What is WebRTC?

What is WebRTC?

WebRTC is a browser-supported set of technologies that lets two clients exchange audio, video, or data directly when the network allows it. It is designed for video calls, screen sharing, and low-latency data transfer without requiring users to install a separate application.

A WebRTC session starts with signaling: participants exchange session metadata through a channel that is often built with WebSocket or HTTP. ICE then tries to discover a workable network path, STUN servers help clients learn their public addresses, and TURN relays traffic when direct peer-to-peer connectivity is blocked. Media is protected with SRTP, while data channels use a secure transport layer.

Where It Is Used

WebRTC is used in customer support calls, remote learning, telehealth, live inspections, file sharing, and browser-based collaboration tools. If the need is only server-to-client updates, SSE may be simpler. For general two-way application messaging, WebSocket may be enough. WebRTC becomes the stronger fit when audio, video, and peer-to-peer low latency matter.

Production reliability depends heavily on network conditions. Corporate firewalls, carrier NAT, and weak mobile connections make TURN capacity, adaptive bitrate handling, permission flows, and monitoring important design choices. Modern protocols such as HTTP/3 can improve web transport performance, but WebRTC’s media negotiation and NAT traversal concerns are a separate architecture topic.