RTP (Multicast and Unicast) Media Channel. More...

#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"

Include dependency graph for chan_rtp.c:

Go to the source code of this file.

Enumerations
enum	{ OPT_RTP_CODEC = (1 << 0) , OPT_RTP_ENGINE = (1 << 1) , OPT_RTP_GLUE = (1 << 2) }

enum	{ OPT_ARG_RTP_CODEC , OPT_ARG_RTP_ENGINE , OPT_ARG_ARRAY_SIZE }

Functions
static void	__reg_module (void)

static void	__unreg_module (void)

struct ast_module *	AST_MODULE_SELF_SYM (void)

static void	chan_rtp_get_codec (struct ast_channel chan, struct ast_format_cap result)
	Function called by RTP engine to get peer capabilities. More...

static enum ast_rtp_glue_result	chan_rtp_get_rtp_peer (struct ast_channel chan, struct ast_rtp_instance *instance)
	Function called by RTP engine to get local audio RTP peer. More...

static enum ast_rtp_glue_result	chan_rtp_get_vrtp_peer (struct ast_channel chan, struct ast_rtp_instance *instance)
	Function called by RTP engine to get local audio RTP peer. More...

static int	chan_rtp_set_rtp_peer (struct ast_channel chan, struct ast_rtp_instance rtp, struct ast_rtp_instance vrtp, struct ast_rtp_instance tpeer, const struct ast_format_cap *cap, int nat_active)
	Function called by RTP engine to change where the remote party should send media. More...

static struct ast_format *	derive_format_from_cap (struct ast_format_cap *cap)

static int	load_module (void)
	Function called when our module is loaded. More...

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)
	Function called when we should prepare to call the multicast destination. More...

static int	rtp_call (struct ast_channel ast, const char dest, int timeout)
	Function called when we should actually call the destination. More...

static int	rtp_hangup (struct ast_channel *ast)
	Function called when we should hang the channel up. More...

static struct ast_frame *	rtp_read (struct ast_channel *ast)
	Function called when we should read a frame from the channel. More...

static int	rtp_write (struct ast_channel ast, struct ast_frame f)
	Function called when we should write a frame to the channel. More...

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)
	Function called when we should prepare to call the unicast destination. More...

static int	unload_module (void)
	Function called when our module is unloaded. More...

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "RTP Media Channel" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_rtp_multicast", }

static const struct ast_module_info *	ast_module_info = &__mod_info

static const struct ast_datastore_info	chan_rtp_datastore_info

static struct ast_channel_tech	multicast_rtp_tech

static struct ast_rtp_glue	unicast_rtp_glue
	Local glue for interacting with the RTP engine core. More...

static const struct ast_app_option	unicast_rtp_options [128] = { [ 'c' ] = { .flag = OPT_RTP_CODEC , .arg_index = OPT_ARG_RTP_CODEC + 1 }, [ 'e' ] = { .flag = OPT_RTP_ENGINE , .arg_index = OPT_ARG_RTP_ENGINE + 1 }, [ 'g' ] = { .flag = OPT_RTP_GLUE }, }

static struct ast_channel_tech	unicast_rtp_tech

Detailed Description

RTP (Multicast and Unicast) Media Channel.

Author: Joshua Colp jcolp.nosp@m.@dig.nosp@m.ium.c.nosp@m.om; Andreas 'MacBrody' Brodmann andre.nosp@m.as.b.nosp@m.rodma.nosp@m.nn@g.nosp@m.mail..nosp@m.com

Definition in file chan_rtp.c.

Enumeration Type Documentation

◆ anonymous enum

anonymous enum

Enumerator
OPT_RTP_CODEC
OPT_RTP_ENGINE
OPT_RTP_GLUE

Definition at line 250 of file chan_rtp.c.

     {
    OPT_RTP_CODEC =  (1 << 0),
    OPT_RTP_ENGINE = (1 << 1),
    OPT_RTP_GLUE = (1 << 2),
};

◆ anonymous enum

anonymous enum

Enumerator
OPT_ARG_RTP_CODEC
OPT_ARG_RTP_ENGINE
OPT_ARG_ARRAY_SIZE

Definition at line 256 of file chan_rtp.c.

     {
    OPT_ARG_RTP_CODEC,
    OPT_ARG_RTP_ENGINE,
    /* note: this entry _MUST_ be the last one in the enum */
    OPT_ARG_ARRAY_SIZE
};

Function Documentation

◆ __reg_module()

static void __reg_module ( void )

static

Definition at line 526 of file chan_rtp.c.

◆ __unreg_module()

static void __unreg_module ( void )

static

Definition at line 526 of file chan_rtp.c.

◆ AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM ( void )

Definition at line 526 of file chan_rtp.c.

◆ chan_rtp_get_codec()

static void chan_rtp_get_codec	(	struct ast_channel *	chan,
		struct ast_format_cap *	result
	)

static

Function called by RTP engine to get peer capabilities.

Definition at line 420 of file chan_rtp.c.

{
    SCOPE_ENTER(1, "%s Native formats %s\n", ast_channel_name(chan),
        ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(ast_channel_nativeformats(chan), &STR_TMP)));
    ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
    SCOPE_EXIT_RTN();
}

References ast_channel_name(), ast_channel_nativeformats(), ast_format_cap_append_from_cap(), ast_format_cap_get_names(), AST_FORMAT_CAP_NAMES_LEN, AST_MEDIA_TYPE_UNKNOWN, ast_str_tmp, result, SCOPE_ENTER, and SCOPE_EXIT_RTN.

◆ chan_rtp_get_rtp_peer()

static enum ast_rtp_glue_result chan_rtp_get_rtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp_instance **	instance
	)

static

Function called by RTP engine to get local audio RTP peer.

Definition at line 444 of file chan_rtp.c.

{
    struct ast_rtp_instance *rtp_instance = ast_channel_tech_pvt(chan);
    struct ast_datastore *datastore;
 
    if (!rtp_instance) {
        return AST_RTP_GLUE_RESULT_FORBID;
    }
 
    if ((datastore = ast_channel_datastore_find(chan, &chan_rtp_datastore_info, NULL))) {
        ao2_ref(datastore, -1);
 
        *instance = rtp_instance;
        ao2_ref(*instance, +1);
 
        return AST_RTP_GLUE_RESULT_LOCAL;
    }
 
    return AST_RTP_GLUE_RESULT_FORBID;
}

References ao2_ref, ast_channel_datastore_find(), ast_channel_tech_pvt(), AST_RTP_GLUE_RESULT_FORBID, AST_RTP_GLUE_RESULT_LOCAL, chan_rtp_datastore_info, and NULL.

◆ chan_rtp_get_vrtp_peer()

static enum ast_rtp_glue_result chan_rtp_get_vrtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp_instance **	instance
	)

static

Function called by RTP engine to get local audio RTP peer.

Definition at line 438 of file chan_rtp.c.

{
    return AST_RTP_GLUE_RESULT_FORBID;
}

References AST_RTP_GLUE_RESULT_FORBID.

◆ chan_rtp_set_rtp_peer()

static int chan_rtp_set_rtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp_instance *	rtp,
		struct ast_rtp_instance *	vrtp,
		struct ast_rtp_instance *	tpeer,
		const struct ast_format_cap *	cap,
		int	nat_active
	)

static

Function called by RTP engine to change where the remote party should send media.

chan_rtp is not able to actually update the peer, so this function has no effect.

Definition at line 432 of file chan_rtp.c.

{
    return -1;
}

◆ derive_format_from_cap()

static struct ast_format * derive_format_from_cap ( struct ast_format_cap * cap )

static

Definition at line 122 of file chan_rtp.c.

{
    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.
         */
        ao2_ref(fmt, -1);
        fmt = ao2_bump(ast_format_ulaw);
    }
 
    return fmt;
}

References ao2_bump, ao2_ref, ast_format_cap_count(), ast_format_cap_get_format(), ast_format_slin, and ast_format_ulaw.

Referenced by multicast_rtp_request(), and unicast_rtp_request().

◆ load_module()

static int load_module ( void )

static

Function called when our module is loaded.

Definition at line 491 of file chan_rtp.c.

{
    if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
        return AST_MODULE_LOAD_DECLINE;
    }
 
    ast_rtp_glue_register(&unicast_rtp_glue);
 
    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;
}

References ast_channel_register(), ast_format_cap_alloc, ast_format_cap_append_by_type(), AST_FORMAT_CAP_FLAG_DEFAULT, ast_log, AST_MEDIA_TYPE_UNKNOWN, AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_SUCCESS, ast_rtp_glue_register, ast_channel_tech::capabilities, LOG_ERROR, multicast_rtp_tech, unicast_rtp_glue, unicast_rtp_tech, and unload_module().

◆ multicast_rtp_request()

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

Function called when we should prepare to call the multicast destination.

Definition at line 140 of file chan_rtp.c.

{
    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/%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, &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;
}

References ao2_cleanup, ao2_ref, args, AST_APP_ARG, AST_CAUSE_FAILURE, ast_channel_alloc, ast_channel_nativeformats_set(), ast_channel_set_rawreadformat(), ast_channel_set_rawwriteformat(), ast_channel_set_readformat(), ast_channel_set_writeformat(), ast_channel_tech_pvt_set(), ast_channel_tech_set(), ast_channel_uniqueid(), ast_channel_unlock, AST_DECLARE_APP_ARGS, ast_format_cap_alloc, ast_format_cap_append, AST_FORMAT_CAP_FLAG_DEFAULT, ast_log, ast_multicast_rtp_create_options(), ast_multicast_rtp_free_options(), ast_multicast_rtp_options_get_format(), AST_NONSTANDARD_APP_ARGS, ast_rtp_instance_destroy(), ast_rtp_instance_new(), ast_rtp_instance_set_channel_id(), ast_rtp_instance_set_remote_address, ast_sockaddr_parse(), ast_sockaddr_setnull(), AST_STATE_DOWN, ast_strdupa, ast_strlen_zero(), derive_format_from_cap(), ast_multicast_rtp_options::fmt, LOG_ERROR, multicast_rtp_tech, NULL, options, PARSE_PORT_REQUIRE, and type.

◆ rtp_call()

static int rtp_call	(	struct ast_channel *	ast,
		const char *	dest,
		int	timeout
	)

static

Function called when we should actually call the destination.

Definition at line 101 of file chan_rtp.c.

{
    struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
 
    ast_queue_control(ast, AST_CONTROL_ANSWER);
 
    return ast_rtp_instance_activate(instance);
}

References ast_channel_tech_pvt(), AST_CONTROL_ANSWER, ast_queue_control(), and ast_rtp_instance_activate().

◆ rtp_hangup()

static int rtp_hangup ( struct ast_channel * ast )

static

Function called when we should hang the channel up.

Definition at line 111 of file chan_rtp.c.

{
    struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
 
    ast_rtp_instance_destroy(instance);
 
    ast_channel_tech_pvt_set(ast, NULL);
 
    return 0;
}

References ast_channel_tech_pvt(), ast_channel_tech_pvt_set(), ast_rtp_instance_destroy(), and NULL.

◆ rtp_read()

static struct ast_frame * rtp_read ( struct ast_channel * ast )

static

Function called when we should read a frame from the channel.

Definition at line 79 of file chan_rtp.c.

{
    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;
    }
}

References ast_channel_fdno(), ast_channel_tech_pvt(), ast_null_frame, and ast_rtp_instance_read().

◆ rtp_write()

static int rtp_write	(	struct ast_channel *	ast,
		struct ast_frame *	f
	)

static

Function called when we should write a frame to the channel.

Definition at line 93 of file chan_rtp.c.

{
    struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
 
    return ast_rtp_instance_write(instance, f);
}

References ast_channel_tech_pvt(), and ast_rtp_instance_write().

◆ unicast_rtp_request()

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

Function called when we should prepare to call the unicast destination.

Definition at line 277 of file chan_rtp.c.

{
    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);
 
    if (ast_test_flag(&opts, OPT_RTP_GLUE)) {
        struct ast_datastore *datastore;
        if ((datastore = ast_datastore_alloc(&chan_rtp_datastore_info, NULL))) {
            ast_channel_datastore_add(chan, datastore);
        }
    }
 
    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_rtp_instance_get_local_address(instance, &local_address);
    pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
        ast_sockaddr_stringify_addr(&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;
}

References ao2_cleanup, ao2_ref, args, AST_APP_ARG, ast_app_parse_options(), AST_CAUSE_FAILURE, ast_channel_alloc, ast_channel_datastore_add(), ast_channel_nativeformats_set(), ast_channel_set_fd(), ast_channel_set_rawreadformat(), ast_channel_set_rawwriteformat(), ast_channel_set_readformat(), ast_channel_set_writeformat(), ast_channel_tech_pvt_set(), ast_channel_tech_set(), ast_channel_uniqueid(), ast_channel_unlock, ast_datastore_alloc, AST_DECLARE_APP_ARGS, ast_dns_resolve_ipv6_and_ipv4(), ast_format_cache_get, ast_format_cap_alloc, ast_format_cap_append, AST_FORMAT_CAP_FLAG_DEFAULT, ast_log, AST_NONSTANDARD_APP_ARGS, ast_ouraddrfor(), ast_rtp_instance_destroy(), ast_rtp_instance_fd(), ast_rtp_instance_get_local_address(), ast_rtp_instance_new(), ast_rtp_instance_set_channel_id(), ast_rtp_instance_set_remote_address, ast_sockaddr_copy(), ast_sockaddr_parse(), ast_sockaddr_split_hostport(), ast_sockaddr_stringify_addr(), ast_sockaddr_stringify_port(), AST_STATE_DOWN, ast_strdupa, ast_strlen_zero(), ast_test_flag, chan_rtp_datastore_info, derive_format_from_cap(), LOG_ERROR, NULL, OPT_ARG_ARRAY_SIZE, OPT_ARG_RTP_CODEC, OPT_ARG_RTP_ENGINE, OPT_RTP_CODEC, OPT_RTP_ENGINE, OPT_RTP_GLUE, options, PARSE_PORT_REQUIRE, pbx_builtin_setvar_helper(), S_COR, S_OR, unicast_rtp_options, and unicast_rtp_tech.

◆ unload_module()

static int unload_module ( void )

static

Function called when our module is unloaded.

Definition at line 475 of file chan_rtp.c.

{
    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;
 
    ast_rtp_glue_unregister(&unicast_rtp_glue);
 
    return 0;
}

References ao2_cleanup, ast_channel_unregister(), ast_rtp_glue_unregister(), ast_channel_tech::capabilities, multicast_rtp_tech, NULL, unicast_rtp_glue, and unicast_rtp_tech.

Referenced by load_module().

Variable Documentation

◆ __mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "RTP Media Channel" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_rtp_multicast", }

static

Definition at line 526 of file chan_rtp.c.

◆ ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 526 of file chan_rtp.c.

◆ chan_rtp_datastore_info

const struct ast_datastore_info chan_rtp_datastore_info

static

Initial value:

= {
    .type = "CHAN_RTP_GLUE",
}

Definition at line 272 of file chan_rtp.c.

Referenced by chan_rtp_get_rtp_peer(), and unicast_rtp_request().

◆ multicast_rtp_tech

struct ast_channel_tech multicast_rtp_tech

static

Definition at line 57 of file chan_rtp.c.

Referenced by load_module(), multicast_rtp_request(), and unload_module().

◆ unicast_rtp_glue

struct ast_rtp_glue unicast_rtp_glue

static

Local glue for interacting with the RTP engine core.

Definition at line 466 of file chan_rtp.c.

Referenced by load_module(), and unload_module().

◆ unicast_rtp_options

const struct ast_app_option unicast_rtp_options[128] = { [ 'c' ] = { .flag = OPT_RTP_CODEC , .arg_index = OPT_ARG_RTP_CODEC + 1 }, [ 'e' ] = { .flag = OPT_RTP_ENGINE , .arg_index = OPT_ARG_RTP_ENGINE + 1 }, [ 'g' ] = { .flag = OPT_RTP_GLUE }, }

static

Set the codec to be used for unicast RTP

Set the RTP engine to use for unicast RTP

Provide RTP glue for the channel

Definition at line 270 of file chan_rtp.c.

Referenced by unicast_rtp_request().

◆ unicast_rtp_tech

struct ast_channel_tech unicast_rtp_tech

static

Definition at line 68 of file chan_rtp.c.

Referenced by load_module(), unicast_rtp_request(), and unload_module().

Enumerations

Functions

Variables

Detailed Description

Enumeration Type Documentation

◆ anonymous enum

◆ anonymous enum

Function Documentation

◆ __reg_module()

◆ __unreg_module()

◆ AST_MODULE_SELF_SYM()

◆ chan_rtp_get_codec()

◆ chan_rtp_get_rtp_peer()

◆ chan_rtp_get_vrtp_peer()

◆ chan_rtp_set_rtp_peer()

◆ derive_format_from_cap()

◆ load_module()

◆ multicast_rtp_request()

◆ rtp_call()

◆ rtp_hangup()

◆ rtp_read()

◆ rtp_write()

◆ unicast_rtp_request()

◆ unload_module()

Variable Documentation

◆ __mod_info

◆ ast_module_info

◆ chan_rtp_datastore_info

◆ multicast_rtp_tech

◆ unicast_rtp_glue

◆ unicast_rtp_options

◆ unicast_rtp_tech