chan_rtp: Backport changes from master.

* Deprecate chan_multicast_rtp.

Change-Id: Ib5a45e58c75ee8abd0b4f9575379b5321feb853e

CHANGES

@@ -17,6 +17,36 @@ Core
  * A channel variable FORWARDERNAME is now set which indicates which channel
    was responsible for a forwarding requests received on dial attempt.
 
+chan_multicast_rtp
+------------------
+ * Deprecated in favor of chan_rtp which is basically chan_multicast_rtp
+   renamed to chan_rtp with UnicastRTP channels added and some internal code
+   improvements.
+
+chan_rtp
+------------------
+ * The format for dialing a unicast RTP channel is:
+   UnicastRTP/<destination-addr>[/[<options>]]
+   Where <destination-addr> is something like '127.0.0.1:5060'.
+   Where <options> are in standard Asterisk flag options format:
+   c(<codec>) - Specify which codec/format to use such as 'ulaw'.
+   e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
+
+ * More options are available over what chan_multicast_rtp supports.
+   The format for dialing a multicast RTP channel is:
+   MulticastRTP/<type>/<destination-addr>[/[<control-addr>][/[<options>]]]
+   Where <type> can be either 'basic' or 'linksys'.
+   Where <destination-addr> is something like '224.0.0.3:5060'.
+   Where <control-addr> is something like '127.0.0.1:5060'.
+   Where <options> are in standard Asterisk flag options format:
+   c(<codec>) - Specify which codec/format to use such as 'ulaw'.
+   i(<address>) - Specify the interface address from which multicast RTP
+     is sent.
+   l(<enable>) - Set whether packets are looped back to the sender.  The
+     enable value can be 0 to set looping to off and non-zero to set
+     looping on.
+   t(<ttl>) - Set the time-to-live (TTL) value for multicast packets.
+
 func_odbc
 ------------------
  * Added new global option "single_db_connection".

channels/chan_multicast_rtp.c

@@ -28,7 +28,8 @@
  */
 
 /*** MODULEINFO
-	<support_level>core</support_level>
+	<support_level>deprecated</support_level>
+	<defaultenabled>no</defaultenabled>
  ***/
 
 #include "asterisk.h"
@@ -215,8 +216,8 @@ static int unload_module(void)
 	return 0;
 }
 
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel (use chan_rtp instead)",
-	.support_level = AST_MODULE_SUPPORT_CORE,
+	.support_level = AST_MODULE_SUPPORT_DEPRECATED,
 	.load = load_module,
 	.unload = unload_module,
 	.load_pri = AST_MODPRI_CHANNEL_DRIVER,

channels/chan_rtp.c

@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 2009, Digium, Inc.
+ * Copyright (C) 2009 - 2014, Digium, Inc.
  *
  * Joshua Colp <jcolp@digium.com>
  * Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
@@ -22,7 +22,7 @@
  * \author Joshua Colp <jcolp@digium.com>
  * \author Andreas 'MacBrody' Broadmann <andreas.brodmann@gmail.com>
  *
- * \brief Multicast RTP Paging Channel
+ * \brief RTP (Multicast and Unicast) Media Channel
  *
  * \ingroup channel_drivers
  */
@@ -33,54 +33,64 @@
 
 #include "asterisk.h"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+ASTERISK_REGISTER_FILE()
 
-#include <fcntl.h>
-#include <sys/signal.h>
-
-#include "asterisk/lock.h"
 #include "asterisk/channel.h"
-#include "asterisk/config.h"
 #include "asterisk/module.h"
 #include "asterisk/pbx.h"
-#include "asterisk/sched.h"
-#include "asterisk/io.h"
 #include "asterisk/acl.h"
-#include "asterisk/callerid.h"
-#include "asterisk/file.h"
-#include "asterisk/cli.h"
 #include "asterisk/app.h"
 #include "asterisk/rtp_engine.h"
 #include "asterisk/causes.h"
-
+#include "asterisk/format_cache.h"
-static const char tdesc[] = "Multicast RTP Paging Channel Driver";
+#include "asterisk/multicast_rtp.h"
 
 /* Forward declarations */
 static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
-static int multicast_rtp_call(struct ast_channel *ast, const char *dest, int timeout);
+static struct ast_channel *unicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
-static int multicast_rtp_hangup(struct ast_channel *ast);
+static int rtp_call(struct ast_channel *ast, const char *dest, int timeout);
-static struct ast_frame *multicast_rtp_read(struct ast_channel *ast);
+static int rtp_hangup(struct ast_channel *ast);
-static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f);
+static struct ast_frame *rtp_read(struct ast_channel *ast);
+static int rtp_write(struct ast_channel *ast, struct ast_frame *f);
 
-/* Channel driver declaration */
+/* Multicast channel driver declaration */
 static struct ast_channel_tech multicast_rtp_tech = {
 	.type = "MulticastRTP",
-	.description = tdesc,
+	.description = "Multicast RTP Paging Channel Driver",
 	.requester = multicast_rtp_request,
-	.call = multicast_rtp_call,
+	.call = rtp_call,
-	.hangup = multicast_rtp_hangup,
+	.hangup = rtp_hangup,
-	.read = multicast_rtp_read,
+	.read = rtp_read,
-	.write = multicast_rtp_write,
+	.write = rtp_write,
+};
+
+/* Unicast channel driver declaration */
+static struct ast_channel_tech unicast_rtp_tech = {
+	.type = "UnicastRTP",
+	.description = "Unicast RTP Media Channel Driver",
+	.requester = unicast_rtp_request,
+	.call = rtp_call,
+	.hangup = rtp_hangup,
+	.read = rtp_read,
+	.write = rtp_write,
 };
 
 /*! \brief Function called when we should read a frame from the channel */
-static struct ast_frame  *multicast_rtp_read(struct ast_channel *ast)
+static struct ast_frame  *rtp_read(struct ast_channel *ast)
 {
-	return &ast_null_frame;
+	struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
+	int fdno = ast_channel_fdno(ast);
+
+	switch (fdno) {
+	case 0:
+		return ast_rtp_instance_read(instance, 0);
+	default:
+		return &ast_null_frame;
+	}
 }
 
 /*! \brief Function called when we should write a frame to the channel */
-static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f)
+static int rtp_write(struct ast_channel *ast, struct ast_frame *f)
 {
 	struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
 
@@ -88,7 +98,7 @@ static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f)
 }
 
 /*! \brief Function called when we should actually call the destination */
-static int multicast_rtp_call(struct ast_channel *ast, const char *dest, int timeout)
+static int rtp_call(struct ast_channel *ast, const char *dest, int timeout)
 {
 	struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
 
@@ -98,7 +108,7 @@ static int multicast_rtp_call(struct ast_channel *ast, const char *dest, int tim
 }
 
 /*! \brief Function called when we should hang the channel up */
-static int multicast_rtp_hangup(struct ast_channel *ast)
+static int rtp_hangup(struct ast_channel *ast)
 {
 	struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
 
@@ -109,41 +119,65 @@ static int multicast_rtp_hangup(struct ast_channel *ast)
 	return 0;
 }
 
-/*! \brief Function called when we should prepare to call the destination */
+/*! \brief Function called when we should prepare to call the multicast destination */
 static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
 {
-	char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
+	char *parse;
 	struct ast_rtp_instance *instance;
 	struct ast_sockaddr control_address;
 	struct ast_sockaddr destination_address;
 	struct ast_channel *chan;
 	struct ast_format_cap *caps = NULL;
 	struct ast_format *fmt = NULL;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(type);
+		AST_APP_ARG(destination);
+		AST_APP_ARG(control);
+		AST_APP_ARG(options);
+	);
+	struct ast_multicast_rtp_options *mcast_options = NULL;
 
-	fmt = ast_format_cap_get_format(cap, 0);
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_ERROR, "A multicast type and destination must be given to the 'MulticastRTP' channel\n");
+		goto failure;
+	}
+	parse = ast_strdupa(data);
+	AST_NONSTANDARD_APP_ARGS(args, parse, '/');
+
+	if (ast_strlen_zero(args.type)) {
+		ast_log(LOG_ERROR, "Type is required for the 'MulticastRTP' channel\n");
+		goto failure;
+	}
+
+	if (ast_strlen_zero(args.destination)) {
+		ast_log(LOG_ERROR, "Destination is required for the 'MulticastRTP' channel\n");
+		goto failure;
+	}
+	if (!ast_sockaddr_parse(&destination_address, args.destination, PARSE_PORT_REQUIRE)) {
+		ast_log(LOG_ERROR, "Destination address '%s' could not be parsed\n",
+			args.destination);
+		goto failure;
+	}
 
 	ast_sockaddr_setnull(&control_address);
-
+	if (!ast_strlen_zero(args.control)
-	/* If no type was given we can't do anything */
+		&& !ast_sockaddr_parse(&control_address, args.control, PARSE_PORT_REQUIRE)) {
-	if (ast_strlen_zero(multicast_type)) {
+		ast_log(LOG_ERROR, "Control address '%s' could not be parsed\n", args.control);
 		goto failure;
 	}
 
-	if (!(destination = strchr(tmp, '/'))) {
+	mcast_options = ast_multicast_rtp_create_options(args.type, args.options);
+	if (!mcast_options) {
 		goto failure;
 	}
-	*destination++ = '\0';
 
-	if ((control = strchr(destination, '/'))) {
+	fmt = ast_multicast_rtp_options_get_format(mcast_options);
-		*control++ = '\0';
+	if (!fmt) {
-		if (!ast_sockaddr_parse(&control_address, control,
+		fmt = ast_format_cap_get_format(cap, 0);
-					PARSE_PORT_REQUIRE)) {
-			goto failure;
-		}
 	}
-
+	if (!fmt) {
-	if (!ast_sockaddr_parse(&destination_address, destination,
+		ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
-				PARSE_PORT_REQUIRE)) {
+			args.destination);
 		goto failure;
 	}
 
@@ -152,11 +186,17 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
 		goto failure;
 	}
 
-	if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, multicast_type))) {
+	instance = ast_rtp_instance_new("multicast", NULL, &control_address, mcast_options);
+	if (!instance) {
+		ast_log(LOG_ERROR,
+			"Could not create '%s' multicast RTP instance for sending media to '%s'\n",
+			args.type, args.destination);
 		goto failure;
 	}
 
-	if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids, requestor, 0, "MulticastRTP/%p", instance))) {
+	chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
+		requestor, 0, "MulticastRTP/%p", instance);
+	if (!chan) {
 		ast_rtp_instance_destroy(instance);
 		goto failure;
 	}
@@ -176,6 +216,144 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
 
 	ast_channel_unlock(chan);
 
+	ao2_ref(fmt, -1);
+	ao2_ref(caps, -1);
+	ast_multicast_rtp_free_options(mcast_options);
+
+	return chan;
+
+failure:
+	ao2_cleanup(fmt);
+	ao2_cleanup(caps);
+	ast_multicast_rtp_free_options(mcast_options);
+	*cause = AST_CAUSE_FAILURE;
+	return NULL;
+}
+
+enum {
+	OPT_RTP_CODEC =  (1 << 0),
+	OPT_RTP_ENGINE = (1 << 1),
+};
+
+enum {
+	OPT_ARG_RTP_CODEC,
+	OPT_ARG_RTP_ENGINE,
+	/* note: this entry _MUST_ be the last one in the enum */
+	OPT_ARG_ARRAY_SIZE
+};
+
+AST_APP_OPTIONS(unicast_rtp_options, BEGIN_OPTIONS
+	/*! Set the codec to be used for unicast RTP */
+	AST_APP_OPTION_ARG('c', OPT_RTP_CODEC, OPT_ARG_RTP_CODEC),
+	/*! Set the RTP engine to use for unicast RTP */
+	AST_APP_OPTION_ARG('e', OPT_RTP_ENGINE, OPT_ARG_RTP_ENGINE),
+END_OPTIONS );
+
+/*! \brief Function called when we should prepare to call the unicast destination */
+static struct ast_channel *unicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
+{
+	char *parse;
+	struct ast_rtp_instance *instance;
+	struct ast_sockaddr address;
+	struct ast_sockaddr local_address;
+	struct ast_channel *chan;
+	struct ast_format_cap *caps = NULL;
+	struct ast_format *fmt = NULL;
+	const char *engine_name;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(destination);
+		AST_APP_ARG(options);
+	);
+	struct ast_flags opts = { 0, };
+	char *opt_args[OPT_ARG_ARRAY_SIZE];
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
+		goto failure;
+	}
+	parse = ast_strdupa(data);
+	AST_NONSTANDARD_APP_ARGS(args, parse, '/');
+
+	if (ast_strlen_zero(args.destination)) {
+		ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
+		goto failure;
+	}
+	if (!ast_sockaddr_parse(&address, args.destination, PARSE_PORT_REQUIRE)) {
+		ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination);
+		goto failure;
+	}
+
+	if (!ast_strlen_zero(args.options)
+		&& ast_app_parse_options(unicast_rtp_options, &opts, opt_args,
+			ast_strdupa(args.options))) {
+		ast_log(LOG_ERROR, "'UnicastRTP' channel options '%s' parse error\n",
+			args.options);
+		goto failure;
+	}
+
+	if (ast_test_flag(&opts, OPT_RTP_CODEC)
+		&& !ast_strlen_zero(opt_args[OPT_ARG_RTP_CODEC])) {
+		fmt = ast_format_cache_get(opt_args[OPT_ARG_RTP_CODEC]);
+		if (!fmt) {
+			ast_log(LOG_ERROR, "Codec '%s' not found for sending RTP to '%s'\n",
+				opt_args[OPT_ARG_RTP_CODEC], args.destination);
+			goto failure;
+		}
+	} else {
+		fmt = ast_format_cap_get_format(cap, 0);
+		if (!fmt) {
+			ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
+				args.destination);
+			goto failure;
+		}
+	}
+
+	caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+	if (!caps) {
+		goto failure;
+	}
+
+	engine_name = S_COR(ast_test_flag(&opts, OPT_RTP_ENGINE),
+		opt_args[OPT_ARG_RTP_ENGINE], NULL);
+
+	ast_ouraddrfor(&address, &local_address);
+	instance = ast_rtp_instance_new(engine_name, NULL, &local_address, NULL);
+	if (!instance) {
+		ast_log(LOG_ERROR,
+			"Could not create %s RTP instance for sending media to '%s'\n",
+			S_OR(engine_name, "default"), args.destination);
+		goto failure;
+	}
+
+	chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
+		requestor, 0, "UnicastRTP/%s-%p", args.destination, instance);
+	if (!chan) {
+		ast_rtp_instance_destroy(instance);
+		goto failure;
+	}
+	ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
+	ast_rtp_instance_set_remote_address(instance, &address);
+	ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(instance, 0));
+
+	ast_channel_tech_set(chan, &unicast_rtp_tech);
+
+	ast_format_cap_append(caps, fmt, 0);
+	ast_channel_nativeformats_set(chan, caps);
+	ast_channel_set_writeformat(chan, fmt);
+	ast_channel_set_rawwriteformat(chan, fmt);
+	ast_channel_set_readformat(chan, fmt);
+	ast_channel_set_rawreadformat(chan, fmt);
+
+	ast_channel_tech_pvt_set(chan, instance);
+
+	pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
+		ast_sockaddr_stringify_addr(&local_address));
+	ast_rtp_instance_get_local_address(instance, &local_address);
+	pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT",
+		ast_sockaddr_stringify_port(&local_address));
+
+	ast_channel_unlock(chan);
+
 	ao2_ref(fmt, -1);
 	ao2_ref(caps, -1);
 
@@ -188,6 +366,20 @@ failure:
 	return NULL;
 }
 
+/*! \brief Function called when our module is unloaded */
+static int unload_module(void)
+{
+	ast_channel_unregister(&multicast_rtp_tech);
+	ao2_cleanup(multicast_rtp_tech.capabilities);
+	multicast_rtp_tech.capabilities = NULL;
+
+	ast_channel_unregister(&unicast_rtp_tech);
+	ao2_cleanup(unicast_rtp_tech.capabilities);
+	unicast_rtp_tech.capabilities = NULL;
+
+	return 0;
+}
+
 /*! \brief Function called when our module is loaded */
 static int load_module(void)
 {
@@ -197,25 +389,25 @@ static int load_module(void)
 	ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
 	if (ast_channel_register(&multicast_rtp_tech)) {
 		ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
-		ao2_ref(multicast_rtp_tech.capabilities, -1);
+		unload_module();
-		multicast_rtp_tech.capabilities = NULL;
+		return AST_MODULE_LOAD_DECLINE;
+	}
+
+	if (!(unicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
+		unload_module();
+		return AST_MODULE_LOAD_DECLINE;
+	}
+	ast_format_cap_append_by_type(unicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
+	if (ast_channel_register(&unicast_rtp_tech)) {
+		ast_log(LOG_ERROR, "Unable to register channel class 'UnicastRTP'\n");
+		unload_module();
 		return AST_MODULE_LOAD_DECLINE;
 	}
 
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
-/*! \brief Function called when our module is unloaded */
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTP Media Channel",
-static int unload_module(void)
-{
-	ast_channel_unregister(&multicast_rtp_tech);
-	ao2_ref(multicast_rtp_tech.capabilities, -1);
-	multicast_rtp_tech.capabilities = NULL;
-
-	return 0;
-}
-
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel",
 	.support_level = AST_MODULE_SUPPORT_CORE,
 	.load = load_module,
 	.unload = unload_module,

include/asterisk/multicast_rtp.h

@@ -0,0 +1,58 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2016, Digium, Inc.
+ *
+ * Mark Michelson <mmichelson@digium.com>
+ *
+ * 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.
+ */
+
+#ifndef MULTICAST_RTP_H_
+#define MULTICAST_RTP_H_
+struct ast_multicast_rtp_options;
+
+/*!
+ * \brief Create multicast RTP options.
+ *
+ * These are passed to the multicast RTP engine on its creation.
+ *
+ * \param type The type of multicast RTP, either "basic" or "linksys"
+ * \param options Miscellaneous options
+ * \retval NULL Failure
+ * \retval non-NULL success
+ */
+struct ast_multicast_rtp_options *ast_multicast_rtp_create_options(const char *type,
+	const char *options);
+
+/*!
+ * \brief Free multicast RTP options
+ *
+ * This function is NULL-tolerant
+ *
+ * \param mcast_options Options to free
+ */
+void ast_multicast_rtp_free_options(struct ast_multicast_rtp_options *mcast_options);
+
+/*!
+ * \brief Get format specified in multicast options
+ *
+ * Multicast options allow for a format to be selected.
+ * This function accesses the selected format and creates
+ * an ast_format structure for it.
+ *
+ * \param mcast_options The options where a codec was specified
+ * \retval NULL No format specified in the options
+ * \revval non-NULL The format to use for communication
+ */
+struct ast_format *ast_multicast_rtp_options_get_format(struct ast_multicast_rtp_options *mcast_options);
+
+#endif /* MULTICAST_RTP_H_ */

res/res_rtp_multicast.c

@@ -54,6 +54,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/module.h"
 #include "asterisk/rtp_engine.h"
 #include "asterisk/format_cache.h"
+#include "asterisk/multicast_rtp.h"
+#include "asterisk/app.h"
 
 /*! Command value used for Linksys paging to indicate we are starting */
 #define LINKSYS_MCAST_STARTCMD 6
@@ -63,8 +65,10 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 /*! \brief Type of paging to do */
 enum multicast_type {
+	/*! Type has not been set yet */
+	MULTICAST_TYPE_UNSPECIFIED = 0,
 	/*! Simple multicast enabled client/receiver paging like Snom and Barix uses */
-	MULTICAST_TYPE_BASIC = 0,
+	MULTICAST_TYPE_BASIC,
 	/*! More advanced Linksys type paging which requires a start and stop packet */
 	MULTICAST_TYPE_LINKSYS,
 };
@@ -95,6 +99,91 @@ struct multicast_rtp {
 	struct timeval txcore;
 };
 
+enum {
+	OPT_CODEC = (1 << 0),
+	OPT_LOOP =  (1 << 1),
+	OPT_TTL =   (1 << 2),
+	OPT_IF =    (1 << 3),
+};
+
+enum {
+	OPT_ARG_CODEC = 0,
+	OPT_ARG_LOOP,
+	OPT_ARG_TTL,
+	OPT_ARG_IF,
+	OPT_ARG_ARRAY_SIZE,
+};
+
+AST_APP_OPTIONS(multicast_rtp_options, BEGIN_OPTIONS
+	/*! Set the codec to be used for multicast RTP */
+	AST_APP_OPTION_ARG('c', OPT_CODEC, OPT_ARG_CODEC),
+	/*! Set whether multicast RTP is looped back to the sender */
+	AST_APP_OPTION_ARG('l', OPT_LOOP, OPT_ARG_LOOP),
+	/*! Set the hop count for multicast RTP */
+	AST_APP_OPTION_ARG('t', OPT_TTL, OPT_ARG_TTL),
+	/*! Set the interface from which multicast RTP is sent */
+	AST_APP_OPTION_ARG('i', OPT_IF, OPT_ARG_IF),
+END_OPTIONS );
+
+struct ast_multicast_rtp_options {
+	char *type;
+	char *options;
+	struct ast_format *fmt;
+	struct ast_flags opts;
+	char *opt_args[OPT_ARG_ARRAY_SIZE];
+	/*! The type and options are stored in this buffer */
+	char buf[0];
+};
+
+struct ast_multicast_rtp_options *ast_multicast_rtp_create_options(const char *type,
+	const char *options)
+{
+	struct ast_multicast_rtp_options *mcast_options;
+	char *pos;
+
+	mcast_options = ast_calloc(1, sizeof(*mcast_options)
+			+ strlen(type)
+			+ strlen(options) + 2);
+	if (!mcast_options) {
+		return NULL;
+	}
+
+	pos = mcast_options->buf;
+
+	/* Safe */
+	strcpy(pos, type);
+	mcast_options->type = pos;
+	pos += strlen(type) + 1;
+
+	/* Safe */
+	strcpy(pos, options);
+	mcast_options->options = pos;
+
+	if (ast_app_parse_options(multicast_rtp_options, &mcast_options->opts,
+		mcast_options->opt_args, mcast_options->options)) {
+		ast_log(LOG_WARNING, "Error parsing multicast RTP options\n");
+		ast_multicast_rtp_free_options(mcast_options);
+		return NULL;
+	}
+
+	return mcast_options;
+}
+
+void ast_multicast_rtp_free_options(struct ast_multicast_rtp_options *mcast_options)
+{
+	ast_free(mcast_options);
+}
+
+struct ast_format *ast_multicast_rtp_options_get_format(struct ast_multicast_rtp_options *mcast_options)
+{
+	if (ast_test_flag(&mcast_options->opts, OPT_CODEC)
+		&& !ast_strlen_zero(mcast_options->opt_args[OPT_ARG_CODEC])) {
+		return ast_format_cache_get(mcast_options->opt_args[OPT_ARG_CODEC]);
+	}
+
+	return NULL;
+}
+
 /* Forward Declarations */
 static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data);
 static int multicast_rtp_activate(struct ast_rtp_instance *instance);
@@ -112,21 +201,93 @@ static struct ast_rtp_engine multicast_rtp_engine = {
 	.read = multicast_rtp_read,
 };
 
-/*! \brief Function called to create a new multicast instance */
+static int set_type(struct multicast_rtp *multicast, const char *type)
-static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
 {
-	struct multicast_rtp *multicast;
-	const char *type = data;
-
-	if (!(multicast = ast_calloc(1, sizeof(*multicast)))) {
-		return -1;
-	}
-
 	if (!strcasecmp(type, "basic")) {
 		multicast->type = MULTICAST_TYPE_BASIC;
 	} else if (!strcasecmp(type, "linksys")) {
 		multicast->type = MULTICAST_TYPE_LINKSYS;
 	} else {
+		ast_log(LOG_WARNING, "Unrecognized multicast type '%s' specified.\n", type);
+		return -1;
+	}
+
+	return 0;
+}
+
+static void set_ttl(int sock, const char *ttl_str)
+{
+	int ttl;
+
+	if (ast_strlen_zero(ttl_str)) {
+		return;
+	}
+
+	ast_debug(3, "Setting multicast TTL to %s\n", ttl_str);
+
+	if (sscanf(ttl_str, "%30d", &ttl) < 1) {
+		ast_log(LOG_WARNING, "Inavlid multicast ttl option '%s'\n", ttl_str);
+		return;
+	}
+
+	if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_TTL, &ttl, sizeof(ttl)) < 0) {
+		ast_log(LOG_WARNING, "Could not set multicast ttl to '%s': %s\n",
+			ttl_str, strerror(errno));
+	}
+}
+
+static void set_loop(int sock, const char *loop_str)
+{
+	unsigned char loop;
+
+	if (ast_strlen_zero(loop_str)) {
+		return;
+	}
+
+	ast_debug(3, "Setting multicast loop to %s\n", loop_str);
+
+	if (sscanf(loop_str, "%30hhu", &loop) < 1) {
+		ast_log(LOG_WARNING, "Invalid multicast loop option '%s'\n", loop_str);
+		return;
+	}
+
+	if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_LOOP, &loop, sizeof(loop)) < 0) {
+		ast_log(LOG_WARNING, "Could not set multicast loop to '%s': %s\n",
+			loop_str, strerror(errno));
+	}
+}
+
+static void set_if(int sock, const char *if_str)
+{
+	struct in_addr iface;
+
+	if (ast_strlen_zero(if_str)) {
+		return;
+	}
+
+	ast_debug(3, "Setting multicast if to %s\n", if_str);
+
+	if (!inet_aton(if_str, &iface)) {
+		ast_log(LOG_WARNING, "Cannot parse if option '%s'\n", if_str);
+	}
+
+	if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_IF, &iface, sizeof(iface)) < 0) {
+		ast_log(LOG_WARNING, "Could not set multicast if to '%s': %s\n",
+			if_str, strerror(errno));
+	}
+}
+
+/*! \brief Function called to create a new multicast instance */
+static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
+{
+	struct multicast_rtp *multicast;
+	struct ast_multicast_rtp_options *mcast_options = data;
+
+	if (!(multicast = ast_calloc(1, sizeof(*multicast)))) {
+		return -1;
+	}
+
+	if (set_type(multicast, mcast_options->type)) {
 		ast_free(multicast);
 		return -1;
 	}
@@ -136,6 +297,18 @@ static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched
 		return -1;
 	}
 
+	if (ast_test_flag(&mcast_options->opts, OPT_LOOP)) {
+		set_loop(multicast->socket, mcast_options->opt_args[OPT_ARG_LOOP]);
+	}
+
+	if (ast_test_flag(&mcast_options->opts, OPT_TTL)) {
+		set_ttl(multicast->socket, mcast_options->opt_args[OPT_ARG_TTL]);
+	}
+
+	if (ast_test_flag(&mcast_options->opts, OPT_IF)) {
+		set_if(multicast->socket, mcast_options->opt_args[OPT_ARG_IF]);
+	}
+
 	multicast->ssrc = ast_random();
 
 	ast_rtp_instance_set_data(instance, multicast);
@@ -314,7 +487,7 @@ static int unload_module(void)
 	return 0;
 }
 
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Engine",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Multicast RTP Engine",
 	.support_level = AST_MODULE_SUPPORT_CORE,
 	.load = load_module,
 	.unload = unload_module,

res/res_rtp_multicast.exports.in

@@ -0,0 +1,6 @@
+{
+	global:
+		LINKER_SYMBOL_PREFIXast_multicast_rtp*;
+	local:
+		*;
+};