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 would expect asterisk itself is pairing the provider
address with the lan address. I was asked to disable all the SIP ALG
on the provider's router in the previous discussion. And it made a big
improvement in the experience.
Marek
2021-09-03 12:19 GMT+02:00, Duncan Turnbull <duncan at
turnbull.co.nz>:> On Fri, Sep 3, 2021 at 8:47 PM Marek Greško <mgresko8 at gmail.com>
wrote:
>
>> Hello,
>>
>> I looked into tcpdumps. When problem starts (after some asterisk
>> reboot) the call looks like this:
>>
>> provider:25298 -> asterisk:5060
>> SIP: SIP/2.0 200 OK
>> Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=...
>> From: <sip:111 at asterisk>;tag=...
>> To: <sip:999 at provider>;tag=...
>> Call-ID: ...
>> CSeq: ... INVITE
>> Contact: <sip:999 at lan:5060>
>> Supported: replaces
>> Allow-Events: message-sumary, refer, ua-profile, talk, check-sync
>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS. INFO, SUBSCRIBE, NOTIFY,
REFER,
>> UPDATE
>> Content-Type: application/sdp
>> Content-Length: ...
>>
>> v=0
>> o=... 5010 ... IN IP4 lan
>> s=Mapping
>>
> This bit here tells where the rtp has to go to. I don't think you want
it
> to be IP4 lan. It would be a lot more helpful if you had the ip address but
> the use of the word LAN suggests its a private IP which asterisk is not
> going to be able to route to
>
>
>> c=IN IP4 lan
>> t=0 0
>> m=audio 5010 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=sendrcv
>> a=ptime:20
>>
>> asterisk:5060 -> provider:25298
>> Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=...
>> From: <sip:111 at asterisk>;tag=...
>> To: <sip:999 at provider>;tag=...
>> Call-ID: ...
>> CSeq: ... ACK
>> Max-Forwards: 70
>> User-Agent: Asterisk PBX 18.2.0
>> Content-Length: 0
>>
>> Then I see RTP packets:
>> asterisk:18892 -> lan:5010
>> provider:25420 -> asterisk:18892
>>
> As above for RTP to work they have to go to/from the end points. Asterisk
> is sending to 18892 instead of the provider 25420
>
> Why is your provider sending you an sdp with rtp with a private ip address?
> Or are they sending the right address and your ALG or something else is
> changing it? Ask your provider what they are sending you? Then find out
> who/what is messing up the SDP
>
>
>>
>> I hear no audio. I heard stream towards the asterisk prior to SIP ALG
>> disabling. Now silence both directions. It should not be a codec
>> problem. After providers router reboot I can hear both directions but
>> it still seems weird:
>>
>> provider:32260 -> asterisk:5060
>> SIP: SIP/2.0 200 OK
>> Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=...
>> From: <sip:111 at asterisk>;tag=...
>> To: <sip:999 at provider>;tag=...
>> Call-ID: ...
>> CSeq: ... INVITE
>> Contact: <sip:999 at lan:5060>
>> Supported: replaces
>> Allow-Events: message-sumary, refer, ua-profile, talk, check-sync
>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS. INFO, SUBSCRIBE, NOTIFY,
REFER,
>> UPDATE
>> Content-Type: application/sdp
>> Content-Length: ...
>>
>> v=0
>> o=... 5016 ... IN IP4 lan
>> s=Mapping
>> c=IN IP4 lan
>> t=0 0
>> m=audio 5016 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=sendrcv
>> a=ptime:20
>>
>> asterisk:5060 -> provider:32260
>> Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=...
>> From: <sip:111 at asterisk>;tag=...
>> To: <sip:999 at provider>;tag=...
>> Call-ID: ...
>> CSeq: ... ACK
>> Max-Forwards: 70
>> User-Agent: Asterisk PBX 18.2.0
>> Content-Length: 0
>>
>> Then I see several RTP packets:
>> asterisk:13786 -> lan:5016
>> provider:32327 -> asterisk:13786
>> for a while and the suddenly
>> asterisk:13786 -> provider:32327
>> provider:32327 -> asterisk:13786
>>
>> The user experience for that scenario is OK.
>>
>> I suspect some configuration error on asterisk side, since also for
>> working scenario I see RTP packets to the lan. But I cannot figure out
>> what it is. When I was using another provider which had working SIP
>> ALG I had no problem even without nat configuration on the asterisk
>> side.
>>
>> The experience is clearly better after disabling SIP ALG, but we still
>> face problems after asterisk side reboots.
>>
>> Could you point me for what should I look in the asterisk
>> configuration? And why the problems are gone after provider's
router
>> reboot?
>>
>> Thanks
>>
>> Marek
>>
>>
>>
>> 2021-08-13 15:31 GMT+02:00, Duncan Turnbull <duncan at
e-simple.co.nz>:
>> >
>> >>Hello,
>> >>
>> >>it triggered again. Even disabling RTSp ALG did not help. My
fantasy
>> >>ends here. It agains seems to be reboot triggered on asterisk
side.
>> >>Not every one. But there was surely one before it was last
working.
>> >>Reboot of the router on the phone side fixes the problem. Any
other
>> >>suggestions?
>> >>
>> > This is where you use sngrep or tcpdump to look at whats actually
>> > happening on the asterisk box. sngrep is focussed on sip dialogs
and is
>> > probably easier than tcpdump when you are just interested in sip
>> >
>> > If you use sngrep on the asterisk server sip port you will see the
SIP
>> > packet flows for registration and call setups. You can check the
>> > addresses given out for rtp to respond to and the codecs. Is an
address
>> > incorrect? Is a code incorrect? You will see in the session
description
>> > protocol what codecs the client is requesting and what the replies
are
>> >
>> > asterisk works well around the world with many nat scenarios so I
>> > imagine its either config or firewall. A firewall with ALGs is
often
>> > problematic but your log suggests a lack of negotiation of agreed
>> > codecs.
>> >
>> > Good luck, you will learn some interesting things.
>> >
>> >
>> >
>> >>
>> >>Thanks
>> >>
>> >>Marek
>> >>
>> >>
>> >>2021-07-26 9:31 GMT+02:00, Marek Greško <mgresko8 at
gmail.com>:
>> >>> I currently disabled also RTSP ALG and rebooted the
router. Fixed
>> >>> for
>> >>> now. I do not know for how long.
>> >>>
>> >>> Marek
>> >>>
>> >>>
>> >>> 2021-07-26 8:54 GMT+02:00, Marek Greško <mgresko8 at
gmail.com>:
>> >>>> Hmm, back to original problem. My happines was
premature. Today one
>> of
>> >>>> the phones have no audio again. I see packets from
lan segment
>> >>>> again.
>> >>>>
>> >>>> I double checked the router configuration. SIP ALG is
disabled.
>> >>>> There
>> >>>> are also another ALGs present:
>> >>>>
>> >>>> NAT ALG
>> >>>> RTSP ALG
>> >>>> PPTP ALG
>> >>>> IPSEC ALG
>> >>>>
>> >>>> Which of them are neede to be disabled?
>> >>>>
>> >>>> As of my observations these problems are triggered by
reboots on
>> >>>> asterisk side. How could this be related? (I may be
wrong.)
>> >>>>
>> >>>> Thanks
>> >>>>
>> >>>> Marek
>> >>>>
>> >>>>
>> >>>>
>> >>>> 2021-07-23 14:54 GMT+02:00, Marek Greško <mgresko8
at gmail.com>:
>> >>>>> I achieved a partial success adding
--use-compact-form option.
>> >>>>>
>> >>>>> Marek
>> >>>>>
>> >>>>>
>> >>>>> 2021-07-23 13:47 GMT+02:00, Marek Greško
<mgresko8 at gmail.com>:
>> >>>>>> Hello,
>> >>>>>>
>> >>>>>> your suggestion to turn off SIP ALG on
provider's router was
>> probably
>> >>>>>> correct. no problem until now. Thank you very
much.
>> >>>>>>
>> >>>>>> I just found out another issue. I had a pjsue
client in that
>> network
>> >>>>>> which called specific number when turned on.
It was working
>> perfectly
>> >>>>>> with the old provider with working SIP ALG.
But now with this
>> >>>>>> provider
>> >>>>>> and SIP ALG disabled I am not able to make
the call using pjsua
>> >>>>>> client.
>> >>>>>>
>> >>>>>> My pjsua config looks like this:
>> >>>>>> --id sip:ext at asterisk.domain
>> >>>>>> --registrar sip:asterisk.domain
>> >>>>>> --proxy sip:asterisk.domain
>> >>>>>> --outbound sip:asterisk.domain
>> >>>>>> --realm *
>> >>>>>> --username username
>> >>>>>> --password password
>> >>>>>> --null-audio
>> >>>>>> --no-tcp
>> >>>>>> --max-calls=1
>> >>>>>> --no-vad
>> >>>>>>
>> >>>>>> The pjsua client successfully registers but
is unable to call.
>> >>>>>>
>> >>>>>> I see the following:
>> >>>>>> IP address change detected for account 1
>> >>>>>> (localip:5060-->nattedip:newport).
Updating registration (using
>> >>>>>> method
>> >>>>>> 4)
>> >>>>>> Temporary failure in sending Request msg
INVITE/cseq=...., will
>> >>>>>> try
>> >>>>>> next server: Unsupported transport
(PJSIP_EUNSUPTRANSPORT)
>> >>>>>>
>> >>>>>> What could be the problem? How can I convince
pjsue to work
>> correctly
>> >>>>>> behind nat?
>> >>>>>>
>> >>>>>> Thanks
>> >>>>>>
>> >>>>>> Marek
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>> 2021-07-10 11:08 GMT+02:00, Marek Greško
<mgresko8 at gmail.com>:
>> >>>>>>> Hello,
>> >>>>>>>
>> >>>>>>> I just disabled. Currently it is working.
I shloud give it some
>> time
>> >>>>>>> to confirm the problem has gone. Maybe
one month would be enough
>> to
>> >>>>>>> confirm.
>> >>>>>>>
>> >>>>>>> Thanks
>> >>>>>>>
>> >>>>>>> Marek
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> 2021-07-09 20:11 GMT+02:00, Abdenasser
Ghomri
>> >>>>>>> <ghomri.nasser at gmail.com>:
>> >>>>>>>> Yes just disable the SIP ALG and see
if it helps, Thanks.
>> >>>>>>>>
>> >>>>>>>> Best Regards,
>> >>>>>>>>
>> >>>>>>>> On Fri, Jul 9, 2021, 09:10 Antony
Stone <
>> >>>>>>>>Antony.Stone at
asterisk.open.source.it> wrote:
>> >>>>>>>>
>> >>>>>>>>> On Friday 09 July 2021 at
08:47:46, Marek Greško wrote:
>> >>>>>>>>>
>> >>>>>>>>> > Hello,
>> >>>>>>>>> >
>> >>>>>>>>> > yes SIP ALG are anbled on
the router. Should I disable?
>> >>>>>>>>>
>> >>>>>>>>> In my opinion, always.
>> >>>>>>>>>
>> >>>>>>>>> Antony.
>> >>>>>>>>>
>> >>>>>>>>> --
>> >>>>>>>>> I don't know, maybe if we all
waited then cosmic rays would
>> write
>> >>>>>>>>> all
>> >>>>>>>>> our
>> >>>>>>>>> software for us. Of course it
might take a while.
>> >>>>>>>>>
>> >>>>>>>>> - Ron Minnich, Los Alamos
National Laboratory
>> >>>>>>>>>
>> >>>>>>>>>
Please
>> >>>>>>>>> reply
>> to
>> >>>>>>>>> the
>> >>>>>>>>> list;
>> >>>>>>>>>
>> >>>>>>>>> please
>> >>>>>>>>> *don't*
>> >>>>>>>>> CC
>> >>>>>>>>> me.
>> >>>>>>>>>
>> >>>>>>>>> --
>> >>>>>>>>>
>> >>>>>>>>>
>> _____________________________________________________________________
>> >>>>>>>>> -- Bandwidth and Colocation
Provided by
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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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>> >>>>>>>>> To UNSUBSCRIBE or update options
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>>
>>>>>>>>>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
>> >>
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>> >>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
>