Make all of the various rtpqos parameters in this branch available from the CHANNEL function.

Also includes a test for retrieving rtpqos parameters, including a NULL RTP
driver.  Additionally, some further separation of the SIP internal API into
headers was necessary.

(closes issue #16652)
 Reported by: kkm
 Patches: 
       20100204__issue16652.diff.txt uploaded by tilghman (license 14)
 
Review: https://reviewboard.asterisk.org/r/501/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247124 f38db490-d61c-443f-a65b-d21fe96a405b

6 files changed, 567 insertions, 3 deletions

diff --git a/channels/sip/dialplan_functions.c b/channels/sip/dialplan_functions.c
new file mode 100644
index 000000000..473199c71
--- /dev/null
+++ b/channels/sip/dialplan_functions.c
@@ -0,0 +1,366 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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
+ * \brief sip channel dialplan functions and unit tests
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <math.h>
+
+#include "asterisk/channel.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/pbx.h"
+
+#include "include/sip.h"
+#include "include/globals.h"
+#include "include/dialog.h"
+#include "include/dialplan_functions.h"
+#include "include/sip_utils.h"
+
+
+int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
+{
+	struct sip_pvt *p = chan->tech_pvt;
+	char *parse = ast_strdupa(preparse);
+	int res = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(param);
+		AST_APP_ARG(type);
+		AST_APP_ARG(field);
+	);
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	/* Sanity check */
+	if (!IS_SIP_TECH(chan->tech)) {
+		ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
+		return 0;
+	}
+
+	memset(buf, 0, buflen);
+
+	if (p == NULL) {
+		return -1;
+	}
+
+	if (!strcasecmp(args.param, "peerip")) {
+		ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", buflen);
+	} else if (!strcasecmp(args.param, "recvip")) {
+		ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", buflen);
+	} else if (!strcasecmp(args.param, "from")) {
+		ast_copy_string(buf, p->from, buflen);
+	} else if (!strcasecmp(args.param, "uri")) {
+		ast_copy_string(buf, p->uri, buflen);
+	} else if (!strcasecmp(args.param, "useragent")) {
+		ast_copy_string(buf, p->useragent, buflen);
+	} else if (!strcasecmp(args.param, "peername")) {
+		ast_copy_string(buf, p->peername, buflen);
+	} else if (!strcasecmp(args.param, "t38passthrough")) {
+		ast_copy_string(buf, (p->t38.state == T38_DISABLED) ? "0" : "1", buflen);
+	} else if (!strcasecmp(args.param, "rtpdest")) {
+		struct sockaddr_in sin;
+
+		if (ast_strlen_zero(args.type))
+			args.type = "audio";
+
+		if (!strcasecmp(args.type, "audio"))
+			ast_rtp_instance_get_remote_address(p->rtp, &sin);
+		else if (!strcasecmp(args.type, "video"))
+			ast_rtp_instance_get_remote_address(p->vrtp, &sin);
+		else if (!strcasecmp(args.type, "text"))
+			ast_rtp_instance_get_remote_address(p->trtp, &sin);
+		else
+			return -1;
+
+		snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
+	} else if (!strcasecmp(args.param, "rtpqos")) {
+		struct ast_rtp_instance *rtp = NULL;
+
+		if (ast_strlen_zero(args.type)) {
+			args.type = "audio";
+		}
+
+		if (!strcasecmp(args.type, "audio")) {
+			rtp = p->rtp;
+		} else if (!strcasecmp(args.type, "video")) {
+			rtp = p->vrtp;
+		} else if (!strcasecmp(args.type, "text")) {
+			rtp = p->trtp;
+		} else {
+			return -1;
+		}
+
+		if (ast_strlen_zero(args.field) || !strcasecmp(args.field, "all")) {
+			char quality_buf[AST_MAX_USER_FIELD], *quality;
+
+			if (!(quality = ast_rtp_instance_get_quality(rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+				return -1;
+			}
+
+			ast_copy_string(buf, quality_buf, buflen);
+			return res;
+		} else {
+			struct ast_rtp_instance_stats stats;
+			int i;
+			struct {
+				const char *name;
+				enum { INT, DBL } type;
+				union {
+					unsigned int *i4;
+					double *d8;
+				};
+			} lookup[] = {
+				{ "txcount",               INT, { .i4 = &stats.txcount, }, },
+				{ "rxcount",               INT, { .i4 = &stats.rxcount, }, },
+				{ "txjitter",              INT, { .i4 = &stats.txjitter, }, },
+				{ "rxjitter",              INT, { .i4 = &stats.rxjitter, }, },
+				{ "remote_maxjitter",      DBL, { .d8 = &stats.remote_maxjitter, }, },
+				{ "remote_minjitter",      DBL, { .d8 = &stats.remote_minjitter, }, },
+				{ "remote_normdevjitter",  DBL, { .d8 = &stats.remote_normdevjitter, }, },
+				{ "remote_stdevjitter",    DBL, { .d8 = &stats.remote_stdevjitter, }, },
+				{ "local_maxjitter",       DBL, { .d8 = &stats.local_maxjitter, }, },
+				{ "local_minjitter",       DBL, { .d8 = &stats.local_minjitter, }, },
+				{ "local_normdevjitter",   DBL, { .d8 = &stats.local_normdevjitter, }, },
+				{ "local_stdevjitter",     DBL, { .d8 = &stats.local_stdevjitter, }, },
+				{ "txploss",               INT, { .i4 = &stats.txploss, }, },
+				{ "rxploss",               INT, { .i4 = &stats.rxploss, }, },
+				{ "remote_maxrxploss",     DBL, { .d8 = &stats.remote_maxrxploss, }, },
+				{ "remote_minrxploss",     DBL, { .d8 = &stats.remote_minrxploss, }, },
+				{ "remote_normdevrxploss", DBL, { .d8 = &stats.remote_normdevrxploss, }, },
+				{ "remote_stdevrxploss",   DBL, { .d8 = &stats.remote_stdevrxploss, }, },
+				{ "local_maxrxploss",      DBL, { .d8 = &stats.local_maxrxploss, }, },
+				{ "local_minrxploss",      DBL, { .d8 = &stats.local_minrxploss, }, },
+				{ "local_normdevrxploss",  DBL, { .d8 = &stats.local_normdevrxploss, }, },
+				{ "local_stdevrxploss",    DBL, { .d8 = &stats.local_stdevrxploss, }, },
+				{ "rtt",                   INT, { .i4 = &stats.rtt, }, },
+				{ "maxrtt",                DBL, { .d8 = &stats.maxrtt, }, },
+				{ "minrtt",                DBL, { .d8 = &stats.minrtt, }, },
+				{ "normdevrtt",            DBL, { .d8 = &stats.normdevrtt, }, },
+				{ "stdevrtt",              DBL, { .d8 = &stats.stdevrtt, }, },
+				{ "local_ssrc",            INT, { .i4 = &stats.local_ssrc, }, },
+				{ "remote_ssrc",           INT, { .i4 = &stats.remote_ssrc, }, },
+				{ NULL, },
+			};
+
+			if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
+				return -1;
+			}
+
+			for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
+				if (!strcasecmp(args.field, lookup[i].name)) {
+					if (lookup[i].type == INT) {
+						snprintf(buf, buflen, "%u", *lookup[i].i4);
+					} else {
+						snprintf(buf, buflen, "%f", *lookup[i].d8);
+					}
+					return 0;
+				}
+			}
+			ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
+			return -1;
+		}
+	} else {
+		res = -1;
+	}
+	return res;
+}
+
+#ifdef TEST_FRAMEWORK
+static int test_sip_rtpqos_1_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data)
+{
+	/* Needed to pass sanity checks */
+	ast_rtp_instance_set_data(instance, data);
+	return 0;
+}
+
+static int test_sip_rtpqos_1_destroy(struct ast_rtp_instance *instance)
+{
+	/* Needed to pass sanity checks */
+	return 0;
+}
+
+static struct ast_frame *test_sip_rtpqos_1_read(struct ast_rtp_instance *instance, int rtcp)
+{
+	/* Needed to pass sanity checks */
+	return &ast_null_frame;
+}
+
+static int test_sip_rtpqos_1_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+	/* Needed to pass sanity checks */
+	return 0;
+}
+
+static int test_sip_rtpqos_1_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
+{
+	struct ast_rtp_instance_stats *s = ast_rtp_instance_get_data(instance);
+	memcpy(stats, s, sizeof(*stats));
+	return 0;
+}
+
+AST_TEST_DEFINE(test_sip_rtpqos_1)
+{
+	int i, res = AST_TEST_PASS;
+	struct ast_rtp_engine test_engine = {
+		.name = "test",
+		.new = test_sip_rtpqos_1_new,
+		.destroy = test_sip_rtpqos_1_destroy,
+		.read = test_sip_rtpqos_1_read,
+		.write = test_sip_rtpqos_1_write,
+		.get_stat = test_sip_rtpqos_1_get_stat,
+	};
+	struct sockaddr_in sin = { .sin_port = 31337, .sin_addr = { 4 * 16777216 + 3 * 65536 + 2 * 256 + 1 } };
+	struct ast_rtp_instance_stats mine = { 0, };
+	struct sip_pvt *p = NULL;
+	struct ast_channel *chan = NULL;
+	struct ast_str *varstr = NULL, *buffer = NULL;
+	struct {
+		const char *name;
+		enum { INT, DBL } type;
+		union {
+			unsigned int *i4;
+			double *d8;
+		};
+	} lookup[] = {
+		{ "txcount",               INT, { .i4 = &mine.txcount, }, },
+		{ "rxcount",               INT, { .i4 = &mine.rxcount, }, },
+		{ "txjitter",              INT, { .i4 = &mine.txjitter, }, },
+		{ "rxjitter",              INT, { .i4 = &mine.rxjitter, }, },
+		{ "remote_maxjitter",      DBL, { .d8 = &mine.remote_maxjitter, }, },
+		{ "remote_minjitter",      DBL, { .d8 = &mine.remote_minjitter, }, },
+		{ "remote_normdevjitter",  DBL, { .d8 = &mine.remote_normdevjitter, }, },
+		{ "remote_stdevjitter",    DBL, { .d8 = &mine.remote_stdevjitter, }, },
+		{ "local_maxjitter",       DBL, { .d8 = &mine.local_maxjitter, }, },
+		{ "local_minjitter",       DBL, { .d8 = &mine.local_minjitter, }, },
+		{ "local_normdevjitter",   DBL, { .d8 = &mine.local_normdevjitter, }, },
+		{ "local_stdevjitter",     DBL, { .d8 = &mine.local_stdevjitter, }, },
+		{ "txploss",               INT, { .i4 = &mine.txploss, }, },
+		{ "rxploss",               INT, { .i4 = &mine.rxploss, }, },
+		{ "remote_maxrxploss",     DBL, { .d8 = &mine.remote_maxrxploss, }, },
+		{ "remote_minrxploss",     DBL, { .d8 = &mine.remote_minrxploss, }, },
+		{ "remote_normdevrxploss", DBL, { .d8 = &mine.remote_normdevrxploss, }, },
+		{ "remote_stdevrxploss",   DBL, { .d8 = &mine.remote_stdevrxploss, }, },
+		{ "local_maxrxploss",      DBL, { .d8 = &mine.local_maxrxploss, }, },
+		{ "local_minrxploss",      DBL, { .d8 = &mine.local_minrxploss, }, },
+		{ "local_normdevrxploss",  DBL, { .d8 = &mine.local_normdevrxploss, }, },
+		{ "local_stdevrxploss",    DBL, { .d8 = &mine.local_stdevrxploss, }, },
+		{ "rtt",                   INT, { .i4 = &mine.rtt, }, },
+		{ "maxrtt",                DBL, { .d8 = &mine.maxrtt, }, },
+		{ "minrtt",                DBL, { .d8 = &mine.minrtt, }, },
+		{ "normdevrtt",            DBL, { .d8 = &mine.normdevrtt, }, },
+		{ "stdevrtt",              DBL, { .d8 = &mine.stdevrtt, }, },
+		{ "local_ssrc",            INT, { .i4 = &mine.local_ssrc, }, },
+		{ "remote_ssrc",           INT, { .i4 = &mine.remote_ssrc, }, },
+		{ NULL, },
+	};
+
+	switch (cmd) {
+	case TEST_INIT:
+		info->name = "test_sip_rtpqos";
+		info->category = "channels/chan_sip/";
+		info->summary = "Test retrieval of SIP RTP QOS stats";
+		info->description =
+			"Verify values in the RTP instance structure can be accessed through the dialplan.";
+		return AST_TEST_NOT_RUN;
+	case TEST_EXECUTE:
+		break;
+	}
+
+	ast_rtp_engine_register2(&test_engine, NULL);
+	/* Have to associate this with a SIP pvt and an ast_channel */
+	if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL))) {
+		res = AST_TEST_NOT_RUN;
+		goto done;
+	}
+	if (!(p->rtp = ast_rtp_instance_new("test", sched, &bindaddr, &mine))) {
+		res = AST_TEST_NOT_RUN;
+		goto done;
+	}
+	ast_rtp_instance_set_remote_address(p->rtp, &sin);
+	if (!(chan = ast_dummy_channel_alloc())) {
+		res = AST_TEST_NOT_RUN;
+		goto done;
+	}
+	chan->tech = &sip_tech;
+	chan->tech_pvt = p;
+	p->owner = chan;
+
+	varstr = ast_str_create(16);
+	buffer = ast_str_create(16);
+	if (!varstr || !buffer) {
+		res = AST_TEST_NOT_RUN;
+		goto done;
+	}
+
+	/* Populate "mine" with values, then retrieve them with the CHANNEL dialplan function */
+	for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
+		ast_str_set(&varstr, 0, "${CHANNEL(rtpqos,audio,%s)}", lookup[i].name);
+		if (lookup[i].type == INT) {
+			int j;
+			char cmpstr[256];
+			for (j = 1; j < 25; j++) {
+				*lookup[i].i4 = j;
+				ast_str_substitute_variables(&buffer, 0, chan, ast_str_buffer(varstr));
+				snprintf(cmpstr, sizeof(cmpstr), "%d", j);
+				if (strcmp(cmpstr, ast_str_buffer(buffer))) {
+					res = AST_TEST_FAIL;
+					ast_test_status_update(test, "%s != %s != %s\n", ast_str_buffer(varstr), cmpstr, ast_str_buffer(buffer));
+					break;
+				}
+			}
+		} else {
+			double j, cmpdbl = 0.0;
+			for (j = 1.0; j < 10.0; j += 0.3) {
+				*lookup[i].d8 = j;
+				ast_str_substitute_variables(&buffer, 0, chan, ast_str_buffer(varstr));
+				if (sscanf(ast_str_buffer(buffer), "%lf", &cmpdbl) != 1 || fabs(j - cmpdbl > .05)) {
+					res = AST_TEST_FAIL;
+					ast_test_status_update(test, "%s != %f != %s\n", ast_str_buffer(varstr), j, ast_str_buffer(buffer));
+					break;
+				}
+			}
+		}
+	}
+
+done:
+	ast_free(varstr);
+	ast_free(buffer);
+
+	/* This unref will take care of destroying the channel, RTP instance, and SIP pvt */
+	if (p) {
+		dialog_unref(p, "Destroy test object");
+	}
+	ast_rtp_engine_unregister(&test_engine);
+	return res;
+}
+#endif
+
+/*! \brief SIP test registration */
+void sip_dialplan_function_register_tests(void)
+{
+	AST_TEST_REGISTER(test_sip_rtpqos_1);
+}
+
+/*! \brief SIP test registration */
+void sip_dialplan_function_unregister_tests(void)
+{
+	AST_TEST_UNREGISTER(test_sip_rtpqos_1);
+}
+
diff --git a/channels/sip/include/config_parser.h b/channels/sip/include/config_parser.h
index 0b86188c6..76fefc2c6 100644
--- a/channels/sip/include/config_parser.h
+++ b/channels/sip/include/config_parser.h
@@ -43,7 +43,7 @@ int sip_parse_register_line(struct sip_registry *reg, int default_expiry, const
  */
 int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport);
 
-/*! 
+/*!
  * \brief register config parsing tests
  */
 void sip_config_parser_register_tests(void);
diff --git a/channels/sip/include/dialog.h b/channels/sip/include/dialog.h
new file mode 100644
index 000000000..554aa5e6a
--- /dev/null
+++ b/channels/sip/include/dialog.h
@@ -0,0 +1,75 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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
+ * \brief sip dialog management header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_DIALOG_H
+#define _SIP_DIALOG_H
+
+/*! \brief
+ * when we create or delete references, make sure to use these
+ * functions so we keep track of the refcounts.
+ * To simplify the code, we allow a NULL to be passed to dialog_unref().
+ */
+#define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
+#define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
+struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func);
+struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func);
+
+struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
+				 int useglobal_nat, const int intended_method, struct sip_request *req);
+void sip_scheddestroy(struct sip_pvt *p, int ms);
+int sip_cancel_destroy(struct sip_pvt *p);
+
+/*! \brief Destroy SIP call structure.
+ * Make it return NULL so the caller can do things like
+ *	foo = sip_destroy(foo);
+ * and reduce the chance of bugs due to dangling pointers.
+ */
+struct sip_pvt *sip_destroy(struct sip_pvt *p);
+
+/*! \brief Destroy SIP call structure.
+ * Make it return NULL so the caller can do things like
+ *	foo = sip_destroy(foo);
+ * and reduce the chance of bugs due to dangling pointers.
+ */
+void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist);
+/*!
+ * \brief Unlink a dialog from the dialogs container, as well as any other places
+ * that it may be currently stored.
+ *
+ * \note A reference to the dialog must be held before calling this function, and this
+ * function does not release that reference.
+ */
+void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist);
+
+/*! \brief Acknowledges receipt of a packet and stops retransmission
+ * called with p locked*/
+int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
+
+/*! \brief Pretend to ack all packets
+ * called with p locked */
+void __sip_pretend_ack(struct sip_pvt *p);
+
+/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
+int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
+
+#endif /* defined(_SIP_DIALOG_H) */
diff --git a/channels/sip/include/dialplan_functions.h b/channels/sip/include/dialplan_functions.h
new file mode 100644
index 000000000..1600d4314
--- /dev/null
+++ b/channels/sip/include/dialplan_functions.h
@@ -0,0 +1,41 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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
+ * \brief SIP dialplan functions header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_DIALPLAN_FUNCTIONS_H
+#define _SIP_DIALPLAN_FUNCTIONS_H
+
+/*!
+ * \brief Channel read dialplan function for SIP
+ */
+int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);
+
+/*!
+ * \brief register dialplan function tests
+ */
+void sip_dialplan_function_register_tests(void);
+/*!
+ * \brief unregister dialplan function tests
+ */
+void sip_dialplan_function_unregister_tests(void);
+
+#endif /* !defined(_SIP_DIALPLAN_FUNCTIONS_H) */
diff --git a/channels/sip/include/globals.h b/channels/sip/include/globals.h
new file mode 100644
index 000000000..0d7131d87
--- /dev/null
+++ b/channels/sip/include/globals.h
@@ -0,0 +1,42 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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
+ * \brief sip global declaration header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_GLOBALS_H
+#define _SIP_GLOBALS_H
+
+extern struct sockaddr_in bindaddr;     /*!< UDP: The address we bind to */
+extern struct sched_context *sched;     /*!< The scheduling context */
+
+/*! \brief Definition of this channel for PBX channel registration */
+extern const struct ast_channel_tech sip_tech;
+
+/*! \brief This version of the sip channel tech has no send_digit_begin
+ * callback so that the core knows that the channel does not want
+ * DTMF BEGIN frames.
+ * The struct is initialized just before registering the channel driver,
+ * and is for use with channels using SIP INFO DTMF.
+ */
+extern struct ast_channel_tech sip_tech_info;
+
+#endif /* !defined(SIP_GLOBALS_H) */
+
diff --git a/channels/sip/include/sip_utils.h b/channels/sip/include/sip_utils.h
index 549f9392f..47297ab68 100644
--- a/channels/sip/include/sip_utils.h
+++ b/channels/sip/include/sip_utils.h
@@ -22,8 +22,11 @@
 #ifndef _SIP_UTILS_H
 #define _SIP_UTILS_H
 
-/*! 
- * \brief converts ascii port to int representation. 
+/* wrapper macro to tell whether t points to one of the sip_tech descriptors */
+#define IS_SIP_TECH(t)  ((t) == &sip_tech || (t) == &sip_tech_info)
+
+/*!
+ * \brief converts ascii port to int representation.
  *
  * \arg pt[in] string that contains a port.
  * \arg standard[in] port to return in case the port string input is NULL
@@ -40,4 +43,41 @@ unsigned int port_str2int(const char *pt, unsigned int standard);
 const char *find_closing_quote(const char *start, const char *lim);
 
 
+/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
+int hangup_sip2cause(int cause);
+
+/*! \brief Convert Asterisk hangup causes to SIP codes
+\verbatim
+ Possible values from causes.h
+        AST_CAUSE_NOTDEFINED    AST_CAUSE_NORMAL        AST_CAUSE_BUSY
+        AST_CAUSE_FAILURE       AST_CAUSE_CONGESTION    AST_CAUSE_UNALLOCATED
+
+	In addition to these, a lot of PRI codes is defined in causes.h
+	...should we take care of them too ?
+
+	Quote RFC 3398
+
+   ISUP Cause value                        SIP response
+   ----------------                        ------------
+   1  unallocated number                   404 Not Found
+   2  no route to network                  404 Not found
+   3  no route to destination              404 Not found
+   16 normal call clearing                 --- (*)
+   17 user busy                            486 Busy here
+   18 no user responding                   408 Request Timeout
+   19 no answer from the user              480 Temporarily unavailable
+   20 subscriber absent                    480 Temporarily unavailable
+   21 call rejected                        403 Forbidden (+)
+   22 number changed (w/o diagnostic)      410 Gone
+   22 number changed (w/ diagnostic)       301 Moved Permanently
+   23 redirection to new destination       410 Gone
+   26 non-selected user clearing           404 Not Found (=)
+   27 destination out of order             502 Bad Gateway
+   28 address incomplete                   484 Address incomplete
+   29 facility rejected                    501 Not implemented
+   31 normal unspecified                   480 Temporarily unavailable
+\endverbatim
+*/
+const char *hangup_cause2sip(int cause);
+
 #endif