At a high level, WebRTC is a way for browsers to have access to video, speakers and microphone directly without an intermediary like Java or Flash.  It’s most common use to-date has been for voice and video calling. WebRTC does NOT have a built-in protocol for call setup.  You can use an existing call setup protocol like SIP or make your own. OnSIP has open-sourced sip.js (https://sipjs.com/) as an OpenSource javascript library to add the SIP protocol to WebRTC. When using SIP as the call setup protocol for WebRTC, WebRTC is simply the conduit to the camera, speakers and microphone.  SIP does all the heavy lifting of notifying each end…