3CX Audio Transcode / Re-Invite

[xmicro]

Hello All,

I am attempting to allow our internal extensions to be able to converse on G722, while all outbound calls are on ulaw. Currently, this is functioning but with a significant drop in quality when going outbound since the PBX is transcoding from G722 to ulaw.

Is it possible to force the SIP phones to use G711 when dialing outbound? On a old Asterisk system I sent re-invites to the sip phones to renegotiate ulaw if it was an outbound call, is something similar possible with 3CX, like the PBX only offering ulaw to the SIP phone if it detects its an outbound call?

Googling around on these forums addresses the fact that the PBX wants to transcode, I was hoping for a more specific fix.

 

[AWS2P]

Why not doing it easier setting phones and Trunk to G711 U in first codec , set like this you don't have audio transcode so audio should be higher quality.
Have you already tried this ?

 

[JohnS_3CX]

Hi xmicro,

The PBX is the audio proxy effectively between the device and the trunk. You cannot avoid transcoding or tell the device to use a different codec here, because there will always be an internal call from the deskphone to the PBX, and then from the PBX to the trunk, merging the whole call.

Even if it was possible, it would only work on outgoing calls, incoming would still be transcoded as the PBX will eventually establish an internal call leg to the endpoint.

The best solution is to use the same codec across devices as your trunk as suggested by @aws2p , unless you must necessarily use G722 for specific reasons.