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authorJoshua Colp <jcolp@digium.com>2009-06-25 18:25:24 +0000
committerJoshua Colp <jcolp@digium.com>2009-06-25 18:25:24 +0000
commitae87ba45b544cd5ee56211d8fcba2f705d445c49 (patch)
tree7c1c2a94fa367bd30f955ba44d4bdc41eea2a53c /channels/chan_multicast_rtp.c
parentca3a181c33cdf7155ffc34a4c8ac79f09b684a13 (diff)
Add support for multicast RTP paging.
(closes issue #11797) Reported by: macbrody Review: https://reviewboard.asterisk.org/r/270/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels/chan_multicast_rtp.c')
-rw-r--r--channels/chan_multicast_rtp.c184
1 files changed, 184 insertions, 0 deletions
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");