On Thu, Oct 2, 2014 at 10:18 AM, Olli Heiskanen <[email protected]> wrote: > Hi, > > Thanks Eric for your reply, yes I know Asterisk replaces the sdp, however it > should create ice lines when calling to a webrtc client, which it is > currently not doing. > > To recap my problem (check previous messages for details); I have 2 webrtc > clients (sip.js on chrome) with realtime information that appears to be > correct. When calling from A to B, INVITE coming to Asterisk contains > correct sdp, but when the INVITE leaves Asterisk, the sdp lacks ice lines. >
Unfortunately, I can't reproduce this. We've been running a lot of tests with a variety of SIP clients over the past week here at SIPit - both with and without ICE - and I haven't had a single instance of Asterisk failing to provide any ICE candidates when it is properly configured to do so. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
