We are using FreePBX 16.0.40
the codec is G722 (verified)
The quality of the sound is not that great.
We started to look around.
Ive noticed that when we make a single call the bandwidth of both RX and TX is 20kb/sec (in the live network usage)
I would expect to see it around 64kb.
Is there a reason for this? I didn’t find any place that limit the bandwidth. We have no issue with our network.
You should expect to see about 100kbps for G.722 for each direction on each leg. There is quite a lot of overhead on RTP.
How are you measuring bandwidth? At normal 20ms packet intervals, 20 kbps is probably totally unusable for SIP/RTP.
There is certainly no place that would limit bandwidth to such a low figure, unless your network was drastically overloaded, or you are trying to run VoIP over V.34 modem. In any case, the OS would not be setting the limit, and nor would Asterisk or FreePBX.
I’m reasonably certain that it is showing 20kB/sec, (twenty kilobytes), which is about right.
Do the quality issues affect internal calls, outgoing calls and/or incoming calls? If outside calls are affected, are they bad for the PBX user, the remote party, or both?
What kind(s) of IP phones, softphones or mobile SIP apps are you using?
If you record the call, is the recording also poor quality?
We monitor the call bandwidth using freepbx dashboard “live network usage” widget
The call recording sounds much better, almost perfect! We tried it using a softphone with a Jabra headset and a Fanvil 6XU. With the softphones, it sounded better than with the Fanvil. With the Fanvil, we got the feeling that some of the frequencies were being cut off.
However, with both options, we felt that there was less “data” than we expected. We had the Panasonic NS1000 for many years, and the sound was great. We were told not to expect such quality, but it still sounds odd to us
This is how it looks like:
That’s 168kbps downlink and 176kbs uplink. There doesn’t appear to be a bandwidth problem.
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