How it is measured
Every call leg reports its own audio quality in RTCP. With signaling capture on (PBX settings → Signaling), AiPBX reads those reports every minute: packet loss and jitter per call leg, per trunk and per extension. Nothing on the PBX is changed.
In is what this PBX received, out what the other side received. A trunk losing audio in points at the carrier's side, out at ours (uplink, firewall, bandwidth); an extension losing audio out is often a phone on poor Wi-Fi.
A leg is poor when two of its reports in one direction reach the loss set below, fair from 1 %. Jitter is in RTP timestamp units (8 = 1 ms for G.711). A trunk or extension losing audio is a problem of the health check "Call quality": it shows in the alert bell and is e-mailed like the other alarms.
Signaling capture is off, so nothing is measured. Turn it on under PBX settings → Signaling →
| Leg | Calls | Poor | Fair | Loss in (avg / max) | Loss out (avg / max) | Jitter max (in / out) |
|---|---|---|---|---|---|---|
No measured call yet. | ||||||
| Time | Level | Leg | Media from | Loss in (max) | Loss out (max) | Reports in / out | Call-ID |
|---|---|---|---|---|---|---|---|
None: every measured call was good. | |||||||