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authorMatthew Jordan <mjordan@digium.com>2014-06-26 12:21:14 +0000
committerMatthew Jordan <mjordan@digium.com>2014-06-26 12:21:14 +0000
commit365ae7523b45f18abb1418f498561cc2c8cbf680 (patch)
tree2d1ce4e889fedf5885299baef55a16df464f7a21 /res/res_http_websocket.c
parentd171e0b2e96ca1cc2cf6c53cdd9d5a3c876be91b (diff)
res_http_websocket: Close websocket correctly and use careful fwrite
When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. Note that this version of the patch, unlike r417310 in Asterisk 11, exposes configuration options beyond just chan_sip's sip.conf. Configuration options to configure the write timeout have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ ........ Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_http_websocket.c')
-rw-r--r--res/res_http_websocket.c22
1 files changed, 16 insertions, 6 deletions
diff --git a/res/res_http_websocket.c b/res/res_http_websocket.c
index 07fcd9e2e..90744a1b3 100644
--- a/res/res_http_websocket.c
+++ b/res/res_http_websocket.c
@@ -81,6 +81,7 @@ struct ast_websocket {
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 */
+ int timeout; /*!< The timeout for operations on the socket */
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 */
unsigned int close_sent:1; /*!< Bit to indicate that the session close opcode has been sent and no further data will be sent */
@@ -260,7 +261,7 @@ int AST_OPTIONAL_API_NAME(ast_websocket_close)(struct ast_websocket *session, ui
session->close_sent = 1;
ao2_lock(session);
- res = (fwrite(frame, 1, 4, session->f) == 4) ? 0 : -1;
+ res = ast_careful_fwrite(session->f, session->fd, frame, 4, session->timeout);
ao2_unlock(session);
return res;
}
@@ -303,13 +304,12 @@ int AST_OPTIONAL_API_NAME(ast_websocket_write)(struct ast_websocket *session, en
ao2_unlock(session);
return -1;
}
-
- if (fwrite(frame, 1, header_size, session->f) != header_size) {
+ if (ast_careful_fwrite(session->f, session->fd, frame, header_size, session->timeout)) {
ao2_unlock(session);
return -1;
}
- if (fwrite(payload, 1, actual_length, session->f) != actual_length) {
+ if (ast_careful_fwrite(session->f, session->fd, payload, actual_length, session->timeout)) {
ao2_unlock(session);
return -1;
}
@@ -371,6 +371,13 @@ int AST_OPTIONAL_API_NAME(ast_websocket_set_nonblock)(struct ast_websocket *sess
return 0;
}
+int AST_OPTIONAL_API_NAME(ast_websocket_set_timeout)(struct ast_websocket *session, int timeout)
+{
+ session->timeout = timeout;
+
+ return 0;
+}
+
/* MAINTENANCE WARNING on ast_websocket_read()!
*
* We have to keep in mind during this function that the fact that session->fd seems ready
@@ -514,8 +521,10 @@ int AST_OPTIONAL_API_NAME(ast_websocket_read)(struct ast_websocket *session, cha
}
/* 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);
+ if ((*opcode == AST_WEBSOCKET_OPCODE_PING) && (ast_websocket_write(session, AST_WEBSOCKET_OPCODE_PONG, *payload, *payload_len))) {
+ *payload_len = 0;
+ ast_websocket_close(session, 1009);
+ return 0;
}
session->payload = new_payload;
@@ -696,6 +705,7 @@ int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instan
ao2_ref(protocol_handler, -1);
return 0;
}
+ session->timeout = AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT;
fprintf(ser->f, "HTTP/1.1 101 Switching Protocols\r\n"
"Upgrade: %s\r\n"