[Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.

[Ken Rice]

Yes its up to the browsers at that point to make the video work properly...
Now if you are seeing something like that, I would try firefox 22 (the
latest at this time) and see if you get the same reaction... If so I would
investigate NAT issues, altho, webrtc requires full ICE support...

Also you can try having both extensions call the mcu or mcu-proxy extensions
to see how the video acts there... The mcu-proxy extension has a media proxy
in play to try and help with nat issues


On 7/8/13 11:59 AM, "Henry Huang" <red.rain.seven at gmail.com> wrote:

> Ken, 
> 
> I will update the wiki after this. And I am not sure how I missed all the
> talks about video. I do remember about the part with conference though. 
> 
> But the test I did earlier was simply extension to extension calls. 2 chrome
> extensions registered on webrtc.freeswitch.org <http://webrtc.freeswitch.org>
> dialing from one to another. The video on callee's screen will be freezed
> while the video on caller's screen works fine. In this case, I suppose it
> should be passthrough and it's all up to the browser to make both side work?
> 
> Thanks,
> 
> Henry
> 
> 
> On Mon, Jul 8, 2013 at 9:39 AM, Ken Rice <krice at freeswitch.org> wrote:
>> Wow this has been asked and answered about 27 times already... Yes Video is
>> supported, no video transcoding is not supported... Now as far as video
>> freezing/doing  other strange things, this is going to depend on what the
>> other end is doing along with various other network issues...
>> 
>> Example: if you are calling into mod_conference to do a multi-party video
>> conference with VP8 its not going to work all that well as VP8 doesn’t have
>> keyframes that are sent on a regular basis as with other codecs like
>> H264/H263 and there is currently limited to no support in mod_conference to
>> automattically handle swapping from participant A to participant B, mixed
>> with the fact that all video on FreeSWITCH is passthru only, there is no
>> “brandy bunch” screens etc...
>> 
>> 
>> 
>> On 7/8/13 11:24 AM, "Henry Huang" <red.rain.seven at gmail.com
>> <http://red.rain.seven@gmail.com> > wrote:
>> 
>>> Is video call currently supported for WebRTC? My experience with the demo
>>> site is that after a few seconds , the video frame freezes while audio
>>> continues to work. Is this the expected behavior for now?
>>> 
>>> Thanks,
>>> 
>>> Henry​
>>> 
>>> 
>>> On Fri, Jul 5, 2013 at 3:15 PM, Michael Jerris <mike at jerris.com
>>> <http://mike@jerris.com> > wrote:
>>>> It is the right thing to do to turn on dtls.  As far as how it should be
>>>> handled in jsssip, you will have to talk to them about that.  I don't think
>>>> we plan at this point to support webrtc without dtls as everything I have
>>>> seen says it will be required by the browsers at some point anyways.
>>>> 
>>>> Mike
>>>> 
>>>> On Jul 5, 2013, at 6:05 PM, Henry Huang <red.rain.seven at gmail.com
>>>> <http://red.rain.seven@gmail.com> > wrote:
>>>> 
>>>>> I confirm that the changes made by Iwan worked and gave me audio. Thanks,
>>>>> Iwan. 
>>>>> 
>>>>> But now the question is that is it a hack on the js side or is it the
>>>>> right thing to do? And is it going to be merged into the JsSIP core?
>>>>> 
>>>>> Thanks,
>>>>> 
>>>>> Henry
>>>>> 
>>>>> 
>>>>> On Fri, Jul 5, 2013 at 2:27 PM, Iwan Budi Kusnanto <ibk at labhijau.net
>>>>> <http://ibk@labhijau.net> > wrote:
>>>>>> Henry,
>>>>>> I can make it jssip demo works by modify this line
>>>>>> 
>>>>>>   this.peerConnection = new
>>>>>> JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);
>>>>>> 
>>>>>> into
>>>>>> 
>>>>>> constraints["optional"] = [];
>>>>>> constraints["optional"].push({'DtlsSrtpKeyAgreement': 'true'});
>>>>>> 
>>>>>> this.peerConnection = new
>>>>>> JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);
>>>>>> 
>>>>>> On Sat, Jul 6, 2013 at 1:12 AM, Anthony Minessale
>>>>>> <anthony.minessale at gmail.com <http://anthony.minessale@gmail.com> >
>>>>>> wrote:
>>>>>>> > Maybe because we hacked in dtls support?
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> > On Fri, Jul 5, 2013 at 12:58 PM, Henry Huang <red.rain.seven at gmail.com
>>>>>>> <http://red.rain.seven@gmail.com> >
>>>>>>> > wrote:
>>>>>>>> >>
>>>>>>>> >> I think I have found the issue. If I use the JsSIP website demo to
>>>>>>>> >> register to webrtc.freeswitch.org <http://webrtc.freeswitch.org>
>>>>>>>> <http://webrtc.freeswitch.org/>  then I will be able to replicate the
no
>>>>>>>> >> audio issue on webrtc.freeswitch.org <http://webrtc.freeswitch.org>
>>>>>>>> <http://webrtc.freeswitch.org/>
>>>>>>>> >>
>>>>>>>> >> After testing out different scenarios, it appears to be only when
>>>>>>>> the
>>>>>>>> >> destination client is the webrtc.freeswitch.org
>>>>>>>> <http://webrtc.freeswitch.org>  <http://webrtc.freeswitch.org/>
>>>>>>>>  version of JsSIP there will
>>>>>>>> >> be audio. Neither sipml5 demo or JsSIP demo website registered with
>>>>>>>> >> webrtc.freeswitch.org <http://webrtc.freeswitch.org>
>>>>>>>> <http://webrtc.freeswitch.org/>  can generate audio. Something in the
>>>>>>>> SDP maybe?
>>>>>>>> >>
>>>>>>>> >> Thanks,
>>>>>>>> >>
>>>>>>>> >> Henry
>>>>>>>> >>
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <http://consulting@freeswitch.org>
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://FreeSWITCH-users@lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://consulting@freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users@lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130708/7a66a20a/attachment-0001.html