<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 id="yui_3_16_0_1_1452819782545_2661">Hi,</div><div id="yui_3_16_0_1_1452819782545_2661"><br></div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">I am working on a Gstreamer based(for Audio Streaming in RTP Session) Media Gateway Development project. </div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">This is my first project that I got involved in Gstreamer based project, and have less knowledge on this domain.</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">Below is the scenario of a Q.SIG to SIP call.</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">Using the deskphone, the Caller A calls callee B's mobile. As per call signalling is concerned, </div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">the call is established between both the parties and RTP ports are activated at Media Gateway. </div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">RTP ports are opened and verified by using netstat command on a Linux box. The issue that I am observing is that </div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">oneway audio was seen from callee to caller.</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">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 id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class=""><br id="yui_3_16_0_1_1452819782545_2720" class=""></div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">User A <---> MG <----> User B - Treating User B and User A as seperate user in a call.</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class=""><br id="yui_3_16_0_1_1452819782545_2724" class=""></div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">when user A speaks, Media Gateway should have done like the below.</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class=""> User A ----->MG</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class=""> ------> User B</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class=""><br id="yui_3_16_0_1_1452819782545_2730" class=""></div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">Could anyone throw some light on how to analyze the RTP issue in the above scenario of Gstreamer?</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">What debug tools are available to verify Gstreamer pipeline in live call scenario?</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">Thank you </div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class=""><br id="yui_3_16_0_1_1452819782545_2736" class=""></div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">Regards</div><div id="yui_3_16_0_1_1452819782545_2661" dir="ltr" class="">Venkatesh</div></div></body></html>