4089 Commits (ac331bff34b037d1ca4dfb431979aaf798dce336)

Author	SHA1	Message	Date
Corey Farrell	fec6e1bd87	core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option. ... Previous to this patch passing a NULL tag to ao2_alloc or ao2_ref based functions would result in the reference not being logged under REF_DEBUG. This could sometimes cause inaccurate logging if NULL was accidentally passed to a reference action. Now reference logging is only disabled by option passed to the allocation method. Change-Id: I3c17d867d901d53f9fcd512bef4d52e342637b54	6 years ago
Friendly Automation	a687c7919d	Merge "func_jitterbuffer: Add audio/video sync support." into 16	6 years ago
Joshua Colp	120f2fb5f9	Merge "core: Add H.265/HEVC passthrough support" into 16	6 years ago
Joshua Colp	6647be69ac	func_jitterbuffer: Add audio/video sync support. ... This change adds support to the JITTERBUFFER dialplan function for audio and video synchronization. When enabled the RTCP SR report is used to produce an NTP timestamp for both the audio and video streams. Using this information the video frames are queued until their NTP timestamp is equal to or behind the NTP timestamp of the audio. The audio jitterbuffer acts as the leader deciding when to shrink/grow the jitterbuffer when adaptive is in use. For both adaptive and fixed the video buffer follows the size of the audio jitterbuffer. ASTERISK-28533 Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492	6 years ago
George Joseph	913c8b48b7	Merge "channels: Allow updating variable value" into 16	6 years ago
Sean Bright	518b6bfb5c	channels: Allow updating variable value ... When modifying an already defined variable in some channel drivers they add a new variable with the same name to the list, but that value is never used, only the first one found. Introduce ast_variable_list_replace() and use it where appropriate. ASTERISK-23756 #close Patches: setvar-multiplie.patch submitted by Michael Goryainov Change-Id: Ie1897a96c82b8945e752733612ee963686f32839	6 years ago
Florian Floimair	f85631cf82	core: Add H.265/HEVC passthrough support ... This change adds H.265/HEVC as a known codec and creates a cached "h265" media format for use. Note that RFC 7798 section 7.2 also describes additional SDP parameters. Handling of these is not yet supported. ASTERISK-28512 Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2	6 years ago
Ben Ford	922d3e02df	res_rtp: Add unit tests for RTCP stats. ... Added unit tests for RTCP video stats. These tests include NACK, REMB, FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR tests are currently disabled due to a bug. We expect to receive a compound packet, but the code sends this out as a single packet, which the browser accepts, but makes Asterisk upset. While writing these tests, I noticed an issue with NACK as well. Where it is handling a received NACK request, it was reading in only the first 8 bits of following packets that were also lost. This has been changed to the correct value of 16 bits. Also made a minor fix to the data buffer unit test. Change-Id: I56107c7411003a247589bbb6086d25c54719901b	6 years ago
George Joseph	6407ccd2d9	dns_core: Create new API ast_dns_resolve_ipv6_and_ipv4 ... The new function takes in a pointer to an ast_sockaddr structure, a hostname and an optional port and then dispatches parallel "AAAA" and "A" record queries. If an "AAAA" record is returned, it's parsed into the ast_sockaddr structure along with the port if it was supplied. If no "AAAA" record was returned, the first "A" record returned (if any) is parsed instead. This is a synchronous call. If you need asynchronous lookups, use ast_dns_query_set_resolve_async and roll your own. Change-Id: I194b0b0e73da94b35cc35263a868ffac3a8d0a95	7 years ago
Sean Bright	51fd43206b	audiohook.c: Substitute silence for unavailable audio frames ... There are 4 scenarios to consider when capturing audio from a channel with an audiohook: 1. There is no rx and no tx audio, so return nothing. 2. There is rx but no tx audio, so return rx. 3. There is tx but no rx audio, so return tx. 4. There is rx and tx audio, so mix them and return. The file passed as the primary argument to MixMonitor will be written to in scenarios 2, 3, and 4. However, if you pass the r() and t() options to MixMonitor, a frame will only be written to the r() file if there was rx audio and a frame will only be written to the t() file if there was tx audio. If you subsequently take the r() and t() files and try to mix them, the sides of the conversation will 'drift' and be non-representative of the user experience. This patch adds a new 'S' option to MixMonitor that injects a frame of silence on either the r() side or the t() side of the channel so that when later mixed, there is no such drift. Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e	7 years ago
Joshua Colp	6350f4e278	cdr / cel: Use event time at event creation instead of processing. ... When updating times on CDR or CEL records using the time at which it is done can result in times being incorrect if the system is heavily loaded and stasis message processing is delayed. This change instead makes it so CDR and CEL use the time at which the stasis messages that drive the systems are created. This allows them to be backed up while still producing correct records. ASTERISK-28498 Change-Id: I6829227e67aefa318efe5e183a94d4a1b4e8500a	7 years ago
Walter Doekes	64d25d36fb	sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread ... When fixing ASTERISK~24212, a change was done so a scheduled callback could not be removed while it was running. The caller of ast_sched_del would have to wait. However, when the caller of ast_sched_del is the callback itself (however wrong this might be), this new check would cause a deadlock: it would wait forever for itself. This changeset introduces an additional check: if ast_sched_del is called by the callback itself, it is immediately rejected (along with an ERROR log and a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the after-ast_sched_del-refcall function is only run if ast_sched_del returned success. This should fix the following spurious race condition found in chan_sip: - thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE) - thread 2: run sip_poke_peer_now - thread 2: blank out sched-ID (too soon!) - thread 1: set sched-ID (too late!) - thread 2: try to delete the currently running sched-ID After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and other madness) should occur. (Thanks Richard Mudgett for reviewing/improving this "scary" change.) Note that this change does not fix the observed race condition: unlocked access to peer->pokeexpire (and potentially other scheduled items in chan_sip), causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the deadlock go away. And in the observed case, it will not have adverse affects (like memory leaks) because the scheduled item is removed through a different path. ASTERISK-28282 Change-Id: Ic26777fa0732725e6ca7010df17af77a012aa856	7 years ago
Friendly Automation	a464847a1b	Merge "app_confbridge: Attended transfer event fixup" into 16	7 years ago
George Joseph	ccc92b6ecb	app_confbridge: Attended transfer event fixup ... When a channel already in a conference bridge is attended transfered to another extension, or when an existing call is attended transferred into a conference bridge, we now generate ConfbridgeJoin and ConfbridgeLeave events for the entering and departing channels. Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1	7 years ago
Joshua Colp	82789aafd6	res_rtp_asterisk: Add support for DTLS packet fragmentation. ... This change adds support for larger TLS certificates by allowing OpenSSL to fragment the DTLS packets according to the configured MTU. By default this is set to 1200. This is accomplished by implementing our own BIO method that supports MTU querying. The configured MTU is returned to OpenSSL which fragments the packet accordingly. When a packet is to be sent it is done directly out the RTP instance. ASTERISK-28018 Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06	7 years ago
Kevin Harwell	edc3e0df1a	conversions.c: Add conversions for largest max sized integer ... Added a conversion for umax (largest maximum sized integer allowed). Adjusted the other current conversion functions (uint and ulong) to be derivatives of the umax conversion since they are simply subsets of umax. Also made the negative check move the pointer on spaces since strtoumax does it anyways. Change-Id: I56c2ef2629d49b524c8df58af12951c181f81f08	7 years ago
Joshua Colp	8357ab7e9a	Merge "app_confbridge: Add "all" variants of REMB behavior." into 16	7 years ago
Friendly Automation	748f1d64a1	Merge "stasis: Call callbacks when imparting fails" into 16	7 years ago
Friendly Automation	7bddfdbfa6	Merge "rtp: Add support for transport-cc in receiver direction." into 16	7 years ago
Holger Hans Peter Freyther	f599ebd29e	stasis: Call callbacks when imparting fails ... After a bridge has been deleted the stasis control will depart the channel and might attempt to re-add it to the dial bridge. The later can fail and this can lead to a situation that the stasis control is unlinked but the after_bridge_cb_failed cb is executed trying to access a dangling control object. Fix it by calling the after_cb's before bridge_channel_impart_signal. ASTERISK-26718 Change-Id: Ib4e8f70d7a21bd54afe3cb51cc6717ef7c355496	7 years ago
Joshua Colp	d861ebdca8	app_confbridge: Add "all" variants of REMB behavior. ... When producing a combined REMB value the normal behavior is to have a REMB value which is unique for each sender based on all of their receivers. This can result in one sender having low bitrate while all the rest are high. This change adds "all" variants which produces a bridge level REMB value instead. All REMB reports are combined together into a single REMB value that is the same for each sender. ASTERISK-28401 Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c	7 years ago
Joshua Colp	5023f02b2d	rtp: Add support for transport-cc in receiver direction. ... The transport-cc draft is a mechanism by which additional information about packet reception can be provided to the sender of packets so they can do sender side bandwidth estimation. This is accomplished by having a transport specific sequence number and an RTCP feedback message. This change implements this in the receiver direction. For each received RTP packet where transport-cc is negotiated we store the time at which the RTP packet was received and its sequence number. At a 1 second interval we go through all packets in that period of time and use the stored time of each in comparison to its preceding packet to calculate its delta. This delta information is placed in the RTCP feedback message, along with indicators for any packets which were not received. The browser then uses this information to better estimate available bandwidth and adjust accordingly. This may result in it lowering the available send bandwidth or adjusting how "bursty" it can be. ASTERISK-28400 Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc	7 years ago
Friendly Automation	f2cb892d7c	Merge "mwi core: Move core MWI functionality into its own files" into 16	7 years ago
Antoni Goldstein	d6b37e2926	app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings ... Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled at the earliest received PROGRESS or RINGING. Added millisecond versions of DIALEDTIME and ANSWEREDTIME. Added millisecond versions of ast_channel_get_up_time and ast_channel_get_duration in channel.c. ASTERISK-28363 Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1	7 years ago
Kevin Harwell	e3a758975d	mwi core: Move core MWI functionality into its own files ... There is enough MWI functionality to warrant it having its own 'c' and header files. This patch moves all current core MWI data structures, and functions into the following files: main/mwi.h main/mwi.c Note, code was simply moved, and not modified. However, this patch is also in preparation for core MWI changes, and additions to come. Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0	7 years ago
Friendly Automation	6ec4b113c1	Merge "loader: support for permanent dlopen()" into 16	7 years ago
Dan Cropp	eca8c440d2	res_pjsip: Added a norefersub configuration setting ... Added a new PJSIP global setting called norefersub. Default is true to keep support working as before. res_pjsip_refer: Configures PJSIP norefersub capability accordingly. Checks the PJSIP global setting value. If it is true (default) it adds the norefersub capability to PJSIP. If it is false (disabled) it does not add the norefersub capability to PJSIP. This is useful for Cisco switches that do not follow RFC4488. ASTERISK-28375 #close Reported-by: Dan Cropp Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9	7 years ago
George Joseph	1988cf795c	Merge "stasis.c: Added topic_all container" into 16	7 years ago
Friendly Automation	777ee8290a	Merge "main/json.c: Added app_name, app_data to channel type" into 16	7 years ago
sungtae kim	d5a318f148	main/json.c: Added app_name, app_data to channel type ... It was difficult to check the channel's current application and parameters using ARI for current channels. Added app_name, app_data items to show the current application information. ASTERISK-28343 Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c	7 years ago
Sebastian Kemper	8ec4de7501	loader: support for permanent dlopen() ... Asterisk assumes that dlopen() will always run the constructor of a shared library and every dlclose() will run its destructor. But dlopen() may be permanent, meaning the constructor will only be run once, as is the case with musl libc. With a permanent dlopen() the Asterisk module loader does not work correctly, because it's expectations regarding when the constructors and destructors are run are not met. In fact a segmentation fault will occur when the first module is "re-opened" that has AST_MODFLAG_GLOBAL_SYMBOLS set (the dlopen() does not call the constructor, resource_being_loaded is not set to NULL, then strlen is called with NULL instead of a string, see issue ASTERISK-28319). This commit adds code to the loader that will manually run the constructors/destructors of the (non-builtin) modules where needed. To achieve this a new ao2 container (linked list) is started and filled with objects that contain the names of the modules and the pointers to their respective info structs. This behavior can be activated when configuring Asterisk (--enable-permanent-dlopen). By default this is disabled, of course. ASTERISK-28319 #close Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net> Change-Id: I86693a0ecf25d5ba81c73773a03df4abc3426875	7 years ago
sungtae kim	5815597a21	stasis.c: Added topic_all container ... Added topic_all container for centralizing the topic. This makes more easier to managing the topics. Added cli commands. stasis show topics : It shows all registered topics. stasis show topic <name> : It shows speicifed topic's detail info. ASTERISK-28264 Change-Id: Ie86d125d2966f93de74ee00f47ae6fbc8c081c5f	7 years ago
sungtae kim	bbc13b1f1f	res/res_ari: Added ARI resource /ari/channels/{channelId}/rtp_statistics ... Added ARI resource for channel statistics. GET /ari/channels/{channelId}/rtp_statistics : It returns given channel's rtp statistics detail. ASTERISK-28320 Change-Id: I4343eec070438cec13f2a4f22e7fd9e574381376	7 years ago
Matthew Fredrickson	ce5bed44e4	main/taskprocessor: Increase max name length of taskprocessors ... Since the new names went in, the maximum taskprocessor name is too short. This patch increases the name field to a length to better handle the new names. Change-Id: I32f32d6926f25c8ef5a91303fd2988d2c2858877	7 years ago
Friendly Automation	c7d29cdcf2	Merge "sorcery.c: Sorcery enhancements for wizard management" into 16	7 years ago
George Joseph	7bdde83f2c	sorcery.c: Sorcery enhancements for wizard management ... Added ability to specifiy a wizard is read-only when applying it to a specific object type. This allows you to specify create, update and delete callbacks for the wizard but limit which object types can use them. Added the ability to allow an object type to have multiple wizards of the same type. This is indicated when a wizard is added to a specific object type. Added 3 new sorcery wizard functions: * ast_sorcery_object_type_insert_wizard which does the same thing as the existing ast_sorcery_insert_wizard_mapping function but accepts the new read-only and allot-duplicates flags and also returns the ast_sorcery_wizard structure used and it's internal data structure. This allows immediate use of the wizard's callbacks without having to register a "wizard mapped" observer. * ast_sorcery_object_type_apply_wizard which does the same thing as the existing ast_sorcery_apply_wizard_mapping function but has the added capabilities of ast_sorcery_object_type_insert_wizard. * ast_sorcery_object_type_remove_wizard which removes a wizard matching both its name and its original argument string. * The original logic in __ast_sorcery_insert_wizard_mapping was moved to __ast_sorcery_object_type_insert_wizard and enhanced for the new capabilities, then __ast_sorcery_insert_wizard_mapping was refactored to just call __ast_sorcery_insert_wizard_mapping. * Added a unit test to test_sorcery.c to test the read-only capability. Change-Id: I40f35840252e4313d99e11dbd80e270a3aa10605	7 years ago
Sean Bright	057c7380be	vector: Add AST_VECTOR_COMPACT() to reclaim wasted space ... This might be useful in situations where you are loading an undetermined number of items into a vector and don't want to keep (potentially) 2x the necessary memory around indefinitely. Change-Id: I9711daa0fe01783fc6f04c5710eba84f2676d7b9	7 years ago
Joshua C. Colp	6f10c9cf4f	Merge "AMI/ARI: Bump non-breaking version numbers" into 16	7 years ago
Joshua C. Colp	3e5dcc9dcf	Merge "stasis: Improve topic/subscription names and statistics." into 16	7 years ago
Kevin Harwell	6a435723f7	AMI/ARI: Bump non-breaking version numbers ... Increasing the non-breaking AMI and ARI version numbers due to changes and additions in those API's. Note, some changes may be forthcoming (will be added between now and the next release of Asterisk), thus not listed here. As well a few changes listed below may have been released in a previous release of Asterisk, but the API version numbers were not increased at that time, so including here. AMI: * app_queue: set the wrapuptime from AddQueueMember application - e806990 * res_pjsip: option for ContactStatus event updates - 7f22c9f ARI: * bridging: Add creation timestamps - 0d70120 * res_stasis: Add ability to switch applications - 50a4b61 * ARI event type filtering - 1c5def4 * Added ARI resource /ari/asterisk/ping - 19fc99a ASTERISK-28314 Change-Id: Iebc813840f8230afa6b20579772e15549064b787	7 years ago
George Joseph	9b15e04011	Merge "jansson: json_pack with new format to verify required runtime version." into 16	7 years ago
Joshua Colp	b886fef66e	stasis: Improve topic/subscription names and statistics. ... Topic names now follow: <subsystem>:<functionality>[/<object>] This ensures that they are all unique, and also provides better insight in to what each topic is for. Subscriber ids now also use the main topic name they are subscribed to and an incrementing integer as their identifier to make it easier to understand what the subscription is primarily responsible for. Both the CLI commands for listing topic and subscription statistics now sort to make it a bit easier to see what is going on. Subscriptions will now show all topics that they are receiving messages from, not just the main topic they were subscribed to. ASTERISK-28335 Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d	7 years ago
Torrey Searle	4ca41caf38	chan_pjsip: add a flag to ignore 183 responses if no SDP present ... chan_sip will always ignore 183 responses that do not contain SDP however, chan_pjsip will currently always translate it into a 183 with SDP. This new flag allows chan_pjsip to have the same behavior as chan_sip. ASTERISK-28322 #close Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a	7 years ago
Corey Farrell	641b6041c7	jansson: json_pack with new format to verify required runtime version. ... Add a json_pack at startup that will fail if runtime links against a library older than jansson-2.11. Change-Id: I101aebafe0f9407650206f7c552dad3d69377b5a	7 years ago
Friendly Automation	afe16021ef	Merge "res_stasis: Add ability to switch applications." into 16	7 years ago
Friendly Automation	7566d9bb61	Merge "bridging: Add creation timestamps" into 16	7 years ago
Ben Ford	50a4b618fc	res_stasis: Add ability to switch applications. ... Added the ability to move between Stasis applications within Stasis. This can be done by calling 'move' in an application, providing (at minimum) the channel's id and the application to switch to. If the application is not registered or active, nothing will happen and the channel will remain in the current application, and an event will be triggered to let the application know that the move failed. The event name is "ApplicationMoveFailed", and provides the "destination" that the channel was attempting to move to, as well as the usual channel information. Optionally, a list of arguments can be passed to the function call for the receiving application. A full example of a 'move' call would look like this: client.channels.move(channelId, app, appArgs) The control object used to control the channel in Stasis can now switch which application it belongs to, rather than belonging to one Stasis application for its lifetime. This allows us to use the same control object instead of having to tear down the current one and create another. ASTERISK-28267 #close Change-Id: I43d12b10045a98a8d42541889b85695be26f288a	7 years ago
Joshua Colp	f9bc474ab0	Merge "res_pjsip_registrar: blocked threads on reliable transport shutdown take 3" into 16	7 years ago
sungtae kim	0d7012044a	bridging: Add creation timestamps ... This small feature will help to checking the bridge's status to figure out which bridge is in old/zombie or not. Also added detail items for the 'bridge show *' cli to provide more detail info. And added creation item to the ARI as well. ASTERISK-28279 Change-Id: I460238c488eca4d216b9176576211cb03286e040	7 years ago
Kevin Harwell	b3ebb51086	res_pjsip_registrar: blocked threads on reliable transport shutdown take 3 ... When a contact was removed by the registrar it did not always check to see if the circumstances involved a monitored reliable transport. For instance, if the 'remove_existing' option was set to 'true' then when existing contacts were removed due to 'max_contacts' being reached, those existing contacts being removed did not unregister the transport monitor. Also, it was possible to add more than one monitor on a reliable transport for a given aor and contact. This patch makes it so all contact removals done by the registrar also remove any associated transport monitors if necessary. It also makes it so duplicate monitors cannot be added for a given transport. ASTERISK-28213 Change-Id: I94b06f9026ed177d6adfd538317c784a42c1b17a	7 years ago