<div dir="ltr">The best way to make video work on webrtc atm is bypass_media=true, then you have the native browsers talking to each other.<div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Jul 8, 2013 at 12:52 PM, Ken Rice <span dir="ltr"><<a href="mailto:krice@freeswitch.org" target="_blank">krice@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<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>
<br>
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<div class="im"><br>
<br>
<br>
On 7/8/13 11:59 AM, "Henry Huang" <<a href="http://red.rain.seven@gmail.com" target="_blank">red.rain.seven@gmail.com</a>> wrote:<br>
<br>
</div></span></font><blockquote><font><font face="Verdana, Helvetica, Arial"><span style="font-size:10pt"><div class="im">Ken, <br>
<br>
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>
<br></div>
But the test I did earlier was simply extension to extension calls. 2 chrome extensions registered on <a href="http://webrtc.freeswitch.org" target="_blank">webrtc.freeswitch.org</a> <<a href="http://webrtc.freeswitch.org" target="_blank">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>
<br>
Thanks,<br>
<br>
Henry<br>
</span></font></font><div class="im"><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
<br>
On Mon, Jul 8, 2013 at 9:39 AM, Ken Rice <<a href="http://krice@freeswitch.org" target="_blank">krice@freeswitch.org</a>> wrote:<br>
</span></font></div><blockquote><font face="Monaco, Courier New"><span style="font-size:11pt"><div class="im">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>
<br>
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>
<br>
<br>
<br></div><div class="im">
On 7/8/13 11:24 AM, "Henry Huang" <<a href="http://red.rain.seven@gmail.com" target="_blank">red.rain.seven@gmail.com</a> <<a href="http://red.rain.seven@gmail.com" target="_blank">http://red.rain.seven@gmail.com</a>> > wrote:<br>
<br>
</div></span></font><blockquote><div class="im"><font face="Monaco, Courier New"><font><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><span style="font-size:10pt"><font face="Verdana, Helvetica, Arial"><br>
Thanks,<br>
<br>
Henry<br>
</font></span></font></div><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
<br>
On Fri, Jul 5, 2013 at 3:15 PM, Michael Jerris <<a href="http://mike@jerris.com" target="_blank">mike@jerris.com</a> <<a href="http://mike@jerris.com" target="_blank">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>
<br>
Mike<br>
<br>
On Jul 5, 2013, at 6:05 PM, Henry Huang <<a href="http://red.rain.seven@gmail.com" target="_blank">red.rain.seven@gmail.com</a> <<a href="http://red.rain.seven@gmail.com" target="_blank">http://red.rain.seven@gmail.com</a>> > wrote:<br>
<br>
</span></font><blockquote><font><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>
<br>
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>
<br>
Thanks,<br>
<br>
Henry<br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
<br>
On Fri, Jul 5, 2013 at 2:27 PM, Iwan Budi Kusnanto <<a href="http://ibk@labhijau.net" target="_blank">ibk@labhijau.net</a> <<a href="http://ibk@labhijau.net" target="_blank">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>
<br>
this.peerConnection = new<br>
JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);<br>
<br>
into<br>
<br>
constraints["optional"] = [];<br>
constraints["optional"].push({'DtlsSrtpKeyAgreement': 'true'});<br>
<br>
this.peerConnection = new<br>
JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);<br>
<br>
On Sat, Jul 6, 2013 at 1:12 AM, Anthony Minessale<br>
<<a href="http://anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a> <<a href="http://anthony.minessale@gmail.com" target="_blank">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="http://red.rain.seven@gmail.com" target="_blank">red.rain.seven@gmail.com</a> <<a href="http://red.rain.seven@gmail.com" target="_blank">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 <a href="http://webrtc.freeswitch.org" target="_blank">webrtc.freeswitch.org</a> <<a href="http://webrtc.freeswitch.org" target="_blank">http://webrtc.freeswitch.org</a>> <<a href="http://webrtc.freeswitch.org/" target="_blank">http://webrtc.freeswitch.org/</a>> then I will be able to replicate the no<br>
>> audio issue on <a href="http://webrtc.freeswitch.org" target="_blank">webrtc.freeswitch.org</a> <<a href="http://webrtc.freeswitch.org" target="_blank">http://webrtc.freeswitch.org</a>> <<a href="http://webrtc.freeswitch.org/" target="_blank">http://webrtc.freeswitch.org/</a>> <br>
>><br>
>> After testing out different scenarios, it appears to be only when the<br>
>> destination client is the <a href="http://webrtc.freeswitch.org" target="_blank">webrtc.freeswitch.org</a> <<a href="http://webrtc.freeswitch.org" target="_blank">http://webrtc.freeswitch.org</a>> <<a href="http://webrtc.freeswitch.org/" target="_blank">http://webrtc.freeswitch.org/</a>> version of JsSIP there will<br>
>> be audio. Neither sipml5 demo or JsSIP demo website registered with<br>
>> <a href="http://webrtc.freeswitch.org" target="_blank">webrtc.freeswitch.org</a> <<a href="http://webrtc.freeswitch.org" target="_blank">http://webrtc.freeswitch.org</a>> <<a href="http://webrtc.freeswitch.org/" target="_blank">http://webrtc.freeswitch.org/</a>> can generate audio. Something in the SDP maybe?<br>
>><br>
>> Thanks,<br>
>><br>
>> Henry<br>
>><br>
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