[asterisk-dev] [Code Review]: DTLS-SRTP Support
Matt Jordan
reviewboard at asterisk.org
Wed Sep 19 11:20:42 CDT 2012
> On Sept. 19, 2012, 11:14 a.m., Mark Michelson wrote:
> > I couldn't see any archtectural issues (e.g. memory leaks, reference leaks, misuses of pointers, etc.) With regards to OpenSSL usage, I don't have the knowledge to say for sure that it's all correct.
For what its worth, Kinsey did a silly amount of testing to verify that the encryption/decryption of RTP works as intended.
- Matt
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On Sept. 14, 2012, 11:43 a.m., Joshua Colp wrote:
>
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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. 14, 2012, 11:43 a.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Summary
> -------
>
> 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
> -----
>
> /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
> -------
>
> 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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