[SR-Users] Problem with TLS

Arik Halperin arik at mobilinq.io
Tue Jun 5 06:56:48 CEST 2018


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 at kamprod.telemessage.com<mailto:972523391991 at kamprod.telemessage.com>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO
                                                                                                               To: <sips:1111111 at kamprod.telemessage.com<mailto:1111111 at kamprod.telemessage.com>>;tag=64H63g861ajHj
                                                                                                               Call-ID: Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2
                                                                                                               CSeq: 8107 INVITE
                                                                                                               Contact: <sip:1111111 at 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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