Asterisk - The Open Source Telephony Project GIT-master-f36a736
transport_websocket.c
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1/*
2 * Asterisk -- An open source telephony toolkit.
3 *
4 * Copyright (C) 2021, Sangoma Technologies Corporation
5 *
6 * Kevin Harwell <kharwell@sangoma.com>
7 *
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
13 *
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
17 */
18
19#include "asterisk.h"
20
22#include "asterisk/utils.h"
23
24#include "logger.h"
25#include "transport.h"
26#include "transport_websocket.h"
27
28#define log_error(obj, fmt, ...) aeap_error(obj, "websocket", fmt, ##__VA_ARGS__)
29
31 /*! Derive from base transport (must be first attribute) */
33 /*! The underlying websocket */
35};
36
37static int websocket_connect(struct aeap_transport *self, const char *url,
38 const char *protocol, int timeout)
39{
40 struct aeap_transport_websocket *transport = (struct aeap_transport_websocket *)self;
41 enum ast_websocket_result ws_result;
42 struct ast_websocket_client_options ws_options = {
43 .uri = url,
44 .protocols = protocol,
45 .timeout = timeout,
46 .tls_cfg = NULL,
47 };
48
49 transport->ws = ast_websocket_client_create_with_options(&ws_options, &ws_result);
50 if (ws_result != WS_OK) {
51 log_error(self, "connect failure (%d)", (int)ws_result);
52 return -1;
53 }
54
55 return 0;
56}
57
58static int websocket_disconnect(struct aeap_transport *self)
59{
60 struct aeap_transport_websocket *transport = (struct aeap_transport_websocket *)self;
61
62 if (transport->ws) {
63 ast_websocket_unref(transport->ws);
64 transport->ws = NULL;
65 }
66
67 return 0;
68}
69
70static void websocket_destroy(struct aeap_transport *self)
71{
72 /*
73 * Disconnect takes care of cleaning up the websocket. Note, disconnect
74 * was called by the base/dispatch interface prior to calling this
75 * function so nothing to do here.
76 */
77}
78
79static intmax_t websocket_read(struct aeap_transport *self, void *buf, intmax_t size,
80 enum AST_AEAP_DATA_TYPE *rtype)
81{
82 struct aeap_transport_websocket *transport = (struct aeap_transport_websocket *)self;
83
84 char *payload;
85 uint64_t bytes_read = 0;
86 uint64_t total_bytes_read = 0;
87 enum ast_websocket_opcode opcode;
88 int fragmented = 0;
89
91
92 if (ast_websocket_fd(transport->ws) < 0) {
93 log_error(self, "unavailable for reading");
94 /* Ensure this transport is in a disconnected state */
96 return -1;
97 }
98
99 /*
100 * This function is called with the read_lock locked. However, the lock needs to be
101 * unlocked while waiting for input otherwise a deadlock can occur during disconnect
102 * (disconnect attempts to grab the lock but can't because read holds it here). So
103 * unlock it prior to waiting.
104 */
105 ast_mutex_unlock(&transport->base.read_lock);
106 while (ast_websocket_wait_for_input(transport->ws, -1) <= 0) {
107 /* If this was poll getting interrupted just go back to waiting */
108 if (errno == EINTR || errno == EAGAIN) {
109 continue;
110 }
111
112 ast_mutex_lock(&transport->base.read_lock);
113 log_error(self, "poll failure: %s", strerror(errno));
114 /* Ensure this transport is in a disconnected state */
116 return -1;
117 }
118 ast_mutex_lock(&transport->base.read_lock);
119
120 if (!transport->ws) {
121 /*
122 * It's possible the transport was told to disconnect while waiting for input.
123 * If so then the websocket will be NULL, so we don't want to continue.
124 */
125 return 0;
126 }
127
128 do {
129 if (ast_websocket_read(transport->ws, &payload, &bytes_read, &opcode,
130 &fragmented) != 0) {
131 log_error(self, "read failure (%d): %s", opcode, strerror(errno));
132 return -1;
133 }
134
135 if (!bytes_read) {
136 continue;
137 }
138
139 if (total_bytes_read + bytes_read > size) {
140 log_error(self, "attempted to read too many bytes into (%jd) sized buffer", size);
141 return -1;
142 }
143
144 memcpy(buf + total_bytes_read, payload, bytes_read);
145 total_bytes_read += bytes_read;
146
147 } while (opcode == AST_WEBSOCKET_OPCODE_CONTINUATION);
148
149 switch (opcode) {
151 log_error(self, "closed");
152 return -1;
155 break;
158
159 /* Append terminator, but check for overflow first */
160 if (total_bytes_read == size) {
161 log_error(self, "unable to write string terminator");
162 return -1;
163 }
164
165 *((char *)(buf + total_bytes_read)) = '\0';
166 break;
167 default:
168 /* Ignore all other message types */
169 return 0;
170 }
171
172 return total_bytes_read;
173}
174
175static intmax_t websocket_write(struct aeap_transport *self, const void *buf, intmax_t size,
176 enum AST_AEAP_DATA_TYPE wtype)
177{
178 struct aeap_transport_websocket *transport = (struct aeap_transport_websocket *)self;
179 intmax_t res = 0;
180
181 switch (wtype) {
184 (char *)buf, size);
185 break;
188 (char *)buf, size);
189 break;
190 default:
191 break;
192 }
193
194 if (res < 0) {
195 log_error(self, "problem writing to websocket (closed)");
196
197 /*
198 * If the underlying socket is closed then ensure the
199 * transport is in a disconnected state as well.
200 */
202
203 return res;
204 }
205
206 return size;
207}
208
210{
211 static struct aeap_transport_vtable websocket_vtable = {
213 .disconnect = websocket_disconnect,
214 .destroy = websocket_destroy,
215 .read = websocket_read,
216 .write = websocket_write,
217 };
218
219 return &websocket_vtable;
220}
221
222/*!
223 * \brief Initialize a transport websocket object, and set its virtual table
224 *
225 * \param transport The transport to initialize
226 *
227 * \returns 0 on success, -1 on error
228 */
230{
231 transport->ws = NULL;
232
233 ((struct aeap_transport *)transport)->vtable = transport_websocket_vtable();
234
235 return 0;
236}
237
239{
240 struct aeap_transport_websocket *transport;
241
242 transport = ast_calloc(1, sizeof(*transport));
243 if (!transport) {
244 ast_log(LOG_ERROR, "AEAP websocket: unable to create transport websocket");
245 return NULL;
246 }
247
248 if (transport_websocket_init(transport)) {
249 ast_free(transport);
250 return NULL;
251 }
252
253 return transport;
254}
Asterisk main include file. File version handling, generic pbx functions.
#define ast_free(a)
Definition: astmm.h:180
#define ast_calloc(num, len)
A wrapper for calloc()
Definition: astmm.h:202
#define ast_log
Definition: astobj2.c:42
char buf[BUFSIZE]
Definition: eagi_proxy.c:66
Support for WebSocket connections within the Asterisk HTTP server and client WebSocket connections to...
int ast_websocket_read(struct ast_websocket *session, char **payload, uint64_t *payload_len, enum ast_websocket_opcode *opcode, int *fragmented)
Read a WebSocket frame and handle it.
int ast_websocket_wait_for_input(struct ast_websocket *session, int timeout)
Wait for the WebSocket session to be ready to be read.
struct ast_websocket * ast_websocket_client_create_with_options(struct ast_websocket_client_options *options, enum ast_websocket_result *result)
Create, and connect, a websocket client using given options.
ast_websocket_result
Result code for a websocket client.
@ WS_OK
ast_websocket_opcode
WebSocket operation codes.
@ AST_WEBSOCKET_OPCODE_CONTINUATION
@ AST_WEBSOCKET_OPCODE_BINARY
@ AST_WEBSOCKET_OPCODE_CLOSE
@ AST_WEBSOCKET_OPCODE_TEXT
void ast_websocket_unref(struct ast_websocket *session)
Decrease the reference count for a WebSocket session.
int ast_websocket_fd(struct ast_websocket *session)
Get the file descriptor for a WebSocket session.
int ast_websocket_write(struct ast_websocket *session, enum ast_websocket_opcode opcode, char *payload, uint64_t payload_size)
Construct and transmit a WebSocket frame.
#define LOG_ERROR
#define ast_mutex_unlock(a)
Definition: lock.h:190
#define ast_mutex_lock(a)
Definition: lock.h:189
int errno
AST_AEAP_DATA_TYPE
Supported Asterisk external application data types.
Definition: res_aeap.h:135
@ AST_AEAP_DATA_TYPE_NONE
Definition: res_aeap.h:136
@ AST_AEAP_DATA_TYPE_BINARY
Definition: res_aeap.h:137
@ AST_AEAP_DATA_TYPE_STRING
Definition: res_aeap.h:138
static char url[512]
#define NULL
Definition: resample.c:96
Asterisk external application transport virtual table.
Definition: transport.h:33
int(* connect)(struct aeap_transport *self, const char *url, const char *protocol, int timeout)
Connect a transport.
Definition: transport.h:44
struct aeap_transport base
struct ast_websocket * ws
Asterisk external application transport structure to be "derived" by specific transport implementatio...
Definition: transport.h:98
ast_mutex_t read_lock
Definition: transport.h:104
Options used for a websocket client.
Structure definition for session.
int aeap_transport_disconnect(struct aeap_transport *transport)
Disconnect a transport.
Definition: transport.c:94
static void websocket_destroy(struct aeap_transport *self)
static int websocket_disconnect(struct aeap_transport *self)
static intmax_t websocket_read(struct aeap_transport *self, void *buf, intmax_t size, enum AST_AEAP_DATA_TYPE *rtype)
static intmax_t websocket_write(struct aeap_transport *self, const void *buf, intmax_t size, enum AST_AEAP_DATA_TYPE wtype)
struct aeap_transport_websocket * aeap_transport_websocket_create(void)
Creates (heap allocated), and initializes a transport websocket.
static int transport_websocket_init(struct aeap_transport_websocket *transport)
Initialize a transport websocket object, and set its virtual table.
#define log_error(obj, fmt,...)
static int websocket_connect(struct aeap_transport *self, const char *url, const char *protocol, int timeout)
static struct aeap_transport_vtable * transport_websocket_vtable(void)
Utility functions.