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 <o...@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 >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users