[svn-commits] file: trunk r368359 - in /trunk: include/asterisk/ main/ res/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Sat Jun 2 16:13:42 CDT 2012


Author: file
Date: Sat Jun  2 16:13:36 2012
New Revision: 368359

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=368359
Log:
Add res_http_websocket module which implements the WebSocket protocol according to RFC 6455.

Review: https://reviewboard.asterisk.org/r/1952/

Added:
    trunk/include/asterisk/http_websocket.h   (with props)
    trunk/res/res_http_websocket.c   (with props)
    trunk/res/res_http_websocket.exports.in   (with props)
Modified:
    trunk/include/asterisk/utils.h
    trunk/main/utils.c

Added: trunk/include/asterisk/http_websocket.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/http_websocket.h?view=auto&rev=368359
==============================================================================
--- trunk/include/asterisk/http_websocket.h (added)
+++ trunk/include/asterisk/http_websocket.h Sat Jun  2 16:13:36 2012
@@ -1,0 +1,280 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at 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 at 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 at 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 at 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 */

Propchange: trunk/include/asterisk/http_websocket.h
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    svn:eol-style = native

Propchange: trunk/include/asterisk/http_websocket.h
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: trunk/include/asterisk/http_websocket.h
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Modified: trunk/include/asterisk/utils.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/utils.h?view=diff&rev=368359&r1=368358&r2=368359
==============================================================================
--- trunk/include/asterisk/utils.h (original)
+++ trunk/include/asterisk/utils.h Sat Jun  2 16:13:36 2012
@@ -219,6 +219,8 @@
 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);
 

Modified: trunk/main/utils.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/utils.c?view=diff&rev=368359&r1=368358&r2=368359
==============================================================================
--- trunk/main/utils.c (original)
+++ trunk/main/utils.c Sat Jun  2 16:13:36 2012
@@ -268,6 +268,18 @@
 	ptr = output;
 	for (x = 0; x < 20; x++)
 		ptr += sprintf(ptr, "%2.2x", Message_Digest[x]);
+}
+
+/*! \brief Produce a 20 byte SHA1 hash of value. */
+void ast_sha1_hash_uint(uint8_t *digest, const char *input)
+{
+        struct SHA1Context sha;
+
+        SHA1Reset(&sha);
+
+        SHA1Input(&sha, (const unsigned char *) input, strlen(input));
+
+        SHA1Result(&sha, digest);
 }
 
 /*! \brief decode BASE64 encoded text */

Added: trunk/res/res_http_websocket.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_http_websocket.c?view=auto&rev=368359
==============================================================================
--- trunk/res/res_http_websocket.c (added)
+++ trunk/res/res_http_websocket.c Sat Jun  2 16:13:36 2012
@@ -1,0 +1,662 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at 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 at digium.com>
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#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/http_websocket.h"
+
+/*! \brief GUID used to compute the accept key, defined in the specifications */
+#define WEBSOCKET_GUID "258EAFA5-E914-47DA-95CA-C5AB0DC85B11"
+
+/*! \brief Number of buckets for registered protocols */
+#define MAX_PROTOCOL_BUCKETS 7
+
+/*! \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 16384
+
+/*! \brief Maximum reconstruction size for multi-frame payload reconstruction. */
+#define MAXIMUM_RECONSTRUCTION_CEILING 16384
+
+/*! \brief Structure definition for session */
+struct ast_websocket {
+	FILE *f;                          /*!< Pointer to the file instance used for writing and reading */
+	int fd;                           /*!< File descriptor for the session, only used for polling */
+	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 */
+	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 */
+};
+
+/*! \brief Structure definition for protocols */
+struct websocket_protocol {
+	char *name;                      /*!< Name of the protocol */
+	ast_websocket_callback callback; /*!< Callback called when a new session is established */
+};
+
+/*! \brief Container for registered protocols */
+static struct ao2_container *protocols;
+
+/*! \brief Hashing function for protocols */
+static int protocol_hash_fn(const void *obj, const int flags)
+{
+	const struct 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 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 websocket_protocol *protocol = obj;
+	ast_free(protocol->name);
+}
+
+/*! \brief Destructor function for sessions */
+static void session_destroy_fn(void *obj)
+{
+	struct ast_websocket *session = obj;
+
+	if (session->f) {
+		fclose(session->f);
+		ast_verb(2, "WebSocket connection from '%s' closed\n", ast_sockaddr_stringify(&session->address));
+	}
+
+	ast_free(session->payload);
+}
+
+int ast_websocket_add_protocol(const char *name, ast_websocket_callback callback)
+{
+	struct websocket_protocol *protocol;
+
+	ao2_lock(protocols);
+
+	/* Ensure a second protocol handler is not registered for the same protocol */
+	if ((protocol = ao2_find(protocols, name, OBJ_KEY | OBJ_NOLOCK))) {
+		ao2_ref(protocol, -1);
+		ao2_unlock(protocols);
+		return -1;
+	}
+
+	if (!(protocol = ao2_alloc(sizeof(*protocol), protocol_destroy_fn))) {
+		ao2_unlock(protocols);
+		return -1;
+	}
+
+	if (!(protocol->name = ast_strdup(name))) {
+		ao2_ref(protocol, -1);
+		ao2_unlock(protocols);
+		return -1;
+	}
+
+	protocol->callback = callback;
+
+	ao2_link_flags(protocols, protocol, OBJ_NOLOCK);
+	ao2_unlock(protocols);
+	ao2_ref(protocol, -1);
+
+	ast_verb(2, "WebSocket registered sub-protocol '%s'\n", name);
+
+	return 0;
+}
+
+int ast_websocket_remove_protocol(const char *name, ast_websocket_callback callback)
+{
+	struct websocket_protocol *protocol;
+
+	if (!(protocol = ao2_find(protocols, name, OBJ_KEY))) {
+		return -1;
+	}
+
+	if (protocol->callback != callback) {
+		ao2_ref(protocol, -1);
+		return -1;
+	}
+
+	ao2_unlink(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_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 */
+
+	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;
+
+	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,
+	);

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Added: trunk/res/res_http_websocket.exports.in
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_http_websocket.exports.in?view=auto&rev=368359
==============================================================================
--- trunk/res/res_http_websocket.exports.in (added)
+++ trunk/res/res_http_websocket.exports.in Sat Jun  2 16:13:36 2012
@@ -1,0 +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:
+		*;
+};

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