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There are a couple of problems with this test: 1) It's quite a bit more complicated than it needs to be. What's actually being tested here is that Asterisk does not send a 503 in addition to a 486 on an INVITE retransmission. This only requires a UA to send and retransmit the INVITE and Asterisk. This should be doable entirely within a SIPp scenario and does not need voxcallcontrol, a Perl AGI load balancer, or grepping any logs. 2) The included sip.conf file is about 95% comments. This makes reviewing the config file much more difficult than it needs to be. I think this test can be accomplished using the SIPpTestCase and defining the test details entirely within test-config.yaml, with no run-test file necessary. If you're unfamiliar with how this is done, there are several examples of this in the testsuite. A simple example can be found at tests/channels/SIP/directrtpsetup/test-config.yaml. In that test, there is a test-modules section that tells the testsuite to use sipp.SIPpTestCase as the main test object for the test (it also has an unnecessary add-test-to-search-path option set. You can ignore that). The corresponding test-object-config provides details about the SIPp scenarios to run. In that test, there are two scenarios run, but I suspect that for your test, you would only need a single scenario to run. If you're curious about what options are available for configuring the SIPpTestCase, you can look in sample-yaml/sipptestcase-config.yaml.sample for some more details. Your test can pass or fail based on whether the SIPp scenario succeeds or fails. - Mark Michelson On Sept. 19, 2014, 8:09 a.m., Torrey Searle wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4006/ > ----------------------------------------------------------- > > (Updated Sept. 19, 2014, 8:09 a.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24335 > https://issues.asterisk.org/jira/browse/ASTERISK-24335 > > > Repository: testsuite > > > Description > ------- > > This is a test for the test suite to reproduce the issue described in > ASTERISK-24335 > > > Diffs > ----- > > /asterisk/trunk/tests/channels/SIP/tests.yaml 5608 > /asterisk/trunk/tests/channels/SIP/invite_retransmit/test-config.yaml > PRE-CREATION > /asterisk/trunk/tests/channels/SIP/invite_retransmit/sipp/A_PARTY.xml > PRE-CREATION > /asterisk/trunk/tests/channels/SIP/invite_retransmit/run-test PRE-CREATION > /asterisk/trunk/tests/channels/SIP/invite_retransmit/configs/ast1/sip.conf > PRE-CREATION > > /asterisk/trunk/tests/channels/SIP/invite_retransmit/configs/ast1/extensions.conf > PRE-CREATION > > Diff: https://reviewboard.asterisk.org/r/4006/diff/ > > > Testing > ------- > > test passes when 4003 patch applied, fails when patch not applied > > > Thanks, > > Torrey Searle > >
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