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








