Have you tried to put the queue before decoding, and limit buffer-size?<br>What are your pipelinies currently?<br><div class="gmail_quote">2010/8/17 Nitin Das <span dir="ltr"><<a href="mailto:nitin.162@gmail.com">nitin.162@gmail.com</a>></span><br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><span style="font-family: arial,sans-serif; font-size: 13px; border-collapse: collapse;">Hello,<div> <br></div><div> i would like to use gstreamer pipeline for video conferencing. Currently i m using v4l2src for camera capture and alsasrc for audio capture then encode to h264 and aac respectively and mux into mpegts and send it over rtp. But this process puts delay of around 3-4 seconds and doesn't give the real feeling of video conferencing system. Is there any combination of pipeline , plugins available so that the delay would be in some milliseconds.</div> <div><br></div><font color="#888888"><div>--nitin</div></font></span> <br>------------------------------------------------------------------------------<br> This SF.net email is sponsored by<br> <br> Make an app they can't live without<br> Enter the BlackBerry Developer Challenge<br> <a href="http://p.sf.net/sfu/RIM-dev2dev" target="_blank">http://p.sf.net/sfu/RIM-dev2dev</a> <br>_______________________________________________<br> gstreamer-devel mailing list<br> <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a><br> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a><br> <br></blockquote></div><br><br clear="all"><br>-- <br>Your Sincerely<br>Michael Joachimiak<br>