Real-time communication with WebRTC
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.