diff options
author | Joshua Colp <jcolp@digium.com> | 2009-06-25 18:25:24 +0000 |
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committer | Joshua Colp <jcolp@digium.com> | 2009-06-25 18:25:24 +0000 |
commit | ae87ba45b544cd5ee56211d8fcba2f705d445c49 (patch) | |
tree | 7c1c2a94fa367bd30f955ba44d4bdc41eea2a53c /channels/chan_multicast_rtp.c | |
parent | ca3a181c33cdf7155ffc34a4c8ac79f09b684a13 (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.c | 184 |
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"); |