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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2185/">https://reviewboard.asterisk.org/r/2185/</a>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<a href="https://reviewboard.asterisk.org/r/2185/diff/1/?file=32116#file32116line4" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">directmedia=yes</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<a href="https://reviewboard.asterisk.org/r/2185/diff/1/?file=32119#file32119line59" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="nt"><recv</span> <span class="na">request=</span><span class="s">"INVITE"</span> <span class="na">optional=</span><span class="s">"true"</span><span class="nt">/></span></pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Why is this here?</pre>
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<p>- Mark</p>
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<p>On November 7th, 2012, 9:46 a.m., Joshua Colp wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Joshua Colp.</div>
<p style="color: grey;"><i>Updated Nov. 7, 2012, 9:46 a.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Ran test to confirm it works, purposely broke test to confirm it fails. Ate some toast. It was crunchy.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/directrtpsetup/run-test <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/tests.yaml <span style="color: grey">(3508)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2185/diff/" style="margin-left: 3em;">View Diff</a></p>
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