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authorJoshua Colp <jcolp@digium.com>2012-06-02 21:13:36 +0000
committerJoshua Colp <jcolp@digium.com>2012-06-02 21:13:36 +0000
commit380c7c5c39b9a0879d90df99196c06d01a70dd92 (patch)
tree13c88e100b85d908c21861746dafe5c48cdc5994 /include
parent91a20ee2f95e2eb8f080bb90568459659960d726 (diff)
Add res_http_websocket module which implements the WebSocket protocol according to RFC 6455.
Review: https://reviewboard.asterisk.org/r/1952/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/http_websocket.h280
-rw-r--r--include/asterisk/utils.h2
2 files changed, 282 insertions, 0 deletions
diff --git a/include/asterisk/http_websocket.h b/include/asterisk/http_websocket.h
new file mode 100644
index 000000000..35962c48b
--- /dev/null
+++ b/include/asterisk/http_websocket.h
@@ -0,0 +1,280 @@
+/*
+ * 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/module.h"
+
+/*!
+ * \file http_websocket.h
+ * \brief Support for WebSocket connections within the Asterisk HTTP server.
+ *
+ * \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 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 Add a sub-protocol handler to the 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
+ */
+int ast_websocket_add_protocol(const char *name, ast_websocket_callback callback);
+
+/*!
+ * \brief Remove a sub-protocol handler from the 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
+ */
+int ast_websocket_remove_protocol(const char *name, ast_websocket_callback callback);
+
+/*!
+ * \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
+ */
+int ast_websocket_read(struct ast_websocket *session, char **payload, uint64_t *payload_len, enum ast_websocket_opcode *opcode, int *fragmented);
+
+/*!
+ * \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
+ */
+int ast_websocket_write(struct ast_websocket *session, enum ast_websocket_opcode opcode, char *payload, uint64_t actual_length);
+
+/*!
+ * \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
+ */
+int ast_websocket_close(struct ast_websocket *session, uint16_t reason);
+
+/*!
+ * \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.
+ */
+void ast_websocket_reconstruct_enable(struct ast_websocket *session, size_t bytes);
+
+/*!
+ * \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.
+ */
+void ast_websocket_reconstruct_disable(struct ast_websocket *session);
+
+/*!
+ * \brief Increase the reference count for a WebSocket session
+ *
+ * \param session Pointer to the WebSocket session
+ */
+void ast_websocket_ref(struct ast_websocket *session);
+
+/*!
+ * \brief Decrease the reference count for a WebSocket session
+ *
+ * \param session Pointer to the WebSocket session
+ */
+void ast_websocket_unref(struct ast_websocket *session);
+
+/*!
+ * \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.
+ */
+int ast_websocket_fd(struct ast_websocket *session);
+
+/*!
+ * \brief Get the remote address for a WebSocket connected session.
+ *
+ * \retval ast_sockaddr Remote address
+ */
+struct ast_sockaddr *ast_websocket_remote_address(struct ast_websocket *session);
+
+/*!
+ * \brief Get whether the WebSocket session is using a secure transport or not.
+ *
+ * \retval 0 if unsecure
+ * \retval 1 if secure
+ */
+int ast_websocket_is_secure(struct ast_websocket *session);
+
+#endif /* _ASTERISK_HTTP_WEBSOCKET_H */
+/*
+ * 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/module.h"
+
+/*!
+ * \file http_websocket.h
+ * \brief Support for WebSocket connections within the Asterisk HTTP server.
+ *
+ * \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 Callback for when a new connection for a sub-protocol is established
+ *
+ * \param f Pointer to the file instance for the session
+ * \param fd File descriptor for the session
+ * \param remote_address The address of the remote party
+ *
+ * \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)(FILE *f, int fd, struct ast_sockaddr *remote_address);
+
+/*!
+ * \brief Add a sub-protocol handler to the 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
+ */
+int ast_websocket_add_protocol(char *name, ast_websocket_callback callback);
+
+/*!
+ * \brief Remove a sub-protocol handler from the 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
+ */
+int ast_websocket_remove_protocol(char *name, ast_websocket_callback callback);
+
+/*!
+ * \brief Read a WebSocket frame and handle it
+ *
+ * \param f Pointer to the file stream, used to respond to certain frames
+ * \param buf Pointer to the buffer containing the frame
+ * \param buflen Size of the buffer
+ * \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 int which will be populated with the opcode of the frame
+ *
+ * \retval NULL if no payload is present
+ * \retval non-NULL if payload is present, returned pointer points to beginning of payload
+ */
+char *ast_websocket_read(FILE *f, char *buf, size_t buflen, uint64_t *payload_len, int *opcode);
+
+/*!
+ * \brief Construct and transmit a WebSocket frame
+ *
+ * \param f Pointer to the file stream which the frame will be sent on
+ * \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)
+ */
+void ast_websocket_write(FILE *f, int op_code, char *payload, uint64_t actual_length);
+
+#endif /* _ASTERISK_HTTP_WEBSOCKET_H */
diff --git a/include/asterisk/utils.h b/include/asterisk/utils.h
index 4ebd3ead8..a7e01153b 100644
--- a/include/asterisk/utils.h
+++ b/include/asterisk/utils.h
@@ -219,6 +219,8 @@ struct hostent *ast_gethostbyname(const char *host, struct ast_hostent *hp);
void ast_md5_hash(char *output, const char *input);
/*! \brief Produces SHA1 hash based on input string */
void ast_sha1_hash(char *output, const char *input);
+/*! \brief Produces SHA1 hash based on input string, stored in uint8_t array */
+void ast_sha1_hash_uint(uint8_t *digest, const char *input);
int ast_base64encode_full(char *dst, const unsigned char *src, int srclen, int max, int linebreaks);