WebRTC, Web Real-Time Communications, is revolutionizing the way web users communicate, both in the consumer and enterprise worlds. WebRTC adds standard APIs and built-in real-time audio and video capabilities and codecs to browsers without a plug-in. With just a few lines of JavaScript, web developers can add high quality peer-to-peer voice, video, and data channel communications to their collaboration, conferencing, telephony, or even gaming site or application. Presented by Dan Burnett, WebRTC Spec editor and co-author of the popular "WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web", and "Dr. Alex" Gouaillard, the king of WebRTC testing, this tutorial rapidly covers the breadth of topics involved in WebRTC, from STUN and TURN, to getUserMedia(), to SDP offer/answer and more, finishing with a status check on today's implementations of the technology. ******