the data are analyzed. In contrast, it is found that WebRTC technology can cope with network fluctuations, and has low latency to meet the requirements of videoconferencing system, thus verifying the feasibility of this scheme.
【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】：2017
【分類號】：TP393.0;TN948.63

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