chan_rtp: Add unicast RTP support.
authorJoshua Colp <jcolp@digium.com>
Fri, 12 Sep 2014 17:42:15 +0000 (17:42 +0000)
committerJoshua Colp <jcolp@digium.com>
Fri, 12 Sep 2014 17:42:15 +0000 (17:42 +0000)
This module supports sending both unicast and multicast RTP
to a specified target. Multicast functionality is the same as
chan_multicast_rtp was. In the case of unicast a specific
IP address and port can be specified, along with optional RTP
engine and format in the form of:

UnicastRTP/<ip address>:<port>/<engine>/<format>

This can be useful for sending a copy of a media stream to
another application for processing.

Review: https://reviewboard.asterisk.org/r/3981/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_multicast_rtp.c [deleted file]
channels/chan_rtp.c [new file with mode: 0644]

diff --git a/channels/chan_multicast_rtp.c b/channels/chan_multicast_rtp.c
deleted file mode 100644 (file)
index 267baab..0000000
+++ /dev/null
@@ -1,223 +0,0 @@
-/*
- * 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,
-);
diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c
new file mode 100644 (file)
index 0000000..97fbf9f
--- /dev/null
@@ -0,0 +1,335 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2009 - 2014, 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 RTP (Multicast and Unicast) Media Channel
+ *
+ * \ingroup channel_drivers
+ */
+
+/*** MODULEINFO
+       <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/channel.h"
+#include "asterisk/module.h"
+#include "asterisk/pbx.h"
+#include "asterisk/acl.h"
+#include "asterisk/app.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/causes.h"
+#include "asterisk/format_cache.h"
+
+/* 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 struct ast_channel *unicast_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 rtp_call(struct ast_channel *ast, const char *dest, int timeout);
+static int rtp_hangup(struct ast_channel *ast);
+static struct ast_frame *rtp_read(struct ast_channel *ast);
+static int rtp_write(struct ast_channel *ast, struct ast_frame *f);
+
+/* Multicast channel driver declaration */
+static struct ast_channel_tech multicast_rtp_tech = {
+       .type = "MulticastRTP",
+       .description = "Multicast RTP Paging Channel Driver",
+       .requester = multicast_rtp_request,
+       .call = rtp_call,
+       .hangup = rtp_hangup,
+       .read = rtp_read,
+       .write = rtp_write,
+};
+
+/* Unicast channel driver declaration */
+static struct ast_channel_tech unicast_rtp_tech = {
+       .type = "UnicastRTP",
+       .description = "Unicast RTP Media Channel Driver",
+       .requester = unicast_rtp_request,
+       .call = rtp_call,
+       .hangup = rtp_hangup,
+       .read = rtp_read,
+       .write = rtp_write,
+};
+
+/*! \brief Function called when we should read a frame from the channel */
+static struct ast_frame  *rtp_read(struct ast_channel *ast)
+{
+       struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
+       int fdno = ast_channel_fdno(ast);
+
+       switch (fdno) {
+       case 0:
+               return ast_rtp_instance_read(instance, 0);
+       default:
+               return &ast_null_frame;
+       }
+}
+
+/*! \brief Function called when we should write a frame to the channel */
+static int 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 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 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 multicast 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 *parse;
+       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;
+       AST_DECLARE_APP_ARGS(args,
+               AST_APP_ARG(type);
+               AST_APP_ARG(destination);
+               AST_APP_ARG(control);
+       );
+
+       if (ast_strlen_zero(data)) {
+               ast_log(LOG_ERROR, "A multicast type and destination must be given to the 'MulticastRTP' channel\n");
+               goto failure;
+       }
+       parse = ast_strdupa(data);
+       AST_NONSTANDARD_APP_ARGS(args, parse, '/');
+
+       fmt = ast_format_cap_get_format(cap, 0);
+
+       ast_sockaddr_setnull(&control_address);
+
+       if (!ast_strlen_zero(args.control) &&
+               !ast_sockaddr_parse(&control_address, args.control, PARSE_PORT_REQUIRE)) {
+               ast_log(LOG_ERROR, "Control address '%s' could not be parsed\n", args.control);
+               goto failure;
+       }
+
+       if (!ast_sockaddr_parse(&destination_address, args.destination,
+                               PARSE_PORT_REQUIRE)) {
+               ast_log(LOG_ERROR, "Destination address '%s' could not be parsed\n", args.destination);
+               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, args.type))) {
+               ast_log(LOG_ERROR, "Could not create RTP instance for sending media to '%s'\n", args.destination);
+               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 we should prepare to call the unicast destination */
+static struct ast_channel *unicast_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 *parse;
+       struct ast_rtp_instance *instance;
+       struct ast_sockaddr address;
+       struct ast_sockaddr local_address;
+       struct ast_channel *chan;
+       struct ast_format_cap *caps = NULL;
+       struct ast_format *fmt = NULL;
+       AST_DECLARE_APP_ARGS(args,
+               AST_APP_ARG(destination);
+               AST_APP_ARG(engine);
+               AST_APP_ARG(format);
+       );
+
+       if (ast_strlen_zero(data)) {
+               goto failure;
+       }
+       parse = ast_strdupa(data);
+       AST_NONSTANDARD_APP_ARGS(args, parse, '/');
+
+       if (!ast_strlen_zero(args.format)) {
+               fmt = ast_format_cache_get(args.format);
+       } else {
+               fmt = ast_format_cap_get_format(cap, 0);
+       }
+
+       if (!fmt) {
+               ast_log(LOG_ERROR, "No format specified for sending RTP to '%s'\n", args.destination);
+               goto failure;
+       }
+
+       if (!ast_sockaddr_parse(&address, args.destination,
+                               PARSE_PORT_REQUIRE)) {
+               ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination);
+               goto failure;
+       }
+
+       caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+       if (!caps) {
+               goto failure;
+       }
+
+       ast_ouraddrfor(&address, &local_address);
+       if (!(instance = ast_rtp_instance_new(args.engine, NULL, &local_address, NULL))) {
+               ast_log(LOG_ERROR, "Could not create RTP instance for sending media to '%s'\n", args.destination);
+               goto failure;
+       }
+
+       if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids, requestor, 0, "UnicastRTP/%s-%p", args.destination, 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, &address);
+       ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(instance, 0));
+
+       ast_channel_tech_set(chan, &unicast_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);
+
+       pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS", ast_sockaddr_stringify_addr(&local_address));
+       ast_rtp_instance_get_local_address(instance, &local_address);
+       pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT", ast_sockaddr_stringify_port(&local_address));
+
+       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 unloaded */
+static int unload_module(void)
+{
+       ast_channel_unregister(&multicast_rtp_tech);
+       ao2_cleanup(multicast_rtp_tech.capabilities);
+       multicast_rtp_tech.capabilities = NULL;
+
+       ast_channel_unregister(&unicast_rtp_tech);
+       ao2_cleanup(unicast_rtp_tech.capabilities);
+       unicast_rtp_tech.capabilities = NULL;
+
+       return 0;
+}
+
+/*! \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");
+               unload_module();
+               return AST_MODULE_LOAD_DECLINE;
+       }
+
+       if (!(unicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
+               unload_module();
+               return AST_MODULE_LOAD_DECLINE;
+       }
+       ast_format_cap_append_by_type(unicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
+       if (ast_channel_register(&unicast_rtp_tech)) {
+               ast_log(LOG_ERROR, "Unable to register channel class 'UnicastRTP'\n");
+               unload_module();
+               return AST_MODULE_LOAD_DECLINE;
+       }
+
+       return AST_MODULE_LOAD_SUCCESS;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTP Media Channel",
+       .support_level = AST_MODULE_SUPPORT_CORE,
+       .load = load_module,
+       .unload = unload_module,
+       .load_pri = AST_MODPRI_CHANNEL_DRIVER,
+);