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 <miconda@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:972523391991@XXXXXXX.com>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO
                                                                                                               To: <sips:1111111@XXXXXX.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


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