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/*
* 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>
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision: 369013 $")
#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);
}
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