chan_rtp.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2009 - 2014, Digium, Inc.
 *
 * Joshua Colp <[email protected]>
 * Andreas 'MacBrody' Brodmann <[email protected]>
 *
 * 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 <[email protected]>
 * \author Andreas 'MacBrody' Broadmann <[email protected]>
 *
 * \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",
);