subject:"\[SR\-Users\] help"


Hmm, it did not fix it (calls still work with my other carriers).
It looks to me like it should work, it does use the external IP for everything.

It generates an error in the log about making your existing address:
topoh [topoh_mod.c:179]: mod_init(): mask address matches myself 
[209.###.###.###]

Here is ther 200 and ACK.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.###.###.###:5060;branch=z9hG4bKf229.9bb425c2.0
Via: SIP/2.0/UDP 
206.###.###.###:5060;rport=5060;received=206.###.###.###;branch=z9hG4bKrs8bqi00cg14535baf70.1
Record-Route: 

Record-Route: 
Record-Route: 
From: "Anonymous" ;tag=as471a1f75
To: ;tag=as199dc3d1
Call-ID: 3510f7167e1a0f6a5423234b1d176a8b@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: 

Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1644013823 1644013823 IN IP4 209.###.###.###
s=Asterisk PBX 16.18.0
c=IN IP4 209.###.###.###
t=0 0
m=audio 19180 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:209.###.###.###;line=sr-1RaGXx7VX8CAKx1yp8oFKfFqKfFqKfFqKfFVXx9GpxF* 
SIP/2.0
Via: SIP/2.0/UDP 64.###.###.###:5060;branch=z9hG4bKf229.9bb425c2.2
Via: SIP/2.0/UDP 
206.###.###.###:5060;rport=5060;received=206.###.###.###;branch=z9hG4bKvtgb6f1048h1g6l9s890.1
Max-Forwards: 67
From: "Anonymous" ;tag=as471a1f75
To: ;tag=as199dc3d1
Contact: 
Call-ID: 3510f7167e1a0f6a5423234b1d176a8b@10.44.###.###:5060
CSeq: 102 ACK
User-Agent: packetrino
Content-Length: 0
Route: 
Route: 


--
^C


On 1/16/22 3:16 PM, Ovidiu Sas wrote:

Use your 209.x external IP.

On Sun, Jan 16, 2022 at 18:07 Chad mailto: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


 > 
>
 >
 > -ovidiu
 >
 > On Sun, Jan 16, 2022 at 16:09 Chad 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 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
 >

Re: [SR-Users] Help with rewriting headers for NAT manually


If I use my external IP do I turn off enable_double_rr?

--
^C


On 1/16/22 3:16 PM, Ovidiu Sas wrote:

Use your 209.x external IP.

On Sun, Jan 16, 2022 at 18:07 Chad mailto: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


 > 
>
 >
 > -ovidiu
 >
 > On Sun, Jan 16, 2022 at 16:09 Chad 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 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:
 >

Re: [SR-Users] Help with rewriting headers for NAT manually


I have been reading a lot more about the problem and it seems my 
mangle/unmangle solution is basically B2BUA.
So I need a B2BUA solution and it seems like Kamailio does not really do B2BUA.
Instead of installing something else I don't know (SEMS or Sippy), it makes more sense to find something that can handle 
it all.
I have read that opensips has B2BUA functionality built in, so I am seriously considering simply replacing Kamailio with 
opensips.
In reality my system has such a low load I can probably replace Kamailio with Asterisk as a B2BUA and it would be fine, 
but from what I have read Asterisk is very inefficient for B2BUA.


--
^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 



-ovidiu

On Sun, Jan 16, 2022 at 16:09 Chad 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 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: 
 >>> 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

Re: [SR-Users] Help with rewriting headers for NAT manually

2022-01-16 Thread Ovidiu Sas

Use your 209.x external IP.

On Sun, Jan 16, 2022 at 18:07 Chad  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>> 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  > 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: 
> >  >>> 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

Re: [SR-Users] Help with rewriting headers for NAT manually

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 



-ovidiu

On Sun, Jan 16, 2022 at 16:09 Chad 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 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: 
 >>> 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

Re: [SR-Users] Help with rewriting headers for NAT manually


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  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: 
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  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: 
Record-Route:

Re: [SR-Users] Help with rewriting headers for NAT manually

2022-01-16 Thread Ovidiu Sas

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  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: 
> > 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  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: 
> >> Record-Route: 
> >> Record-Route: 
> >> From: "Anonymous" ;tag=as04035ef0
> >> To: ;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, 
>

Re: [SR-Users] Help with rewriting headers for NAT manually

e same point, I just added a little bit more information for you to fight with your SIP 
provider. It is better to ask what RFC they are following on the basis of which they are writing the Kamailio public IP 
in the 200 OK ACK message?


Regards,
Shah Hussain


*From:* sr-users  on behalf of Chad 

*Sent:* Sunday, January 16, 2022 10:14 AM
*To:* Ovidiu Sas 
*Cc:* Kamailio (SER) - Users Mailing List 
*Subject:* Re: [SR-Users] Help with rewriting headers for NAT manually
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: 
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  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 

<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 
<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: 
Record-Route: 
Record-Route: 
From: "Anonymous" ;tag=as04035ef0
To: ;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: 
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/8

Re: [SR-Users] Help with rewriting headers for NAT manually

2022-01-16 Thread Shah Hussain Khattak

Hi There,

If I look at the latest SIP trace you shared:

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: 
Record-Route: 
Record-Route: 
From: "Anonymous" ;tag=as04035ef0
To: ;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: 
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" ;tag=as04035ef0
To: ;tag=as7047ed05
Contact: 
Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###:5060
CSeq: 102 ACK
User-Agent: packetrino
Content-Length: 0
Route: 
Route: 

The ACK is getting sent to the Kamailio with correct Route information:

Route: 
Route: 

The Kamailio server should strip the 1st and 2nd Route(s) header from the ACK 
and should relay it towards the next-hop as per the request URI. Please note 
that Kamailio is sending Double RR headers ( When a Proxy receives a request on 
one network interface and sends it onwards using a different interface e.g. WAN 
to LAN, this will normally require the addition of an extra Record-Route 
header. i.e. the Proxy must add two RR headers where you might normally expect 
it to add one.)

Record-Route: 
Record-Route: 

The problem is, the peer behavior is not compliant with the specs. It is 
sending the ACK with RURI set to:

ACK sip:928###@209.###.###.###:5060 SIP/2.0

Ideally, it should have sent the ACK with the following Request-URI:

 ACK sip:928###@10.###.###.104:5060 SIP/2.0

Once this ACK will be received on Kamailio, it will relay it towards the 
Asterisk IP, which is 10.###.###.104.

For further understanding of the ACK routing, you can refer to the following 
post:

https://lists.cs.columbia.edu/pipermail/sip-implementors/2019-February/031229.html

The peer is not copying the 200 OK Contact header URI into the ACK message and 
it is a problem.

Lastly, the trace might be showing only part of the puzzle, it is also 
suggested to get a capture on the remote peer end, because it is sending the 
209.###.###.### IP in the ACK, which seems to be the public interface of the 
Kamailio server. I am not sure if there is some device in the path, that is 
changing the contact IP in the 200 OK to the Kamailio public IP?

Ovidiu also explained the same point, I just added a little bit more 
information for you to fight with your SIP provider. It is better to ask what 
RFC they are following on the basis of which they are writing the Kamailio 
public IP in the 200 OK ACK message?

Regards,
Shah Hussain

From: sr-users  on behalf of Chad 

Sent: Sunday, January 16, 2022 10:14 AM
To: Ovidiu Sas 
Cc: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Help with rewriting headers for NAT manually

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: 
> The carrier should set the RURI in ACK like this:
> ACK sip:928###@10.###

Re: [SR-Users] Help with rewriting headers for NAT manually

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: 
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  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: 
Record-Route: 
Record-Route: 
From: "Anonymous" ;tag=as04035ef0
To: ;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: 
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" ;tag=as04035ef0
To: ;tag=as7047ed05
Contact: 
Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###:5060
CSeq: 102 ACK
User-Agent: packetrino
Content-Length: 0
Route: 
Route: 


--
^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

Re: [SR-Users] Help with rewriting headers for NAT manually

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: 
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  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: 
> Record-Route: 
> Record-Route: 
> From: "Anonymous" ;tag=as04035ef0
> To: ;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: 
> 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" ;tag=as04035ef0
> To: ;tag=as7047ed05
> Contact: 
> Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###:5060
> CSeq: 102 ACK
> User-Agent: packetrino
> Content-Length: 0
> Route: 
> Route: 
>
>
> --
> ^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  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

Re: [SR-Users] Help with rewriting headers for NAT manually


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: 
Record-Route: 
Record-Route: 
From: "Anonymous" ;tag=as04035ef0
To: ;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: 
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" ;tag=as04035ef0
To: ;tag=as7047ed05
Contact: 
Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###:5060
CSeq: 102 ACK
User-Agent: packetrino
Content-Length: 0
Route: 
Route: 


--
^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  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  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:

Re: [SR-Users] Help with rewriting headers for NAT manually

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  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  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  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 >

Re: [SR-Users] Help with rewriting headers for NAT manually

Oops ... copy/paste mistake:
You need to replace:
listen=LISTEN_UDP_PRIVATE advertise MY_PUBLIC_IP:5060
with
listen=LISTEN_UDP_PRIVATE

-ovidiu

On Sat, Jan 15, 2022 at 1:07 PM Ovidiu Sas  wrote:
>
> It doesn't look good because you have the public IP twice in the
> Record-Route header.
> You need to replace the
> listen=LISTEN_UDP_PRIVATE advertise MY_PUBLIC_IP:5060
> with
> listen=LISTEN_UDP_PRIVATE advertise MY_PUBLIC_IP:5060
> and all should be good.
>
> If your carrier is telling you that the IP address in Contact should
> be public, then you need to find an RFC compliant carrier (or mangle
> the Contact to make them happy).
> Most of the time I fight with them until they have this fixed.
> Sometimes it's a lost battle and you just need to hack your config to
> make it work.
>
> I have deployed kamailio using this setup and if you deal with RFC
> compliant end-point (carriers, softphones, hardphones) then all is
> good.
>
> -ovidiu
>
> On Sat, Jan 15, 2022 at 12:14 PM Chad  wrote:
> >
> > It would be great if you are right and I am simply doing something else 
> > wrong in the config file!
> >
> > Here is the 200 OK (note I have enable_double_rr enabled):
> > SIP/2.0 200 OK
> > Via: SIP/2.0/UDP 18.###.###.###:5060;branch=z9hG4bK22b2.6b6d30e5.0
> > Via: SIP/2.0/UDP 66.###.###.###:5060;branch=z9hG4bK22b2.d15ac8a.0
> > Record-Route: 
> > Record-Route: 
> > Record-Route: 
> > From: ;tag=gK0e16642e
> > To: ;tag=as488a6fb4
> > Call-ID: 202251204_54250714@206.###.###.###
> > CSeq: 710596 INVITE
> > Server: Asterisk PBX 16.18.0
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
> > PUBLISH, MESSAGE
> > Supported: replaces, timer
> > Session-Expires: 1800;refresher=uas
> > Contact: 
> > Content-Type: application/sdp
> > Require: timer
> > Content-Length: 272
> >
> > v=0
> > o=root 153822920 153822920 IN IP4 209.###.###.###
> > s=Asterisk PBX 16.18.0
> > c=IN IP4 209.###.###.###
> > t=0 0
> > m=audio 17198 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
> >
> >
> > --
> > ^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  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  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

Re: [SR-Users] Help with rewriting headers for NAT manually

It doesn't look good because you have the public IP twice in the
Record-Route header.
You need to replace the
listen=LISTEN_UDP_PRIVATE advertise MY_PUBLIC_IP:5060
with
listen=LISTEN_UDP_PRIVATE advertise MY_PUBLIC_IP:5060
and all should be good.

If your carrier is telling you that the IP address in Contact should
be public, then you need to find an RFC compliant carrier (or mangle
the Contact to make them happy).
Most of the time I fight with them until they have this fixed.
Sometimes it's a lost battle and you just need to hack your config to
make it work.

I have deployed kamailio using this setup and if you deal with RFC
compliant end-point (carriers, softphones, hardphones) then all is
good.

-ovidiu

On Sat, Jan 15, 2022 at 12:14 PM Chad  wrote:
>
> It would be great if you are right and I am simply doing something else wrong 
> in the config file!
>
> Here is the 200 OK (note I have enable_double_rr enabled):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 18.###.###.###:5060;branch=z9hG4bK22b2.6b6d30e5.0
> Via: SIP/2.0/UDP 66.###.###.###:5060;branch=z9hG4bK22b2.d15ac8a.0
> Record-Route: 
> Record-Route: 
> Record-Route: 
> From: ;tag=gK0e16642e
> To: ;tag=as488a6fb4
> Call-ID: 202251204_54250714@206.###.###.###
> CSeq: 710596 INVITE
> Server: Asterisk PBX 16.18.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
> PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: 
> Content-Type: application/sdp
> Require: timer
> Content-Length: 272
>
> v=0
> o=root 153822920 153822920 IN IP4 209.###.###.###
> s=Asterisk PBX 16.18.0
> c=IN IP4 209.###.###.###
> t=0 0
> m=audio 17198 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
>
>
> --
> ^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  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  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:
>

Re: [SR-Users] Help with rewriting headers for NAT manually


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  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  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  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: 

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

Re: [SR-Users] Help with rewriting headers for NAT manually


It would be great if you are right and I am simply doing something else wrong 
in the config file!

Here is the 200 OK (note I have enable_double_rr enabled):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 18.###.###.###:5060;branch=z9hG4bK22b2.6b6d30e5.0
Via: SIP/2.0/UDP 66.###.###.###:5060;branch=z9hG4bK22b2.d15ac8a.0
Record-Route: 
Record-Route: 
Record-Route: 
From: ;tag=gK0e16642e
To: ;tag=as488a6fb4
Call-ID: 202251204_54250714@206.###.###.###
CSeq: 710596 INVITE
Server: Asterisk PBX 16.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: 
Content-Type: application/sdp
Require: timer
Content-Length: 272

v=0
o=root 153822920 153822920 IN IP4 209.###.###.###
s=Asterisk PBX 16.18.0
c=IN IP4 209.###.###.###
t=0 0
m=audio 17198 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


--
^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  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  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  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: 

There are 2 possible solutions to this:
1. Make changes to linux, keepalived and/or Kamailio so that Kamailio

Re: [SR-Users] Help with rewriting headers for NAT manually

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  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  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  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: 
> 
>  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

Re: [SR-Users] Help with rewriting headers for NAT manually


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  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  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: 

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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Important: keep the mailing list in the recipients, do not reply only to the 
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Edit mailing list options or unsubscribe:
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--
VoIP Embedded, Inc.
http://www.voipembedded.com

__
Kamailio - Users Mailing List - Non Commercial Discussions
* sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the 
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Re: [SR-Users] Help with rewriting headers for NAT manually

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  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  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: 
> >>
> >> 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
> >> 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@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

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Re: [SR-Users] Help with rewriting headers for NAT manually

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 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:

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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--
VoIP Embedded, Inc.
http://www.voipembedded.com

Re: [SR-Users] Help with rewriting headers for NAT manually

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  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: 
>
> 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
> 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

__
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Important: keep the mailing list in the recipients, do not reply only to the 
sender!
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[SR-Users] Help with rewriting headers for NAT manually