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

Fri May 29 18:19:09 MSD 2015

Thanks for the thoughtful response Ken. Answers inline.

On Thu, May 28, 2015 at 10:21 PM, Ken Rice <krice at freeswitch.org> wrote:

>  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.
>
> 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.
>
> Each of the following things could be any number of issues.
>
>    1. did you follow the setup documentation on Debian Jessie to the
>    letter?
>
> yup

>
>    1.
>    2. Are you using the Latest Chrome? (FireFox may work but  no
>    guarentees, IE and Safari don’t even support WebRTC)
>
> yup

>
>    1.
>    2. 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
>
> ok - I'm going to try using the FS verto demo to troubleshoot and see if
my problems are with sip.js

>
>    1.
>    2. 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
>
> Prolly not. It has to be explicitly compiled into the build, right?

>
>
>
> On 5/28/15, 8:42 PM, "Chris Mandra" <mandra at 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 <
> http://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 ideas
> 3. 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-hash
>
> Should 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 at 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 at freeswitch.org <http://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.ClueCon.com> http://www.freeswitch.org
> <http://www.freeswitch.org> http://www.OSTAG.org <http://www.OSTAG.org> *
> irc.freenode.net <http://irc.freenode.net>  #freeswitch
> Twitter: @FreeSWITCH
>          @ClueCon
>
>
> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com <
> http://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>  <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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> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
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>
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-- 
mandra
c:410.258.5281
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