[asterisk-users] problems with natted phones
Duncan Turnbull
duncan at turnbull.co.nz
Sun Sep 5 19:51:03 CDT 2021
Hi Marek
I didn't understand your setup originally.
Can you confirm this is correct:
You provide asterisk for a number of remote phones. I assume they register
to the asterisk
Asterisk ( 198.51.100.1) <==> Phone Provider ( 192.0.2.0/24 ) <==> Phone (
192.168.100.235 )
Your call that fail is coming from asterisk to the phone offering G711A,
G729, iLBC, GSM, G723 and rtp on port 18892
Its unclear to me still whether the remote provider has a SIP device in
front of the phones or just a firewall. The user agent for the reply is
A540 which I am not familiar with
The call that works shows the Asterisk sending to the internal ip until it
receives rtp from the remote phone from which it learns its address and
port and redirects the rtp to. This is fairly standard
For the case of the call that doesn't work, your asterisk receives the rtp
with the external address but doesn't learn from it.
You haven't provided the full call data including the close down of the
call and the registrations would have been helpful too but no matter.
The question is why your asterisk didn't learn the external address and
port from the received rtp packet
You can look at your logs with debug to see what decisions its making. You
can see if different rtp ports have different results.
Your phone provider has rtp on 5010 unsuccessfully and 5016 successfully.
Your asterisk uses rtp 13786 successfully and fails when using 18892. Is it
possible your firewall is blocking port 18892 and so asterisk never sees
the returned packet and can't learn from it?
In any event you should put your debug on and look at your logs in asterisk
to see what it sees and why it doesn't react to the rtp packet, if it gets
it
Have fun, its all good learning.
On Sun, Sep 5, 2021 at 6:27 PM Marek Greško <mgresko8 at gmail.com> wrote:
> Hello,
>
> regarding the ipv6, you see nothing about that it should be some type
> of ipv6 tunnelling, because also MTU is lower than expected. You
> should not see any ipv6 related communication in the sniff. Phone is
> not aware of it.
>
> The asterisk's static public ip address is 198.51.100.1.
> The remote provider's dynamic nat pool is 192.0.2.0/24. By provider we
> mean internet provider the remote phones are behind. We are not
> complaining about voip provider, we have no problem with that. Only
> communication between asterisk and remote phones behind some internet
> provider. This is the only conversation to look at.
> The phone private address is 192.168.100.235.
>
> Thanks
>
> Marek
>
>
> 2021-09-05 1:11 GMT+02:00, Duncan Turnbull <duncan at e-simple.co.nz>:
> >
> >
> >> On 5/09/2021, at 10:21 AM, Marek Greško <mgresko8 at gmail.com> wrote:
> >>
> >> Hello,
> >>
> >> could you please answer my previous question about anonymizing several
> >> parameters? I have the data ready, but will post after answer. I have
> >> no clue whether I could disclose some important data not deleting
> >> them.
> >>
> >> Regarding sdp, the address will be the internal one, since the phone
> >> is behind nat and it is not aware of the nat. The provider's nat
> >> device is configured as dump nat, no application tweaking is done. So
> >> the asterisk will see the lan address in the sip.
> >>
> > There are two conversations to look at
> > Provider to Asterisk
> > Asterisk to Phone
> > You need the packet captures of both.
> >
> > Your statements are mixing them up
> >
> > I don’t know what you mean by LAN address, that’s an ambiguous term. The
> ip
> > your asterisk receives from the provider should be the providers
> external ip
> > or in the sdp the external address of the media server which may or may
> not
> > be the same device
> >
> >> In the working scenario it is sending rtp packets to the internal
> >> address which is wrong, but after receiving cca 5 rtp packets from the
> >> phone it somehow discovers correct nat ip/port and switches to it. In
> >> non-working scenario it never switches and still sends to the lan
> >> address. Strange there is no audio, even one direction. Another
> >> strange thing is there are 2 phones (different vendors) behind the
> >> same nat and the problem appearance on them is independent, sometimes
> >> the first has problem, sometimes the second and sometimes both.
> >>
> >> The tcpdumps are made on the asterisk side. I have currently no means
> >> of capturing on phone side.
> >>
> >> Marek
> >>
> >> 2021-09-04 23:56 GMT+02:00, Antony Stone
> >> <Antony.Stone at asterisk.open.source.it>:
> >>>> On Saturday 04 September 2021 at 22:13:32, Marek Greško wrote:
> >>>>
> >>>> Hello,
> >>>>
> >>>> I agree my knowledge of SIP itself is poor, but I have quite well
> >>>> general tcp/ip understanding. What sip parameters should be
> >>>> anonymized? How about tag, branch, call-id, cseq values?
> >>>
> >>> Show us your packet captures with meaningful addresses (not necessarily
> >>> accurate ones, but at least unambiguous - see my previous suggestion re
> >>> RFC5737) and we can help you to understand them and what they mean.
> >>>
> >>>
> >>> Antony.
> >>>
> >>> --
> >>> Heisenberg, Gödel, and Chomsky walk in to a bar.
> >>> Heisenberg says, "Clearly this is a joke, but how can we work out if
> it's
> >>> funny or not?"
> >>> Gödel replies, "We can't know that because we're inside the joke."
> >>> Chomsky says, "Of course it's funny. You're just saying it wrong."
> >>>
> >>> Please reply to the
> >>> list;
> >>> please *don't*
> CC
> >>> me.
> >>>
> >>> --
> >>> _____________________________________________________________________
> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>>
> >>> Check out the new Asterisk community forum at:
> >>> https://community.asterisk.org/
> >>>
> >>> New to Asterisk? Start here:
> >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >>>
> >>> asterisk-users mailing list
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> >>
> >> --
> >> _____________________________________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> Check out the new Asterisk community forum at:
> >> https://community.asterisk.org/
> >>
> >> New to Asterisk? Start here:
> >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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