[asterisk-dev] [Code Review] 4518: testsuite: Add PJSIP test for new rpid_immediate option.

[Corey Farrell]

> On March 21, 2015, 12:54 a.m., Corey Farrell wrote:
> > Is a sipp scenario needed to verify Asterisk sends the correct SIP packets for rpid_immediate on and off?
> 
> rmudgett wrote:
>     A sipp scenario is not needed.  The test is to check *when* connected line is passed not how since that is what the rpid_immediate option controls.

>From r4473 Testing Done:
* Made a call chain as follows: 100 -> * -> * -> * -> 200.  With the patch
there are no unnecessary messages.  Without the patch there were several
"180 Ringing" messages sent back to the caller.

This is what I was asking about, if sipp is the caller it will get a different number of SIP messages depending on rpid_immediate.


- Corey


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On March 23, 2015, 1:21 p.m., rmudgett wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4518/
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> 
> (Updated March 23, 2015, 1:21 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-24781
>     https://issues.asterisk.org/jira/browse/ASTERISK-24781
> 
> 
> Repository: testsuite
> 
> 
> Description
> -------
> 
> Test the rpid_immediate option when enabled and disabled.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/pjsip/tests.yaml 6547 
>   /asterisk/trunk/tests/channels/pjsip/rpid_immediate/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/pjsip/rpid_immediate/configs/ast1/pjsip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/pjsip/rpid_immediate/configs/ast1/extensions.conf PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/4518/diff/
> 
> 
> Testing
> -------
> 
> This is the testsuite test to test the rpid_immediate option added by review:
> https://reviewboard.asterisk.org/r/4473/
> 
> The test passes.
> 
> 
> Thanks,
> 
> rmudgett
> 
>

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