Realtime

WebRTC

A peer-to-peer realtime communication framework for audio, video, and arbitrary data channels, standardized by the W3C and IETF. WebRTC defines an SDP-based offer/answer signaling model, ICE/STUN/TURN for NAT traversal, SRTP for media, and SCTP-based data channels. It powers video conferencing (LiveKit, Daily, Agora, Twilio Video, Zoom Web), live audio rooms, and low-latency game streaming.

GitHub

Documentation

📖
Documentation
https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API

Other Resources

🔗
Specification
https://www.w3.org/TR/webrtc/
🔗
Portal
https://webrtc.org/

API entry from apis.yml

apis.yml Raw ↑ 
aid: realtime:webrtc
name: WebRTC
description: A peer-to-peer realtime communication framework for audio, video, and arbitrary data channels,
  standardized by the W3C and IETF. WebRTC defines an SDP-based offer/answer signaling model, ICE/STUN/TURN
  for NAT traversal, SRTP for media, and SCTP-based data channels. It powers video conferencing (LiveKit,
  Daily, Agora, Twilio Video, Zoom Web), live audio rooms, and low-latency game streaming.
humanURL: https://webrtc.org/
tags:
- WebRTC
- Peer-to-Peer
- Video
- Audio
- Data Channel
- Protocol
- Standard
properties:
- type: Specification
  url: https://www.w3.org/TR/webrtc/
- type: Documentation
  url: https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API
- type: Portal
  url: https://webrtc.org/