[svn-commits] file: trunk r423004 - in /trunk/channels: chan_multicast_rtp.c chan_rtp.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Sep 12 12:42:19 CDT 2014
Author: file
Date: Fri Sep 12 12:42:15 2014
New Revision: 423004
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=423004
Log:
chan_rtp: Add unicast RTP support.
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/
Added:
trunk/channels/chan_rtp.c (with props)
Removed:
trunk/channels/chan_multicast_rtp.c
Added: trunk/channels/chan_rtp.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_rtp.c?view=auto&rev=423004
==============================================================================
--- trunk/channels/chan_rtp.c (added)
+++ trunk/channels/chan_rtp.c Fri Sep 12 12:42:15 2014
@@ -1,0 +1,335 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2009 - 2014, 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 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,
+);
Propchange: trunk/channels/chan_rtp.c
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Propchange: trunk/channels/chan_rtp.c
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Propchange: trunk/channels/chan_rtp.c
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