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

dotnetdub dotnetdub at gmail.com
Tue Feb 4 10:42:50 MSK 2014


Thanks for your help Anthony / Michael and thanks for making software
so amazing.

Brian


On 4 February 2014 00:02, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 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
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > _________________________________________________________________________
>> >> > Professional FreeSWITCH Consulting Services:
>> >> > consulting at freeswitch.org
>> >> > http://www.freeswitchsolutions.com
>> >> >
>> >> > 
>> >> > 
>> >> >
>> >> > Official FreeSWITCH Sites
>> >> > http://www.freeswitch.org
>> >> > http://wiki.freeswitch.org
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>> >> >
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>> >> >
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>> >> > http://www.freeswitch.org
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
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>> >>
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>> >
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>> >
>> > ☞ http://freeswitch.org/http://cluecon.com/>> > http://twitter.com/FreeSWITCH
>> > ☞ irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+
>> >
>> > ClueCon Weekly Development Call
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>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>
>
>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
>http://freeswitch.org/http://cluecon.com/> http://twitter.com/FreeSWITCH
> ☞ irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+
>
> ClueCon Weekly Development Call
> ☎ sip:888 at conference.freeswitch.org  ☎ +19193869900
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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>
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