11 Jun
2018
11 Jun
'18
10:32 a.m.
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
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference -- www.kamailioworld.com
On 11 Jun 2018, at 9:43, Daniel-Constantin Mierla
<miconda(a)gmail.com
<mailto: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
<mailto:972523391991@kamprod.telemessage.com>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO
To:
<sips:1111111@XXXXXX.com
<mailto:1111111@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
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--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference -- www.kamailioworld.com