mirror of
https://github.com/asterisk/asterisk.git
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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
This commit is contained in:
Joshua Colp
10
CHANGES
10
CHANGES
@@ -173,6 +173,16 @@ Calendaring for Asterisk
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iCalendar, CalDAV, and Exchange Server calendars are supported (Exchange support
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only tested on Exchange Server 2003 with no support for forms-based authentication).
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Multicast RTP Support
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---------------------
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* A new RTP engine and channel driver have been added which supports Multicast RTP.
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The channel driver can be used with the Page application to perform multicast RTP
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paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
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Type can be either basic or linksys.
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Destination is the IP address and port for the RTP packets.
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Control address is specific to the linksys type and is used for sending the control
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packets unique to them.
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
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------------------------------------------------------------------------------
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184
channels/chan_multicast_rtp.c
Normal file
184
channels/chan_multicast_rtp.c
Normal file
@@ -0,0 +1,184 @@
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/*
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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, int format, 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 const struct ast_channel_tech multicast_rtp_tech = {
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.type = "MulticastRTP",
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.description = tdesc,
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.capabilities = -1,
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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, int format, 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 sockaddr_in control_address = { .sin_family = AF_INET, }, destination_address = { .sin_family = AF_INET, };
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struct ast_channel *chan;
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int fmt = ast_best_codec(format);
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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 (ast_parse_arg(destination, PARSE_INADDR | PARSE_PORT_REQUIRE, &destination_address)) {
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goto failure;
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}
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if ((control = strchr(destination, '/'))) {
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*control++ = '\0';
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if (ast_parse_arg(control, PARSE_INADDR | PARSE_PORT_REQUIRE, &control_address)) {
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goto failure;
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}
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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, "", "", "", "", "", 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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chan->nativeformats = fmt;
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chan->writeformat = fmt;
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chan->readformat = fmt;
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chan->rawwriteformat = fmt;
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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 (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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return 0;
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multicast RTP Paging Channel");
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261
res/res_rtp_multicast.c
Normal file
261
res/res_rtp_multicast.c
Normal file
@@ -0,0 +1,261 @@
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/*
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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
|
||||
* channels for your use.
|
||||
*
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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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/*!
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* \file
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*
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* \brief Multicast RTP Engine
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*
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* \author Joshua Colp <jcolp@digium.com>
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* \author Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
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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 <sys/time.h>
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#include <signal.h>
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#include <fcntl.h>
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#include <math.h>
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#include "asterisk/pbx.h"
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#include "asterisk/frame.h"
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#include "asterisk/channel.h"
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#include "asterisk/acl.h"
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#include "asterisk/config.h"
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#include "asterisk/lock.h"
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#include "asterisk/utils.h"
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#include "asterisk/netsock.h"
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#include "asterisk/cli.h"
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#include "asterisk/manager.h"
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#include "asterisk/unaligned.h"
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#include "asterisk/module.h"
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#include "asterisk/rtp_engine.h"
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/*! Command value used for Linksys paging to indicate we are starting */
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#define LINKSYS_MCAST_STARTCMD 6
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/*! Command value used for Linksys paging to indicate we are stopping */
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#define LINKSYS_MCAST_STOPCMD 7
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/*! \brief Type of paging to do */
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enum multicast_type {
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/*! Simple multicast enabled client/receiver paging like Snom and Barix uses */
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MULTICAST_TYPE_BASIC = 0,
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/*! More advanced Linksys type paging which requires a start and stop packet */
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MULTICAST_TYPE_LINKSYS,
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};
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/*! \brief Structure for a Linksys control packet */
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struct multicast_control_packet {
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/*! Unique identifier for the control packet */
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uint32_t unique_id;
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/*! Actual command in the control packet */
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uint32_t command;
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/*! IP address for the RTP */
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uint32_t ip;
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/*! Port for the RTP */
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uint32_t port;
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};
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/*! \brief Structure for a multicast paging instance */
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struct multicast_rtp {
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/*! TYpe of multicast paging this instance is doing */
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enum multicast_type type;
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/*! Socket used for sending the audio on */
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int socket;
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/*! Synchronization source value, used when creating/sending the RTP packet */
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unsigned int ssrc;
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/*! Sequence number, used when creating/sending the RTP packet */
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unsigned int seqno;
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};
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/* Forward Declarations */
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static int multicast_rtp_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data);
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static int multicast_rtp_activate(struct ast_rtp_instance *instance);
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static int multicast_rtp_destroy(struct ast_rtp_instance *instance);
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||||
static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
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static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
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/* RTP Engine Declaration */
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static struct ast_rtp_engine multicast_rtp_engine = {
|
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.name = "multicast",
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.new = multicast_rtp_new,
|
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.activate = multicast_rtp_activate,
|
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.destroy = multicast_rtp_destroy,
|
||||
.write = multicast_rtp_write,
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.read = multicast_rtp_read,
|
||||
};
|
||||
|
||||
/*! \brief Function called to create a new multicast instance */
|
||||
static int multicast_rtp_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data)
|
||||
{
|
||||
struct multicast_rtp *multicast;
|
||||
const char *type = data;
|
||||
|
||||
if (!(multicast = ast_calloc(1, sizeof(*multicast)))) {
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (!strcasecmp(type, "basic")) {
|
||||
multicast->type = MULTICAST_TYPE_BASIC;
|
||||
} else if (!strcasecmp(type, "linksys")) {
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||||
multicast->type = MULTICAST_TYPE_LINKSYS;
|
||||
} else {
|
||||
ast_free(multicast);
|
||||
return -1;
|
||||
}
|
||||
|
||||
if ((multicast->socket = socket(AF_INET, SOCK_DGRAM, 0)) < 0) {
|
||||
ast_free(multicast);
|
||||
return -1;
|
||||
}
|
||||
|
||||
multicast->ssrc = ast_random();
|
||||
|
||||
ast_rtp_instance_set_data(instance, multicast);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/*! \brief Helper function which populates a control packet with useful information and sends it */
|
||||
static int multicast_send_control_packet(struct ast_rtp_instance *instance, struct multicast_rtp *multicast, int command)
|
||||
{
|
||||
struct multicast_control_packet control_packet = { .unique_id = htonl((u_long)time(NULL)),
|
||||
.command = htonl(command),
|
||||
};
|
||||
struct sockaddr_in control_address, remote_address;
|
||||
|
||||
ast_rtp_instance_get_local_address(instance, &control_address);
|
||||
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
||||
|
||||
/* Ensure the user of us have given us both the control address and destination address */
|
||||
if (!control_address.sin_addr.s_addr || !remote_address.sin_addr.s_addr) {
|
||||
return -1;
|
||||
}
|
||||
|
||||
control_packet.ip = remote_address.sin_addr.s_addr;
|
||||
control_packet.port = htonl(ntohs(remote_address.sin_port));
|
||||
|
||||
/* Based on a recommendation by Brian West who did the FreeSWITCH implementation we send control packets twice */
|
||||
sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, (struct sockaddr *)&control_address, sizeof(control_address));
|
||||
sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, (struct sockaddr *)&control_address, sizeof(control_address));
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/*! \brief Function called to indicate that audio is now going to flow */
|
||||
static int multicast_rtp_activate(struct ast_rtp_instance *instance)
|
||||
{
|
||||
struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
|
||||
|
||||
if (multicast->type != MULTICAST_TYPE_LINKSYS) {
|
||||
return 0;
|
||||
}
|
||||
|
||||
return multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STARTCMD);
|
||||
}
|
||||
|
||||
/*! \brief Function called to destroy a multicast instance */
|
||||
static int multicast_rtp_destroy(struct ast_rtp_instance *instance)
|
||||
{
|
||||
struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
|
||||
|
||||
if (multicast->type == MULTICAST_TYPE_LINKSYS) {
|
||||
multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STOPCMD);
|
||||
}
|
||||
|
||||
close(multicast->socket);
|
||||
|
||||
ast_free(multicast);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/*! \brief Function called to broadcast some audio on a multicast instance */
|
||||
static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
|
||||
{
|
||||
struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
|
||||
struct ast_frame *f = frame;
|
||||
struct sockaddr_in remote_address;
|
||||
int hdrlen = 12, res, codec;
|
||||
unsigned char *rtpheader;
|
||||
|
||||
/* We only accept audio, nothing else */
|
||||
if (frame->frametype != AST_FRAME_VOICE) {
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Grab the actual payload number for when we create the RTP packet */
|
||||
if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, frame->subclass)) < 0) {
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* If we do not have space to construct an RTP header duplicate the frame so we get some */
|
||||
if (frame->offset < hdrlen) {
|
||||
f = ast_frdup(frame);
|
||||
}
|
||||
|
||||
/* Construct an RTP header for our packet */
|
||||
rtpheader = (unsigned char *)(f->data.ptr - hdrlen);
|
||||
put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno++) | (0 << 23)));
|
||||
put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
|
||||
put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));
|
||||
|
||||
/* Finally send it out to the eager phones listening for us */
|
||||
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
||||
res = sendto(multicast->socket, (void *) rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *) &remote_address, sizeof(remote_address));
|
||||
|
||||
if (res < 0) {
|
||||
ast_log(LOG_ERROR, "Multicast RTP Transmission error to %s:%u: %s\n",
|
||||
ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port), strerror(errno));
|
||||
}
|
||||
|
||||
/* If we were forced to duplicate the frame free the new one */
|
||||
if (frame != f) {
|
||||
ast_frfree(f);
|
||||
}
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/*! \brief Function called to read from a multicast instance */
|
||||
static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
|
||||
{
|
||||
return &ast_null_frame;
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
if (ast_rtp_engine_register(&multicast_rtp_engine)) {
|
||||
return AST_MODULE_LOAD_DECLINE;
|
||||
}
|
||||
|
||||
return AST_MODULE_LOAD_SUCCESS;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
ast_rtp_engine_unregister(&multicast_rtp_engine);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multicast RTP Engine");
|
||||
Reference in New Issue
Block a user