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feat: AudioSocket channel, application, and ARI support.
This commit adds support for [AudioSocket]( https://wiki.asterisk.org/wiki/display/AST/AudioSocket), a very simple bidirectional audio streaming protocol. There are both channel and application interfaces. A description of the protocol can be found on the above referenced GitHub page. A short talk about the reasons and implementation can be found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from CommCon 2019. ARI support has also been added via the existing "externalMedia" ARI functionality. The UUID is specified using the arbitrary "data" field. ASTERISK-28484 #close Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5
This commit is contained in:
Seán C McCord
Ben Ford
302
channels/chan_audiosocket.c
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302
channels/chan_audiosocket.c
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2019, CyCore Systems, Inc
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*
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* Seán C McCord <scm@cycoresys.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 Seán C McCord <scm@cycoresys.com>
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*
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* \brief AudioSocket Channel
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*
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* \ingroup channel_drivers
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*/
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/*** MODULEINFO
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<depend>res_audiosocket</depend>
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<support_level>extended</support_level>
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***/
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#include "asterisk.h"
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#include <uuid/uuid.h>
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#include "asterisk/channel.h"
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#include "asterisk/module.h"
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#include "asterisk/res_audiosocket.h"
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#include "asterisk/pbx.h"
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#include "asterisk/acl.h"
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#include "asterisk/app.h"
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#include "asterisk/causes.h"
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#include "asterisk/format_cache.h"
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#define FD_OUTPUT 1 /* A fd of -1 means an error, 0 is stdin */
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struct audiosocket_instance {
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int svc; /* The file descriptor for the AudioSocket instance */
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char id[38]; /* The UUID identifying this AudioSocket instance */
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} audiosocket_instance;
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/* Forward declarations */
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static struct ast_channel *audiosocket_request(const char *type,
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struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids,
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const struct ast_channel *requestor, const char *data, int *cause);
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static int audiosocket_call(struct ast_channel *ast, const char *dest, int timeout);
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static int audiosocket_hangup(struct ast_channel *ast);
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static struct ast_frame *audiosocket_read(struct ast_channel *ast);
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static int audiosocket_write(struct ast_channel *ast, struct ast_frame *f);
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/* AudioSocket channel driver declaration */
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static struct ast_channel_tech audiosocket_channel_tech = {
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.type = "AudioSocket",
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.description = "AudioSocket Channel Driver",
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.requester = audiosocket_request,
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.call = audiosocket_call,
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.hangup = audiosocket_hangup,
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.read = audiosocket_read,
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.write = audiosocket_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 *audiosocket_read(struct ast_channel *ast)
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{
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struct audiosocket_instance *instance;
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/* The channel should always be present from the API */
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instance = ast_channel_tech_pvt(ast);
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if (instance == NULL || instance->svc < FD_OUTPUT) {
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return NULL;
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}
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return ast_audiosocket_receive_frame(instance->svc);
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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 audiosocket_write(struct ast_channel *ast, struct ast_frame *f)
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{
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struct audiosocket_instance *instance;
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/* The channel should always be present from the API */
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instance = ast_channel_tech_pvt(ast);
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if (instance == NULL || instance->svc < 1) {
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return -1;
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}
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return ast_audiosocket_send_frame(instance->svc, 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 audiosocket_call(struct ast_channel *ast, const char *dest, int timeout)
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{
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struct audiosocket_instance *instance = ast_channel_tech_pvt(ast);
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ast_queue_control(ast, AST_CONTROL_ANSWER);
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return ast_audiosocket_init(instance->svc, instance->id);
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}
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/*! \brief Function called when we should hang the channel up */
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static int audiosocket_hangup(struct ast_channel *ast)
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{
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struct audiosocket_instance *instance;
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/* The channel should always be present from the API */
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instance = ast_channel_tech_pvt(ast);
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if (instance != NULL && instance->svc > 0) {
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close(instance->svc);
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}
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ast_channel_tech_pvt_set(ast, NULL);
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ast_free(instance);
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return 0;
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}
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enum {
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OPT_AUDIOSOCKET_CODEC = (1 << 0),
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};
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enum {
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OPT_ARG_AUDIOSOCKET_CODEC = (1 << 0),
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OPT_ARG_ARRAY_SIZE
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};
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AST_APP_OPTIONS(audiosocket_options, BEGIN_OPTIONS
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AST_APP_OPTION_ARG('c', OPT_AUDIOSOCKET_CODEC, OPT_ARG_AUDIOSOCKET_CODEC),
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END_OPTIONS );
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/*! \brief Function called when we should prepare to call the unicast destination */
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static struct ast_channel *audiosocket_request(const char *type,
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struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids,
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const struct ast_channel *requestor, const char *data, int *cause)
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{
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char *parse;
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struct audiosocket_instance *instance = NULL;
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struct ast_sockaddr address;
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struct ast_channel *chan;
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struct ast_format_cap *caps = NULL;
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struct ast_format *fmt = NULL;
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uuid_t uu;
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int fd;
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(destination);
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AST_APP_ARG(idStr);
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AST_APP_ARG(options);
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);
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struct ast_flags opts = { 0, };
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char *opt_args[OPT_ARG_ARRAY_SIZE];
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if (ast_strlen_zero(data)) {
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ast_log(LOG_ERROR, "Destination is required for the 'AudioSocket' channel\n");
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goto failure;
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}
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parse = ast_strdupa(data);
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AST_NONSTANDARD_APP_ARGS(args, parse, '/');
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if (ast_strlen_zero(args.destination)) {
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ast_log(LOG_ERROR, "Destination is required for the 'AudioSocket' channel\n");
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goto failure;
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}
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if (ast_sockaddr_resolve_first_af
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(&address, args.destination, PARSE_PORT_REQUIRE, AST_AF_UNSPEC)) {
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ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination);
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goto failure;
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}
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if (ast_strlen_zero(args.idStr)) {
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ast_log(LOG_ERROR, "UUID is required for the 'AudioSocket' channel\n");
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goto failure;
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}
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if (uuid_parse(args.idStr, uu)) {
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ast_log(LOG_ERROR, "Failed to parse UUID '%s'\n", args.idStr);
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goto failure;
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}
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if (!ast_strlen_zero(args.options)
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&& ast_app_parse_options(audiosocket_options, &opts, opt_args,
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ast_strdupa(args.options))) {
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ast_log(LOG_ERROR, "'AudioSocket' channel options '%s' parse error\n",
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args.options);
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goto failure;
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}
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if (ast_test_flag(&opts, OPT_AUDIOSOCKET_CODEC)
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&& !ast_strlen_zero(opt_args[OPT_ARG_AUDIOSOCKET_CODEC])) {
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fmt = ast_format_cache_get(opt_args[OPT_ARG_AUDIOSOCKET_CODEC]);
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if (!fmt) {
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ast_log(LOG_ERROR, "Codec '%s' not found for AudioSocket connection to '%s'\n",
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opt_args[OPT_ARG_AUDIOSOCKET_CODEC], args.destination);
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goto failure;
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}
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} else {
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fmt = ast_format_cap_get_format(cap, 0);
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if (!fmt) {
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ast_log(LOG_ERROR, "No codec available for AudioSocket connection to '%s'\n",
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args.destination);
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goto failure;
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}
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}
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caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
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if (!caps) {
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goto failure;
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}
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instance = ast_calloc(1, sizeof(*instance));
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if (!instance) {
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ast_log(LOG_ERROR, "Failed to allocate AudioSocket channel pvt\n");
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goto failure;
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}
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ast_copy_string(instance->id, args.idStr, sizeof(instance->id));
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if ((fd = ast_audiosocket_connect(args.destination, NULL)) < 0) {
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goto failure;
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}
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instance->svc = fd;
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chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
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requestor, 0, "AudioSocket/%s-%s", args.destination, args.idStr);
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if (!chan) {
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goto failure;
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}
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ast_channel_set_fd(chan, 0, fd);
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ast_channel_tech_set(chan, &audiosocket_channel_tech);
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ast_format_cap_append(caps, fmt, 0);
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ast_channel_nativeformats_set(chan, caps);
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ast_channel_set_writeformat(chan, fmt);
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ast_channel_set_rawwriteformat(chan, fmt);
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ast_channel_set_readformat(chan, fmt);
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ast_channel_set_rawreadformat(chan, fmt);
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ast_channel_tech_pvt_set(chan, instance);
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pbx_builtin_setvar_helper(chan, "AUDIOSOCKET_UUID", args.idStr);
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pbx_builtin_setvar_helper(chan, "AUDIOSOCKET_SERVICE", args.destination);
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ast_channel_unlock(chan);
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ao2_ref(fmt, -1);
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ao2_ref(caps, -1);
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return chan;
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failure:
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*cause = AST_CAUSE_FAILURE;
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ao2_cleanup(fmt);
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ao2_cleanup(caps);
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if (instance != NULL) {
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ast_free(instance);
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if (fd >= 0) {
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close(fd);
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}
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}
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return NULL;
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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(&audiosocket_channel_tech);
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ao2_cleanup(audiosocket_channel_tech.capabilities);
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audiosocket_channel_tech.capabilities = NULL;
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return 0;
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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 (!(audiosocket_channel_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
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return AST_MODULE_LOAD_DECLINE;
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}
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ast_format_cap_append_by_type(audiosocket_channel_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
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if (ast_channel_register(&audiosocket_channel_tech)) {
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ast_log(LOG_ERROR, "Unable to register channel class AudioSocket");
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ao2_ref(audiosocket_channel_tech.capabilities, -1);
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audiosocket_channel_tech.capabilities = NULL;
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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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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER,
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"AudioSocket Channel",
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.support_level = AST_MODULE_SUPPORT_EXTENDED,
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_CHANNEL_DRIVER,
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.requires = "res_audiosocket",
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);
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