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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1280/">https://reviewboard.asterisk.org/r/1280/</a>
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<div>Review request for Asterisk Developers, Paul Belanger and jrose.</div>
<div>By rmudgett.</div>
<p style="color: grey;"><i>Updated July 12, 2011, 4:36 p.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</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;">Address reviewboard comments.
* Removed unneeded tcpenable and tcpbindaddr in sip.conf files.
* Made authenticate all calls.
* Made ast2 use 127.0.0.2 IP address instead of a different port.
* Made secure_bridge_media test use "channel originate" instead of
"console dial" CLI command for better portability on test machines.</pre>
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<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;">These tests check the ability to get SRTP connections:
1) sip_srtp test establishes a SIP call with SRTP to see if the call can get connected.
2) noload_res_srtp test checks to see if a normal SIP call can still be done when SRTP is not loaded.
3) noload_res_srtp_attemtp_srtp test checks to see if the call will fail if SRTP is not enabled and an incoming call requests it.
4) secure_bridge_media test checks to see if SRTP can be requested dynamically for an outgoing call.</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;">The tests pass and the debug output shows what is expected for each test.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> (updated)</h1>
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<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/modules.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/manager.general.conf.inc <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/modules.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/secure_bridge_media/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/secure_bridge_media/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/secure_bridge_media/configs/ast2/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/secure_bridge_media/configs/ast2/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/secure_bridge_media/run-test <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/secure_bridge_media/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast2/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast2/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_srtp/run-test <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_srtp/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/tests.yaml <span style="color: grey">(1740)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1280/diff/" style="margin-left: 3em;">View Diff</a></p>
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