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

5 files changed, 3 insertions, 520 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 */
diff --git a/channels/sip/sdp_crypto.c b/channels/sip/sdp_crypto.c
deleted file mode 100644
index c27e882c2..000000000
--- a/channels/sip/sdp_crypto.c
+++ /dev/null
@@ -1,318 +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.c
- *
- * \brief SDP Security descriptions
- *
- * Specified in RFC 4568
- *
- * \author Mikael Magnusson <mikma@users.sourceforge.net>
- */
-
-/*** MODULEINFO
-	<support_level>core</support_level>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/options.h"
-#include "asterisk/utils.h"
-#include "include/sdp_crypto.h"
-#include "include/srtp.h"
-
-#define SRTP_MASTER_LEN 30
-#define SRTP_MASTERKEY_LEN 16
-#define SRTP_MASTERSALT_LEN ((SRTP_MASTER_LEN) - (SRTP_MASTERKEY_LEN))
-#define SRTP_MASTER_LEN64 (((SRTP_MASTER_LEN) * 8 + 5) / 6 + 1)
-
-extern struct ast_srtp_res *res_srtp;
-extern struct ast_srtp_policy_res *res_srtp_policy;
-
-struct sdp_crypto {
-	char *a_crypto;
-	unsigned char local_key[SRTP_MASTER_LEN];
-	char *tag;
-	char local_key64[SRTP_MASTER_LEN64];
-	unsigned char remote_key[SRTP_MASTER_LEN];
-};
-
-static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, unsigned long ssrc, int inbound);
-
-static struct sdp_crypto *sdp_crypto_alloc(void)
-{
-	return ast_calloc(1, sizeof(struct sdp_crypto));
-}
-
-void sdp_crypto_destroy(struct sdp_crypto *crypto)
-{
-	ast_free(crypto->a_crypto);
-	crypto->a_crypto = NULL;
-	ast_free(crypto->tag);
-	crypto->tag = NULL;
-	ast_free(crypto);
-}
-
-struct sdp_crypto *sdp_crypto_setup(void)
-{
-	struct sdp_crypto *p;
-	int key_len;
-	unsigned char remote_key[SRTP_MASTER_LEN];
-
-	if (!ast_rtp_engine_srtp_is_registered()) {
-		return NULL;
-	}
-
-	if (!(p = sdp_crypto_alloc())) {
-		return NULL;
-	}
-
-	if (res_srtp->get_random(p->local_key, sizeof(p->local_key)) < 0) {
-		sdp_crypto_destroy(p);
-		return NULL;
-	}
-
-	ast_base64encode(p->local_key64, p->local_key, SRTP_MASTER_LEN, sizeof(p->local_key64));
-
-	key_len = ast_base64decode(remote_key, p->local_key64, sizeof(remote_key));
-
-	if (key_len != SRTP_MASTER_LEN) {
-		ast_log(LOG_ERROR, "base64 encode/decode bad len %d != %d\n", key_len, SRTP_MASTER_LEN);
-		ast_free(p);
-		return NULL;
-	}
-
-	if (memcmp(remote_key, p->local_key, SRTP_MASTER_LEN)) {
-		ast_log(LOG_ERROR, "base64 encode/decode bad key\n");
-		ast_free(p);
-		return NULL;
-	}
-
-	ast_debug(1 , "local_key64 %s len %zu\n", p->local_key64, strlen(p->local_key64));
-
-	return p;
-}
-
-static int set_crypto_policy(struct ast_srtp_policy *policy, int suite_val, const unsigned char *master_key, unsigned long ssrc, int inbound)
-{
-	const unsigned char *master_salt = NULL;
-
-	if (!ast_rtp_engine_srtp_is_registered()) {
-		return -1;
-	}
-
-	master_salt = master_key + SRTP_MASTERKEY_LEN;
-	if (res_srtp_policy->set_master_key(policy, master_key, SRTP_MASTERKEY_LEN, master_salt, SRTP_MASTERSALT_LEN) < 0) {
-		return -1;
-	}
-
-	if (res_srtp_policy->set_suite(policy, suite_val)) {
-		ast_log(LOG_WARNING, "Could not set remote SRTP suite\n");
-		return -1;
-	}
-
-	res_srtp_policy->set_ssrc(policy, ssrc, inbound);
-
-	return 0;
-}
-
-static int sdp_crypto_activate(struct sdp_crypto *p, int suite_val, unsigned char *remote_key, struct ast_rtp_instance *rtp)
-{
-	struct ast_srtp_policy *local_policy = NULL;
-	struct ast_srtp_policy *remote_policy = NULL;
-	struct ast_rtp_instance_stats stats = {0,};
-	int res = -1;
-
-	if (!ast_rtp_engine_srtp_is_registered()) {
-		return -1;
-	}
-
-	if (!p) {
-		return -1;
-	}
-
-	if (!(local_policy = res_srtp_policy->alloc())) {
-		return -1;
-	}
-
-	if (!(remote_policy = res_srtp_policy->alloc())) {
-		goto err;
-	}
-
-	if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_LOCAL_SSRC)) {
-		goto err;
-	}
-
-	if (set_crypto_policy(local_policy, suite_val, p->local_key, stats.local_ssrc, 0) < 0) {
-		goto err;
-	}
-
-	if (set_crypto_policy(remote_policy, suite_val, remote_key, 0, 1) < 0) {
-		goto err;
-	}
-
-	/* Add the SRTP policies */
-	if (ast_rtp_instance_add_srtp_policy(rtp, remote_policy, local_policy)) {
-		ast_log(LOG_WARNING, "Could not set SRTP policies\n");
-		goto err;
-	}
-
-	ast_debug(1 , "SRTP policy activated\n");
-	res = 0;
-
-err:
-	if (local_policy) {
-		res_srtp_policy->destroy(local_policy);
-	}
-
-	if (remote_policy) {
-		res_srtp_policy->destroy(remote_policy);
-	}
-
-	return res;
-}
-
-int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp)
-{
-	char *str = NULL;
-	char *tag = NULL;
-	char *suite = NULL;
-	char *key_params = NULL;
-	char *key_param = NULL;
-	char *session_params = NULL;
-	char *key_salt = NULL;
-	char *lifetime = NULL;
-	int found = 0;
-	int key_len = 0;
-	int suite_val = 0;
-	unsigned char remote_key[SRTP_MASTER_LEN];
-	int taglen = 0;
-
-	if (!ast_rtp_engine_srtp_is_registered()) {
-		return -1;
-	}
-
-	str = ast_strdupa(attr);
-
-	strsep(&str, ":");
-	tag = strsep(&str, " ");
-	suite = strsep(&str, " ");
-	key_params = strsep(&str, " ");
-	session_params = strsep(&str, " ");
-
-	if (!tag || !suite) {
-		ast_log(LOG_WARNING, "Unrecognized a=%s", attr);
-		return -1;
-	}
-
-	if (!ast_strlen_zero(session_params)) {
-		ast_log(LOG_WARNING, "Unsupported crypto parameters: %s", session_params);
-		return -1;
-	}
-
-	if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_80")) {
-		suite_val = AST_AES_CM_128_HMAC_SHA1_80;
-		ast_set_flag(srtp, SRTP_CRYPTO_TAG_80);
-		taglen = 80;
-	} else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) {
-		suite_val = AST_AES_CM_128_HMAC_SHA1_32;
-		ast_set_flag(srtp, SRTP_CRYPTO_TAG_32);
-		taglen = 32;
-	} else {
-		ast_log(LOG_WARNING, "Unsupported crypto suite: %s\n", suite);
-		return -1;
-	}
-
-	while ((key_param = strsep(&key_params, ";"))) {
-		char *method = NULL;
-		char *info = NULL;
-
-		method = strsep(&key_param, ":");
-		info = strsep(&key_param, ";");
-
-		if (!strcmp(method, "inline")) {
-			key_salt = strsep(&info, "|");
-			lifetime = strsep(&info, "|");
-
-			if (lifetime) {
-				ast_log(LOG_NOTICE, "Crypto life time unsupported: %s\n", attr);
-				continue;
-			}
-
-			found = 1;
-			break;
-		}
-	}
-
-	if (!found) {
-		ast_log(LOG_NOTICE, "SRTP crypto offer not acceptable\n");
-		return -1;
-	}
-
-	if ((key_len = ast_base64decode(remote_key, key_salt, sizeof(remote_key))) != SRTP_MASTER_LEN) {
-		ast_log(LOG_WARNING, "SRTP descriptions key %d != %d\n", key_len, SRTP_MASTER_LEN);
-		return -1;
-	}
-
-	if (!memcmp(p->remote_key, remote_key, sizeof(p->remote_key))) {
-		ast_debug(1, "SRTP remote key unchanged; maintaining current policy\n");
-		return 0;
-	}
-	memcpy(p->remote_key, remote_key, sizeof(p->remote_key));
-
-	if (sdp_crypto_activate(p, suite_val, remote_key, rtp) < 0) {
-		return -1;
-	}
-
-	if (!p->tag) {
-		ast_log(LOG_DEBUG, "Accepting crypto tag %s\n", tag);
-		p->tag = ast_strdup(tag);
-		if (!p->tag) {
-			ast_log(LOG_ERROR, "Could not allocate memory for tag\n");
-			return -1;
-		}
-	}
-
-	/* Finally, rebuild the crypto line */
-	return sdp_crypto_offer(p, taglen);
-}
-
-int sdp_crypto_offer(struct sdp_crypto *p, int taglen)
-{
-	/* Rebuild the crypto line */
-	if (p->a_crypto) {
-		ast_free(p->a_crypto);
-	}
-
-	if (ast_asprintf(&p->a_crypto, "a=crypto:%s AES_CM_128_HMAC_SHA1_%i inline:%s\r\n",
-			 p->tag ? p->tag : "1", taglen, p->local_key64) == -1) {
-			ast_log(LOG_ERROR, "Could not allocate memory for crypto line\n");
-		return -1;
-	}
-
-	ast_log(LOG_DEBUG, "Crypto line: %s", p->a_crypto);
-
-	return 0;
-}
-
-const char *sdp_crypto_attrib(struct sdp_crypto *p)
-{
-	return p->a_crypto;
-}
diff --git a/channels/sip/srtp.c b/channels/sip/srtp.c
deleted file mode 100644
index 8b2718fc3..000000000
--- a/channels/sip/srtp.c
+++ /dev/null
@@ -1,55 +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.c
- *
- * \brief SIP Secure RTP (SRTP)
- *
- * Specified in RFC 3711
- *
- * \author Mikael Magnusson <mikma@users.sourceforge.net>
- */
-
-/*** MODULEINFO
-	<support_level>core</support_level>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/utils.h"
-#include "include/srtp.h"
-
-struct sip_srtp *sip_srtp_alloc(void)
-{
-	struct sip_srtp *srtp;
-
-	srtp = ast_calloc(1, sizeof(*srtp));
-
-	return srtp;
-}
-
-void sip_srtp_destroy(struct sip_srtp *srtp)
-{
-	if (srtp->crypto) {
-		sdp_crypto_destroy(srtp->crypto);
-	}
-	srtp->crypto = NULL;
-	ast_free(srtp);
-}