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

[Michael Jerris]

On Thursday, May 28, 2015, 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.
>

This is always running current code daily.  We always dogfood our bleeding
edge code.


> 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?)
>
>
Your using chrome canary I think.  This is a bug in canary.  Our latest
code addresses this chrome bug.


> 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
>

Look at packet captures, maybe firewall blocking things?

> 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?
>

This is because you don't have a dtls handshake yet.

> 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?
>

Is one leg of your call passing in zrtp stuff ?  You would not be running
zrtp on a webrtc call as no webrtc end points I know of support it.

> Lastly, it looks like we're not getting any sip responses from FS - Any
> ideas why?
>
>
> Maybe firewall, maybe misconfigured ip or acl?  If you are using verto,
where does sip come in to play?




> All answers/help/hints are appreciated.
>
> chris
>
>
>
>
> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','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>
>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
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>>
>>
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