Real-time communication with WebRTC: Google I/O 2013

Justin Uberti, Sam Dutton Presentation slides: http://io13webrtc.appspot.com 01:41 WebRTC across platforms 03:37 WebRTC APIs 04:32 MediaStream 09:44 RTCPeerConnection 12:28 RTCDataChannel 16:31 Signaling 18:43 STUN and TURN 22:18 Security 23:21 Architectures 25:03 Beyond browsers 27:15 Building a WebRTC app 29:56 Chris Wilson LIVE! 31:16 Links and resources WebRTC implements open standards for realtime, plugin-free video, audio and data communication. The core WebRTC APIs getUserMedia, RTCPeerConnection and DataChannel have now been implemented across Chrome and Firefox. In this session, we show you how to get started with building a WebRTC app: what's a MediaStream (aka getUserMedia) and how can I use it? resolution constraints signalling: what is it and how can I set it up? servers: what do I need? RTCPeerConnection: WebRTC s most powerful API RTCDataChannel: realtime communication of arbitrary data integrating WebRTC with Web Audio interoperability security During the session, we talk through code examples, live demos and production apps. For all I/O 2013 sessions, go to https://developers.google.com/live.