What is WebRTC?

WebRTC is the technology that lets you make a live video or voice call right inside your browser, with no app to install and no plugin. You have almost certainly used it without ever hearing the name.

You already use it constantly

You click a link and you are in a video call. Instantly. No download, no account, nothing to set up. Google Meet, Discord, a telehealth appointment, the "join call" button on some website: all of them run on WebRTC.

There is a name for the thing that makes that possible, and almost nobody who uses it every day has ever heard it. It is WebRTC, which stands for Web Real-Time Communication.

Your browser is secretly a phone system

Here is the surprising part. The same Chrome or Safari you are reading this in has a full real-time communication system built into it: microphone access, camera, live audio and video, all standard, all already there.

For most of the internet's life this was impossible. You needed Flash, a plugin, or a downloaded app to do live video. Since around 2013, browsers just do it natively. Most people never noticed the moment it became built in.

The calls go directly between people

WebRTC calls do not work the way you might guess. When you are on a browser video call with one other person, your audio and video often travel straight to their device. Not stored on a company server in the middle. Browser to browser, directly across the internet.

That is why it can feel so fast, and why it is private by design. The two of you form a direct line, and the media never has to sit on anyone else's computer.

The hard part: finding each other

Two strangers' devices cannot simply find one another on the open internet. Both are hidden behind home routers, each with only a private address that means nothing to the outside world.

So at the very start of a call there is a frantic negotiation. Each side gathers every possible address it might be reachable on, shares them, and the two devices try combinations until one path connects. This process has a name: ICE, Interactive Connectivity Establishment.

You never see it. It happens in the second before the call rings. And when it fails, that is the "connecting..." that never connects.

Where it goes from here

Once you understand that a WebRTC call is really "two browsers finding a path to each other and then sending media directly," the rest of the topic opens up: how that path is discovered (ICE), what happens when a direct path is blocked (STUN and TURN), and how the two sides agree on formats (SDP).

Every one of those is just a piece of the same story: getting your voice from one browser to another, reliably, in real time.