You might try using the fix_nated_sdp function from the nathelper module to rewrite the c line in the SDP body.
___ John Petrini NOC Systems Administrator // *CoreDial, LLC* // coredial.com // [image: Twitter] <https://twitter.com/coredial> [image: LinkedIn] <http://www.linkedin.com/company/99631> [image: Google Plus] <https://plus.google.com/104062177220750809525/posts> [image: Blog] <http://success.coredial.com/blog> Hillcrest I, 751 Arbor Way, Suite 150, Blue Bell PA, 19422 *P: *215.297.4400 x232 // *F: *215.297.4401 // *E: * [email protected] Interested in sponsoring PartnerConnex 2017? Learn more. <http://success.coredial.com/partnerconnex-2017-sponsorship> The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. On Wed, Jun 14, 2017 at 5:00 PM, Pascal Poudrier < [email protected]> wrote: > Hi, first of all, here's the architecture of what I'm trying to do : > > Public client --> Kamailio and there is a private freeswitch behind > kamailio. I use rtpproxy to connect the client and the freeswitch that is > used at media server. > > 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. > > 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. > > Here's the original INVITE Packet : > > 13:21:22.824290 IP 8.8.8.16.30893 > 8.8.8.18.5060: SIP: INVITE sip: > [email protected] SIP/2.0 > Eh....@.?...........x......mINVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.236:5060;branch=z9hG4bK-3e3b9e22 > From: 102 <sip:[email protected]>;tag=cacbdfac40ddd17ao0 > To: <sip:[email protected]> > Remote-Party-ID: 102 <sip:[email protected]>;screen=yes;party=calling > Call-ID: [email protected] > CSeq: 101 INVITE > Max-Forwards: 70 > Contact: 102 <sip:[email protected]:5060;ref=102> > Expires: 240 > User-Agent: Cisco/SPA122-1.3.5r(003) > Content-Length: 263 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: replaces > Content-Type: application/sdp > > v=0 > o=- 16833207 16833207 IN IP4 192.168.0.236 > s=- > c=IN IP4 192.168.0.236 > t=0 0 > m=audio 16396 RTP/AVP 0 100 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > > 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. > > Here's the INVITE packet sent from kamailio to Freeswitch > > 13:21:32.947196 IP 172.16.0.18.5060 > 172.16.0.19.5080: SIP: INVITE > sip:[email protected]:5080 SIP/2.0 > E... ...@. ...e...e.......P1INVITE sip:[email protected]. > 19:5080 SIP/2.0 > Record-Route: <sip:8.8.8.18;lr=on;ftag=cacbdfac40ddd17ao0;did=f3f.205> > Test-Pascal: SDP avec freeswitch > Via: SIP/2.0/UDP 172.16.0.18;branch=z9hG4bK31ea. > e52b370f8248032680211fc7c253902a.1 > Via: SIP/2.0/UDP > 192.168.0.236:5060;rport=30893;received=8.8.8.16;branch=z9hG4bK-3e3b9e22 > > From: 102 <sip:[email protected]>;tag=cacbdfac40ddd17ao0 > To: <sip:[email protected]> > Remote-Party-ID: 102 <sip:[email protected]>;screen=yes;party=calling > Call-ID: [email protected] > CSeq: 101 INVITE > Max-Forwards: 69 > Contact: 102 <sip:[email protected]:5060;ref=102;alias=8.8.8.16~30893~1> > Expires: 240 > User-Agent: Cisco/SPA122-1.3.5r(003) > Content-Length: 279 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: replaces > Content-Type: application/sdp > > v=0 > o=- 16833207 16833207 IN IP4 8.8.8.18 > s=- > c=IN IP4 8.8.8.18 > t=0 0 > m=audio 46134 RTP/AVP 0 100 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > a=nortpproxy:yes > > Note that 172.16.0.18 is the internal IP of kamailio and 172.16.0.19 is > the internal IP of FreeSwitch > > Here's the revealing par of kamailio configuration that us use when > falling back to freeswitch : > > failure_route[TOVOICEMAIL] { > > $ru = "sip:666666666666666@" + $sel(cfg_get.freeswitch.bindip) > + ":" + $sel(cfg_get.freeswitch.bindport); > $du = "sip:666666666666666@"+ $sel(cfg_get.freeswitch.bindip) > + ":" + $sel(cfg_get.freeswitch.bindport); > force_send_socket(udp:172.16.0.18:5060); > > if (is_request()) { > xlog("L_INFO","$ft -- NATMANAGE -- Request"); > if(has_totag()) { > if(check_route_param("nat=yes")) { > setbflag(FLT_NATS); > } > } > } > > if(nat_uac_test("8")) { > > xlog("L_INFO","---------------------------------------------------"); > > > xlog("L_INFO","---------------------------------------------------"); > > xlog("L_INFO","SDP Body avant : $fu --> $du | $ru"); > > xlog("L_INFO","---------------------------------------------------"); > > xlog("L_INFO","$ft -- RTPMANAGE(FreeSwitch) : NAT SDP"); > rtpproxy_manage("1cowei",'192.168.101.18'); > insert_hf("Test-Pascal: SDP avec freeswitch\r\n"); > > xlog("L_INFO","---------------------------------------------------"); > > } > else { > xlog("L_INFO","$ft -- RTPMANAGE(FreeSwitch) : PAS DE NAT SDP"); > rtpproxy_manage("cowrei"); > } > if (is_reply()) { > xlog("L_INFO","$ft -- NATMANAGE -- Reply"); > if(isbflagset(FLT_NATS)) { > if(is_first_hop()) > set_contact_alias(); > > } > if ( $si == $sel(cfg_get.freeswitch.bindip) && > is_method("SUBSCRIBE") ) { > remove_hf("Contact"); > insert_hf("Contact: \$fU\ <sip:[email protected]:5060>\r\n","Expires"); > > > } > else { > fix_nated_contact(); > } > > } > > if ( is_method("INVITE") ) > setflag(FLT_DIALOG); > > if (!t_relay()) { > xlog("L_INFO","$ft -- ERROR Relay : $rc"); > sl_reply_error(); > } > > ------------------------------------------------ > > This configuration is working if the call is not going through > failure_route > > I use Kamailio version 5.0.1 on Debian. > > Thank you > > > Pascal > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >
_______________________________________________ Kamailio (SER) - Users Mailing List [email protected] https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
