11 Jun
2018
11 Jun
'18
12:58 p.m.
Daniel Hello,
Pasted below, 200 OK and Following ACK(Recorded at the client side via wireshark
configured with private key)
BR,
Arik
Session Initiation Protocol (200)
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/TLS
192.168.2.2:48182;received=82.80.164.63;rport=33898;branch=z9hG4bKPjVppvYKQb4X5lJrYpod1wUN.j3KVLrEiT;alias
Record-Route:
<sips:10.168.10.227:5099;r2=on;lr=on;ftag=ZmXcXh6ReoLbMco46J0fCpKOHkUR1sWF;nat=yes>
Record-Route:
<sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=ZmXcXh6ReoLbMco46J0fCpKOHkUR1sWF;nat=yes>
From: "number"
<sips:17813000000@XXXXXX.com<mailto:17813000000@XXXXXX.com>>;tag=ZmXcXh6ReoLbMco46J0fCpKOHkUR1sWF
To:
<sips:1111111@XXXXXX.com<mailto:1111111@XXXXXX.com>>;tag=7t2StmvUeNpQD
Call-ID: yekcL-0b2PhpgdQo52l921tjX1Z8wErH
CSeq: 10885 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: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 1056
Remote-Party-ID: "1111111"
<sip:1111111@XXXXXX.com>;party=calling;privacy=off;screen=no
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): FreeSWITCH 1528683321 1528683322 IN IP4
70.36.25.66
Session Name (s): FreeSWITCH
Connection Information (c): IN IP4 70.36.25.66
Time Description, active time (t): 0 0
Session Attribute (a): msid-semantic: WMS V60mDk4CUtzxt4H5xDQPB48KjzMcYE1K
Media Description, name and address (m): audio 37680 RTP/SAVP 107 96
Media Attribute (a): ice-ufrag:b6TC1SdbiQd6k5GL
Media Attribute (a): ice-pwd:NtGGa3jbPjvwRLASIklz2oAa
Media Attribute (a): candidate:5807878115 1 udp 659136 10.168.10.200 38056 typ
host generation 0
Media Attribute (a): candidate:5807878115 2 udp 659135 10.168.10.200 38057 typ
host generation 0
Media Attribute (a): ssrc:3542382753 cname:ASW42RxMaWauQHpe
Media Attribute (a): ssrc:3542382753 msid:V60mDk4CUtzxt4H5xDQPB48KjzMcYE1K
a0
Media Attribute (a): ssrc:3542382753
mslabel:V60mDk4CUtzxt4H5xDQPB48KjzMcYE1K
Media Attribute (a): ssrc:3542382753
label:V60mDk4CUtzxt4H5xDQPB48KjzMcYE1Ka0
Media Attribute (a): rtpmap:107 opus/48000/2
Media Attribute (a): rtpmap:96 telephone-event/8000
Media Attribute (a): fmtp:107 useinbandfec=1; minptime=10; maxptime=40
Media Attribute (a): fmtp:96 0-16
Media Attribute (a): sendrecv
Media Attribute (a): rtcp:37681
Media Attribute (a): crypto:1 AES_CM_128_HMAC_SHA1_80
inline:/KCNveJuRh5lQ+g3YWnyb2QwQhl0GgdmxtKAJ5G3
Media Attribute (a): ptime:20
Media Attribute (a): candidate:K6gXQsPK0KD4MsGa 1 UDP 2130706431 70.36.25.66
37680 typ host
Media Attribute (a): candidate:K6gXQsPK0KD4MsGa 2 UDP 2130706430 70.36.25.66
37681 typ host
Media Attribute (a): end-of-candidates
1201 272.987349 192.168.2.2 70.36.25.65 SIP 695
Request: ACK sip:1111111@10.168.10.200:5080;transport=tls | 1201
Frame 1201: 695 bytes on wire (5560 bits), 695 bytes captured (5560 bits) on interface 0
Ethernet II, Src: Htc_50:62:7b (ac:37:43:50:62:7b), Dst: 9a:01:a7:d9:66:64
(9a:01:a7:d9:66:64)
Internet Protocol Version 4, Src: 192.168.2.2, Dst: 70.36.25.65
Transmission Control Protocol, Src Port: 48182, Dst Port: 443, Seq: 8791, Ack: 10303, Len:
629
Secure Sockets Layer
Session Initiation Protocol (ACK)
Request-Line: ACK sip:1111111@10.168.10.200:5080;transport=tls SIP/2.0
Message Header
Via: SIP/2.0/TLS
192.168.2.2:48182;rport;branch=z9hG4bKPjFpv1IqHt9ON8nS6zOYuUZ5HxhNTDTBq7;alias
Max-Forwards: 70
From: "number"
<sips:17813000000@XXXXXXXX.com<mailto:17813000000@XXXXXXXX.com>>;tag=ZmXcXh6ReoLbMco46J0fCpKOHkUR1sWF
To: sips:1111111@XXXXXXX.com<mailto:1111111@XXXXXXX.com>;tag=7t2StmvUeNpQD
Call-ID: yekcL-0b2PhpgdQo52l921tjX1Z8wErH
CSeq: 10885 ACK
Route:
<sips:70.36.25.65:443;transport=tls;lr;r2=on;ftag=ZmXcXh6ReoLbMco46J0fCpKOHkUR1sWF;nat=yes>
Route:
<sips:10.168.10.227:5099;lr;r2=on;ftag=ZmXcXh6ReoLbMco46J0fCpKOHkUR1sWF;nat=yes>
Content-Length: 0
On 11 Jun 2018, at 13:32, Daniel-Constantin Mierla
<miconda@gmail.com<mailto:miconda@gmail.com>> wrote:
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
<miconda@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<http://www.asipto.com/>
www.twitter.com/miconda<http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>
Kamailio World Conference -- www.kamailioworld.com<http://www.kamailioworld.com/>
--
Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com/>
www.twitter.com/miconda<http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>
Kamailio World Conference -- www.kamailioworld.com<http://www.kamailioworld.com/>