res_audiosocket.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2019, CyCore Systems, Inc
 *
 * Seán C McCord <scm@cycoresys.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
 *
 * \brief AudioSocket support for Asterisk
 *
 * \author Seán C McCord <scm@cycoresys.com>
 *
 */
 
/*** MODULEINFO
    <support_level>extended</support_level>
 ***/
 
#include "asterisk.h"
#include "errno.h"
#include <uuid/uuid.h>
 
#include "asterisk/file.h"
#include "asterisk/res_audiosocket.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/uuid.h"
#include "asterisk/format_cache.h"
 
#define MODULE_DESCRIPTION  "AudioSocket support functions for Asterisk"
 
#define MAX_CONNECT_TIMEOUT_MSEC 2000
 
/*!
 * \internal
 * \brief Attempt to complete the audiosocket connection.
 *
 * \param server Url that we are trying to connect to.
 * \param addr Address that host was resolved to.
 * \param netsockfd File descriptor of socket.
 *
 * \retval 0 when connection is succesful.
 * \retval 1 when there is an error.
 */
static int handle_audiosocket_connection(const char *server,
    const struct ast_sockaddr addr, const int netsockfd)
{
    struct pollfd pfds[1];
    int res, conresult;
    socklen_t reslen;
 
    reslen = sizeof(conresult);
 
    pfds[0].fd = netsockfd;
    pfds[0].events = POLLOUT;
 
    while ((res = ast_poll(pfds, 1, MAX_CONNECT_TIMEOUT_MSEC)) != 1) {
        if (errno != EINTR) {
            if (!res) {
                ast_log(LOG_WARNING, "AudioSocket connection to '%s' timed"
                    "out after MAX_CONNECT_TIMEOUT_MSEC (%d) milliseconds.\n",
                    server, MAX_CONNECT_TIMEOUT_MSEC);
            } else {
                ast_log(LOG_WARNING, "Connect to '%s' failed: %s\n", server,
                    strerror(errno));
            }
 
            return -1;
        }
    }
 
    if (getsockopt(pfds[0].fd, SOL_SOCKET, SO_ERROR, &conresult, &reslen) < 0) {
        ast_log(LOG_WARNING, "Connection to %s failed with error: %s\n",
            ast_sockaddr_stringify(&addr), strerror(errno));
        return -1;
    }
 
    if (conresult) {
        ast_log(LOG_WARNING, "Connecting to '%s' failed for url '%s': %s\n",
            ast_sockaddr_stringify(&addr), server, strerror(conresult));
        return -1;
    }
 
    return 0;
}
 
const int ast_audiosocket_connect(const char *server, struct ast_channel *chan)
{
    int s = -1;
    struct ast_sockaddr *addrs = NULL;
    int num_addrs = 0, i = 0;
 
    if (chan && ast_autoservice_start(chan) < 0) {
        ast_log(LOG_WARNING, "Failed to start autoservice for channel "
            "%s\n", ast_channel_name(chan));
        goto end;
    }
 
    if (ast_strlen_zero(server)) {
        ast_log(LOG_ERROR, "No AudioSocket server provided\n");
        goto end;
    }
 
    if (!(num_addrs = ast_sockaddr_resolve(&addrs, server, PARSE_PORT_REQUIRE,
        AST_AF_UNSPEC))) {
        ast_log(LOG_ERROR, "Failed to resolve AudioSocket service using %s - "
            "requires a valid hostname and port\n", server);
        goto end;
    }
 
    /* Connect to AudioSocket service */
    for (i = 0; i < num_addrs; i++) {
 
        if (!ast_sockaddr_port(&addrs[i])) {
            /* If there's no port, other addresses should have the
             * same problem. Stop here.
             */
            ast_log(LOG_ERROR, "No port provided for %s\n",
                ast_sockaddr_stringify(&addrs[i]));
            s = -1;
            goto end;
        }
 
        if ((s = ast_socket_nonblock(addrs[i].ss.ss_family, SOCK_STREAM,
            IPPROTO_TCP)) < 0) {
            ast_log(LOG_WARNING, "Unable to create socket: %s\n", strerror(errno));
            continue;
        }
 
        if (ast_connect(s, &addrs[i]) && errno == EINPROGRESS) {
 
            if (handle_audiosocket_connection(server, addrs[i], s)) {
                close(s);
                continue;
            }
 
        } else {
            ast_log(LOG_ERROR, "Connection to %s failed with unexpected error: %s\n",
                ast_sockaddr_stringify(&addrs[i]), strerror(errno));
            close(s);
            s = -1;
        }
 
        break;
    }
 
end:
    if (addrs) {
        ast_free(addrs);
    }
 
    if (chan && ast_autoservice_stop(chan) < 0) {
        ast_log(LOG_WARNING, "Failed to stop autoservice for channel %s\n",
        ast_channel_name(chan));
        close(s);
        return -1;
    }
 
    if (i == num_addrs) {
        ast_log(LOG_ERROR, "Failed to connect to AudioSocket service\n");
        close(s);
        return -1;
    }
 
    return s;
}
 
const int ast_audiosocket_init(const int svc, const char *id)
{
    uuid_t uu;
    int ret = 0;
    uint8_t buf[3 + 16];
 
    if (ast_strlen_zero(id)) {
        ast_log(LOG_ERROR, "No UUID for AudioSocket\n");
        return -1;
    }
 
    if (uuid_parse(id, uu)) {
        ast_log(LOG_ERROR, "Failed to parse UUID '%s'\n", id);
        return -1;
    }
 
    buf[0] = 0x01;
    buf[1] = 0x00;
    buf[2] = 0x10;
    memcpy(buf + 3, uu, 16);
 
    if (write(svc, buf, 3 + 16) != 3 + 16) {
        ast_log(LOG_WARNING, "Failed to write data to AudioSocket\n");
        ret = -1;
    }
 
    return ret;
}
 
const int ast_audiosocket_send_frame(const int svc, const struct ast_frame *f)
{
    int ret = 0;
    uint8_t kind = 0x10;    /* always 16-bit, 8kHz signed linear mono, for now */
    uint8_t *p;
    uint8_t buf[3 + f->datalen];
 
    p = buf;
 
    *(p++) = kind;
    *(p++) = f->datalen >> 8;
    *(p++) = f->datalen & 0xff;
    memcpy(p, f->data.ptr, f->datalen);
 
    if (write(svc, buf, 3 + f->datalen) != 3 + f->datalen) {
        ast_log(LOG_WARNING, "Failed to write data to AudioSocket\n");
        ret = -1;
    }
 
    return ret;
}
 
struct ast_frame *ast_audiosocket_receive_frame(const int svc)
{
 
    int i = 0, n = 0, ret = 0, not_audio = 0;
    struct ast_frame f = {
        .frametype = AST_FRAME_VOICE,
        .subclass.format = ast_format_slin,
        .src = "AudioSocket",
        .mallocd = AST_MALLOCD_DATA,
    };
    uint8_t kind;
    uint8_t len_high;
    uint8_t len_low;
    uint16_t len = 0;
    uint8_t *data;
 
    n = read(svc, &kind, 1);
    if (n < 0 && errno == EAGAIN) {
        return &ast_null_frame;
    }
    if (n == 0) {
        return &ast_null_frame;
    }
    if (n != 1) {
        ast_log(LOG_WARNING, "Failed to read type header from AudioSocket\n");
        return NULL;
    }
    if (kind == 0x00) {
        /* AudioSocket ended by remote */
        return NULL;
    }
    if (kind != 0x10) {
        /* read but ignore non-audio message */
        ast_log(LOG_WARNING, "Received non-audio AudioSocket message\n");
        not_audio = 1;
    }
 
    n = read(svc, &len_high, 1);
    if (n != 1) {
        ast_log(LOG_WARNING, "Failed to read data length from AudioSocket\n");
        return NULL;
    }
    len += len_high * 256;
    n = read(svc, &len_low, 1);
    if (n != 1) {
        ast_log(LOG_WARNING, "Failed to read data length from AudioSocket\n");
        return NULL;
    }
    len += len_low;
 
    if (len < 1) {
        return &ast_null_frame;
    }
 
    data = ast_malloc(len);
    if (!data) {
        ast_log(LOG_ERROR, "Failed to allocate for data from AudioSocket\n");
        return NULL;
    }
 
    ret = 0;
    n = 0;
    i = 0;
    while (i < len) {
        n = read(svc, data + i, len - i);
        if (n < 0) {
            ast_log(LOG_ERROR, "Failed to read data from AudioSocket\n");
            ret = n;
            break;
        }
        if (n == 0) {
            ast_log(LOG_ERROR, "Insufficient data read from AudioSocket\n");
            ret = -1;
            break;
        }
        i += n;
    }
 
    if (ret != 0) {
        ast_free(data);
        return NULL;
    }
 
    if (not_audio) {
        ast_free(data);
        return &ast_null_frame;
    }
 
    f.data.ptr = data;
    f.datalen = len;
    f.samples = len / 2;
 
    /* The frame steals data, so it doesn't need to be freed here */
    return ast_frisolate(&f);
}
 
static int load_module(void)
{
    ast_verb(1, "Loading AudioSocket Support module\n");
    return AST_MODULE_LOAD_SUCCESS;
}
 
static int unload_module(void)
{
    ast_verb(1, "Unloading AudioSocket Support module\n");
    return AST_MODULE_LOAD_SUCCESS;
}
 
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "AudioSocket support",
    .support_level = AST_MODULE_SUPPORT_EXTENDED,
    .load = load_module,
    .unload = unload_module,
    .load_pri = AST_MODPRI_CHANNEL_DEPEND,
);