WebRTC: how browser video calls work

What is WebRTC?

WebRTC is an open web standard that lets browsers send audio, video and data directly to each other in real time, without plugins or a downloaded app. It is the technology behind most browser-based video calls.

How does WebRTC work?

Each browser captures its camera and microphone, and the two sides exchange connection details through a signalling server. They then try to connect directly, and a WebRTC peer connection carries the media between them, adjusting quality to the available bandwidth as the call goes on.

STUN and TURN servers

Most devices sit behind routers that hide their address. STUN servers help a browser discover its public address so a direct connection can be made. When a direct path is impossible, as on some company networks, a TURN server relays the encrypted traffic between participants.

WebRTC encryption

Encryption is mandatory in WebRTC: media is encrypted with SRTP, using keys negotiated over DTLS between the participants. A relay server only forwards packets it cannot read.

WebRTC browser support

All major desktop and mobile browsers support WebRTC, which is what makes video calls without a download possible. Arkenso Meet is built on it: audio and video travel straight between the people in the call rather than through our servers, and guests join from a link in their browser.

Last reviewed 2026-11-01.

Related terms

  • Peer-to-peer video call — A peer-to-peer video call sends audio and video directly between participants’ devices instead of routing it through a central media server.
  • Video conferencing — Video conferencing is a live meeting between people in different places using video and audio over the internet, usually with screen sharing and chat, from a browser, an app or a meeting-room system.
  • Screen sharing — Screen sharing shows what is on your screen, a single window or a browser tab to the other people in a call, so they can follow a document, a demo or a problem as it happens.

All glossary terms →