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?
Thanks Marek 2021-09-04 12:36 GMT+02:00, Duncan Turnbull <dun...@turnbull.co.nz>: > > >> On 4/09/2021, at 8:55 PM, Marek Greško <mgres...@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.st...@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 <mgres...@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 >>> 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
