I have a FreePBX 188.8.131.52/Asterisk 13.18.4 system that’s been in production for almost a year now. Overall I hear few complaints about audio quality. Our PSTN connectivity is through a SIP provider. I’ve had maybe one complaint a week, and we consistently average 4-5 concurrent calls throughout most of the workday. We have around 75 phones.
One user said this morning when he called his airline, the agent on the remote end said that his audio was coming in and out. When he notified me of this, I placed a SIP debug on his extension and asked him to call back in. After being on the call for about 16 minutes, he said the remote agent said that his audio was fading in and out.
When I look at the MOS type stats, the call quality looks good. I’ll paste those below.
AverageLatency = 23.620 # in whole milliseconds
Jitter = 0.053 # in whole milliseconds
TxPacketLoss = 6.0 # in packet counts
RxPacketLoss = 0.0 # in packet counts
TxPackets = 48744.0 # in packet counts
RxPackets = 48740.0 # in packet counts
I am pretty sure that one-sided audio where one party hears total silence is typically due to a codec mismatch or NAT issues. When it’s call quality then typically packet loss, excessive latency, jitter overruns, etc. can be the culprit. But I don’t see this according to the stats.
If I provide the debug logs, can one of y’all experts perhaps take a peek and let me know your opinion? Perhaps it’s a remote side issue, since I’m coming up blank on my end so far at least!