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

3 files changed, 3 insertions, 147 deletions

diff --git a/channels/sip/include/sdp_crypto.h b/channels/sip/include/sdp_crypto.h
deleted file mode 100644
index da1035e87..000000000
--- a/channels/sip/include/sdp_crypto.h
+++ /dev/null
@@ -1,85 +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 sdp_crypto.h
- *
- * \brief SDP Security descriptions
- *
- * Specified in RFC 4568
- *
- * \author Mikael Magnusson <mikma@users.sourceforge.net>
- */
-
-#ifndef _SDP_CRYPTO_H
-#define _SDP_CRYPTO_H
-
-#include <asterisk/rtp_engine.h>
-
-struct sdp_crypto;
-struct sip_srtp;
-
-/*! \brief Initialize an return an sdp_crypto struct
- *
- * \details
- * This function allocates a new sdp_crypto struct and initializes its values
- *
- * \retval NULL on failure
- * \retval a pointer to a  new sdp_crypto structure
- */
-struct sdp_crypto *sdp_crypto_setup(void);
-
-/*! \brief Destroy a previously allocated sdp_crypto struct */
-void sdp_crypto_destroy(struct sdp_crypto *crypto);
-
-/*! \brief Parse the a=crypto line from SDP and set appropriate values on the
- * sdp_crypto struct.
- *
- * \param p A valid sdp_crypto struct
- * \param attr the a:crypto line from SDP
- * \param rtp The rtp instance associated with the SDP being parsed
- * \param srtp SRTP structure
- *
- * \retval 0 success
- * \retval nonzero failure
- */
-int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp);
-
-
-/*! \brief Generate an SRTP a=crypto offer
- *
- * \details
- * The offer is stored on the sdp_crypto struct in a_crypto
- *
- * \param p A valid sdp_crypto struct
- * \param taglen Length
- *
- * \retval 0 success
- * \retval nonzero failure
- */
-int sdp_crypto_offer(struct sdp_crypto *p, int taglen);
-
-
-/*! \brief Return the a_crypto value of the sdp_crypto struct
- *
- * \param p An sdp_crypto struct that has had sdp_crypto_offer called
- *
- * \retval The value of the a_crypto for p
- */
-const char *sdp_crypto_attrib(struct sdp_crypto *p);
-
-#endif	/* _SDP_CRYPTO_H */
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 0adde37f2..8b4672b25 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -1165,9 +1165,9 @@ struct sip_pvt {
 	AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */
 	struct sip_invite_param *options;   /*!< Options for INVITE */
 	struct sip_st_dlg *stimer;          /*!< SIP Session-Timers */
-	struct sip_srtp *srtp;              /*!< Structure to hold Secure RTP session data for audio */
-	struct sip_srtp *vsrtp;             /*!< Structure to hold Secure RTP session data for video */
-	struct sip_srtp *tsrtp;             /*!< Structure to hold Secure RTP session data for text */
+	struct ast_sdp_srtp *srtp;              /*!< Structure to hold Secure RTP session data for audio */
+	struct ast_sdp_srtp *vsrtp;             /*!< Structure to hold Secure RTP session data for video */
+	struct ast_sdp_srtp *tsrtp;             /*!< Structure to hold Secure RTP session data for text */
 
 	int red;                            /*!< T.140 RTP Redundancy */
 	int hangupcause;                    /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
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 */