[asterisk-users] problems with natted phones

Marek Greško mgresko8 at gmail.com
Mon Sep 6 15:28:41 CDT 2021 

Hello,

it is only local nftables with nf_conntrack_sip on the asterisk
server. Probably a kernel bug? It did not trigger with previous
providers since they had working SIP ALG. Now I hear no audio in both
directions because outgoing rtp stream from asterisk goes to private
address space and incoming stream is blocked. So the outgoing rtp
could not be learnt to send to nat addess.

Marek


2021-09-06 22:17 GMT+02:00, Duncan Turnbull <duncan at turnbull.co.nz>:
>
>
>> On 7/09/2021, at 3:08 AM, Marek Greško <mgresko8 at gmail.com> wrote:
>>
>> Hello,
>>
>> so when debugging RTP in asterisk there was no rtp income from the
>> remote site. I did check remote nat ip address and it was same as the
>> one in the pjsip show aors. So it is not due to ip address change. It
>> seems the local firewall sip module does not allow rtp stream to get
>> into. It was working previously with the other provider because of
>> working SIP ALG on their gateways. But now with this provider and
>> disabled SIP ALG it is not allowed. As I remeber in the past these
>> setups did work. What are your experiences on this?
>>
> You would need to provide a lot more explanation here. What is your
> firewall? I am assuming you configure it so find the configuration that’s
> blocking the ports and change it.
>
> My experience as before was that something is blocking rtp, now you know
> what that something is and it’s under your control so you need to check it’s
> configuration and fix it. I don’t use a sip firewall. If I have external sip
> clients I use a proxy.
>
>> Thanks
>>
>> Marek
>>
>>
>> 2021-09-06 11:50 GMT+02:00, Marek Greško <mgresko8 at gmail.com>:
>>> Sorry rtp set debug on showed something. So let try for the problem to
>>> arise again.
>>>
>>> Marek
>>>
>>>
>>> 2021-09-06 11:48 GMT+02:00, Marek Greško <mgresko8 at gmail.com>:
>>>> Hello,
>>>>
>>>>>> I would expect that when asterisk is aware of nat, it does not send
>>>>>> the rtp until it receives rtp from other side to learn the port, but
>>>>>> OK, no problem to accept the behavior.
>>>>>>
>>>>> That’s not how things work. You should google how sip rtp and Nat work
>>>>> as
>>>>> it
>>>>> will help you
>>>>
>>>> no problem if it is intended.
>>>>
>>>>>>
>>>>>>> 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?
>>>>>>
>>>>>> It is very unprobable. I see no reason for blocking the port. The
>>>>>> problem is asterisk never learns the correct port, so there is nothing
>>>>>> to block.
>>>>> It wasn’t what is probable, look at the asterisk logs and see what it’s
>>>>> actually doing. If asterisk never sees the reply then you will know
>>>>> something is blocking or stealing the port for some other service
>>>>
>>>> If it is stolen port for rtp, the next call would solve it, since it
>>>> will use different one, and it does not solve 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
>>>>>>
>>>>>> Could you point me how the debug should be conducted?
>>>>>
>>>>> Using the asterisk cli turn on debug for the peer and rtp and see what
>>>>> happens. Match it with the asterisk processes. You have to do this, you
>>>>> can
>>>>> look at cli or the log files, follow it through to see the rtp packet
>>>>> being
>>>>> received. Lots of debug advice on google.
>>>>
>>>> Asterisk cli did not show anything interesting. I tried pjsip set
>>>> logger verbose on, but no logs showed anywhere. What am I doing wrong?
>>>>
>>>> Marek
>>>>
>>>>
>>>>>>
>>>>>> Is my suspection that the problem could be caused by nat ip addres
>>>>>> changing reasonable? How should asterisk handle the situation?
>>>>> I can’t see anything to support that. Everything is looking normal
>>>>> except
>>>>> asterisk doesn’t appear to beseeing the rtp packet
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>> Marek
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> 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.
>>>>>>>>>>>>
>>>>>>>>>>>> --
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>>>>>>>>>>>> -- Bandwidth and Colocation Provided by
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>>>>>>>>>>>> --
>>>>>>>>>>>>
>>>>>>>>>>>> Check out the new Asterisk community forum at:
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>>>>>>>>>>>>
>>>>>>>>>>>> New to Asterisk? Start here:
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>>>>>>>>>>>>
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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
>>>>>>>>>>>
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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
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
>>>>>> -- 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
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>>   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
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