> 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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>>>>>>>>>>> --
>>>>>>>>>>>
>>>>>>>>>>> Check out the new Asterisk
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>>>>>>>>>>> https://community.asterisk.org/
>>>>>>>>>>>
>>>>>>>>>>> New to Asterisk? Start here:
>>>>>>>>>>>
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>>>>>>>>>>>
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>>>>>>>>>>> To UNSUBSCRIBE or update options
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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
>>>>>>>>>>
>>>>>>>>>> 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
>>>>> To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>
> --
> _____________________________________________________________________
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>
> 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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