diff options
author | Terry Wilson <twilson@digium.com> | 2010-06-08 05:29:08 +0000 |
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committer | Terry Wilson <twilson@digium.com> | 2010-06-08 05:29:08 +0000 |
commit | 857814f4354fb26255d4d5db6e06e90749e9bad0 (patch) | |
tree | ecc27fc0db142ea1cd335a74cd1265f993fecd11 /channels/sip/srtp.c | |
parent | ebbf166c2d15fd233ee307e760b2a88c46d19f6b (diff) |
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
Diffstat (limited to 'channels/sip/srtp.c')
-rw-r--r-- | channels/sip/srtp.c | 51 |
1 files changed, 51 insertions, 0 deletions
diff --git a/channels/sip/srtp.c b/channels/sip/srtp.c new file mode 100644 index 000000000..3b55106ab --- /dev/null +++ b/channels/sip/srtp.c @@ -0,0 +1,51 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2006 - 2007, Mikael Magnusson + * + * Mikael Magnusson <mikma@users.sourceforge.net> + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*! \file sip_srtp.c + * + * \brief SIP Secure RTP (SRTP) + * + * Specified in RFC 3711 + * + * \author Mikael Magnusson <mikma@users.sourceforge.net> + */ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include "asterisk/utils.h" +#include "include/srtp.h" + +struct sip_srtp *sip_srtp_alloc(void) +{ + struct sip_srtp *srtp; + + srtp = ast_calloc(1, sizeof(*srtp)); + + return srtp; +} + +void sip_srtp_destroy(struct sip_srtp *srtp) +{ + if (srtp->crypto) { + sdp_crypto_destroy(srtp->crypto); + } + srtp->crypto = NULL; + ast_free(srtp); +} |