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

[Chris Mandra]

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 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 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 #freeswitch
> Twitter: @FreeSWITCH
>          @ClueCon
>
>
> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at 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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> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
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