----------------------------------------------------------- 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
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev