[asterisk-dev] [Code Review]: 'directrtpsetup' option test

[jcolp]

> On Nov. 7, 2012, 1:49 p.m., Mark Michelson wrote:
> > Have you seen the SIPpTestCase in the test suite? It's tailor-made for tests like this one where you simply need to run some SIPp scenarios. All you have to do is configure it in your test-config.yaml so that the tests are run simultaneously and one is started before the other. I would prefer that this test use the SIPpTestCase. If there were problems using the SIPpTestCase with this test, then I think the SIPpTestCase should be modified to be able to accommodate this sort of test rather than having another run-test file that just runs through SIPp scenarios.

I shall take a gander!


> On Nov. 7, 2012, 1:49 p.m., Mark Michelson wrote:
> > /asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf, line 4
> > <https://reviewboard.asterisk.org/r/2185/diff/1/?file=32116#file32116line4>
> >
> >     Is directmedia=yes required for directrtpsetup to work? If not, then I'd disable it so that there is no potential of Asterisk sending out reinvites to the endpoints.

Yes, it be required yarrrr.


> On Nov. 7, 2012, 1:49 p.m., Mark Michelson wrote:
> > /asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml, line 59
> > <https://reviewboard.asterisk.org/r/2185/diff/1/?file=32119#file32119line59>
> >
> >     Why is this here?

It's possible for a reinvite back to Asterisk to occur, depending on how fast stuff shuts down.


- jcolp


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On Nov. 7, 2012, 9:46 a.m., Joshua Colp wrote:
> 
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> 
> (Updated Nov. 7, 2012, 9:46 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This test covers the 'directrtpsetup' option present within chan_sip. It creates a call going through Asterisk and examines the SDP to ensure that the address for media is of the caller, and not that of Asterisk. It also examines the answer SDP to confirm that the address for media is that of the called party.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 3508 
> 
> Diff: https://reviewboard.asterisk.org/r/2185/diff
> 
> 
> Testing
> -------
> 
> Ran test to confirm it works, purposely broke test to confirm it fails. Ate some toast. It was crunchy.
> 
> 
> Thanks,
> 
> Joshua
> 
>

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