~/talk

WebRTC: Connecting Embedded, Mobile and Desktop with the Lowest Latencies

slides:none on file

WebRTC provides sub-second peer-to-peer audio/video using RTP/ICE/DTLS/SRTP and SRTP keying, with common stacks like libwebrtc (Google), Pion (Go) and a GStreamer webrtc element; server choices include TURN (relay), SFU (Jitsi/Janus/mediasoup) and MCU (FreeSWITCH). TURN bandwidth, not CPU, usually saturates first; TLS/AES encryption is mandatory so Cortex A5 too slow for 1080p without hardware crypto. Browsers move on a ~four‑week cycle, so continuous CI and rapid device updates are required.

AI-written summary · about

  1. 0:00Intro
  2. 1:42JavaScript
  3. 2:50Components IETF standards
  4. 5:01Interactive Connectivity Establishment (ICE)
  5. 6:45Datagram TLS
  6. 8:30Other features
  7. 9:25Data channels
  8. 10:52Selective Forwarding Unit
  9. 11:42Multi-Conference Unit
  10. 12:42WebRTC endpoints
  11. 13:58GStreamer
  12. 16:08WebRTC isn't for everyone
  13. 16:59WebRTC is for you
  14. 18:03Performance required
  15. 20:26Browsers move fast
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