<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px"><div id="yui_3_16_0_1_1430151272637_365754" dir="ltr"><span id="yui_3_16_0_1_1430151272637_366191"><font id="yui_3_16_0_1_1430151272637_366190"><font id="yui_3_16_0_1_1430151272637_366189">Thanks Sebastian,</font></font></span></div><div id="yui_3_16_0_1_1430151272637_365754" dir="ltr"><span><font><font><br></font></font></span></div><div id="yui_3_16_0_1_1430151272637_365754" dir="ltr"><span id="yui_3_16_0_1_1430151272637_366711"><font id="yui_3_16_0_1_1430151272637_366710"><font id="yui_3_16_0_1_1430151272637_366709"><font id="yui_3_16_0_1_1430151272637_366708"><font id="yui_3_16_0_1_1430151272637_366707">Can you give an example on how to use gst_rtp_buffer_map with respect to my pipeline code.</font></font></font></font></span></div><div id="yui_3_16_0_1_1430151272637_365754" dir="ltr"><br></div> <br><div class="qtdSeparateBR"><br><br></div><div class="yahoo_quoted" style="display: block;"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div dir="ltr"> <font size="2" face="Arial"> On Saturday, 2 May 2015, 9:02, Sebastian Dröge <sebastian@centricular.com> wrote:<br> </font> </div> <br><br> <div class="y_msg_container">On Do, 2015-04-30 at 17:39 +0000, Raheeb Muzaffar wrote:<br>> Hello,<br>> I am multicasting video stream using RTP (rtph264pay) and UDP (udpsink) from a source node. At the client end the video stream is decoded and Displayed As They stream. I have coded this using gstreamer in c / c ++. I now want to<br>> 1. At the source, capture the sequence no., Packet size (in bytes) and timestamp of each packet of the video stream being Transmitted over the network.<br>> 2. At receiver, capture the sequence no. And timestamp of the received packets so as to measure delay and packet loss.<br>> It would be great if someone Could help me suggest a way to implement this in C / C ++. I do not want to use GST debug since I have to do some processing with the video packets. Here is the code for the server and client.<br><br>Why don't you want to use GStreamer for that?<br><br>But in any case, if you don't you'll have to parse the RTP packets<br>yourself. See <a href="https://tools.ietf.org/html/rfc3550" target="_blank">https://tools.ietf.org/html/rfc3550 </a>section 5 for the<br>packet format.<br><br>If you use GStreamer, you could use the gst_rtp_buffer_map() function<br>from libgstrtp.<br><br>-- <br>Sebastian Dröge, Centricular Ltd · <a href="http://www.centricular.com/" target="_blank">http://www.centricular.com</a><br>_______________________________________________<br>gstreamer-devel mailing list<br><a ymailto="mailto:gstreamer-devel@lists.freedesktop.org" 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><br></div> </div> </div> </div></div></body></html>