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Browser Diagnostics

WebRTC Test

Real-time communication readiness

WebRTC Test

Real-time communication capabilities

not supported
🔗
RTCPeerConnection

P2P connection

Missing
🎤
getUserMedia

Media access

Missing
📹
MediaDevices

Device enumeration

Missing
📡
Data Channel

Binary data transfer

Missing

What we check

We verify getUserMedia, RTCPeerConnection, data channels, ICE candidate gathering, and device permissions so you know whether video calls will connect without turn-server fallbacks. The report highlights blocked camera/mic access or missing codecs that commonly break Google Meet and Teams.

If permissions are denied, clear them and reload. Network-level firewalls that block UDP/3478 can still cause one-way audio; in that case try a wired network or enable TURN on your service.

Need a fix fast?

Our WebRTC troubleshooting guide covers camera, microphone, and network steps.

FAQ

Why does the test pass but calls still fail?
Many apps also need STUN/TURN access. If UDP is blocked, media may fail even though the APIs work. Check corporate firewalls or switch networks.
Do I need H.264 for WebRTC?
Most modern services use VP8/VP9/AV1, but some enterprise setups require H.264. Use our Codec Test if a service insists on H.264.
Why is my camera listed as 'in use'?
Another tab/app may be locking the device. Close other video apps and restart the browser, then re-run the test.