[OpenSIPS-Users] dialog bye_on_timeout and other issues

Alex Massover alex at jajah.com
Mon Jun 21 15:16:07 CEST 2010 

One more observation. In my test UAS answers with 180 and 200OK to INVITE without delay, i.e. the delay between 180 and 200 is only about some ms.

BAD scenario (rare):
180 is not processed by OpenSIPS - received from UAS but not forwarded to UAC
route_set appears at dialog level
BYE contains Route header and is corrupted

GOOD scenario (common):
180 is processed - received and forwarded
Route_set does not appears at dialog level
BYE does not contain Route header and is not corrupted


After putting 2s delay between 180 and 200 in UAS the BAD scenario is not reproduced anymore.

BTW, yes, I intercept BYE with local route to account it.


> -----Original Message-----
> From: users-bounces at lists.opensips.org [mailto:users-
> bounces at lists.opensips.org] On Behalf Of Alex Massover
> Sent: יום ב 21 יוני 2010 15:52
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] dialog bye_on_timeout and other issues
> 
> 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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> > >
> >
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> >
> 
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