Random chat explained
What Is WebRTC? A Plain-English Guide
What is WebRTC? It's the technology built into modern browsers that lets websites make live video and voice calls with no plugin or download. WebRTC, short for Web Real-Time Communication, handles three jobs: getting your camera and microphone with your permission, connecting your browser to another person's, and encrypting everything that flows between you.
It's an open standard published by the W3C and the IETF, and it works in Chrome, Safari, Edge and Firefox. If you've ever joined a video call from a browser tab, you've almost certainly used it.
Where WebRTC came from
Before WebRTC, video in the browser usually meant installing a plugin such as Adobe Flash, or downloading a separate app. In 2011 Google released WebRTC as open-source code, and browser makers and standards bodies spent the following decade turning it into a shared standard. It became an official W3C and IETF standard in January 2021.
Apple added WebRTC to Safari in 2017, which made browser video calls practical on iPhone as well.
That history is why a service like Omlet can work as a plain website. There's nothing to download or install, because the calling technology is already in your browser.
How WebRTC works, in three jobs
1. Camera and microphone access
A website can't switch on your camera by itself. It asks the browser, and the browser asks you, with a prompt to allow or block. You can change that choice later in your browser's site settings, and most browsers show an indicator while the camera is in use. Browsers only offer this on secure (HTTPS) pages.
2. Connecting two browsers
WebRTC doesn't decide who talks to whom; the website does. Once it has paired two people, it passes a few setup messages between their browsers, a step called signaling. Each browser shares which formats it supports and a list of possible network addresses. A process called ICE then tries those routes until one works: directly if possible, with help from a STUN server, or through a TURN relay if not.
For the direct-versus-relay part, see peer-to-peer video chat explained simply and what a TURN server is.
3. Encryption
Encryption isn't optional in WebRTC. Audio and video are protected with a protocol called SRTP, using keys the two browsers agree on directly through a handshake called DTLS. A relay in the middle passes the scrambled data along but doesn't have the keys to read it. Our guide on whether video calls are encrypted explains what that does and doesn't protect.
WebRTC explained in one table
These are the terms you'll run into most, in plain words.
| Term | What it means |
|---|---|
| Signaling | The website passing setup messages between two browsers before a call |
| Peer connection | The live link between two browsers that carries the call |
| ICE | The process of testing possible routes and picking one that works |
| NAT | How your router shares one public address among all the devices in your home |
| STUN | A server that tells your device its public address, so a direct route can be tried |
| TURN | A relay server that carries the call when no direct route works |
| Codec | The method used to compress video or audio, such as VP8, H.264 or Opus |
| Data channel | A WebRTC link for sending text or files between browsers |
| DTLS-SRTP | The encryption that protects WebRTC audio and video |
What is a WebRTC leak?
A WebRTC leak is when a website learns your real IP address through WebRTC even though you're using a VPN or proxy that's supposed to hide it. It happens because, to find a route for a call, the browser gathers possible addresses for your device, and in some setups that includes addresses outside the VPN tunnel.
- Browser proxy extensions often cover normal web traffic but not WebRTC, which makes them a common cause.
- Some VPN setups, such as split tunneling that excludes your browser, can let WebRTC traffic bypass the tunnel.
- Local addresses, the private ones your router hands out, used to be exposed widely. Modern browsers now hide them in most situations.
On a video chat site there's also an expected version of this: a direct call shares your IP address with the person you're talking to. That isn't a bug; it's how a direct connection works. Omlet's connection can share your IP with your chat partner, which is why a VPN is worth considering if that matters to you.
How to reduce WebRTC leaks
- Use a full VPN app on your device rather than a browser proxy extension, and don't exclude your browser from the tunnel.
- Turn on your VPN's leak protection setting if it has one.
- If you never make calls in the browser, you can limit WebRTC instead. Firefox lets you switch it off in about:config (media.peerconnection.enabled), and extensions that restrict it exist for Chromium-based browsers. This breaks video chat sites, so switch it back on when you want to call.
A VPN hides your IP from chat partners but can add delay or block calls. If that happens, see VPN causing video chat connection problems.
Is WebRTC safe to use?
WebRTC is a widely reviewed standard built into every major browser, with safety designed in:
- Camera and microphone access always needs your permission.
- It only works on secure HTTPS pages.
- Audio and video are always encrypted between the browsers.
- Your browser shows when your camera or mic is in use.
What WebRTC can't do is make the person on the other end trustworthy. They can still record their own screen, and on a direct call their device can see your IP address. So the usual habits apply: keep personal details private and leave any chat that feels wrong.
WebRTC on Omlet
Omlet runs in the browser on WebRTC. When you press Start, your browser asks for camera and microphone access, Omlet pairs you with the next person, and your browsers connect: peer to peer when possible, relayed through Omlet's network provider when not. Calls are encrypted, and Omlet doesn't record or store them.
No download, no account: start a video chat from any modern browser on your phone, tablet or computer.