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: WebRTCs 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.