Home / Blog / How to Build a Real-Time WebRTC Video Chat
Building a real-time video chat might sound like high-level wizardry, but thanks to a technology called WebRTC (Web Real-Time Communication), it is more accessible than ever.
The first step is getting permission. Your application needs to ask the user for access to their hardware. Modern browsers have a built-in "handshake" that allows the website to grab your local video and audio stream. At this stage, you’re essentially just showing the user their own face in a video box on the screen.
WebRTC is "peer-to-peer," meaning data travels directly between callers. However, two computers on opposite sides of the world don't just know how to find each other. You need a Signaling Server. Think of this as a middleman or a receptionist. It doesn't handle the video itself
Most computers are hidden behind firewalls or routers (NAT). To get through these barriers, WebRTC uses two types of "helper" servers. STUN Servers & TURN Servers
Once the "handshake" is complete and the path is found, the signaling server steps aside. The video and audio data now flow directly between the two browsers. This is why WebRTC is so fast—there is no central server slowing down the data or "watching" the call, which also makes it incredibly secure and private.
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