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

[Mark Michelson]

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Ship it!


Excellent!

As a final wrap-up to this, you should probably check the phone B SIPp scenarios being used in the non-direct media hold tests and be sure they are also include appropriate SDPs in their 200 OKs.

- Mark Michelson


On Aug. 28, 2013, 4:37 p.m., jrose wrote:
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> 
> (Updated Aug. 28, 2013, 4:37 p.m.)
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> 
> Review request for Asterisk Developers, Joshua Colp, kmoore, Matt Jordan, and Mark Michelson.
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> Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-22217
>     https://issues.asterisk.org/jira/browse/https://issues.asterisk.org/jira/browse/ASTERISK-22217
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> 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_media_restrict.xml 4089 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_restrict.xml 4089 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_media_restrict.xml 4089 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_IP_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test 4089 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/inject_bypass.csv 4089 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A.xml 4089 
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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.
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> Repeated this process many times to make sure the results were consistent.
> Tracked the invites in 12 against the code that was generating them.
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> 
> Thanks,
> 
> jrose
> 
>

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