res_audiosocket.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2019, CyCore Systems, Inc
 *
 * Seán C McCord <[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
 *
 * \brief AudioSocket support for Asterisk
 *
 * \author Seán C McCord <[email protected]>
 *
 */

/*** 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,
);