[asterisk-users] problems with natted phones

[Duncan Turnbull]

> 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;
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>> me.
>> 
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> -- 
> _____________________________________________________________________
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> 
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
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