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 <mgres...@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 <mgres...@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 <dun...@e-simple.co.nz>: >>>>>> >>>>>> >>>>>>> On 5/09/2021, at 10:21 AM, Marek Greško <mgres...@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.st...@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. >>>>>>>> >>>>>>>> -- >>>>>>>> _____________________________________________________________________ >>>>>>>> -- 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 >> >> -- >> _____________________________________________________________________ >> -- 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
