asterisk/channels/chan_rtp.c

452 lines
13 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2009 - 2014, 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' Brodmann <andreas.brodmann@gmail.com>
*
* \brief RTP (Multicast and Unicast) Media Channel
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<depend>res_rtp_multicast</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/acl.h"
#include "asterisk/app.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/causes.h"
#include "asterisk/format_cache.h"
#include "asterisk/multicast_rtp.h"
#include "asterisk/dns_core.h"
/* 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 struct ast_channel *unicast_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 rtp_call(struct ast_channel *ast, const char *dest, int timeout);
static int rtp_hangup(struct ast_channel *ast);
static struct ast_frame *rtp_read(struct ast_channel *ast);
static int rtp_write(struct ast_channel *ast, struct ast_frame *f);
/* Multicast channel driver declaration */
static struct ast_channel_tech multicast_rtp_tech = {
.type = "MulticastRTP",
.description = "Multicast RTP Paging Channel Driver",
.requester = multicast_rtp_request,
.call = rtp_call,
.hangup = rtp_hangup,
.read = rtp_read,
.write = rtp_write,
};
/* Unicast channel driver declaration */
static struct ast_channel_tech unicast_rtp_tech = {
.type = "UnicastRTP",
.description = "Unicast RTP Media Channel Driver",
.requester = unicast_rtp_request,
.call = rtp_call,
.hangup = rtp_hangup,
.read = rtp_read,
.write = rtp_write,
};
/*! \brief Function called when we should read a frame from the channel */
static struct ast_frame *rtp_read(struct ast_channel *ast)
{
struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
int fdno = ast_channel_fdno(ast);
switch (fdno) {
case 0:
return ast_rtp_instance_read(instance, 0);
default:
return &ast_null_frame;
}
}
/*! \brief Function called when we should write a frame to the channel */
static int 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 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 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;
}
static struct ast_format *derive_format_from_cap(struct ast_format_cap *cap)
{
struct ast_format *fmt = ast_format_cap_get_format(cap, 0);
if (ast_format_cap_count(cap) == 1 && fmt == ast_format_slin) {
/*
* Because we have no SDP, we must use one of the static RTP payload
* assignments. Signed linear @ 8kHz does not map, so if that is our
* only capability, we force μ-law instead.
*/
fmt = ast_format_ulaw;
}
return fmt;
}
/*! \brief Function called when we should prepare to call the multicast 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 *parse;
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;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(type);
AST_APP_ARG(destination);
AST_APP_ARG(control);
AST_APP_ARG(options);
);
struct ast_multicast_rtp_options *mcast_options = NULL;
if (ast_strlen_zero(data)) {
ast_log(LOG_ERROR, "A multicast type and destination must be given to the 'MulticastRTP' channel\n");
goto failure;
}
parse = ast_strdupa(data);
AST_NONSTANDARD_APP_ARGS(args, parse, '/');
if (ast_strlen_zero(args.type)) {
ast_log(LOG_ERROR, "Type is required for the 'MulticastRTP' channel\n");
goto failure;
}
if (ast_strlen_zero(args.destination)) {
ast_log(LOG_ERROR, "Destination is required for the 'MulticastRTP' channel\n");
goto failure;
}
if (!ast_sockaddr_parse(&destination_address, args.destination, PARSE_PORT_REQUIRE)) {
ast_log(LOG_ERROR, "Destination address '%s' could not be parsed\n",
args.destination);
goto failure;
}
ast_sockaddr_setnull(&control_address);
if (!ast_strlen_zero(args.control)
&& !ast_sockaddr_parse(&control_address, args.control, PARSE_PORT_REQUIRE)) {
ast_log(LOG_ERROR, "Control address '%s' could not be parsed\n", args.control);
goto failure;
}
mcast_options = ast_multicast_rtp_create_options(args.type, args.options);
if (!mcast_options) {
goto failure;
}
fmt = ast_multicast_rtp_options_get_format(mcast_options);
if (!fmt) {
fmt = derive_format_from_cap(cap);
}
if (!fmt) {
ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
args.destination);
goto failure;
}
caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!caps) {
goto failure;
}
instance = ast_rtp_instance_new("multicast", NULL, &control_address, mcast_options);
if (!instance) {
ast_log(LOG_ERROR,
"Could not create '%s' multicast RTP instance for sending media to '%s'\n",
args.type, args.destination);
goto failure;
}
chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
requestor, 0, "MulticastRTP/%p", instance);
if (!chan) {
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);
ast_multicast_rtp_free_options(mcast_options);
return chan;
failure:
ao2_cleanup(fmt);
ao2_cleanup(caps);
ast_multicast_rtp_free_options(mcast_options);
*cause = AST_CAUSE_FAILURE;
return NULL;
}
enum {
OPT_RTP_CODEC = (1 << 0),
OPT_RTP_ENGINE = (1 << 1),
};
enum {
OPT_ARG_RTP_CODEC,
OPT_ARG_RTP_ENGINE,
/* note: this entry _MUST_ be the last one in the enum */
OPT_ARG_ARRAY_SIZE
};
AST_APP_OPTIONS(unicast_rtp_options, BEGIN_OPTIONS
/*! Set the codec to be used for unicast RTP */
AST_APP_OPTION_ARG('c', OPT_RTP_CODEC, OPT_ARG_RTP_CODEC),
/*! Set the RTP engine to use for unicast RTP */
AST_APP_OPTION_ARG('e', OPT_RTP_ENGINE, OPT_ARG_RTP_ENGINE),
END_OPTIONS );
/*! \brief Function called when we should prepare to call the unicast destination */
static struct ast_channel *unicast_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 *parse;
struct ast_rtp_instance *instance;
struct ast_sockaddr address;
struct ast_sockaddr local_address;
struct ast_channel *chan;
struct ast_format_cap *caps = NULL;
struct ast_format *fmt = NULL;
const char *engine_name;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(destination);
AST_APP_ARG(options);
);
struct ast_flags opts = { 0, };
char *opt_args[OPT_ARG_ARRAY_SIZE];
if (ast_strlen_zero(data)) {
ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
goto failure;
}
parse = ast_strdupa(data);
AST_NONSTANDARD_APP_ARGS(args, parse, '/');
if (ast_strlen_zero(args.destination)) {
ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
goto failure;
}
if (!ast_sockaddr_parse(&address, args.destination, PARSE_PORT_REQUIRE)) {
int rc;
char *host;
char *port;
rc = ast_sockaddr_split_hostport(args.destination, &host, &port, PARSE_PORT_REQUIRE);
if (!rc) {
ast_log(LOG_ERROR, "Unable to parse destination '%s' into host and port\n", args.destination);
goto failure;
}
rc = ast_dns_resolve_ipv6_and_ipv4(&address, host, port);
if (rc != 0) {
ast_log(LOG_ERROR, "Unable to resolve host '%s'\n", host);
goto failure;
}
}
if (!ast_strlen_zero(args.options)
&& ast_app_parse_options(unicast_rtp_options, &opts, opt_args,
ast_strdupa(args.options))) {
ast_log(LOG_ERROR, "'UnicastRTP' channel options '%s' parse error\n",
args.options);
goto failure;
}
if (ast_test_flag(&opts, OPT_RTP_CODEC)
&& !ast_strlen_zero(opt_args[OPT_ARG_RTP_CODEC])) {
fmt = ast_format_cache_get(opt_args[OPT_ARG_RTP_CODEC]);
if (!fmt) {
ast_log(LOG_ERROR, "Codec '%s' not found for sending RTP to '%s'\n",
opt_args[OPT_ARG_RTP_CODEC], args.destination);
goto failure;
}
} else {
fmt = derive_format_from_cap(cap);
if (!fmt) {
ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
args.destination);
goto failure;
}
}
caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!caps) {
goto failure;
}
engine_name = S_COR(ast_test_flag(&opts, OPT_RTP_ENGINE),
opt_args[OPT_ARG_RTP_ENGINE], "asterisk");
ast_sockaddr_copy(&local_address, &address);
if (ast_ouraddrfor(&address, &local_address)) {
ast_log(LOG_ERROR, "Could not get our address for sending media to '%s'\n",
args.destination);
goto failure;
}
instance = ast_rtp_instance_new(engine_name, NULL, &local_address, NULL);
if (!instance) {
ast_log(LOG_ERROR,
"Could not create %s RTP instance for sending media to '%s'\n",
S_OR(engine_name, "default"), args.destination);
goto failure;
}
chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
requestor, 0, "UnicastRTP/%s-%p", args.destination, instance);
if (!chan) {
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, &address);
ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(instance, 0));
ast_channel_tech_set(chan, &unicast_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);
pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
ast_sockaddr_stringify_addr(&local_address));
ast_rtp_instance_get_local_address(instance, &local_address);
pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT",
ast_sockaddr_stringify_port(&local_address));
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 unloaded */
static int unload_module(void)
{
ast_channel_unregister(&multicast_rtp_tech);
ao2_cleanup(multicast_rtp_tech.capabilities);
multicast_rtp_tech.capabilities = NULL;
ast_channel_unregister(&unicast_rtp_tech);
ao2_cleanup(unicast_rtp_tech.capabilities);
unicast_rtp_tech.capabilities = NULL;
return 0;
}
/*! \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");
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
if (!(unicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
ast_format_cap_append_by_type(unicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
if (ast_channel_register(&unicast_rtp_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class 'UnicastRTP'\n");
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
return AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTP Media Channel",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
.requires = "res_rtp_multicast",
);