WebRTC, also known as RTCWeb, puts two-way media streaming capabilities into the web browser and provides an API to manage them (starting and stopping calls) from the JavaScript embedded in any web page. The technology has been pioneered in the two major browsers, Mozilla Firefox and Google Chrome. Other browsers have been following their lead. There was some instability in the early years of WebRTC but since mid-2014 the technology has stabilised significantly. There have been some pseudo-WebRTC solutions as well, specifically, browser plugins that offer behavior similar to WebRTC with an emphasis on a specific provider. These solutions are not true WebRTC and they are largely becoming irrelevant now that most users have upgraded to browsers with genuine WebRTC support built in. WebRTC provides a mechanism for peer-to-peer media streaming (audio or video) but it does not specify the use of any particular signalling system, the mechanism responsible for locating other users and routing calls to them. The talk is going to include how to develop FreeRTC plugins for third party applications. Code snippets will be shared. Now it is possible to talk to each other from different cms systems.
Speakers: Mesut Can Gurle