[OpenSIPS-Users] dialog bye_on_timeout and other issues
Alex Massover
alex at jajah.com
Wed Jun 23 10:08:46 CEST 2010
Hi Bogdan,
Have you received the trace and the log I sent?
Also one more observation. The problem is not reproducible with children=1 but is reproducible with children>=1.
I noticed long time ago that children>=1 misbehaves:
a) performance is worse instead of getting better
b) it can hang
c) with big number of children it hangs/deadlocks very easily
I might be mistaken but it looks for me that something is wrong either with interprocessing or with locks or with both of them.
A corrupted BYE is just a symptom of bigger problem.
For example in this scenario 180 is processed within one child and 200 within another child, there's race condition/locks misbehave/something wrong with memory, and as a result something gets corrupted and wrong BYE is sent.
> -----Original Message-----
> From: users-bounces at lists.opensips.org [mailto:users-
> bounces at lists.opensips.org] On Behalf Of Alex Massover
> Sent: יום ג 22 יוני 2010 12:04
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] dialog bye_on_timeout and other issues
>
> Hi,
>
> > You mentioned you generate these calls with SIPP - right ? are all of
> > them the same ?
> >
> [Alex] Yes, they all are the same.
>
> > looking at signalling, we should see why that calls have the
> > callee_route_set (some RR done after your proxy)..
> >
> [Alex] UAS is also a SIPP script. But it's not the reason for OpenSIPS
> to send corrupted BYE anyway ;)
>
> > also, have you checked the patch I sent you?
>
> [Alex]
> Yes, I get no warnings.
>
> > Regards,
> > Bogdan
> >
> > Alex Massover wrote:
> > > Hi Bogdan,
> > >
> > > At signaling level 200 OK to INVITE contains RR header (always):
> > >
> > > Record-Route: <sip:212.179.159.9:7640>;lr
> > >
> > > But at dialog level I have only rare appearance of route set:
> > >
> > > callee_route_set:: <sip:212.179.159.9:7640>;lr
> > >
> > > absolutely most of the dialogs do not have it:
> > >
> > > opensipsctl fifo dlg_list | grep route_set gives:
> > >
> > > caller_route_set::
> > > callee_route_set::
> > > caller_route_set::
> > > callee_route_set::
> > > caller_route_set::
> > > callee_route_set::
> > > caller_route_set::
> > > callee_route_set::
> > > caller_route_set::
> > > callee_route_set::
> > > caller_route_set::
> > > callee_route_set::
> > > caller_route_set::
> > > callee_route_set::
> > > caller_route_set::
> > > callee_route_set::
> > > caller_route_set::
> > > callee_route_set:: <sip:212.179.159.9:7640>;lr
> > > caller_route_set::
> > > callee_route_set::
> > > caller_route_set::
> > > callee_route_set::
> > >
> > > And looks that it corresponds with the corrupted BYEs. Most of the
> > BYEs do not have Route headers and they are not corrupted, but some
> of
> > them have it and they are corrupted.
> > >
> > > And there's no warning after applying the patch.
> > >
> > >
> > >
> > >> -----Original Message-----
> > >> From: users-bounces at lists.opensips.org [mailto:users-
> > >> bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
> > >> Sent: יום ב 21 יוני 2010 15:12
> > >> To: OpenSIPS users mailling list
> > >> Subject: Re: [OpenSIPS-Users] dialog bye_on_timeout and other
> issues
> > >>
> > >> Hi Alex,
> > >>
> > >> First, about the Route hdr - opensips adds a Route hdr in BYE only
> > if
> > >> the dialog (on that specific leg) received a 200 OK INVITE with RR
> > >> header - can you confirm this at (1) signalling level and (2) at
> > dialog
> > >> level (do a dlg_list via MI).
> > >>
> > >> Now, about the bogus BYE - indeed, it is strange - do you use a
> > local
> > >> route for accessing the BYEs requests? Attached is a small
> debugging
> > >> patch - please apply it a nd see if you get any WARNINGs at
> runtime.
> > >>
> > >> Regards,
> > >> Bogdan
> > >>
> > >> --
> > >> Bogdan-Andrei Iancu
> > >> OpenSIPS Bootcamp
> > >> 20 - 24 September 2010, Frankfurt, Germany www.voice-system.ro
> > >>
> > >>
> > >>
> > >> Alex Massover wrote:
> > >>
> > >>> Hi,
> > >>>
> > >>> I have a strange behavior of OpenSIPS 1.6.2. First dialog module
> > >>> _/sometimes/_ sends a wrong bye (generated by dialog module on
> > >>>
> > >> timeout):
> > >>
> > >>> Here’s a correct one:
> > >>>
> > >>> BYE sip:972123456789 at 212.179.159.9:7640;transport=UDP SIP/2.0
> > >>>
> > >>> Via: SIP/2.0/UDP 212.179.159.18;branch=z9hG4bKd6c7.7f7a3d36.0
> > >>>
> > >>> To: <sip:+496925420990 at 212.179.159.18:5060>;tag=8548
> > >>>
> > >>> From: <sip:+972542384166 at 212.179.159.9:5061>;tag=8547
> > >>>
> > >>> CSeq: 2 BYE
> > >>>
> > >>> Call-ID: 8547-15512 at 212.179.159.9
> > >>>
> > >>> Content-Length: 0
> > >>>
> > >>> Max-Forwards: 70
> > >>>
> > >>> And here’s a wrong one:
> > >>>
> > >>> BYE sip:212.179.159.9:7640 SIP/2.0
> > >>>
> > >>> Via: SIP/2.0/UDP 212.179.159.18;branch=z9hG4bKc6c7.7ecb1057.0
> > >>>
> > >>> To: <sip:+496925420989 at 212.179.159.18:5060>;tag=8547
> > >>>
> > >>> From: <sip:+972542384166 at 212.179.159.9:5061>;tag=8546
> > >>>
> > >>> CSeq: 2 BYE
> > >>>
> > >>> Call-ID: 8546-15512 at 212.179.159.9
> > >>>
> > >>> Route: <sip:972123456789 at 212.179.159.9:7640;transport=UDP>
> > >>>
> > >>> Content-Length: 0
> > >>>
> > >>> Max-Forwards
> > >>>
> > >>> In a wrong one there’s Route header inserted (by mistake?) and
> the
> > >>> message is cut at Max-Forwards line. It’s missing “:70\r\n”.
> > >>>
> > >>> Both of the BYEs above I got just by running test with SIPP. This
> > can
> > >>> happen even with single call, not related to stress. I.e. one
> call
> > it
> > >>> might send a correct BYE and another call a corrupted BYE,
> without
> > >>>
> > >> any
> > >>
> > >>> reason, because calls are exactly the same.
> > >>>
> > >>> Another issue is, looks like t_newtran() is unable to handle
> > >>> retransmissions. In this test UAC and UAS are in the same machine
> > >>> (.9), and you can’t see INVITE from OpenSIPS (.18) to UAS because
> > >>>
> > >> it’s
> > >>
> > >>> fragmented.
> > >>>
> > >>> |Time | x.x.x.9 | x.x.x.18 |
> > >>>
> > >>> |13.501 | INVITE SDP ( MP4V-ES) |SIP From:
> > >>> sip:+xxxxxxxxx166 at x.x.x.9:5061 To:sip:+xxxxxxxxxx82 at x.x.x.18:5060
> > >>>
> > >>> | |(5061) ------------------> (5060) |
> > >>>
> > >>> |14.003 | INVITE SDP ( MP4V-ES) |SIP From:
> > >>> sip:+xxxxxxxxx166 at x.x.x.9:5061 To:sip:+xxxxxxxxxx82 at x.x.x.18:5060
> > >>>
> > >>> | |(5061) ------------------> (5060) |
> > >>>
> > >>> |15.005 | INVITE SDP ( MP4V-ES) |SIP From:
> > >>> sip:+xxxxxxxxx166 at x.x.x.9:5061 To:sip:+xxxxxxxxxx82 at x.x.x.18:5060
> > >>>
> > >>> | |(5061) ------------------> (5060) |
> > >>>
> > >>> |15.743 | 100 Trying| |SIP Status
> > >>>
> > >>> | |(5061) <------------------ (5060) |
> > >>>
> > >>> |15.800 | 181 Call is being forwarded |SIP Status
> > >>>
> > >>> | |(5061) <------------------ (5060) |
> > >>>
> > >>> |15.801 | 100 Trying| |SIP Status
> > >>>
> > >>> | |(7640) ------------------> (5060) |
> > >>>
> > >>> |15.801 | 180 Ringing |SIP Status
> > >>>
> > >>> | |(7640) ------------------> (5060) |
> > >>>
> > >>> |15.801 | 200 OK SDP ( G723) |SIP Status
> > >>>
> > >>> | |(7640) ------------------> (5060) |
> > >>>
> > >>> |15.840 | 181 Call is being forwarded |SIP Status
> > >>>
> > >>> | |(5061) <------------------ (5060) |
> > >>>
> > >>> |16.041 | 181 Call is being forwarded |SIP Status
> > >>>
> > >>> | |(5061) <------------------ (5060) |
> > >>>
> > >>> |16.188 | 180 Ringing |SIP Status
> > >>>
> > >>> | |(5061) <------------------ (5060) |
> > >>>
> > >>> |16.188 | 200 OK SDP ( G723) |SIP Status
> > >>>
> > >>> | |(5061) <------------------ (5060) |
> > >>>
> > >>> |16.189 | ACK | |SIP Request
> > >>>
> > >>> | |(5061) ------------------> (5060) |
> > >>>
> > >>> |16.302 | 200 OK SDP ( G723) |SIP Status
> > >>>
> > >>> | |(7640) ------------------> (5060) |
> > >>>
> > >>> |16.357 | ACK | |SIP Request
> > >>>
> > >>> | |(7640) <------------------ (5060) |
> > >>>
> > >>> |16.651 | 200 OK SDP ( G723) |SIP Status
> > >>>
> > >>> | |(5061) <------------------ (5060) |
> > >>>
> > >>> |16.652 | ACK | |SIP Request
> > >>>
> > >>> | |(5061) ------------------> (5060) |
> > >>>
> > >>> |17.075 | ACK | |SIP Request
> > >>>
> > >>> | |(7640) <------------------ (5060) |
> > >>>
> > >>> |36.730 | BYE | |SIP Request
> > >>>
> > >>> | |(5061) <------------------ (5060) |
> > >>>
> > >>> |36.731 | BYE | |SIP Request
> > >>>
> > >>> | |(7640) <------------------ (5060) |
> > >>>
> > >>> |36.731 | 200 OK | |SIP Status
> > >>>
> > >>> | |(5061) ------------------> (5060) |
> > >>>
> > >>> |36.731 | 200 OK | |SIP Status
> > >>>
> > >>> | |(7640) ------------------> (5060) |
> > >>>
> > >>> This issue happens during stress test.
> > >>>
> > >>> Any ideas, please? The OpenSIPS 1.6.2 is compiled with system
> > malloc
> > >>> and runs over VMware.
> > >>>
> > >>> --
> > >>>
> > >>> Best Regards,
> > >>>
> > >>> Alex Massover
> > >>>
> > >>>
> > >>>
> > >>> This mail was sent via Mail-SeCure System.
> > >>> -----------------------------------------------------------------
> --
> > --
> > >>>
> > >> ---
> > >>
> > >>> _______________________________________________
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> > >>> Users at lists.opensips.org
> > >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >>>
> > >>>
> > >> This mail was received via Mail-SeCure System.
> > >>
> > >>
> > >
> > >
> > > This mail was sent via Mail-SeCure System.
> > >
> > >
> > >
> > > _______________________________________________
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> > >
> >
> >
> > --
> > Bogdan-Andrei Iancu
> > OpenSIPS Bootcamp
> > 20 - 24 September 2010, Frankfurt, Germany
> > www.voice-system.ro
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
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