kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2026-07-23 16:05:30 -07:00
Code Issues Packages Projects Releases Wiki Activity
Files
20
asterisk/res/res_http_websocket.c
T
Thomas Guebels 0330d37340 res_http_websocket: Unref session when it fails to establish. 
Two error paths in res_http_websocket could leak an ast_websocket session by
failing to drop a reference after the session had been allocated.

The first occurs when session ID generation fails. While this is largely
theoretical, as it would require an out-of-memory condition, the session
reference should still be released correctly.

The second occurs when the ast_websocket_pre_callback session_attempted callback
returns an error. This path is reachable through ari_websockets, which
implements this callback and can legitimately fail under certain conditions.

Fixes: #2020
2026-07-09 16:49:59 +00:00

2222 lines
73 KiB
C
Raw Permalink Blame History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2012, Digium, Inc.
*
* Joshua Colp <jcolp@digium.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 WebSocket support for the Asterisk internal HTTP server
*
* \author Joshua Colp <jcolp@digium.com>
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/module.h"
#include "asterisk/http.h"
#include "asterisk/astobj2.h"
#include "asterisk/strings.h"
#include "asterisk/file.h"
#include "asterisk/sched.h"
#include "asterisk/unaligned.h"
#include "asterisk/uri.h"
#include "asterisk/uuid.h"
#define AST_API_MODULE
#include "asterisk/http_websocket.h"
/*! \brief GUID used to compute the accept key, defined in the specifications */
#define WEBSOCKET_GUID "258EAFA5-E914-47DA-95CA-C5AB0DC85B11"
/*! \brief Length of a websocket's client key */
#define CLIENT_KEY_SIZE 16
/*! \brief Number of buckets for registered protocols */
#define MAX_PROTOCOL_BUCKETS 7
#ifdef LOW_MEMORY
/*! \brief Default reconstruction size for multi-frame payload reconstruction. If exceeded the next frame will start a
* payload.
*/
#define DEFAULT_RECONSTRUCTION_CEILING AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE
/*! \brief Maximum reconstruction size for multi-frame payload reconstruction. */
#define MAXIMUM_RECONSTRUCTION_CEILING AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE
#else
/*! \brief Default reconstruction size for multi-frame payload reconstruction. If exceeded the next frame will start a
* payload.
*/
#define DEFAULT_RECONSTRUCTION_CEILING AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE
/*! \brief Maximum reconstruction size for multi-frame payload reconstruction. */
#define MAXIMUM_RECONSTRUCTION_CEILING AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE
#endif
/*! \brief Maximum size of a websocket frame header
* 1 byte flags and opcode
* 1 byte mask flag + payload len
* 8 bytes max extended length
* 4 bytes optional masking key
* ... payload follows ...
* */
#define MAX_WS_HDR_SZ 14
#define MIN_WS_HDR_SZ 2
/*! \brief WS_PING_PAYLOAD
* It's possible that a user of this API could be sending their own PINGs
* and expecting to see PONGs so we use the PING_PAYLOAD in the PINGs we
* send so we can detect that any PONGs we receive are from our own PINGs.
*/
#define WS_PING_PAYLOAD "WS_CLIENT_PING"
#define WS_PING_PAYLOAD_LEN 14
static struct ast_sched_context *ping_scheduler;
enum ws_closed_by {
WS_NOT_CLOSED = 0,
WS_CLOSED_BY_REMOTE,
WS_CLOSED_BY_US,
};
struct websocket_client {
/*! Options used to create the client */
struct ast_websocket_client_options *options;
/*! host portion of client uri */
char *host;
/*! path for logical websocket connection */
struct ast_str *resource_name;
/*! unique key used during server handshaking */
char *key;
/*! container for registered protocols */
char *protocols;
/*! the protocol accepted by the server */
char *accept_protocol;
/*! websocket protocol version */
int version;
/*! tcptls connection arguments */
struct ast_tcptls_session_args *args;
/*! tcptls connection instance */
struct ast_tcptls_session_instance *ser;
/*! Authentication userid:password */
char *userinfo;
/*! Suppress connection log messages */
int suppress_connection_msgs;
/*! Proxy-Authentication userid:password */
char *proxy_userinfo;
/*! The ping scheduler timer id */
int ping_sched_timer;
/*! How many missed pong responses currently */
int missed_pong_count;
};
/*! \brief Structure definition for session */
struct ast_websocket {
struct ast_iostream *stream; /*!< iostream of the connection */
struct ast_sockaddr remote_address; /*!< Address of the remote client */
struct ast_sockaddr local_address; /*!< Our local address */
enum ast_websocket_opcode opcode; /*!< Cached opcode for multi-frame messages */
size_t payload_len; /*!< Length of the payload */
char *payload; /*!< Pointer to the payload */
size_t reconstruct; /*!< Number of bytes before a reconstructed payload will be returned and a new one started */
int timeout; /*!< The timeout for operations on the socket */
unsigned int secure:1; /*!< Bit to indicate that the transport is secure */
unsigned int closing:1; /*!< Bit to indicate that the session is in the process of being closed */
unsigned int close_sent:1; /*!< Bit to indicate that the session close opcode has been sent and no further data will be sent */
unsigned int non_blocking:1; /*!< Bit to indicate that the socket is non-blocking */
struct websocket_client *client; /*!< Client object when connected as a client websocket */
char session_id[AST_UUID_STR_LEN]; /*!< The identifier for the websocket session */
uint16_t close_status_code; /*!< Status code sent in a CLOSE frame upon shutdown */
enum ws_closed_by closed_by; /*!< Who's closing the websocket? */
char buf[AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE]; /*!< Fixed buffer for reading data into */
};
#define WS_SESSION_REMOTE(_session) (_session ? (_session->client ? _session->client->options->uri : ast_sockaddr_stringify(&_session->remote_address)) : "NULL")
#define ARE_PINGPONGS_ENABLED(_session) (_session && _session->client && _session->client->options->pingpongs && _session->client->ping_sched_timer >= 0)
static const char *closed_by_str[] = {
[WS_NOT_CLOSED] = "not closed",
[WS_CLOSED_BY_REMOTE] = "remote",
[WS_CLOSED_BY_US] = "local"
};
static const char *closed_by_to_str(enum ws_closed_by closed_by)
{
if (!ARRAY_IN_BOUNDS(closed_by, closed_by_str)) {
return "unknown";
}
return closed_by_str[closed_by];
}
const char *ast_websocket_type_to_str(enum ast_websocket_type type)
{
switch (type) {
case AST_WS_TYPE_CLIENT_PERSISTENT:
return "persistent";
case AST_WS_TYPE_CLIENT_PER_CALL:
return "per_call";
case AST_WS_TYPE_CLIENT_PER_CALL_CONFIG:
return "per_call_config";
case AST_WS_TYPE_CLIENT:
return "client";
case AST_WS_TYPE_INBOUND:
return "inbound";
case AST_WS_TYPE_SERVER:
return "server";
case AST_WS_TYPE_ANY:
return "any";
default:
return "unknown";
}
}
/*! \brief Hashing function for protocols */
static int protocol_hash_fn(const void *obj, const int flags)
{
const struct ast_websocket_protocol *protocol = obj;
const char *name = obj;
return ast_str_case_hash(flags & OBJ_KEY ? name : protocol->name);
}
/*! \brief Comparison function for protocols */
static int protocol_cmp_fn(void *obj, void *arg, int flags)
{
const struct ast_websocket_protocol *protocol1 = obj, *protocol2 = arg;
const char *protocol = arg;
return !strcasecmp(protocol1->name, flags & OBJ_KEY ? protocol : protocol2->name) ? CMP_MATCH | CMP_STOP : 0;
}
/*! \brief Destructor function for protocols */
static void protocol_destroy_fn(void *obj)
{
struct ast_websocket_protocol *protocol = obj;
ast_free(protocol->name);
}
/*! \brief Structure for a WebSocket server */
struct ast_websocket_server {
struct ao2_container *protocols; /*!< Container for registered protocols */
};
static void websocket_server_dtor(void *obj)
{
struct ast_websocket_server *server = obj;
ao2_cleanup(server->protocols);
server->protocols = NULL;
}
static struct ast_websocket_server *websocket_server_create_impl(void)
{
RAII_VAR(struct ast_websocket_server *, server, NULL, ao2_cleanup);
server = ao2_alloc(sizeof(*server), websocket_server_dtor);
if (!server) {
return NULL;
}
server->protocols = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
MAX_PROTOCOL_BUCKETS, protocol_hash_fn, NULL, protocol_cmp_fn);
if (!server->protocols) {
return NULL;
}
ao2_ref(server, +1);
return server;
}
static struct ast_websocket_server *websocket_server_internal_create(void)
{
return websocket_server_create_impl();
}
struct ast_websocket_server *AST_OPTIONAL_API_NAME(ast_websocket_server_create)(void)
{
return websocket_server_create_impl();
}
/*! \brief Destructor function for sessions */
static void session_destroy_fn(void *obj)
{
struct ast_websocket *session = obj;
char *id = ast_strdupa(WS_SESSION_REMOTE(session));
SCOPE_ENTER(2, "%s: Session %p destructor\n", id, obj);
if (session->stream) {
ast_websocket_close(session, session->close_status_code);
if (session->stream) {
ast_iostream_close(session->stream);
session->stream = NULL;
ast_trace(-1, "%s: WebSocket connection closed\n", WS_SESSION_REMOTE(session));
}
}
ao2_cleanup(session->client);
ast_free(session->payload);
SCOPE_EXIT_RTN("%s; Session %p destructor complete\n", id, obj);
}
struct ast_websocket_protocol *AST_OPTIONAL_API_NAME(ast_websocket_sub_protocol_alloc)(const char *name)
{
struct ast_websocket_protocol *protocol;
protocol = ao2_alloc(sizeof(*protocol), protocol_destroy_fn);
if (!protocol) {
return NULL;
}
protocol->name = ast_strdup(name);
if (!protocol->name) {
ao2_ref(protocol, -1);
return NULL;
}
protocol->version = AST_WEBSOCKET_PROTOCOL_VERSION;
return protocol;
}
int AST_OPTIONAL_API_NAME(ast_websocket_server_add_protocol)(struct ast_websocket_server *server, const char *name, ast_websocket_callback callback)
{
struct ast_websocket_protocol *protocol;
if (!server->protocols) {
return -1;
}
protocol = ast_websocket_sub_protocol_alloc(name);
if (!protocol) {
return -1;
}
protocol->session_established = callback;
if (ast_websocket_server_add_protocol2(server, protocol)) {
ao2_ref(protocol, -1);
return -1;
}
return 0;
}
int AST_OPTIONAL_API_NAME(ast_websocket_server_add_protocol2)(struct ast_websocket_server *server, struct ast_websocket_protocol *protocol)
{
struct ast_websocket_protocol *existing;
if (!server->protocols) {
return -1;
}
if (protocol->version != AST_WEBSOCKET_PROTOCOL_VERSION) {
ast_log(LOG_WARNING, "WebSocket could not register sub-protocol '%s': "
"expected version '%u', got version '%u'\n",
protocol->name, AST_WEBSOCKET_PROTOCOL_VERSION, protocol->version);
return -1;
}
ao2_lock(server->protocols);
/* Ensure a second protocol handler is not registered for the same protocol */
existing = ao2_find(server->protocols, protocol->name, OBJ_KEY | OBJ_NOLOCK);
if (existing) {
ao2_ref(existing, -1);
ao2_unlock(server->protocols);
return -1;
}
ao2_link_flags(server->protocols, protocol, OBJ_NOLOCK);
ao2_unlock(server->protocols);
ast_debug(1, "WebSocket registered sub-protocol '%s'\n", protocol->name);
ao2_ref(protocol, -1);
return 0;
}
int AST_OPTIONAL_API_NAME(ast_websocket_server_remove_protocol)(struct ast_websocket_server *server, const char *name, ast_websocket_callback callback)
{
struct ast_websocket_protocol *protocol;
if (!(protocol = ao2_find(server->protocols, name, OBJ_KEY))) {
return -1;
}
if (protocol->session_established != callback) {
ao2_ref(protocol, -1);
return -1;
}
ao2_unlink(server->protocols, protocol);
ao2_ref(protocol, -1);
ast_debug(1, "WebSocket unregistered sub-protocol '%s'\n", name);
return 0;
}
/*! \brief Perform payload masking for client sessions */
static void websocket_mask_payload(struct ast_websocket *session, char *frame, char *payload, uint64_t payload_size)
{
/* RFC 6455 5.1 - clients MUST mask frame data */
if (session->client) {
uint64_t i;
uint8_t mask_key_idx;
uint32_t mask_key = ast_random();
uint8_t length = frame[1] & 0x7f;
frame[1] |= 0x80; /* set mask bit to 1 */
/* The mask key octet position depends on the length */
mask_key_idx = length == 126 ? 4 : length == 127 ? 10 : 2;
put_unaligned_uint32(&frame[mask_key_idx], mask_key);
for (i = 0; i < payload_size; i++) {
payload[i] ^= ((char *)&mask_key)[i % 4];
}
}
}
static const char *opcode_map[] = {
[AST_WEBSOCKET_OPCODE_CONTINUATION] = "continuation",
[AST_WEBSOCKET_OPCODE_TEXT] = "text",
[AST_WEBSOCKET_OPCODE_BINARY] = "binary",
[AST_WEBSOCKET_OPCODE_CLOSE] = "close",
[AST_WEBSOCKET_OPCODE_PING] = "ping",
[AST_WEBSOCKET_OPCODE_PONG] = "pong",
};
static const char *websocket_opcode2str(enum ast_websocket_opcode opcode)
{
if (!ARRAY_IN_BOUNDS(opcode, opcode_map)) {
return "<unknown>";
}
return opcode_map[opcode];
}
static void ping_scheduler_cancel(struct ast_websocket *session)
{
int enabled = ARE_PINGPONGS_ENABLED(session);
SCOPE_ENTER(2, "%s: Cancelling PING/PONG keepalives\n", WS_SESSION_REMOTE(session));
if (!enabled) {
SCOPE_EXIT_RTN("%s: Not enabled, cancel not needed\n", WS_SESSION_REMOTE(session));
}
AST_SCHED_DEL(ping_scheduler, session->client->ping_sched_timer);
ao2_ref(session, -1);
SCOPE_EXIT_RTN("%s: Cancelled PING/PONG keepalives\n", WS_SESSION_REMOTE(session));
}
static int websocket_close(struct ast_websocket *session, uint16_t reason, int force)
{
enum ast_websocket_opcode opcode = AST_WEBSOCKET_OPCODE_CLOSE;
/* The header is either 2 or 6 bytes and the
* reason code takes up another 2 bytes */
char frame[8] = { 0, };
int header_size, fsize, res = 0;
int fd = session->stream ? ast_iostream_get_fd(session->stream) : -1;
SCOPE_ENTER(2, "%s: Close requested. Reason: %s (%d) Closed by: %s Force: %s\n", WS_SESSION_REMOTE(session),
ast_websocket_status_to_str(reason), reason, closed_by_to_str(session->closed_by), AST_YESNO(force));
ping_scheduler_cancel(session);
ao2_lock(session);
if (session->closing) {
ao2_unlock(session);
SCOPE_EXIT_RTN_VALUE(0, "%s: Close already sent\n", WS_SESSION_REMOTE(session));
}
session->closing = 1;
if (!session->stream) {
ao2_unlock(session);
SCOPE_EXIT_RTN_VALUE(-1, "%s: WebSocket stream already closed\n", WS_SESSION_REMOTE(session));
}
if (force) {
ast_trace(-1, "%s: Forcing close. Handle: %p FD: %d\n", WS_SESSION_REMOTE(session),
session->stream, fd);
ast_iostream_close(session->stream);
session->stream = NULL;
ao2_unlock(session);
SCOPE_EXIT_RTN_VALUE(-1, "%s: Forced close\n", WS_SESSION_REMOTE(session));
}
/* clients need space for an additional 4 byte masking key */
header_size = session->client ? 6 : 2;
fsize = header_size + 2;
session->close_sent = 1;
frame[0] = opcode | 0x80;
frame[1] = 2; /* The reason code is always 2 bytes */
/*
* If the remote initiated the close we should respond with the same
* reason code they sent.
*/
if (session->closed_by == WS_CLOSED_BY_REMOTE) {
reason = session->close_status_code;
}
/* If no reason has been specified assume 1000 which is normal closure */
put_unaligned_uint16(&frame[header_size], htons(reason ? reason : 1000));
websocket_mask_payload(session, frame, &frame[header_size], 2);
ast_trace(-1, "%s: Writing %sCLOSE frame with reason %s (%d). fd: %d\n", WS_SESSION_REMOTE(session),
session->closed_by == WS_CLOSED_BY_REMOTE ? "reply " : "", ast_websocket_status_to_str(reason), reason, fd);
ast_iostream_set_timeout_inactivity(session->stream, session->timeout);
res = ast_iostream_write(session->stream, frame, fsize);
ast_iostream_set_timeout_disable(session->stream);
/*
* If the remote initiated the close or we failed to send,
* we can close the socket. If we just sent a CLOSE of our own, we need to wait
* until we get the close reply from the remote.
*/
if (session->closed_by == WS_CLOSED_BY_REMOTE || res != fsize) {
ast_trace(-1, "%s: %s Closing socket. fd: %d\n",
session->closed_by == WS_CLOSED_BY_REMOTE ? "Wrote CLOSE reply." : "Writing CLOSE failed.",
WS_SESSION_REMOTE(session), fd);
ast_iostream_close(session->stream);
session->stream = NULL;
ao2_unlock(session);
SCOPE_EXIT_RTN_VALUE(0, "%s: Socket closed after %s\n", WS_SESSION_REMOTE(session),
session->closed_by == WS_CLOSED_BY_REMOTE ? "reply" : "write failure");
}
ao2_unlock(session);
SCOPE_EXIT_RTN_VALUE(res == sizeof(frame), "%s: Close done\n", WS_SESSION_REMOTE(session));
}
static int websocket_handled_pong_or_close(struct ast_websocket *session, char *payload, uint64_t payload_len,
enum ast_websocket_opcode opcode)
{
SCOPE_ENTER(4, "%s: Opcode: %s\n", WS_SESSION_REMOTE(session), websocket_opcode2str(opcode));
if (opcode == AST_WEBSOCKET_OPCODE_PONG) {
/*
* If it's from our own PING, reset the missed count.
*/
if (session->client->missed_pong_count
&& payload_len == WS_PING_PAYLOAD_LEN
&& strncmp(payload, WS_PING_PAYLOAD, payload_len) == 0) {
int mpc = session->client->missed_pong_count;
session->client->missed_pong_count = 0;
SCOPE_EXIT_RTN_VALUE(1, "%s: Received PONG from our own PING. Missed count was: %d. Cleared.\n",
WS_SESSION_REMOTE(session), mpc);
} else {
SCOPE_EXIT_RTN_VALUE(0, "%s: Received PONG. Passing up to client.\n", WS_SESSION_REMOTE(session));
}
}
if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
if (payload_len >= 2) {
session->close_status_code = ntohs(get_unaligned_uint16(payload));
}
if (session->closed_by == WS_NOT_CLOSED) {
session->closed_by = WS_CLOSED_BY_REMOTE;
SCOPE_EXIT_RTN_VALUE(1, "%s: Handled CLOSE request by remote with reason %s (%d)\n", WS_SESSION_REMOTE(session),
ast_websocket_status_to_str(session->close_status_code), session->close_status_code);
}
ast_trace(-1, "%s: Received CLOSE response from remote with reason: %s (%d)\n",
WS_SESSION_REMOTE(session), ast_websocket_status_to_str(session->close_status_code),
session->close_status_code);
/*
* We got the close response so we can now clean up the socket.
*/
websocket_close(session, session->close_status_code, 1);
SCOPE_EXIT_RTN_VALUE(1, "%s: Handled CLOSE\n", WS_SESSION_REMOTE(session));
}
SCOPE_EXIT_RTN_VALUE(0, "%s: Unhandled %s opcode\n", WS_SESSION_REMOTE(session), websocket_opcode2str(opcode));
}
/*! \brief Close function for websocket session */
int AST_OPTIONAL_API_NAME(ast_websocket_close)(struct ast_websocket *session, uint16_t reason)
{
if (session->closed_by == WS_NOT_CLOSED) {
session->closed_by = WS_CLOSED_BY_US;
}
return websocket_close(session, reason, 0);
}
/*! \brief Write function for websocket traffic */
int AST_OPTIONAL_API_NAME(ast_websocket_write)(struct ast_websocket *session, enum ast_websocket_opcode opcode,
char *payload, uint64_t payload_size)
{
size_t header_size = 2; /* The minimum size of a websocket frame is 2 bytes */
char *frame;
uint64_t length;
uint64_t frame_size;
SCOPE_ENTER(4, "%s: Opcode: %s Length: %"PRIu64"\n",
WS_SESSION_REMOTE(session), websocket_opcode2str(opcode), payload_size);
if (payload_size < 126) {
length = payload_size;
} else if (payload_size < (1 << 16)) {
length = 126;
/* We need an additional 2 bytes to store the extended length */
header_size += 2;
} else {
length = 127;
/* We need an additional 8 bytes to store the really really extended length */
header_size += 8;
}
if (session->client) {
/* Additional 4 bytes for the client masking key */
header_size += 4;
}
frame_size = header_size + payload_size;
frame = ast_alloca(frame_size + 1);
memset(frame, 0, frame_size + 1);
frame[0] = opcode | 0x80;
frame[1] = length;
/* Use the additional available bytes to store the length */
if (length == 126) {
put_unaligned_uint16(&frame[2], htons(payload_size));
} else if (length == 127) {
put_unaligned_uint64(&frame[2], htonll(payload_size));
}
memcpy(&frame[header_size], payload, payload_size);
websocket_mask_payload(session, frame, &frame[header_size], payload_size);
ao2_lock(session);
if (session->closing) {
ao2_unlock(session);
SCOPE_EXIT_RTN_VALUE(-1, "%s: Websocket already closing\n", WS_SESSION_REMOTE(session));
}
ast_iostream_set_timeout_sequence(session->stream, ast_tvnow(), session->timeout);
if (ast_iostream_write(session->stream, frame, frame_size) != frame_size) {
ao2_unlock(session);
/* 1011 - server terminating connection due to not being able to fulfill the request */
ast_trace(-1, "%s: Closing WS with 1011 because we can't fulfill a write request\n",
WS_SESSION_REMOTE(session));
websocket_close(session, 1011, 1);
SCOPE_EXIT_RTN_VALUE(-1, "%s: Closed WS with 1011 because we couldn't fulfill a write request\n",
WS_SESSION_REMOTE(session));
}
ast_iostream_set_timeout_disable(session->stream);
ao2_unlock(session);
SCOPE_EXIT_RTN_VALUE(0, "%s: Wrote opcode: %s length: %"PRIu64"\n",
WS_SESSION_REMOTE(session), websocket_opcode2str(opcode), payload_size);
}
void AST_OPTIONAL_API_NAME(ast_websocket_reconstruct_enable)(struct ast_websocket *session, size_t bytes)
{
session->reconstruct = MIN(bytes, MAXIMUM_RECONSTRUCTION_CEILING);
}
void AST_OPTIONAL_API_NAME(ast_websocket_reconstruct_disable)(struct ast_websocket *session)
{
session->reconstruct = 0;
}
void AST_OPTIONAL_API_NAME(ast_websocket_ref)(struct ast_websocket *session)
{
char *id = ast_strdupa(WS_SESSION_REMOTE(session));
int refcount = session ? ao2_ref(session, 0) : 0;
SCOPE_ENTER(2, "%s: Reffing. Refcount: %d\n", id, refcount);
ao2_ref(session, +1);
SCOPE_EXIT("%s: Reffed. Refcount: %d\n", id, session ? refcount - 1 : 0);
}
void AST_OPTIONAL_API_NAME(ast_websocket_unref)(struct ast_websocket *session)
{
char *id = ast_strdupa(WS_SESSION_REMOTE(session));
int refcount = session ? ao2_ref(session, 0) : 0;
SCOPE_ENTER(2, "%s: Unreffing. Refcount: %d\n", id, refcount);
ao2_cleanup(session);
SCOPE_EXIT("%s: Unreffed. Refcount: %d\n", id, session ? refcount - 1 : 0);
}
int AST_OPTIONAL_API_NAME(ast_websocket_fd)(struct ast_websocket *session)
{
return session->closing ? -1 : ast_iostream_get_fd(session->stream);
}
int AST_OPTIONAL_API_NAME(ast_websocket_wait_for_input)(struct ast_websocket *session, int timeout)
{
return session->closing ? -1 : ast_iostream_wait_for_input(session->stream, timeout);
}
struct ast_sockaddr * AST_OPTIONAL_API_NAME(ast_websocket_remote_address)(struct ast_websocket *session)
{
return &session->remote_address;
}
struct ast_sockaddr * AST_OPTIONAL_API_NAME(ast_websocket_local_address)(struct ast_websocket *session)
{
return &session->local_address;
}
int AST_OPTIONAL_API_NAME(ast_websocket_is_secure)(struct ast_websocket *session)
{
return session->secure;
}
int AST_OPTIONAL_API_NAME(ast_websocket_set_nonblock)(struct ast_websocket *session)
{
ast_iostream_nonblock(session->stream);
ast_iostream_set_exclusive_input(session->stream, 0);
return 0;
}
int AST_OPTIONAL_API_NAME(ast_websocket_set_timeout)(struct ast_websocket *session, int timeout)
{
session->timeout = timeout;
return 0;
}
const char * AST_OPTIONAL_API_NAME(ast_websocket_session_id)(struct ast_websocket *session)
{
return session->session_id;
}
/* MAINTENANCE WARNING on ast_websocket_read()!
*
* We have to keep in mind during this function that the fact that session->fd seems ready
* (via poll) does not necessarily mean we have application data ready, because in the case
* of an SSL socket, there is some encryption data overhead that needs to be read from the
* TCP socket, so poll() may say there are bytes to be read, but whether it is just 1 byte
* or N bytes we do not know that, and we do not know how many of those bytes (if any) are
* for application data (for us) and not just for the SSL protocol consumption
*
* There used to be a couple of nasty bugs here that were fixed in last refactoring but I
* want to document them so the constraints are clear and we do not re-introduce them:
*
* - This function would incorrectly assume that fread() would necessarily return more than
* 1 byte of data, just because a websocket frame is always >= 2 bytes, but the thing
* is we're dealing with a TCP bitstream here, we could read just one byte and that's normal.
* The problem before was that if just one byte was read, the function bailed out and returned
* an error, effectively dropping the first byte of a websocket frame header!
*
* - Another subtle bug was that it would just read up to MAX_WS_HDR_SZ (14 bytes) via fread()
* then assume that executing poll() would tell you if there is more to read, but since
* we're dealing with a buffered stream (session->f is a FILE*), poll would say there is
* nothing else to read (in the real tcp socket session->fd) and we would get stuck here
* without processing the rest of the data in session->f internal buffers until another packet
* came on the network to unblock us!
*
* Note during the header parsing stage we try to read in small chunks just what we need, this
* is buffered data anyways, no expensive syscall required most of the time ...
*/
static inline int ws_safe_read(struct ast_websocket *session, char *buf, size_t len, enum ast_websocket_opcode *opcode)
{
ssize_t rlen;
int xlen = len;
char *rbuf = buf;
int sanity = 10;
ast_assert(len > 0);
if (!len) {
errno = EINVAL;
return -1;
}
ao2_lock(session);
if (!session->stream) {
ao2_unlock(session);
errno = ECONNABORTED;
return -1;
}
for (;;) {
rlen = ast_iostream_read(session->stream, rbuf, xlen);
if (rlen != xlen) {
if (rlen == 0) {
ast_log(LOG_WARNING, "Web socket closed abruptly\n");
*opcode = AST_WEBSOCKET_OPCODE_CLOSE;
session->closing = 1;
ao2_unlock(session);
return -1;
}
if (rlen < 0 && errno != EAGAIN) {
ast_log(LOG_ERROR, "Error reading from web socket: %s\n", strerror(errno));
*opcode = AST_WEBSOCKET_OPCODE_CLOSE;
session->closing = 1;
ao2_unlock(session);
return -1;
}
if (!--sanity) {
ast_log(LOG_WARNING, "Websocket seems unresponsive, disconnecting ...\n");
*opcode = AST_WEBSOCKET_OPCODE_CLOSE;
session->closing = 1;
ao2_unlock(session);
return -1;
}
}
if (rlen > 0) {
xlen = xlen - rlen;
rbuf = rbuf + rlen;
if (!xlen) {
break;
}
}
if (ast_iostream_wait_for_input(session->stream, 1000) < 0) {
ast_log(LOG_ERROR, "ast_iostream_wait_for_input returned err: %s\n", strerror(errno));
*opcode = AST_WEBSOCKET_OPCODE_CLOSE;
session->closing = 1;
ao2_unlock(session);
return -1;
}
}
ao2_unlock(session);
return 0;
}
int AST_OPTIONAL_API_NAME(ast_websocket_read)(struct ast_websocket *session, char **payload, uint64_t *payload_len, enum ast_websocket_opcode *opcode, int *fragmented)
{
int fin = 0;
int mask_present = 0;
char *mask = NULL, *new_payload = NULL;
size_t options_len = 0, frame_size = 0;
SCOPE_ENTER(4, "%s: Reading\n", WS_SESSION_REMOTE(session));
*payload = NULL;
*payload_len = 0;
*fragmented = 0;
if (ws_safe_read(session, &session->buf[0], MIN_WS_HDR_SZ, opcode)) {
SCOPE_EXIT_RTN_VALUE(-1, "%s: Initial ws_safe_read failed\n", WS_SESSION_REMOTE(session));
}
frame_size += MIN_WS_HDR_SZ;
/* ok, now we have the first 2 bytes, so we know some flags, opcode and payload length (or whether payload length extension will be required) */
*opcode = session->buf[0] & 0xf;
*payload_len = session->buf[1] & 0x7f;
if (*opcode == AST_WEBSOCKET_OPCODE_TEXT || *opcode == AST_WEBSOCKET_OPCODE_BINARY || *opcode == AST_WEBSOCKET_OPCODE_CONTINUATION ||
*opcode == AST_WEBSOCKET_OPCODE_PING || *opcode == AST_WEBSOCKET_OPCODE_PONG || *opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
fin = (session->buf[0] >> 7) & 1;
mask_present = (session->buf[1] >> 7) & 1;
/* Based on the mask flag and payload length, determine how much more we need to read before start parsing the rest of the header */
options_len += mask_present ? 4 : 0;
options_len += (*payload_len == 126) ? 2 : (*payload_len == 127) ? 8 : 0;
if (options_len) {
/* read the rest of the header options */
if (ws_safe_read(session, &session->buf[frame_size], options_len, opcode)) {
SCOPE_EXIT_RTN_VALUE(-1, "%s: ws_safe_read of options failed\n", WS_SESSION_REMOTE(session));
}
frame_size += options_len;
}
if (*payload_len == 126) {
/* Grab the 2-byte payload length */
*payload_len = ntohs(get_unaligned_uint16(&session->buf[2]));
mask = &session->buf[4];
} else if (*payload_len == 127) {
/* Grab the 8-byte payload length */
*payload_len = ntohll(get_unaligned_uint64(&session->buf[2]));
mask = &session->buf[10];
} else {
/* Just set the mask after the small 2-byte header */
mask = &session->buf[2];
}
/* Now read the rest of the payload */
*payload = &session->buf[frame_size]; /* payload will start here, at the end of the options, if any */
frame_size = frame_size + (*payload_len); /* final frame size is header + optional headers + payload data */
if (frame_size > AST_WEBSOCKET_MAX_RX_PAYLOAD_SIZE) {
/* The frame won't fit :-( */
ast_websocket_close(session, 1009);
SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Cannot fit huge websocket frame of %zu bytes\n",
WS_SESSION_REMOTE(session), frame_size);
}
if (*payload_len) {
if (ws_safe_read(session, *payload, *payload_len, opcode)) {
SCOPE_EXIT_RTN_VALUE(-1, "%s: ws_safe_read of payload failed\n", WS_SESSION_REMOTE(session));
}
}
/* If a mask is present unmask the payload */
if (mask_present) {
unsigned int pos;
for (pos = 0; pos < *payload_len; pos++) {
(*payload)[pos] ^= mask[pos % 4];
}
}
/* Per the RFC for PING we need to send back an opcode with the application data as received */
if (*opcode == AST_WEBSOCKET_OPCODE_PING) {
if (ast_websocket_write(session, AST_WEBSOCKET_OPCODE_PONG, *payload, *payload_len)) {
ast_websocket_close(session, 1009);
}
*payload_len = 0;
SCOPE_EXIT_RTN_VALUE(0, "%s: PING received. Sent PONG\n", WS_SESSION_REMOTE(session));
}
if (websocket_handled_pong_or_close(session, *payload, *payload_len, *opcode)) {
SCOPE_EXIT_RTN_VALUE(0, "%s: Handled PONG or CLOSE\n", WS_SESSION_REMOTE(session));
}
/* Below this point we are handling TEXT, BINARY or CONTINUATION opcodes */
if (*payload_len) {
if (!(new_payload = ast_realloc(session->payload, (session->payload_len + *payload_len)))) {
*payload_len = 0;
ast_websocket_close(session, 1009);
SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Failed allocation: %p, %zu, %"PRIu64"\n",
WS_SESSION_REMOTE(session), session->payload, session->payload_len, *payload_len);
}
session->payload = new_payload;
memcpy((session->payload + session->payload_len), (*payload), (*payload_len));
session->payload_len += *payload_len;
} else if (!session->payload_len && session->payload) {
ast_free(session->payload);
session->payload = NULL;
}
if (!fin && session->reconstruct && (session->payload_len < session->reconstruct)) {
/* If this is not a final message we need to defer returning it until later */
if (*opcode != AST_WEBSOCKET_OPCODE_CONTINUATION) {
session->opcode = *opcode;
}
*opcode = AST_WEBSOCKET_OPCODE_CONTINUATION;
*payload_len = 0;
*payload = NULL;
} else {
if (*opcode == AST_WEBSOCKET_OPCODE_CONTINUATION) {
if (!fin) {
/* If this was not actually the final message tell the user it is fragmented so they can deal with it accordingly */
*fragmented = 1;
} else {
/* Final frame in multi-frame so push up the actual opcode */
*opcode = session->opcode;
}
}
*payload_len = session->payload_len;
*payload = session->payload;
session->payload_len = 0;
}
} else {
ast_log(LOG_WARNING, "WebSocket unknown opcode %u\n", *opcode);
/* We received an opcode that we don't understand, the RFC states that 1003 is for a type of data that can't be accepted... opcodes
* fit that, I think. */
ast_websocket_close(session, 1003);
}
SCOPE_EXIT_RTN_VALUE(0, "%s: Read complete. Opcode: %s Length: %"PRIu64"\n",
WS_SESSION_REMOTE(session), websocket_opcode2str(*opcode), *payload_len);
}
/*!
* \brief If the server has exactly one configured protocol, return it.
*/
static struct ast_websocket_protocol *one_protocol(
struct ast_websocket_server *server)
{
SCOPED_AO2LOCK(lock, server->protocols);
if (ao2_container_count(server->protocols) != 1) {
return NULL;
}
return ao2_callback(server->protocols, OBJ_NOLOCK, NULL, NULL);
}
static char *websocket_combine_key(const char *key, char *res, int res_size)
{
char *combined;
unsigned combined_length = strlen(key) + strlen(WEBSOCKET_GUID) + 1;
uint8_t sha[20];
combined = ast_alloca(combined_length);
snprintf(combined, combined_length, "%s%s", key, WEBSOCKET_GUID);
ast_sha1_hash_uint(sha, combined);
ast_base64encode(res, (const unsigned char*)sha, 20, res_size);
return res;
}
static void websocket_bad_request(struct ast_tcptls_session_instance *ser)
{
struct ast_str *http_header = ast_str_create(64);
if (!http_header) {
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 500, "Server Error", "Out of memory");
return;
}
ast_str_set(&http_header, 0, "Sec-WebSocket-Version: 7, 8, 13\r\n");
ast_http_send(ser, AST_HTTP_UNKNOWN, 400, "Bad Request", http_header, NULL, 0, 0);
}
int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instance *ser, const struct ast_http_uri *urih, const char *uri, enum ast_http_method method, struct ast_variable *get_vars, struct ast_variable *headers)
{
struct ast_variable *v;
const char *upgrade = NULL, *key = NULL, *key1 = NULL, *key2 = NULL, *protos = NULL;
char *requested_protocols = NULL, *protocol = NULL;
int version = 0, flags = 1;
struct ast_websocket_protocol *protocol_handler = NULL;
struct ast_websocket *session;
struct ast_websocket_server *server;
SCOPED_MODULE_USE(ast_module_info->self);
/* Upgrade requests are only permitted on GET methods */
if (method != AST_HTTP_GET) {
ast_http_error(ser, 501, "Not Implemented", "Attempt to use unimplemented / unsupported method");
return 0;
}
server = urih->data;
/* Get the minimum headers required to satisfy our needs */
for (v = headers; v; v = v->next) {
if (!strcasecmp(v->name, "Upgrade")) {
upgrade = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Key")) {
key = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Key1")) {
key1 = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Key2")) {
key2 = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Protocol")) {
protos = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Version")) {
if (sscanf(v->value, "%30d", &version) != 1) {
version = 0;
}
}
}
/* If this is not a websocket upgrade abort */
if (!upgrade || strcasecmp(upgrade, "websocket")) {
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - did not request WebSocket\n",
ast_sockaddr_stringify(&ser->remote_address));
ast_http_error(ser, 426, "Upgrade Required", NULL);
return 0;
} else if (ast_strlen_zero(protos)) {
/* If there's only a single protocol registered, and the
* client doesn't specify what protocol it's using, go ahead
* and accept the connection */
protocol_handler = one_protocol(server);
if (!protocol_handler) {
/* Multiple registered subprotocols; client must specify */
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - no protocols requested\n",
ast_sockaddr_stringify(&ser->remote_address));
websocket_bad_request(ser);
return 0;
}
} else if (key1 && key2) {
/* Specification defined in http://tools.ietf.org/html/draft-hixie-thewebsocketprotocol-76 and
* http://tools.ietf.org/html/draft-ietf-hybi-thewebsocketprotocol-00 -- not currently supported*/
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - unsupported version '00/76' chosen\n",
ast_sockaddr_stringify(&ser->remote_address));
websocket_bad_request(ser);
return 0;
}
if (!protocol_handler && protos) {
requested_protocols = ast_strdupa(protos);
/* Iterate through the requested protocols trying to find one that we have a handler for */
while (!protocol_handler && (protocol = strsep(&requested_protocols, ","))) {
protocol_handler = ao2_find(server->protocols, ast_strip(protocol), OBJ_KEY);
}
}
/* If no protocol handler exists bump this back to the requester */
if (!protocol_handler) {
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - no protocols out of '%s' supported\n",
ast_sockaddr_stringify(&ser->remote_address), protos);
websocket_bad_request(ser);
return 0;
}
/* Determine how to respond depending on the version */
if (version == 7 || version == 8 || version == 13) {
char base64[64];
if (!key || strlen(key) + strlen(WEBSOCKET_GUID) + 1 > 8192) { /* no stack overflows please */
websocket_bad_request(ser);
ao2_ref(protocol_handler, -1);
return 0;
}
if (ast_http_body_discard(ser)) {
websocket_bad_request(ser);
ao2_ref(protocol_handler, -1);
return 0;
}
if (!(session = ao2_alloc(sizeof(*session) + AST_UUID_STR_LEN + 1, session_destroy_fn))) {
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted\n",
ast_sockaddr_stringify(&ser->remote_address));
websocket_bad_request(ser);
ao2_ref(protocol_handler, -1);
return 0;
}
session->timeout = AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT;
/* Generate the session id */
if (!ast_uuid_generate_str(session->session_id, sizeof(session->session_id))) {
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - failed to generate a session id\n",
ast_sockaddr_stringify(&ser->remote_address));
ast_http_error(ser, 500, "Internal Server Error", "Allocation failed");
ao2_ref(session, -1);
ao2_ref(protocol_handler, -1);
return 0;
}
if (protocol_handler->session_attempted
&& protocol_handler->session_attempted(ser, get_vars, headers, session->session_id)) {
ast_debug(3, "WebSocket connection from '%s' rejected by protocol handler '%s'\n",
ast_sockaddr_stringify(&ser->remote_address), protocol_handler->name);
websocket_bad_request(ser);
ao2_ref(session, -1);
ao2_ref(protocol_handler, -1);
return 0;
}
/* RFC 6455, Section 4.1:
*
* 6. If the response includes a |Sec-WebSocket-Protocol| header
* field and this header field indicates the use of a
* subprotocol that was not present in the client's handshake
* (the server has indicated a subprotocol not requested by
* the client), the client MUST _Fail the WebSocket
* Connection_.
*/
if (protocol) {
ast_iostream_printf(ser->stream,
"HTTP/1.1 101 Switching Protocols\r\n"
"Upgrade: %s\r\n"
"Connection: Upgrade\r\n"
"Sec-WebSocket-Accept: %s\r\n"
"Sec-WebSocket-Protocol: %s\r\n\r\n",
upgrade,
websocket_combine_key(key, base64, sizeof(base64)),
protocol);
} else {
ast_iostream_printf(ser->stream,
"HTTP/1.1 101 Switching Protocols\r\n"
"Upgrade: %s\r\n"
"Connection: Upgrade\r\n"
"Sec-WebSocket-Accept: %s\r\n\r\n",
upgrade,
websocket_combine_key(key, base64, sizeof(base64)));
}
} else {
/* Specification defined in http://tools.ietf.org/html/draft-hixie-thewebsocketprotocol-75 or completely unknown */
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - unsupported version '%d' chosen\n",
ast_sockaddr_stringify(&ser->remote_address), version ? version : 75);
websocket_bad_request(ser);
ao2_ref(protocol_handler, -1);
return 0;
}
/* Enable keepalive on all sessions so the underlying user does not have to */
if (setsockopt(ast_iostream_get_fd(ser->stream), SOL_SOCKET, SO_KEEPALIVE, &flags, sizeof(flags))) {
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - failed to enable keepalive\n",
ast_sockaddr_stringify(&ser->remote_address));
websocket_bad_request(ser);
ao2_ref(session, -1);
ao2_ref(protocol_handler, -1);
return 0;
}
/* Get our local address for the connected socket */
if (ast_getsockname(ast_iostream_get_fd(ser->stream), &session->local_address)) {
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - failed to get local address\n",
ast_sockaddr_stringify(&ser->remote_address));
websocket_bad_request(ser);
ao2_ref(session, -1);
ao2_ref(protocol_handler, -1);
return 0;
}
ast_debug(3, "WebSocket connection from '%s' for protocol '%s' accepted using version '%d'\n", ast_sockaddr_stringify(&ser->remote_address), protocol ? : "", version);
/* Populate the session with all the needed details */
session->stream = ser->stream;
ast_sockaddr_copy(&session->remote_address, &ser->remote_address);
session->opcode = -1;
session->reconstruct = DEFAULT_RECONSTRUCTION_CEILING;
session->secure = ast_iostream_get_ssl(ser->stream) ? 1 : 0;
/* Give up ownership of the socket and pass it to the protocol handler */
ast_iostream_set_exclusive_input(session->stream, 0);
protocol_handler->session_established(session, get_vars, headers);
ao2_ref(protocol_handler, -1);
/*
* By dropping the stream from the session the connection
* won't get closed when the HTTP server cleans up because we
* passed the connection to the protocol handler.
*/
ser->stream = NULL;
return 0;
}
static struct ast_http_uri websocketuri = {
.callback = AST_OPTIONAL_API_NAME(ast_websocket_uri_cb),
.description = "Asterisk HTTP WebSocket",
.uri = "ws",
.has_subtree = 0,
.data = NULL,
.key = __FILE__,
};
/*! \brief Simple echo implementation which echoes received text and binary frames */
static void websocket_echo_callback(struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers)
{
int res;
ast_debug(1, "Entering WebSocket echo loop\n");
if (ast_fd_set_flags(ast_websocket_fd(session), O_NONBLOCK)) {
goto end;
}
while ((res = ast_websocket_wait_for_input(session, -1)) > 0) {
char *payload;
uint64_t payload_len;
enum ast_websocket_opcode opcode;
int fragmented;
if (ast_websocket_read(session, &payload, &payload_len, &opcode, &fragmented)) {
/* We err on the side of caution and terminate the session if any error occurs */
ast_log(LOG_WARNING, "Read failure during WebSocket echo loop\n");
break;
}
if (opcode == AST_WEBSOCKET_OPCODE_TEXT || opcode == AST_WEBSOCKET_OPCODE_BINARY) {
ast_websocket_write(session, opcode, payload, payload_len);
} else if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
break;
} else {
ast_debug(1, "Ignored WebSocket opcode %u\n", opcode);
}
}
end:
ast_debug(1, "Exiting WebSocket echo loop\n");
ast_websocket_unref(session);
}
static int websocket_add_protocol_internal(const char *name, ast_websocket_callback callback)
{
struct ast_websocket_server *ws_server = websocketuri.data;
if (!ws_server) {
return -1;
}
return ast_websocket_server_add_protocol(ws_server, name, callback);
}
int AST_OPTIONAL_API_NAME(ast_websocket_add_protocol)(const char *name, ast_websocket_callback callback)
{
return websocket_add_protocol_internal(name, callback);
}
int AST_OPTIONAL_API_NAME(ast_websocket_add_protocol2)(struct ast_websocket_protocol *protocol)
{
struct ast_websocket_server *ws_server = websocketuri.data;
if (!ws_server) {
return -1;
}
if (ast_websocket_server_add_protocol2(ws_server, protocol)) {
return -1;
}
return 0;
}
static int websocket_remove_protocol_internal(const char *name, ast_websocket_callback callback)
{
struct ast_websocket_server *ws_server = websocketuri.data;
if (!ws_server) {
return -1;
}
return ast_websocket_server_remove_protocol(ws_server, name, callback);
}
int AST_OPTIONAL_API_NAME(ast_websocket_remove_protocol)(const char *name, ast_websocket_callback callback)
{
return websocket_remove_protocol_internal(name, callback);
}
/*! \brief Parse the given uri into a path and remote address.
*
* Expected uri form:
* \verbatim [ws[s]]://<host>[:port][/<path>] \endverbatim
*
* The returned host will contain the address and optional port while
* path will contain everything after the address/port if included.
*/
static int websocket_client_parse_uri(const char *uri, char **host,
struct ast_str **path, char **userinfo, int proxy)
{
struct ast_uri *parsed_uri = proxy ? ast_uri_parse_http(uri) : ast_uri_parse_websocket(uri);
if (!parsed_uri) {
return -1;
}
*host = ast_uri_make_host_with_port(parsed_uri);
*userinfo = ast_strdup(ast_uri_user_info(parsed_uri));
if (ast_uri_path(parsed_uri) || ast_uri_query(parsed_uri)) {
*path = ast_str_create(64);
if (!*path) {
ao2_ref(parsed_uri, -1);
return -1;
}
if (ast_uri_path(parsed_uri)) {
ast_str_set(path, 0, "%s", ast_uri_path(parsed_uri));
}
if (ast_uri_query(parsed_uri)) {
ast_str_append(path, 0, "?%s", ast_uri_query(parsed_uri));
}
}
ao2_ref(parsed_uri, -1);
return 0;
}
static void websocket_client_args_destroy(void *obj)
{
struct ast_tcptls_session_args *args = obj;
if (args->tls_cfg) {
ast_free(args->tls_cfg->certfile);
ast_free(args->tls_cfg->pvtfile);
ast_free(args->tls_cfg->cipher);
ast_free(args->tls_cfg->cafile);
ast_free(args->tls_cfg->capath);
ast_ssl_teardown(args->tls_cfg);
}
ast_free(args->tls_cfg);
}
static struct ast_tcptls_session_args *websocket_client_args_create(struct ast_websocket *ws,
struct ast_websocket_client_options *options, enum ast_websocket_result *result)
{
struct ast_sockaddr *addr;
struct ast_tcptls_session_args *args = ao2_alloc(
sizeof(*args), websocket_client_args_destroy);
const char *resolve_host = NULL;
if (!args) {
*result = WS_ALLOCATE_ERROR;
return NULL;
}
args->accept_fd = -1;
args->tls_cfg = options->tls_cfg;
args->name = "websocket client";
if (!ast_strlen_zero(ws->client->options->proxy_host)) {
resolve_host = ws->client->options->proxy_host;
} else {
resolve_host = ws->client->host;
}
if (!ast_sockaddr_resolve(&addr, resolve_host, 0, 0)) {
ast_log(LOG_ERROR, "Unable to resolve address %s\n",
resolve_host);
ao2_ref(args, -1);
*result = WS_URI_RESOLVE_ERROR;
return NULL;
}
ast_sockaddr_copy(&args->remote_address, addr);
ast_free(addr);
/* We need to save off the hostname but it may contain a port spec */
snprintf(args->hostname, sizeof(args->hostname),
"%.*s",
(int) strcspn(ws->client->host, ":"), ws->client->host);
return args;
}
static char *websocket_client_create_key(void)
{
static int encoded_size = CLIENT_KEY_SIZE * 2 * sizeof(char) + 1;
/* key is randomly selected 16-byte base64 encoded value */
unsigned char key[CLIENT_KEY_SIZE + sizeof(long) - 1];
char *encoded = ast_malloc(encoded_size);
long i = 0;
if (!encoded) {
ast_log(LOG_ERROR, "Unable to allocate client websocket key\n");
return NULL;
}
while (i < CLIENT_KEY_SIZE) {
long num = ast_random();
memcpy(key + i, &num, sizeof(long));
i += sizeof(long);
}
ast_base64encode(encoded, key, CLIENT_KEY_SIZE, encoded_size);
return encoded;
}
static void websocket_client_destroy(void *obj)
{
struct websocket_client *client = obj;
char *id = ast_strdupa(client->options->uri);
SCOPE_ENTER(2, "%s: Client destructor %p\n", id, obj);
ao2_cleanup(client->options);
ao2_cleanup(client->ser);
ao2_cleanup(client->args);
ast_free(client->accept_protocol);
ast_free(client->protocols);
ast_free(client->key);
ast_free(client->resource_name);
ast_free(client->host);
ast_free(client->userinfo);
ast_free(client->proxy_userinfo);
SCOPE_EXIT_RTN("%s: Client destructor complete\n", id);
}
static void client_options_destroy(void *obj)
{
struct ast_websocket_client_options *clone = obj;
ast_free((char *)clone->uri);
ast_free((char *)clone->protocols);
ast_free((char *)clone->username);
ast_free((char *)clone->password);
ast_free((char *)clone->proxy_host);
ast_free((char *)clone->proxy_username);
ast_free((char *)clone->proxy_password);
ast_free(clone->tls_cfg);
}
#define SAFE_STRDUP_WITH_ERROR_RTN(_clone, _str) \
({ \
char *_duped = NULL; \
if (_str) { \
_duped = ast_strdup(_str); \
if (!_duped) { \
ao2_cleanup(_clone); \
return NULL; \
} \
} \
_duped; \
})
static struct ast_websocket_client_options *client_options_clone(
struct ast_websocket_client_options *options)
{
struct ast_websocket_client_options *clone = NULL;
clone = ao2_alloc(sizeof(*clone), client_options_destroy);
if (!clone) {
ast_log(LOG_ERROR, "Unable to clone client options\n");
return NULL;
}
memcpy(clone, options, sizeof(*options));
clone->uri = SAFE_STRDUP_WITH_ERROR_RTN(clone, options->uri);
clone->protocols = SAFE_STRDUP_WITH_ERROR_RTN(clone, options->protocols);
clone->username = SAFE_STRDUP_WITH_ERROR_RTN(clone, options->username);
clone->password = SAFE_STRDUP_WITH_ERROR_RTN(clone, options->password);
clone->proxy_host = SAFE_STRDUP_WITH_ERROR_RTN(clone, options->proxy_host);
clone->proxy_username = SAFE_STRDUP_WITH_ERROR_RTN(clone, options->proxy_username);
clone->proxy_password = SAFE_STRDUP_WITH_ERROR_RTN(clone, options->proxy_password);
if (options->tls_cfg) {
clone->tls_cfg = ast_calloc(1, sizeof(*options->tls_cfg));
if (!clone->tls_cfg) {
ao2_cleanup(clone);
return NULL;
}
memcpy(clone->tls_cfg, options->tls_cfg, sizeof(*options->tls_cfg));
}
return clone;
}
static struct ast_websocket * websocket_client_create(
struct ast_websocket_client_options *options, enum ast_websocket_result *result)
{
struct ast_websocket *ws = NULL;
ast_debug(2, "%s: Creating client\n", options->uri);
ws = ao2_alloc(sizeof(*ws), session_destroy_fn);
if (!ws) {
ast_log(LOG_ERROR, "Unable to allocate websocket\n");
*result = WS_ALLOCATE_ERROR;
return NULL;
}
if (!ast_uuid_generate_str(ws->session_id, sizeof(ws->session_id))) {
ast_log(LOG_ERROR, "Unable to allocate websocket session_id\n");
ao2_ref(ws, -1);
*result = WS_ALLOCATE_ERROR;
return NULL;
}
if (!(ws->client = ao2_alloc(
sizeof(*ws->client), websocket_client_destroy))) {
ast_log(LOG_ERROR, "Unable to allocate websocket client\n");
ao2_ref(ws, -1);
*result = WS_ALLOCATE_ERROR;
return NULL;
}
ws->client->ping_sched_timer = -1;
ws->client->options = client_options_clone(options);
if (!ws->client->options) {
ast_log(LOG_ERROR, "Unable to clone client options\n");
ao2_ref(ws, -1);
*result = WS_ALLOCATE_ERROR;
return NULL;
}
if (!(ws->client->key = websocket_client_create_key())) {
ao2_ref(ws, -1);
*result = WS_KEY_ERROR;
return NULL;
}
if (websocket_client_parse_uri(
options->uri, &ws->client->host, &ws->client->resource_name,
&ws->client->userinfo, 0)) {
ao2_ref(ws, -1);
*result = WS_URI_PARSE_ERROR;
return NULL;
}
ast_debug(2, "%s: host: %s resource: %s userinfo: %s\n", options->uri, ws->client->host,
ws->client->resource_name ? ast_str_buffer(ws->client->resource_name) : "",
ws->client->userinfo);
if (ast_strlen_zero(ws->client->userinfo)
&& !ast_strlen_zero(options->username)
&& !ast_strlen_zero(options->password)) {
ast_asprintf(&ws->client->userinfo, "%s:%s", options->username,
options->password);
}
if (!ast_strlen_zero(options->proxy_host)) {
ast_debug(2, "%s: Proxy host: %s userinfo: %s\n", options->uri, ws->client->options->proxy_host,
ws->client->proxy_userinfo);
if (ast_strlen_zero(ws->client->proxy_userinfo)
&& !ast_strlen_zero(options->proxy_username)
&& !ast_strlen_zero(options->proxy_password)) {
ast_asprintf(&ws->client->proxy_userinfo, "%s:%s", options->proxy_username,
options->proxy_password);
}
}
if (!(ws->client->args = websocket_client_args_create(ws, options, result))) {
ao2_ref(ws, -1);
return NULL;
}
ws->client->suppress_connection_msgs = options->suppress_connection_msgs;
ws->client->args->suppress_connection_msgs = options->suppress_connection_msgs;
ws->client->protocols = ast_strdup(options->protocols);
ws->client->version = 13;
ws->opcode = -1;
ws->reconstruct = DEFAULT_RECONSTRUCTION_CEILING;
ws->timeout = options->timeout;
return ws;
}
const char * AST_OPTIONAL_API_NAME(
ast_websocket_client_accept_protocol)(struct ast_websocket *ws)
{
return ws->client->accept_protocol;
}
static enum ast_websocket_result websocket_client_handle_response_code(
struct websocket_client *client, int response_code, int proxy)
{
if (response_code <= 0) {
return WS_INVALID_RESPONSE;
}
switch (response_code) {
case 101:
if (!proxy) {
return WS_OK;
}
break;
case 200:
if (proxy) {
return WS_OK;
}
break;
case 400:
if (!client->suppress_connection_msgs) {
ast_log(LOG_ERROR, "Received response 400 - Bad Request "
"- from %s\n", client->host);
}
return WS_BAD_REQUEST;
case 401:
if (!client->suppress_connection_msgs) {
ast_log(LOG_ERROR, "Received response 401 - Unauthorized "
"- from %s\n", client->host);
}
return WS_UNAUTHORIZED;
case 404:
if (!client->suppress_connection_msgs) {
ast_log(LOG_ERROR, "Received response 404 - Request URL not "
"found - from %s\n", client->host);
}
return WS_URL_NOT_FOUND;
}
if (!client->suppress_connection_msgs) {
ast_log(LOG_ERROR, "Invalid HTTP response code %d from %s\n",
response_code, proxy ? client->options->proxy_host : client->host);
}
return WS_INVALID_RESPONSE;
}
static enum ast_websocket_result websocket_client_handshake_get_response(
struct websocket_client *client, int proxy)
{
enum ast_websocket_result res;
char buf[4096];
char base64[64];
int has_upgrade = 0;
int has_connection = 0;
int has_accept = 0;
int has_protocol = 0;
int status_code = 0;
SCOPE_ENTER(2, "%s: Proxy? %s Proxy host: %s\n", client->options->uri, AST_YESNO(proxy),
S_OR(client->options->proxy_host, "N/A"));
while (ast_iostream_gets(client->ser->stream, buf, sizeof(buf)) <= 0) {
SCOPE_EXIT_LOG_RTN_VALUE((errno == EINTR || errno == EAGAIN) ? WS_CLIENT_START_ERROR : WS_BAD_STATUS,
LOG_ERROR, "%s: %s waiting for HTTP status line", client->options->uri,
(errno == EINTR || errno == EAGAIN) ? "Timeout" : "Error");
}
status_code = ast_http_response_status_line(buf, "HTTP/1.1", 101);
ast_trace(-1, "%s: Status code: %d\n", client->options->uri, status_code);
res = websocket_client_handle_response_code(client, status_code, proxy);
if (res != WS_OK) {
SCOPE_EXIT_RTN_VALUE(res, "%s: HTTP status line result: %d/%s", client->options->uri,
res, ast_websocket_result_to_str(res));
}
/* Ignoring line folding - assuming header field values are contained
within a single line */
while (1) {
ssize_t len = ast_iostream_gets(client->ser->stream, buf, sizeof(buf));
char *name, *value;
int parsed;
if (len <= 0) {
if (errno == EINTR || errno == EAGAIN) {
SCOPE_EXIT_LOG_RTN_VALUE((errno == EINTR || errno == EAGAIN) ? WS_CLIENT_START_ERROR : WS_BAD_STATUS,
LOG_ERROR, "%s: %s waiting for HTTP header", client->options->uri,
(errno == EINTR || errno == EAGAIN) ? "Timeout" : "Error");
} else {
ast_trace(-1, "%s: Blank line received\n", client->options->uri);
}
break;
}
parsed = ast_http_header_parse(buf, &name, &value);
if (parsed < 0) {
break;
}
if (proxy || parsed > 0) {
continue;
}
if (!has_upgrade &&
(has_upgrade = ast_http_header_match(
name, "upgrade", value, "websocket")) < 0) {
SCOPE_EXIT_RTN_VALUE(WS_HEADER_MISMATCH);
} else if (!has_connection &&
(has_connection = ast_http_header_match(
name, "connection", value, "upgrade")) < 0) {
SCOPE_EXIT_RTN_VALUE(WS_HEADER_MISMATCH);
} else if (!has_accept &&
(has_accept = ast_http_header_match(
name, "sec-websocket-accept", value,
websocket_combine_key(
client->key, base64, sizeof(base64)))) < 0) {
SCOPE_EXIT_RTN_VALUE(WS_HEADER_MISMATCH);
} else if (!has_protocol &&
(has_protocol = ast_http_header_match_in(
name, "sec-websocket-protocol", value, client->protocols))) {
if (has_protocol < 0) {
SCOPE_EXIT_RTN_VALUE(WS_HEADER_MISMATCH);
}
client->accept_protocol = ast_strdup(value);
} else if (!strcasecmp(name, "sec-websocket-extensions")) {
ast_log(LOG_ERROR, "Extensions received, but not "
"supported by client\n");
SCOPE_EXIT_RTN_VALUE(WS_NOT_SUPPORTED);
}
}
if (status_code == 408) {
res = WS_CLIENT_START_ERROR;
} else {
if (proxy) {
res = WS_OK;
} else {
res = has_upgrade && has_connection && has_accept ?
WS_OK : WS_HEADER_MISSING;
}
}
SCOPE_EXIT_RTN_VALUE(res, "%s: Status code: %d\n", client->options->uri, status_code);
}
static void websocket_client_start_handshake_timer(struct websocket_client *client)
{
/*
* ast_iostream_gets (called above in websocket_client_handshake_get_response) is
* a blocking call which means that if the TCP/TLS connection succeeds but the remote doesn't
* actually respond to the proxy CONNECT (if proxy is configured) or GET requests, the process
* can hang indefinitely and escalate to a deadlock. To get ast_iostream_gets to timeout,
* we need to make the following 3 calls on the iostream.
*
* Since the write of the CONNECT and/or GET request is included in the timeout, we'll
* double the timeout set in the websocket client's "connect_timeout" parameter to give
* the server enough time to respond.
*/
ast_iostream_nonblock(client->ser->stream);
ast_iostream_set_exclusive_input(client->ser->stream, 1);
ast_iostream_set_timeout_sequence(client->ser->stream, ast_tvnow(), client->options->timeout * 2);
}
static void websocket_client_stop_handshake_timer(struct websocket_client *client)
{
/*
* Once the handshake is complete, we need to undo what we did in
* websocket_client_start_handshake_timer.
*/
ast_iostream_set_timeout_disable(client->ser->stream);
ast_iostream_set_exclusive_input(client->ser->stream, 0);
ast_iostream_blocking(client->ser->stream);
}
#define optional_header_spec "%s%s%s"
#define print_optional_header(test, name, value) \
test ? name : "", \
test ? value : "", \
test ? "\r\n" : ""
static enum ast_websocket_result websocket_proxy_handshake(
struct websocket_client *client)
{
struct ast_variable *auth_header = NULL;
enum ast_websocket_result res = WS_OK;
size_t bytes_written = 0;
SCOPE_ENTER(2, "%s: Handshaking with proxy %s\n", client->options->uri, client->options->proxy_host);
if (!ast_strlen_zero(client->proxy_userinfo)) {
auth_header = ast_http_create_basic_auth_header(client->proxy_userinfo, NULL);
if (!auth_header) {
SCOPE_EXIT_LOG_RTN_VALUE(WS_ALLOCATE_ERROR, LOG_ERROR, "Unable to allocate client websocket userinfo\n");
}
}
websocket_client_start_handshake_timer(client);
bytes_written = ast_iostream_printf(client->ser->stream,
"CONNECT %s HTTP/1.1\r\n"
"Host: %s\r\n"
optional_header_spec
"Proxy-Connection: Keep-Alive\r\n"
"\r\n",
client->host,
client->host,
print_optional_header(auth_header, "Proxy-Authorization: ", auth_header->value)
);
ast_variables_destroy(auth_header);
if (bytes_written < 0) {
websocket_client_stop_handshake_timer(client);
SCOPE_EXIT_LOG_RTN_VALUE(WS_WRITE_ERROR, LOG_ERROR, "Failed to send handshake\n");
}
/* wait for a response before doing anything else */
res = websocket_client_handshake_get_response(client, 1);
websocket_client_stop_handshake_timer(client);
SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_websocket_result_to_str(res));
}
static enum ast_websocket_result websocket_client_handshake(
struct websocket_client *client)
{
size_t protocols_len = 0;
struct ast_variable *auth_header = NULL;
size_t res;
SCOPE_ENTER(2, "%s: Handshaking with server\n", client->options->uri);
if (!ast_strlen_zero(client->userinfo)) {
auth_header = ast_http_create_basic_auth_header(client->userinfo, NULL);
if (!auth_header) {
SCOPE_EXIT_LOG_RTN_VALUE(WS_ALLOCATE_ERROR, LOG_ERROR, "Unable to allocate client websocket userinfo\n");
}
}
protocols_len = client->protocols ? strlen(client->protocols) : 0;
websocket_client_start_handshake_timer(client);
res = ast_iostream_printf(client->ser->stream,
"GET /%s HTTP/1.1\r\n"
"Sec-WebSocket-Version: %d\r\n"
"Upgrade: websocket\r\n"
"Connection: Upgrade\r\n"
"Host: %s\r\n"
optional_header_spec
optional_header_spec
"Sec-WebSocket-Key: %s\r\n"
"\r\n",
client->resource_name ? ast_str_buffer(client->resource_name) : "",
client->version,
client->host,
print_optional_header(auth_header, "Authorization: ", auth_header->value),
print_optional_header(protocols_len, "Sec-WebSocket-Protocol: ", client->protocols),
client->key
);
ast_variables_destroy(auth_header);
if (res < 0) {
websocket_client_stop_handshake_timer(client);
SCOPE_EXIT_LOG_RTN_VALUE(WS_WRITE_ERROR, LOG_ERROR, "Failed to send handshake\n");
}
/* wait for a response before doing anything else */
res = websocket_client_handshake_get_response(client, 0);
websocket_client_stop_handshake_timer(client);
SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_websocket_result_to_str(res));
}
static enum ast_websocket_result websocket_client_connect(struct ast_websocket *ws,
struct ast_websocket_client_options *options)
{
enum ast_websocket_result res;
int original_tls_enabled = ws->client->args->tls_cfg ? ws->client->args->tls_cfg->enabled : 0;
int proxy = !ast_strlen_zero(ws->client->options->proxy_host);
SCOPE_ENTER(2, "%s: proxy: %s tls_enabled: %s\n", options->uri, S_OR(ws->client->options->proxy_host, "N/A"),
AST_YESNO(original_tls_enabled));
/* create and connect the client - note client_start
releases the session instance on failure */
if (proxy && original_tls_enabled ) {
ast_trace(-1, "%s: Disabling TLS while handshaking with proxy\n", options->uri);
if (ws->client->args->tls_cfg) {
ws->client->args->tls_cfg->enabled = 0;
}
}
ast_trace(-1, "%s: Creating tcptls client\n", options->uri);
ws->client->ser = ast_tcptls_client_create(ws->client->args);
if (!ws->client->ser) {
SCOPE_EXIT_LOG_RTN_VALUE(WS_CLIENT_START_ERROR, LOG_ERROR, "%s: Unable to create tcptls client\n", options->uri);
}
ast_trace(-1, "%s: Connecting%s%s\n", options->uri, proxy ? " via proxy " : "", S_OR(ws->client->options->proxy_host, ""));
if (!ast_tcptls_client_start_timeout(ws->client->ser, options->timeout)) {
ws->client->ser = NULL;
SCOPE_EXIT_LOG_RTN_VALUE(WS_CLIENT_START_ERROR, LOG_ERROR, "%s: Unable to connect%s%s\n", options->uri,
proxy ? " via proxy " : "", S_OR(ws->client->options->proxy_host, ""));
}
ast_trace(-1, "%s: Connected%s%s\n", options->uri, proxy ? " via proxy " : "", S_OR(ws->client->options->proxy_host, ""));
if (proxy) {
res = websocket_proxy_handshake(ws->client);
if (res != WS_OK) {
ao2_ref(ws->client->ser, -1);
ws->client->ser = NULL;
SCOPE_EXIT_LOG_RTN_VALUE(res, LOG_ERROR, "%s: Unable to perform proxy handshake with %s\n", options->uri,
ws->client->options->proxy_host);
}
}
if (proxy && original_tls_enabled && !ws->client->args->tls_cfg->enabled) {
int rc = 0;
ast_trace(-1, "%s: Re-enabling TLS after handshaking with proxy %s\n", options->uri, ws->client->options->proxy_host);
ws->client->args->tls_cfg->enabled = 1;
rc = ast_ssl_setup_client(ws->client->args->tls_cfg);
if (rc != 1) {
ao2_cleanup(ws->client->ser);
ws->client->ser = NULL;
SCOPE_EXIT_LOG_RTN_VALUE(WS_TLS_ERROR, LOG_ERROR, "%s: TLS context setup failed after handshake with %s\n",
options->uri, ws->client->options->proxy_host);
}
if (!ast_tcptls_start_tls(ws->client->ser)) {
ws->client->ser = NULL;
SCOPE_EXIT_LOG_RTN_VALUE(WS_TLS_ERROR, LOG_ERROR, "%s: TLS with websocket server failed after handshake with %s\n",
options->uri, ws->client->options->proxy_host);
}
}
if ((res = websocket_client_handshake(ws->client)) != WS_OK) {
ao2_ref(ws->client->ser, -1);
ws->client->ser = NULL;
SCOPE_EXIT_LOG_RTN_VALUE(res, LOG_ERROR, "%s: Unable to perform websocket handshake\n", options->uri);
}
ws->stream = ws->client->ser->stream;
ws->secure = ast_iostream_get_ssl(ws->stream) ? 1 : 0;
ws->client->ser->stream = NULL;
ast_sockaddr_copy(&ws->remote_address, &ws->client->ser->remote_address);
SCOPE_EXIT_RTN_VALUE(WS_OK, "%s: Handshake complete\n", options->uri);
}
struct ast_websocket *AST_OPTIONAL_API_NAME(ast_websocket_client_create)
(const char *uri, const char *protocols, struct ast_tls_config *tls_cfg,
enum ast_websocket_result *result)
{
struct ast_websocket_client_options options = {
.uri = uri,
.protocols = protocols,
.timeout = -1,
.tls_cfg = tls_cfg,
};
return ast_websocket_client_create_with_options(&options, result);
}
static int ping_scheduler_callback(const void *obj)
{
struct ast_websocket *session = (struct ast_websocket *)obj;
if (!session->client) {
/*
* We should never get here because we can only start pingpongs from a client
* but just in case...
*/
return 0;
}
ao2_lock(session);
if (session->closing) {
ao2_unlock(session);
return 0;
}
if (session->client->missed_pong_count > 1) {
ast_debug(2, "%s: Missed PONG count is now %d\n", WS_SESSION_REMOTE(session),
session->client->missed_pong_count);
}
if (session->client->missed_pong_count >= session->client->options->pingpong_probes) {
ao2_unlock(session);
ast_log(LOG_WARNING, "%s: %d missed PONGs. Closing connection.\n",
WS_SESSION_REMOTE(session), session->client->missed_pong_count);
session->client->ping_sched_timer = -1;
websocket_close(session, AST_WEBSOCKET_STATUS_GOING_AWAY, 1);
return 0;
}
ast_websocket_write(session, AST_WEBSOCKET_OPCODE_PING, WS_PING_PAYLOAD, WS_PING_PAYLOAD_LEN);
session->client->missed_pong_count++;
ao2_unlock(session);
return session->client->options->pingpong_interval * 1000;
}
/*
* Notes on client session lifecycle:
*
* Historically, the lifecycle of a client session was fairly simple. Ownership was
* transferred to the caller with the return of the create and when the caller released
* their last reference, the session was closed by the destructor. There was one issue
* with this however, if the remote end sent us a CLOSE opcode, we were destroying
* everything without sending the required CLOSE reply. This could cause issues on
* the remote end. The addition of WebSocket PING/PONG capability also doesn't work well
* with that pattern because it adds a scheduler and callback which means that a reference
* must be held by this module as long as the scheduler is active. This means that a caller
* can't just unref the websocket and expect it to be automatically closed.
*
* So now...
*
* Once the websocket is created and connected, the usual pattern is for the higher
* level client to be in a loop that blocks waiting on a websocket frame to be
* available. The client owns a reference to the websocket while the loop is
* active and if we're sending PINGs, the scheduled task also holds a reference.
*
* When the higher level client wants to close the websocket, it calls
* ast_websocket_close() from another thread. We mark the session as CLOSED_BY_US,
* stop the scheduled PING task and release its reference, send a CLOSE opcode to the
* remote, then return to the caller. The client's read thread is still blocked
* at this point. When we receive the CLOSE reply from the remote, we use the
* CLOSED_BY_US flag to indicate that we're done and can close the socket and stream.
* This causes the client's read thread to unblock with a CLOSE opcode which then calls
* ast_websocket_close() (which is a NoOp because we already did the close) and
* ast_websocket_unref() which triggers the session destructor.
*
* When the remote sends us a CLOSE opcode, we mark the session as CLOSED_BY_REMOTE
* and return the CLOSE opcode in the ast_websocket_read that the caller should have
* been blocked on. The caller must then call ast_websocket_close() just as above.
* In this case however, it's not a NoOp. We stop the scheduled PING task and
* release its reference, send a CLOSE reply to the remote and since the transaction
* is done, we close the socket and stream and return to the caller. The caller then
* calls ast_websocket_unref() which triggers the session destructor.
*/
struct ast_websocket *AST_OPTIONAL_API_NAME(ast_websocket_client_create_with_options)
(struct ast_websocket_client_options *options, enum ast_websocket_result *result)
{
struct ast_websocket *ws = NULL;
SCOPE_ENTER(2, "%s: Creating client\n", options->uri);
ws = websocket_client_create(options, result);
if (!ws) {
SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR, "%s: Failed to create: %s\n", options->uri, ast_websocket_result_to_str(*result));
}
ast_trace(-1, "%s: Connecting\n", ws->client->options->uri);
if ((*result = websocket_client_connect(ws, options)) != WS_OK) {
ao2_ref(ws, -1);
SCOPE_EXIT_RTN_VALUE(NULL, "%s: Failed to connect: %s\n", options->uri, ast_websocket_result_to_str(*result));
}
ast_trace(-1, "%s: Connected\n", ws->client->options->uri);
if (ws->client->options->tcp_keepalives) {
int sockfd = ast_iostream_get_fd(ws->stream);
int enabled = 1;
setsockopt(sockfd, SOL_SOCKET, SO_KEEPALIVE, &enabled, sizeof(enabled));
setsockopt(sockfd, IPPROTO_TCP, TCP_KEEPIDLE, &options->tcp_keepalive_time, sizeof(options->tcp_keepalive_time));
setsockopt(sockfd, IPPROTO_TCP, TCP_KEEPINTVL, &options->tcp_keepalive_interval, sizeof(&options->tcp_keepalive_interval));
setsockopt(sockfd, IPPROTO_TCP, TCP_KEEPCNT, &options->tcp_keepalive_probes, sizeof(&options->tcp_keepalive_probes));
ast_trace(-1, "%s: Enabled TCP keepalives\n", ws->client->options->uri);
}
if (ws->client->options->pingpongs) {
ws->client->ping_sched_timer = ast_sched_add(ping_scheduler, ws->client->options->pingpong_interval * 1000,
ping_scheduler_callback, ao2_bump(ws));
if (ws->client->ping_sched_timer < 0) {
ast_log(LOG_WARNING, "%s: Unable to schedule PING/PONG keepalives\n", ws->client->options->uri);
ao2_ref(ws, -1);
} else {
ast_trace(-1, "%s: Enabled PING/PONG keepalives\n", ws->client->options->uri);
}
}
SCOPE_EXIT_RTN_VALUE(ws, "%s: Client created and connected %p %p\n", options->uri, ws, ws->client);
}
int AST_OPTIONAL_API_NAME(ast_websocket_read_string)
(struct ast_websocket *ws, char **buf)
{
char *payload;
uint64_t payload_len;
enum ast_websocket_opcode opcode;
int fragmented = 1;
while (fragmented) {
if (ast_websocket_read(ws, &payload, &payload_len,
&opcode, &fragmented)) {
ast_log(LOG_ERROR, "Client WebSocket string read - "
"error reading string data\n");
return -1;
}
if (opcode == AST_WEBSOCKET_OPCODE_PING) {
/* Try read again, we have sent pong already */
fragmented = 1;
continue;
}
if (opcode == AST_WEBSOCKET_OPCODE_CONTINUATION) {
continue;
}
if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
return -1;
}
if (opcode != AST_WEBSOCKET_OPCODE_TEXT) {
ast_log(LOG_ERROR, "Client WebSocket string read - "
"non string data received\n");
return -1;
}
}
if (!(*buf = ast_strndup(payload, payload_len))) {
return -1;
}
return payload_len + 1;
}
int AST_OPTIONAL_API_NAME(ast_websocket_write_string)
(struct ast_websocket *ws, const char *buf)
{
uint64_t len = strlen(buf);
ast_debug(3, "Writing websocket string of length %" PRIu64 "\n", len);
/* We do not pass strlen(buf) to ast_websocket_write() directly because the
* size_t returned by strlen() may not require the same storage size
* as the uint64_t that ast_websocket_write() uses. This normally
* would not cause a problem, but since ast_websocket_write() uses
* the optional API, this function call goes through a series of macros
* that may cause a 32-bit to 64-bit conversion to go awry.
*/
return ast_websocket_write(ws, AST_WEBSOCKET_OPCODE_TEXT,
(char *)buf, len);
}
const char *websocket_result_string_map[] = {
[WS_OK] = "OK",
[WS_ALLOCATE_ERROR] = "Allocation error",
[WS_KEY_ERROR] = "Key error",
[WS_URI_PARSE_ERROR] = "URI parse error",
[WS_URI_RESOLVE_ERROR] = "URI resolve error",
[WS_BAD_STATUS] = "Bad status line",
[WS_INVALID_RESPONSE] = "Invalid response code",
[WS_BAD_REQUEST] = "Bad request",
[WS_URL_NOT_FOUND] = "URL not found",
[WS_HEADER_MISMATCH] = "Header mismatch",
[WS_HEADER_MISSING] = "Header missing",
[WS_NOT_SUPPORTED] = "Not supported",
[WS_WRITE_ERROR] = "Write error",
[WS_CLIENT_START_ERROR] = "Client start error",
[WS_UNAUTHORIZED] = "Unauthorized"
};
const char *AST_OPTIONAL_API_NAME(ast_websocket_result_to_str)
(enum ast_websocket_result result)
{
if (!ARRAY_IN_BOUNDS(result, websocket_result_string_map)) {
return "unknown";
}
return websocket_result_string_map[result];
}
struct status_map {
enum ast_websocket_status_code code;
const char *desc;
};
static const struct status_map websocket_status_map[] = {
{ AST_WEBSOCKET_STATUS_NORMAL, "Normal" },
{ AST_WEBSOCKET_STATUS_GOING_AWAY, "Going away" },
{ AST_WEBSOCKET_STATUS_PROTOCOL_ERROR, "Protocol error" },
{ AST_WEBSOCKET_STATUS_UNSUPPORTED_DATA, "Unsupported data" },
{ AST_WEBSOCKET_STATUS_RESERVED_1004, "reserved 1004" },
{ AST_WEBSOCKET_STATUS_RESERVED_1005, "reserved 1005" },
{ AST_WEBSOCKET_STATUS_RESERVED_1006, "reserved 1006" },
{ AST_WEBSOCKET_STATUS_INVALID_FRAME, "Invalid frame" },
{ AST_WEBSOCKET_STATUS_POLICY_VIOLATION, "Policy violation" },
{ AST_WEBSOCKET_STATUS_TOO_BIG, "Data too big" },
{ AST_WEBSOCKET_STATUS_MANDATORY_EXT, "Mandatory extension" },
{ AST_WEBSOCKET_STATUS_INTERNAL_ERROR, "Internal error" },
{ AST_WEBSOCKET_STATUS_RESERVED_1012, "reserved 1012" },
{ AST_WEBSOCKET_STATUS_RESERVED_1013, "reserved 1013" },
{ AST_WEBSOCKET_STATUS_BAD_GATEWAY, "Bad gateway" },
{ AST_WEBSOCKET_STATUS_RESERVED_1015, "reserved 1015" },
};
const char *AST_OPTIONAL_API_NAME(ast_websocket_status_to_str)
(enum ast_websocket_status_code code)
{
int i;
for (i = 0; i < ARRAY_LEN(websocket_status_map); i++) {
if (websocket_status_map[i].code == code)
return websocket_status_map[i].desc;
}
return "Unknown";
}
static int unload_module(void)
{
if (ping_scheduler) {
ast_sched_context_destroy(ping_scheduler);
ping_scheduler = NULL;
}
websocket_remove_protocol_internal("echo", websocket_echo_callback);
ast_http_uri_unlink(&websocketuri);
ao2_ref(websocketuri.data, -1);
websocketuri.data = NULL;
return 0;
}
static int load_module(void)
{
websocketuri.data = websocket_server_internal_create();
if (!websocketuri.data) {
return AST_MODULE_LOAD_DECLINE;
}
ast_http_uri_link(&websocketuri);
websocket_add_protocol_internal("echo", websocket_echo_callback);
ping_scheduler = ast_sched_context_create();
if (!ping_scheduler) {
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
if (ast_sched_start_thread(ping_scheduler)) {
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "HTTP WebSocket Support",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
.requires = "http",
);
Reference in New Issue View Git Blame Copy Permalink