Hello,
I
have a problem with one of my sip trunks. The provider
uses g729 with a ptime of 40ms and all incoming calls that
are being recorded or go to the conference have a big
delay and choppy sound, like a mismatch of ptime.
I
have already tried to put rtp-autofix-timing=false into
the profiles and even changed the inbound-codec…. to
g729@40i, but nothing helps.
We
have done a test with the provider switching for a test
call to 20ms and the sound was perfect.
The
issue now is that the provider can’t keep this setting
only for us, he wants to send with 40ms.
Is
there a way to configure it on Freeswitch?
Thank
you
Anestis