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<TITLE>Re: [Freeswitch-users] WEBRTC was working with 1.4, having no luck with fs_video</TITLE>
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<FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Conference.freeswitch.org has been running this code for a while and its used quite heavily. We use it every week for the past month or so for video with ClueCon Weekly. <BR>
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The point of paid assistance is you said you have to have this working now. The best way to get it working right now is to get paid support.<BR>
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Each of the following things could be any number of issues.<BR>
</SPAN></FONT><OL><LI><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>did you follow the setup documentation on Debian Jessie to the letter?
</SPAN></FONT><LI><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Are you using the Latest Chrome? (FireFox may work but no guarentees, IE and Safari don’t even support WebRTC)
</SPAN></FONT><LI><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>theres not enough information to even start saying anything about FS not responding to SIP messages... You’ll have to review the logs and configs to figure out why
</SPAN></FONT><LI><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>As far as ZRTP goes do you need it? We cant answer that, only you can... If you don’t need it you can disable it<BR>
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On 5/28/15, 8:42 PM, "Chris Mandra" <<a href="mandra@gmail.com">mandra@gmail.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Hi Ken, thanks for the reply. <BR>
Does <a href="https://conference.freeswitch.org/verto/">https://conference.freeswitch.org/verto/</a> run fs_video? I kind of thought it was running an older version. <BR>
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I'm hoping to get some help from the community bc, well, I'm probably not the only person with these problems and, like many people, I can't really afford paid assistance. <BR>
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I have a few more questions: <BR>
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I'm getting the followingf messages I could use some clarification on:<BR>
1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by candidate to set remote rtcp audio addr to 205.215.241.76:59980 <<a href="http://205.215.241.76:59980">http://205.215.241.76:59980</a>> but this is rtcp-mux so no thanks (is this affecting anything?)<BR>
<BR>
2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video DTLS state from OFF to HANDSHAKE <BR>
- it never gets beyond handshake - any ideas<BR>
3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video packet 0 bytes (ice not ready @ line 4241!)<BR>
- ice not ready - what's causing this?<BR>
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2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for zrtp-hash<BR>
2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding whether to pass zrtp-hash between legs<BR>
2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash<BR>
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Should we be using ZRTP? if it weren't enabled at all would I see these messages?<BR>
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Lastly, it looks like we're not getting any sip responses from FS - Any ideas why?<BR>
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All answers/help/hints are appreciated.<BR>
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chris<BR>
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On Thursday, May 28, 2015, Ken Rice <<a href="krice@freeswitch.org">krice@freeswitch.org</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Hi Chris,<BR>
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There is always <a href="https://conference.freeswitch.org/verto/">https://conference.freeswitch.org/verto/</a> where we have been demoing this for quite a while<BR>
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If you need this working like now I would suggest emailing <a href="consulting@freeswitch.org">consulting@freeswitch.org</a> <<a href="http://consulting@freeswitch.org">http://consulting@freeswitch.org</a>> where you can get paid realtime assistance from one of the Core Dev Team<BR>
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See you @ ClueCon!<BR>
Ken<BR>
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On 5/28/15, 5:20 PM, "Chris Mandra" <<a href="mandra@gmail.com">mandra@gmail.com</a> <<a href="http://mandra@gmail.com">http://mandra@gmail.com</a>> > wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>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 <<a href="http://apprtc.appspot.com">http://apprtc.appspot.com</a>> <<a href="http://apprtc.appspot.com">http://apprtc.appspot.com</a>> demo.)<BR>
I have followed the instructions as well I am able on <a href="https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video">https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video</a><BR>
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<BR>
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a few questions: <BR>
1. Is there any known problem using wildcard certs from godaddy?<BR>
2. is there other configuration beyond what's on the web page needed to make webrtc work? (I have the apropos ports open etc) <BR>
3. Are there any known issues with fs_video and sip.js?<BR>
4. I feel like it's getting hung op on dtls, as it never passes <BR>
"2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS state from OFF to HANDSHAKE"<BR>
(it repeats that twice with:<BR>
"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?<BR>
5. is there a working webrtc / fs_video demo I can look at?<BR>
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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. <BR>
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Thanks for all you do,<BR>
chris<BR>
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</SPAN></FONT></FONT><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>-- <BR>
Ken<BR>
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