Hi Chris, There is always https://conference.freeswitch.org/verto/ where we have been demoing this for quite a while If you need this working like now I would suggest emailing consulting@freeswitch.org where you can get paid realtime assistance from one of the Core Dev Team See you @ ClueCon! Ken http://www.ClueCon.com http://www.freeswitch.org http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon On 5/28/15, 5:20 PM, "Chris Mandra" <mandra@gmail.com> wrote:
Hi guys. I had been able to use webrtc via chrome with FS 1.4 for connections and confenerce calls without incident (although the bandwidth and video quality seemed rather low as compared to the apprtc.appspot.com <http://apprtc.appspot.com> demo.) I have followed the instructions as well I am able on https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video but so far I can't make an actual connection; the certs pass the cert test without issue, but I have no joy using webrtc and fs_video
a few questions: 1. Is there any known problem using wildcard certs from godaddy? 2. is there other configuration beyond what's on the web page needed to make webrtc work? (I have the apropos ports open etc) 3. Are there any known issues with fs_video and sip.js? 4. I feel like it's getting hung op on dtls, as it never passes "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS state from OFF to HANDSHAKE" (it repeats that twice with: "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video packet 0 bytes (ice not ready @ line 4241!)" not long after - does that speak to something obvious to you guys? 5. is there a working webrtc / fs_video demo I can look at?
I realize this may all be solved by the release of 1.6, but I need this working now. I've been looking forward to this bc I know there have been some issues with webrtc, rtcp, freeswitch and solid BWE.
Thanks for all you do, chris
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