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asterisk/channels/sip
twilson 9b1a36a294 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: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08 05:29:08 +00:00
..
include Add SRTP support for Asterisk 2010-06-08 05:29:08 +00:00
config_parser.c do all sip registry parsing before transmit_register 2010-05-26 19:46:49 +00:00
dialplan_functions.c Add SRTP support for Asterisk 2010-06-08 05:29:08 +00:00
reqresp_parser.c Add routines for parsing SIP URIs consistently. 2010-04-09 16:04:16 +00:00
sdp_crypto.c Add SRTP support for Asterisk 2010-06-08 05:29:08 +00:00
srtp.c Add SRTP support for Asterisk 2010-06-08 05:29:08 +00:00