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

[dotnetdub]

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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>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
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>
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