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: [email protected] [mailto:users- > >> [email protected]] 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:[email protected]:7640;transport=UDP SIP/2.0 > >>> > >>> Via: SIP/2.0/UDP 212.179.159.18;branch=z9hG4bKd6c7.7f7a3d36.0 > >>> > >>> To: <sip:[email protected]:5060>;tag=8548 > >>> > >>> From: <sip:[email protected]:5061>;tag=8547 > >>> > >>> CSeq: 2 BYE > >>> > >>> Call-ID: [email protected] > >>> > >>> 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:[email protected]:5060>;tag=8547 > >>> > >>> From: <sip:[email protected]:5061>;tag=8546 > >>> > >>> CSeq: 2 BYE > >>> > >>> Call-ID: [email protected] > >>> > >>> Route: <sip:[email protected]: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:[email protected]:5061 To:sip:[email protected]:5060 > >>> > >>> | |(5061) ------------------> (5060) | > >>> > >>> |14.003 | INVITE SDP ( MP4V-ES) |SIP From: > >>> sip:[email protected]:5061 To:sip:[email protected]:5060 > >>> > >>> | |(5061) ------------------> (5060) | > >>> > >>> |15.005 | INVITE SDP ( MP4V-ES) |SIP From: > >>> sip:[email protected]:5061 To:sip:[email protected]: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. > >>> ------------------------------------------------------------------- > -- > >>> > >> --- > >> > >>> _______________________________________________ > >>> Users mailing list > >>> [email protected] > >>> 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. > > > > > > > > _______________________________________________ > > Users mailing list > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Bogdan-Andrei Iancu > OpenSIPS Bootcamp > 20 - 24 September 2010, Frankfurt, Germany > www.voice-system.ro > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > This mail was received via Mail-SeCure System. 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