[asterisk-users] Timer1 RFC and SIP.CONF

[Olle E. Johansson]

6 jul 2012 kl. 09:29 skrev Elliot Murdock:

> Hello,
> 
> Thank you for the clarification.
> 
> Just a few questions:
> 1. What is the Timer1 used for?
Timer1 is the base for many other SIP timers and it's an estimate of the roundtrip time for a packet
between two SIP devices or servers. TimerB is based on T1, like the retransmit timers.

> 
> 2. Since timerb is for all responses, final and provisional, the
> comment in sip.conf perhaps should point that out instead of implying
> only for provisional responses: "If a provisional response is not
> received in this amount of time, the call will autocongest"
Yes, that should propably change.

/O
> 
> Thanks,
> Elliot
> 
> On Thu, Jul 5, 2012 at 12:31 PM, Olle E. Johansson <oej at edvina.net> wrote:
>> 
>> 4 jul 2012 kl. 13:32 skrev Elliot Murdock:
>> 
>>> Hello,
>>> 
>>> I am trying to get clarity with the sip.conf timer configuration.  The
>>> current configuration states:
>>> 
>>> ;--------------------------- SIP timers
>>> ----------------------------------------------------
>>> ; These timers are used primarily in INVITE transactions.
>>> ; The default for Timer T1 is 500 ms or the measured run-trip time between
>>> ; Asterisk and the device if you have qualify=yes for the device.
>>> ;
>>> ;t1min=100                      ; Minimum roundtrip time for messages
>>> to monitored hosts
>>>                               ; Defaults to 100 ms
>>> ;timert1=500                    ; Default T1 timer
>>>                               ; Defaults to 500 ms or the measured round-trip
>>>                               ; time to a peer (qualify=yes).
>>> ;timerb=32000                   ; Call setup timer. If a provisional
>>> response is not received
>>>                               ; in this amount of time, the call
>>> will autocongest
>>>                               ; Defaults to 64*timert1
>>> 
>>> However, according to RFC 3261:
>>> 
>>> (EXCERPT 17.1.1.1)
>>> T1 is an estimate of the round-trip time (RTT), and
>>>  it defaults to 500 ms.  Nearly all of the transaction timers
>>>  described here scale with T1, and changing T1 adjusts their values.
>>>  The request is not retransmitted over reliable transports.  After
>>>  receiving a 1xx response, any retransmissions cease altogether, and
>>>  the client waits for further responses.  The server transaction can
>>>  send additional 1xx responses, which are not transmitted reliably by
>>>  the server transaction.  Eventually, the server transaction decides
>>>  to send a final response.
>>> 
>>> (EXCERPT 13.2.2.4 2xx Responses)
>>> The UAC core considers the INVITE transaction completed 64*T1 seconds
>>>  after the reception of the first 2xx response.
>>> 
>>> According to the RFC, the 64*t1 timeout is not for provisional
>>> responses, but for final responses.  This seems to be in contradiction
>>> to what is stated in the sip.conf file.
>> 
>> Unless you have PRACK support, which Asterisk not yet has, there's
>> no timeout in the SIP protocol for provisional responses more than
>> the need to update your provisional response at least every minute
>> not to hit a 3 minute timeout in the SIP transaction state in a proxy.
>> 
>> Now, the timerb is used to get ANY response from a server out there,
>> including provisional responses. If we don't get ANY response within
>> TIMERB, the SIP transaction dies and in a UA with a transaction
>> layer we generate a local 408 timeout. In Asterisk, we congest the call.
>> 
>> So the 64*T1 is for any response, including final response. It's there
>> to decide whether or not you have intelligent SIP life forms handling
>> your SIP request in the network universe.
>> 
>> I hope this clears up your confusion.
>> 
>> Regards,
>> /Olle
>> --
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* Olle E Johansson - oej at edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden