[asterisk-users] problems with natted phones

[Marek Greško]

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.

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.
>
> --
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>
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