<div dir="ltr">Its a matter of understanding how vp8 transitions are handled. The codec is very picky about the data and probably changing the source does causes some errors. There are some rtcp signals to manipulate the video that we may need to add.<div>
<br></div><div>When it freezes try sending the command uuid_video_refresh <uuid> on the leg who is sending the video and see if it has any effect.</div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Fri, Jun 28, 2013 at 10:12 AM, Phil Dunks <span dir="ltr"><<a href="mailto:phil.dunks@netdev.co.uk" target="_blank">phil.dunks@netdev.co.uk</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">Hi FS Users<div><br><div>I made a simple web application using sipML5, which connects directly to 1.4.0+git~20130623T182400Z~2f08e40fce, and goes straight into a conference.</div><div>The audio works fine, but am experiencing problems with the video freezing or not being present.</div>
<div>I tried out <a href="https://webrtc.freeswitch.org/webrtc/portal.html" target="_blank">https://webrtc.freeswitch.org/webrtc/portal.html</a> </div></div><div>Enabled video, and dialed 3300 for the local 48K conference.</div>
<div>Seeing the same problem there. Audio fine. Video fine for first member, but goes wrong when second person joins.</div><div><span style="color:rgb(48,57,66);font-family:'Lucida Grande',sans-serif">Should this work for me, or is the webrtc conference video not quite ready yet?</span></div>
<div><span style="color:rgb(48,57,66);font-family:'Lucida Grande',sans-serif">Thanks, </span></div><span class="HOEnZb"><font color="#888888"><div><span style="color:rgb(48,57,66);font-family:'Lucida Grande',sans-serif">Phil</span></div>
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