58 lines
1.4 KiB
C
58 lines
1.4 KiB
C
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/*
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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.h
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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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#ifndef _SIP_SRTP_H
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#define _SIP_SRTP_H
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#include "sdp_crypto.h"
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/* SRTP flags */
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#define SRTP_ENCR_OPTIONAL (1 << 1) /* SRTP encryption optional */
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#define SRTP_CRYPTO_ENABLE (1 << 2)
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#define SRTP_CRYPTO_OFFER_OK (1 << 3)
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/*! \brief structure for secure RTP audio */
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struct sip_srtp {
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unsigned int flags;
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struct sdp_crypto *crypto;
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};
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/*!
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* \brief allocate a sip_srtp structure
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* \retval a new malloc'd sip_srtp structure on success
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* \retval NULL on failure
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*/
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struct sip_srtp *sip_srtp_alloc(void);
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/*!
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* \brief free a sip_srtp structure
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* \param srtp a sip_srtp structure
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*/
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void sip_srtp_destroy(struct sip_srtp *srtp);
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#endif /* _SIP_SRTP_H */
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