What calling does
The calling subsystem supports voice and video sessions through WebRTC. The application includes signalling, incoming-call handling, accept/decline, hang-up and media controls. Browser microphone and camera permissions remain essential.
What WebRTC provides
WebRTC is the browser platform for real-time audio and video. Chat256 adds an application signalling layer to exchange the information required to establish sessions. A call can have working signalling and still fail to establish a media path.
Why TURN matters
NATs, carrier networks and firewalls can prevent direct browser connectivity. TURN provides a relay path when direct ICE candidates fail. TURN configuration supports relay connectivity when direct browser paths are unavailable, helping calls work across restrictive networks.
Calling security
| Layer | Observed design |
|---|---|
| Signalling | Authenticated, encrypted signalling envelopes with sender verification. |
| Session setup | ICE with STUN/TURN support. |
| Media | WebRTC encrypted media transport. |
| Controls | Call rate limits, TTL cleanup and browser media controls. |
| Network traversal | ICE, STUN and TURN support for real-world connectivity. |
Troubleshooting
- Check microphone/camera permissions.
- Test two separate devices.
- Test Wi-Fi and mobile networks.
- Check ICE candidates and TURN reachability.
- Capture the exact browser error before changing server security controls.
What happens when a call starts
- Your browser requests microphone and/or camera permission.
- Chat256 signalling authenticates the participants and exchanges the information needed to establish the call.
- WebRTC performs connectivity checks using ICE.
- STUN can help discover a viable network path; TURN can relay traffic when direct connectivity is not possible.
- The browsers establish encrypted WebRTC media transport.
The exact path depends on network topology, browser support and whether direct peer connectivity succeeds.
Call controls and privacy
Users can control microphone and camera access from the browser and operating system. Closing a call stops the active media session, but it cannot undo information another participant already heard, recorded or saved.
Why a call may fail
Common causes include blocked UDP traffic, restrictive corporate or mobile networks, denied browser permissions, stale browser versions, unavailable TURN capacity and firewall policies. If direct connectivity fails, a correctly configured TURN service can provide a relay path. See the private calling guide for a practical checklist.