Add support for multicast RTP paging.

(closes issue #11797)
Reported by: macbrody

Review: https://reviewboard.asterisk.org/r/270/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203227 f38db490-d61c-443f-a65b-d21fe96a405b

3 files changed, 455 insertions, 0 deletions

diff --git a/CHANGES b/CHANGES
index 8a387244d..c66bf68ca 100644
--- a/CHANGES
+++ b/CHANGES
@@ -173,6 +173,16 @@ Calendaring for Asterisk
    iCalendar, CalDAV, and Exchange Server calendars are supported (Exchange support
    only tested on Exchange Server 2003 with no support for forms-based authentication).
 
+Multicast RTP Support
+---------------------
+ * A new RTP engine and channel driver have been added which supports Multicast RTP.
+   The channel driver can be used with the Page application to perform multicast RTP
+   paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
+   Type can be either basic or linksys.
+   Destination is the IP address and port for the RTP packets.
+   Control address is specific to the linksys type and is used for sending the control
+   packets unique to them.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2  -------------
 ------------------------------------------------------------------------------
diff --git a/channels/chan_multicast_rtp.c b/channels/chan_multicast_rtp.c
new file mode 100644
index 000000000..589fa1008
--- /dev/null
+++ b/channels/chan_multicast_rtp.c
@@ -0,0 +1,184 @@
+/*
+ * 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
+ *
+ * \author Joshua Colp <jcolp@digium.com>
+ * \author Andreas 'MacBrody' Broadmann <andreas.brodmann@gmail.com>
+ *
+ * \brief Multicast RTP Paging Channel
+ *
+ * \ingroup channel_drivers
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#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"
+
+static const char tdesc[] = "Multicast RTP Paging Channel Driver";
+
+/* Forward declarations */
+static struct ast_channel *multicast_rtp_request(const char *type, int format, void *data, int *cause);
+static int multicast_rtp_call(struct ast_channel *ast, char *dest, int timeout);
+static int multicast_rtp_hangup(struct ast_channel *ast);
+static struct ast_frame *multicast_rtp_read(struct ast_channel *ast);
+static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f);
+
+/* Channel driver declaration */
+static const struct ast_channel_tech multicast_rtp_tech = {
+	.type = "MulticastRTP",
+	.description = tdesc,
+	.capabilities = -1,
+	.requester = multicast_rtp_request,
+	.call = multicast_rtp_call,
+	.hangup = multicast_rtp_hangup,
+	.read = multicast_rtp_read,
+	.write = multicast_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)
+{
+	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)
+{
+	struct ast_rtp_instance *instance = ast->tech_pvt;
+
+	return ast_rtp_instance_write(instance, f);
+}
+
+/*! \brief Function called when we should actually call the destination */
+static int multicast_rtp_call(struct ast_channel *ast, char *dest, int timeout)
+{
+	struct ast_rtp_instance *instance = ast->tech_pvt;
+
+	ast_queue_control(ast, AST_CONTROL_ANSWER);
+
+	return ast_rtp_instance_activate(instance);
+}
+
+/*! \brief Function called when we should hang the channel up */
+static int multicast_rtp_hangup(struct ast_channel *ast)
+{
+	struct ast_rtp_instance *instance = ast->tech_pvt;
+
+	ast_rtp_instance_destroy(instance);
+
+	ast->tech_pvt = NULL;
+
+	return 0;
+}
+
+/*! \brief Function called when we should prepare to call the destination */
+static struct ast_channel *multicast_rtp_request(const char *type, int format, void *data, int *cause)
+{
+	char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
+	struct ast_rtp_instance *instance;
+	struct sockaddr_in control_address = { .sin_family = AF_INET, }, destination_address = { .sin_family = AF_INET, };
+	struct ast_channel *chan;
+	int fmt = ast_best_codec(format);
+
+	/* If no type was given we can't do anything */
+	if (ast_strlen_zero(multicast_type)) {
+		goto failure;
+	}
+
+	if (!(destination = strchr(tmp, '/'))) {
+		goto failure;
+	}
+	*destination++ = '\0';
+
+	if (ast_parse_arg(destination, PARSE_INADDR | PARSE_PORT_REQUIRE, &destination_address)) {
+		goto failure;
+	}
+
+	if ((control = strchr(destination, '/'))) {
+		*control++ = '\0';
+		if (ast_parse_arg(control, PARSE_INADDR | PARSE_PORT_REQUIRE, &control_address)) {
+			goto failure;
+		}
+	}
+
+	if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, multicast_type))) {
+		goto failure;
+	}
+
+	if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", 0, "MulticastRTP/%p", instance))) {
+		ast_rtp_instance_destroy(instance);
+		goto failure;
+	}
+
+	ast_rtp_instance_set_remote_address(instance, &destination_address);
+
+	chan->tech = &multicast_rtp_tech;
+	chan->nativeformats = fmt;
+	chan->writeformat = fmt;
+	chan->readformat = fmt;
+	chan->rawwriteformat = fmt;
+	chan->rawreadformat = fmt;
+	chan->tech_pvt = instance;
+
+	return chan;
+
+failure:
+	*cause = AST_CAUSE_FAILURE;
+	return NULL;
+}
+
+/*! \brief Function called when our module is loaded */
+static int load_module(void)
+{
+	if (ast_channel_register(&multicast_rtp_tech)) {
+		ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
+		return AST_MODULE_LOAD_DECLINE;
+	}
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+/*! \brief Function called when our module is unloaded */
+static int unload_module(void)
+{
+	ast_channel_unregister(&multicast_rtp_tech);
+
+	return 0;
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multicast RTP Paging Channel");
diff --git a/res/res_rtp_multicast.c b/res/res_rtp_multicast.c
new file mode 100644
index 000000000..3752ebefc
--- /dev/null
+++ b/res/res_rtp_multicast.c
@@ -0,0 +1,261 @@
+/*
+ * 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>
+ */
+
+#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/netsock.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 sched_context *sched, struct sockaddr_in *sin, 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 sched_context *sched, struct sockaddr_in *sin, 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 sockaddr_in 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 (!control_address.sin_addr.s_addr || !remote_address.sin_addr.s_addr) {
+		return -1;
+	}
+
+	control_packet.ip = remote_address.sin_addr.s_addr;
+	control_packet.port = htonl(ntohs(remote_address.sin_port));
+
+	/* Based on a recommendation by Brian West who did the FreeSWITCH implementation we send control packets twice */
+	sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, (struct sockaddr *)&control_address, sizeof(control_address));
+	sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, (struct sockaddr *)&control_address, sizeof(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 sockaddr_in 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)) < 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 = sendto(multicast->socket, (void *) rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *) &remote_address, sizeof(remote_address));
+
+	if (res < 0) {
+		ast_log(LOG_ERROR, "Multicast RTP Transmission error to %s:%u: %s\n",
+			ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port), 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_STANDARD(ASTERISK_GPL_KEY, "Multicast RTP Engine");