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	Expression 'rlen < 0' is always false. Unsigned type value is never < 0. Change-Id: Id9f393ff25b009a6c4a6e40b95f561a9369e4585
		
			
				
	
	
		
			1466 lines
		
	
	
		
			44 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			1466 lines
		
	
	
		
			44 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
 | |
|  * 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/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 Size of the pre-determined buffer for WebSocket frames */
 | |
| #define MAXIMUM_FRAME_SIZE 8192
 | |
| 
 | |
| /*! \brief Default reconstruction size for multi-frame payload reconstruction. If exceeded the next frame will start a
 | |
|  *         payload.
 | |
|  */
 | |
| #define DEFAULT_RECONSTRUCTION_CEILING 8192
 | |
| 
 | |
| /*! \brief Maximum reconstruction size for multi-frame payload reconstruction. */
 | |
| #define MAXIMUM_RECONSTRUCTION_CEILING 8192
 | |
| #else
 | |
| /*! \brief Size of the pre-determined buffer for WebSocket frames */
 | |
| #define MAXIMUM_FRAME_SIZE 32768
 | |
| 
 | |
| /*! \brief Default reconstruction size for multi-frame payload reconstruction. If exceeded the next frame will start a
 | |
|  *         payload.
 | |
|  */
 | |
| #define DEFAULT_RECONSTRUCTION_CEILING 32768
 | |
| 
 | |
| /*! \brief Maximum reconstruction size for multi-frame payload reconstruction. */
 | |
| #define MAXIMUM_RECONSTRUCTION_CEILING 32768
 | |
| #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 Structure definition for session */
 | |
| struct ast_websocket {
 | |
| 	struct ast_iostream *stream;       /*!< iostream of the connection */
 | |
| 	struct ast_sockaddr address;       /*!< Address of the remote client */
 | |
| 	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 */
 | |
| 	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 */
 | |
| };
 | |
| 
 | |
| /*! \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_internal_dtor(void *obj)
 | |
| {
 | |
| 	struct ast_websocket_server *server = obj;
 | |
| 	ao2_cleanup(server->protocols);
 | |
| 	server->protocols = NULL;
 | |
| }
 | |
| 
 | |
| static void websocket_server_dtor(void *obj)
 | |
| {
 | |
| 	websocket_server_internal_dtor(obj);
 | |
| 	ast_module_unref(ast_module_info->self);
 | |
| }
 | |
| 
 | |
| static struct ast_websocket_server *websocket_server_create_impl(void (*dtor)(void *))
 | |
| {
 | |
| 	RAII_VAR(struct ast_websocket_server *, server, NULL, ao2_cleanup);
 | |
| 
 | |
| 	server = ao2_alloc(sizeof(*server), dtor);
 | |
| 	if (!server) {
 | |
| 		return NULL;
 | |
| 	}
 | |
| 
 | |
| 	server->protocols = ao2_container_alloc(MAX_PROTOCOL_BUCKETS, protocol_hash_fn, 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(websocket_server_internal_dtor);
 | |
| }
 | |
| 
 | |
| struct ast_websocket_server *AST_OPTIONAL_API_NAME(ast_websocket_server_create)(void)
 | |
| {
 | |
| 	ast_module_ref(ast_module_info->self);
 | |
| 	return websocket_server_create_impl(websocket_server_dtor);
 | |
| }
 | |
| 
 | |
| /*! \brief Destructor function for sessions */
 | |
| static void session_destroy_fn(void *obj)
 | |
| {
 | |
| 	struct ast_websocket *session = obj;
 | |
| 
 | |
| 	if (session->stream) {
 | |
| 		ast_websocket_close(session, 0);
 | |
| 		if (session->stream) {
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| 			ast_iostream_close(session->stream);
 | |
| 			session->stream = NULL;
 | |
| 			ast_verb(2, "WebSocket connection %s '%s' closed\n", session->client ? "to" : "from",
 | |
| 				ast_sockaddr_stringify(&session->address));
 | |
| 		}
 | |
| 	}
 | |
| 
 | |
| 	ao2_cleanup(session->client);
 | |
| 	ast_free(session->payload);
 | |
| }
 | |
| 
 | |
| 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;
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| 
 | |
| 	if (ast_websocket_server_add_protocol2(server, protocol)) {
 | |
| 		ao2_ref(protocol, -1);
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| 		return -1;
 | |
| 	}
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| 
 | |
| 	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",
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| 			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_verb(2, "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_verb(2, "WebSocket unregistered sub-protocol '%s'\n", name);
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| /*! \brief Close function for websocket session */
 | |
| int AST_OPTIONAL_API_NAME(ast_websocket_close)(struct ast_websocket *session, uint16_t reason)
 | |
| {
 | |
| 	char frame[4] = { 0, }; /* The header is 2 bytes and the reason code takes up another 2 bytes */
 | |
| 	int res;
 | |
| 
 | |
| 	if (session->close_sent) {
 | |
| 		return 0;
 | |
| 	}
 | |
| 
 | |
| 	frame[0] = AST_WEBSOCKET_OPCODE_CLOSE | 0x80;
 | |
| 	frame[1] = 2; /* The reason code is always 2 bytes */
 | |
| 
 | |
| 	/* If no reason has been specified assume 1000 which is normal closure */
 | |
| 	put_unaligned_uint16(&frame[2], htons(reason ? reason : 1000));
 | |
| 
 | |
| 	session->closing = 1;
 | |
| 	session->close_sent = 1;
 | |
| 
 | |
| 	ao2_lock(session);
 | |
| 	ast_iostream_set_timeout_inactivity(session->stream, session->timeout);
 | |
| 	res = ast_iostream_write(session->stream, frame, sizeof(frame));
 | |
| 	ast_iostream_set_timeout_disable(session->stream);
 | |
| 
 | |
| 	/* If an error occurred when trying to close this connection explicitly terminate it now.
 | |
| 	 * Doing so will cause the thread polling on it to wake up and terminate.
 | |
| 	 */
 | |
| 	if (res != sizeof(frame)) {
 | |
| 		ast_iostream_close(session->stream);
 | |
| 		session->stream = NULL;
 | |
| 		ast_verb(2, "WebSocket connection %s '%s' forcefully closed due to fatal write error\n",
 | |
| 			session->client ? "to" : "from", ast_sockaddr_stringify(&session->address));
 | |
| 	}
 | |
| 
 | |
| 	ao2_unlock(session);
 | |
| 	return res == sizeof(frame);
 | |
| }
 | |
| 
 | |
| 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 (opcode < AST_WEBSOCKET_OPCODE_CONTINUATION ||
 | |
| 			opcode > AST_WEBSOCKET_OPCODE_PONG) {
 | |
| 		return "<unknown>";
 | |
| 	} else {
 | |
| 		return opcode_map[opcode];
 | |
| 	}
 | |
| }
 | |
| 
 | |
| /*! \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;
 | |
| 
 | |
| 	ast_debug(3, "Writing websocket %s frame, length %" PRIu64 "\n",
 | |
| 			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;
 | |
| 	}
 | |
| 
 | |
| 	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);
 | |
| 
 | |
| 	ao2_lock(session);
 | |
| 	if (session->closing) {
 | |
| 		ao2_unlock(session);
 | |
| 		return -1;
 | |
| 	}
 | |
| 
 | |
| 	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_debug(1, "Closing WS with 1011 because we can't fulfill a write request\n");
 | |
| 		ast_websocket_close(session, 1011);
 | |
| 		return -1;
 | |
| 	}
 | |
| 
 | |
| 	ast_iostream_set_timeout_disable(session->stream);
 | |
| 	ao2_unlock(session);
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| 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)
 | |
| {
 | |
| 	ao2_ref(session, +1);
 | |
| }
 | |
| 
 | |
| void AST_OPTIONAL_API_NAME(ast_websocket_unref)(struct ast_websocket *session)
 | |
| {
 | |
| 	ao2_cleanup(session);
 | |
| }
 | |
| 
 | |
| int AST_OPTIONAL_API_NAME(ast_websocket_fd)(struct ast_websocket *session)
 | |
| {
 | |
| 	return session->closing ? -1 : ast_iostream_get_fd(session->stream);
 | |
| }
 | |
| 
 | |
| struct ast_sockaddr * AST_OPTIONAL_API_NAME(ast_websocket_remote_address)(struct ast_websocket *session)
 | |
| {
 | |
| 	return &session->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, int len, enum ast_websocket_opcode *opcode)
 | |
| {
 | |
| 	ssize_t rlen;
 | |
| 	int xlen = len;
 | |
| 	char *rbuf = buf;
 | |
| 	int sanity = 10;
 | |
| 
 | |
| 	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;
 | |
| 			}
 | |
| 		}
 | |
| 		xlen = xlen - rlen;
 | |
| 		rbuf = rbuf + rlen;
 | |
| 		if (!xlen) {
 | |
| 			break;
 | |
| 		}
 | |
| 		if (ast_wait_for_input(ast_iostream_get_fd(session->stream), 1000) < 0) {
 | |
| 			ast_log(LOG_ERROR, "ast_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)
 | |
| {
 | |
| 	char buf[MAXIMUM_FRAME_SIZE] = "";
 | |
| 	int fin = 0;
 | |
| 	int mask_present = 0;
 | |
| 	char *mask = NULL, *new_payload = NULL;
 | |
| 	size_t options_len = 0, frame_size = 0;
 | |
| 
 | |
| 	*payload = NULL;
 | |
| 	*payload_len = 0;
 | |
| 	*fragmented = 0;
 | |
| 
 | |
| 	if (ws_safe_read(session, &buf[0], MIN_WS_HDR_SZ, opcode)) {
 | |
| 		return -1;
 | |
| 	}
 | |
| 	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 = buf[0] & 0xf;
 | |
| 	*payload_len = 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) {
 | |
| 		fin = (buf[0] >> 7) & 1;
 | |
| 		mask_present = (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, &buf[frame_size], options_len, opcode)) {
 | |
| 				return -1;
 | |
| 			}
 | |
| 			frame_size += options_len;
 | |
| 		}
 | |
| 
 | |
| 		if (*payload_len == 126) {
 | |
| 			/* Grab the 2-byte payload length  */
 | |
| 			*payload_len = ntohs(get_unaligned_uint16(&buf[2]));
 | |
| 			mask = &buf[4];
 | |
| 		} else if (*payload_len == 127) {
 | |
| 			/* Grab the 8-byte payload length  */
 | |
| 			*payload_len = ntohl(get_unaligned_uint64(&buf[2]));
 | |
| 			mask = &buf[10];
 | |
| 		} else {
 | |
| 			/* Just set the mask after the small 2-byte header */
 | |
| 			mask = &buf[2];
 | |
| 		}
 | |
| 
 | |
| 		/* Now read the rest of the payload */
 | |
| 		*payload = &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 > MAXIMUM_FRAME_SIZE) {
 | |
| 			ast_log(LOG_WARNING, "Cannot fit huge websocket frame of %zu bytes\n", frame_size);
 | |
| 			/* The frame won't fit :-( */
 | |
| 			ast_websocket_close(session, 1009);
 | |
| 			return -1;
 | |
| 		}
 | |
| 
 | |
| 		if (ws_safe_read(session, *payload, *payload_len, opcode)) {
 | |
| 			return -1;
 | |
| 		}
 | |
| 		/* 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) && (ast_websocket_write(session, AST_WEBSOCKET_OPCODE_PONG, *payload, *payload_len))) {
 | |
| 			*payload_len = 0;
 | |
| 			ast_websocket_close(session, 1009);
 | |
| 			return 0;
 | |
| 		}
 | |
| 
 | |
| 		if (*payload_len) {
 | |
| 			if (!(new_payload = ast_realloc(session->payload, (session->payload_len + *payload_len)))) {
 | |
| 				ast_log(LOG_WARNING, "Failed allocation: %p, %zu, %"PRIu64"\n",
 | |
| 					session->payload, session->payload_len, *payload_len);
 | |
| 				*payload_len = 0;
 | |
| 				ast_websocket_close(session, 1009);
 | |
| 				return -1;
 | |
| 			}
 | |
| 
 | |
| 			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 if (*opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
 | |
| 		/* Make the payload available so the user can look at the reason code if they so desire */
 | |
| 		if ((*payload_len) && (new_payload = ast_realloc(session->payload, *payload_len))) {
 | |
| 			if (ws_safe_read(session, &buf[frame_size], (*payload_len), opcode)) {
 | |
| 				return -1;
 | |
| 			}
 | |
| 			session->payload = new_payload;
 | |
| 			memcpy(session->payload, &buf[frame_size], *payload_len);
 | |
| 			*payload = session->payload;
 | |
| 			frame_size += (*payload_len);
 | |
| 		}
 | |
| 
 | |
| 		session->closing = 1;
 | |
| 	} 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);
 | |
| 	}
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| /*!
 | |
|  * \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;
 | |
| 	char *upgrade = NULL, *key = NULL, *key1 = NULL, *key2 = NULL, *protos = NULL, *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 = ast_strip(ast_strdupa(v->value));
 | |
| 		} else if (!strcasecmp(v->name, "Sec-WebSocket-Key")) {
 | |
| 			key = ast_strip(ast_strdupa(v->value));
 | |
| 		} else if (!strcasecmp(v->name, "Sec-WebSocket-Key1")) {
 | |
| 			key1 = ast_strip(ast_strdupa(v->value));
 | |
| 		} else if (!strcasecmp(v->name, "Sec-WebSocket-Key2")) {
 | |
| 			key2 = ast_strip(ast_strdupa(v->value));
 | |
| 		} else if (!strcasecmp(v->name, "Sec-WebSocket-Protocol")) {
 | |
| 			requested_protocols = ast_strip(ast_strdupa(v->value));
 | |
| 			protos = ast_strdupa(requested_protocols);
 | |
| 		} 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(requested_protocols)) {
 | |
| 		/* 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;
 | |
| 	}
 | |
| 
 | |
| 	/* 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(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(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;
 | |
| 	}
 | |
| 
 | |
| 	ast_verb(2, "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->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 flags, res;
 | |
| 
 | |
| 	ast_debug(1, "Entering WebSocket echo loop\n");
 | |
| 
 | |
| 	if ((flags = fcntl(ast_websocket_fd(session), F_GETFL)) == -1) {
 | |
| 		goto end;
 | |
| 	}
 | |
| 
 | |
| 	flags |= O_NONBLOCK;
 | |
| 
 | |
| 	if (fcntl(ast_websocket_fd(session), F_SETFL, flags) == -1) {
 | |
| 		goto end;
 | |
| 	}
 | |
| 
 | |
| 	while ((res = ast_wait_for_input(ast_websocket_fd(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)
 | |
| {
 | |
| 	int res = websocket_add_protocol_internal(name, callback);
 | |
| 	if (res == 0) {
 | |
| 		ast_module_ref(ast_module_info->self);
 | |
| 	}
 | |
| 	return res;
 | |
| }
 | |
| 
 | |
| 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;
 | |
| 	}
 | |
| 
 | |
| 	ast_module_ref(ast_module_info->self);
 | |
| 	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)
 | |
| {
 | |
| 	int res = websocket_remove_protocol_internal(name, callback);
 | |
| 	if (res == 0) {
 | |
| 		ast_module_unref(ast_module_info->self);
 | |
| 	}
 | |
| 	return res;
 | |
| }
 | |
| 
 | |
| /*! \brief Parse the given uri into a path and remote address.
 | |
|  *
 | |
|  * Expected uri form: [ws[s]]://<host>[:port][/<path>]
 | |
|  *
 | |
|  * 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)
 | |
| {
 | |
| 	struct ast_uri *parsed_uri = ast_uri_parse_websocket(uri);
 | |
| 
 | |
| 	if (!parsed_uri) {
 | |
| 		return -1;
 | |
| 	}
 | |
| 
 | |
| 	*host = ast_uri_make_host_with_port(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(
 | |
| 	const char *host, struct ast_tls_config *tls_cfg,
 | |
| 	enum ast_websocket_result *result)
 | |
| {
 | |
| 	struct ast_sockaddr *addr;
 | |
| 	struct ast_tcptls_session_args *args = ao2_alloc(
 | |
| 		sizeof(*args), websocket_client_args_destroy);
 | |
| 
 | |
| 	if (!args) {
 | |
| 		*result = WS_ALLOCATE_ERROR;
 | |
| 		return NULL;
 | |
| 	}
 | |
| 
 | |
| 	args->accept_fd = -1;
 | |
| 	args->tls_cfg = tls_cfg;
 | |
| 	args->name = "websocket client";
 | |
| 
 | |
| 	if (!ast_sockaddr_resolve(&addr, host, 0, 0)) {
 | |
| 		ast_log(LOG_ERROR, "Unable to resolve address %s\n",
 | |
| 			host);
 | |
| 		ao2_ref(args, -1);
 | |
| 		*result = WS_URI_RESOLVE_ERROR;
 | |
| 		return NULL;
 | |
| 	}
 | |
| 	ast_sockaddr_copy(&args->remote_address, addr);
 | |
| 	ast_free(addr);
 | |
| 	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;
 | |
| }
 | |
| 
 | |
| struct websocket_client {
 | |
| 	/*! 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;
 | |
| };
 | |
| 
 | |
| static void websocket_client_destroy(void *obj)
 | |
| {
 | |
| 	struct websocket_client *client = obj;
 | |
| 
 | |
| 	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);
 | |
| }
 | |
| 
 | |
| static struct ast_websocket * websocket_client_create(
 | |
| 	const char *uri, const char *protocols,	struct ast_tls_config *tls_cfg,
 | |
| 	enum ast_websocket_result *result)
 | |
| {
 | |
| 	struct ast_websocket *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 (!(ws->client = ao2_alloc(
 | |
| 		      sizeof(*ws->client), websocket_client_destroy))) {
 | |
| 		ast_log(LOG_ERROR, "Unable to allocate websocket client\n");
 | |
| 		*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(
 | |
| 		    uri, &ws->client->host, &ws->client->resource_name)) {
 | |
| 		ao2_ref(ws, -1);
 | |
| 		*result = WS_URI_PARSE_ERROR;
 | |
| 		return NULL;
 | |
| 	}
 | |
| 
 | |
| 	if (!(ws->client->args = websocket_client_args_create(
 | |
| 		      ws->client->host, tls_cfg, result))) {
 | |
| 		ao2_ref(ws, -1);
 | |
| 		return NULL;
 | |
| 	}
 | |
| 	ws->client->protocols = ast_strdup(protocols);
 | |
| 
 | |
| 	ws->client->version = 13;
 | |
| 	ws->opcode = -1;
 | |
| 	ws->reconstruct = DEFAULT_RECONSTRUCTION_CEILING;
 | |
| 	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)
 | |
| {
 | |
| 	if (response_code <= 0) {
 | |
| 		return WS_INVALID_RESPONSE;
 | |
| 	}
 | |
| 
 | |
| 	switch (response_code) {
 | |
| 	case 101:
 | |
| 		return 0;
 | |
| 	case 400:
 | |
| 		ast_log(LOG_ERROR, "Received response 400 - Bad Request "
 | |
| 			"- from %s\n", client->host);
 | |
| 		return WS_BAD_REQUEST;
 | |
| 	case 404:
 | |
| 		ast_log(LOG_ERROR, "Received response 404 - Request URL not "
 | |
| 			"found - from %s\n", client->host);
 | |
| 		return WS_URL_NOT_FOUND;
 | |
| 	}
 | |
| 
 | |
| 	ast_log(LOG_ERROR, "Invalid HTTP response code %d from %s\n",
 | |
| 		response_code, client->host);
 | |
| 	return WS_INVALID_RESPONSE;
 | |
| }
 | |
| 
 | |
| static enum ast_websocket_result websocket_client_handshake_get_response(
 | |
| 	struct websocket_client *client)
 | |
| {
 | |
| 	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;
 | |
| 
 | |
| 	if (ast_iostream_gets(client->ser->stream, buf, sizeof(buf)) <= 0) {
 | |
| 		ast_log(LOG_ERROR, "Unable to retrieve HTTP status line.");
 | |
| 		return WS_BAD_STATUS;
 | |
| 	}
 | |
| 
 | |
| 	if ((res = websocket_client_handle_response_code(client,
 | |
| 		    ast_http_response_status_line(
 | |
| 			    buf, "HTTP/1.1", 101))) != WS_OK) {
 | |
| 		return res;
 | |
| 	}
 | |
| 
 | |
| 	/* Ignoring line folding - assuming header field values are contained
 | |
| 	   within a single line */
 | |
| 	while (ast_iostream_gets(client->ser->stream, buf, sizeof(buf)) > 0) {
 | |
| 		char *name, *value;
 | |
| 		int parsed = ast_http_header_parse(buf, &name, &value);
 | |
| 
 | |
| 		if (parsed < 0) {
 | |
| 			break;
 | |
| 		}
 | |
| 
 | |
| 		if (parsed > 0) {
 | |
| 			continue;
 | |
| 		}
 | |
| 
 | |
| 		if (!has_upgrade &&
 | |
| 		    (has_upgrade = ast_http_header_match(
 | |
| 			    name, "upgrade", value, "websocket")) < 0) {
 | |
| 			return WS_HEADER_MISMATCH;
 | |
| 		} else if (!has_connection &&
 | |
| 			   (has_connection = ast_http_header_match(
 | |
| 				   name, "connection", value, "upgrade")) < 0) {
 | |
| 			return 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) {
 | |
| 			return 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) {
 | |
| 				return 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");
 | |
| 			return WS_NOT_SUPPORTED;
 | |
| 		}
 | |
| 	}
 | |
| 	return has_upgrade && has_connection && has_accept ?
 | |
| 		WS_OK : WS_HEADER_MISSING;
 | |
| }
 | |
| 
 | |
| static enum ast_websocket_result websocket_client_handshake(
 | |
| 	struct websocket_client *client)
 | |
| {
 | |
| 	char protocols[100] = "";
 | |
| 
 | |
| 	if (!ast_strlen_zero(client->protocols)) {
 | |
| 		sprintf(protocols, "Sec-WebSocket-Protocol: %s\r\n",
 | |
| 			client->protocols);
 | |
| 	}
 | |
| 
 | |
| 	if (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"
 | |
| 			"Sec-WebSocket-Key: %s\r\n"
 | |
| 			"%s\r\n",
 | |
| 			client->resource_name ? ast_str_buffer(client->resource_name) : "",
 | |
| 			client->version,
 | |
| 			client->host,
 | |
| 			client->key,
 | |
| 			protocols) < 0) {
 | |
| 		ast_log(LOG_ERROR, "Failed to send handshake.\n");
 | |
| 		return WS_WRITE_ERROR;
 | |
| 	}
 | |
| 	/* wait for a response before doing anything else */
 | |
| 	return websocket_client_handshake_get_response(client);
 | |
| }
 | |
| 
 | |
| static enum ast_websocket_result websocket_client_connect(struct ast_websocket *ws)
 | |
| {
 | |
| 	enum ast_websocket_result res;
 | |
| 	/* create and connect the client - note client_start
 | |
| 	   releases the session instance on failure */
 | |
| 	if (!(ws->client->ser = ast_tcptls_client_start(
 | |
| 		      ast_tcptls_client_create(ws->client->args)))) {
 | |
| 		return WS_CLIENT_START_ERROR;
 | |
| 	}
 | |
| 
 | |
| 	if ((res = websocket_client_handshake(ws->client)) != WS_OK) {
 | |
| 		ao2_ref(ws->client->ser, -1);
 | |
| 		ws->client->ser = NULL;
 | |
| 		return res;
 | |
| 	}
 | |
| 
 | |
| 	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->address, &ws->client->ser->remote_address);
 | |
| 	return WS_OK;
 | |
| }
 | |
| 
 | |
| 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 *ws = websocket_client_create(
 | |
| 		uri, protocols, tls_cfg, result);
 | |
| 
 | |
| 	if (!ws) {
 | |
| 		return NULL;
 | |
| 	}
 | |
| 
 | |
| 	if ((*result = websocket_client_connect(ws)) != WS_OK) {
 | |
| 		ao2_ref(ws, -1);
 | |
| 		return NULL;
 | |
| 	}
 | |
| 
 | |
| 	return ws;
 | |
| }
 | |
| 
 | |
| 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_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_malloc(payload_len + 1))) {
 | |
| 		return -1;
 | |
| 	}
 | |
| 
 | |
| 	ast_copy_string(*buf, payload, payload_len + 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);
 | |
| }
 | |
| 
 | |
| static int load_module(void)
 | |
| {
 | |
| 	websocketuri.data = websocket_server_internal_create();
 | |
| 	if (!websocketuri.data) {
 | |
| 		return AST_MODULE_LOAD_FAILURE;
 | |
| 	}
 | |
| 	ast_http_uri_link(&websocketuri);
 | |
| 	websocket_add_protocol_internal("echo", websocket_echo_callback);
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| static int unload_module(void)
 | |
| {
 | |
| 	websocket_remove_protocol_internal("echo", websocket_echo_callback);
 | |
| 	ast_http_uri_unlink(&websocketuri);
 | |
| 	ao2_ref(websocketuri.data, -1);
 | |
| 	websocketuri.data = NULL;
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "HTTP WebSocket Support",
 | |
| 	.support_level = AST_MODULE_SUPPORT_EXTENDED,
 | |
| 	.load = load_module,
 | |
| 	.unload = unload_module,
 | |
| 	.load_pri = AST_MODPRI_CHANNEL_DEPEND,
 | |
| );
 |