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<TITLE>Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.</TITLE>
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<FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>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...<BR>
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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<BR>
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On 7/8/13 11:59 AM, "Henry Huang" <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>Ken, <BR>
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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. <BR>
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But the test I did earlier was simply extension to extension calls. 2 chrome extensions registered on webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> 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?<BR>
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Thanks,<BR>
<BR>
Henry<BR>
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On Mon, Jul 8, 2013 at 9:39 AM, Ken Rice <<a href="krice@freeswitch.org">krice@freeswitch.org</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>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... <BR>
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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... <BR>
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On 7/8/13 11:24 AM, "Henry Huang" <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a> <<a href="http://red.rain.seven@gmail.com">http://red.rain.seven@gmail.com</a>> > wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><FONT SIZE="2"><SPAN STYLE='font-size:10pt'>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?<BR>
</SPAN></FONT></FONT><FONT SIZE="2"><SPAN STYLE='font-size:10pt'><FONT FACE="Verdana, Helvetica, Arial"><BR>
Thanks,<BR>
<BR>
Henry<BR>
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On Fri, Jul 5, 2013 at 3:15 PM, Michael Jerris <<a href="mike@jerris.com">mike@jerris.com</a> <<a href="http://mike@jerris.com">http://mike@jerris.com</a>> > wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>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.<BR>
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Mike<BR>
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On Jul 5, 2013, at 6:05 PM, Henry Huang <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a> <<a href="http://red.rain.seven@gmail.com">http://red.rain.seven@gmail.com</a>> > wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>I confirm that the changes made by Iwan worked and gave me audio. Thanks, Iwan. <BR>
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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?<BR>
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Thanks,<BR>
<BR>
Henry<BR>
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On Fri, Jul 5, 2013 at 2:27 PM, Iwan Budi Kusnanto <<a href="ibk@labhijau.net">ibk@labhijau.net</a> <<a href="http://ibk@labhijau.net">http://ibk@labhijau.net</a>> > wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Henry,<BR>
I can make it jssip demo works by modify this line<BR>
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this.peerConnection = new<BR>
JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);<BR>
<BR>
into<BR>
<BR>
constraints["optional"] = [];<BR>
constraints["optional"].push({'DtlsSrtpKeyAgreement': 'true'});<BR>
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this.peerConnection = new<BR>
JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);<BR>
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On Sat, Jul 6, 2013 at 1:12 AM, Anthony Minessale<BR>
<<a href="anthony.minessale@gmail.com">anthony.minessale@gmail.com</a> <<a href="http://anthony.minessale@gmail.com">http://anthony.minessale@gmail.com</a>> > wrote:<BR>
> Maybe because we hacked in dtls support?<BR>
><BR>
><BR>
><BR>
> On Fri, Jul 5, 2013 at 12:58 PM, Henry Huang <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a> <<a href="http://red.rain.seven@gmail.com">http://red.rain.seven@gmail.com</a>> ><BR>
> wrote:<BR>
>><BR>
>> I think I have found the issue. If I use the JsSIP website demo to<BR>
>> register to webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> <<a href="http://webrtc.freeswitch.org/">http://webrtc.freeswitch.org/</a>> then I will be able to replicate the no<BR>
>> audio issue on webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> <<a href="http://webrtc.freeswitch.org/">http://webrtc.freeswitch.org/</a>> <BR>
>><BR>
>> After testing out different scenarios, it appears to be only when the<BR>
>> destination client is the webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> <<a href="http://webrtc.freeswitch.org/">http://webrtc.freeswitch.org/</a>> version of JsSIP there will<BR>
>> be audio. Neither sipml5 demo or JsSIP demo website registered with<BR>
>> webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> <<a href="http://webrtc.freeswitch.org/">http://webrtc.freeswitch.org/</a>> can generate audio. Something in the SDP maybe?<BR>
>><BR>
>> Thanks,<BR>
>><BR>
>> Henry<BR>
>><BR>
</SPAN></FONT></BLOCKQUOTE></BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'><BR>
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</FONT></SPAN></FONT></FONT><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>-- <BR>
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