17 Jan
2022
17 Jan
'22
12:38 a.m.
Have you tried using the mask_ip param:
https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask…
-ovidiu
On Sun, Jan 16, 2022 at 16:09 Chad <ccolumbu(a)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:
--
VoIP Embedded, Inc.
http://www.voipembedded.com
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(a)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.enco…
>>
https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.en…
>>
https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.co…
>>
>>> Regards,
>>> Ovidiu Sas
>>
>>> On Sat, Jan 15, 2022
at 1:28 PM Chad <ccolumbu(a)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 for seqno 102
(Critical
Response) -- See
>>>
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 - no
>>> reply to our critical packet (see
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(a)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(a)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(a)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(a)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(a)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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> VoIP Embedded, Inc.
>>>>>>>>>> http://www.voipembedded.com
>>>>>>>>>>
>>>>>>>>>>
__________________________________________________________
>>>>>>>>>> Kamailio - Users Mailing List - Non Commercial
Discussions
>>>>>>>>>> * sr-users(a)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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>
>>>>>
>>>>>
>>
>>
>>
>
>
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