asterisk/channels/sip/include/srtp.h

60 lines
1.4 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 srtp.h
*
* \brief SIP Secure RTP (SRTP)
*
* Specified in RFC 3711
*
* \author Mikael Magnusson <mikma@users.sourceforge.net>
*/
#ifndef _SIP_SRTP_H
#define _SIP_SRTP_H
#include "sdp_crypto.h"
/* SRTP flags */
#define SRTP_ENCR_OPTIONAL (1 << 1) /* SRTP encryption optional */
#define SRTP_CRYPTO_ENABLE (1 << 2)
#define SRTP_CRYPTO_OFFER_OK (1 << 3)
#define SRTP_CRYPTO_TAG_32 (1 << 4)
#define SRTP_CRYPTO_TAG_80 (1 << 5)
/*! \brief structure for secure RTP audio */
struct sip_srtp {
unsigned int flags;
struct sdp_crypto *crypto;
};
/*!
* \brief allocate a sip_srtp structure
* \retval a new malloc'd sip_srtp structure on success
* \retval NULL on failure
*/
struct sip_srtp *sip_srtp_alloc(void);
/*!
* \brief free a sip_srtp structure
* \param srtp a sip_srtp structure
*/
void sip_srtp_destroy(struct sip_srtp *srtp);
#endif /* _SIP_SRTP_H */