We had to pull a call the other day, which led me to realize that while our side sounds great, the other party’s voice is being turned into a garbled mess. I reviewed a few other recordings, and it’s the same across the board.
I then tried switching the recording format from gsm to wav…while it’s clearer, same issue. It’s definitely not just poor connection quality, as they sound fine on the phone, but you can’t even really make out what they’re saying in the recording.
Where might I start to find a solution?
Try using the CLI command “rtcp set debug on” and looking at jitter statistics. The phone may be applying a large jitter buffer to compensate for network problems.
Thank you. How do I check jitter stats once the debug flag is on?
If I’ve got the right command, they appear in the full log.
Does this help?
I would not be surprised if it is related to the desktop client we use, Bria. It’s never been great for us unfortunately, but we haven’t found anything better since eliminating our desk phones (closed the office).
Appreciate your help so far!
Sangoma has a great Desktop soft phone.
I would totally go for it, but it looks like we’re being acquired and probably being switched to something else in the near future (they use Teams I believe).
But if that doesn’t happen, does the software work with an existing FreePBX distro install?
You only have one sample, from one call leg, of inbound jitter. That is very low., but might be an outlier.
Actually, the first one contains statistics on inbound traffic, as well, also low.
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