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

Опубликовано 20 мая 2013, 5:31

Justin Uberti, Sam Dutton

Presentation slides: 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 developers.google.com/live.