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@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@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@gmail.com> wrote:
Maybe because we hacked in dtls support?
On Fri, Jul 5, 2013 at 12:58 PM, Henry Huang <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 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