[svn-commits] file: branch file/issue11797 r187719 - /team/file/issue11797/channels/

[SVN commits to the Digium repositories]

Author: file
Date: Fri Apr 10 11:37:13 2009
New Revision: 187719

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=187719
Log:
Add a channel driver which uses the multicast RTP engine to do multicast RTP paging.

Added:
    team/file/issue11797/channels/chan_multicast_rtp.c   (with props)

Added: team/file/issue11797/channels/chan_multicast_rtp.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/issue11797/channels/chan_multicast_rtp.c?view=auto&rev=187719
==============================================================================
--- team/file/issue11797/channels/chan_multicast_rtp.c (added)
+++ team/file/issue11797/channels/chan_multicast_rtp.c Fri Apr 10 11:37:13 2009
@@ -1,0 +1,184 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2009, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at digium.com>
+ * Andreas 'MacBrody' Brodmann <andreas.brodmann at 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 at digium.com>
+ * \author Andreas 'MacBrody' Broadmann <andreas.brodmann at 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 = { 0, }, destination_address = { 0, };
+	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_FAILURE;
+	}
+
+	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");

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