Basar Daldal, Ibrahim Bilgin, Dogac Basaran, Selin Metin
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引用次数: 4
摘要
基于webrtc的应用程序被认为主要基于点对点通信,其中应用程序的一个实例与另一个实例进行通信。然而,情况并非总是如此:基于webbrtc的应用程序与传统VoIP设备或媒体服务器(如视频会议服务器)通信也很常见。能够进行两种类型的通信(WebRTC到WebRTC和WebRTC到传统VoIP)是WebRTC产品的一个区别因素。本文就如何利用诸如RESTful Web Services之类的行业标准方法在信令层上实现这一点提出了建议。
Using Web Services for WebRTC signaling interoperability
WebRTC-based applications are assumed to be based mostly on peer-to-peer communication, where an instance of the application is talking to another instance. However, this is not always the case: a WebRTC-based application communicating with a legacy VoIP device or Media Server like a Video Conference Server is also very common. The ability to make both types of communications (WebRTC to WebRTC and WebRTC to Legacy VoIP) is a differentiating factor for a WebRTC product. This paper makes a proposal on how this could be achieved on the signaling layer by leveraging an industry standard method such as the RESTful Web Services.