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

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