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Mark Michelson wrote:
<blockquote cite="mid:4A3A6F22.4070301@digium.com" type="cite">
<pre wrap="">Kristian Kielhofner wrote:
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<pre wrap="">On Thu, Jun 18, 2009 at 12:03 PM, Mark Michelson <a class="moz-txt-link-rfc2396E" href="mailto:mmichelson@digium.com"><mmichelson@digium.com></a> wrote:
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<pre wrap="">David Hansen wrote:
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<pre wrap="">Are your PBX and phone on same subnet? Is the phone SIP capable and listening on port 5060? You mention T1, but this has no relationship to SIP convos.
David Hansen
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<pre wrap="">In SIP, there is a timer called T1 used as a base for calculating retransmission
times.
In answer to the originally-posted question, the SIP T1 timer defaults to 500
ms. Looking at the code in sip_reliable_xmit, we set a value called siptimer_a
to be 2 * pkt->timer_t1. This value is what is passed to the scheduler for the
retransmission time.
Reading RFC 3261, section 17.1.1.2:
"If an unreliable transport is being
used, the client transaction MUST start timer A with a value of T1.
If a reliable transport is being used, the client transaction SHOULD
NOT start timer A (Timer A controls request retransmissions). For
any transport, the client transaction MUST start timer B with a value
of 64*T1 seconds (Timer B controls transaction timeouts).
When timer A fires, the client transaction MUST retransmit the
request by passing it to the transport layer, and MUST reset the
timer with a value of 2*T1."
So it appears that this may be an instance where we are doing things a bit
out-of-spec here. I wonder if there was some sort of justification for this when
it was written...
Mark Michelson
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<pre wrap="">IIRC the T1 value is calculated based off of whatever qualify returns
(if enabled). I don't remember what happens if you have qualify
disabled...
This may have changed in recent versions but I remember it being
correct at some point in time.
--
Kristian Kielhofner
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<pre wrap=""><!---->
You are correct regarding the use of the qualify time as timer T1 since it is
the estimated RTT.
If you have qualify disabled, then T1 will default to 500 ms. Additionally, in
Asterisk 1.6 versions the initial value for T1 is configurable using the
"timert1" option in sip.conf.
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1.4 versions has t1min parametr<br>
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