mirror of
https://github.com/asterisk/asterisk.git
synced 2025-09-23 14:44:28 +00:00
857814f435
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
58 lines
1.4 KiB
C
58 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 sip_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)
|
|
|
|
/*! \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 */
|