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
__________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org