Hello, can you paste here the 200OK for INVITE sent out by kamailio and the ACK received by kamailio?
Cheers, Daniel On 11.06.18 09:51, Arik Halperin wrote: > Daniel, Thank you! > > You are right about this. > > I configured PJSIP not to check whether the contact contains SIPS. > > This solved the problem on one of my setups where I have one NIC that > has a public IP. > > However on the original setup, the kamailio has one public IP and one > private IP. In that setup, the ACK to the 200 OK is not forwarded over > the private IP to the freeswitch. This only happens in TLS, when I > work with TCP it works well. I believe it is somehow connected to the > record route, and I’m looking into PJSIP to try to find the answer, > but is there anything I could do in the kamailio? > > I have the same problems with other SIP clients(Bria for example) > > > Thanks, > Arik Halperin > >> On 11 Jun 2018, at 9:43, Daniel-Constantin Mierla <mico...@gmail.com >> <mailto:mico...@gmail.com>> wrote: >> >> Hello, >> >> Kamailio is not involved in the issue reported here. Practically, >> pjsip expects sips: scheme in the contact URI, which was set by >> FreeSwitch in 200ok. Maybe there is an option that you have to turn >> on for FreeSwitch to use sips: scheme. >> >> Otherwise, you can try to replace sip with sips in kamailio config >> and do the reverse the other way. >> >> Cheers, >> Daniel >> >> >> On 05.06.18 06:56, Arik Halperin wrote: >>> Hello, >>> >>> I’m using TLS >>> >>> After receiving 200OK from kamailio: >>> >>> r2voip.clear2voipdialer I/(NativeSdk_2_0) 1528174138320 PJSIP: >>> (NativeSdk_2_0) 1528174138320 PJSIP:2018-05 07:48:58.319 >>> pjsua_core.c RX 2203 bytes Response msg 200/INVITE/cseq=8107 >>> (rdata0x7a2c56fb38) from TLS 70.36.25.65:443: >>> >>> SIP/2.0 200 OK >>> >>> Via: SIP/2.0/TLS >>> 10.134.232.109:44097;received=109.253.173.146;rport=31373;branch=z9hG4bKPj4MV5llP9SW5ufk-OcFB-Qh78PmIQFrRk;alias >>> >>> Record-Route: >>> <sips:10.168.10.227:5099;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> >>> >>> Record-Route: >>> <sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> >>> >>> From: "number" >>> <sips:972523391...@xxxxxxx.com >>> <mailto:972523391...@kamprod.telemessage.com>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO >>> >>> To: >>> <sips:1111...@xxxxxx.com >>> <mailto:1111...@kamprod.telemessage.com>>;tag=64H63g861ajHj >>> >>> Call-ID: >>> Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2 >>> >>> CSeq: 8107 INVITE >>> >>> Contact: >>> <sip:1111111@10.168.10.200:5080;transport=tls> >>> >>> User-Agent: >>> FreeSWITCH-mod_sofia/1.6.20+git~20180123T214909Z~987c9b9a2a~64bit >>> >>> Accept: application/sdp >>> >>> Allow: INVITE, ACK, BYE, >>> CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY >>> >>> Require: timer >>> >>> Supported: ti >>> >>> >>> *PJSIP responds with:* >>> * >>> * >>> *Secure dialog requires SIPS scheme in Contact and Record-Route >>> headers, ending the session* >>> * >>> * >>> What is the reason for this? How can I fix this issue? >>> >>> Thanks, >>> Arik Halperin >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users@lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla -- www.asipto.com >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio World Conference -- www.kamailioworld.com > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users