Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
authorJoshua Colp <jcolp@digium.com>
Thu, 20 Sep 2012 18:27:28 +0000 (18:27 +0000)
committerJoshua Colp <jcolp@digium.com>
Thu, 20 Sep 2012 18:27:28 +0000 (18:27 +0000)
commite8380afc8a147ee299c3881423b2e0b27c4cfc0d
tree9930ca060cafb0821bd7f2d977f1aede33a67877
parentf1fb120f5d62933cac50dc47810418ebf535af7c
Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.

As mentioned on the review for this, WebRTC has moved towards choosing
DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds
support for this but makes it available for normal SIP clients as well.

Testing has been done to ensure that this introduces no regressions with
existing behavior and also that it functions as expected.

Review: https://reviewboard.asterisk.org/r/2113/
........

Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c
channels/sip/include/sip.h
configs/sip.conf.sample
configure
configure.ac
include/asterisk/autoconfig.h.in
include/asterisk/rtp_engine.h
main/rtp_engine.c
res/res_rtp_asterisk.c