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

[Anthony Minessale]

The best way to make video work on webrtc atm is bypass_media=true, then
you have the native browsers talking to each other.



On Mon, Jul 8, 2013 at 12:52 PM, Ken Rice <krice at freeswitch.org> wrote:

>  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
> >>
>
>
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>
> --
> Ken
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>


-- 
Anthony Minessale II

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