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This is an automatically generated e-mail. To reply, visit:
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<p style="margin-top: 0;">On July 6th, 2011, 12:37 p.m., <b>Paul Belanger</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1280/diff/2/?file=17222#file17222line10" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/extensions.conf</a>
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(Diff revision 2)
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">;; Give the other side a chance to get its test result</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">;exten => 1000,n,Wait(5)</pre></td>
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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;">Remove</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I would rather leave this comment commented out dialplan code. There is an associated comment with commented out code in the run-test script about not dropping the AGI connection. If that code is uncommented then this dialplan code would be needed.
These comments are more about documenting what should not be done to get a more reliable test.</pre>
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<p style="margin-top: 0;">On July 6th, 2011, 12:37 p.m., <b>Paul Belanger</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1280/diff/2/?file=17223#file17223line7" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf</a>
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(Diff revision 2)
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">tcpenable=no ; Enable server for incoming TCP connections (default is no)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)</pre></td>
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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;">same, can be removed</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">done</pre>
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<p style="margin-top: 0;">On July 6th, 2011, 12:37 p.m., <b>Paul Belanger</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1280/diff/2/?file=17223#file17223line37" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf</a>
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(Diff revision 2)
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">[1000](sip-trunk,my-codecs)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">callerid="Ast1Name" <1000></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">host=127.0.0.1 ; we have a static but private IP address</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> ; No registration allowed</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">port=5060</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">[2000](sip-trunk,my-codecs)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">callerid="Ast2Name" <2000></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">host=127.0.0.1 ; we have a static but private IP address</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> ; No registration allowed</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">port=5061</pre></td>
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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;">I'd like to see 'secret' added, more coverage.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">ok</pre>
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<p style="margin-top: 0;">On July 6th, 2011, 12:37 p.m., <b>Paul Belanger</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1280/diff/2/?file=17225#file17225line5" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">udpbindaddr=127.0.0.1:5061 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)</pre></td>
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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;">Rather then changing the port, increment the loopback adapter.
udpbindaddr=127.0.0.2:5060</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Is there some benefit to using a different IP address rather than attaching to a different port?
Using a different port should cause different code paths to be executed.</pre>
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<p style="margin-top: 0;">On July 6th, 2011, 12:37 p.m., <b>Paul Belanger</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1280/diff/2/?file=17225#file17225line7" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">tcpenable=no ; Enable server for incoming TCP connections (default is no)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)</pre></td>
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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;">remove</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">done</pre>
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<p style="margin-top: 0;">On July 6th, 2011, 12:37 p.m., <b>Paul Belanger</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1280/diff/2/?file=17231#file17231line7" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">tcpenable=no ; Enable server for incoming TCP connections (default is no)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)</pre></td>
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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;">same, remove</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">done</pre>
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<p style="margin-top: 0;">On July 6th, 2011, 12:37 p.m., <b>Paul Belanger</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1280/diff/2/?file=17241#file17241line42" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/secure_bridge_media/run-test</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> self.ast[0].cli_exec("console dial 1000@siptest1")</pre></td>
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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;">convert this to originate, console requires a soundcard and not all remote agents have them.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">This test is about using Set(CHANNEL(secure_bridge_media)=1) to initiate a SIP SRTP call.
I suppose using originate to a local channel might be possible. Not sure.</pre>
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<p>- rmudgett</p>
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<p>On June 24th, 2011, 12:41 p.m., rmudgett wrote:</p>
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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 June 24, 2011, 12:41 p.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;">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> </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">(1679)</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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