Poor gateway audio quality

[nufay]

I set up a few local and external sip endpoints and they work fine.
Then i tried to configure PSTN gateway (i'm using openvox, it's in the same network) and audio quality for remote side is very bad, voice sounds more metal, but local extension sound is good.

endpoints codec - 722,alaw,ulaw
same codec settings is used for gateways

Why remote sound quality is that bad in this scenario?
if any configuration is needed to determine the reason why - ask.

 

[Dan]

What cell network is your OpenVox connected to, and what model OpenVox is it? Its likely the OpenVox is connecting to the cellular network and using the GSM codec (or AMR), then transcoding to G722. Transcoding can severely degrade audio quality, especially with older transcoding software like what is in the OpenVox.

Try GSM end to end!

 

[nufay]

What cell network is your OpenVox connected to, and what model OpenVox is it? Its likely the OpenVox is connecting to the cellular network and using the GSM codec (or AMR), then transcoding to G722. Transcoding can severely degrade audio quality, especially with older transcoding software like what is in the OpenVox.

Try GSM end to end!

I currently use my Openvox with Asterisk and all works perfectly, i wanted the same with Fusionpbx.
Actually Openvox is 3G and transcoding to G722 in both cases.
I'm using VS-GWM420W.

 

[Dan]

I currently use my Openvox with Asterisk and all works perfectly, i wanted the same with Fusionpbx.

What phones are you using with your Asterisk box? Have you tried pass-through media on FusionPBX (since everything is in the local network)?

 

[nufay]

I'm using Digium phones (D65) in local network and Bria softphone for remote phone connection.
No, i have not tried pass-through media.
Where can i set it up?

 

[Dan]

No, i have not tried pass-through media.
Where can i set it up?

Take a look at inbound-bypass-media on ReadTheDocs & the Freeswitch Confluence if setting SIP Bypass Media on the extensions doesn't work: https://docs.fusionpbx.com/en/latest/accounts/extensions.html

 

[nufay]

None of this option solves poor audio quality problem for me.

 

[Dan]

Is FusionPBX transcoding the call before handing it off to your Digium D65?

 

[nufay]

Is FusionPBX transcoding the call before handing it off to your Digium D65?

both of them configured to use 722 codec
how can i check if there's any transcoding?

 

[DigitalDaz]

Sngrep will give you the whole story of a call.

 

[nufay]

Sngrep will give you the whole story of a call.

I assume, that i need to check this section of configuration?
What exact part of sngrep output do i need to check to understand if there were any transcoding?

v=0
o=- 258567464 258567464 IN IP4 192.168.0.24
s=digphn
b=AS:84
t=0 0
a=X-nat:0
m=audio 4014 RTP/AVP 107 9 8 0 111 96
c=IN IP4 192.168.0.24
b=TIAS:64000
a=rtcp:4015 IN IP4 192.168.0.24
a=sendrecv
a=rtpmap:107 opus/48000/2
a=fmtp:107 cbr=1;maxptime=20;maxplaybackrate=16000;maxaveragebitrate=20000;sprop-maxcapturerate=
000;usedtx=0
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

 

[nufay]

Sngrep will give you the whole story of a call.

What exact part of sngrep output do i need to check to understand if there were any transcoding?

 

[DigitalDaz]

The sdp in the invite and the 200ok, see if they have matching codecs.

 

[DigitalDaz]

Actually.....

Look in the fs_cli log of t a call, you should see the match happening both from the phone side and the remote side.

 

[nufay]

Actually.....

Look in the fs_cli log of t a call, you should see the match happening both from the phone side and the remote side.

ok, i'll try to check it today if i can, otherwise i can only do it tomorrow.