From e8380afc8a147ee299c3881423b2e0b27c4cfc0d Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Thu, 20 Sep 2012 18:27:28 +0000 Subject: 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 --- include/asterisk/autoconfig.h.in | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include/asterisk/autoconfig.h.in') diff --git a/include/asterisk/autoconfig.h.in b/include/asterisk/autoconfig.h.in index 9288cce54..e6835a030 100644 --- a/include/asterisk/autoconfig.h.in +++ b/include/asterisk/autoconfig.h.in @@ -527,6 +527,9 @@ /* Define to 1 if you have the OpenSSL Secure Sockets Layer library. */ #undef HAVE_OPENSSL +/* Define to 1 if CRYPTO has the OpenSSL SRTP Extension Support feature. */ +#undef HAVE_OPENSSL_SRTP + /* Define this to indicate the ${OSPTK_DESCRIP} library */ #undef HAVE_OSPTK -- cgit v1.2.3