Files
asterisk/channels/sip/include/srtp.h
Gregory Nietsky 8493c46308 Merged revisions 336936 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
  
  
  Allow Setting Auth Tag Bit length Based on invite or config option
  
  Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
  Curently only 80 bit is supported.
  
  The outgoing invite will use the taglen of the incoming invite preventing
  one-way audio.
  
  (Closes issue ASTERISK-17895)
  
  Review: https://reviewboard.asterisk.org/r/1173/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 16:56:11 +00:00

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 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)
#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 */