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audiosocket: fix timeout, fix dialplan app exit, server address in logs

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- Correct wait timeout logic in the dialplan application.
- Include server address in log messages for better traceability.
- Allow dialplan app to exit gracefully on hangup messages and socket closure.
- Optimize I/O by reducing redundant read()/write() operations.

Co-authored-by: Florent CHAUVEAU <florentch@pm.me>
(cherry picked from commit 218c64c4eb)
This commit is contained in:
Norm Harrison
2023-04-03 21:36:11 -05:00
committed by Asterisk Development Team
parent 993e26ccf9
commit 428c14d1f1
4 changed files with 171 additions and 125 deletions
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127
res/res_audiosocket.c
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@@ -31,6 +31,7 @@
#include "asterisk.h"
#include "errno.h"
#include <uuid/uuid.h>
#include <arpa/inet.h> /* For byte-order conversion. */
#include "asterisk/file.h"
#include "asterisk/res_audiosocket.h"
@@ -82,7 +83,7 @@ static int handle_audiosocket_connection(const char *server,
}
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_log(LOG_WARNING, "Connection to '%s' failed with error: %s\n",
ast_sockaddr_stringify(&addr), strerror(errno));
return -1;
}
@@ -104,7 +105,7 @@ const int ast_audiosocket_connect(const char *server, struct ast_channel *chan)
if (chan && ast_autoservice_start(chan) < 0) {
ast_log(LOG_WARNING, "Failed to start autoservice for channel "
"%s\n", ast_channel_name(chan));
"'%s'\n", ast_channel_name(chan));
goto end;
}
@@ -115,7 +116,7 @@ const int ast_audiosocket_connect(const char *server, struct ast_channel *chan)
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 - "
ast_log(LOG_ERROR, "Failed to resolve AudioSocket service using '%s' - "
"requires a valid hostname and port\n", server);
goto end;
}
@@ -127,7 +128,7 @@ const int ast_audiosocket_connect(const char *server, struct ast_channel *chan)
/* 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_log(LOG_ERROR, "No port provided for '%s'\n",
ast_sockaddr_stringify(&addrs[i]));
s = -1;
goto end;
@@ -135,7 +136,7 @@ const int ast_audiosocket_connect(const char *server, struct ast_channel *chan)
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));
ast_log(LOG_WARNING, "Unable to create socket: '%s'\n", strerror(errno));
continue;
}
@@ -147,7 +148,7 @@ const int ast_audiosocket_connect(const char *server, struct ast_channel *chan)
}
} else {
ast_log(LOG_ERROR, "Connection to %s failed with unexpected error: %s\n",
ast_log(LOG_ERROR, "Connection to '%s' failed with unexpected error: %s\n",
ast_sockaddr_stringify(&addrs[i]), strerror(errno));
close(s);
s = -1;
@@ -162,7 +163,7 @@ end:
}
if (chan && ast_autoservice_stop(chan) < 0) {
ast_log(LOG_WARNING, "Failed to stop autoservice for channel %s\n",
ast_log(LOG_WARNING, "Failed to stop autoservice for channel '%s'\n",
ast_channel_name(chan));
close(s);
return -1;
@@ -193,13 +194,13 @@ const int ast_audiosocket_init(const int svc, const char *id)
return -1;
}
buf[0] = 0x01;
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\n");
ast_log(LOG_WARNING, "Failed to write data to AudioSocket because: %s\n", strerror(errno));
ret = -1;
}
@@ -208,81 +209,74 @@ const int ast_audiosocket_init(const int svc, const char *id)
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];
uint16_t *length = (uint16_t *) &buf[1];
p = buf;
*(p++) = kind;
*(p++) = f->datalen >> 8;
*(p++) = f->datalen & 0xff;
memcpy(p, f->data.ptr, f->datalen);
/* Audio format is 16-bit, 8kHz signed linear mono for dialplan app,
depends on agreed upon audio codec for channel driver interface. */
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO;
*length = htons(f->datalen);
memcpy(&buf[3], 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;
ast_log(LOG_WARNING, "Failed to write data to AudioSocket because: %s\n", strerror(errno));
return -1;
}
return ret;
return 0;
}
struct ast_frame *ast_audiosocket_receive_frame(const int svc)
{
return ast_audiosocket_receive_frame_with_hangup(svc, NULL);
}
int i = 0, n = 0, ret = 0, not_audio = 0;
struct ast_frame *ast_audiosocket_receive_frame_with_hangup(const int svc,
int *const hangup)
{
int i = 0, n = 0, ret = 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 header[3];
uint8_t *kind = &header[0];
uint16_t *length = (uint16_t *) &header[1];
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;
if (hangup) {
*hangup = 0;
}
n = read(svc, &len_high, 1);
if (n != 1) {
ast_log(LOG_WARNING, "Failed to read data length from AudioSocket\n");
n = read(svc, &header, 3);
if (n == -1) {
ast_log(LOG_WARNING, "Failed to read header from AudioSocket because: %s\n", strerror(errno));
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");
if (n == 0 || *kind == AST_AUDIOSOCKET_KIND_HANGUP) {
/* Socket closure or requested hangup. */
if (hangup) {
*hangup = 1;
}
return NULL;
}
len += len_low;
if (len < 1) {
return &ast_null_frame;
if (*kind != AST_AUDIOSOCKET_KIND_AUDIO) {
ast_log(LOG_ERROR, "Received AudioSocket message other than hangup or audio, refer to protocol specification for valid message types\n");
return NULL;
}
data = ast_malloc(len);
/* 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;
@@ -291,15 +285,15 @@ struct ast_frame *ast_audiosocket_receive_frame(const int svc)
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;
while (i < *length) {
n = read(svc, data + i, *length - i);
if (n == -1) {
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 data read from AudioSocket\n");
ast_log(LOG_ERROR, "Insufficient payload read from AudioSocket\n");
ret = -1;
break;
}
@@ -311,14 +305,9 @@ struct ast_frame *ast_audiosocket_receive_frame(const int svc)
return NULL;
}
if (not_audio) {
ast_free(data);
return &ast_null_frame;
}
f.data.ptr = data;
f.datalen = len;
f.samples = len / 2;
f.datalen = *length;
f.samples = *length / 2;
/* The frame steals data, so it doesn't need to be freed here */
return ast_frisolate(&f);