chan_rtp.c

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/*
 * 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.
         */
        ao2_ref(fmt, -1);
        fmt = ao2_bump(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/%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;
}
 
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,
    /* 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),
    /*! Provide RTP glue for the channel */
    AST_APP_OPTION('g', OPT_RTP_GLUE),
END_OPTIONS );
 
static const struct ast_datastore_info chan_rtp_datastore_info = {
    .type = "CHAN_RTP_GLUE",
};
 
/*! \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);
 
    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;
}
 
/*! \brief Function called by RTP engine to get peer capabilities */
static void chan_rtp_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
{
    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();
}
 
/*! \brief 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.
 * */
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)
{
    return -1;
}
 
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result chan_rtp_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
    return AST_RTP_GLUE_RESULT_FORBID;
}
 
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result chan_rtp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
    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;
}
 
/*! \brief Local glue for interacting with the RTP engine core */
static struct ast_rtp_glue unicast_rtp_glue = {
    .type = "UnicastRTP",
    .get_rtp_info = chan_rtp_get_rtp_peer,
    .get_vrtp_info = chan_rtp_get_vrtp_peer,
    .get_codec = chan_rtp_get_codec,
    .update_peer = chan_rtp_set_rtp_peer,
};
 
/*! \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;
 
    ast_rtp_glue_unregister(&unicast_rtp_glue);
 
    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_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;
}
 
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",
);