Files
asterisk/include/asterisk/http_websocket.h
Matthew Jordan 365ae7523b res_http_websocket: Close websocket correctly and use careful fwrite
When a client takes a long time to process information received from Asterisk,
a write operation using fwrite may fail to write all information. This causes
the underlying file stream to be in an unknown state, such that the socket
must be disconnected. Unfortunately, there are two problems with this in
Asterisk's existing websocket code:
1. Periodically, during the read loop, Asterisk must write to the connected
   websocket to respond to pings. As such, Asterisk maintains a reference to
   the session during the loop. When ast_http_websocket_write fails, it may
   cause the session to decrement its ref count, but this in and of itself
   does not break the read loop. The read loop's write, on the other hand,
   does not break the loop if it fails. This causes the socket to get in a
   'stuck' state, preventing the client from reconnecting to the server.
2. More importantly, however, is that the fwrite in ast_http_websocket_write
   fails with a large volume of data when the client takes awhile to process
   the information. When it does fail, it fails writing only a portion of
   the bytes. With some debugging, it was shown that this was failing in a
   similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite
   with a long enough timeout solved the problem.

Note that this version of the patch, unlike r417310 in Asterisk 11, exposes
configuration options beyond just chan_sip's sip.conf. Configuration options
to configure the write timeout have also been added to pjsip.conf and ari.conf.

#ASTERISK-23917 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3624/
........

Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-26 12:21:14 +00:00

346 lines
12 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.
*/
#ifndef _ASTERISK_HTTP_WEBSOCKET_H
#define _ASTERISK_HTTP_WEBSOCKET_H
#include "asterisk/http.h"
#include "asterisk/optional_api.h"
#include <errno.h>
/*! \brief Default websocket write timeout, in ms */
#define AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT 100
/*! \brief Default websocket write timeout, in ms (as a string) */
#define AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT_STR "100"
/*!
* \file http_websocket.h
* \brief Support for WebSocket connections within the Asterisk HTTP server and client
* WebSocket connections to a server.
*
* Supported WebSocket versions in server implementation:
* Version 7 defined in specification http://tools.ietf.org/html/draft-ietf-hybi-thewebsocketprotocol-07
* Version 8 defined in specification http://tools.ietf.org/html/draft-ietf-hybi-thewebsocketprotocol-10
* Version 13 defined in specification http://tools.ietf.org/html/rfc6455
* Supported WebSocket versions in client implementation:
* Version 13 defined in specification http://tools.ietf.org/html/rfc6455
*
* \author Joshua Colp <jcolp@digium.com>
*
*/
/*! \brief WebSocket operation codes */
enum ast_websocket_opcode {
AST_WEBSOCKET_OPCODE_TEXT = 0x1, /*!< Text frame */
AST_WEBSOCKET_OPCODE_BINARY = 0x2, /*!< Binary frame */
AST_WEBSOCKET_OPCODE_PING = 0x9, /*!< Request that the other side respond with a pong */
AST_WEBSOCKET_OPCODE_PONG = 0xA, /*!< Response to a ping */
AST_WEBSOCKET_OPCODE_CLOSE = 0x8, /*!< Connection is being closed */
AST_WEBSOCKET_OPCODE_CONTINUATION = 0x0, /*!< Continuation of a previous frame */
};
/*!
* \brief Opaque structure for WebSocket server.
* \since 12
*/
struct ast_websocket_server;
/*!
* \brief Opaque structure for WebSocket sessions.
*/
struct ast_websocket;
/*!
* \brief Callback for when a new connection for a sub-protocol is established
*
* \param session A WebSocket session structure
* \param parameters Parameters extracted from the request URI
* \param headers Headers included in the request
*
* \note Once called the ownership of the session is transferred to the sub-protocol handler. It
* is responsible for closing and cleaning up.
*
*/
typedef void (*ast_websocket_callback)(struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers);
/*!
* \brief Creates a \ref websocket_server
*
* \retval New \ref websocket_server instance
* \retval \c NULL on error
* \since 12
*/
AST_OPTIONAL_API(struct ast_websocket_server *, ast_websocket_server_create, (void), { return NULL; });
/*!
* \brief Callback suitable for use with a \ref ast_http_uri.
*
* Set the data field of the ast_http_uri to \ref ast_websocket_server.
* \since 12
*/
AST_OPTIONAL_API(int, 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), { return -1; });
/*!
* \brief Add a sub-protocol handler to the default /ws server
*
* \param name Name of the sub-protocol to register
* \param callback Callback called when a new connection requesting the sub-protocol is established
*
* \retval 0 success
* \retval -1 if sub-protocol handler could not be registered
*/
AST_OPTIONAL_API(int, ast_websocket_add_protocol, (const char *name, ast_websocket_callback callback), {return -1;});
/*!
* \brief Remove a sub-protocol handler from the default /ws server.
*
* \param name Name of the sub-protocol to unregister
* \param callback Callback that was previously registered with the sub-protocol
*
* \retval 0 success
* \retval -1 if sub-protocol was not found or if callback did not match
*/
AST_OPTIONAL_API(int, ast_websocket_remove_protocol, (const char *name, ast_websocket_callback callback), {return -1;});
/*!
* \brief Add a sub-protocol handler to the given server.
*
* \param name Name of the sub-protocol to register
* \param callback Callback called when a new connection requesting the sub-protocol is established
*
* \retval 0 success
* \retval -1 if sub-protocol handler could not be registered
* \since 12
*/
AST_OPTIONAL_API(int, ast_websocket_server_add_protocol, (struct ast_websocket_server *server, const char *name, ast_websocket_callback callback), {return -1;});
/*!
* \brief Remove a sub-protocol handler from the given server.
*
* \param name Name of the sub-protocol to unregister
* \param callback Callback that was previously registered with the sub-protocol
*
* \retval 0 success
* \retval -1 if sub-protocol was not found or if callback did not match
* \since 12
*/
AST_OPTIONAL_API(int, ast_websocket_server_remove_protocol, (struct ast_websocket_server *server, const char *name, ast_websocket_callback callback), {return -1;});
/*!
* \brief Read a WebSocket frame and handle it
*
* \param session Pointer to the WebSocket session
* \param payload Pointer to a char* which will be populated with a pointer to the payload if present
* \param payload_len Pointer to a uint64_t which will be populated with the length of the payload if present
* \param opcode Pointer to an enum which will be populated with the opcode of the frame
* \param fragmented Pointer to an int which is set to 1 if payload is fragmented and 0 if not
*
* \retval -1 on error
* \retval 0 on success
*
* \note Once an AST_WEBSOCKET_OPCODE_CLOSE opcode is received the socket will be closed
*/
AST_OPTIONAL_API(int, ast_websocket_read, (struct ast_websocket *session, char **payload, uint64_t *payload_len, enum ast_websocket_opcode *opcode, int *fragmented), { errno = ENOSYS; return -1;});
/*!
* \brief Read a WebSocket frame containing string data.
*
* \note The caller is responsible for freeing the output "buf".
*
* \param ws pointer to the websocket
* \param buf string buffer to populate with data read from socket
* \retval -1 on error
* \retval number of bytes read on success
*
* \note Once an AST_WEBSOCKET_OPCODE_CLOSE opcode is received the socket will be closed
*/
AST_OPTIONAL_API(int, ast_websocket_read_string,
(struct ast_websocket *ws, char **buf),
{ errno = ENOSYS; return -1;});
/*!
* \brief Construct and transmit a WebSocket frame
*
* \param session Pointer to the WebSocket session
* \param opcode WebSocket operation code to place in the frame
* \param payload Optional pointer to a payload to add to the frame
* \param actual_length Length of the payload (0 if no payload)
*
* \retval 0 if successfully written
* \retval -1 if error occurred
*/
AST_OPTIONAL_API(int, ast_websocket_write, (struct ast_websocket *session, enum ast_websocket_opcode opcode, char *payload, uint64_t actual_length), { errno = ENOSYS; return -1;});
/*!
* \brief Construct and transmit a WebSocket frame containing string data.
*
* \param ws pointer to the websocket
* \param buf string data to write to socket
* \retval 0 if successfully written
* \retval -1 if error occurred
*/
AST_OPTIONAL_API(int, ast_websocket_write_string,
(struct ast_websocket *ws, const char *buf),
{ errno = ENOSYS; return -1;});
/*!
* \brief Close a WebSocket session by sending a message with the CLOSE opcode and an optional code
*
* \param session Pointer to the WebSocket session
* \param reason Reason code for closing the session as defined in the RFC
*
* \retval 0 if successfully written
* \retval -1 if error occurred
*/
AST_OPTIONAL_API(int, ast_websocket_close, (struct ast_websocket *session, uint16_t reason), { errno = ENOSYS; return -1;});
/*!
* \brief Enable multi-frame reconstruction up to a certain number of bytes
*
* \param session Pointer to the WebSocket session
* \param bytes If a reconstructed payload exceeds the specified number of bytes the payload will be returned
* and upon reception of the next multi-frame a new reconstructed payload will begin.
*/
AST_OPTIONAL_API(void, ast_websocket_reconstruct_enable, (struct ast_websocket *session, size_t bytes), {return;});
/*!
* \brief Disable multi-frame reconstruction
*
* \param session Pointer to the WebSocket session
*
* \note If reconstruction is disabled each message that is part of a multi-frame message will be sent up to
* the user when ast_websocket_read is called.
*/
AST_OPTIONAL_API(void, ast_websocket_reconstruct_disable, (struct ast_websocket *session), {return;});
/*!
* \brief Increase the reference count for a WebSocket session
*
* \param session Pointer to the WebSocket session
*/
AST_OPTIONAL_API(void, ast_websocket_ref, (struct ast_websocket *session), {return;});
/*!
* \brief Decrease the reference count for a WebSocket session
*
* \param session Pointer to the WebSocket session
*/
AST_OPTIONAL_API(void, ast_websocket_unref, (struct ast_websocket *session), {return;});
/*!
* \brief Get the file descriptor for a WebSocket session.
*
* \retval file descriptor
*
* \note You must *not* directly read from or write to this file descriptor. It should only be used for polling.
*/
AST_OPTIONAL_API(int, ast_websocket_fd, (struct ast_websocket *session), { errno = ENOSYS; return -1;});
/*!
* \brief Get the remote address for a WebSocket connected session.
*
* \retval ast_sockaddr Remote address
*/
AST_OPTIONAL_API(struct ast_sockaddr *, ast_websocket_remote_address, (struct ast_websocket *session), {return NULL;});
/*!
* \brief Get whether the WebSocket session is using a secure transport or not.
*
* \retval 0 if unsecure
* \retval 1 if secure
*/
AST_OPTIONAL_API(int, ast_websocket_is_secure, (struct ast_websocket *session), { errno = ENOSYS; return -1;});
/*!
* \brief Set the socket of a WebSocket session to be non-blocking.
*
* \retval 0 on success
* \retval -1 on failure
*/
AST_OPTIONAL_API(int, ast_websocket_set_nonblock, (struct ast_websocket *session), { errno = ENOSYS; return -1;});
/*!
* \brief Result code for a websocket client.
*/
enum ast_websocket_result {
WS_OK,
WS_ALLOCATE_ERROR,
WS_KEY_ERROR,
WS_URI_PARSE_ERROR,
WS_URI_RESOLVE_ERROR,
WS_BAD_STATUS,
WS_INVALID_RESPONSE,
WS_BAD_REQUEST,
WS_URL_NOT_FOUND,
WS_HEADER_MISMATCH,
WS_HEADER_MISSING,
WS_NOT_SUPPORTED,
WS_WRITE_ERROR,
WS_CLIENT_START_ERROR,
};
/*!
* \brief Create, and connect, a websocket client.
*
* \detail If the client websocket successfully connects, then the accepted protocol
* can be checked via a call to ast_websocket_client_accept_protocol.
*
* \note While connecting this *will* block until a response is
* received from the remote host.
* \note Expected uri form: ws[s]://<address>[:port][/<path>] The address (can be a
* host name) and port are parsed out and used to connect to the remote server.
* If multiple IPs are returned during address resolution then the first one is
* chosen.
*
* \param uri uri to connect to
* \param protocols a comma separated string of supported protocols
* \param tls_cfg secure websocket credentials
* \param result result code set on client failure
* \retval a client websocket.
* \retval NULL if object could not be created or connected
* \since 13
*/
AST_OPTIONAL_API(struct ast_websocket *, ast_websocket_client_create,
(const char *uri, const char *protocols,
struct ast_tls_config *tls_cfg,
enum ast_websocket_result *result), { return NULL;});
/*!
* \brief Retrieve the server accepted sub-protocol on the client.
*
* \param ws the websocket client
* \retval the accepted client sub-protocol.
* \since 13
*/
AST_OPTIONAL_API(const char *, ast_websocket_client_accept_protocol,
(struct ast_websocket *ws), { return NULL;});
/*!
* \brief Set the timeout on a non-blocking WebSocket session.
*
* \since 11.11.0
* \since 12.4.0
*
* \retval 0 on success
* \retval -1 on failure
*/
AST_OPTIONAL_API(int, ast_websocket_set_timeout, (struct ast_websocket *session, int timeout), {return -1;});
#endif