<div dir="ltr"><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">Ken, </div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">
<br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">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. </div>
<div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">But the test I did earlier was simply extension to extension calls. 2 chrome extensions registered on <a href="http://webrtc.freeswitch.org">webrtc.freeswitch.org</a> dialing from one to another. The video on callee&#39;s screen will be freezed while the video on caller&#39;s screen works fine. In this case, I suppose it should be passthrough and it&#39;s all up to the browser to make both side work?</div>
<div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">Thanks,</div>
<div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">Henry</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Jul 8, 2013 at 9:39 AM, Ken Rice <span dir="ltr">&lt;<a href="mailto:krice@freeswitch.org" target="_blank">krice@freeswitch.org</a>&gt;</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">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>
<div><div class="h5">
<br>
<br>
<br>
On 7/8/13 11:24 AM, &quot;Henry Huang&quot; &lt;<a href="http://red.rain.seven@gmail.com" target="_blank">red.rain.seven@gmail.com</a>&gt; wrote:<br>
<br>
</div></div></span></font><blockquote><div><div class="h5"><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><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
<br>
On Fri, Jul 5, 2013 at 3:15 PM, Michael Jerris &lt;<a href="http://mike@jerris.com" target="_blank">mike@jerris.com</a>&gt; wrote:<br>
</span></font></div></div><blockquote><div><div class="h5"><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&#39;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 &lt;<a href="http://red.rain.seven@gmail.com" target="_blank">red.rain.seven@gmail.com</a>&gt; wrote:<br>
<br>
</span></font></div></div><blockquote><div><div class="h5"><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 &lt;<a href="http://ibk@labhijau.net" target="_blank">ibk@labhijau.net</a>&gt; wrote:<br>
</span></font></div></div><blockquote><font face="Monaco, Courier New"><span style="font-size:11pt"><div><div class="h5">Henry,<br>
I can make it jssip demo works by modify this line<br>
<br>
  this.peerConnection = new<br>
JsSIP.WebRTC.RTCPeerConnection({&#39;iceServers&#39;: servers}, constraints);<br>
<br>
into<br>
<br>
constraints[&quot;optional&quot;] = [];<br>
constraints[&quot;optional&quot;].push({&#39;DtlsSrtpKeyAgreement&#39;: &#39;true&#39;});<br>
<br>
this.peerConnection = new<br>
JsSIP.WebRTC.RTCPeerConnection({&#39;iceServers&#39;: servers}, constraints);<br>
<br>
On Sat, Jul 6, 2013 at 1:12 AM, Anthony Minessale<br>
&lt;<a href="http://anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>&gt; wrote:<br>
&gt; Maybe because we hacked in dtls support?<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Fri, Jul 5, 2013 at 12:58 PM, Henry Huang &lt;<a href="http://red.rain.seven@gmail.com" target="_blank">red.rain.seven@gmail.com</a>&gt;<br>
&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; I think I have found the issue. If I use the JsSIP website demo to<br></div></div>
&gt;&gt; register to <a href="http://webrtc.freeswitch.org" target="_blank">webrtc.freeswitch.org</a> &lt;<a href="http://webrtc.freeswitch.org/" target="_blank">http://webrtc.freeswitch.org/</a>&gt;  then I will be able to replicate the no<br>

&gt;&gt; audio issue on <a href="http://webrtc.freeswitch.org" target="_blank">webrtc.freeswitch.org</a> &lt;<a href="http://webrtc.freeswitch.org/" target="_blank">http://webrtc.freeswitch.org/</a>&gt; <br>
&gt;&gt;<br>
&gt;&gt; After testing out different scenarios, it appears to be only when the<br>
&gt;&gt; destination client is the <a href="http://webrtc.freeswitch.org" target="_blank">webrtc.freeswitch.org</a> &lt;<a href="http://webrtc.freeswitch.org/" target="_blank">http://webrtc.freeswitch.org/</a>&gt;  version of JsSIP there will<br>

&gt;&gt; be audio. Neither sipml5 demo or JsSIP demo website registered with<br>
&gt;&gt; <a href="http://webrtc.freeswitch.org" target="_blank">webrtc.freeswitch.org</a> &lt;<a href="http://webrtc.freeswitch.org/" target="_blank">http://webrtc.freeswitch.org/</a>&gt;  can generate audio. Something in the SDP maybe?<br>

&gt;&gt;<br>
&gt;&gt; Thanks,<br>
&gt;&gt;<br>
&gt;&gt; Henry<br>
&gt;&gt;<br>
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</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt">-- <br>
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