[asterisk-users] problems with natted phones

Duncan Turnbull duncan at turnbull.co.nz
Fri Sep 3 05:19:13 CDT 2021 

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
> 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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