This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
201 lines
5.7 KiB
C
201 lines
5.7 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2009, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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* Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \author Joshua Colp <jcolp@digium.com>
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* \author Andreas 'MacBrody' Broadmann <andreas.brodmann@gmail.com>
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*
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* \brief Multicast RTP Paging Channel
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*
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* \ingroup channel_drivers
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <fcntl.h>
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#include <sys/signal.h>
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#include "asterisk/lock.h"
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#include "asterisk/channel.h"
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#include "asterisk/config.h"
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#include "asterisk/module.h"
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#include "asterisk/pbx.h"
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#include "asterisk/sched.h"
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#include "asterisk/io.h"
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#include "asterisk/acl.h"
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#include "asterisk/callerid.h"
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#include "asterisk/file.h"
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#include "asterisk/cli.h"
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#include "asterisk/app.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/causes.h"
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static const char tdesc[] = "Multicast RTP Paging Channel Driver";
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/* Forward declarations */
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static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause);
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static int multicast_rtp_call(struct ast_channel *ast, char *dest, int timeout);
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static int multicast_rtp_hangup(struct ast_channel *ast);
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static struct ast_frame *multicast_rtp_read(struct ast_channel *ast);
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static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f);
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/* Channel driver declaration */
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static struct ast_channel_tech multicast_rtp_tech = {
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.type = "MulticastRTP",
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.description = tdesc,
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.requester = multicast_rtp_request,
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.call = multicast_rtp_call,
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.hangup = multicast_rtp_hangup,
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.read = multicast_rtp_read,
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.write = multicast_rtp_write,
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};
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/*! \brief Function called when we should read a frame from the channel */
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static struct ast_frame *multicast_rtp_read(struct ast_channel *ast)
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{
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return &ast_null_frame;
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}
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/*! \brief Function called when we should write a frame to the channel */
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static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f)
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{
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struct ast_rtp_instance *instance = ast->tech_pvt;
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return ast_rtp_instance_write(instance, f);
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}
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/*! \brief Function called when we should actually call the destination */
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static int multicast_rtp_call(struct ast_channel *ast, char *dest, int timeout)
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{
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struct ast_rtp_instance *instance = ast->tech_pvt;
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ast_queue_control(ast, AST_CONTROL_ANSWER);
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return ast_rtp_instance_activate(instance);
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}
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/*! \brief Function called when we should hang the channel up */
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static int multicast_rtp_hangup(struct ast_channel *ast)
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{
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struct ast_rtp_instance *instance = ast->tech_pvt;
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ast_rtp_instance_destroy(instance);
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ast->tech_pvt = NULL;
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return 0;
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}
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/*! \brief Function called when we should prepare to call the destination */
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static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause)
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{
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char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
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struct ast_rtp_instance *instance;
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struct ast_sockaddr control_address;
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struct ast_sockaddr destination_address;
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struct ast_channel *chan;
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struct ast_format fmt;
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ast_best_codec(cap, &fmt);
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ast_sockaddr_setnull(&control_address);
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/* If no type was given we can't do anything */
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if (ast_strlen_zero(multicast_type)) {
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goto failure;
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}
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if (!(destination = strchr(tmp, '/'))) {
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goto failure;
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}
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*destination++ = '\0';
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if ((control = strchr(destination, '/'))) {
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*control++ = '\0';
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if (!ast_sockaddr_parse(&control_address, control,
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PARSE_PORT_REQUIRE)) {
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goto failure;
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}
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}
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if (!ast_sockaddr_parse(&destination_address, destination,
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PARSE_PORT_REQUIRE)) {
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goto failure;
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}
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if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, multicast_type))) {
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goto failure;
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}
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if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", requestor ? requestor->linkedid : "", 0, "MulticastRTP/%p", instance))) {
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ast_rtp_instance_destroy(instance);
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goto failure;
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}
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ast_rtp_instance_set_remote_address(instance, &destination_address);
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chan->tech = &multicast_rtp_tech;
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ast_format_cap_add(chan->nativeformats, &fmt);
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ast_format_copy(&chan->writeformat, &fmt);
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ast_format_copy(&chan->rawwriteformat, &fmt);
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ast_format_copy(&chan->readformat, &fmt);
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ast_format_copy(&chan->rawreadformat, &fmt);
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chan->tech_pvt = instance;
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return chan;
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failure:
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*cause = AST_CAUSE_FAILURE;
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return NULL;
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}
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/*! \brief Function called when our module is loaded */
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static int load_module(void)
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{
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if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc())) {
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return AST_MODULE_LOAD_DECLINE;
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}
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ast_format_cap_add_all(multicast_rtp_tech.capabilities);
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if (ast_channel_register(&multicast_rtp_tech)) {
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ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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/*! \brief Function called when our module is unloaded */
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static int unload_module(void)
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{
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ast_channel_unregister(&multicast_rtp_tech);
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multicast_rtp_tech.capabilities = ast_format_cap_destroy(multicast_rtp_tech.capabilities);
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return 0;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel",
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_CHANNEL_DRIVER,
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);
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