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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’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> </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> </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> </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> </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 & 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 & 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>