[asterisk-dev] [Code Review] DTLS-SRTP Support
Joshua Colp
reviewboard at asterisk.org
Fri Sep 14 11:43:40 CDT 2012
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2113/
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Review request for Asterisk Developers.
Summary
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WebRTC has migrated to using DTLS-SRTP as the method for securing media streams. This patch adds support for it using OpenSSL. DTLS is used between both sides with the keying material for SRTP extracted from that negotiation.
Diffs
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/configure UNKNOWN
/trunk/channels/chan_sip.c 373058
/trunk/channels/sip/include/sip.h 373058
/trunk/configs/sip.conf.sample 373058
/trunk/configure.ac 373058
/trunk/include/asterisk/autoconfig.h.in 373058
/trunk/include/asterisk/rtp_engine.h 373058
/trunk/main/rtp_engine.c 373058
/trunk/res/res_rtp_asterisk.c 373058
Diff: https://reviewboard.asterisk.org/r/2113/diff
Testing
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Tested various configurations between two Asterisk instances. Rekeying, verification, etc all appear to work. Unfortunately there are very few DTLS-SRTP implementations in the wild so testing against another implementation has not yet occurred.
Thanks,
Joshua
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