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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2080/">https://reviewboard.asterisk.org/r/2080/</a>
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<p style="margin-top: 0;">On August 22nd, 2012, 7:35 a.m., <b>lathama</b> wrote:</p>
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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;">Can you expand the SRTP content or link to https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial</pre>
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<p>On August 22nd, 2012, 7:38 a.m., <b>lathama</b> wrote:</p>
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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;">Also note that manager needs enabled as does webenabled. Maybe the webserver should work without the manager at some point.</pre>
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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;">Expand it with what? Really all you have to do is ensure that SRTP support is built and set encryption=yes. The vast majority of what is mentioned on the secure calling tutorial is for SIP TLS setup.</pre>
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<p>- Joshua</p>
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<p>On August 22nd, 2012, 6:34 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 Aug. 22, 2012, 6:34 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;">I've created a wiki page at https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support which describes the WebRTC a bit, along with the configuration items and potential issues that may occur. Any feedback is welcome!</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<p><a href="https://reviewboard.asterisk.org/r/2080/diff/" style="margin-left: 3em;">View Diff</a></p>
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