WebRTC enables real-time, peer-to-peer audio, video, and data communication across web browsers and native applications. WebRTC is standardized by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF) and is implemented in major browsers including @Google Chrome, Firefox, Safari, and Edge. It exposes @JavaScript APIs for media capture (getUserMedia), connection setup and control (RTCPeerConnection), and low-latency data transport (RTCDataChannel). Connectivity uses Interactive Connectivity Establishment (ICE) for path selection and NAT traversal, often with Session Traversal Utilities for NAT (STUN) and Traversal Using Relays around NAT (TURN), while media is encrypted with DTLS-SRTP. Typical codecs include Opus for audio and VP8, VP9, or AV1 for video, with hardware acceleration when available. Primary use cases include video conferencing, live customer support, co-browsing, multiplayer synchronization, IoT control, and peer-to-peer file transfer. The open-source project and developer resources are available at webrtc.org, and key specifications include the W3C WebRTC API and the IETF RTCWEB suite, which define both the browser-facing interfaces and the underlying transport and security protocols.

Contexts

• #topia (See: @Topia)