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

[Henry Huang]

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> 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> 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>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> 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>
>> > wrote:
>> >>
>> >> I think I have found the issue. If I use the JsSIP website demo to
>> >> register to webrtc.freeswitch.org then I will be able to replicate
>> the no
>> >> audio issue on webrtc.freeswitch.org
>> >>
>> >> After testing out different scenarios, it appears to be only when the
>> >> destination client is the webrtc.freeswitch.org version of JsSIP
>> there will
>> >> be audio. Neither sipml5 demo or JsSIP demo website registered with
>> >> webrtc.freeswitch.org can generate audio. Something in the SDP maybe?
>> >>
>> >> Thanks,
>> >>
>> >> Henry
>> >>
>>
>
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