res_audiosocket.c File Reference

AudioSocket support for Asterisk. More...

#include "asterisk.h"
#include "errno.h"
#include <uuid/uuid.h>
#include <arpa/inet.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"

Include dependency graph for res_audiosocket.c:

Go to the source code of this file.

Macros
#define	MAX_CONNECT_TIMEOUT_MSEC   2000
#define	MODULE_DESCRIPTION   "AudioSocket support functions for Asterisk"

Functions
static void	__reg_module (void)
static void	__unreg_module (void)
const int	ast_audiosocket_connect (const char *server, struct ast_channel *chan)
	Send the initial message to an AudioSocket server.
const int	ast_audiosocket_init (const int svc, const char *id)
	Send the initial message to an AudioSocket server.
struct ast_frame *	ast_audiosocket_receive_frame (const int svc)
	Receive an Asterisk frame from an AudioSocket server.
struct ast_frame *	ast_audiosocket_receive_frame_with_hangup (const int svc, int *const hangup)
	Receive an Asterisk frame from an AudioSocket server.
const int	ast_audiosocket_send_frame (const int svc, const struct ast_frame *f)
	Send an Asterisk audio frame to an AudioSocket server.
struct ast_module *	AST_MODULE_SELF_SYM (void)
static int	handle_audiosocket_connection (const char *server, const struct ast_sockaddr addr, const int netsockfd)
static int	load_module (void)
static int	unload_module (void)

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER , .description = "AudioSocket support" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_EXTENDED, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DEPEND, }
static const struct ast_module_info *	ast_module_info = &__mod_info

Detailed Description

AudioSocket support for Asterisk.

Author

Seán C McCord scm@c.nosp@m.ycor.nosp@m.esys..nosp@m.com

Definition in file res_audiosocket.c.

Macro Definition Documentation

MAX_CONNECT_TIMEOUT_MSEC

#define MAX_CONNECT_TIMEOUT_MSEC   2000

Definition at line 45 of file res_audiosocket.c.

MODULE_DESCRIPTION

#define MODULE_DESCRIPTION   "AudioSocket support functions for Asterisk"

Definition at line 43 of file res_audiosocket.c.

Function Documentation

__reg_module()

static void __reg_module ( void  )

static

Definition at line 444 of file res_audiosocket.c.

__unreg_module()

static void __unreg_module ( void  )

static

Definition at line 444 of file res_audiosocket.c.

ast_audiosocket_connect()

const int ast_audiosocket_connect	(	const char *	server,
		struct ast_channel *	chan
	)

Send the initial message to an AudioSocket server.

Parameters

server	The server address, including port.
chan	An optional channel which will be put into autoservice during the connection period. If there is no channel to be autoserviced, pass NULL instead.

Return values

socket	file descriptor for AudioSocket on success
-1	on error

Definition at line 100 of file res_audiosocket.c.

{
    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;
        }
 
        /*
         * Disable Nagle's algorithm by setting the TCP_NODELAY socket option.
         * This reduces latency by preventing delays caused by packet buffering.
         */
        if (setsockopt(s, IPPROTO_TCP, TCP_NODELAY, &(int){1}, sizeof(int)) < 0) {
            ast_log(LOG_ERROR, "Failed to set TCP_NODELAY on AudioSocket: %s\n", strerror(errno));
        }
 
        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;
}

References AST_AF_UNSPEC, ast_autoservice_start(), ast_autoservice_stop(), ast_channel_name(), ast_connect(), ast_free, ast_log, ast_sockaddr_port, ast_sockaddr_resolve(), ast_sockaddr_stringify(), ast_socket_nonblock, ast_strlen_zero(), end, errno, handle_audiosocket_connection(), LOG_ERROR, LOG_WARNING, NULL, PARSE_PORT_REQUIRE, and ast_sockaddr::ss.

Referenced by audiosocket_exec(), and audiosocket_request().

ast_audiosocket_init()

const int ast_audiosocket_init	(	const int	svc,
		const char *	id
	)

Send the initial message to an AudioSocket server.

Parameters

svc	The file descriptor of the network socket to the AudioSocket server.
id	The UUID to send to the AudioSocket server to uniquely identify this connection.

Return values

0	on success
-1	on error

Definition at line 189 of file res_audiosocket.c.

{
    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] = AST_AUDIOSOCKET_KIND_UUID;
    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 because: %s\n", strerror(errno));
        ret = -1;
    }
 
    return ret;
}

References AST_AUDIOSOCKET_KIND_UUID, ast_log, ast_strlen_zero(), buf, errno, LOG_ERROR, and LOG_WARNING.

Referenced by audiosocket_call(), and audiosocket_run().

ast_audiosocket_receive_frame()

struct ast_frame * ast_audiosocket_receive_frame

(

const int

svc

)

Receive an Asterisk frame from an AudioSocket server.

This returned object is a pointer to an Asterisk frame which must be manually freed by the caller.

Parameters

svc	The file descriptor of the network socket to the AudioSocket server.

Return values

A	ast_frame on success
NULL	on error

Definition at line 282 of file res_audiosocket.c.

{
    return ast_audiosocket_receive_frame_with_hangup(svc, NULL);
}

References ast_audiosocket_receive_frame_with_hangup(), and NULL.

ast_audiosocket_receive_frame_with_hangup()

struct ast_frame * ast_audiosocket_receive_frame_with_hangup	(	const int	svc,
		int *const	hangup
	)

Receive an Asterisk frame from an AudioSocket server.

This returned object is a pointer to an Asterisk frame which must be manually freed by the caller.

Parameters

svc	The file descriptor of the network socket to the AudioSocket server.
hangup	Will be true if the AudioSocket server requested the channel be hung up, otherwise false. Used as an out-parameter only, pass NULL if not needed. The function return value will always be NULL when true.

Return values

A	ast_frame on success
NULL	on error or when the hungup parameter is true.

Definition at line 287 of file res_audiosocket.c.

{
    int i = 0, n = 0, ret = 0;
    struct ast_frame f = {
        .frametype = AST_FRAME_VOICE,
        .src = "AudioSocket",
        .mallocd = AST_MALLOCD_DATA,
    };
    uint8_t header[3];
    uint8_t *kind = &header[0];
    uint16_t *length = (uint16_t *) &header[1];
    uint8_t *data;
 
    if (hangup) {
        *hangup = 0;
    }
 
    while (i < 3) {
        n = read(svc, header + i, 3 - i);
        if (n == -1) {
            if (errno == EAGAIN || errno == EWOULDBLOCK) {
                int poll_result = ast_wait_for_input(svc, 5);
 
                if (poll_result == 1) {
                    continue;
                } else if (poll_result == 0) {
                    ast_debug(1, "Poll timed out while waiting for header data\n");
                    continue;
                } else {
                    ast_log(LOG_WARNING, "Poll error: %s\n", strerror(errno));
                }
            }
 
            ast_log(LOG_ERROR, "Failed to read header from AudioSocket because: %s\n", strerror(errno));
            return NULL;
        }
        if (n == 0) {
            break;
        }
        i += n;
    }
 
    if (n == 0 || *kind == AST_AUDIOSOCKET_KIND_HANGUP) {
        /* Socket closure or requested hangup. */
        if (hangup) {
            *hangup = 1;
        }
        return NULL;
    }
 
    switch (*kind) {
        case AST_AUDIOSOCKET_KIND_AUDIO:
            f.subclass.format = ast_format_slin;
            break;
        case AST_AUDIOSOCKET_KIND_AUDIO_SLIN12:
            f.subclass.format = ast_format_slin12;
            break;
        case AST_AUDIOSOCKET_KIND_AUDIO_SLIN16:
            f.subclass.format = ast_format_slin16;
            break;
        case AST_AUDIOSOCKET_KIND_AUDIO_SLIN24:
            f.subclass.format = ast_format_slin24;
            break;
        case AST_AUDIOSOCKET_KIND_AUDIO_SLIN32:
            f.subclass.format = ast_format_slin32;
            break;
        case AST_AUDIOSOCKET_KIND_AUDIO_SLIN44:
            f.subclass.format = ast_format_slin44;
            break;
        case AST_AUDIOSOCKET_KIND_AUDIO_SLIN48:
            f.subclass.format = ast_format_slin48;
            break;
        case AST_AUDIOSOCKET_KIND_AUDIO_SLIN96:
            f.subclass.format = ast_format_slin96;
            break;
        case AST_AUDIOSOCKET_KIND_AUDIO_SLIN192:
            f.subclass.format = ast_format_slin192;
            break;
        default:
            ast_log(LOG_ERROR, "Received AudioSocket message other than hangup or audio, refer to protocol specification for valid message types\n");
            return NULL;
    }
 
    /* Swap endianess of length if needed. */
    *length = ntohs(*length);
    if (*length < 1) {
        ast_log(LOG_ERROR, "Invalid message length received from AudioSocket server. \n");
        return NULL;
    }
 
    data = ast_malloc(*length);
    if (!data) {
        ast_log(LOG_ERROR, "Failed to allocate for data from AudioSocket\n");
        return NULL;
    }
 
    ret = 0;
    n = 0;
    i = 0;
    while (i < *length) {
        n = read(svc, data + i, *length - i);
        if (n == -1) {
            if (errno == EAGAIN || errno == EWOULDBLOCK) {
                int poll_result = ast_wait_for_input(svc, 5);
 
                if (poll_result == 1) {
                    continue;
                } else if (poll_result == 0) {
                    ast_log(LOG_WARNING, "Poll timed out while waiting for data\n");
                } else {
                    ast_log(LOG_WARNING, "Poll error: %s\n", strerror(errno));
                }
            }
 
            ast_log(LOG_ERROR, "Failed to read payload from AudioSocket: %s\n", strerror(errno));
            ret = -1;
            break;
        }
        if (n == 0) {
            ast_log(LOG_ERROR, "Insufficient payload read from AudioSocket\n");
            ret = -1;
            break;
        }
        i += n;
    }
 
    if (ret != 0) {
        ast_free(data);
        return NULL;
    }
 
    f.data.ptr = data;
    f.datalen = *length;
    f.samples = *length / 2;
 
    /* The frame steals data, so it doesn't need to be freed here */
    return ast_frisolate(&f);
}

References AST_AUDIOSOCKET_KIND_AUDIO, AST_AUDIOSOCKET_KIND_AUDIO_SLIN12, AST_AUDIOSOCKET_KIND_AUDIO_SLIN16, AST_AUDIOSOCKET_KIND_AUDIO_SLIN192, AST_AUDIOSOCKET_KIND_AUDIO_SLIN24, AST_AUDIOSOCKET_KIND_AUDIO_SLIN32, AST_AUDIOSOCKET_KIND_AUDIO_SLIN44, AST_AUDIOSOCKET_KIND_AUDIO_SLIN48, AST_AUDIOSOCKET_KIND_AUDIO_SLIN96, AST_AUDIOSOCKET_KIND_HANGUP, ast_debug, ast_format_slin, ast_format_slin12, ast_format_slin16, ast_format_slin192, ast_format_slin24, ast_format_slin32, ast_format_slin44, ast_format_slin48, ast_format_slin96, AST_FRAME_VOICE, ast_free, ast_frisolate, ast_log, ast_malloc, AST_MALLOCD_DATA, ast_wait_for_input(), ast_frame::data, ast_frame::datalen, errno, ast_frame_subclass::format, ast_frame::frametype, hangup(), LOG_ERROR, LOG_WARNING, NULL, ast_frame::ptr, ast_frame::samples, and ast_frame::subclass.

Referenced by ast_audiosocket_receive_frame(), audiosocket_read(), and audiosocket_run().

ast_audiosocket_send_frame()

const int ast_audiosocket_send_frame	(	const int	svc,
		const struct ast_frame *	f
	)

Send an Asterisk audio frame to an AudioSocket server.

Parameters

svc	The file descriptor of the network socket to the AudioSocket server.
f	The Asterisk audio frame to send.

Return values

0	on success
-1	on error

Definition at line 218 of file res_audiosocket.c.

{
    int datalen = f->datalen;
    if (f->frametype == AST_FRAME_DTMF) {
        datalen = 1;
    }
 
    {
        uint8_t buf[3 + datalen];
        uint16_t *length = (uint16_t *) &buf[1];
 
        /* Audio format is 16-bit, 8kHz signed linear mono for dialplan app,
            depends on agreed upon audio codec for channel driver interface. */
        switch (f->frametype) {
            case AST_FRAME_VOICE:
                if (ast_format_cmp(f->subclass.format, ast_format_slin) == AST_FORMAT_CMP_EQUAL) {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO;
                } else if (ast_format_cmp(f->subclass.format, ast_format_slin12) == AST_FORMAT_CMP_EQUAL) {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN12;
                } else if (ast_format_cmp(f->subclass.format, ast_format_slin16) == AST_FORMAT_CMP_EQUAL) {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN16;
                } else if (ast_format_cmp(f->subclass.format, ast_format_slin24) == AST_FORMAT_CMP_EQUAL) {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN24;
                } else if (ast_format_cmp(f->subclass.format, ast_format_slin32) == AST_FORMAT_CMP_EQUAL) {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN32;
                } else if (ast_format_cmp(f->subclass.format, ast_format_slin44) == AST_FORMAT_CMP_EQUAL) {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN44;
                } else if (ast_format_cmp(f->subclass.format, ast_format_slin48) == AST_FORMAT_CMP_EQUAL) {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN48;
                } else if (ast_format_cmp(f->subclass.format, ast_format_slin96) == AST_FORMAT_CMP_EQUAL) {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN96;
                } else if (ast_format_cmp(f->subclass.format, ast_format_slin192) == AST_FORMAT_CMP_EQUAL) {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN192;
                } else {
                    buf[0] = AST_AUDIOSOCKET_KIND_AUDIO;
                }
 
                *length = htons(datalen);
                memcpy(&buf[3], f->data.ptr, datalen);
                break;
            case AST_FRAME_DTMF:
                buf[0] = AST_AUDIOSOCKET_KIND_DTMF;
                buf[3] = (uint8_t) f->subclass.integer;
                *length = htons(1);
                break;
            case AST_FRAME_CNG:
                return 0;
            default: {
                char frame_type[32];
                ast_frame_type2str(f->frametype, frame_type, sizeof(frame_type));
                ast_log(LOG_WARNING, "Unsupported frame type %s (%d) for AudioSocket\n", frame_type, f->frametype);
                return 0;
            }
        }
 
        if (write(svc, buf, 3 + datalen) != 3 + datalen) {
            ast_log(LOG_WARNING, "Failed to write data to AudioSocket because: %s\n", strerror(errno));
            return -1;
        }
    }
 
    return 0;
}

References AST_AUDIOSOCKET_KIND_AUDIO, AST_AUDIOSOCKET_KIND_AUDIO_SLIN12, AST_AUDIOSOCKET_KIND_AUDIO_SLIN16, AST_AUDIOSOCKET_KIND_AUDIO_SLIN192, AST_AUDIOSOCKET_KIND_AUDIO_SLIN24, AST_AUDIOSOCKET_KIND_AUDIO_SLIN32, AST_AUDIOSOCKET_KIND_AUDIO_SLIN44, AST_AUDIOSOCKET_KIND_AUDIO_SLIN48, AST_AUDIOSOCKET_KIND_AUDIO_SLIN96, AST_AUDIOSOCKET_KIND_DTMF, ast_format_cmp(), AST_FORMAT_CMP_EQUAL, ast_format_slin, ast_format_slin12, ast_format_slin16, ast_format_slin192, ast_format_slin24, ast_format_slin32, ast_format_slin44, ast_format_slin48, ast_format_slin96, AST_FRAME_CNG, AST_FRAME_DTMF, ast_frame_type2str(), AST_FRAME_VOICE, ast_log, buf, ast_frame::data, ast_frame::datalen, errno, ast_frame_subclass::format, ast_frame::frametype, ast_frame_subclass::integer, LOG_WARNING, ast_frame::ptr, and ast_frame::subclass.

Referenced by audiosocket_run(), audiosocket_send_dtmf(), and audiosocket_write().

AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM

(

void

)

Definition at line 444 of file res_audiosocket.c.

handle_audiosocket_connection()

static int handle_audiosocket_connection ( const char *  server, const struct ast_sockaddr  addr, const int  netsockfd  )

static

Definition at line 58 of file res_audiosocket.c.

{
    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;
}

References ast_log, ast_poll, ast_sockaddr_stringify(), errno, LOG_WARNING, and MAX_CONNECT_TIMEOUT_MSEC.

Referenced by ast_audiosocket_connect().

load_module()

static int load_module ( void  )

static

Definition at line 427 of file res_audiosocket.c.

{
    ast_verb(5, "Loading AudioSocket Support module\n");
    return AST_MODULE_LOAD_SUCCESS;
}

References AST_MODULE_LOAD_SUCCESS, and ast_verb.

unload_module()

static int unload_module ( void  )

static

Definition at line 433 of file res_audiosocket.c.

{
    ast_verb(5, "Unloading AudioSocket Support module\n");
    return AST_MODULE_LOAD_SUCCESS;
}

References AST_MODULE_LOAD_SUCCESS, and ast_verb.

Variable Documentation

__mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER , .description = "AudioSocket support" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_EXTENDED, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DEPEND, }

static

Definition at line 444 of file res_audiosocket.c.

ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 444 of file res_audiosocket.c.