Use your 209.x external IP. On Sun, Jan 16, 2022 at 18:07 Chad <ccolu...@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 <ccolu...@hotmail.com <mailto: > ccolu...@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 <ccolu...@hotmail.com > <mailto:ccolu...@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 <ccolu...@hotmail.com > <mailto:ccolu...@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 <ccolu...@hotmail.com > <mailto:ccolu...@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 <ccolu...@hotmail.com > <mailto:ccolu...@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 < > ccolu...@hotmail.com <mailto:ccolu...@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 < > ccolu...@hotmail.com <mailto:ccolu...@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> <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> > -- VoIP Embedded, Inc. http://www.voipembedded.com
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