跨异构终端的WebRTC移动多媒体技术研究
发布时间：2018-07-30 07:14
【摘要】：自从2011年 Google 将 WebRTC( Web Real-Time Communication)开源以后,各种基于WebRTC的Web应用增长迅速,也有越来越多的浏览器支持WebRTC。人们只需打开浏览器就可以进行音视频聊天,而无需安装具体的PC应用。但是在移动互联网的背景下,WebRTC在移动端的发展则相对缓慢,当今市场更是没有一款基于WebRTC的应用能够支持Web和移动端的联合通信。本文重点关注基于WebRTC的异构终端之间的通信问题,异构终端指Web端和Android端。我们模拟视频会议的场景,不同终端的成员都可以加入到同一个会议室同其他终端音视频聊天。本文结合WebRTC端到端传输特点将其应用到视频会议系统环境中,并设计一套方案实现视频会议中成员的加入和离开。我们选择XMPP作为信令的承载协议,以及XMPP的扩展协议Jingle作为会话控制协议,利用WebRTC提供的API实现本地音视频流的采集、传输与播放。每个客户端需要先登录到XMPP服务器,然后输入房间号以加入视频会议房间,当有其他成员加入时,媒体流服务器便会转发媒体流给房间内的其他成员。信令服务器还会维持每个客户端的状态,客户端通过心跳机制一直向服务器发送消息来保证自己在线状态,当服务器一定时间收不到该客户端发送消息便会认为该客户端离线。客户端和信令服务器之间的信道区别于传输媒体流所用的信道。根据本文设计,我们搭建服务器端,并开发Android客户端应用。最终成功完成此视频会议系统,实现各异构终端的两两连接以及跨终端连接。本文将给出Android客户端详细设计与实现。最后,本文对该系统丢包率、时延和帧率进行了测试,经过对数据的分析对比,发现WebRTC技术可以应对网络波动情况,有较低的时延,能满足视频会议系统的要求,从而验证了此方案的可行性。
[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.
【学位授予单位】：北京邮电大学
【学位级别】：硕士
【学位授予年份】：2017
【分类号】：TP393.0;TN948.63
[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.
【学位授予单位】：北京邮电大学
【学位级别】：硕士
【学位授予年份】：2017
【分类号】：TP393.0;TN948.63
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