1. 2. Offline play of video file is it fine? if not Please check you are able to dump exact data what you actually streamed. May be your are lossing info before decoding. Regards, Sandeep.Yedire ---------------------------------------------------------- <div class="gmail_quote">2009/10/22 Jyoti <<a href="mailto:jyoti.d@allaboutif.com">jyoti.d@allaboutif.com</a>> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> Hi All, I am trying implement a multicast UDP Server Client module using gstreamer. The video plays very bad. Could someone please give some suggestions on the same? The server and client are built using pipelines shown below: Server: gst-launch -v gstrtpbin name=rtpbin filesrc location=~/workdir/filesys/opt/data/collateral.ts ! mpegtsdemux name=d d. ! queue ! rtph264pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5000 host=127.0.0.1 ts-offset=0 rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=127.0.0.1 sync=false async=false udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 d. ! queue ! rtpmpapay ! rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink port=5002 host=127.0.0.1 ts-offset=0 rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 <div> Client: gst-launch gstrtpbin name=rtpbin latency=200 udpsrc caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, profile-level-id=(string)4d400d, payload=(int)96, clock-base=(guint)3013157687, seqnum-base=(guint)28981" port=5000 name=vrtpsrc ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! videoscale ! queue ! xvimagesink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=127.0.0.1 sync=false async=false udpsrc caps="application/x-rtp, media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA, ssrc=(guint)316404369, payload=(int)96, clock-base=(guint)810575426, seqnum-base=(guint)9183" port=5002 name=artpsrc ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpmpadepay ! mad ! audioconvert ! audioresample ! queue ! alsasink udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=127.0.0.1 sync=false async=false Thanks, Jyoti </div> ------------------------------------------------------------------------------ Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! <a href="http://p.sf.net/sfu/devconference" target="_blank">http://p.sf.net/sfu/devconference</a> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> </blockquote></div>