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							224 lines
						
					
					
						
							6.4 KiB
						
					
					
				| /*
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|  * Asterisk -- An open source telephony toolkit.
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|  *
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|  * Copyright (C) 2009, Digium, Inc.
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|  *
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|  * Joshua Colp <jcolp@digium.com>
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|  * Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
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|  *
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|  * See http://www.asterisk.org for more information about
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|  * the Asterisk project. Please do not directly contact
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|  * any of the maintainers of this project for assistance;
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|  * the project provides a web site, mailing lists and IRC
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|  * channels for your use.
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|  *
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|  * This program is free software, distributed under the terms of
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|  * the GNU General Public License Version 2. See the LICENSE file
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|  * at the top of the source tree.
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|  */
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| 
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| /*! \file
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|  *
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|  * \author Joshua Colp <jcolp@digium.com>
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|  * \author Andreas 'MacBrody' Broadmann <andreas.brodmann@gmail.com>
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|  *
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|  * \brief Multicast RTP Paging Channel
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|  *
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|  * \ingroup channel_drivers
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|  */
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| 
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| /*** MODULEINFO
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| 	<support_level>core</support_level>
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|  ***/
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| 
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| #include "asterisk.h"
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| 
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| ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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| 
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| #include <fcntl.h>
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| #include <sys/signal.h>
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| 
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| #include "asterisk/lock.h"
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| #include "asterisk/channel.h"
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| #include "asterisk/config.h"
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| #include "asterisk/module.h"
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| #include "asterisk/pbx.h"
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| #include "asterisk/sched.h"
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| #include "asterisk/io.h"
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| #include "asterisk/acl.h"
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| #include "asterisk/callerid.h"
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| #include "asterisk/file.h"
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| #include "asterisk/cli.h"
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| #include "asterisk/app.h"
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| #include "asterisk/rtp_engine.h"
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| #include "asterisk/causes.h"
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| 
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| static const char tdesc[] = "Multicast RTP Paging Channel Driver";
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| 
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| /* Forward declarations */
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| 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);
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| static int multicast_rtp_call(struct ast_channel *ast, const char *dest, int timeout);
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| static int multicast_rtp_hangup(struct ast_channel *ast);
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| static struct ast_frame *multicast_rtp_read(struct ast_channel *ast);
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| static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f);
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| 
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| /* Channel driver declaration */
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| static struct ast_channel_tech multicast_rtp_tech = {
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| 	.type = "MulticastRTP",
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| 	.description = tdesc,
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| 	.requester = multicast_rtp_request,
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| 	.call = multicast_rtp_call,
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| 	.hangup = multicast_rtp_hangup,
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| 	.read = multicast_rtp_read,
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| 	.write = multicast_rtp_write,
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| };
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| 
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| /*! \brief Function called when we should read a frame from the channel */
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| static struct ast_frame  *multicast_rtp_read(struct ast_channel *ast)
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| {
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| 	return &ast_null_frame;
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| }
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| 
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| /*! \brief Function called when we should write a frame to the channel */
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| static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f)
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| {
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| 	struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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| 
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| 	return ast_rtp_instance_write(instance, f);
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| }
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| 
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| /*! \brief Function called when we should actually call the destination */
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| static int multicast_rtp_call(struct ast_channel *ast, const char *dest, int timeout)
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| {
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| 	struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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| 
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| 	ast_queue_control(ast, AST_CONTROL_ANSWER);
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| 
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| 	return ast_rtp_instance_activate(instance);
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| }
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| 
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| /*! \brief Function called when we should hang the channel up */
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| static int multicast_rtp_hangup(struct ast_channel *ast)
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| {
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| 	struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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| 
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| 	ast_rtp_instance_destroy(instance);
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| 
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| 	ast_channel_tech_pvt_set(ast, NULL);
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| 
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| 	return 0;
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| }
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| 
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| /*! \brief Function called when we should prepare to call the destination */
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| 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)
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| {
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| 	char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
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| 	struct ast_rtp_instance *instance;
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| 	struct ast_sockaddr control_address;
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| 	struct ast_sockaddr destination_address;
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| 	struct ast_channel *chan;
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| 	struct ast_format_cap *caps = NULL;
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| 	struct ast_format *fmt = NULL;
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| 
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| 	fmt = ast_format_cap_get_format(cap, 0);
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| 
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| 	ast_sockaddr_setnull(&control_address);
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| 
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| 	/* If no type was given we can't do anything */
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| 	if (ast_strlen_zero(multicast_type)) {
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| 		goto failure;
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| 	}
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| 
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| 	if (!(destination = strchr(tmp, '/'))) {
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| 		goto failure;
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| 	}
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| 	*destination++ = '\0';
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| 
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| 	if ((control = strchr(destination, '/'))) {
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| 		*control++ = '\0';
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| 		if (!ast_sockaddr_parse(&control_address, control,
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| 					PARSE_PORT_REQUIRE)) {
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| 			goto failure;
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| 		}
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| 	}
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| 
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| 	if (!ast_sockaddr_parse(&destination_address, destination,
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| 				PARSE_PORT_REQUIRE)) {
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| 		goto failure;
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| 	}
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| 
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| 	caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
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| 	if (!caps) {
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| 		goto failure;
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| 	}
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| 
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| 	if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, multicast_type))) {
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| 		goto failure;
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| 	}
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| 
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| 	if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids, requestor, 0, "MulticastRTP/%p", instance))) {
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| 		ast_rtp_instance_destroy(instance);
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| 		goto failure;
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| 	}
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| 	ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
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| 	ast_rtp_instance_set_remote_address(instance, &destination_address);
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| 
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| 	ast_channel_tech_set(chan, &multicast_rtp_tech);
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| 
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| 	ast_format_cap_append(caps, fmt, 0);
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| 	ast_channel_nativeformats_set(chan, caps);
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| 	ast_channel_set_writeformat(chan, fmt);
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| 	ast_channel_set_rawwriteformat(chan, fmt);
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| 	ast_channel_set_readformat(chan, fmt);
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| 	ast_channel_set_rawreadformat(chan, fmt);
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| 
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| 	ast_channel_tech_pvt_set(chan, instance);
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| 
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| 	ast_channel_unlock(chan);
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| 
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| 	ao2_ref(fmt, -1);
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| 	ao2_ref(caps, -1);
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| 
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| 	return chan;
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| 
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| failure:
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| 	ao2_cleanup(fmt);
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| 	ao2_cleanup(caps);
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| 	*cause = AST_CAUSE_FAILURE;
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| 	return NULL;
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| }
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| 
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| /*! \brief Function called when our module is loaded */
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| static int load_module(void)
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| {
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| 	if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
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| 		return AST_MODULE_LOAD_DECLINE;
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| 	}
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| 	ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
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| 	if (ast_channel_register(&multicast_rtp_tech)) {
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| 		ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
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| 		ao2_ref(multicast_rtp_tech.capabilities, -1);
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| 		multicast_rtp_tech.capabilities = NULL;
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| 		return AST_MODULE_LOAD_DECLINE;
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| 	}
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| 
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| 	return AST_MODULE_LOAD_SUCCESS;
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| }
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| 
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| /*! \brief Function called when our module is unloaded */
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| static int unload_module(void)
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| {
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| 	ast_channel_unregister(&multicast_rtp_tech);
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| 	ao2_ref(multicast_rtp_tech.capabilities, -1);
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| 	multicast_rtp_tech.capabilities = NULL;
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| 
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| 	return 0;
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| }
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| 
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| AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel",
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| 	.support_level = AST_MODULE_SUPPORT_CORE,
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| 	.load = load_module,
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| 	.unload = unload_module,
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| 	.load_pri = AST_MODPRI_CHANNEL_DRIVER,
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| );
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