[asterisk-dev] [Code Review] DTLS-SRTP Support

[Joshua Colp]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2113/
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(Updated Sept. 19, 2012, 2:19 p.m.)


Review request for Asterisk Developers.


Changes
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Incorporated feedback.


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 (updated)
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  /configure UNKNOWN 
  /trunk/channels/chan_sip.c 373057 
  /trunk/channels/sip/include/sip.h 373057 
  /trunk/configs/sip.conf.sample 373057 
  /trunk/configure.ac 373057 
  /trunk/include/asterisk/autoconfig.h.in 373057 
  /trunk/include/asterisk/rtp_engine.h 373057 
  /trunk/main/rtp_engine.c 373057 
  /trunk/res/res_rtp_asterisk.c 373057 

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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