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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=DE link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi Răzvan,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Ok, thanks for your answer! <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Unfortunately we are offering „early media“ to our customers (call center, radio station, and other companies) and lots of them like to play a free-of-charge announcement in the beginning. But if we started to get cheated, maybe we need to go for this workaround.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>But apart from that: Mostly the SDP is NOT repeated in the 200OK. Can I call rtpproxy_answer() when receiving the 200OK anyway?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Thanks and best regards<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> Marco<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";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 lang=EN-GB style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>Von:</span></b><span lang=EN-GB style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> users-bounces@lists.opensips.org [mailto:users-bounces@lists.opensips.org] <b>Im Auftrag von </b>Razvan Crainea<br><b>Gesendet:</b> Donners</span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>tag, 22. Januar 2015 09:36<br><b>An:</b> users@lists.opensips.org<br><b>Betreff:</b> Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><tt><span style='font-size:10.0pt'>Hi, Marco!</span></tt><span style='font-size:10.0pt;font-family:"Courier New"'><br><br><tt>From RTPProxy point of view, you can't differentiate between SIP replies, because for all of them you call the same function - rtpproxy_answer().</tt><br><tt>Now, if the client decides to send RTP for 183 (and indeed, I've seen this several times), there's not that much that you can do. Although it's kind of a hack, all I can think of is to not call rtpproxy_answer() for 180/183 and strip the body to prevent the client from sending RTP directly to the callee.</tt><br><tt>I hope this works for you.</tt><br><br><tt>Best regards,</tt><br><br></span><o:p></o:p></p><pre>Răzvan Crainea<o:p></o:p></pre><pre>OpenSIPS Solutions<o:p></o:p></pre><pre><a href="http://www.opensips-solutions.com">www.opensips-solutions.com</a><o:p></o:p></pre><p class=MsoNormal>On 01/21/2015 04:07 PM, Marco Hierl wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Dear all,</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>first of all I need to apologize that I was not able to find information about this issue although I’m sure that I’m not the first one complaining!</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The caller is sending an INVITE via OpenSIPS and rtpproxy_offer() is executed, callee answers with REPLY 180 or REPLY 183 (with SDP) and rtpproxy_answer() is made. In this status it should be ok that the rtp stream from callee to caller is transferred via the rtpproxy (e.g. for announcements), but I can see that rtp stream from caller to callee is transferred too!!! This means that there can be a conversation without receiving the 200OK and what is the real problem: that means (at least for me) they can talk to each other without any charging !! A timer will stop the conversion after the a while, but this can take time.</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>How can I overcome this problem? How can prevent RTP to be send to the callee before REPLY 200 is received?</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I can’t find any help in the RTPproxy protocol <a href="http://www.b2bua.org/wiki/RTPproxy/Protocol">http://www.b2bua.org/wiki/RTPproxy/Protocol</a>, nor in the rtpproxy module description in OpenSIPS.</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Thanks for your ideas, and best regards</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> Marco</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div></div><p class=MsoNormal><br><br><br><o:p></o:p></p><pre>_______________________________________________<o:p></o:p></pre><pre>Users mailing list<o:p></o:p></pre><pre><a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><o:p></o:p></pre><pre><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><o:p></o:p></pre></blockquote><p class=MsoNormal><o:p> </o:p></p></div></body></html>