Real-time communication with WebRTC
Sam Dutton, Justin Uberti
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.
Sam Dutton is a Developer Advocate for Chrome in London, specialising in WebRTC, OWP, media APIs, and the mobile web. Sam has designed and coded numerous websites as well as native desktop and mobile applications. He previously worked for BBC R&D, ITN, Decca Records and Picador Books, and is a governor at Ravenstone School in south London.
Justin Uberti is one of the co-creators of the WebRTC initiative, and leads the WebRTC engineering team at Google. Previously, Justin helped create Google+ Hangouts.