[asterisk-dev] [Code Review] 2796: Testsuite: sip_hold_direct_media adaptations for Asterisk 12

[jrose]

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/asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test
<https://reviewboard.asterisk.org/r/2796/#comment18634>

    NOTE: The first three scenarios from this test were trimmed on account of them not using direct media. They still exist in the SIP/sip_hold test and were never part of the failures.
    
    Speaking of which, I'm going to go ahead and remove INJECT_FILE_BYPASS from runtest and inject_bypass.csv from the sipp folder.



/asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test
<https://reviewboard.asterisk.org/r/2796/#comment18635>

    These two conditions were added to prevent the next test from starting if the previous test didn't pass.  This was leading to events being received when they weren't expected due to the test continuing for a short while after failing.


- jrose


On Aug. 26, 2013, 8:18 p.m., jrose wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
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> 
> (Updated Aug. 26, 2013, 8:18 p.m.)
> 
> 
> Review request for Asterisk Developers, Joshua Colp, kmoore, Matt Jordan, and Mark Michelson.
> 
> 
> Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-22217
>     https://issues.asterisk.org/jira/browse/https://issues.asterisk.org/jira/browse/ASTERISK-22217
> 
> 
> Repository: testsuite
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> 
> Description
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> On Friday I committed a patch which addressed some bugs with holding in Asterisk 12 while using native RTP bridges and directmedia. As part of that effort, the SIP hold tests in the testsuite were split up and divided into tests which used direct media and tests which didn't use direct media. At the time, Asterisk 12 failed the tests which used direct media. This patch fixes those tests by making the test use Asterisk 12 specific sipp scenarios (which were based on the existing scenarios). The main difference between the Asterisk 12 scenarios and their older counterparts was always just the addition of more expected invites and responses to those invites on account of how directmedia is established in Asterisk 12 both during the initial setup.
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> 
> Diffs
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>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_12_media_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_12_IP_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_12_IP_media_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_12_type2.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_12.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test 4044 
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> Diff: https://reviewboard.asterisk.org/r/2796/diff/
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> 
> Testing
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> Ran modified tests against Asterisk 12.
> Ran modified tests against Asterisk 11.
> 
> Repeated this process many times to make sure the results were consistent.
> Tracked the invites in 12 against the code that was generating them.
> 
> 
> Thanks,
> 
> jrose
> 
>

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