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> 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/siputils.html#siputils.f.encode_contact
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> 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 <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> 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> 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> 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>
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
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