[Freeswitch-users] Asterisk / Proxy / Freeswitch reinvites race condition

Tue Feb 4 03:02:43 MSK 2014

So, it's doing it because the cseq is out of order not because its a
request pending.
That clarifies the reason for the 500 so that may alter OEJ's hypothesis.

On Mon, Feb 3, 2014 at 5:44 PM, dotnetdub <dotnetdub at gmail.com> wrote:

> Hi Anthony - thanks - that is quite another level of logging! Wow..
>
> So the problem freeswitch is giving out about is this:
>
> nta.c:2967 agent_recv_request() nta: INVITE (102) going to existing leg
> nta.c:1341 set_timeout() nta: timer shortened to 200 ms
> nta.c:5441 incoming_callback() nta_leg(0xb6656790): out-of-order
> INVITE (102 < 103)
>
> It then sends the 500
>
> The CSEQ on the reinvite is 102 - the CSEQ on the UPDATE that asterisk
> sent just before was 103 ....
>
> Does this help or do you require more information.
>
> Many Thanks
> Brian
>
>
>
>
>
> On 3 February 2014 22:50, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> > What you want is logs from FS side of things to try to catch more info.
> >
> > sofia global siptrace on
> > sofia tracelevel alert
> > sofia loglevel all 9
> > console loglevel debug
> >
> > And you have to catch the problem at the beginning of when it first
> starts
> > happening.
> >
> >
> >
> >
> >
> >
> > On Mon, Feb 3, 2014 at 1:08 PM, dotnetdub <dotnetdub at gmail.com> wrote:
> >>
> >> Hi Mike,
> >>
> >> Thanks for looking at this.
> >>
> >> I have the log level at 7 and this is all that appears immediately
> >> before the invite and 500
> >>
> >> [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/012345678
> >> [BREAK]
> >>
> >> I can't see anything else relating to the reinvite.
> >>
> >> Thanks
> >> Brian
> >>
> >>
> >> On 3 February 2014 16:20, Michael Jerris <mike at jerris.com> wrote:
> >> > Have you looked at the debug log from freeswitch to see why it is
> >> > sending the 500?
> >> >
> >> > On Feb 2, 2014, at 7:35 AM, dotnetdub <dotnetdub at gmail.com> wrote:
> >> >
> >> >> Hi All,
> >> >>
> >> >> We are facing an issue where we have a customer using asterisk which
> >> >> insists on sending connected line updates via SIP UPDATES and
> >> >> REINVITES.
> >> >>
> >> >> I have tried to mitigate this on the customer Asterisk by using all
> >> >> the config instructions you would expect to work but there are still
> >> >> some scenarios where asterisk insists on sending REINVITES with
> caller
> >> >> ID update to the trunk. I've also tried a number of things on the
> >> >> freeswitch side to try and persuade Asterisk not to send UPDATES or
> >> >> REINVITES without success...
> >> >>
> >> >> SIP Dialog always starts off well but quickly descends into
>  thousands
> >> >> upon thousands of UPDATES and INVITES
> >> >>
> >> >> Freeswitch sends OK to every UPDATE and an OK to the first REINVITE
> >> >> but then starts sending 500 . Asterisk does ACK these errors but
> >> >> immediately sends another UPDATE (again FS sends ok) and INVITE which
> >> >> again we send back a 500 - you can see where I'm going here.
> >> >>
> >> >> This can end up with 1000s of UPDATE,INVITE,500 going on in a SIP
> >> >> dialog and customer reports it is affecting the call audio while this
> >> >> is happening.. I still am unsure of why the audio is affected, the
> >> >> asterisk invites always have SDP so maybe asterisk is opening/closing
> >> >> the port very quickly or something...
> >> >>
> >> >> I've attached a pastie of the first 48 packets of the SIP Trace. This
> >> >> particular call has 2067 SIP packets , mostly INVITE, 500
> >> >>
> >> >> Olle (OEJ) comments on the Kamailio mailing list:
> >> >>
> >> >> 'If freeswitch believes it already has an open INVITE transaction it
> >> >> should
> >> >> not respond with 500, it should respond with 491 request pending. In
> >> >> that
> >> >> case Asterisk will back off and retry.
> >> >>
> >> >> Please check with the FreeSwitch people and file a bug report so that
> >> >> they
> >> >> can fix this issue. That's the long term solution, all the rest is
> >> >> just quick and
> >> >> dirty fixes. Seems like if this problem is still around, no one filed
> >> >> a bug report.'
> >> >>
> >> >> Should I file a bug report or what are freeswitch users / dev
> thoughts
> >> >> on this?
> >> >>
> >> >> Maybe there is something I can do in config or is this something that
> >> >> I must address with Asterisk... With older versions that some
> >> >> customers use I was able to stop reinvites completely. With version
> 11
> >> >> this functionality is still documented but doesn't work so well...
> >> >>
> >> >> traces are http://pastebin.freeswitch.org/21924 truncated after 48
> >> >> packets.
> >> >>
> >> >> Many Thanks,
> >> >> Brian
> >> >
> >> >
> >> >
> >> >
> _________________________________________________________________________
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> >> >
> >> > 
> >> > 
> >> >
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> >>
> _________________________________________________________________________
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> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
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> >
> >
> >
> > --
> > Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
> >
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> >
> > _________________________________________________________________________
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> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
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> >
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>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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