From dde58df3182b7f99714e446a684d8ddb81f759bb Mon Sep 17 00:00:00 2001 From: Richard Mudgett Date: Fri, 10 Jun 2016 16:13:04 -0500 Subject: chan_rtp.c: Copy file from chan_multicast_rtp.c Change-Id: I1119b53f2152ab1cbec74b5be7ea44844dbda8ef --- channels/chan_rtp.c | 223 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 223 insertions(+) create mode 100644 channels/chan_rtp.c (limited to 'channels') diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c new file mode 100644 index 000000000..267baabf1 --- /dev/null +++ b/channels/chan_rtp.c @@ -0,0 +1,223 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2009, Digium, Inc. + * + * Joshua Colp + * Andreas 'MacBrody' Brodmann + * + * 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 + * \author Andreas 'MacBrody' Broadmann + * + * \brief Multicast RTP Paging Channel + * + * \ingroup channel_drivers + */ + +/*** MODULEINFO + core + ***/ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include +#include + +#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, +); -- cgit v1.2.3