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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I think you will need to sort out the signaling first, as you&#8217;ll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>-Michael<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</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>On Behalf Of </b>Phillip Jones<br><b>Sent:</b> Friday, December 04, 2009 2:29 PM<br><b>To:</b> freeswitch-users@lists.freeswitch.org<br><b>Subject:</b> [Freeswitch-users] Bridging to a non SIP based system<o:p></o:p></span></p><p class=MsoNormal><o:p>&nbsp;</o:p></p><p class=MsoNormal>Hi All,<br><br>Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started.<br><br>What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port &amp; CODEC of the RTP stream. They use speex rtp sessions over TCP.<br><br>So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it.<br><br>The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port &amp; CODEC of the RTP stream.<br><br>But perhaps I missing a key bit of knowledge/understanding here.<br><br>I would be grateful for any advise here. <br><br>Thanks a lot,<br><br><br>Phil<o:p></o:p></p></div></body></html>