WebRTC A Quick Introduction
summary
Live demo uses meet.jit.si/krankygeek and recommends adapter.js for cross‑browser quirks. Covers the three concrete APIs: getUserMedia returns a MediaStream with separate video and audio tracks (use deviceId via enumerateDevices), RTCPeerConnection for SDP, ICE/STUN/TURN and signaling (signaling is left to implement), and RTCDataChannel reliable or unreliable like TCP/UDP for low‑latency P2P data. Notes codecs (H.264, VP8/“BPA”, Opus, G.711), getStats, ORTC work-in-progress on Edge, mobile SDK choices, HTTPS and scaling (MCU/SFU) requirements.
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chapters (16)
- 0:00Intro
- 1:42Real-time communication without plugins
- 2:14Three main tasks
- 3:05MediaStream (aka getUserMedia) • Allows the capture of media input devices (cameras and microphones)
- 5:53MediaStream object
- 6:24Media device enumeration & selection
- 7:04MediaStream Demo
- 8:16RTCPeerConnection
- 9:53Basic call flow
- 11:54RTCDataChannel
- 14:36Object APIs - ORTC
- 15:18WebRTC Media Codecs
- 16:31WebRTC support in desktop browsers
- 16:59WebRTC support in mobile browsers
- 17:23WebRTC on native mobile apps How to get the WebRTC library?
- 18:42Deployment considerations
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