29326 Commits (19a79ae12c73992508910d2c7cddc059e10bc48c)

 

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Author	SHA1	Message	Date
Joshua Colp	19a79ae12c	sdp: Add support for T.38 ... This change adds a T.38 format which can be used in a stream topology to specify that a UDPTL stream needs to be created. The SDP API has been changed to understand T.38 and create the UDPTL session, add the attributes, and parse the attributes. This change does not change the boundary of the T.38 state machine. It is still up to the channel driver to implement and act on it (such as queueing control frames or reacting to them). ASTERISK-26949 Change-Id: If28956762ccb8ead562ac6c03d162d3d6014f2c7	9 years ago
Mark Michelson	32b3e36c68	SDP: Ensure SDPs "merge" properly. ... The gist of this work ensures that when a remote SDP is received, it is merged properly with the local capabilities. The remote SDP is converted into a stream topology. That topology is then merged with the current local topology on the SDP state. That new merged topology is then used to create an SDP. Finally, adjustments are made to RTP instances based on knowledge gained from the remote SDP. There are also a battery of tests in this commit that ensure that some basic SDP merges work as expected. While this may not sound like a big change, it has the property that it caused lots of ancillary changes. * The remote SDP is no longer stored on the SDP state. Biggest reason: there's no need for it. The remote SDP is used at the time it is being set and nowhere else. * Some new SDP APIs were added in order to find attributes and convert generic SDP attributes into rtpmap structures. * Writing tests made me realize that retrieving a value from an SDP options structure, the SDP options needs to be made const. * The SDP state machine was essentially gutted by a previous commit. Initially, I attempted to reinstate it, but I found that as it had been defined, it was not all that useful. What was more useful was knowing the role we play in SDP negotiation, so the SDP state machine has been transformed into an indicator of role. * Rather than storing separate local and joint stream state capabilities, it makes more sense to keep track of current stream state and update it as things change. Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d	9 years ago
Sean Bright	59203c51cc	core: Use eventfd for alert pipes on Linux when possible ... The primary win of switching to eventfd when possible is that it only uses a single file descriptor while pipe() will use two. This means for each bridge channel we're reducing the number of required file descriptors by 1, and - if you're using timerfd - we also now have 1 less file descriptor per Asterisk channel. The API is not ideal (passing int arrays), but this is the cleanest approach I could come up with to maintain API/ABI. I've also removed what I believe to be an erroneous code block that checked the non-blocking flag on the pipe ends for each read. If the file descriptor is 'losing' its non-blocking mode, it is because of a bug somewhere else in our code. In my testing I haven't seen any measurable difference in performance. Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d	9 years ago
George Joseph	dc6654d969	Merge "pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified"	9 years ago
George Joseph	cebfe85aff	Merge "rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes."	9 years ago
George Joseph	d373ebc428	Merge "build: Update config.guess and config.sub"	9 years ago
George Joseph	dd239e9f91	Merge "res_stun_monitor: Don't fail to load if DNS resolution fails"	9 years ago
George Joseph	b18bfb7672	Merge "make ari-stubs so doc periodic jobs can run"	9 years ago
Sean Bright	c47b3e74d2	pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified ... Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE is passed to these functions, the calling thread will be blocked until the newly created channel has been hung up. After this patch, we run the dial on the current thread rather than spawning a new one. The only in-tree code that passes AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced thread usage if you are using .call files. Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913	9 years ago
Richard Mudgett	d165079cbc	rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes. ... The struct ast_rtp_instance has historically been indirectly protected from reentrancy issues by the channel lock because early channel drivers held the lock for really long times. Holding the channel lock for such a long time has caused many deadlock problems in the past. Along comes chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock because sometimes there may not be an associated channel created yet or the channel pointer isn't available. In the case of ASTERISK-26835 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up changing the rtp->rtcp->local_addr_str string and interfering with each other. The classic reentrancy problem resulted in a crash. In the case of ASTERISK-26853 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up processing ICE candidates in PJPROJECT and interfering with each other. The classic reentrancy problem resulted in a crash. * rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP instance struct. * rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP instance struct for the API call. * res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy problem with rtp->rtcp->local_addr_str in the scheduler thread running ast_rtcp_write(). * res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in bridge_p2p_rtp_write() because there are two RTP instance structs involved. * res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler callbacks. We cannot hold the instance lock when trying to stop a scheduler callback. * res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads to prevent a race condition between starting and stopping a timeout timer. The race condition is no longer present between dtls_perform_handshake() and __rtp_recvfrom() because the instance lock prevents these functions from overlapping each other with regards to the timeout timer. * res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads using a condition signal to know when TURN negotiations complete. * res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN ioqueue_worker_thread(). We cannot hold the instance lock when trying to create or shut down the worker thread without a risk of deadlock. This patch exposed a race condition between a PJSIP serializer thread setting up an ICE session in ice_create() and another thread reading RTP packets. * res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we have re-locked the RTP instance to prevent the other thread from trying to process ICE packets on an incomplete ICE session setup. A similar race condition is between a PJSIP serializer thread resetting up an ICE session in ice_create() and the timer_worker_thread() processing the completion of the previous ICE session. * res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an uninitialized/null remote_address after calling update_address_with_ice_candidate(). * res_rtp_asterisk.c: Eliminate the chance of ice_reset_session() destroying and setting the rtp->ice pointer to NULL while other threads are using it by adding an ao2 wrapper around the PJPROJECT ice pointer. Now when we have to unlock the RTP instance object to call a PJPROJECT ICE function we will hold a ref to the wrapper. Also added some rtp->ice NULL checks after we relock the RTP instance and have to do something with the ICE structure. ASTERISK-26835 #close ASTERISK-26853 #close Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4	9 years ago
Sean Bright	b8b3380944	build: Update config.guess and config.sub ... Change-Id: Id078a1df07a771808775e1053cdfe1d99c8fb172	9 years ago
Joshua Colp	b84abac144	Merge "format_wav: Read 16khz wav samples properly"	9 years ago
Joshua Colp	e756a9e063	Merge "format_ogg_vorbis: Clear ogg/vorbis data structures on close"	9 years ago
Joshua Colp	b0c72da6fc	Merge "Revert "bridging: Ensure successful T.38 negotation""	9 years ago
Sean Bright	6c0ab9afa7	format_wav: Read 16khz wav samples properly ... When opening a PCM wave file for reading, we aren't tracking the frequency of the opened file, so we treat 16khz files as 8khz and do half reads. This patch also cleans up some of the data types and an unnecessarily complex `if` expression. ASTERISK-26613 #close Reported by: Vitaly K Change-Id: I05f8b263058dc573ea8ffe0c62e7964506e11815	9 years ago
George Joseph	b55d21ad91	make ari-stubs so doc periodic jobs can run ... The periodic doc job does a make ari-stubs and checks that there are no changes before generating the docs. Since I changed the mustache template (and the generated code directly) recently and forgot to regenerate the stubs, the doc job thinks they're out of date. Change-Id: I94b97035311eccf52b0101b8590223265a7881d4	9 years ago
Sean Bright	4fb9f5d60e	format_ogg_vorbis: Clear ogg/vorbis data structures on close ... On filestream close, we need to clear out the ogg & vorbis data structures to prevent a memory leak. ASTERISK-26169 #close Reported by: Ivan Myalkin Change-Id: Iee94c5a5d5bdafbf8b181c5c064d15d90ace8274	9 years ago
Richard Mudgett	a3e623dd70	Revert "bridging: Ensure successful T.38 negotation" ... This reverts commit 7819f95791. Change-Id: Ib91a7e6c9856f5f41329e42f40ba2394fee861a4	9 years ago
Sean Bright	f6600f2c2e	res_stun_monitor: Don't fail to load if DNS resolution fails ... res_stun_monitor will fail to load if DNS resolution of the STUN server fails. Instead, we continue without the STUN server being resolved and we will re-attempt the resolution on the STUN refresh interval. ASTERISK-21856 #close Reported by: Jeremy Kister Change-Id: I6334c54a1cc798f8a836b4b47948e0bb4ef59254	9 years ago
Sean Bright	be71be7ed2	format_pcm: Track actual header size of .au files ... Sun's Au file format has a minimum data offset 24 bytes, but this offset is encoded in each .au file. Instead of assuming the minimum, read the actual value and store it for later use. ASTERISK-20984 #close Reported by: Roman S. Patches: asterisk-1.8.20.0-au-clicks-2.diff (license #6474) patch uploaded by Roman S. Change-Id: I524022fb19ff2fd5af2cc2d669d27a780ab2057c	9 years ago
George Joseph	2e6075c51f	modules: change module LOAD_FAILUREs to LOAD_DECLINES (master) ... Change-Id: Iac40ecb20e10513d67bf0eaf61807f306067b258	9 years ago
Joshua Colp	49535c17fc	Merge "modules: change module LOAD_FAILUREs to LOAD_DECLINES (14)"	9 years ago
zuul	e803eafb65	Merge "modules: change module LOAD_FAILUREs to LOAD_DECLINES"	9 years ago
Joshua Colp	f4ed700ca2	Merge "strings.h: Avoid overflows in the string hash functions"	9 years ago
George Joseph	6db0939b96	modules: change module LOAD_FAILUREs to LOAD_DECLINES (14) ... Change-Id: If99e3b4fc2d7e86fc3e61182aa6c835b407ed49e	9 years ago
George Joseph	747beb1ed1	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
zuul	60fd01f7f4	Merge "bridging: Ensure successful T.38 negotation"	9 years ago
zuul	1d15e95440	Merge "res_rtp_asterisk.c: Add stun_blacklist option"	9 years ago
Torrey Searle	7819f95791	bridging: Ensure successful T.38 negotation ... When a T.38 happens immediatly after call establishment, the control frame can be lost because the other leg is not yet in the bridge. This patch detects this case an makes sure T.38 negotation happens when the 2nd leg is being made compatible with the negotating first leg ASTERISK-26923 #close Change-Id: If334125ee61ed63550d242fc9efe7987e37e1d94	9 years ago
Joshua Colp	c5536eaee9	Merge "stun.c: Fix ast_stun_request() erratic timeout."	9 years ago
zuul	3c32680a8a	Merge "sorcery.c: Speed up ast_sorcery_retrieve_by_id()"	9 years ago
zuul	ecde4f1593	Merge "res_pjsip: Fix pointer use after unref."	9 years ago
zuul	a0220877fa	Merge "res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member."	9 years ago
Torrey Searle	7901225261	strings.h: Avoid overflows in the string hash functions ... On 2's compliment machines abs(INT_MIN) behavior is undefined and results in a negative value still being returnd. This results in negative hash codes that can result in crashes. ASTERISK-26528 #close Change-Id: Idff550145ca2133792a61a2e212b4a3e82c6517b	9 years ago
Richard Mudgett	7312cbe803	res_rtp_asterisk.c: Add stun_blacklist option ... Added the stun_blacklist option to rtp.conf. Some multihomed servers have IP interfaces that cannot reach the STUN server specified by stunaddr. Blacklist those interface subnets from trying to send a STUN packet to find the external IP address. Attempting to send the STUN packet needlessly delays processing incoming and outgoing SIP INVITEs because we will wait for a response that can never come until we give up on the response. Multiple subnets may be listed. ASTERISK-26890 #close Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342	9 years ago
Richard Mudgett	7c37365f03	stun.c: Fix ast_stun_request() erratic timeout. ... If ast_stun_request() receives packets other than a STUN response then we could conceivably never exit if we continue to receive packets with less than three seconds between them. * Fix poll timeout to keep track of the time when we sent the STUN request. We will now send a STUN request every three seconds regardless of how many other packets we receive while waiting for a response until we have completed three STUN request transmission cycles. Change-Id: Ib606cb08585e06eb50877f67b8d3bd385a85c266	9 years ago
Richard Mudgett	8d323c74fa	sorcery.c: Speed up ast_sorcery_retrieve_by_id() ... Return early if ast_sorcery_retrieve_by_id() is not passed an id to find. Also eliminated the RAII_VAR() usage in the function. Change-Id: I871dbe162a301b5ced8b4393cec27180c7c6b218	9 years ago
Richard Mudgett	5b4e2ec267	res_pjsip: Fix pointer use after unref. ... Change-Id: I4b6e1b0070563eeaee223cb58326f1b962ed5bc1	9 years ago
Richard Mudgett	6f793ac149	res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member. ... * create_rtp(): Eliminate use of deprecated transport struct member. That member and several others in the transport structure were deprecated because of an infinite loop created when using realtime configuration. See 2451d4e455 ASTERISK-26851 Change-Id: I0533aa13c9ce3c6cc394e0fd2b5bf1cd1b2ef3bc	9 years ago
Richard Mudgett	d76bc0565c	tcptls.c: Cleanup TCP/TLS listener thread on abnormal exit. ... Temporarily running out of file descriptors should not terminate the listener thread. Otherwise, when there becomes more file descriptors available, nothing is listening. * Added EMFILE exception to abnormal thread exit. * Added an abnormal TCP/TLS listener exit error message. * Closed the TCP/TLS listener socket on abnormal exit so Asterisk does not appear dead if something tries to connect to the socket. ASTERISK-26903 #close Change-Id: I10f2f784065136277f271159f0925927194581b5	9 years ago
Walter Doekes	2b8dbc9e00	samples: Undo removal of include from canonicalize-app-names commit. ... This include was accidentally removed in changeset Ia79aea64de89531362e993e34230c2044a70aa93. My bad. Change-Id: I1d716c7f9590b4e97909fb8bca1f2ed9bd0e4082	9 years ago
Joshua Colp	62386dd1df	Merge "pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete()."	9 years ago
zuul	6ac1c686ca	Merge "pjsip: Add Alembic for PUBLISH support."	9 years ago
Joshua Colp	a1a2e16f67	Merge "samples: Canonicalize app names in extensions.conf.sample."	9 years ago
Joshua Colp	0e7d29501d	Merge "core: Improve/simplify handling of required headers."	9 years ago
Joshua Colp	270b485f04	pjsip: Add Alembic for PUBLISH support. ... This change adds database tables for the PUBLISH support so it can be configured using realtime. A minor fix to the res_pjsip_publish_asterisk module was done so that it read the sorcery configuration from the correct section. Finally the sample configuration files have been updated. ASTERISK-26928 Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952	9 years ago
Alexander Traud	7a46cd7433	pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). ... When the Asterisk channel driver res_pjsip offers SIP-over-TLS, sometimes, not reproducible, Asterisk crashed in pj_ssl_sock_get_info() because a NULL pointer was read. This change avoids this crash. ASTERISK-26927 #close Change-Id: I24a6011b44d1426d159742ff4421cf806a52938b	9 years ago
Walter Doekes	e6ae3651b8	samples: Canonicalize app names in extensions.conf.sample. ... This takes care of warnings by ossobv/asterisklint. Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93	9 years ago
Joshua Colp	33f29240da	Merge "pjproject_bundled: Add 3 upstream patches"	9 years ago
Joshua Colp	b12eeddb35	Merge "chan_sip: Session Timers required but refused wrongly."	9 years ago