27218 Commits (482f2fc5ff67b3a7fd26b5df174ec847c05f4643)

 

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Author	SHA1	Message	Date
Joshua Colp	495dfb24b7	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	e1e37e47fd	Merge "res_http_websocket: Forcefully terminate on write errors."	10 years ago
Mark Michelson	58edd2dddc	Merge "chan_dahdi.c: Lock private struct for ast_write()."	10 years ago
Mark Michelson	318b97fd1e	Merge "chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF."	10 years ago
Joshua Colp	1aa722b3ce	Merge "chan_sip: Fix negotiation of iLBC 30."	10 years ago
Mark Michelson	cf45868984	Merge "res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message."	10 years ago
Joshua Colp	7e65be4ecd	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
Matt Jordan	a87e2dd254	res/res_format_attr_silk: Expose format attributes to other modules ... This patch adds the .get callback to the format attribute module, such that the Asterisk core or other third party modules can query for the negotiated format attributes. Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c	10 years ago
Richard Mudgett	87c92d2aee	chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF. ... Pressing DTMF digits on a phone to go out on a DAHDI channel can result in the digit not being recognized or even heard by the peer. Phone -> Asterisk -> DAHDI/channel Turns out the DAHDI behavior with DTMF generation (and any other generated tones) is exposed by the "buffers=" setting in chan_dahdi.conf. When Asterisk requests to start sending DTMF then DAHDI waits until its write buffer is empty before generating any samples for the DTMF tones. When Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI immediately stops generating the DTMF samples. As a result, the more samples there are in the DAHDI write buffer the shorter the time DTMF actually gets sent on the wire. If there are more samples in the write buffer than the time DTMF is supposed to be sent then no DTMF gets sent on the wire. With the "buffers=12,half" setting and each buffer representing 20 ms of samples then the DAHDI write buffer is going to contain around 120 ms of samples. For DTMF to be recognized by the peer the actual sent DTMF duration needs to be a minimum of 40 ms. Therefore, the intended duration needs to be a minimum of 160 ms for the peer to receive the minimum DTMF digit duration to recognize it. A simple and effective solution to work around the DAHDI behavior is for Asterisk to flush the DAHDI write buffer when sending DTMF so the full duration of DTMF is actually sent on the wire. When someone is going to send DTMF they are not likely to be talking before sending the tones so the flushed write samples are expected to just contain silence. * Made dahdi_digit_begin() flush the DAHDI write buffer after requesting to send a DTMF digit. ASTERISK-25315 #close Reported by John Hardin Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a	10 years ago
Richard Mudgett	b9b957d4e9	chan_dahdi.c: Lock private struct for ast_write(). ... There is a window of opportunity for DTMF to not go out if an audio frame is in the process of being written to DAHDI while another thread starts sending DTMF. The thread sending the audio frame could be past the currently dialing check before being preempted by another thread starting a DTMF generation request. When the thread sending the audio frame resumes it will then cause DAHDI to stop the DTMF tone generation. The result is no DTMF goes out. * Made dahdi_write() lock the private struct before writing to the DAHDI file descriptor. ASTERISK-25315 Reported by John Hardin Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb	10 years ago
Richard Mudgett	f3f5b45d57	res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message. ... If the saved SUBSCRIBE message is not parseable for whatever reason then Asterisk could crash when libpjsip tries to parse the message and adds an error message to the parse error list. * Made ast_sip_create_rdata() initialize the parse error rdata list. The list is checked after parsing to see that it remains empty for the function to return successful. ASTERISK-25306 Reported by Mark Michelson Change-Id: Ie0677f69f707503b1a37df18723bd59418085256	10 years ago
Alexander Traud	991d4da1eb	chan_sip: Fix negotiation of iLBC 30. ... iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5, only iLBC 30 is negotiated now. ASTERISK-25309 #close Change-Id: I92d724600a183eec3114da0ac607b994b1a793da	10 years ago
Matt Jordan	e188192ad1	main/format: Add an API call for retrieving format attributes ... Some codecs that may be a third party library to Asterisk need to have knowledge of the format attributes that were negotiated. Unfortunately, when the great format migration of Asterisk 13 occurred, that ability was lost. This patch adds an API call, ast_format_attribute_get, to the core format API, along with updates to the unit test to check the new API call. A new callback is also now available for format attribute modules, such that they can provide the format attribute values they manage. Note that the API returns a void *. This is done as the format attribute modules themselves may store format attributes in any particular manner they like. Care should be taken by consumers of the API to check the return value before casting and dereferencing. Consumers will obviously need to have a priori knowledge of the type of the format attribute as well. Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3	10 years ago
Joshua Colp	feffafbd55	Merge "Replace htobe64 with htonll"	10 years ago
Joshua Colp	ff36b5482b	Merge "Replaces clock_gettime() with ast_tsnow()"	10 years ago
Matt Jordan	35fe8f74fe	Merge "rtp_engine.c: Fix performance issue with several channel drivers that use RTP."	10 years ago
David M. Lee	d5f0c27122	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
David M. Lee	40caf0ad9b	Replaces clock_gettime() with ast_tsnow() ... clock_gettime() is, unfortunately, not portable. But I did like that over our usual `ts.tv_nsec = tv.tv_usec * 1000` copy/paste code we usually do when we want a timespec and all we have is ast_tvnow(). This patch adds ast_tsnow(), which mimics ast_tvnow(), but returns a timespec. If clock_gettime() is available, it will use that. Otherwise ast_tsnow() falls back to using ast_tvnow(). Change-Id: Ibb1ee67ccf4826b9b76d5a5eb62e90b29b6c456e	10 years ago
Scott Emidy	12e6f5ac01	ARI: Retrieve existing log channels ... An http request can be sent to get the existing Asterisk logs. The command "curl -v -u user:pass -X GET 'http://localhost:8088 /ari/asterisk/logging'" can be run in the terminal to access the newly implemented functionality. * Retrieve all existing log channels ASTERISK-25252 Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808	10 years ago
Scott Emidy	b91ca7ba49	ARI: Creating log channels ... An http request can be sent to create a log channel in Asterisk. The command "curl -v -u user:pass -X POST 'http://localhost:088/ari/asterisk/logging/mylog? configuration=notice,warning'" can be run in the terminal to access the newly implemented functionality for ARI. * Ability to create log channels using ARI ASTERISK-25252 Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782	10 years ago
Joshua Colp	ecd4cde521	Merge "ARI: Deleting log channels"	10 years ago
Joshua Colp	1b89cbb3b0	Merge "res_pjsip: Ensure sanitized XML is NULL terminated."	10 years ago
Joshua Colp	12d7c8a740	Merge "res_pjsip_pubsub: More accurately persist packet."	10 years ago
Joshua Colp	58effbc3f6	Merge "res_rtp_asterisk.c: Fix off-nominal crash potential."	10 years ago
Scott Emidy	f19c4930c2	ARI: Deleting log channels ... An http request can be sent to delete a log channel in Asterisk. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/logging/mylog'" can be run in the terminal to access the newly implemented functionally for ARI. * Able to delete log channels using ARI ASTERISK-25252 Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6	10 years ago
Mark Michelson	382334cc06	res_pjsip_pubsub: More accurately persist packet. ... The pjsip_rx_data structure has a pkt_info.packet field on it that is the packet that was read from the transport. For datagram transports, the packet read from the transport will correspond to the SIP message that arrived. For streamed transports, however, it is possible to read multiple SIP messages in one packet. In a recent case, Asterisk crashed on a system where TCP was being used. This is because at some point, a read from the TCP socket resulted in a 200 OK response as well as an incoming SUBSCRIBE request being stored in rdata->pkt_info.packet. When the SUBSCRIBE was processed, the combination 200 OK and SUBSCRIBE was saved in persistent storage. Later, a restart of Asterisk resulted in the crash because the persistent subscription recreation code ended up building the 200 OK response instead of a SUBSCRIBE request, and we attempted to access request-specific data. The fix here is to use the pjsip_msg_print() function in order to persist SUBSCRIBE requests. This way, rather than using the raw socket data, we use the parsed SIP message that PJSIP has given us. If we receive multiple SIP messages from a single read, we will be sure only to save off the relevant SIP message. There also is a safeguard put in place to make sure that if we do end up reconstructing a SIP response, it will not cause a crash. ASTERISK-25306 #close Reported by Mark Michelson Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2	10 years ago
Joshua Colp	ca84a4b235	Merge "rtp_engine.c: Must protect mime_types_len with mime_types_lock."	10 years ago
Joshua Colp	7ebca6795b	Merge "res_pjsip_sdp_rtp.c: Fixup some whitespace."	10 years ago
Joshua Colp	4b6c657a82	res_pjsip: Ensure sanitized XML is NULL terminated. ... The ast_sip_sanitize_xml function is used to sanitize a string for placement into XML. This is done by examining an input string and then appending values to an output buffer. The function used by its implementation, strncat, has specific behavior that was not taken into account. If the size of the input string exceeded the available output buffer size it was possible for the sanitization function to write past the output buffer itself causing a crash. The crash would either occur because it was writing into memory it shouldn't be or because the resulting string was not NULL terminated. This change keeps count of how much remaining space is available in the output buffer for text and only allows strncat to use that amount. Since this was exposed by the res_pjsip_pidf_digium_body_supplement module attempting to send a large message the maximum allowed message size has also been increased in it. A unit test has also been added which confirms that the ast_sip_sanitize_xml function is providing NULL terminated output even when the input length exceeds the output buffer size. ASTERISK-25304 #close Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302	10 years ago
Joshua Colp	3d4db97253	Merge "res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list."	10 years ago
Joshua Colp	8a48c9e6cb	Merge "res_http_websocket: Debug write lengths."	10 years ago
Joshua Colp	7351d33a1f	res_rtp_asterisk: Don't leak temporary key when enabling PFS. ... A change recently went in which enabled perfect forward secrecy for DTLS in res_rtp_asterisk. This was accomplished two different ways depending on the availability of a feature in OpenSSL. The fallback method created a temporary instance of a key but did not free it. This change fixes that. ASTERISK-25265 Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396	10 years ago
Mark Michelson	c63316eec1	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
Matt Jordan	3ba6099a9e	Merge "res_http_websocket: Avoid passing strlen() to ast_websocket_write()."	10 years ago
Matt Jordan	8672f0bbbd	Merge "res/res_rtp_asterisk: Add ECDH support"	10 years ago
Mark Michelson	35a98161df	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
Joshua Colp	8eef7392c4	Merge topic 'misc_rtp_tweaks' ... * changes: rtp_engine.h: No sense allowing payload types larger than RFC allows. rtp_engine.c: Minor tweaks. rtp_engine.h: Misc comment fixes.	10 years ago
Joshua Colp	4cf21fed27	Merge "chan_sip.c: Tweak glue->update_peer() parameter nil value."	10 years ago
Mark Michelson	b002e09214	Merge "ARI: Channels added to Stasis application during WebSocket creation ..."	10 years ago
Mark Michelson	92ddda68aa	Merge "ARI: Rotate log channels."	10 years ago
Joshua Colp	94f7427b17	Merge "res_pjsip_session.c: Fix crashes seen when call cancelled."	10 years ago
Benjamin Ford	1f02d20da4	ARI: Rotate log channels. ... An http request can be sent to rotate a specified log channel. If the channel does not exist, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/logging/logChannelName/rotate'" can be run in the terminal to access this new functionality. * Added the ability to rotate log files through ARI ASTERISK-25252 Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01	10 years ago
Joshua Colp	d61ea4bcc2	Merge "dns_core: Allow zero-length DNS responses."	10 years ago
Ashley Sanders	fe804b09b3	ARI: Channels added to Stasis application during WebSocket creation ... ... Prior to ASTERISK-24988, the WebSocket handshake was resolved before Stasis applications were registered. This was done such that the WebSocket would be ready when an application is registered. However, by creating the WebSocket first, the client had the ability to make requests for the Stasis application it thought had been created with the initial handshake request. The inevitable conclusion of this scenario was the cart being put before the horse. ASTERISK-24988 resolved half of the problem by ensuring that the applications were created and registered with Stasis prior to completing the handshake with the client. While this meant that Stasis was ready when the client received the green-light from Asterisk, it also meant that the WebSocket was not yet ready for Stasis to dispatch messages. This patch introduces a message queuing mechanism for delaying messages from Stasis applications while the WebSocket is being constructed. When the ARI event processor receives the message from the WebSocket that it is being created, the event processor instantiates an event session which contains a message queue. It then tries to create and register the requested applications with Stasis. Messages that are dispatched from Stasis between this point and the point at which the event processor is notified the WebSocket is ready, are stashed in the queue. Once the WebSocket has been built, the queue's messages are dispatched in the order in which they were originally received and the queue is concurrently cleared. ASTERISK-25181 #close Reported By: Matt Jordan Change-Id: Iafef7b85a2e0bf78c114db4c87ffc3d16d671a17	10 years ago
Mark Michelson	86034227ca	dns_core: Allow zero-length DNS responses. ... A testsuite test recently failed due to a crash that occurred in the DNS core. The problem was that the test could not resolve an address, did not set a result on the DNS query, and then indicated the query was completed. The DNS core does not handle the case of a query with no result gracefully, and so there is a crash. This changeset makes the DNS system resolver set a result with a zero-length answer in the case that a DNS resolution failure occurs early. The DNS core now also will accept such a response without treating it as invalid input. A unit test was updated to no longer treat setting a zero-length response as off-nominal. Change-Id: Ie56641e22debdaa61459e1c9a042e23b78affbf6	10 years ago
Richard Mudgett	f49bef08a2	rtp_engine.c: Fix performance issue with several channel drivers that use RTP. ... ast_rtp_codecs_get_payload() gets called once or twice for every received RTP frame so it would be nice to not allocate an ao2 object to then have it destroyed shortly thereafter. The ao2 object gets allocated only if the payload type is not set by the channel driver as a negotiated value. The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323. * Made static_RTP_PT[] an array of ao2 objects that ast_rtp_codecs_get_payload() can return instead of an array of structs that must be copied into a created ao2 object. ASTERISK-25296 #close Reported by: Richard Mudgett Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0	10 years ago
Richard Mudgett	33a465249b	res_rtp_asterisk.c: Fix off-nominal crash potential. ... ASTERISK-25296 Reported by: Richard Mudgett Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b	10 years ago
Richard Mudgett	5f925d48b7	rtp_engine.c: Must protect mime_types_len with mime_types_lock. ... Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e	10 years ago
Richard Mudgett	ba7dd38470	res_pjsip_sdp_rtp.c: Fixup some whitespace. ... Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973	10 years ago
Richard Mudgett	3751bf0971	res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list. ... Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2	10 years ago