2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS state from OFF to HANDSHAKE
2. not getting sip messages back from freeswitch ( we're using sip.js and not verto. This was all working (more or less) with 1.4)
thanks,
chris
On Fri, May 29, 2015 at 3:42 AM, Chris Mandra <mandra@gmail.com> wrote:Hi Ken, thanks for the reply.Does https://conference.freeswitch.org/verto/ run fs_video? I kind of thought it was running an older version.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.I have a few more questions:I'm getting the followingf messages I could use some clarification on: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 but this is rtcp-mux so no thanks (is this affecting anything?)2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video DTLS state from OFF to HANDSHAKE- it never gets beyond handshake - any ideas3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video packet 0 bytes (ice not ready @ line 4241!)
- ice not ready - what's causing this?2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for zrtp-hash
2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding whether to pass zrtp-hash between legs
2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hashShould we be using ZRTP? if it weren't enabled at all would I see these messages?
Lastly, it looks like we're not getting any sip responses from FS - Any ideas why?
All answers/help/hints are appreciated.
chris
On Thursday, May 28, 2015, Ken Rice <krice@freeswitch.org> wrote: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
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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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_________________________________________________________________________
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