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</o:shapelayout></xml><![endif]--></head><body lang=DE link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi Patrik,<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 lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>thanks for this idea!<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'>I did not say clear enough: I’m afraid that anybody can cheat us. My intention is to assure that our interconnection partners (or their customers) do not have the possibility to make a conversation without being charged. <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'>Sending the indication “a:sendonly” only means, that the client is told not to send RTP, but IF it send RTP anyway then the RTPproxy leads in on to the callee. So, it is not in my hands then!<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'>Best regards from Hamburg<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><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><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"'> users-bounces@lists.opensips.org [mailto:users-bounces@lists.opensips.org] <b>Im Auftrag von </b>Patrick Wakano<br><b>Gesendet:</b> Donnerstag, 22. Januar 2015 11:16<br><b>An:</b> OpenSIPS users mailling list<br><b>Betreff:</b> Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>Have you tried to insert a "a:sendonly" line in your SDP body when sending it to the caller?<br>If the client receives such line it should not send media...<o:p></o:p></p></div><div><p class=MsoNormal>Then in the 200Ok you can put an "a:sendrecv" line to establish full media path!<o:p></o:p></p></div><div><p class=MsoNormal>It's just an idea, I'm not sure if it will really work...<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Patrick<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Thu, Jan 22, 2015 at 6:51 AM, Marco Hierl <<a href="mailto:marco.hierl@mrnetgroup.com" target="_blank">marco.hierl@mrnetgroup.com</a>> wrote:<o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi Răzvan,</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Ok, thanks for your answer! </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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?</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Thanks and best regards</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span lang=EN-GB style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Von:</span></b><span lang=EN-GB style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> <a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a> [mailto:<a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a>] <b>Im Auftrag von </b>Razvan Crainea<br><b>Gesendet:</b> Donners</span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>tag, 22. Januar 2015 09:36<br><b>An:</b> <a href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a><br><b>Betreff:</b> Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK</span><o:p></o:p></p></div></div><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><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></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" target="_blank">www.opensips-solutions.com</a><o:p></o:p></pre><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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" target="_blank">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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><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" target="_blank">Users@lists.opensips.org</a><o:p></o:p></pre><pre><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><o:p></o:p></pre></blockquote><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div></div></div><p class=MsoNormal style='margin-bottom:12.0pt'><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>