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@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@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@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@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@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@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@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@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 >> >>
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