aboutsummaryrefslogtreecommitdiffstats
path: root/res/res_rtp_multicast.c
blob: 0e930d61f3504ec15efec649c52b262290eaf150 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2009, Digium, Inc.
 *
 * Joshua Colp <jcolp@digium.com>
 * Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.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.
 */

/*!
 * \file
 *
 * \brief Multicast RTP Engine
 *
 * \author Joshua Colp <jcolp@digium.com>
 * \author Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
 *
 * \ingroup rtp_engines
 */

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include <sys/time.h>
#include <signal.h>
#include <fcntl.h>
#include <math.h>

#include "asterisk/pbx.h"
#include "asterisk/frame.h"
#include "asterisk/channel.h"
#include "asterisk/acl.h"
#include "asterisk/config.h"
#include "asterisk/lock.h"
#include "asterisk/utils.h"
#include "asterisk/cli.h"
#include "asterisk/manager.h"
#include "asterisk/unaligned.h"
#include "asterisk/module.h"
#include "asterisk/rtp_engine.h"

/*! Command value used for Linksys paging to indicate we are starting */
#define LINKSYS_MCAST_STARTCMD 6

/*! Command value used for Linksys paging to indicate we are stopping */
#define LINKSYS_MCAST_STOPCMD 7

/*! \brief Type of paging to do */
enum multicast_type {
	/*! Simple multicast enabled client/receiver paging like Snom and Barix uses */
	MULTICAST_TYPE_BASIC = 0,
	/*! More advanced Linksys type paging which requires a start and stop packet */
	MULTICAST_TYPE_LINKSYS,
};

/*! \brief Structure for a Linksys control packet */
struct multicast_control_packet {
	/*! Unique identifier for the control packet */
	uint32_t unique_id;
	/*! Actual command in the control packet */
	uint32_t command;
	/*! IP address for the RTP */
	uint32_t ip;
	/*! Port for the RTP */
	uint32_t port;
};

/*! \brief Structure for a multicast paging instance */
struct multicast_rtp {
	/*! TYpe of multicast paging this instance is doing */
	enum multicast_type type;
	/*! Socket used for sending the audio on */
	int socket;
	/*! Synchronization source value, used when creating/sending the RTP packet */
	unsigned int ssrc;
	/*! Sequence number, used when creating/sending the RTP packet */
	unsigned int seqno;
};

/* 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);
static int multicast_rtp_destroy(struct ast_rtp_instance *instance);
static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp);

/* RTP Engine Declaration */
static struct ast_rtp_engine multicast_rtp_engine = {
	.name = "multicast",
	.new = multicast_rtp_new,
	.activate = multicast_rtp_activate,
	.destroy = multicast_rtp_destroy,
	.write = multicast_rtp_write,
	.read = multicast_rtp_read,
};

/*! \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;
	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_free(multicast);
		return -1;
	}

	if ((multicast->socket = socket(AF_INET, SOCK_DGRAM, 0)) < 0) {
		ast_free(multicast);
		return -1;
	}

	multicast->ssrc = ast_random();

	ast_rtp_instance_set_data(instance, multicast);

	return 0;
}

/*! \brief Helper function which populates a control packet with useful information and sends it */
static int multicast_send_control_packet(struct ast_rtp_instance *instance, struct multicast_rtp *multicast, int command)
{
	struct multicast_control_packet control_packet = { .unique_id = htonl((u_long)time(NULL)),
							   .command = htonl(command),
	};
	struct ast_sockaddr control_address, remote_address;

	ast_rtp_instance_get_local_address(instance, &control_address);
	ast_rtp_instance_get_remote_address(instance, &remote_address);

	/* Ensure the user of us have given us both the control address and destination address */
	if (ast_sockaddr_isnull(&control_address) ||
	    ast_sockaddr_isnull(&remote_address)) {
		return -1;
	}

	/* The protocol only supports IPv4. */
	if (ast_sockaddr_is_ipv6(&remote_address)) {
		ast_log(LOG_WARNING, "Cannot send control packet for IPv6 "
			"remote address.\n");
		return -1;
	}

	control_packet.ip = htonl(ast_sockaddr_ipv4(&remote_address));
	control_packet.port = htonl(ast_sockaddr_port(&remote_address));

	/* Based on a recommendation by Brian West who did the FreeSWITCH implementation we send control packets twice */
	ast_sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, &control_address);
	ast_sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, &control_address);

	return 0;
}

/*! \brief Function called to indicate that audio is now going to flow */
static int multicast_rtp_activate(struct ast_rtp_instance *instance)
{
	struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);

	if (multicast->type != MULTICAST_TYPE_LINKSYS) {
		return 0;
	}

	return multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STARTCMD);
}

/*! \brief Function called to destroy a multicast instance */
static int multicast_rtp_destroy(struct ast_rtp_instance *instance)
{
	struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);

	if (multicast->type == MULTICAST_TYPE_LINKSYS) {
		multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STOPCMD);
	}

	close(multicast->socket);

	ast_free(multicast);

	return 0;
}

/*! \brief Function called to broadcast some audio on a multicast instance */
static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
{
	struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
	struct ast_frame *f = frame;
	struct ast_sockaddr remote_address;
	int hdrlen = 12, res, codec;
	unsigned char *rtpheader;

	/* We only accept audio, nothing else */
	if (frame->frametype != AST_FRAME_VOICE) {
		return 0;
	}

	/* Grab the actual payload number for when we create the RTP packet */
	if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, &frame->subclass.format, 0)) < 0) {
		return -1;
	}

	/* If we do not have space to construct an RTP header duplicate the frame so we get some */
	if (frame->offset < hdrlen) {
		f = ast_frdup(frame);
	}

	/* Construct an RTP header for our packet */
	rtpheader = (unsigned char *)(f->data.ptr - hdrlen);
	put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno++) | (0 << 23)));
	put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
	put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));

	/* Finally send it out to the eager phones listening for us */
	ast_rtp_instance_get_remote_address(instance, &remote_address);
	res = ast_sendto(multicast->socket, (void *) rtpheader, f->datalen + hdrlen, 0, &remote_address);

	if (res < 0) {
		ast_log(LOG_ERROR, "Multicast RTP Transmission error to %s: %s\n",
			ast_sockaddr_stringify(&remote_address),
			strerror(errno));
	}

	/* If we were forced to duplicate the frame free the new one */
	if (frame != f) {
		ast_frfree(f);
	}

	return res;
}

/*! \brief Function called to read from a multicast instance */
static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
{
	return &ast_null_frame;
}

static int load_module(void)
{
	if (ast_rtp_engine_register(&multicast_rtp_engine)) {
		return AST_MODULE_LOAD_DECLINE;
	}

	return AST_MODULE_LOAD_SUCCESS;
}

static int unload_module(void)
{
	ast_rtp_engine_unregister(&multicast_rtp_engine);

	return 0;
}

AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Engine",
	.load = load_module,
	.unload = unload_module,
	.load_pri = AST_MODPRI_CHANNEL_DEPEND,
);