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	This module supports sending both unicast and multicast RTP to a specified target. Multicast functionality is the same as chan_multicast_rtp was. In the case of unicast a specific IP address and port can be specified, along with optional RTP engine and format in the form of: UnicastRTP/<ip address>:<port>/<engine>/<format> This can be useful for sending a copy of a media stream to another application for processing. Review: https://reviewboard.asterisk.org/r/3981/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
		
			
				
	
	
		
			336 lines
		
	
	
		
			10 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			336 lines
		
	
	
		
			10 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * Asterisk -- An open source telephony toolkit.
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|  *
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|  * Copyright (C) 2009 - 2014, 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 RTP (Multicast and Unicast) Media 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 "asterisk/channel.h"
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| #include "asterisk/module.h"
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| #include "asterisk/pbx.h"
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| #include "asterisk/acl.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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| #include "asterisk/format_cache.h"
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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 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);
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| static int rtp_call(struct ast_channel *ast, const char *dest, int timeout);
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| static int rtp_hangup(struct ast_channel *ast);
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| static struct ast_frame *rtp_read(struct ast_channel *ast);
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| static int rtp_write(struct ast_channel *ast, struct ast_frame *f);
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| 
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| /* Multicast 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 = "Multicast RTP Paging Channel Driver",
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| 	.requester = multicast_rtp_request,
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| 	.call = rtp_call,
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| 	.hangup = rtp_hangup,
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| 	.read = rtp_read,
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| 	.write = rtp_write,
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| };
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| 
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| /* Unicast channel driver declaration */
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| static struct ast_channel_tech unicast_rtp_tech = {
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| 	.type = "UnicastRTP",
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| 	.description = "Unicast RTP Media Channel Driver",
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| 	.requester = unicast_rtp_request,
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| 	.call = rtp_call,
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| 	.hangup = rtp_hangup,
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| 	.read = rtp_read,
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| 	.write = 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  *rtp_read(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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| 	int fdno = ast_channel_fdno(ast);
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| 
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| 	switch (fdno) {
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| 	case 0:
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| 		return ast_rtp_instance_read(instance, 0);
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| 	default:
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| 		return &ast_null_frame;
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| 	}
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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 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 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 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 multicast 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 *parse;
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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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| 	AST_DECLARE_APP_ARGS(args,
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| 		AST_APP_ARG(type);
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| 		AST_APP_ARG(destination);
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| 		AST_APP_ARG(control);
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| 	);
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| 
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| 	if (ast_strlen_zero(data)) {
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| 		ast_log(LOG_ERROR, "A multicast type and destination must be given to the 'MulticastRTP' channel\n");
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| 		goto failure;
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| 	}
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| 	parse = ast_strdupa(data);
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| 	AST_NONSTANDARD_APP_ARGS(args, parse, '/');
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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 (!ast_strlen_zero(args.control) &&
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| 		!ast_sockaddr_parse(&control_address, args.control, PARSE_PORT_REQUIRE)) {
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| 		ast_log(LOG_ERROR, "Control address '%s' could not be parsed\n", args.control);
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| 		goto failure;
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| 	}
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| 
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| 	if (!ast_sockaddr_parse(&destination_address, args.destination,
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| 				PARSE_PORT_REQUIRE)) {
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| 		ast_log(LOG_ERROR, "Destination address '%s' could not be parsed\n", args.destination);
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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, args.type))) {
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| 		ast_log(LOG_ERROR, "Could not create RTP instance for sending media to '%s'\n", args.destination);
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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 we should prepare to call the unicast destination */
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| 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)
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| {
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| 	char *parse;
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| 	struct ast_rtp_instance *instance;
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| 	struct ast_sockaddr address;
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| 	struct ast_sockaddr local_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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| 	AST_DECLARE_APP_ARGS(args,
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| 		AST_APP_ARG(destination);
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| 		AST_APP_ARG(engine);
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| 		AST_APP_ARG(format);
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| 	);
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| 
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| 	if (ast_strlen_zero(data)) {
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| 		goto failure;
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| 	}
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| 	parse = ast_strdupa(data);
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| 	AST_NONSTANDARD_APP_ARGS(args, parse, '/');
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| 
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| 	if (!ast_strlen_zero(args.format)) {
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| 		fmt = ast_format_cache_get(args.format);
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| 	} else {
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| 		fmt = ast_format_cap_get_format(cap, 0);
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| 	}
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| 
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| 	if (!fmt) {
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| 		ast_log(LOG_ERROR, "No format specified for sending RTP to '%s'\n", args.destination);
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| 		goto failure;
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| 	}
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| 
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| 	if (!ast_sockaddr_parse(&address, args.destination,
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| 				PARSE_PORT_REQUIRE)) {
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| 		ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination);
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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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| 	ast_ouraddrfor(&address, &local_address);
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| 	if (!(instance = ast_rtp_instance_new(args.engine, NULL, &local_address, NULL))) {
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| 		ast_log(LOG_ERROR, "Could not create RTP instance for sending media to '%s'\n", args.destination);
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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, "UnicastRTP/%s-%p", args.destination, 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, &address);
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| 	ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(instance, 0));
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| 
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| 	ast_channel_tech_set(chan, &unicast_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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| 	pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS", ast_sockaddr_stringify_addr(&local_address));
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| 	ast_rtp_instance_get_local_address(instance, &local_address);
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| 	pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT", ast_sockaddr_stringify_port(&local_address));
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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 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_cleanup(multicast_rtp_tech.capabilities);
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| 	multicast_rtp_tech.capabilities = NULL;
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| 
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| 	ast_channel_unregister(&unicast_rtp_tech);
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| 	ao2_cleanup(unicast_rtp_tech.capabilities);
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| 	unicast_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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| /*! \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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| 		unload_module();
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| 		return AST_MODULE_LOAD_DECLINE;
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| 	}
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| 
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| 	if (!(unicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
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| 		unload_module();
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| 		return AST_MODULE_LOAD_DECLINE;
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| 	}
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| 	ast_format_cap_append_by_type(unicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
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| 	if (ast_channel_register(&unicast_rtp_tech)) {
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| 		ast_log(LOG_ERROR, "Unable to register channel class 'UnicastRTP'\n");
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| 		unload_module();
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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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| AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTP Media 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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