Hi, Thanks Matthew for trying to reproduce the problem, I appreciate your efforts very much.
There must be something off in my setup in one way or another. I could just discard this server and build a new one, but I think it's not good practice to leave a problem unsolved, so I'll continue trying to figure this out. One thing I noticed - don't know if it's relevant or not - due to a repo mismatch, I had problems with updating libgdiplus and libgdiplus-devel package, had to disable a repo and reinstall those and my mono installation (which is making me lose my hair). Is there a way to debug Asterisk itself? Or find the code that parses the outbound sdp? I figured there must be an if statement or more that determines whether or not to parse the ice lines into the sdp body. Finding that/those statements that produce the kind of sdp I'm seeing Asterisk send out, might tell something about what's wrong with my setup. As my c is not exactly fluent I wasn't sure which code files to search, can you guys help out with that? cheers, Olli 2014-10-03 11:31 GMT+03:00 Matthew Jordan <[email protected]>: > 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 >
-- _____________________________________________________________________ -- 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
