Agents are reporting ‘choppy’ audio on calls, but I don’t know how to diagnose the issue.
Is there a built-in diagnostic tool in GC that shows audio quality metrics per call?
Yes! Go to Performance > Interactions > select a conversation > Voice Quality tab.
The Voice Quality tab shows per-second metrics including: jitter (ms), packet loss (%), round-trip time (ms), and MOS score. Look for jitter spikes above 30ms or packet loss above 1% - these are the primary causes of choppy audio.
We also use chrome://webrtc-internals for real-time debugging during active calls.
The WebRTC internals page shows live RTP statistics including: packets sent/received, bytes transferred, NACK counts, and FEC recovery rates. If the agent opens this page while on a call, you can watch the audio stream health in real-time. Screenshot the graphs for troubleshooting tickets.