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</o:shapelayout></xml><![endif]--></head><body lang=SV link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;color:#1F497D'>If FS lowers the timestamp FS activates the mark bit, which should cause the client to be able to handle this correctly.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;color:#1F497D'>/Peter<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Från:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>För </b>Chris Martineau<br><b>Skickat:</b> den 21 december 2011 10:33<br><b>Till:</b> freeswitch-users@lists.freeswitch.org<br><b>Ämne:</b> [Freeswitch-users] rtp timestamp issue on pickup<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-GB>Hi,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>My specific application requires an incoming call to pickup an existing waiting call.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>The first call is terminated and parked.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>The second call rings in and uses intercept to then connect it to the waiting call.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>The problem I have is that the second call has a lower rtp timestamp than the waiting call causing the client codec to wait until it catches up?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>i.e client incoming rtp stream timestamp effectively looks like this ...<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>existing stream from freeswitch<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>37920<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>38080<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>38240<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>On intercept the timestamps then become that of the other calls stream<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>3040<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>3200<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>...<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>The client then doesn’t get any incoming voice until the new timestamps catch up!<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>Is there any way to set freeswitch to maintain the timestamps?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>Should the client allow for this?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>Any ideas would be greatly appreciated.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>Regards<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>Chris <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:12.0pt;font-family:"Times New Roman","serif"'>!DSPAM:4ef1a81732761796211860! <o:p></o:p></span></p></div></body></html>