WebRTC系统中信令子系统的设计与实现
发布时间：2018-11-27 19:30
【摘要】：RTCWeb(Real-Time Communications in WEB-browsers)是指Web应用通过调用浏览器提供的API,在不需要插件的情况下实现浏览器之间的实时音视频通信。WebRTC系统是基于浏览器的集音视频通信、及时消息、通讯录、好友分组于一体,且能够与使用SIP协议的VoIP系统进行互通的系统。 本文关注WebRTC系统中信令子系统的设计与实现,信令子系统是WebRTC系统中至关重要的组成部分,其主要功能是负责音视频通信的媒体协商和会话控制。浏览器之间的实时通信采用的信令协议为WebSocket承载的ROAP(RTCWeb Offer/Answer Protocol)协议,WebRTC应用与WebRTC服务器之间建立WebSocket连接,然后由服务器负责ROAP消息的转发和会话控制。WebRTC系统与其他VoIP系统的互通由WebRTC网关实现,WebRTC网关能够与任何支持SIP协议的VoIP系统互通,其主要功能是ROAP协议与SIP协议的相互转换以及会话控制。本文详细设计了WebRTC系统中信令子系统的网络架构、业务流程以及具体模块,根据设计实现了信令子系统,最后对信令子系统进行了功能测试,验证了可行性。
[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.
【学位授予单位】：北京邮电大学
【学位级别】：硕士
【学位授予年份】：2014
【分类号】：TP393.092
本文编号：2361816
[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.
【学位授予单位】：北京邮电大学
【学位级别】：硕士
【学位授予年份】：2014
【分类号】：TP393.092
【参考文献】
相关期刊论文 前1条
1 乐利锋;彭晋;段晓东;;RTCWeb及其与IMS的融合研究[J];电信科学;2013年01期
，本文编号：2361816
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