<div dir="ltr"><div><div><div><div>It just seems like there are way too many pieces in your pipeline to me. There are several things that you need to do.<br><br></div>1. What OS are you using? That is important.<br></div>2. You can turn on a lot of debug tracing. That helps you find where and why it fails. Search on 'gstreamer debugging' and 'gstinfo'.<br>
</div><div>3. Upgrade to the 1.x GStreamer if you can. The online docs default to the current release and it is different than the 0.10 code. Here is a link to the 0.10.14 doc.<br> <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/manual/html/index.html">http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/manual/html/index.html</a><br>
</div>4. Here is a Linux script that I use for this testing. I think that the payload should be 33 (which you have). Please note that this does not handle audio.<br><br>#!/bin/sh<br><br>gst-launch udpsrc port=$1 ! 'application/x-rtp,media=video,payload=96,clock-rate=90000,encoding-name=H264'\<br>
! rtph264depay byte-stream=true ! video/x-h264 ! ffdec_h264 \<br> ! ffmpegcolorspace ! xvimagesink sync=false<br><br></div>The '$1' is the first parameter to the script which is the port. This should be a good start. You can expand from here. You can also use wireshark (or tcpdump or tshark) to capture and then examine the packets generated to verify that they are in the format that you expect.<br>
<div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Jul 23, 2014 at 6:46 AM, Ravikumar B <span dir="ltr"><<a href="mailto:ravikumar.embed@gmail.com" target="_blank">ravikumar.embed@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div><div><div><div>Dear experts.,<br><br></div> I am new to the gstreamer. i am trying to test / play video file using RTP on client side.<br>
<br></div> Following setup i have done on my Host machine:<br>
<br></div> Server:<br></div> Run RTP streaming using VLC player.<br><br></div> Client:<br></div> How to test RTP on client side.. <br> <br></div><div> I am using following pipeline on client side. for testing on RTP.<br>
<b> gst-launch-0.10 -v gstrtpbin name=rtpbin latency=2000 rtcp-sync=2 buffer-mode=0 udpsrc caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, payload=(int)33, seqnum-base=(int)49316, clock-base=(int)470688351, encoder-name=(string)MP2T-ES " auto-multicast=1 port=1234 buffer-size=524288 do-timestamp=true blocksize=8192 ! rtpbin.recv_rtp_sink_0 rtpbin. ! typefind ! rtpmp2tdepay ! mpegtsdemux name=demux demux. ! queue2 ! mpegvideoparse ! ffdec_mpeg2video ! ffmpegcolorspace qos=true ! videoscale ! autovideosink demux. ! queue2 ! ffdec_ac3 ! autoaudiosink<br>
</b></div><div> but this pipeline is not working><br></div><div> <br></div> Can you please suggest me how can i make pipeline or test application for testing RTP.<span class="HOEnZb"><font color="#888888"><br>
<div><div><div><div><div><div><div> <br clear="all"><div><div><div><br>-- <br>...with regards<br> ravikumar
</div></div></div></div></div></div></div></div></div></div></font></span></div>
<br>_______________________________________________<br>
gstreamer-devel mailing list<br>
<a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a><br>
<a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a><br>
<br></blockquote></div><br></div>