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

Fri May 29 18:32:55 MSD 2015

yes still defaults to off, but I think we still do pass through even when
it's built without it

On Friday, May 29, 2015, Chris Mandra <mandra at gmail.com> wrote:

> 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
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>
>>
>>
>> On Thursday, May 28, 2015, Chris Mandra <mandra at gmail.com
>> <javascript:_e(%7B%7D,'cvml','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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>
>
>
> --
> mandra
> c:410.258.5281
>
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