WebRTC

WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications capabilities via simple JavaScript APIs, enabling peer-to-peer audio, video, and data communication directly between devices without requiring plugins or third-party downloads. The project is jointly maintained by Google, Mozilla, Apple, and Microsoft, with the W3C WebRTC Working Group responsible for the official API specification and the IETF handling the underlying protocol standardization including RTCWeb. The three core JavaScript APIs are getUserMedia for accessing device cameras and microphones with configurable resolution, frame rate, and device constraints, RTCPeerConnection for establishing secure peer connections with ICE, STUN, and TURN servers handling NAT traversal, codec negotiation, bandwidth estimation, and adaptive bitrate streaming, and RTCDataChannel for bidirectional arbitrary data transfer with low latency comparable to WebSocket but operating over the peer-to-peer SCTP protocol. The media pipeline supports modern video codecs including VP8, VP9, AV1, and H.264, audio codecs including Opus, G.711, and the Lyra neural audio codec, with echo cancellation, noise suppression, and automatic gain control built into the audio processing module. The technology powers video conferencing at scale for applications including Google Meet, Discord, Microsoft Teams web client, Facebook Messenger, Amazon Chime, and hundreds of telehealth and remote support platforms. Server-side implementations exist for Node.js via node-webrtc, Go, Rust, Python, and Java, while cloud services like Twilio, Vonage, Daily.co, and Agora provide managed WebRTC infrastructure with SFU and MCU media servers for multi-party calls. The project also maintains adapter.js for cross-browser API normalization. Copyright 2011-2026 The WebRTC Project Authors.