Hi Brian, I thought as much, WebRTC isn’t something we are trying to use at the moment (although im sure we’ll find a use for it in the not too distant future.) Invites are created using the bridge application in XML dialplan. I have SIP pcaps but I don’t have and Freeswitch traces at the moment as the issue is only appearing once in say 500 calls on our production system so it can be a little awkward to pin down detailed tracing. I’m working on getting an example today and will post back as soon as possible. Is there any way to disable WebRTC entirely? That could be worth a try whilst I get a test setup for this scenario. From: Brian West <brian@freeswitch.org<mailto:brian@freeswitch.org>> Date: Mon, Mar 2, 2015 at 6:15 PM Subject: Re: [Freeswitch-users] ICE SDP Params To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> This is all related to WebRTC, how are you creating the invite? Logs would be helpful./b