You make calls using SIP over TLS and it's OK.
You make calls using SIPS and it's not OK.

The 200-OK Contact is this
- Contact: <sip:172.16.0.2:5060>

The RURI in the ACK is this.
- sips:172.16.0.2:5060;transport=tcp SIP/2.0

The client should be using _exactly_ the same URI in the ACK as was in the Contact in the 200-OK response. The client is getting it wrong (https://datatracker.ietf.org/doc/html/rfc3261#section-12.2.1.1).

With the ACK, the Routeset is this.
- ACK sips:172.16.0.2:5060;transport=tcp SIP/2.0
- Route: <sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
- Route: <sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>

This is asking kamailio to send the ACK to 172.16.0.2 over TLS on port 5060 (from the RURI), but use a UDP socket to do it (the second Route header field). This can't work. The client should be fixed.

(Else you may try getting asterisk to use sips, but maybe that's not going to be easy.)

James

On Thu, 13 Oct 2022 at 06:30, beer Ll <llcfhllml@gmail.com> wrote:

Hi everyone, 

I'm using Kamailio as TLS gateway/filter for an internal Asterisk server

the network schema is  :

UAC (tls) --- INTERNET  --- (tls) KAMAILIO (sip udp) --- LAN --- (sip udp) ASTERISK

 

with kamailio in multi-homed mode

WAN network interface for sip tls

LAN network interface for sip udp  to asterisk server

UAC address 80.0.0.1

KAMAILIO Wan address 80.0.0.2

KAMAILIO Lan address 172.16.0.2

ASTERISK Lan address 172.16.0.3

SIP-TLS call example

If the UAC use tls(sip) all works good

SIPS call example

If the same UAC use his default settings tls(sips)  , there are problems with ACK and BYE packet

the SIP OK SDP packet from kamailio to UAC  is

2022/10/10 09:28:47.854721 80.0.0.2:5061 -> 80.0.0.1:49992
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.0.1:49992;rport=49992;received=80.0.0.1;branch=z9hG4bKM01j360VrBdH5VSV
Record-Route: <sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
Record-Route: <sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7
From: <sips:200@pbx.voip.com>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F
To: <sips:*43@pbx.voip.com>;tag=961d0e22-a4f0-453c-9870-6a41578afc96
CSeq: 2 INVITE
Contact: <sip:172.16.0.2:5060>
P-Asserted-Identity: "xxxxxxxxx" <sips:*43@pbx.voip.com>
Content-Type: application/sdp

and the UAC send the ACK and BYE from a different tcp port and to: sips:172.16.0.2:5060;transport=tcp

2022/10/10 09:28:48.495365 80.0.0.1:49996 -> 80.0.0.2:5061
ACK sips:172.16.0.2:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TLS 192.168.0.1:49996;branch=z9hG4bKppftdQze20lnwT41;rport
Route: <sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
Route: <sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
Max-Forwards: 70
To: <sips:*43@pbx.voip.com>;tag=961d0e22-a4f0-453c-9870-6a41578afc96
From: <sips:200@pbx.voip.com>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F
Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7
CSeq: 2 ACK

kamailio error log

WARNING:  <core> [core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced udp:172.16.0.2:5060, to tls:172.16.0.3:5060)

How can I solve this ?

Best Regards

Leo

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