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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3350/">https://reviewboard.asterisk.org/r/3350/</a>
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<p style="margin-top: 0;">On March 14th, 2014, 4:12 a.m. UTC, <b>Matt Jordan</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;">Ship It!</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;">I appreciate the enthusiasm but I don't think it's ready quite yet...
The code isn't conditional in any way and as I said various autoconf checks will need to be included to test for feature-openssl support in system libsrtp as well as AES-GCM support in OpenSSL. If I find myself with some time on my hands I may look into both of these but I'm not sure when that will happen (it almost never does, for some reason)!</pre>
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<p>- Kristian</p>
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<p>On March 13th, 2014, 5:54 p.m. UTC, Kristian Kielhofner wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Kristian Kielhofner.</div>
<p style="color: grey;"><i>Updated March 13, 2014, 5:54 p.m.</i></p>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22832">ASTERISK-22832</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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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;">There is a version of libsrtp that supports AES-NI and AES-GCM mode:
https://github.com/cisco/libsrtp/pull/34
More on AES-GCM mode:
http://tools.ietf.org/html/draft-ietf-avtcore-srtp-aes-gcm-10
http://2013.diac.cr.yp.to/slides/gueron.pdf
AES-GCM mode improves the performance of SRTP on systems with and without support for the AES-NI instruction set.
This patch implements 128 bit AES GCM mode with SRTP. Significantly more work will be required to support 192 and 256 bit AES regardless of mode. Various build stuffs will also need to be updated with the required checks for AES-GCM support in libsrtp and OpenSSL.
"Big AES" (including 256 GCM) should probably be implemented with a separate patch/bug/review:
http://tools.ietf.org/html/rfc6188</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;">Successfully tested call setup and audio exchange with patched pjsip client and FreeSWITCH.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/trunk/res/res_srtp.c <span style="color: grey">(402525)</span></li>
<li>/trunk/main/sdp_srtp.c <span style="color: grey">(402525)</span></li>
<li>/trunk/include/asterisk/sdp_srtp.h <span style="color: grey">(402525)</span></li>
<li>/trunk/include/asterisk/res_srtp.h <span style="color: grey">(402525)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3350/diff/" style="margin-left: 3em;">View Diff</a></p>
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