Hi James Thanks , You confirm my idea . I will write to the UAC developers and check this problem asking to fix it. Is safer use sip-tls or sips-tcp ? Best Regards Leo On Wed, Oct 19, 2022 at 2:07 PM James Browne <james(a)frideo.com> wrote:

Hello, the „sips“ URI scheme is not really useful nowadays and IMHO should not be used. You probably find in the archives some old discussions about it. Use “sip” and “transport=tls” if you want to have a secure connection. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users &lt;sr-users-bounces(a)lists.kamailio.org&gt; On Behalf Of beer Ll Sent: Thursday, October 20, 2022 9:22 AM To: Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.kamailio.org&gt; Subject: Re: [SR-Users] relay sip-tls vs sips-tcp Hi James Thanks , You confirm my idea . I will write to the UAC developers and check this problem asking to fix it. Is safer use sip-tls or sips-tcp ? Best Regards Leo On Wed, Oct 19, 2022 at 2:07 PM James Browne <james@frideo.com<mailto:james@frideo.com>> wrote: 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<http://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<mailto: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 [cid:image001.jpg@01D8E543.1E954C20] SIPS call example If the same UAC use his default settings tls(sips) , there are problems with ACK and BYE packet [cid:image002.jpg@01D8E543.1E954C20] the SIP OK SDP packet from kamailio to UAC is 2022/10/10 09:28:47.854721 80.0.0.2:5061<http://80.0.0.2:5061> -> 80.0.0.1:49992<http://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<mailto:sips%3A200@pbx.voip.com>>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F To: <sips:*43@pbx.voip.com<mailto:43@pbx.voip.com>>;tag=961d0e22-a4f0-453c-9870-6a41578afc96 CSeq: 2 INVITE Contact: <sip:172.16.0.2:5060<http://172.16.0.2:5060>> P-Asserted-Identity: "xxxxxxxxx" <sips:*43@pbx.voip.com<mailto: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<http://80.0.0.1:49996> -> 80.0.0.2:5061<http://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<mailto:43@pbx.voip.com>>;tag=961d0e22-a4f0-453c-9870-6a41578afc96 From: <sips:200@pbx.voip.com<mailto:sips%3A200@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<http://172.16.0.2:5060>, to tls:172.16.0.3:5060<http://172.16.0.3:5060>) How can I solve this ? Best Regards Leo __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users