Always Transcode from G729 to PCMA

[Skeelkat]

Hi Everyone,

I need some help, one of my Upstream SBC's has stopped accepting G.729. Whenever a call is transmitted sent to the SBC, I receive a 400 Bad Request right after the 183 Session Progress.

When I change the Codec priorities around on the phone to ensure that PCMA or PCMU is used first, the call is completed without any issues. So I am attempting to achieve that my User Agents connect to FusionPBX with G.729 and the FusionPBX connects to the Gateway using PCMA.

I have tried all the settings I can find and various combinations of them... no luck. Anyone here that has a working step-by-step for this?

Version	4.5.21
Git Information	Branch: master Commit: 313704745a1bd4f2186d3cf5ac3453577a473754 Origin: https://github.com/fusionpbx/fusionpbx Status: Your branch is up to date with 'origin/master'. +0 days ago
Project Path	/var/www/fusionpbx
Switch Version	1.10.5 (64bit)
PHP Version	7.3.19-1~deb10u1

 

[glennbtn]

Does it not work if you set it in Variables

outbound_codec_prefs PCMA,PCMU

 

[Skeelkat]

No.

I have added the values to the gateway configuration and in variables. I even hardcoded them in the Sofia profiles

No matter what I try I get no joy.

 

[Davesworld]

I believe that you cannot transcode g.729 since it is passthrough only as are all non-free codecs. The only benefit at all for using this codec is to save on bandwidth, it is inferior in every other way, even packet loss and it does add extra milliseconds to the latency. In my own setup, I go the opposite direction, I use g.722(HD Voice) between my sip phones and my PBX, even if the rest of the call is PCMU from the PBX on out, you still get the benefits of lost packet concealment from the one leg of g.722 even though it is narrowband audio but when hearing voice prompts, I am hearing the very lovely Allison Smith in HD telling me to put my password in or howto delete etc. Isristel, a CLEC in Canada, supports g.722. One person I know had to call a Canadian business who happened to use Isristel as their CLEC and he chose Iristel as his outgoing CLEC so the call never went through the PSTN or even outside the CLEC so it was g.722 end to end. You'd never get that through Bandwidth which all my incoming calls come through since they have my DID. I can choose any I want on outgoing but cannot choose which CLEC they come in from. Iristel does not port in US DIDs, believe me I asked.