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
52 lines
1.1 KiB
C
52 lines
1.1 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2006 - 2007, Mikael Magnusson
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*
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* Mikael Magnusson <mikma@users.sourceforge.net>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file sip_srtp.c
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*
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* \brief SIP Secure RTP (SRTP)
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*
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* Specified in RFC 3711
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*
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* \author Mikael Magnusson <mikma@users.sourceforge.net>
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/utils.h"
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#include "include/srtp.h"
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struct sip_srtp *sip_srtp_alloc(void)
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{
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struct sip_srtp *srtp;
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srtp = ast_calloc(1, sizeof(*srtp));
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return srtp;
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}
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void sip_srtp_destroy(struct sip_srtp *srtp)
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{
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if (srtp->crypto) {
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sdp_crypto_destroy(srtp->crypto);
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}
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srtp->crypto = NULL;
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ast_free(srtp);
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}
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