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authorJoshua Colp <jcolp@digium.com>2013-06-22 14:03:22 +0000
committerJoshua Colp <jcolp@digium.com>2013-06-22 14:03:22 +0000
commit77002bc377f19ea11e60732c486b6ef371688773 (patch)
treec19fd245c519c6d7905403849a7af9c7e4a4be3e /channels/sip/include/srtp.h
parentea03516cb5426915d183526335d3a7d662ea29dc (diff)
Merge in current pimp_my_sip work, including:
1. Security events 2. Websocket support 3. Diversion header + redirecting support 4. An anonymous endpoint identifier 5. Inbound extension state subscription support 6. PIDF notify generation 7. One touch recording support (special thanks Sean Bright!) 8. Blind and attended transfer support 9. Automatic inbound registration expiration 10. SRTP support 11. Media offer control dialplan function 12. Connected line support 13. SendText() support 14. Qualify support 15. Inband DTMF detection 16. Call and pickup groups 17. Messaging support Thanks everyone! Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels/sip/include/srtp.h')
-rw-r--r--channels/sip/include/srtp.h59
1 files changed, 0 insertions, 59 deletions
diff --git a/channels/sip/include/srtp.h b/channels/sip/include/srtp.h
deleted file mode 100644
index a4ded62ca..000000000
--- a/channels/sip/include/srtp.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * 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 */