asterisk/channels/chan_multicast_rtp.c

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/*
* 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 fmt;
ast_best_codec(cap, &fmt);
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;
}
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;
}
Refactor RTCP events over to Stasis; associate with channels This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
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_add(ast_channel_nativeformats(chan), &fmt);
ast_format_copy(ast_channel_writeformat(chan), &fmt);
ast_format_copy(ast_channel_rawwriteformat(chan), &fmt);
ast_format_copy(ast_channel_readformat(chan), &fmt);
ast_format_copy(ast_channel_rawreadformat(chan), &fmt);
ast_channel_tech_pvt_set(chan, instance);
ast_channel_unlock(chan);
return chan;
failure:
*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(0))) {
return AST_MODULE_LOAD_DECLINE;
}
ast_format_cap_add_all(multicast_rtp_tech.capabilities);
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);
multicast_rtp_tech.capabilities = ast_format_cap_destroy(multicast_rtp_tech.capabilities);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
);