[asterisk-dev] [Code Review] 3350: Add AES-GCM support for SRTP

Matt Jordan reviewboard at asterisk.org
Thu Mar 13 23:12:58 CDT 2014


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Ship it!


Ship It!

- Matt Jordan


On March 13, 2014, 12:54 p.m., Kristian Kielhofner wrote:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3350/
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> (Updated March 13, 2014, 12:54 p.m.)
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> 
> Review request for Asterisk Developers.
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> Bugs: ASTERISK-22832
>     https://issues.asterisk.org/jira/browse/ASTERISK-22832
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> Repository: Asterisk
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> Description
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> There is a version of libsrtp that supports AES-NI and AES-GCM mode:
> https://github.com/cisco/libsrtp/pull/34
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> 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
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> 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.
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> "Big AES" (including 256 GCM) should probably be implemented with a separate patch/bug/review:
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> http://tools.ietf.org/html/rfc6188
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> Diffs
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>   /trunk/res/res_srtp.c 402525 
>   /trunk/main/sdp_srtp.c 402525 
>   /trunk/include/asterisk/sdp_srtp.h 402525 
>   /trunk/include/asterisk/res_srtp.h 402525 
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> Diff: https://reviewboard.asterisk.org/r/3350/diff/
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> Testing
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> Successfully tested call setup and audio exchange with patched pjsip client and FreeSWITCH.
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> 
> Thanks,
> 
> Kristian Kielhofner
> 
>

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