Network lab
Measured and configured in this browser only. Every RTT is a round trip: not one-way delay and not microphone-to-speaker latency. WebSocket/HTTP RTT is the signaling path (browser ↔ server) and says nothing about the media route; do not tune TURN or media from it.
Live
WebRTC peers
Settings
2-minute recording
Median / p95 of everything above. Works without a call (signaling only). Results are kept until reload; copy them to compare settings.
Details
Selected ICE candidate pair
transport.selectedCandidatePairId → localCandidateId / remoteCandidateId. RTT = STUN round trip on that pair.
Inbound audio
Interval values cover the last 2 s. Jitter-buffer delay = Δ jitterBufferDelay / Δ jitterBufferEmittedCount × 1000.
Video and bitrates
Invitations (owner)
The link below is shown once. Share it privately.