[Freeswitch-users] FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone?

[Alan Frisch]

Hi,

Just tinkering around with FreeSWITCH pretty much using the default
configuration, with some customization to connect my Polycom 430 and
my ITSP.  The problem I am having is that the first few seconds of
calls (or early media if available) have no audio.

When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP
with timestamps, I can see that FreeSwitch is holding onto the 183 and
subsequent 200 OK for about 2 seconds before relaying it onto the
Polycom.  FWIW, server is an Intel Atom 330.

Any idea why this is... I am sure it must be a configuration issue
somewhere?  I did not have this issue with Asterisk, so I am hoping to
troubleshoot it and move on with FreeSWITCH

Thanks.

AF.