WebRTC Stats Viewer
Inspect live WebRTC getStats() metrics in your browser. View bitrate, packets, bytes, frames, codec, candidate pair, RTT and copy the raw stats JSON instantly.
Live getStats() metrics
Stopped| Codec | — |
|---|---|
| Candidate pair | — |
| RTT | — |
Raw stats snapshot
Start the test to inspect RTCPeerConnection.getStats() output.
Inspect RTCPeerConnection.getStats() online
This viewer creates a local WebRTC media connection and polls the browser's RTCPeerConnection.getStats() report once per second. It surfaces useful outbound RTP and candidate-pair information while keeping a raw JSON snapshot available for debugging.
Bitrate uses byte deltas over time
The displayed bitrate is calculated from the change in bytesSent between samples divided by elapsed time. Packet and byte totals remain cumulative counters. This distinction matters because cumulative bytes alone are not a rate.
Use stats with related WebRTC diagnostics
Check browser codec capabilities before debugging negotiated media, inspect an offer or answer with the SDP parser, and use the RTCDataChannel test when you want a non-media local peer-connection diagnostic.
WebRTC stats FAQ
Where do these WebRTC stats come from?
The viewer calls RTCPeerConnection.getStats() on a local WebRTC peer connection and reads browser-reported outbound RTP, codec and candidate-pair statistics.
How is bitrate calculated?
Bitrate is derived from the change in bytes sent between polling intervals. The tool does not label cumulative bytes as bitrate.
Is the RTT shown here an Internet latency test?
No. This first version uses a local browser loopback, so candidate-pair RTT is a browser-level diagnostic rather than a realistic Internet or geographically remote RTT measurement.