[Freeswitch-users] WEBRTC was working with 1.4, having no luck with fs_video

[Chris Mandra]

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
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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