Add SRTP support for Asterisk
authorTerry Wilson <twilson@digium.com>
Tue, 8 Jun 2010 05:29:08 +0000 (05:29 +0000)
committerTerry Wilson <twilson@digium.com>
Tue, 8 Jun 2010 05:29:08 +0000 (05:29 +0000)
commit857814f4354fb26255d4d5db6e06e90749e9bad0
treeecc27fc0db142ea1cd335a74cd1265f993fecd11
parentebbf166c2d15fd233ee307e760b2a88c46d19f6b
Add SRTP support for Asterisk

After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.

Original patch by mikma, updated for trunk and revised by me.

(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel

Review: https://reviewboard.asterisk.org/r/191/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
28 files changed:
CHANGES
build_tools/menuselect-deps.in
channels/chan_iax2.c
channels/chan_sip.c
channels/sip/dialplan_functions.c
channels/sip/include/sdp_crypto.h [new file with mode: 0644]
channels/sip/include/sip.h
channels/sip/include/srtp.h [new file with mode: 0644]
channels/sip/sdp_crypto.c [new file with mode: 0644]
channels/sip/srtp.c [new file with mode: 0644]
configure
configure.ac
doc/tex/asterisk.tex
doc/tex/secure-calls.tex [new file with mode: 0644]
funcs/func_channel.c
include/asterisk/autoconfig.h.in
include/asterisk/frame.h
include/asterisk/global_datastores.h
include/asterisk/res_srtp.h [new file with mode: 0644]
include/asterisk/rtp_engine.h
main/asterisk.exports.in
main/channel.c
main/global_datastores.c
main/rtp_engine.c
makeopts.in
res/res_rtp_asterisk.c
res/res_srtp.c [new file with mode: 0644]
res/res_srtp.exports.in [new file with mode: 0644]