<html><head></head><body><div style="color:#000; background-color:#fff; font-family:Helvetica Neue-Light, Helvetica Neue Light, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px"><div><span>Dear Group,</span></div><div><span></span><br></div><div><span></span><br></div><div id="yui_3_16_0_1_1453637888981_3153"><span></span><br></div> <div class="qtdSeparateBR"><br><br></div><div class="yahoo_quoted" style="display: block;"> <div style="font-family: Helvetica Neue-Light, Helvetica Neue Light, 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 face="Arial" size="2"> On Friday, 15 January 2016 6:39 AM, Venkatesh Adiga <rv_adiga@yahoo.com> wrote:<br></font></div> <br><br> <div class="y_msg_container"><div id="yiv0194497470"><div><div style="color: rgb(0, 0, 0); font-family: Helvetica Neue-Light, Helvetica Neue Light, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px; background-color: rgb(255, 255, 255);"><div id="yiv0194497470yui_3_16_0_1_1452819782545_2661">Hi,</div><div id="yiv0194497470yui_3_16_0_1_1452819782545_2661"><br></div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">I am working on a Gstreamer based(for Audio Streaming in RTP Session) Media Gateway Development project. </div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">This is my first project that I got involved in Gstreamer based project, and have less knowledge on this domain.</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">Below is the scenario of a Q.SIG to SIP call.</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">Using the deskphone, the Caller A calls callee B's mobile. As per call signalling is concerned, </div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">the call is established between both the parties and RTP ports are activated at Media Gateway. </div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">RTP ports are opened and verified by using netstat command on a Linux box. The issue that I am observing is that </div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">oneway audio was seen from callee to caller.</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">As per wireshark trace, I am observing that Media Gateway performs the RTP packet transfer to individual call owner, but not forwarding from Caller or Callee or visa versa.</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr"><br class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2720"></div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">User A <---> MG <----> User B - Treating User B and User A as seperate user in a call.</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr"><br class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2724"></div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">when user A speaks, Media Gateway should have done like the below.</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr"> User A ----->MG</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr"> ------> User B</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr"><br class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2730"></div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">Could anyone throw some light on how to analyze the RTP issue in the above scenario of Gstreamer?</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">What debug tools are available to verify Gstreamer pipeline in live call scenario?</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">Thank you </div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr"><br class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2736"></div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">Regards</div><div class="yiv0194497470" id="yiv0194497470yui_3_16_0_1_1452819782545_2661" dir="ltr">Venkatesh</div></div></div></div><br><br></div> </div> </div> </div></div></body></html>