I am completely new here, so excuse my silly questions.
I came across RasPBX and installed it (current version is raspbx-10-10-2020) on my
RaspberryPi 3. OS (based on
Asterisk 16.18.0 and
FreePBX 22.214.171.124. I am located in Slovenia, Europe.
I am also using Huawei USB dongle and have set up trunk to mobile network via that dongle.
For clients I am using Zoiper 5 (on Linux, iOS and Android).
Another thing. I had problems with UDP - after 30 seconds (of established connection) call has been dropped. I suppose there has been a problem with SIP ALG, I switched it off on my router, but the problem persisted. So I switched to TCP connection and now everything works just great. However, I am not using TLS/SRTP/ZRTP encryption, but my RasPBX device and clients are operating within secure OpenVPN network. BTW - OpenVPN server is located somewhere else (in Slovenia), so VPN connection between OpenVPN server and RaspPBX and other clients is going through the internet.
Now the problem.
I have tested the incoming and outgoing call via USB dongle and first test were quite good. I am able to call from my GSM to RasPBX and the client can answer the call and sound is of a good quality. I can also call from a Zoiper client to the outside number and call quality is good.
So I decided for some more tests.
I have asked a colleague from USA, Ilinois for a test. Call from one Zoiper client to the other (through the VPN) is working very good. Sound quality is really good.
But then I asked a colleague to call my GSM through USB dongle.
In that case call quality was bad, it was really hard to understand what he was talking.
I am not an expert, but have a feeling this is codec related. This is the list of my codecs under
Asterisk SIP Settings:
In a logs, I can see lines like this:
2021-10-27 19:50:07] NOTICE[C-0000000b] translate.c: 2604 lost frame(s) 2605/0 ([email protected])->([email protected])
I assume that means
g722 is being used, I have also found out Asterisk is internally using
slin coded and then performs transcoding, but don’t know why or if this could affect the sound quality.
What is weird to me is that call from one PJSIP extension and the other has great sound quality, but if call goes through USB dongle, sound is much much worse.