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Ship it! Would it be worthwhile to incorporate the SIPp scenario into the Test Suite? - Matt Jordan On May 23, 2014, 5:03 a.m., wdoekes wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3562/ > ----------------------------------------------------------- > > (Updated May 23, 2014, 5:03 a.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-22551 > https://issues.asterisk.org/jira/browse/ASTERISK-22551 > > > Repository: Asterisk > > > Description > ------- > > RFC says Asterisk should start counting session timers at 200. Asterisk > starts at INVITE. > > For short intervals (e.g. 90 secs) and long ringing times (e.g. 30 secs), > this means that > a caller with refresher=uac will get disconnected by Asterisk before it has a > chance to > send a refreshing reINVITE. > > Reproduced using: > https://github.com/ossobv/sipp-scenarios/blob/master/INVITE-test-session-refresher-uac.xml > > > Diffs > ----- > > /branches/1.8/channels/chan_sip.c 414344 > > Diff: https://reviewboard.asterisk.org/r/3562/diff/ > > > Testing > ------- > > Tested against the scenario mentioned above. > > Before the patch, Asterisk aborts the call during <pause > milliseconds="45000"/> > After the patch, the scenario completes succesfully. > > The scenario tests both: > - asterisk not killing the dialog too early, and > - killing it when expected > > After the scenario and a reasonable time, no excess 'sip show objects' were > seen. > > > Thanks, > > wdoekes > >
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