[asterisk-users] Timer1 RFC and SIP.CONF

[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.

Thanks,
Elliot