WebRTC技术初探

A Preliminary Exploration of Arising WebRTC Technology

传统的实时通信应用需要通过下载客户端或安装控件才能使用,这给用户造成一些不便。基于Web的实时通信技术简称WebRTC(Web real-time communication),是最近兴起的一种在Web浏览器内支持音/视频实时通信的新技术。基于该技术,开发者仅需调用简单的JavaScript API即可获得音/视频实时通信能力。该技术仍处在发展的初期阶段,本文将分析和探讨其产生背景、发展现状、技术架构以及对电信运营商的可能影响,希望能从一个较为宏观的角度描绘其发展脉络。

The traditional way to acquire RTC ability is to install a software client or a browser plug-in. This may take an user some tedious steps and prevent the popularization of RTC applications. WebRTC is an arising new technology affiliated to the HTML5 standard which intends to provide pervasive access to RTC ability from an arbitrary browser. It works by integrating a fully functional audio/video RTC architecture into browsers. With this new technology, web application developers can implement powerful browser-based audio/video RTC applications by simply invoking a few JavaScript API. Currently, WebRTC is still at its early stage of development. This paper will give a preliminary exploration to WebRTC by addressing its background, current status, technical architecture, and the probable influence to the telecoms. It can give the reader a portrait of the development roadmap of WebRTC.