[asterisk-dev] [Code Review] 3562: chan_sip: Start session timer at 200, not at INVITE.

[Matt Jordan]

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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:
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> (Updated May 23, 2014, 5:03 a.m.)
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> 
> Review request for Asterisk Developers.
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> Bugs: ASTERISK-22551
>     https://issues.asterisk.org/jira/browse/ASTERISK-22551
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> Repository: Asterisk
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> Description
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> 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
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> 
> Diffs
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>   /branches/1.8/channels/chan_sip.c 414344 
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> Diff: https://reviewboard.asterisk.org/r/3562/diff/
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> Testing
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> Tested against the scenario mentioned above.
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> Before the patch, Asterisk aborts the call during <pause milliseconds="45000"/>
> After the patch, the scenario completes succesfully.
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> The scenario tests both:
> - asterisk not killing the dialog too early, and
> - killing it when expected
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> After the scenario and a reasonable time, no excess 'sip show objects' were seen.
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> 
> Thanks,
> 
> wdoekes
> 
>

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