On Sun, Jan 16, 2022 at 18:07 Chad <ccolumbu@hotmail.com> wrote:

Yes I am using a 172.16.x.x IP and it works, it rewrites the headers, but again because 172.16.x.x is also a private IP
it is the same as using my real 10.x.x.x IP. The carrier's ACK throws away the local IP and sends the response to my
209.x external IP.

--
^C

On 1/16/22 1:38 PM, Ovidiu Sas wrote:
> Have you tried using the mask_ip param:
> https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip
> <https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip>
>
> -ovidiu
>
> On Sun, Jan 16, 2022 at 16:09 Chad <ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>> wrote:
>
> I found a sample config file using topoh, which I copied (with some changes) and added the topoh module to my config.
> It works fine, but it does not solve the problem.
> In fact it has the exact same problem, because all the topoh module does is replace one private IP with another in the
> 2nd (top most) Record-Route header.
> So the carrier still changes the ACK to the public IP and the call is still broken in the exact same way.
> It was super easy to add, but does not work, 1 possible solution down.
>
> --
> ^C
>
>
> On 1/16/22 8:26 AM, Ovidiu Sas wrote:
> > Most of the time, if you get the right person on the carrier's side
> > and you explain the situation, they will come up with a solution.
> > If not, you need to break the RFC in a way that will counterpart their breakage.
> >
> > The carrier is also using a SIP proxy (maybe kamailio, who knows).
> > In the old days, the default kamailio config was using
> > fix_nated_contact() to deal with NATed devices and this is exactly the
> > behavior that you are seeing.
> > The recommended way to deal with NATed devices is to use
> > add_contact_alias([ip_addr, port, proto]) which is RFC compliant.
> >
> > There are several solution for this scenario:
> > - mangle the signaling to allow proper routing on your end
> > - use a B2BUA in between your kamailio and carrier
> > - configure kamailio to use one of the topology hiding modules:
> > topoh, topos, topos_redis
> > - maybe something else ... :)
> >
> > There's no right or wrong approach, one must be comfortable with the
> > chosen solution to be able to maintain it.
> >
> > -ovidiu
> >
> > On Sat, Jan 15, 2022 at 9:14 PM Chad <ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>> wrote:
> >>
> >> Ok so in short I was not doing anything wrong (although I had some miss-configurations), but the carrier is
> (i.e. they
> >> are a bad actor). When they said I was doing it wrong, they did not mean in the RFC sense they meant in the "to work
> >> with us" sense. Now in order for me to get it to work with their SBC I have to mangle the contact on the way out an
> >> unmangle it on the return in Kamailio somehow, as I originally purposed.
> >> However I have no idea how to do that :)
> >>
> >> Shouldn't we (the Kamailio community) assume there are lots of bad actors out there and possibly many Kamailio users
> >> with this exact same issue (I personally know of at least 2 bad actor carriers right now) and create some kind of
> >> template or snippet that we can publicly publish on the Kamailio docs or wiki for all of the Kamailio community
> to use
> >> for this use case?
> >>
> >> I have been fighting with carriers about this for years and they always said I was doing it wrong and I don't
> know the
> >> SIP RFC well enough to fight back. So why not build a solution for everyone out there that has to deal with a
> bad actor?
> >>
> >> --
> >> ^C
> >>
> >>
> >> On 1/15/22 11:40 AM, Ovidiu Sas wrote:
> >>> As expected, your carrier is bogus and "thinks" it knows better.
> >>> Your carrier is treating your setup as a dumb endpoint and is
> >>> re-writing the Contact header:
> >>> You provide this contact header in 200 OK:
> >>> Contact: <sip:928#######@10.###.###.104:5060>
> >>> The carrier should set the RURI in ACK like this:
> >>> ACK sip:928#######@10.###.###.104:5060 SIP/2.0
> >>> Instead, your ACK is sent to you like this:
> >>> ACK sip:928#######@209.###.###.###:5060 SIP/2.0
> >>>
> >>> The RURI in ACK should point to the private IP of the asterisk server,
> >>> not to the public IP of the kamailio server.
> >>> You need to ask the carrier to follow the SIP RFC and not treat your
> >>> endpoints like dumb SIP endpoints.
> >>>
> >>> There's a high chance that they won't do it :)
> >>> Your best chance is to manually mangle the URI in Contact in the 200
> >>> OK in a way that when you receive the ACK with the mangled RURI, you
> >>> can restore the original URI and let kamailio do the proper routing to
> >>> the private IP of the asterisk serverr.
> >>> You should be able to achieve this by using one of the following functions:
> >>> https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact
> <https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact>
> >>> https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact
> <https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact>
> >>> https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode
> <https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode>
> >>>
> >>> Regards,
> >>> Ovidiu Sas
> >>>
> >>> On Sat, Jan 15, 2022 at 1:28 PM Chad <ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>> wrote:
> >>>>
> >>>> I changed the listen per your advice and here is the 200 and ACK.
> >>>> I get no audio and the the call disconnects and I see this is the Asterisk log:
> >>>> [Jan 15 10:17:13] WARNING[29953] chan_sip.c: Retransmission timeout reached on transmission
> >>>> 5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060
> <http://5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060> for seqno 102 (Critical Response) -- See
> >>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> <https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions>
> >>>> Packet timed out after 6401ms with no response
> >>>> [Jan 15 10:17:13] WARNING[29953] chan_sip.c: Hanging up call
> 5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060 <http://5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060> - no
> >>>> reply to our critical packet (see https://wiki.asterisk.org/wik <https://wiki.asterisk.org/wik>
> >>>>
> >>>> FYI 10.###.###.254 is the private virtual IP on the Kamailio server and 10.###.###.104 is the asterisk box.
> >>>>
> >>>> SIP/2.0 200 OK
> >>>> Via: SIP/2.0/UDP 64.###.###.###:5060;branch=z9hG4bK26ab.5547ac15.0
> >>>> Via: SIP/2.0/UDP 206.###.###.###:5060;rport=5060;received=206.###.###.###;branch=z9hG4bK6gj48a00dolcl3jm2gq0.1
> >>>> Record-Route: <sip:10.###.###.254;r2=on;lr=on;ftag=as04035ef0>
> >>>> Record-Route: <sip:209.###.###.###;r2=on;lr=on;ftag=as04035ef0>
> >>>> Record-Route: <sip:64.###.###.###;lr;ftag=as04035ef0>
> >>>> From: "Anonymous" <sip:anonymous@anonymous.invalid:5060>;tag=as04035ef0
> >>>> To: <sip:928#######@64.###.###.###:5060>;tag=as7047ed05
> >>>> Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###:5060
> >>>> CSeq: 102 INVITE
> >>>> Server: Asterisk PBX 16.18.0
> >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> >>>> Supported: replaces, timer
> >>>> Contact: <sip:928#######@10.###.###.104:5060>
> >>>> Content-Type: application/sdp
> >>>> Content-Length: 274
> >>>>
> >>>> v=0
> >>>> o=root 1911037741 1911037741 IN IP4 209.###.###.###
> >>>> s=Asterisk PBX 16.18.0
> >>>> c=IN IP4 209.###.###.###
> >>>> t=0 0
> >>>> m=audio 11384 RTP/AVP 0 101
> >>>> a=rtpmap:0 PCMU/8000
> >>>> a=rtpmap:101 telephone-event/8000
> >>>> a=fmtp:101 0-16
> >>>> a=ptime:20
> >>>> a=maxptime:150
> >>>> a=sendrecv
> >>>> a=nortpproxy:yes
> >>>>
> >>>> ACK sip:928#######@209.###.###.###:5060 SIP/2.0
> >>>> Via: SIP/2.0/UDP 64.###.###.###:5060;branch=z9hG4bK26ab.5547ac15.2
> >>>> Via: SIP/2.0/UDP 206.###.###.###:5060;rport=5060;received=206.###.###.###;branch=z9hG4bK91l3it006gr9oiulcqn0.1
> >>>> Max-Forwards: 67
> >>>> From: "Anonymous" <sip:anonymous@anonymous.invalid:5060>;tag=as04035ef0
> >>>> To: <sip:928#######@64.###.###.###:5060>;tag=as7047ed05
> >>>> Contact: <sip:anonymous@206.###.###.###:5060;transport=udp>
> >>>> Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###:5060
> >>>> CSeq: 102 ACK
> >>>> User-Agent: packetrino
> >>>> Content-Length: 0
> >>>> Route: <sip:209.###.###.###;r2=on;lr=on;ftag=as04035ef0>
> >>>> Route: <sip:10.###.###.254;r2=on;lr=on;ftag=as04035ef0>
> >>>>
> >>>>
> >>>> --
> >>>> ^C
> >>>>
> >>>>
> >>>> On 1/15/22 10:21 AM, Ovidiu Sas wrote:
> >>>>> This is false. The IP in the Contact header must be routable by the
> >>>>> SIP hop from the top Record-Route header in the reply.
> >>>>> The carrier (and it seems that they have a PROXY also) must be able to
> >>>>> route to their adjacent SIP hop, which is your public IP (the IP in
> >>>>> the second Record-Route header).
> >>>>> It seems that the carrier is not taking into account that they might
> >>>>> interface with other proxies.
> >>>>> Most likely, your carrier expects to interface with a simple SIP UA,
> >>>>> not with another proxy. This is a pretty common setup for most of the
> >>>>> carriers, although many new carrier implementations are taking care of
> >>>>> the proxy to proxy calls.
> >>>>>
> >>>>> It would be helpful to see the ACK that is sent by the carrier in
> >>>>> response to your 200ok (after you fix your config and you have your
> >>>>> private IP listed in the Record-Route header).
> >>>>>
> >>>>> -ovidiu
> >>>>>
> >>>>> On Sat, Jan 15, 2022 at 12:33 PM Chad <ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>> wrote:
> >>>>>>
> >>>>>> Hmm, I don't think you are right that the Contact header can be a private IP even if the RR is correct.
> >>>>>> I did some research on it and I found several places saying it must be a routable IP which is what the
> carrier also said.
> >>>>>>
> >>>>>> "The Contact header contains the SIP URI where the client wants to be contacted for subsequent requests.
> That means that
> >>>>>> the host part of the URI must be globally reachable by anyone.
> >>>>>> If your contact contains a private IP (behind a NAT?) then it is wrong, because other peers cannot reach you
> with that."
> >>>>>>
> >>>>>>
> >>>>>> --
> >>>>>> ^C
> >>>>>>
> >>>>>>
> >>>>>> On 1/15/22 9:05 AM, Ovidiu Sas wrote:
> >>>>>>> You have a different problem then.
> >>>>>>> Having private IPs in Contact is fine. You need to lose route the
> >>>>>>> calls (kamailio will add two Record-Route headers) and the origination
> >>>>>>> server will set the RURI to the private IP from Contact, but it will
> >>>>>>> send the in-dialog requests to the public IP of kamailio. This has
> >>>>>>> nothing to do with virtual IPs.
> >>>>>>> Maybe you have a buggy client that doesn't do proper loose routing.
> >>>>>>>
> >>>>>>> -ovidiu
> >>>>>>>
> >>>>>>> On Sat, Jan 15, 2022 at 11:50 AM Chad <ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>> wrote:
> >>>>>>>>
> >>>>>>>> Ovidiu,
> >>>>>>>> Thank you again for your response.
> >>>>>>>> One is public (an internet IP) and one is private (a 10.x ip).
> >>>>>>>> Apparently this is a known problem with virtual IPs, it does not work.
> >>>>>>>> When the asterisk server responds to the invite it sends a contact header with the private IP and Kamailio
> does not
> >>>>>>>> rewrite it to the advertised public IP. So the originating server sees the private IP in the Contact
> header and tries to
> >>>>>>>> send the traffic to the 10.x IP (which is non-routable) and the call dies.
> >>>>>>>> I have been trying things for a long time to fix this (years) what you are saying will not fix it because
> of the virtual
> >>>>>>>> IPs.
> >>>>>>>> If it was a normal IP it would work fine. It has something to do with the routing table and how mhomed
> detects networks.
> >>>>>>>>
> >>>>>>>> --
> >>>>>>>> ^C
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> On 1/15/22 8:36 AM, Ovidiu Sas wrote:
> >>>>>>>>> Hello Chad,
> >>>>>>>>>
> >>>>>>>>> The floating IPs that you have, are they both private IPs or one
> >>>>>>>>> private IP and the other one a public IP?
> >>>>>>>>>
> >>>>>>>>> If you have to two floating private IPs, then you need a config like this:
> >>>>>>>>> listen=FLOATING_UDP_PRIVATE1 advertise PUBLIC_UDP_IP
> >>>>>>>>> listen=FLOATING_UDP_PRIVATE2
> >>>>>>>>>
> >>>>>>>>> In the config, before relaying the initial INVITE you need to detect
> >>>>>>>>> the direction of the call and set $fs accordingly:
> >>>>>>>>> if (CAL_FROM_PRIVATE_TO_PUBLIC) {
> >>>>>>>>> $fs = udp:FLOATING_UDP_PRIVATE1
> >>>>>>>>> }
> >>>>>>>>> else {
> >>>>>>>>> $fs = udp:FLOATING_UDP_PRIVATE2
> >>>>>>>>> }
> >>>>>>>>>
> >>>>>>>>> If you have a floating private IPs and a floating public IP, then you
> >>>>>>>>> need a config like this:
> >>>>>>>>> listen=FLOATING_UDP_PRIVATE
> >>>>>>>>> listen=FLOATING_UDP_PUBLIC
> >>>>>>>>>
> >>>>>>>>> There should be no need to force the socket, but if you do, there's no
> >>>>>>>>> harm (actually it's better and faster).
> >>>>>>>>>
> >>>>>>>>> Hope this clarifies things and helps,
> >>>>>>>>> -ovidiu
> >>>>>>>>>
> >>>>>>>>> On Sat, Jan 15, 2022 at 9:48 AM Chad <ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>> wrote:
> >>>>>>>>>>
> >>>>>>>>>> Ovidiu,
> >>>>>>>>>> Thank you for your response.
> >>>>>>>>>>
> >>>>>>>>>> I have done that, in addition to the linux ip_nonlocal_bind I have also set the Kamailio ip_free_bind=1
> and it does not
> >>>>>>>>>> work.
> >>>>>>>>>> Here are my relevant config lines:
> >>>>>>>>>> listen=LISTEN_UDP_PRIVATE advertise MY_PUBLIC_IP:5060
> >>>>>>>>>> listen=LISTEN_UDP_PUBLIC
> >>>>>>>>>>
> >>>>>>>>>> mhomed=1
> >>>>>>>>>> ip_free_bind=1
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>> In my /etc/sysctl.conf I have (yes I applied it with sysctl -p, and I have been using it for a long time
> and have
> >>>>>>>>>> rebooted as well):
> >>>>>>>>>> net.ipv4.ip_nonlocal_bind=1
> >>>>>>>>>> --
> >>>>>>>>>> ^C
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>> On 1/15/22 4:55 AM, Ovidiu Sas wrote:
> >>>>>>>>>>> Hello Chad,
> >>>>>>>>>>>
> >>>>>>>>>>> You can add a listen directive to your config for the virtual IPs
> >>>>>>>>>>> (both public and private) and then you don't need to manually modify
> >>>>>>>>>>> any headers or use force_send_socket().
> >>>>>>>>>>> You need to enable non local IP binding so kamailio can start on the
> >>>>>>>>>>> server that doesn't have the virtual IP:
> >>>>>>>>>>> echo 1 > /proc/sys/net/ipv4/ip_nonlocal_bind
> >>>>>>>>>>> To make the change permanent, edit your sysctl.conf file and enable it there:
> >>>>>>>>>>> net/ipv4/ip_nonlocal_bind = 1
> >>>>>>>>>>>
> >>>>>>>>>>> Regards
> >>>>>>>>>>> Ovidiu Sas
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>> On Sat, Jan 15, 2022 at 4:16 AM Chad <ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>> wrote:
> >>>>>>>>>>>>
> >>>>>>>>>>>> We are looking for some help (possibly a paid consultant) to help us with our Kamailio setup.
> >>>>>>>>>>>> To keep this as short as possible: we use Kamailio as a NAT proxy to bridge our external IP and our
> private IP asterisk
> >>>>>>>>>>>> servers (via dispatcher).
> >>>>>>>>>>>> However both the external IP and the internal IP that the Kamailio server uses are virtual IPs created
> by keepalived.
> >>>>>>>>>>>> Because of that neither mhomed nor fix_nated_contact work, and we use force_send_socket to direct the
> traffic.
> >>>>>>>>>>>> We run linux Debian 10 for the OS.
> >>>>>>>>>>>> Also we do not use a DB at all, everything is done with local config files.
> >>>>>>>>>>>>
> >>>>>>>>>>>> The problem is that when traffic goes out the Contact header has a private IP in it, like:
> >>>>>>>>>>>> Contact: <sip:##########@10.10.10.###]:5060 <http://10.10.10.#%23%23]:5060>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> There are 2 possible solutions to this:
> >>>>>>>>>>>> 1. Make changes to linux, keepalived and/or Kamailio so that Kamailio recognize the virtual IPs so
> that mhomed and
> >>>>>>>>>>>> fix_nated_contact work as usual.
> >>>>>>>>>>>>
> >>>>>>>>>>>> 2. Create a manual header rewrite system.
> >>>>>>>>>>>>
> >>>>>>>>>>>> If solution #2:
> >>>>>>>>>>>> What we need to do is create a way to rewrite the contact header to the external IP on the way out,
> and on the way back
> >>>>>>>>>>>> rewrite it back to the internal server that the call is already connected to.
> >>>>>>>>>>>>
> >>>>>>>>>>>> Not sure if we will need to store those paths on the server or if we can do some kind of cheat with
> another persistant
> >>>>>>>>>>>> header like P-Preferred-Identity or P-Asserted-Identity (i.e. store the internal IP in the name field
> or something).
> >>>>>>>>>>>>
> >>>>>>>>>>>> If anyone out there know of a way to do this or wants to give it a try please reach out to me.
> >>>>>>>>>>>>
> >>>>>>>>>>>> Thank you all for your time.
> >>>>>>>>>>>>
> >>>>>>>>>>>> --
> >>>>>>>>>>>> ^C
> >>>>>>>>>>>> Chad
> >>>>>>>>>>>>
> >>>>>>>>>>>> __________________________________________________________
> >>>>>>>>>>>> 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
> <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>> --
> >>>>>>>>>>> VoIP Embedded, Inc.
> >>>>>>>>>>> http://www.voipembedded.com <http://www.voipembedded.com>
> >>>>>>>>>>>
> >>>>>>>>>>> __________________________________________________________
> >>>>>>>>>>> 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
> <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>
> >>>
> >>>
> >
> >
> >
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com <http://www.voipembedded.com>