[asterisk-dev] Configurable T1 default

[Martin Vít]

Johansson Olle E wrote:
>
> 29 okt 2006 kl. 14.56 skrev Martin Vít:
>
>> Hello,
>>
>> I'had some times ago (with 1.2) issues with gxp2000 that sip qualify 
>> was about 8ms but some actions as invite, reinvite tooks 50ms and 
>> sometime 100-200ms. But when qualify was low it drops packets because 
>> of timeout. Solution was turn off qualify. Maybe there should be 
>> option how qualify affects T1 (fixed, based on qualify or so) Whats 
>> your opinion?
>
> There's a T1 minimum setting that should help this situation.
Yes, but for asterisk >= 1.4 only. So for 1.2 i can not use qualify 
feature because of its unespected behaviour with t1 timer if RTT qualify 
is less then 10ms.

 From RFC 3261:

The default value for T1 is 500 ms.  T1 is an estimate of the RTT
between the client and server transactions.  Elements MAY (though it
is *NOT RECOMMENDED*) use smaller values of T1 within closed, private
networks that do not permit general Internet connection.  T1 MAY be
chosen larger, and this is RECOMMENDED if it is known in advance
(such as on high latency access links) that the RTT is larger.
Whatever the value of T1, the exponential backoffs on retransmissions
described in this section MUST be used.

It would be nice to backport minimun T1 timer into 1.2 as fix. Whats your opinion Olle? 
I think that many people dont realize that qualify may cause this unexpected things. 


>
> I'll look into the possibility to change the default setting, which 
> now is 500 ms according to the RFC.
>
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-- 
Martin Vít
LAM plus s.r.o.
http://www.vasesit.cz/
mobil: 605 267 610

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