<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">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">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><div><span style="color: rgb(48, 57, 66); font-family: 'Lucida Grande', sans-serif; ">Phil</span></div><div><span style="color: rgb(48, 57, 66); font-family: 'Lucida Grande', sans-serif; "><br></span></div><div><span style="color: rgb(48, 57, 66); font-family: 'Lucida Grande', sans-serif; "><br></span></div><div><br></div><div><br></div><div><br></div><div><font color="#303942" face="Lucida Grande, sans-serif"><br></font></div></body></html>