WebRTC
What is WebRTC?
It is the reason a guest can click a link and talk on camera without installing anything: the browser handles the camera, the microphone, encryption and the network connection.
On this page
How does WebRTC work?
WebRTC does three jobs that a video call needs. First, it captures the camera and microphone, but only after the browser has asked you for permission. Second, it connects: the two ends exchange connection details through the website's own server, then test possible network routes and keep the best one that works. Third, it transports the media: audio and video are compressed with codecs, encrypted and sent as a steady stream of small packets.
| Part | What it does |
|---|---|
| Device access | The browser asks before a site can use your camera, microphone or screen. |
| Signalling | The service's server passes connection details between the two ends; WebRTC itself does not define it. |
| ICE, STUN and TURN | Find a working network route: direct where possible, through a relay when a firewall blocks it. |
| Codecs | Compress audio and video so a call fits an ordinary internet connection. |
| DTLS-SRTP | Encrypts the media on the way; WebRTC does not allow unencrypted media. |
Because these parts are standard, the same call can include a laptop in Chrome, a phone in Safari and a native app, as long as each one speaks WebRTC.
Why does WebRTC matter in a video call?
For the person you invite, WebRTC means a meeting link is enough. There is no installer to approve, no plugin to update and no account to create just to talk. For the person running the call, it means the quality depends mostly on the network and the device rather than on which software the guest happens to have.
Features follow what the browser supports: screen capture is a computer-browser feature, background effects rely on the browser's graphics features, and a strict network can block the routes WebRTC prefers. That is why good meeting services add a relay for blocked networks and adjust quality as conditions change.
How does Meeto use WebRTC?
Every Meeto call runs on WebRTC, in the browser and in the iPhone and Android apps. Guests open a link and join from the browser lobby without an account or a download; you can try it by creating a meeting link.
Each participant connects to Meeto's media servers, an SFU, and every call, one-to-one included, runs through them. Calls are encrypted in transit (WebRTC, DTLS-SRTP) through Meeto's media servers in Europe and South-East Asia. When a network blocks the usual routes, a TURN relay over TLS on port 443 carries the call.
A Free meeting holds up to 10 people for 60 minutes, counted once 2 or more people are in it; pricing shows what Pro and Team add.
Frequently asked questions
Is WebRTC a product I need to install?
No. WebRTC is a set of open web standards with a widely used open-source implementation, and it already ships inside Chrome, Edge, Firefox and Safari. Meeting services such as Meeto build their calls on top of it, so there is nothing separate to download.
Is a WebRTC call always peer to peer?
No. WebRTC can connect two devices directly, but many services route media through their own servers so that group calls, quality adjustments and recording work reliably. Meeto routes every call, including one-to-one calls, through its media servers.
Can a company network block WebRTC?
Yes. A firewall or VPN that blocks the UDP traffic real-time media prefers can stop a call from connecting even though the website loads. A TURN relay over port 443 usually gets through; the firewall troubleshooting guide lists what to check.
Does WebRTC work on a phone?
Yes. Current mobile browsers support WebRTC calls, so a guest can join a Meeto meeting from a phone browser, watch a shared screen and draw on it. To share a phone's own screen, use the Meeto app for iPhone or Android.
Try your first browser meeting
Create a room, share the link and invite someone to join you.