What Is WebRTC? The Technology Behind Browser Calls, Explained Simply

If you have ever joined a video call by clicking a link, with no download, you have used WebRTC. It is one of the quiet success stories of the web: a technology that lets any browser or app send live video, voice and data straight to another one. It rarely gets mentioned by name, yet it sits underneath much of what people do online together. Here is what it is, in plain words.

The name

WebRTC stands for Web Real-Time Communication. "Real-time" is the key: unlike a web page, which loads and then sits still, a call is a continuous stream where delays of even half a second feel wrong. WebRTC is designed for that: low delay, constant flow, and the ability to adapt as the network changes.

What it can send

• Audio: your voice, with echo and noise handling built in.
• Video: your camera, or your screen when you share it.
• Data: any information at all, such as chat messages, files or the mouse movements in a remote session.

That third item matters a lot. Because WebRTC can carry arbitrary data directly between two devices, it powers peer-to-peer file transfer as well as calls.

How it works in three steps

1. Introduction. The two devices first need to find each other. They contact a server (called a signalling server), which passes small messages between them: "here is how to reach me".
2. Connection. Using that information, the devices try to open a direct path through their routers and firewalls. If that is impossible, they use a relay. See STUN, TURN and ICE explained.
3. Streaming. Once connected, audio, video and data flow directly between the two devices, encrypted.

After the introduction, the server is no longer in the path of the media in the direct case. That is the reason peer-to-peer calls can be fast and private.

Why it is built into browsers

Before WebRTC, a call in a browser needed a plug-in or an installed program. Browser makers agreed on a standard instead, so every modern browser can capture a camera, handle audio and open a direct connection. This is why a guest can join a PeerFlow meeting from a link, on a laptop or a phone, with nothing to install.

Is it secure?

WebRTC makes encryption mandatory. Media is protected with standard encryption between the two ends, and data channels use encryption as well. Browsers also insist on your permission before a page can use the camera or microphone, and show an indicator while they are in use. That does not make every app built on it safe, since an app can still misuse what you allow, but the foundation is sound. Read the security overview for how PeerFlow uses it.

What WebRTC does not do for you

• It does not provide the introduction server. Every app needs its own.
• It does not store anything. If you want a recording, an app has to make one.
• It does not solve identity. The app decides who a person is.
• It does not guarantee a direct path. Strict networks need a relay.

Where you meet it

Limits and trade-offs

In a group of many people, direct connections between everyone would be too heavy, so group meetings often route the streams through a media server that forwards them. PeerFlow meetings do this for groups of up to 15. One-to-one calls, file transfers and remote sessions go directly between the devices whenever the network allows.

Why it matters to you

For an everyday user, the point is simple: you can have a private, direct, high-quality conversation without installing special software, and without every second of it passing through someone's data centre. For someone building a product, it removes a mountain of work. For privacy, it offers a design that does not need to store your conversations.

Learn more

If you want to know how devices get through routers, read STUN, TURN and ICE explained simply. For the practical side, see how PeerFlow keeps data and what it does not, and install the Android and Windows apps.