asterisk/channels/sip/srtp.c
Terry Wilson 857814f435 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
2010-06-08 05:29:08 +00:00

52 lines
1.1 KiB
C

/*
* 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);
}