[asterisk-users] problems with natted phones

Duncan Turnbull
Sat, 04 Sep 2021 10:36:42 -0000

> On 4/09/2021, at 8:55 PM, Marek Greško <mgresko8 at gmail.com> wrote:
>
> ﻿Ok,
>
> let substitute lan for 192.168.100.235, provider with 192.0.2.1 and
> asterisk with 198.51.100.1.

Can you provide the previous packet details with these addresses filled
in
>
> In the working scenario understand that asterisk is not aware of the
> providers ip address
If the call goes provider - asterisk - phone then asterisk is absolutely aware
of the provider ip. I think you need to get more familiar with sip and rtp

> 192.0.2.1 in the sip protocol, and it should pick
> it from the network layer. It is harder to calcutale port, so it
> should probably listen for incoming rtp stream?

The sdp in the sip packet tells the rtp ip and port to connect
to
> Until then it is just
> sending to private address? But I thing it is futile, since it is
> known from the sip protocol there is nat involved and thus the packets
> are destined to nowhere.

You need to realise that this works normally everyday all over the place so what
you are imagining is incorrect
>
> But I still cannot imagine what goes wrong in non working scenario and
> how the asterisk reboot (not every one and not sure this is the real
> trigger). The sip communication seems identical to me. The provider's
> router does not touch SIP now as observed after disabling SIP ALG.

It is very unclear as to how you are justifying these statements. You don’t yet
understand how sip and call setup with media works. If you provide the whole sip
packet capture with the substituted ips it should be easier to point out where
the error is

You need to be really clear on what’s ip
is what and where the conversations are captured

It will become clear once you provide all the details


>
> Thanks
>
> Marek
>
> 2021-09-04 0:40 GMT+02:00, Antony Stone <Antony.Stone at
asterisk.open.source.it>:
>> On Saturday 04 September 2021 at 00:34:49, Duncan Turnbull wrote:
>>
>>>>> On 4/09/2021, at 7:53 AM, Marek Greško <mgresko8 at
gmail.com> wrote:
>>>>>
>>>>> ﻿So you suspect something is messing up SIP protocol? Maybe
the phone
>>>>> itself is not working properly. The phone itself is not
aware of the
>>>>> internet address, so is sending lan private address in the
sip
>>>>> protocol.
>>>
>>> I doubt it’s the phone. You need to check your data again and
ideally
>>> explain what you mean by the names you have substituted for the ip
>>> addresses
>>
>> My advice (regarding the IP addresses) is:
>>
>> - where you have https://tools.ietf.org/html/rfc1918 addresses, leave
them
>> as
>> they are - you're not giving away any sensitive information by
telling us
>> about your internal addresses which can't be routed over the
Internet
>>
>> - where you have public addresses and would prefer not to reveal what
these
>> are, substitute with https://tools.ietf.org/html/rfc5737 addresses
instead.
>>
>> - always ensure that you substitute address A in the same way each
time,
>> and
>> address B, etc.
>>
>>
>> Antony.
>>
>> --
>> You can spend the whole of your life trying to be popular,
>> but at the end of the day the size of the crowd at your funeral
>> will be largely dictated by the weather.
>>
>> - Frank Skinner
>>
>>                                                   Please reply to the
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>>                                                         please
*don't* CC
>> me.
>>
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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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