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<p>Hello,</p>
<p>can you paste here the 200OK for INVITE sent out by kamailio and
the ACK received by kamailio?</p>
<p>Cheers,<br>
Daniel<br>
</p>
<br>
<div class="moz-cite-prefix">On 11.06.18 09:51, Arik Halperin wrote:<br>
</div>
<blockquote type="cite"
cite="mid:1623CA64-0131-4549-8CC4-1FFB41A4F365@mobilinq.io">
<meta http-equiv="Content-Type" content="text/html; charset=utf-8">
Daniel, Thank you!
<div class=""><br class="">
</div>
<div class="">You are right about this.
<div class=""><br class="">
</div>
<div class="">I configured PJSIP not to check whether the
contact contains SIPS. </div>
<div class=""><br class="">
</div>
<div class="">This solved the problem on one of my setups where
I have one NIC that has a public IP.</div>
<div class=""><br class="">
</div>
<div class="">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? </div>
<div class=""><br class="">
</div>
<div class="">I have the same problems with other SIP
clients(Bria for example)</div>
<div class=""><br class="">
</div>
<div class=""><br class="">
</div>
<div class="">Thanks,</div>
<div class="">Arik Halperin<br class="">
<div><br class="">
<blockquote type="cite" class="">
<div class="">On 11 Jun 2018, at 9:43, Daniel-Constantin
Mierla <<a href="mailto:miconda@gmail.com" class=""
moz-do-not-send="true">miconda@gmail.com</a>>
wrote:</div>
<br class="Apple-interchange-newline">
<div class="">
<div text="#000000" bgcolor="#FFFFFF" class="">
<p class="">Hello,</p>
<p class="">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.</p>
<p class="">Otherwise, you can try to replace sip with
sips in kamailio config and do the reverse the other
way.</p>
<p class="">Cheers,<br class="">
Daniel<br class="">
</p>
<br class="">
<div class="moz-cite-prefix">On 05.06.18 06:56, Arik
Halperin wrote:<br class="">
</div>
<blockquote type="cite"
cite="mid:140EA09C-16A9-4FF2-9370-D03E6AAD2B08@mobilinq.io"
class="">
Hello,
<div class=""><br class="">
</div>
<div class="">I’m using TLS</div>
<div class=""><br class="">
</div>
<div class="">After receiving 200OK from kamailio:</div>
<div class=""><br class="">
</div>
<div class="">
<div class="">
<div class="">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:</div>
<div class="">
SIP/2.0 200 OK</div>
<div class="">
Via:
SIP/2.0/TLS
10.134.232.109:44097;received=109.253.173.146;rport=31373;branch=z9hG4bKPj4MV5llP9SW5ufk-OcFB-Qh78PmIQFrRk;alias</div>
<div class="">
Record-Route:
<sips:10.168.10.227:5099;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes></div>
<div class="">
Record-Route:
<sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes></div>
<div class="">
From: "number"
<sips:<a
href="mailto:972523391991@kamprod.telemessage.com"
class="" moz-do-not-send="true">972523391991@XXXXXXX.com</a>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO</div>
<div class="">
To: <sips:<a
href="mailto:1111111@kamprod.telemessage.com" class=""
moz-do-not-send="true">1111111@XXXXXX.com</a>>;tag=64H63g861ajHj</div>
<div class="">
Call-ID:
Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2</div>
<div class="">
CSeq: 8107
INVITE</div>
<div class="">
Contact: <<a
href="sip:1111111@10.168.10.200:5080;transport=tls" class=""
moz-do-not-send="true">sip:1111111@10.168.10.200:5080;transport=tls</a>></div>
<div class="">
User-Agent:
FreeSWITCH-mod_sofia/1.6.20+git~20180123T214909Z~987c9b9a2a~64bit</div>
<div class="">
Accept:
application/sdp</div>
<div class="">
Allow: INVITE,
ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY</div>
<div class="">
Require: timer</div>
<div class="">
Supported: ti</div>
</div>
<div class=""><br class="">
</div>
<div class=""><br class="">
</div>
<div class=""><b class="">PJSIP responds with:</b></div>
<div class=""><b class=""><br class="">
</b></div>
<div class=""><b class="">Secure dialog requires
SIPS scheme in Contact and Record-Route
headers, ending the session</b></div>
</div>
<div class=""><b class=""><br class="">
</b></div>
<div class="">What is the reason for this? How can I
fix this issue?</div>
<div class=""><br class="">
</div>
<div class="">Thanks,</div>
<div class="">Arik Halperin</div>
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<br class="">
<pre class="" wrap="">_______________________________________________
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</pre>
</blockquote>
<br class="">
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla -- <a class="moz-txt-link-abbreviated" href="http://www.asipto.com/" moz-do-not-send="true">www.asipto.com</a>
<a class="moz-txt-link-abbreviated" href="http://www.twitter.com/miconda" moz-do-not-send="true">www.twitter.com/miconda</a> -- <a class="moz-txt-link-abbreviated" href="http://www.linkedin.com/in/miconda" moz-do-not-send="true">www.linkedin.com/in/miconda</a>
Kamailio World Conference -- <a class="moz-txt-link-abbreviated" href="http://www.kamailioworld.com/" moz-do-not-send="true">www.kamailioworld.com</a></pre>
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</blockquote>
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</div>
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</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla -- <a class="moz-txt-link-abbreviated" href="http://www.asipto.com">www.asipto.com</a>
<a class="moz-txt-link-abbreviated" href="http://www.twitter.com/miconda">www.twitter.com/miconda</a> -- <a class="moz-txt-link-abbreviated" href="http://www.linkedin.com/in/miconda">www.linkedin.com/in/miconda</a>
Kamailio World Conference -- <a class="moz-txt-link-abbreviated" href="http://www.kamailioworld.com">www.kamailioworld.com</a></pre>
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