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</o:shapelayout></xml><![endif]--></head><body lang=DE link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Thanks – I will have a deeper look and asking concrete questions afterword.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Thanks!<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>MatzeMuc86<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"'>Von:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> devel-bounces@lists.opensips.org [mailto:devel-bounces@lists.opensips.org] <b>Im Auftrag von </b>Dave Singer<br><b>Gesendet:</b> Freitag, 25. Februar 2011 18:26<br><b>An:</b> OpenSIPS devel mailling list<br><b>Cc:</b> MatzeMuc86<br><b>Betreff:</b> Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span class=apple-style-span><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>MatzeMuc86,</span></span><o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Opensips handles just the SIP signaling which contains the information about where the RTP should connect to. The modules media_proxy and nat_helper can be used to communicate to the external applications media_proxy and rtp_proxy respectively that setup proxying of the RTP and return to the opensips module the connection setup it has prepared for the RTP. The module then alters the SDP appropriately and the opensips script continues on deciding where to send and sending the SIP message to the next server.<o:p></o:p></p><div><p class=MsoNormal>I believe rtp_proxy can be setup to stream audio from a file and it might be a starting point for you to mix the stereo audio.<o:p></o:p></p></div><div><p class=MsoNormal>However I'm not sure if opensips can be the endpoint of a call (SIP) without writing or extending a module for that purpose. The B2B module is something you might look at to see how much tweaking it would take to make it do what you want.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'>Dave<o:p></o:p></p><div><p class=MsoNormal>On Fri, Feb 25, 2011 at 1:15 AM, MatzeMuc86 <<a href="mailto:matzemuc86@gmail.com">matzemuc86@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Hi,<br><br>as I already invested a lot of time, I try to be sure to check out all<br>possibilities. I already know about PJSIP but that does not mean that<br>OpenSIPS could not be the right project for me - I wanted to be sure.<br>Anyway: Thanks A LOT for your very nice support and searching time!!!<br><br>-----Ursprüngliche Nachricht-----<br>Von: <a href="mailto:devel-bounces@lists.opensips.org">devel-bounces@lists.opensips.org</a><br>[mailto:<a href="mailto:devel-bounces@lists.opensips.org">devel-bounces@lists.opensips.org</a>] Im Auftrag von Saúl Ibarra<br>Corretgé<br>Gesendet: Freitag, 25. Februar 2011 10:08<br>An: OpenSIPS devel mailling list<o:p></o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'>Betreff: Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server<o:p></o:p></p></div><div><div><p class=MsoNormal>Hi,<br><br>On 02/25/2011 10:02 AM, MatzeMuc86 wrote:<br>> Hello Adrian,<br>><br>> I know that I need some RTP part which receives, mixes and sends the<br>> media stream. I thought I can do all these things with OpenSIPS? I saw<br>> that SDP is implemented, but, of course, this is transported by SIP.<br>> Maybe I am wrong about RTP and OpenSIPS - sorry.<br>><br><br>SDP is just signaling, OpenSIPS doesn't deal with RTP at all.<br><br>> Refering to FreeSWITCH: IT can handle RTP, conferences, B2BUA eg. -<br>> but no stereo. TO implement this, the project is that big that - after<br>> talking with the freeswitch developers - this seems to be a very big<br>> project. As it is only a Bachelor Thesis I thought about finding an<br>> easier way to implement my idea.<br>><br><br>IIRC, PJSIP does have stereo support to some degree. A quick search returned<br>this: <a href="http://blog.pjsip.org/2008/03/31/doing-it-in-stereo/" target="_blank">http://blog.pjsip.org/2008/03/31/doing-it-in-stereo/</a><br><br><br>Regards,<br><br>--<br>Saúl Ibarra Corretgé<br>AG Projects<br><br>_______________________________________________<br>Devel mailing list<br><a href="mailto:Devel@lists.opensips.org">Devel@lists.opensips.org</a><br><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/devel" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/devel</a><br><br><br>_______________________________________________<br>Devel mailing list<br><a href="mailto:Devel@lists.opensips.org">Devel@lists.opensips.org</a><br><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/devel" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/devel</a><o:p></o:p></p></div></div></div><p class=MsoNormal><o:p> </o:p></p></div></div></div></body></html>