I have a problem with outgoing call voice quality.
People who call tell me that my voice seems to fade away and come back.
Incoming call voice quality is OK.
This occurs on long distance call - I have tested it with calling from a local land-line phone to my FreePBX and then it works OK.
I have following:
FreePBx 5.211.65-16
Codecs alaw,ulaw
Sip phone: Zoipher Android (same codecs as FreePBX)
What kind of problem is it:
Codecs ?
FreePBX Settings ?
Sip Trunk from my carrier ?
Sip Phone ?
I have not been able to solve this problem - my users are very annoyed by the bad voice quality.
Appreciate if someone with deep knowledge in FreePBX could help - I am loosing clients & business
VoIP will not vary in amplitude, so you need to get a clearer description of what is actually wrong with the audio.
If it really is fading, the problem is probably the phone, as the only common analogue part of the system. Most other problems are network related, although similar problems can be caused by using VMs.
You almost certainly need to get a network capture (e.g. use tcpdump and then analyze with wireshark) to locate where in the network things are going wrong. Listen to the captured audio, and also check for high levels of jitter, or packet loss.
If it’s truly “fading away” by getting softer and then louder again, then the problem is not VOIP related. VOIP issues will cause you to come and go, pop in and out, or sound like you’re underwater. Audio volume problems are likely caused by your microphone or PC.
Try an actual phone and see if the problem goes away.
Thank you for the answers.
It is good to know that it probably is not a VoIP issue - that will eliminate many error sources.
I have tried two separate sip phones and the same problem (incoming calls OK / outgoing calls fade).
My sip phones are Android mobiles running Zoiper 1.18.5 app.
As I understand from your answers one problem could be in the Zoiper software or ?
Can different codecs have something to do with fading calls - I have alaw&ulaw (in that order) in PBX and the sip phones.
Don’t have experience in wireshark so I really don’t know what to look for in the tcpdump.
Will read the documentation.
Actual fading happens in the analogue part of the phone. It is not a software problem (and that includes codecs). I think it much more likely that your quality problem is not fading, but, in that case you need to characterise it properly.
What do you recommend to characterise it properly ?
Is wireshark the way to got - I understand that the audio stream can be played back as RTP captured packages.