Technical overview
How Random Voice Chat works
Random Voice Chat combines a browser interface, a small signaling service, and WebRTC audio to connect two people for a voice-only call.
1. Microphone permission
The browser asks for microphone access before a call can start. The app needs a local audio stream so it can attach your microphone to the WebRTC connection.
2. Matchmaking
When you press Start, the web app opens a WebSocket connection to the signaling service. The service places available callers in a queue and pairs two compatible waiting clients.
3. WebRTC signaling
After a match, the two browsers exchange WebRTC offers, answers, and ICE candidates through the signaling server. The server coordinates setup; it is not the normal audio path.
4. Audio path
When possible, audio flows directly peer to peer. If a network blocks the direct path, TURN relay fallback can carry encrypted media so the call still has a chance to connect.
FAQ
Does the signaling server carry the audio?
Normally no. The signaling server helps browsers exchange connection details; WebRTC audio then tries to flow peer to peer.
Why does WebRTC need TURN fallback?
Some networks block direct peer-to-peer paths. TURN can relay encrypted media when a direct path is not available.
Why does the browser ask for microphone permission?
The browser needs permission before it can attach your microphone stream to a WebRTC voice call.