WebRTC Test
Real-time communication readiness
WebRTC Test
Real-time communication capabilities
P2P connection
Media access
Device enumeration
Binary data transfer
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.