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

[Michael Jerris]

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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