48 Commits (4603e246091d779e539c211274fd31a7e2773808)

Author	SHA1	Message	Date
Sean Bright	5ebc94da02	res_http_websocket: Don't leak memory on read failure ... Change-Id: Ic449ea832bc81a1671c0e910c5fbe8c683e3da89	8 years ago
Alexander Traud	9865e689d2	General: Avoid implicit conversion to char when changes value to negative. ... clang 5.0 warned about this. ASTERISK-27557 Change-Id: I7cceaa88e147cbdf81a3a7beec5c1c20210fa41e	8 years ago
Sean Bright	f726f11974	utils: Add convenience function for setting fd flags ... There are many places in the code base where we ignore the return value of fcntl() when getting/setting file descriptior flags. This patch introduces a convenience function that allows setting or clearing file descriptor flags and will also log an error on failure for later analysis. Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7	8 years ago
Joshua Colp	c2dddb001a	pjsip / hep: Provide correct local address for Websockets. ... Previously for PJSIP the local address of WebSocket connections was set to the remote address. For logging purposes this is not particularly useful. The WebSocket API has been extended to allow the local address to be queried and this is used in PJSIP to set the local address to the correct value. The PJSIP HEP support has also been tweaked so that reliable transports always use the local address on the transport and do not try to (wrongly) guess. As they are connection based it is impossible for the source to be anything else. ASTERISK-26758 ASTERISK-27363 Change-Id: Icd305fd038ad755e2682ab2786e381f6bf29e8ca	8 years ago
Corey Farrell	73a5e9f0e9	Prevent unload of modules which implement an Optional API. ... Once an Optional API module is loaded it should stay loaded. Unloading an optional API module runs the risk of a crash if something else is using it. This patch causes all optional API providers to tell the module loader not to unload except at shutdown. ASTERISK-27389 Change-Id: Ia07786fe655681aec49cc8d3d96e06483b11f5e6	8 years ago
George Joseph	f882ca2572	modules: change module LOAD_FAILUREs to LOAD_DECLINES ... In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting if a module can't be loaded. If the user wishes to retain the FAILURE behavior for a specific module, they can use the "require" or "preload-require" keyword in modules.conf. A new API was added to logger: ast_is_logger_initialized(). This allows asterisk.c/check_init() to print to the error log once the logger subsystem is ready instead of just to stdout. If something does fail before the logger is initialized, we now print to stderr instead of stdout. Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25	9 years ago
Matt Jordan	e79acaeb75	res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems ... Not surprisingly, using Respoke (and possibly other systems) it is possible to blow past the 16k limit for a WebSocket packet size. This patch bumps it up to 32k, which, at least for Respoke, is sufficient. For now. Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that matter), this patch adds a LOW_MEMORY directive that sets the buffer to 8k for systems who have asked for their reduced memory availability to be considered. Change-Id: Id235902537091b58608196844dc4b045e383cd2e	9 years ago
Joshua Colp	cb56ef8069	res_http_websocket: Make core supported. ... Websockets are a core part of ARI support and as such this module should also be core supported. Change-Id: I8f9283c6a167152761b92984779bb39e3db51a9c	10 years ago
Dade Brandon	136c537695	res_http_websocket.c: prevent avoidable disconnections caused by write errors ... Updated ast_websocket_write to encode the entire frame in to one write operation, to ensure that we don't end up with a situation where the websocket header has been sent, while the body can not be written. Previous to August's patch in commit b9bd3c14, certain network conditions could cause the header to be written, and then the sub-sequent body to fail - which would cause the next successful write to contain a new header, and a new body (resulting in the peer receiving two headers - the second of which would be read as part of the body for the first header). This was patched to have both write operations individually fail by closing the websocket. In a case available to the submitter of this patch, the same body which would consistently fail to write, would succeed if written at the same time as the header. This update merges the two operations in to one, adds debug messages indicating the reason for a websocket connection being closed during a write operation, and clarifies some variable names for code legibility. Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598	10 years ago
Richard Mudgett	77518d5434	res_http_websocket.c: Fix some off nominal path cleanup. ... * Remove extraneous unlock on off-nominal path. * Add missing HTTP error reply. Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b	10 years ago
Joshua Colp	e18c300550	res_http_websocket: When shutting down a session don't close closed socket ... Due to the use of ast_websocket_close in session termination it is possible for the underlying socket to already be closed when the session is terminated. This occurs when the close frame is attempted to be written out but fails. Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b	10 years ago
Joshua Colp	b4e9416138	res_http_websocket: Forcefully terminate on write errors. ... The res_http_websocket module will currently attempt to close the WebSocket connection if fatal cases occur, such as when attempting to write out data and being unable to. When the fatal cases occur the code attempts to write a WebSocket close frame out to have the remote side close the connection. If writing this fails then the connection is not terminated. This change forcefully terminates the connection if the WebSocket is to be closed but is unable to send the close frame. ASTERISK-25312 #close Change-Id: I10973086671cc192a76424060d9ec8e688602845	10 years ago
David M. Lee	26f0559a94	Replace htobe64 with htonll ... We don't have a compatability function to fill in a missing htobe64; but we already have one for the identical htonll. Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac	10 years ago
Mark Michelson	27dc2094e9	res_http_websocket: Debug write lengths. ... Commit 39cc28f6ea attempted to fix a test failure observed on 32 bit test agents by ensuring that a cast from a 32 bit unsigned integer to a 64 bit unsigned integer was happening in a predictable place. As it turns out, this did not cause test runs to succeed. This commit adds several redundant debug messages that print the payload lengths of websocket frames. The idea here is that this commit will not cause tests to succeed for the faulty test agent, but we might deduce where the fault lies more easily this way by observing at what point the expected value (537) changes to some ungangly huge number. If you are wondering why something like this is being committed to the branch, keep in mind that in commit 39cc28f6ea I noted that the observed test failures only happen when automated tests are run. Attempts to run the tests by hand manually on the test agent result in the tests passing. Change-Id: I14a65c19d8af40dadcdbd52348de3b0016e1ae8d	10 years ago
Mark Michelson	39cc28f6ea	res_http_websocket: Avoid passing strlen() to ast_websocket_write(). ... We have seen a rash of test failures on a 32-bit build agent. Commit 48698a5e21 solved an obvious problem where we were not encoding a 64-bit value correctly over the wire. This commit, however, did not solve the test failures. In the failing tests, ARI is attempting to send a 537 byte text frame over a websocket. When sending a frame this small, 16 bits are all that is required in order to encode the payload length on the websocket frame. However, ast_websocket_write() thinks that the payload length is greater than 65535 and therefore writes out a 64 bit payload length. Inspecting this payload length, the lower 32 bits are exactly what we would expect it to be, 537 in hex. The upper 32 bits, are junk values that are not expected to be there. In the failure, we are passing the result of strlen() to a function that expects a uint64_t parameter to be passed in. strlen() returns a size_t, which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit unsigned value to somewhere where a 64-bit unsigned value is expected would cause no problems. In fact, in manual runs of failing tests, this works just fine. However, ast_websocket_write() uses the Asterisk optional API, which means that rather than a simple function call, there are a series of macros that are used for its declaration and implementation. These macros may be causing some sort of error to occur when converting from a 32 bit quantity to a 64 bit quantity. This commit changes the logic by making existing ast_websocket_write() calls use ast_websocket_write_string() instead. Within ast_websocket_write_string(), the 64-bit converted strlen is saved in a local variable, and that variable is passed to ast_websocket_write() instead. Note that this commit message is full of speculation rather than certainty. This is because the observed test failures, while always present in automated test runs, never occur when tests are manually attempted on the same test agent. The idea behind this commit is to fix a theoretical issue by performing changes that should, at the least, cause no harm. If it turns out that this change does not fix the failing tests, then this commit should be reverted. Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67	10 years ago
Mark Michelson	48698a5e21	res_http_websocket: Properly encode 64 bit payload ... A test agent was continuously failing all ARI tests when run against Asterisk 13. As it turns out, the reason for this is that on those test runs, for some reason we decided to use the super extended 64 bit payload length for websocket text frames instead of the extended 16 bit payload length. For 64-bit payloads, the expected byte order over the network is 7, 6, 5, 4, 3, 2, 1, 0 However, we were sending the payload as 3, 2, 1, 0, 7, 6, 5, 4 This meant that we were saying to expect an absolutely MASSIVE payload to arrive. Since we did not follow through on this expected payload size, the client would sit patiently waiting for the rest of the payload to arrive until the test would time out. With this change, we use the htobe64() function instead of htonl() so that a 64-bit byte-swap is performed instead of a 32 bit byte-swap. Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a	10 years ago
Joshua Colp	55137c3d12	res/res_http_websocket: Don't send HTTP response fragmented. ... This change makes it so that when accepting a WebSocket connection the HTTP response is sent as one packet instead of fragmented. Browsers don't like it when you send it fragmented. ASTERISK-25103 Change-Id: I9b82c4ec2949b0bce692ad0bf6f7cea9709e7f69	10 years ago
Matt Jordan	31cc24aad6	res/res_http_websocket: Add a pre-session established callback ... This patch updates http_websocket and its corresponding implementation with a pre-session established callback. This callback allows for WebSocket server consumers to be notified when a WebSocket connection is attempted, but before we accept it. Consumers can choose to reject the connection, if their application specific logic allows for it. As a result, this patch pulls out the previously private websocket_protocol struct and makes it public, as ast_websocket_protocol. In order to preserve backwards compatibility with existing modules, the existing APIs were left as-is, and new APIs were added for the creation of the ast_websocket_protocol as well as for adding a sub-protocol to a WebSocket server. In particular, the following new API calls were added: * ast_websocket_add_protocol2 - add a protocol to the core WebSocket server * ast_websocket_server_add_protocol2 - add a protocol to a specific WebSocket server * ast_websocket_sub_protocol_alloc - allocate a sub-protocol object. Consumers can populate this with whatever callbacks they wish to support, then add it to the core server or a specified server. ASTERISK-24988 Reported by: Joshua Colp Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2	11 years ago
David M. Lee	43a3e80be1	Increase WebSocket frame size and improve large read handling ... Some WebSocket applications, like [chan_respoke][], require a larger frame size than the default 8k; this patch bumps the default to 16k. This patch also fixes some problems exacerbated by large frames. The sanity counter was decremented on every fread attempt in ws_safe_read(), regardless of whether data was read from the socket or not. For large frames, this could result in loss of sanity prior to reading the entire frame. (16k frame / 1448 bytes per segment = 12 segments). This patch changes the sanity counter so that it only decrements when fread() doesn't read any bytes. This more closely matches the original intention of ws_safe_read(), given that the error message is "Websocket seems unresponsive". This patch also properly logs EOF conditions, so disconnects are no longer confused with unresponsive connections. [chan_respoke]: https://github.com/respoke/chan_respoke Review: https://reviewboard.asterisk.org/r/4431/ ........ Merged revisions 432236 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	72e5ba2ce8	res_http_websocket: websocket write timeout fails to fully disconnect ... When writing to a websocket if a timeout occurred the underlying socket did not get closed/disconnected. This patch makes sure the websocket gets disconnected on a write timeout. Also a notice is logged stating that the websocket was disconnected. ASTERISK-24701 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/ ........ Merged revisions 431669 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4b054bdc6d	res_http_websocket.c: Fix incorrect use of sizeof in ast_websocket_write(). ... This won't fix the reported issue but it is an incorrect use of sizeof. ASTERISK-24566 Reported by: Badalian Vyacheslav ........ Merged revisions 429867 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	fb768ec33a	res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero. ... Frames with a payload length of 0 were incorrectly handled in res_http_websocket. Provided a frame with a payload had been received prior it was possible for a double free to occur. The realloc operation would succeed (thus freeing the payload) but be treated as an error. When the session was then torn down the payload would be freed again causing a crash. The read function now takes this into account. This change also fixes assumptions made by users of res_http_websocket. There is no guarantee that a frame received from it will be NULL terminated. ASTERISK-24472 #close Reported by: Badalian Vyacheslav Review: https://reviewboard.asterisk.org/r/4220/ Review: https://reviewboard.asterisk.org/r/4219/ ........ Merged revisions 429270 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429272 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	3e29fa3686	res_http_websockets: Fix extra unref of module ... In websocket_add_protocol_internal is used to add the "echo" protocol, but ast_websocket_remove_protocol is used to remove it. This causes an extra call to ast_module_unref. ASTERISK-24480 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4140/ ........ Merged revisions 427200 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b23580afe6	res/res_http_websocket: Fix minor nits found by wdoekes on r409681 ... When Moises committed the fixes for WSS (which was a great patch), wdoekes had a few style nits that were on the review that got missed. This patch resolves what I *think* were all of the ones that were still on the review. Thanks to both moy for the patch, and wdoekes for the reviews. Review: https://reviewboard.asterisk.org/r/3248/ ........ Merged revisions 426209 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426210 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	66fb08e26d	res_http_websocket: Include query parameters in client connection requests. ... Review: https://reviewboard.asterisk.org/r/3914/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ... ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	dbec5e0d8d	HTTP: Add persistent connection support. ... Persistent HTTP connection support is needed due to the increased usage of the Asterisk core HTTP transport and the frequency at which REST API calls are going to be issued. * Add http.conf session_keep_alive option to enable persistent connections. * Parse and discard optional chunked body extension information and trailing request headers. * Increased the maximum application/json and application/x-www-form-urlencoded body size allowed to 4k. The previous 1k was kind of small. * Removed a couple inlined versions of ast_http_manid_from_vars() by calling the function. manager.c:generic_http_callback() and res_http_post.c:http_post_callback() * Add missing va_end() in ast_ari_response_error(). * Eliminated unnecessary RAII_VAR() use in http.c:auth_create(). ASTERISK-23552 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/3691/ ........ Merged revisions 417880 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	365ae7523b	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	12 years ago
Kevin Harwell	bd0aa4fb04	res_http_websocket: read/write string fixup ... There was a problem when reading a string from the websocket. It assumed the received data had a null terminator and tried to write the data to an ast_str. This of course could/would read past the end of the given buffer while writing the data to the internal buffer of ast_str. Modified the the code to correctly place a null terminator on the result string. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416394 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4ca5745dbe	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. ... Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	e763d70470	res_http_websocket: Create a websocket client ... Added a websocket server client in Asterisk. Asterisk has a websocket server, but not a client. The ability to have Asterisk be able to connect to a websocket server can potentially be useful for future work (for instance this could allow ARI to connect back to some external system, although more work would be needed in order to incorporate that). Also a couple of things to note - proxy connection support has not been implemented and there is limited http response code handling (basically, it is connect or not). Also added an initial new URI handling mechanism to core. Internet type URI's are parsed into a data structure that contains pointers to the various parts of the URI. (closes issue ASTERISK-23742) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/3541/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	abd3e4040b	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	a7fc217837	Websocket: Add session locking and delay close ... This resolves a race condition where data could be written to a NULL FILE pointer causing a crash as a websocket connection was in the process of shutting down by adding locking to websocket session writes and by deferring session teardown until session destruction. (closes issue ASTERISK-23605) Review: https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan ........ Merged revisions 413123 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413124 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Moises Silva	bcb0f94604	Fix res/res_http_websocket.c build failure in 32bit due to incorrect print format for uint64_t ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Moises Silva	e4e5cfa9b8	Fix WebRTC over WSS not working ... Several fixes for the WebSockets implementation in res/res_http_websocket.c * Flush the websocket session FILE* as fwrite() may not actually guarantee sending the data to the network. If we do not flush, it seems that buffering on the SSL socket for outbound messages causes issues * Refactored ast_websocket_read to take into account that SSL file descriptors may be ready to read via fread() but poll() will not actually say so because the data was already read from the network buffers and is now in the libc buffers (closes issue ASTERISK-23099) (closes issue ASTERISK-21930) Review: https://reviewboard.asterisk.org/r/3248/ ........ Merged revisions 409681 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409697 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	f56796a539	ARI: Fix WebSocket response when subprotocol isn't specified ... When I moved the ARI WebSocket from /ws to /ari/events, I added code to allow a WebSocket to connect without specifying the subprotocol if there's only one subprotocol handler registered for the WebSocket. Naively, I coded it to always respond with the subprotocol in use. Unfortunately, according to RFC 6455, if the server's response includes a subprotocol header field that "indicates the use of a subprotocol that was not present in the client's handshake [...], the client MUST _Fail the WebSocket Connection_.", emphasis theirs. This patch correctly omits the Sec-WebSocket-Protocol if one is not specified by the client. (closes issue ASTERISK-22441) Review: https://reviewboard.asterisk.org/r/2828/ ........ Merged revisions 399039 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	9bed50db41	optional_api: Fix linking problems between modules that export global symbols ... With the new work in Asterisk 12, there are some uses of the optional_api that are prone to failure. The details are rather involved, and captured on [the wiki][1]. This patch addresses the issue by removing almost all of the magic from the optional API implementation. Instead of relying on weak symbol resolution, a new optional_api.c module was added to Asterisk core. For modules providing an optional API, the pointer to the implementation function is registered with the core. For modules that use an optional API, a pointer to a stub function, along with a optional_ref function pointer are registered with the core. The optional_ref function pointers is set to the implementation function when it's provided, or the stub function when it's now. Since the implementation no longer relies on magic, it is now supported on all platforms. In the spirit of choice, an OPTIONAL_API flag was added, so we can disable the optional_api if needed (maybe it's buggy on some bizarre platform I haven't tested on) The AST_OPTIONAL_API*() macros themselves remained unchanged, so existing code could remain unchanged. But to help with debugging the optional_api, the patch limits the #include of optional API's to just the modules using the API. This also reduces resource waste maintaining optional_ref pointers that aren't used. Other changes made as a part of this patch: * The stubs for http_websocket that wrap system calls set errno to ENOSYS. * res_http_websocket now properly increments module use count. * In loader.c, the while() wrappers around dlclose() were removed. The while(!dlclose()) is actually an anti-pattern, which can lead to infinite loops if the module you're attempting to unload exports a symbol that was directly linked to. * The special handling of nonoptreq on systems without weak symbol support was removed, since we no longer rely on weak symbols for optional_api. [1]: https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue ASTERISK-22296) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2797/ ........ Merged revisions 397989 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	3c86832f9f	Fixed null dereference when WebSocket subprotocol isn't specified ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	80dd0229f1	Fixed null dereference when WebSocket protocol is omitted ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	dcf03554a0	Shuffle RESTful URL's around. ... This patch moves the RESTful URL's around to more appropriate locations for release. The /stasis URL's are moved to /ari, since Asterisk REST Interface was a more appropriate name than Stasis-HTTP. (Most of the code still has stasis_http references, but they will be cleaned up after there are no more outstanding branches that would have merge conflicts with such a change). A larger change was moving the ARI events WebSocket off of the shared /ws URL to its permanent home on /ari/events. The Swagger code generator was extended to handle "upgrade: websocket" and "websocketProtocol:" attributes on an operation. The WebSocket module was modified to better handle WebSocket servers that have a single registered protocol handler. If a client connections does not specify the Sec-WebSocket-Protocol header, and the server has a single protocol handler registered, the WebSocket server will go ahead and accept the client for that subprotocol. (closes issue ASTERISK-21857) Review: https://reviewboard.asterisk.org/r/2621/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	1e9faaf78a	Fix segfault for certain invalid WebSocket input. ... The WebSocket code would allocate, on the stack, a string large enough to hold a key provided by the client, and the WEBSOCKET_GUID. If the key is NULL, this causes a segfault. If the key is too large, it could overflow the stack. This patch checks the key for NULL and checks the length of the key to avoid stack smashing nastiness. (closes issue ASTERISK-21825) Reported by: Alfred Farrugia Tested by: Alfred Farrugia, David M. Lee Patches: issueA21825_check_if_key_is_sent.patch uploaded by Walter Doekes (license 5674) ........ Merged revisions 391560 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	9c696e665f	Allow WebSocket connections on more URL's ... This patch adds the concept of ast_websocket_server to res_http_websocket, allowing WebSocket connections on URL's more more than /ws. The existing funcitons for managing the WebSocket subprotocols on /ws still work, so this patch should be completely backward compatible. (closes issue ASTERISK-21279) Review: https://reviewboard.asterisk.org/r/2453/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a5df2542c3	Don't attempt a websocket protocol removal if res_http_websocket isn't there ... This patch sets the protocols container provided by res_http_websocket to NULL when the module gets unloaded and adds the necessary checks when adding/ removing a websocket protocol. This prevents some FRACKing on an invalid pointer to the disposed container if a module that uses res_http_websocket is unloaded after it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	cc01a79463	Added missing newlines to websocket ast_logs. ... Without these newlines, log messages just continue tacking onto the same line, and do not flush immediately. ........ Merged revisions 376561 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	9f55e5e928	Make res_http_websocket an optional dependency on supported platforms for chan_sip. ... (closes issue ASTERISK-20439) Reported by: sruffell Patches: 0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded by sruffell (license 5417) ........ Merged revisions 373914 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	9b16c8b0f6	Clean up and ensure proper usage of alloca() ... This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e938737570	Add support for SIP over WebSocket. ... This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb. Review: https://reviewboard.asterisk.org/r/2008 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	380c7c5c39	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	14 years ago