mirror of https://github.com/asterisk/asterisk
Review: https://reviewboard.asterisk.org/r/1952/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368359 65c4cc65-6c06-0410-ace0-fbb531ad65f3certified/11.2
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2012, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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#ifndef _ASTERISK_HTTP_WEBSOCKET_H
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#define _ASTERISK_HTTP_WEBSOCKET_H
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#include "asterisk/module.h"
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/*!
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* \file http_websocket.h
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* \brief Support for WebSocket connections within the Asterisk HTTP server.
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*
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* \author Joshua Colp <jcolp@digium.com>
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*
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*/
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/*! \brief WebSocket operation codes */
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enum ast_websocket_opcode {
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AST_WEBSOCKET_OPCODE_TEXT = 0x1, /*!< Text frame */
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AST_WEBSOCKET_OPCODE_BINARY = 0x2, /*!< Binary frame */
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AST_WEBSOCKET_OPCODE_PING = 0x9, /*!< Request that the other side respond with a pong */
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AST_WEBSOCKET_OPCODE_PONG = 0xA, /*!< Response to a ping */
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AST_WEBSOCKET_OPCODE_CLOSE = 0x8, /*!< Connection is being closed */
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AST_WEBSOCKET_OPCODE_CONTINUATION = 0x0, /*!< Continuation of a previous frame */
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};
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/*!
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* \brief Opaque structure for WebSocket sessions
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*/
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struct ast_websocket;
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/*!
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* \brief Callback for when a new connection for a sub-protocol is established
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*
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* \param session A WebSocket session structure
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* \param parameters Parameters extracted from the request URI
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* \param headers Headers included in the request
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*
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* \note Once called the ownership of the session is transferred to the sub-protocol handler. It
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* is responsible for closing and cleaning up.
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*
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*/
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typedef void (*ast_websocket_callback)(struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers);
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/*!
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* \brief Add a sub-protocol handler to the server
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*
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* \param name Name of the sub-protocol to register
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* \param callback Callback called when a new connection requesting the sub-protocol is established
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*
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* \retval 0 success
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* \retval -1 if sub-protocol handler could not be registered
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*/
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int ast_websocket_add_protocol(const char *name, ast_websocket_callback callback);
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/*!
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* \brief Remove a sub-protocol handler from the server
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*
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* \param name Name of the sub-protocol to unregister
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* \param callback Callback that was previously registered with the sub-protocol
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*
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* \retval 0 success
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* \retval -1 if sub-protocol was not found or if callback did not match
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*/
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int ast_websocket_remove_protocol(const char *name, ast_websocket_callback callback);
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/*!
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* \brief Read a WebSocket frame and handle it
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*
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* \param session Pointer to the WebSocket session
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* \param payload Pointer to a char* which will be populated with a pointer to the payload if present
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* \param payload_len Pointer to a uint64_t which will be populated with the length of the payload if present
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* \param opcode Pointer to an enum which will be populated with the opcode of the frame
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* \param fragmented Pointer to an int which is set to 1 if payload is fragmented and 0 if not
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*
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* \retval -1 on error
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* \retval 0 on success
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*
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* \note Once an AST_WEBSOCKET_OPCODE_CLOSE opcode is received the socket will be closed
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*/
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int ast_websocket_read(struct ast_websocket *session, char **payload, uint64_t *payload_len, enum ast_websocket_opcode *opcode, int *fragmented);
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/*!
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* \brief Construct and transmit a WebSocket frame
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*
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* \param session Pointer to the WebSocket session
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* \param opcode WebSocket operation code to place in the frame
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* \param payload Optional pointer to a payload to add to the frame
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* \param actual_length Length of the payload (0 if no payload)
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*
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* \retval 0 if successfully written
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* \retval -1 if error occurred
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*/
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int ast_websocket_write(struct ast_websocket *session, enum ast_websocket_opcode opcode, char *payload, uint64_t actual_length);
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/*!
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* \brief Close a WebSocket session by sending a message with the CLOSE opcode and an optional code
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*
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* \param session Pointer to the WebSocket session
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* \param reason Reason code for closing the session as defined in the RFC
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*
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* \retval 0 if successfully written
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* \retval -1 if error occurred
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*/
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int ast_websocket_close(struct ast_websocket *session, uint16_t reason);
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/*!
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* \brief Enable multi-frame reconstruction up to a certain number of bytes
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*
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* \param session Pointer to the WebSocket session
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* \param bytes If a reconstructed payload exceeds the specified number of bytes the payload will be returned
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* and upon reception of the next multi-frame a new reconstructed payload will begin.
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*/
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void ast_websocket_reconstruct_enable(struct ast_websocket *session, size_t bytes);
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/*!
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* \brief Disable multi-frame reconstruction
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*
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* \param session Pointer to the WebSocket session
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*
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* \note If reconstruction is disabled each message that is part of a multi-frame message will be sent up to
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* the user when ast_websocket_read is called.
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*/
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void ast_websocket_reconstruct_disable(struct ast_websocket *session);
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/*!
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* \brief Increase the reference count for a WebSocket session
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*
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* \param session Pointer to the WebSocket session
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*/
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void ast_websocket_ref(struct ast_websocket *session);
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/*!
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* \brief Decrease the reference count for a WebSocket session
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*
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* \param session Pointer to the WebSocket session
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*/
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void ast_websocket_unref(struct ast_websocket *session);
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/*!
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* \brief Get the file descriptor for a WebSocket session.
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*
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* \retval file descriptor
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*
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* \note You must *not* directly read from or write to this file descriptor. It should only be used for polling.
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*/
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int ast_websocket_fd(struct ast_websocket *session);
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/*!
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* \brief Get the remote address for a WebSocket connected session.
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*
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* \retval ast_sockaddr Remote address
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*/
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struct ast_sockaddr *ast_websocket_remote_address(struct ast_websocket *session);
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/*!
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* \brief Get whether the WebSocket session is using a secure transport or not.
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*
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* \retval 0 if unsecure
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* \retval 1 if secure
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*/
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int ast_websocket_is_secure(struct ast_websocket *session);
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#endif /* _ASTERISK_HTTP_WEBSOCKET_H */
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2012, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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#ifndef _ASTERISK_HTTP_WEBSOCKET_H
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#define _ASTERISK_HTTP_WEBSOCKET_H
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#include "asterisk/module.h"
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/*!
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* \file http_websocket.h
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* \brief Support for WebSocket connections within the Asterisk HTTP server.
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*
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* \author Joshua Colp <jcolp@digium.com>
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*
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*/
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/*! \brief WebSocket operation codes */
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enum ast_websocket_opcode {
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AST_WEBSOCKET_OPCODE_TEXT = 0x1, /*!< Text frame */
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AST_WEBSOCKET_OPCODE_BINARY = 0x2, /*!< Binary frame */
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AST_WEBSOCKET_OPCODE_PING = 0x9, /*!< Request that the other side respond with a pong */
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AST_WEBSOCKET_OPCODE_PONG = 0xA, /*!< Response to a ping */
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AST_WEBSOCKET_OPCODE_CLOSE = 0x8, /*!< Connection is being closed */
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AST_WEBSOCKET_OPCODE_CONTINUATION = 0x0, /*!< Continuation of a previous frame */
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};
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/*!
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* \brief Callback for when a new connection for a sub-protocol is established
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*
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* \param f Pointer to the file instance for the session
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* \param fd File descriptor for the session
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* \param remote_address The address of the remote party
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*
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* \note Once called the ownership of the session is transferred to the sub-protocol handler. It
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* is responsible for closing and cleaning up.
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*
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*/
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typedef void (*ast_websocket_callback)(FILE *f, int fd, struct ast_sockaddr *remote_address);
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/*!
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* \brief Add a sub-protocol handler to the server
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*
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* \param name Name of the sub-protocol to register
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* \param callback Callback called when a new connection requesting the sub-protocol is established
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*
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* \retval 0 success
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* \retval -1 if sub-protocol handler could not be registered
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*/
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int ast_websocket_add_protocol(char *name, ast_websocket_callback callback);
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/*!
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* \brief Remove a sub-protocol handler from the server
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*
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* \param name Name of the sub-protocol to unregister
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* \param callback Callback that was previously registered with the sub-protocol
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*
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* \retval 0 success
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* \retval -1 if sub-protocol was not found or if callback did not match
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*/
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int ast_websocket_remove_protocol(char *name, ast_websocket_callback callback);
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/*!
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* \brief Read a WebSocket frame and handle it
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*
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* \param f Pointer to the file stream, used to respond to certain frames
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* \param buf Pointer to the buffer containing the frame
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* \param buflen Size of the buffer
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* \param payload_len Pointer to a uint64_t which will be populated with the length of the payload if present
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* \param opcode Pointer to an int which will be populated with the opcode of the frame
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*
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* \retval NULL if no payload is present
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* \retval non-NULL if payload is present, returned pointer points to beginning of payload
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*/
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char *ast_websocket_read(FILE *f, char *buf, size_t buflen, uint64_t *payload_len, int *opcode);
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/*!
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* \brief Construct and transmit a WebSocket frame
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*
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* \param f Pointer to the file stream which the frame will be sent on
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* \param opcode WebSocket operation code to place in the frame
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* \param payload Optional pointer to a payload to add to the frame
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* \param actual_length Length of the payload (0 if no payload)
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*/
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void ast_websocket_write(FILE *f, int op_code, char *payload, uint64_t actual_length);
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#endif /* _ASTERISK_HTTP_WEBSOCKET_H */
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2012, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief WebSocket support for the Asterisk internal HTTP server
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*
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* \author Joshua Colp <jcolp@digium.com>
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*/
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/*** MODULEINFO
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<support_level>extended</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/module.h"
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#include "asterisk/http.h"
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#include "asterisk/astobj2.h"
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#include "asterisk/strings.h"
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#include "asterisk/file.h"
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#include "asterisk/unaligned.h"
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#include "asterisk/http_websocket.h"
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/*! \brief GUID used to compute the accept key, defined in the specifications */
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#define WEBSOCKET_GUID "258EAFA5-E914-47DA-95CA-C5AB0DC85B11"
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/*! \brief Number of buckets for registered protocols */
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#define MAX_PROTOCOL_BUCKETS 7
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/*! \brief Size of the pre-determined buffer for WebSocket frames */
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#define MAXIMUM_FRAME_SIZE 8192
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/*! \brief Default reconstruction size for multi-frame payload reconstruction. If exceeded the next frame will start a
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* payload.
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*/
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#define DEFAULT_RECONSTRUCTION_CEILING 16384
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/*! \brief Maximum reconstruction size for multi-frame payload reconstruction. */
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#define MAXIMUM_RECONSTRUCTION_CEILING 16384
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/*! \brief Structure definition for session */
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struct ast_websocket {
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FILE *f; /*!< Pointer to the file instance used for writing and reading */
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int fd; /*!< File descriptor for the session, only used for polling */
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struct ast_sockaddr address; /*!< Address of the remote client */
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enum ast_websocket_opcode opcode; /*!< Cached opcode for multi-frame messages */
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size_t payload_len; /*!< Length of the payload */
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char *payload; /*!< Pointer to the payload */
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size_t reconstruct; /*!< Number of bytes before a reconstructed payload will be returned and a new one started */
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unsigned int secure:1; /*!< Bit to indicate that the transport is secure */
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unsigned int closing:1; /*!< Bit to indicate that the session is in the process of being closed */
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};
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/*! \brief Structure definition for protocols */
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struct websocket_protocol {
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char *name; /*!< Name of the protocol */
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ast_websocket_callback callback; /*!< Callback called when a new session is established */
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};
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/*! \brief Container for registered protocols */
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static struct ao2_container *protocols;
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/*! \brief Hashing function for protocols */
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static int protocol_hash_fn(const void *obj, const int flags)
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{
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const struct websocket_protocol *protocol = obj;
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const char *name = obj;
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return ast_str_case_hash(flags & OBJ_KEY ? name : protocol->name);
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}
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/*! \brief Comparison function for protocols */
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static int protocol_cmp_fn(void *obj, void *arg, int flags)
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{
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const struct websocket_protocol *protocol1 = obj, *protocol2 = arg;
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const char *protocol = arg;
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return !strcasecmp(protocol1->name, flags & OBJ_KEY ? protocol : protocol2->name) ? CMP_MATCH | CMP_STOP : 0;
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}
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/*! \brief Destructor function for protocols */
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static void protocol_destroy_fn(void *obj)
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{
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struct websocket_protocol *protocol = obj;
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ast_free(protocol->name);
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}
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/*! \brief Destructor function for sessions */
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static void session_destroy_fn(void *obj)
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{
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struct ast_websocket *session = obj;
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if (session->f) {
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fclose(session->f);
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ast_verb(2, "WebSocket connection from '%s' closed\n", ast_sockaddr_stringify(&session->address));
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}
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ast_free(session->payload);
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}
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int ast_websocket_add_protocol(const char *name, ast_websocket_callback callback)
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{
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struct websocket_protocol *protocol;
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ao2_lock(protocols);
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/* Ensure a second protocol handler is not registered for the same protocol */
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if ((protocol = ao2_find(protocols, name, OBJ_KEY | OBJ_NOLOCK))) {
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ao2_ref(protocol, -1);
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ao2_unlock(protocols);
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return -1;
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}
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||||
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if (!(protocol = ao2_alloc(sizeof(*protocol), protocol_destroy_fn))) {
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ao2_unlock(protocols);
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return -1;
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||||
}
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||||
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||||
if (!(protocol->name = ast_strdup(name))) {
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ao2_ref(protocol, -1);
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ao2_unlock(protocols);
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return -1;
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}
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protocol->callback = callback;
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ao2_link_flags(protocols, protocol, OBJ_NOLOCK);
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ao2_unlock(protocols);
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ao2_ref(protocol, -1);
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ast_verb(2, "WebSocket registered sub-protocol '%s'\n", name);
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return 0;
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}
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int ast_websocket_remove_protocol(const char *name, ast_websocket_callback callback)
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{
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struct websocket_protocol *protocol;
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if (!(protocol = ao2_find(protocols, name, OBJ_KEY))) {
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return -1;
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}
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if (protocol->callback != callback) {
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ao2_ref(protocol, -1);
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return -1;
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}
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ao2_unlink(protocols, protocol);
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ao2_ref(protocol, -1);
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||||
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ast_verb(2, "WebSocket unregistered sub-protocol '%s'\n", name);
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||||
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return 0;
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}
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/*! \brief Close function for websocket session */
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||||
int ast_websocket_close(struct ast_websocket *session, uint16_t reason)
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||||
{
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char frame[4] = { 0, }; /* The header is 2 bytes and the reason code takes up another 2 bytes */
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||||
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||||
frame[0] = AST_WEBSOCKET_OPCODE_CLOSE | 0x80;
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frame[1] = 2; /* The reason code is always 2 bytes */
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||||
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||||
/* If no reason has been specified assume 1000 which is normal closure */
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||||
put_unaligned_uint16(&frame[2], htons(reason ? reason : 1000));
|
||||
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||||
session->closing = 1;
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||||
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||||
return (fwrite(frame, 1, 4, session->f) == 4) ? 0 : -1;
|
||||
}
|
||||
|
||||
|
||||
/*! \brief Write function for websocket traffic */
|
||||
int ast_websocket_write(struct ast_websocket *session, enum ast_websocket_opcode opcode, char *payload, uint64_t actual_length)
|
||||
{
|
||||
size_t header_size = 2; /* The minimum size of a websocket frame is 2 bytes */
|
||||
char *frame;
|
||||
uint64_t length = 0;
|
||||
|
||||
if (actual_length < 126) {
|
||||
length = actual_length;
|
||||
} else if (actual_length < (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 = alloca(header_size);
|
||||
memset(frame, 0, sizeof(*frame));
|
||||
|
||||
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(actual_length));
|
||||
} else if (length == 127) {
|
||||
put_unaligned_uint64(&frame[2], htonl(actual_length));
|
||||
}
|
||||
|
||||
if (fwrite(frame, 1, header_size, session->f) != header_size) {
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (fwrite(payload, 1, actual_length, session->f) != actual_length) {
|
||||
return -1;
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
void ast_websocket_reconstruct_enable(struct ast_websocket *session, size_t bytes)
|
||||
{
|
||||
session->reconstruct = MIN(bytes, MAXIMUM_RECONSTRUCTION_CEILING);
|
||||
}
|
||||
|
||||
void ast_websocket_reconstruct_disable(struct ast_websocket *session)
|
||||
{
|
||||
session->reconstruct = 0;
|
||||
}
|
||||
|
||||
void ast_websocket_ref(struct ast_websocket *session)
|
||||
{
|
||||
ao2_ref(session, +1);
|
||||
}
|
||||
|
||||
void ast_websocket_unref(struct ast_websocket *session)
|
||||
{
|
||||
ao2_ref(session, -1);
|
||||
}
|
||||
|
||||
int ast_websocket_fd(struct ast_websocket *session)
|
||||
{
|
||||
return session->closing ? -1 : session->fd;
|
||||
}
|
||||
|
||||
struct ast_sockaddr *ast_websocket_remote_address(struct ast_websocket *session)
|
||||
{
|
||||
return &session->address;
|
||||
}
|
||||
|
||||
int ast_websocket_is_secure(struct ast_websocket *session)
|
||||
{
|
||||
return session->secure;
|
||||
}
|
||||
|
||||
int 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] = "";
|
||||
size_t frame_size, expected = 2;
|
||||
|
||||
*payload = NULL;
|
||||
*payload_len = 0;
|
||||
*fragmented = 0;
|
||||
|
||||
/* We try to read in 14 bytes, which is the largest possible WebSocket header */
|
||||
if ((frame_size = fread(&buf, 1, 14, session->f)) < 1) {
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* The minimum size for a WebSocket frame is 2 bytes */
|
||||
if (frame_size < expected) {
|
||||
return -1;
|
||||
}
|
||||
|
||||
*opcode = buf[0] & 0xf;
|
||||
|
||||
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) {
|
||||
int fin = (buf[0] >> 7) & 1;
|
||||
int mask_present = (buf[1] >> 7) & 1;
|
||||
char *mask = NULL, *new_payload;
|
||||
size_t remaining;
|
||||
|
||||
if (mask_present) {
|
||||
/* The mask should take up 4 bytes */
|
||||
expected += 4;
|
||||
|
||||
if (frame_size < expected) {
|
||||
/* Per the RFC 1009 means we received a message that was too large for us to process */
|
||||
ast_websocket_close(session, 1009);
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
|
||||
/* Assume no extended length and no masking at the beginning */
|
||||
*payload_len = buf[1] & 0x7f;
|
||||
*payload = &buf[2];
|
||||
|
||||
/* Determine if extended length is being used */
|
||||
if (*payload_len == 126) {
|
||||
/* Use the next 2 bytes to get a uint16_t */
|
||||
expected += 2;
|
||||
*payload += 2;
|
||||
|
||||
if (frame_size < expected) {
|
||||
ast_websocket_close(session, 1009);
|
||||
return 0;
|
||||
}
|
||||
|
||||
*payload_len = ntohs(get_unaligned_uint16(&buf[2]));
|
||||
} else if (*payload_len == 127) {
|
||||
/* Use the next 8 bytes to get a uint64_t */
|
||||
expected += 8;
|
||||
*payload += 8;
|
||||
|
||||
if (frame_size < expected) {
|
||||
ast_websocket_close(session, 1009);
|
||||
return 0;
|
||||
}
|
||||
|
||||
*payload_len = ntohl(get_unaligned_uint64(&buf[2]));
|
||||
}
|
||||
|
||||
/* If masking is present the payload currently points to the mask, so move it over 4 bytes to the actual payload */
|
||||
if (mask_present) {
|
||||
mask = *payload;
|
||||
*payload += 4;
|
||||
}
|
||||
|
||||
/* Determine how much payload we need to read in as we may have already read some in */
|
||||
remaining = *payload_len - (frame_size - expected);
|
||||
|
||||
/* If how much payload they want us to read in exceeds what we are capable of close the session, things
|
||||
* will fail no matter what most likely */
|
||||
if (remaining > (MAXIMUM_FRAME_SIZE - frame_size)) {
|
||||
ast_websocket_close(session, 1009);
|
||||
return 0;
|
||||
}
|
||||
|
||||
new_payload = *payload + (frame_size - expected);
|
||||
|
||||
/* Read in the remaining payload */
|
||||
while (remaining > 0) {
|
||||
size_t payload_read;
|
||||
|
||||
/* Wait for data to come in */
|
||||
if (ast_wait_for_input(session->fd, -1) <= 0) {
|
||||
*opcode = AST_WEBSOCKET_OPCODE_CLOSE;
|
||||
*payload = NULL;
|
||||
session->closing = 1;
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* If some sort of failure occurs notify the caller */
|
||||
if ((payload_read = fread(new_payload, 1, remaining, session->f)) < 1) {
|
||||
return -1;
|
||||
}
|
||||
|
||||
remaining -= payload_read;
|
||||
new_payload += payload_read;
|
||||
}
|
||||
|
||||
/* 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];
|
||||
}
|
||||
}
|
||||
|
||||
if (!(new_payload = ast_realloc(session->payload, session->payload_len + *payload_len))) {
|
||||
*payload_len = 0;
|
||||
ast_websocket_close(session, 1009);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* 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);
|
||||
}
|
||||
|
||||
session->payload = new_payload;
|
||||
memcpy(session->payload + session->payload_len, *payload, *payload_len);
|
||||
session->payload_len += *payload_len;
|
||||
|
||||
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) {
|
||||
char *new_payload;
|
||||
|
||||
*payload_len = buf[1] & 0x7f;
|
||||
|
||||
/* 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))) {
|
||||
session->payload = new_payload;
|
||||
memcpy(session->payload, &buf[2], *payload_len);
|
||||
*payload = session->payload;
|
||||
}
|
||||
|
||||
if (!session->closing) {
|
||||
ast_websocket_close(session, 0);
|
||||
}
|
||||
|
||||
fclose(session->f);
|
||||
session->f = NULL;
|
||||
ast_verb(2, "WebSocket connection from '%s' closed\n", ast_sockaddr_stringify(&session->address));
|
||||
} else {
|
||||
/* 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 Callback that is executed everytime an HTTP request is received by this module */
|
||||
static int websocket_callback(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 websocket_protocol *protocol_handler = NULL;
|
||||
struct ast_websocket *session;
|
||||
|
||||
/* 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 -1;
|
||||
}
|
||||
|
||||
/* 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",
|
||||
ast_sockaddr_stringify(&ser->remote_address));
|
||||
ast_http_error(ser, 426, "Upgrade Required", NULL);
|
||||
return -1;
|
||||
} else if (ast_strlen_zero(requested_protocols)) {
|
||||
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - no protocols requested",
|
||||
ast_sockaddr_stringify(&ser->remote_address));
|
||||
fputs("HTTP/1.1 400 Bad Request\r\n"
|
||||
"Sec-WebSocket-Version: 7, 8, 13\r\n\r\n", ser->f);
|
||||
return -1;
|
||||
} 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",
|
||||
ast_sockaddr_stringify(&ser->remote_address));
|
||||
fputs("HTTP/1.1 400 Bad Request\r\n"
|
||||
"Sec-WebSocket-Version: 7, 8, 13\r\n\r\n", ser->f);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Iterate through the requested protocols trying to find one that we have a handler for */
|
||||
while ((protocol = strsep(&requested_protocols, ","))) {
|
||||
if ((protocol_handler = ao2_find(protocols, ast_strip(protocol), OBJ_KEY))) {
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/* 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);
|
||||
fputs("HTTP/1.1 400 Bad Request\r\n"
|
||||
"Sec-WebSocket-Version: 7, 8, 13\r\n\r\n", ser->f);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Determine how to respond depending on the version */
|
||||
if (version == 7 || version == 8 || version == 13) {
|
||||
/* 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 */
|
||||
char combined[strlen(key) + strlen(WEBSOCKET_GUID) + 1], base64[64];
|
||||
uint8_t sha[20];
|
||||
|
||||
if (!(session = ao2_alloc(sizeof(*session), session_destroy_fn))) {
|
||||
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted",
|
||||
ast_sockaddr_stringify(&ser->remote_address));
|
||||
fputs("HTTP/1.1 400 Bad Request\r\n"
|
||||
"Sec-WebSocket-Version: 7, 8, 13\r\n\r\n", ser->f);
|
||||
ao2_ref(protocol_handler, -1);
|
||||
return 0;
|
||||
}
|
||||
|
||||
snprintf(combined, sizeof(combined), "%s%s", key, WEBSOCKET_GUID);
|
||||
ast_sha1_hash_uint(sha, combined);
|
||||
ast_base64encode(base64, (const unsigned char*)sha, 20, sizeof(base64));
|
||||
|
||||
fprintf(ser->f, "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,
|
||||
base64,
|
||||
protocol);
|
||||
} 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",
|
||||
ast_sockaddr_stringify(&ser->remote_address), version ? version : 75);
|
||||
fputs("HTTP/1.1 400 Bad Request\r\n"
|
||||
"Sec-WebSocket-Version: 7, 8, 13\r\n\r\n", ser->f);
|
||||
ao2_ref(protocol_handler, -1);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Enable keepalive on all sessions so the underlying user does not have to */
|
||||
if (setsockopt(ser->fd, SOL_SOCKET, SO_KEEPALIVE, &flags, sizeof(flags))) {
|
||||
ast_log(LOG_WARNING, "WebSocket connection from '%s' could not be accepted - failed to enable keepalive",
|
||||
ast_sockaddr_stringify(&ser->remote_address));
|
||||
fputs("HTTP/1.1 400 Bad Request\r\n"
|
||||
"Sec-WebSocket-Version: 7, 8, 13\r\n\r\n", ser->f);
|
||||
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->f = ser->f;
|
||||
session->fd = ser->fd;
|
||||
ast_sockaddr_copy(&session->address, &ser->remote_address);
|
||||
session->opcode = -1;
|
||||
session->reconstruct = DEFAULT_RECONSTRUCTION_CEILING;
|
||||
session->secure = ser->ssl ? 1 : 0;
|
||||
|
||||
/* Give up ownership of the socket and pass it to the protocol handler */
|
||||
protocol_handler->callback(session, get_vars, headers);
|
||||
ao2_ref(protocol_handler, -1);
|
||||
|
||||
/* By dropping the FILE* from the session it won't get closed when the HTTP server cleans up */
|
||||
ser->f = NULL;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_http_uri websocketuri = {
|
||||
.callback = websocket_callback,
|
||||
.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;
|
||||
|
||||
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 */
|
||||
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;
|
||||
}
|
||||
}
|
||||
|
||||
end:
|
||||
ast_websocket_unref(session);
|
||||
}
|
||||
|
||||
static int load_module(void)
|
||||
{
|
||||
protocols = ao2_container_alloc(MAX_PROTOCOL_BUCKETS, protocol_hash_fn, protocol_cmp_fn);
|
||||
ast_http_uri_link(&websocketuri);
|
||||
ast_websocket_add_protocol("echo", websocket_echo_callback);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
ast_websocket_remove_protocol("echo", websocket_echo_callback);
|
||||
ast_http_uri_unlink(&websocketuri);
|
||||
ao2_ref(protocols, -1);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "HTTP WebSocket Support",
|
||||
.load = load_module,
|
||||
.unload = unload_module,
|
||||
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
|
||||
);
|
@ -0,0 +1,17 @@
|
||||
{
|
||||
global:
|
||||
LINKER_SYMBOL_PREFIXast_websocket_add_protocol;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_remove_protocol;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_read;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_write;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_close;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_reconstruct_enable;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_reconstruct_disable;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_ref;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_unref;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_fd;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_remote_address;
|
||||
LINKER_SYMBOL_PREFIXast_websocket_is_secure;
|
||||
local:
|
||||
*;
|
||||
};
|
Loading…
Reference in new issue