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</head><body><p></p><div class="io-ox-signature"></div><div class="io-ox-signature">Thank you John for your idea but it seem that when rtpproxy_manage change the SDP, it also change the RTP port that fit the rtpproxy session.</div><div class="io-ox-signature"><br></div><div class="io-ox-signature">So basically, I need to call rtpproxy_manage with the invite to change the SDP attribute but when falling back to the failure_route, I need to destroy the current rtpproxy session and create a new one base on the new parameters of the session (the changed user and destination ip) for the RTP stream.</div><div class="io-ox-signature"><br></div><div class="io-ox-signature">So to manage this correctly, I may need to destroy the first created rtpproxy session and create a new one with the new invite parameters.</div><div class="io-ox-signature"><br></div><div class="io-ox-signature">I will try to work on this.</div><div class="io-ox-signature"><br></div><blockquote type="cite">On June 14, 2017 at 5:19 PM John Petrini <jpetrini@coredial.com> wrote:<br><br><div dir="ltr">You might try using the fix_nated_sdp function from the nathelper module to rewrite the c line in the SDP body.</div><div class="ox-f8fad8a1a7-gmail_extra"><br clear="all"><div><div class="ox-f8fad8a1a7-gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div dir="ltr"><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><p style="margin-top: 8px; margin-bottom: 8px; font-family: Helvetica,Verdana,Arial,sans-serif; font-size: 0.85em; color: #333333;"><span style="font-size: 0.85em;">___</span><br></p><p style="margin-top: 8px; margin-bottom: 8px; font-family: Helvetica,Verdana,Arial,sans-serif; font-size: 1.1em; color: #333333;">John Petrini</p><p style="margin-top: 8px; margin-bottom: 8px; font-family: Helvetica,Verdana,Arial,sans-serif; font-size: 0.85em; color: #333333;">NOC Systems Administrator // <strong>CoreDial, LLC</strong> // <a href="http://coredial.com/" style="color: #2368b3; text-decoration: none;" target="_blank">coredial.com</a> // <a href="https://twitter.com/coredial" target="_blank"><img src="http://cdn2.hubspot.net/hub/210539/file-2353637622-gif/email/twitter-1.gif" alt="Twitter" style="vertical-align: text-bottom;"></a> <a href="http://www.linkedin.com/company/99631" target="_blank"><img src="http://cdn2.hubspot.net/hub/210539/file-2351103565-gif/email/linkedin-1.gif" alt="LinkedIn" style="vertical-align: text-bottom;"></a> <a href="https://plus.google.com/104062177220750809525/posts" target="_blank"><img src="http://cdn2.hubspot.net/hub/210539/file-2353637617-gif/email/googleplus-1.gif" alt="Google Plus" style="vertical-align: text-bottom;"></a> <a href="http://success.coredial.com/blog" target="_blank"><img src="http://cdn2.hubspot.net/hub/210539/file-2351103560-gif/email/rss-1.gif" alt="Blog" style="vertical-align: text-bottom;"></a> <br>Hillcrest I, 751 Arbor Way, Suite 150, Blue Bell PA, 19422 <br><strong>P: </strong><a style="color: #2368b3;">215.297.4400</a> x232 // <strong>F: </strong><a style="color: #2368b3;">215.297.4401</a> // <strong>E: </strong><a href="mailto:jpetrini@coredial.com" style="color: #2368b3; text-decoration: none;" target="_blank">jpetrini@coredial.com</a></p><p><img src="http://success.coredial.com/hubfs/images/CD16-PC17-Email-Signature-Final-Mar-v2-1.gif"><br></p><p style="margin-top: 10px; margin-bottom: 8px; font-family: Helvetica,Verdana,Arial,sans-serif; font-size: 0.85em; color: #333333;">Interested in sponsoring PartnerConnex 2017? <a href="http://success.coredial.com/partnerconnex-2017-sponsorship" style="color: #2368b3; text-decoration: none;" target="_blank">Learn more.</a></p><p style="padding-bottom: 15px; max-width: 600px; font-family: Helvetica,Verdana,Arial,sans-serif; font-size: 0.6em; color: #333333;">The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. 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I use rtpproxy to connect the client and the freeswitch that is used at media server.</p><p>If a call is not answered from userA to userB and userB doesn't have a voicemail configured, I route the Invite to FreeSwitch and I change the destination user to 666666 that take care of playing a nice message that the user doesn't have a voicemail.</p><p>My problem is that the SDP Connection doesn't get updated to the right IP. It get changed to the external IP instead of the internal.</p><p>Here's the original INVITE Packet :</p><p>13:21:22.824290 IP 8.8.8.16.30893 > 8.8.8.18.5060: SIP: INVITE sip:<a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:101@mydomain.com" target="_blank">101@mydomain.com</a> SIP/2.0 <br>Eh....@.?...........x......<wbr/>mINVITE sip:<a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:101@mydomain.com" target="_blank">101@mydomain.com</a> SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.0.236:5060;branch=<wbr/>z9hG4bK-3e3b9e22 <br>From: <a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:sip:102@mydomain.com" target="_blank">102</a>;tag=cacbdfac40ddd17ao0 <br>To: <sip:<a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:101@mydomain.com" target="_blank">101@mydomain.com</a>> <br>Remote-Party-ID: <a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:sip:102@mydomain.com" target="_blank">102</a>;screen=yes;party=calling <br>Call-ID: <a href="mailto:3ae5b914-591b8732@192.168.0.236" target="_blank">3ae5b914-591b8732@192.168.0.<wbr/>236</a> <br>CSeq: 101 INVITE <br>Max-Forwards: 70 <br>Contact: <a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:sip:102@192.168.0.236:5060;ref=102" target="_blank">102</a> <br>Expires: 240 <br>User-Agent: Cisco/SPA122-1.3.5r(003) <br>Content-Length: 263 <br>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER <br>Supported: replaces <br>Content-Type: application/sdp <br><br>v=0 <br>o=- 16833207 16833207 IN IP4 192.168.0.236 <br>s=- <br>c=IN IP4 192.168.0.236 <br>t=0 0 <br>m=audio 16396 RTP/AVP 0 100 101 <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:100 NSE/8000 <br>a=fmtp:100 192-193 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-15 <br>a=ptime:20 <br>a=sendrecv<br><br></p><p>Note that 192.168.0.x in the internal network of the clients and 8.8.8.16 is the public IP of the client and 8.8.8.18 is the kamailio ip.</p><p>Here's the INVITE packet sent from kamailio to Freeswitch</p><p>13:21:32.947196 IP 172.16.0.18.5060 > 172.16.0.19.5080: SIP: INVITE <a href="http://sip:666666666666666@172.16.0.19:5080" target="_blank">sip:666666666666666@172.16.0.<wbr/>19:5080</a> SIP/2.0 <br>E... ...@. ...e...e.......P1INVITE <a href="http://sip:666666666666666@172.16.0.19:5080" target="_blank">sip:666666666666666@172.16.0.<wbr/>19:5080</a> SIP/2.0 <br>Record-Route: <sip:8.8.8.18;lr=on;ftag=<wbr/>cacbdfac40ddd17ao0;did=f3f.<wbr/>205> <br>Test-Pascal: SDP avec freeswitch <br>Via: SIP/2.0/UDP 172.16.0.18;branch=<wbr/>z9hG4bK31ea.<wbr/>e52b370f8248032680211fc7c25390<wbr/>2a.1 <br>Via: SIP/2.0/UDP 192.168.0.236:5060;rport=<wbr/>30893;received=8.8.8.16;<wbr/>branch=z9hG4bK-3e3b9e22 <br>From: <a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:sip:102@mydomain.com" target="_blank">102</a>;tag=cacbdfac40ddd17ao0 <br>To: <sip:<a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:101@mydomain.com" target="_blank">101@mydomain.com</a>> <br>Remote-Party-ID: <a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:sip:102@mydomain.com" target="_blank">102</a>;screen=yes;party=calling <br>Call-ID: <a href="mailto:3ae5b914-591b8732@192.168.0.236" target="_blank">3ae5b914-591b8732@192.168.0.<wbr/>236</a> <br>CSeq: 101 INVITE <br>Max-Forwards: 69 <br>Contact: <a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:sip:102@192.168.0.236:5060;ref=102;alias=8.8.8.16~30893~1" target="_blank">102</a> <br>Expires: 240 <br>User-Agent: Cisco/SPA122-1.3.5r(003) <br>Content-Length: 279 <br>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER <br>Supported: replaces <br>Content-Type: application/sdp <br><br>v=0 <br>o=- 16833207 16833207 IN IP4 8.8.8.18 <br>s=- <br>c=IN IP4 8.8.8.18 <br>t=0 0 <br>m=audio 46134 RTP/AVP 0 100 101 <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:100 NSE/8000 <br>a=fmtp:100 192-193 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-15 <br>a=ptime:20 <br>a=sendrecv <br>a=nortpproxy:yes<br></p><p>Note that 172.16.0.18 is the internal IP of kamailio and 172.16.0.19 is the internal IP of FreeSwitch</p><p>Here's the revealing par of kamailio configuration that us use when falling back to freeswitch :</p><p>failure_route[TOVOICEMAIL] {</p><p>$ru = "sip:666666666666666@" + $sel(cfg_get.freeswitch.<wbr/>bindip)<br>+ ":" + $sel(cfg_get.freeswitch.<wbr/>bindport);<br>$du = "sip:666666666666666@"+ $sel(cfg_get.freeswitch.<wbr/>bindip)<br>+ ":" + $sel(cfg_get.freeswitch.<wbr/>bindport);<br>force_send_socket(udp:<a href="http://172.16.0.18:5060" target="_blank">172.16.<wbr/>0.18:5060</a>);<br></p><p>if (is_request()) { <br> xlog("L_INFO","$ft -- NATMANAGE -- Request"); <br> if(has_totag()) { <br> if(check_route_<wbr/>param("nat=yes")) { <br> setbflag(FLT_<wbr/>NATS); <br> } <br> } <br> }<br><br> if(nat_uac_test("8")) { <br> xlog("L_INFO","----<wbr/>------------------------------<wbr/>-----------------"); <br> xlog("L_INFO","----<wbr/>------------------------------<wbr/>-----------------"); <br> xlog("L_INFO","SDP Body avant : $fu --> $du | $ru"); <br> xlog("L_INFO","----<wbr/>------------------------------<wbr/>-----------------"); <br> xlog("L_INFO","$ft -- RTPMANAGE(FreeSwitch) : NAT SDP"); <br> rtpproxy_manage("<wbr/>1cowei",'192.168.101.18'); <br> insert_hf("Test-<wbr/>Pascal: SDP avec freeswitch\r\n"); <br> xlog("L_INFO","----<wbr/>------------------------------<wbr/>-----------------"); <br> } <br> else { <br> xlog("L_INFO","$ft -- RTPMANAGE(FreeSwitch) : PAS DE NAT SDP"); <br> rtpproxy_manage("<wbr/>cowrei"); <br> } <br> if (is_reply()) { <br> xlog("L_INFO","$ft -- NATMANAGE -- Reply"); <br> if(isbflagset(FLT_NATS)<wbr/>) { <br> if(is_first_hop()) <br> set_contact_<wbr/>alias(); <br><br> } <br> if ( $si == $sel(cfg_get.freeswitch.<wbr/>bindip) && is_method("SUBSCRIBE") ) { <br> remove_hf("Contact"<wbr/>); <br> insert_hf("Contact: \<a class="ox-f8fad8a1a7-m_-8244530551436555704mailto-link" href="mailto:sip:$fU@8.8.8.18:5060" target="_blank">$fU\</a>\r\n","Expires"); <br><br> } <br> else { <br> fix_nated_contact()<wbr/>; <br> } <br><br> }<br><br>if ( is_method("INVITE") )<br> setflag(FLT_DIALOG);<br><br>if (!t_relay()) { <br> xlog("L_INFO","$ft -- ERROR Relay : $rc"); <br> sl_reply_error(); <br> }<br><br>------------------------------<wbr/>------------------</p><p>This configuration is working if the call is not going through failure_route</p><p>I use Kamailio version 5.0.1 on Debian.</p><p>Thank you</p><p><br></p><p>Pascal</p></div><br>______________________________<wbr/>_________________<br> Kamailio (SER) - Users Mailing List<br> <a href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a><br> <a href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">https://lists.kamailio.org/<wbr/>cgi-bin/mailman/listinfo/sr-<wbr/>users</a><br> <br></blockquote></div><br></div></blockquote></body></html>