18 Oct
2022
18 Oct
'22
9:36 p.m.
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(a)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
[image: sip-ok-small.jpeg]
SIPS call example
If the same UAC use his default settings tls(sips) , there are problems
with ACK and BYE packet
[image: sip-ko-small.jpeg]
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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