What is WebRTC
WebRTC is the technology we use to ensure that as many people as possible can have a frictionless video consultation experience. WebRTC is a cross-platform standard for conducting video calls.
WebRTC (Web Real-Time Communications) is a technology which enables Web applications and sites to capture and optionally stream audio and/or video media, as well as to exchange arbitrary data between browsers without requiring an intermediary. The set of standards that comprises WebRTC makes it possible to share data and perform teleconferencing peer-to-peer, without requiring that the user install plug-ins or any other third-party software. Source
WebRTC is a cross-platform, cross-vendor project that’s being standardized through the W3C and IETF. There are a number of sub-protocols within the WebRTC stack, but the general workflow is that a signaling session is established via a server, and then a media channel directly between peers. The full handshake process is described here: https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Connectivity
For additional detail about how WebRTC operates in detail, please see the IETF Memo Web Real-Time Communication (WebRTC): Media Transport and Use of RTP.