chan_rtp.c: Copy file from chan_multicast_rtp.c

Change-Id: I1119b53f2152ab1cbec74b5be7ea44844dbda8ef

diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c
new file mode 100644 (file)
index 0000000..267baab
--- /dev/null
+++ b/channels/chan_rtp.c
@@ -0,0 +1,223 @@
+/*
+ * 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
+ */
+
+/*** MODULEINFO
+       <support_level>core</support_level>
+ ***/
+
+#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, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
+static int multicast_rtp_call(struct ast_channel *ast, const 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 struct ast_channel_tech multicast_rtp_tech = {
+       .type = "MulticastRTP",
+       .description = tdesc,
+       .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_channel_tech_pvt(ast);
+
+       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, const char *dest, int timeout)
+{
+       struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
+
+       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_channel_tech_pvt(ast);
+
+       ast_rtp_instance_destroy(instance);
+
+       ast_channel_tech_pvt_set(ast, NULL);
+
+       return 0;
+}
+
+/*! \brief Function called when we should prepare to call the destination */
+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)
+{
+       char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
+       struct ast_rtp_instance *instance;
+       struct ast_sockaddr control_address;
+       struct ast_sockaddr destination_address;
+       struct ast_channel *chan;
+       struct ast_format_cap *caps = NULL;
+       struct ast_format *fmt = NULL;
+
+       fmt = ast_format_cap_get_format(cap, 0);
+
+       ast_sockaddr_setnull(&control_address);
+
+       /* 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 ((control = strchr(destination, '/'))) {
+               *control++ = '\0';
+               if (!ast_sockaddr_parse(&control_address, control,
+                                       PARSE_PORT_REQUIRE)) {
+                       goto failure;
+               }
+       }
+
+       if (!ast_sockaddr_parse(&destination_address, destination,
+                               PARSE_PORT_REQUIRE)) {
+               goto failure;
+       }
+
+       caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+       if (!caps) {
+               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, "", "", "", "", "", assignedids, requestor, 0, "MulticastRTP/%p", instance))) {
+               ast_rtp_instance_destroy(instance);
+               goto failure;
+       }
+       ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
+       ast_rtp_instance_set_remote_address(instance, &destination_address);
+
+       ast_channel_tech_set(chan, &multicast_rtp_tech);
+
+       ast_format_cap_append(caps, fmt, 0);
+       ast_channel_nativeformats_set(chan, caps);
+       ast_channel_set_writeformat(chan, fmt);
+       ast_channel_set_rawwriteformat(chan, fmt);
+       ast_channel_set_readformat(chan, fmt);
+       ast_channel_set_rawreadformat(chan, fmt);
+
+       ast_channel_tech_pvt_set(chan, instance);
+
+       ast_channel_unlock(chan);
+
+       ao2_ref(fmt, -1);
+       ao2_ref(caps, -1);
+
+       return chan;
+
+failure:
+       ao2_cleanup(fmt);
+       ao2_cleanup(caps);
+       *cause = AST_CAUSE_FAILURE;
+       return NULL;
+}
+
+/*! \brief Function called when our module is loaded */
+static int load_module(void)
+{
+       if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
+               return AST_MODULE_LOAD_DECLINE;
+       }
+       ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
+       if (ast_channel_register(&multicast_rtp_tech)) {
+               ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
+               ao2_ref(multicast_rtp_tech.capabilities, -1);
+               multicast_rtp_tech.capabilities = NULL;
+               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);
+       ao2_ref(multicast_rtp_tech.capabilities, -1);
+       multicast_rtp_tech.capabilities = NULL;
+
+       return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel",
+       .support_level = AST_MODULE_SUPPORT_CORE,
+       .load = load_module,
+       .unload = unload_module,
+       .load_pri = AST_MODPRI_CHANNEL_DRIVER,
+);