[Freeswitch-users] Significant audio clipping when bridging webrtc to sip

[Joegen Baclor]

Hi,

I am using FreeSwitch 1.4-beta to bridge WebRTC->SIP.  All is working 
great except for the fact that audio is almost always clipped at the 
beginning of the call.  Looking at the logs, it seems to me that this is 
due to the delay incurred when performing the DTLS handshake on receipt 
of media.  I am doing a "pre_answer" before bridging.  My question is, 
is it possible to do the DTLS handshake prior to originate and right 
after pre_answer?  Log is in pastebin:  
http://pastebin.freeswitch.org/22037.  The dropped media happens during 
the gap between lines 310 and 311.  Thanks for any pointers.

Joegen