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

[Chris Mandra]

Thanks for this Mike. I'm going to test with verto (using sip.js currently.)
other responses inline.

On Thu, May 28, 2015 at 10:29 PM, Michael Jerris <mike at jerris.com> wrote:

>
>
> 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.
>
> good to know. thanks!

>
>> 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.
>

great.

>
>
>> 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?
>
I'll check.

> 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.
>
makes sense

> 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.
>
I don't think so, but I see messages abut it. I've seen docs mention it has
to be turned on in the build - I haven't done that. Does it default to off
still?

> 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?
>
> using sip.js and not verto. will try the fs demo with verto.

>
>
>
>> 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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>>>
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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
>>> Twitter: @FreeSWITCH
>>>
>>>
> _________________________________________________________________________
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>
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-- 
mandra
c:410.258.5281
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