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 ->
80.0.0.1:49992SIP/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 ->
80.0.0.2:5061ACK 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
How can I solve this ?
Best Regards
Leo