[asterisk-dev] [Code Review] SDP Media Attribute Test

Matt Jordan reviewboard at asterisk.org
Tue Jul 10 17:23:28 CDT 2012


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/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test
<https://reviewboard.asterisk.org/r/2029/#comment12650>

    To go off of Mark's comment a bit here...
    
    By injecting the TestCase object into the SIPpScenario objects, you guarantee that if the scenario fails, the entire test will fail.  The SIPpScenario object will set passed to False, then call stop_reactor.
    
    If all of the test pass/fail criteria can be fulfilled by the pass/fail status of the SIPp scenarios, then this is sufficient, and you don't need __check_scenario_a/__check_scenario_b.  Usually, you only need intermediate callbacks in one of the following scenarios:
    (a) execute the next scenario, as you're doing here,
    (b) verify AMI events or some other Asterisk information in between each scenario, or
    (c) set the overall test passed status to False if a SIPpScenario failed, but continue to execute subsequent SIPpScenarios in order to gain additional test coverage.  This is useful if the scenarios are relatively independent, and you want to gain as much information as possible if a test fails
    
    



/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test
<https://reviewboard.asterisk.org/r/2029/#comment12648>

    the start_asterisk/stop_asterisk calls are no longer necessary


- Matt


On July 10, 2012, 2:30 p.m., Joshua Colp wrote:
> 
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> (Updated July 10, 2012, 2:30 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This test ensures that attribute information is transported through Asterisk.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_speex.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_speex.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 3307 
> 
> Diff: https://reviewboard.asterisk.org/r/2029/diff
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> 
> Testing
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> Ran it, checked to make sure it worked okay. Tweaked test to fail and made sure it failed as expected.
> 
> 
> Thanks,
> 
> Joshua
> 
>

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