WebRTC is a browser-native framework for real-time audio, video, and data communication between peers without plugins or intermediate servers for media relay.

What is WebRTC? Definition, examples and how it works

WebRTC (Web Real-Time Communication) is an open standard and set of JavaScript APIs that enable peer-to-peer audio, video, and arbitrary data exchange directly between browsers and other endpoints. Developed by the W3C and IETF, it was first standardized in 2021 (W3C Recommendation) and is supported in all major browsers including Chrome, Firefox, Safari, and Edge. It eliminates the need for proprietary plugins like Flash or Java applets for real-time communications.

WebRTC works by establishing a direct peer-to-peer connection using a signaling channel (typically over WebSocket or HTTP) to exchange session descriptions and network candidates via the Session Description Protocol (SDP) and ICE (Interactive Connectivity Establishment). ICE uses STUN servers to discover public IP addresses and ports, and falls back to TURN servers when symmetric NATs or firewalls block direct connections. Once the connection is established, media is encrypted using SRTP (Secure Real-Time Transport Protocol) with DTLS-SRTP keying, and data channels use SCTP over DTLS. The APIs include getUserMedia for accessing cameras and microphones, RTCPeerConnection for managing the peer connection, and RTCDataChannel for low-latency data transfer.

In the wider stack, WebRTC sits as a transport-agnostic application layer protocol that can run over UDP, TCP, or TLS. It is commonly used in video conferencing applications (e.g., Google Meet, Zoom via browser), live streaming, screen sharing, and peer-to-peer file transfer. Its design prioritizes low latency and strong security, with mandatory encryption for all media and data. WebRTC is not limited to browsers; native implementations exist for iOS, Android, and desktop applications via libraries like Google's libwebrtc.

Key facts

Standardized by W3C and IETF; W3C Recommendation published January 2021.
Uses ICE with STUN and TURN to traverse NATs and firewalls.
All media and data channels are encrypted by default using DTLS-SRTP.
Operates over UDP primarily, with fallback to TCP or TLS.
Supported in Chrome, Firefox, Safari, Edge, and many mobile browsers.

How it works in practice

A user opens a video call in a web browser. The browser uses getUserMedia to capture the local camera feed. It then exchanges SDP offers and answers with the remote peer via a signaling server. ICE candidates are gathered using a STUN server to find the public IP and port. Once the peer-to-peer connection is established, encrypted video and audio flow directly between the two browsers without passing through a media server, reducing latency and server costs.

Related terms

ICE STUN TURN SDP DTLS SRTP SCTP

References

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