WebRTC

WebRTC stands for Web Real-Time Communication. It is a collection of browser APIs and protocols that enable peer-to-peer audio, video, and data communication directly between devices, without the need for plugins. The project is standardized by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF) and is implemented in major browsers.

The core APIs are getUserMedia, RTCPeerConnection, and RTCDataChannel. getUserMedia grants access to local media devices; RTCPeerConnection manages the media path and network negotiation; RTCDataChannel provides peer-to-peer data transfer. Signaling—exchanging offers, answers, and ICE candidates to establish the connection—is outside the WebRTC specification and is implemented by developers in their own signaling system.

WebRTC uses NAT traversal with ICE, aided by STUN and TURN servers. Media streams are secured by DTLS-SRTP, and all data and media are encrypted in transit. Connections typically run over secure contexts (HTTPS or secure origins).

Codecs commonly negotiated include video codecs like VP8, VP9, and H.264, and audio codecs such as Opus and G.711. Data channels use SCTP over DTLS. WebRTC supports one-to-one and multi-party scenarios; for group calls, servers such as selective forwarding units (SFUs) or media servers are often used to scale.

Usage and compatibility: WebRTC is supported in Chrome, Firefox, Safari, and Edge, with ongoing improvements in latency, reliability, and accessibility. It is used in video conferencing, live streaming, file transfer, and real-time gaming contexts.