Looking for advice on interpreting WebRTC performance metrics in the Performance dashboard. Our EU-FR region shows a 400ms latency spike during peak hours, specifically affecting the ‘Premium Support’ queue. The softphone logs indicate packet loss, but agent adherence reports remain stable. Is there a specific view to correlate network jitter with conversation detail data? We need to determine if this is a regional infrastructure issue or a configuration error within the routing strategy.
This is caused by network jitter affecting the WebRTC handshake, not the recording pipeline. For legal discovery, ensure the exported transcripts include the network_quality metadata tag to document the degraded session. The Recording API separates media health from transcript integrity.
Note: Verify S3 bucket permissions before bulk exporting these high-latency sessions.
The documentation actually says… that while network jitter is the culprit, the configuration on the trunk side often gets overlooked. Since i’m primarily an IVR designer, i don’t usually dig into the telephony plumbing, but here is what worked for a similar containment issue we faced with a client.
- Check the trunk settings in Admin > Telephony.
- Look for the “Secure Pause” indicator. It’s not enabled by default.
- Enable it if it’s off. This sometimes forces a renegotiation of the media stream that clears up the jitter.
It’s a weird fix, but it worked. The logs showed the same packet loss, but once the secure pause tone was forced, the latency dropped back to normal levels. Not sure why, but it’s worth a shot. Don’t forget to check your S3 permissions if you’re exporting those transcripts, as As noted above. We lost a whole day because of a permission error. Just a heads up.
You need to stop guessing and hit /api/v2/analytics/queues/realtime instead of staring at the dashboard. It gives you the actual networkQuality metrics in real-time, not some lagged aggregate.
Note: make sure your OAuth token has analytics:view scope or it’ll 403 you instantly.
Make sure you check the WebRTC stats object directly in the client logs rather than relying on the dashboard aggregates. the latency spikes in EU-FR are usually tied to how the SDK handles the ICE candidate gathering when the TURN servers are overloaded. i’ve seen this exact 400ms jump when the fallback path kicks in. here’s how to capture the actual jitter values in the session log:
const stats = await peerConnection.getStats();
stats.forEach(report => {
if (report.type === 'inbound-rtp') {
console.log('jitter:', report.jitter);
console.log('packetsLost:', report.packetsLost);
}
});
if the jitter is high but packet loss is low, it’s likely a routing issue within the region, not your trunk config. also, check if your agents are on mobile data with aggressive battery saving modes. that kills background WebRTC sockets faster than you’d think.
- ICE candidate gathering timeout
- TURN server connectivity
- Mobile background app refresh settings
- Network interface switching events