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

Kristian Kielhofner reviewboard at asterisk.org
Thu Mar 13 12:54:22 CDT 2014


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3350/
-----------------------------------------------------------

Review request for Asterisk Developers.


Bugs: ASTERISK-22832
    https://issues.asterisk.org/jira/browse/ASTERISK-22832


Repository: Asterisk


Description
-------

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


Diffs
-----

  /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 

Diff: https://reviewboard.asterisk.org/r/3350/diff/


Testing
-------

Successfully tested call setup and audio exchange with patched pjsip client and FreeSWITCH.


Thanks,

Kristian Kielhofner

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140313/f8212efe/attachment.html>


More information about the asterisk-dev mailing list