asterisk

Commit Graph

Author	SHA1	Message	Date
Jenkins2	1bec535df2	Merge "Core: Add support for systemd socket activation."	9 years ago
Joshua Colp	b610295b62	Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues."	9 years ago
Joshua Colp	8412cc1e07	Merge "SDP: Rework SDP offer/answer model and update capabilities merges."	9 years ago
Joshua Colp	a6eb9ee7d2	core: Add VP9 passthrough support. This change adds VP9 as a known codec and creates a cached "vp9" media format for use. Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc	9 years ago
Matthew Fredrickson	2697e45157	format.h: Fix a few minor errors in comments. A few minor problems were found in comments in format.h. This patch fixes them. Change-Id: I07f0bdb47b93359b361c4c3d8ecc87cd3199dd94	9 years ago
George Joseph	3e8d628c0e	Update AMI and ARI versions for master/15 and update UPDATE.txt AMI goes from 3.2.0 to 4.0.0 ARI goes from 2.0.0 to 3.0.0 Copied UPGRADE.txt -> UPGRADE-15.txt Created new UPGRADE.txt Removed a log file that was accidentally checked in a while ago Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7	9 years ago
Joshua Colp	f43fc91911	Merge "core: Add digit filtering to ast_waitfordigit_full"	9 years ago
Jenkins2	62c381afdb	Merge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO."	9 years ago
Joshua Colp	680c491a62	bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. This change does a few things to improve packet loss and renegotiation: 1. On outgoing RTP streams we will now properly reflect out of order packets and packet loss in the sequence number. This allows the remote jitterbuffer to better reorder things. 2. Video updates can now be discarded for a period of time after one has been sent to prevent flooding of clients. 3. For declined and removed streams we will now release any media session resources associated with them. This was not previously done and caused an issue where old state was being used for a new stream. 4. RTP bundling was not actually removing bundled RTP instances from the parent. This has been resolved by removing based on the RTP instance itself and not the SSRC. 5. The code did not properly handle explicitly unbundling an RTP instance from its parent. This now works as expected. ASTERISK-27143 Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45	9 years ago
Jenkins2	594c7a50af	Merge "bridge/core_unreal: Fix SFU bugs with forwarding frames."	9 years ago
Kevin Harwell	7da6ddda30	res_pjsip: Add "webrtc" configuration option This patch creates a new configuration option called "webrtc". When enabled it defaults and enables the following options that are needed in order for webrtc to work in Asterisk: rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled media_encryption=dtls dtls_verify=fingerprint dtls_setup=actpass When "webrtc" is enabled, this patch also parses the "msid" media level attribute from an SDP. It will also appropriately add it onto the outgoing session when applicable. Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent. ASTERISK-27119 #close Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd	9 years ago
Jenkins2	0f45c979a3	Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."	9 years ago
Corey Farrell	78a50b0343	core: Add PARSE_TIMELEN support to ast_parse_arg and ACO. This adds support for parsing timelen values from config files. This includes support for all flags which apply to PARSE_INT32. Support for this parser is added to ACO via the OPT_TIMELEN_T option type. Fixes an issue where extra characters provided to ast_app_parse_timelen were ignored, they now cause an error. Testing is included. ASTERISK-27117 #close Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554	9 years ago
Joshua Colp	065c3005ad	res_rtp_asterisk / res_pjsip: Add support for BUNDLE. BUNDLE is a specification used in WebRTC to allow multiple streams to use the same underlying transport. This reduces the number of ICE and DTLS negotiations that has to occur to 1 normally. This change implements this by adding support for it to the RTP SDP module in PJSIP. BUNDLE can be turned on using the "bundle" option and on an offer we will offer to bundle streams together. On an answer we will accept any bundle groups provided. Once accepted each stream is bundled to another RTP instance for transport. For the res_rtp_asterisk changes the ability to bundle an RTP instance to another based on the SSRC received from the remote side has been added. For outgoing traffic if an RTP instance is bundled to another we will use the other RTP instance for any transport related things. For incoming traffic received from the transport instance we look up the correct instance based on the SSRC and use it for any non-transport related data. ASTERISK-27118 Change-Id: I96c0920b9f9aca7382256484765a239017973c11	9 years ago
Corey Farrell	6b138046e7	core: Add digit filtering to ast_waitfordigit_full This adds a parameter to ast_waitfordigit_full which can be used to only stop waiting when certain expected digits are received. Any unexpected DTMF digits are simply ignored. This also creates a new dialplan application WaitDigit. ASTERISK-27129 #close Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9	9 years ago
Joshua Colp	7f09fd2c2f	bridge/core_unreal: Fix SFU bugs with forwarding frames. This change fixes a few things uncovered during SFU testing. 1. Unreal channels incorrectly forwarded video frames when no video stream was present on them. This caused a crash when they were read as the core requires a stream to exist for the underlying media type. The Unreal channel will now ensure a stream exists for the media type before forwarding the frame and if no stream exists then the frame is dropped. 2. Mapping of frames during bridging from the stream number of the underlying channel to the stream number of the bridge was done in the wrong location. This resulted in the frame getting dropped. This mapping now occurs on reading of the frame from the channel. 3. Bridging was using the wrong ast_read function resulting in it living in a non-multistream world. 4. In bridge_softmix when adding new streams to existing channels the wrong stream topology was copied resulting in no streams being added. Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8	9 years ago
Jenkins2	d6c08cc559	Merge "core: Remove 'Data Retrieval API'"	9 years ago
George Joseph	7a306468f4	Merge "bridge_native_rtp: Keep rtp instance refs on bridge_channel"	9 years ago
Sean Bright	325eeced6a	core: Remove 'Data Retrieval API' This API was not actively maintained, was not added to new modules (such as res_pjsip), and there exist better alternatives to acquire the same information, such as the ARI. Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83	9 years ago
Torrey Searle	fb7247c57c	res_pjsip: Add DTMF INFO Failback mode The existing auto dtmf mode reverts to inband if 4733 fails to be negotiated. This patch adds a new mode auto_info which will switch to INFO instead of inband if 4733 is not available. ASTERISK-27066 #close Change-Id: Id185b11e84afd9191a2f269e8443019047765e91	9 years ago
Mark Michelson	45df25a579	chan_pjsip: Add support for multiple streams of the same type. The stream topology (list of streams and order) is now stored with the configured PJSIP endpoints and used during the negotiation process. Media negotiation state information has been changed to be stored in a separate object. Two of these objects exist at any one time on a session. The active media state information is what was previously negotiated and the pending media state information is what the media state will become if negotiation succeeds. Streams and other state information is stored in this object using the index (or position) of each individual stream for easy lookup. The ability for a media type handler to specify a callback for writing has been added as well as the ability to add file descriptors with a callback which is invoked when data is available to be read on them. This allows media logic to live outside of the chan_pjsip module. Direct media has been changed so that only the first audio and video stream are directly connected. In the future once the RTP engine glue API has been updated to know about streams each individual stream can be directly connected as appropriate. Media negotiation itself will currently answer all the provided streams on an offer within configured limits and on an offer will use the topology created as a result of the disallow/allow codec lines. If a stream has been removed or declined we will now mark it as such within the resulting SDP. Applications can now also request that the stream topology change. If we are told to do so we will limit any provided formats to the ones configured on the endpoint and send a re-invite with the new topology. Two new configuration options have also been added to PJSIP endpoints: max_audio_streams: determines the maximum number of audio streams to offer/accept from an endpoint. Defaults to 1. max_video_streams: determines the maximum number of video streams to offer/accept from an endpoint. Defaults to 1. ASTERISK-27076 Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7	9 years ago
George Joseph	80e11bd79b	bridge_native_rtp: Keep rtp instance refs on bridge_channel There have been reports of deadlocks caused by an attempt to send a frame to a channel's rtp instance after the channel has left the native bridge and been destroyed. This patch effectively causes the bridge channel to keep a reference to the glue and both the audio and video rtp instances so what gets started will get stopped. ASTERISK-26978 #close Reported-by: Ross Beer Change-Id: I9e1ac49fa4af68d64826ccccd152593cf8cdb21a	9 years ago
Kevin Harwell	27dae55fb6	core_local: local channel data not being properly unref'ed and unlocked In an earlier version of Asterisk a local channel [un]lock all functions were added in order to keep a crash from occurring when a channel hung up too early during an attended transfer. Unfortunately, when a transfer failure occurs and depending on the timing, the local channels sometime do not get properly unlocked and deref'ed after being locked and ref'ed. This happens because the underlying local channel structure gets NULLed out before unlocking. This patch reworks those [un]lock functions and makes sure the values that get locked and ref'ed later get unlocked and deref'ed. ASTERISK-27074 #close Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09	9 years ago
Richard Mudgett	3a18a09030	SDP: Rework SDP offer/answer model and update capabilities merges. The SDP offer/answer model requires an answer to an offer before a new SDP can be processed. This allows our local SDP creation to be deferred until we know that we need to create an offer or an answer SDP. Once the local SDP is created it won't change until the SDP negotiation is restarted. An offer SDP in an initial SIP INVITE can receive more than one answer SDP. In this case, we need to merge each answer SDP with our original offer capabilities to get the currently negotiated capabilities. To satisfy this requirement means that we cannot update our proposed capabilities until the negotiations are restarted. Local topology updates from ast_sdp_state_update_local_topology() are merged together until the next offer SDP is created. These accumulated updates are then merged with the current negotiated capabilities to create the new proposed capabilities that the offer SDP is built. Local topology updates are merged in several passes to attempt to be smart about how streams from the system are matched with the previously negotiated stream slots. To allow for T.38 support when merging, type matching considers audio and image types to be equivalent. First streams are matched by stream name and type. Then streams are matched by stream type only. Any remaining unmatched existing streams are declined. Any new active streams are either backfilled into pre-merge declined slots or appended onto the end of the merged topology. Any excess new streams above the maximum supported number of streams are simply discarded. Remote topology negotiation merges depend if the topology is an offer or answer. An offer remote topology negotiation dictates the stream slot ordering and new streams can be added. A remote offer can do anything to the previously negotiated streams except reduce the number of stream slots. An answer remote topology negotiation is limited to what our offer requested. The answer can only decline streams, pick codecs from the offered list, or indicate the remote's stream hold state. I had originally kept the RTP instance if the remote offer SDP changed a stream type between audio and video since they both use RTP. However, I later removed this support in favor of simply creating a new RTP instance since the stream's purpose has to be changing anyway. Any RTP packets from the old stream type might cause mischief for the bridged peer. * Added ast_sdp_state_restart_negotiations() to restart the SDP offer/answer negotiations. We will thus know to create a new local SDP when it is time to create an offer or answer. * Removed ast_sdp_state_reset(). Save the current topology before starting T.38. To recover from T.38 simply update the local topology to the saved topology and restart the SDP negotiations to get the offer SDP renegotiating the previous configuration. * Allow initial topology for ast_sdp_state_alloc() to be NULL so an initial remote offer SDP can dictate the streams we start with. We can always update the local topology later if it turns out we need to offer SDP first because the remote chose to defer sending us a SDP. * Made the ast_sdp_state_alloc() initial topology limit to max_streams, limit to configured codecs, handle declined streams, and discard unsupported types. * Convert struct ast_sdp to ao2 object. Needed to easily save off a remote SDP to refer to later for various reasons such as generating declined m= lines in the local SDP. * Improve converting remote SDP streams to a topology including stream state. A stream state of AST_STREAM_STATE_REMOVED indicates the stream is declined/dead. * Improve merging streams to take into account the stream state. * Added query for remote hold state. * Added maximum streams allowed SDP config option. * Added ability to create new streams as needed. New streams are created with configured default audio, video, or image codecs depending on stream type. * Added global locally_held state along with a per stream local hold state. Historically, Asterisk only has a global locally held state because when the we put the remote on hold we do it for all active streams. * Added queries for a rejected offer and current SDP negotiation role. The rejected query allows the using module to know how to respond to a failed remote SDP set. Should the using module respond with a 488 Not Acceptable Here or 500 Internal Error to the offer SDP? * Moved sdp_state_capabilities.connection_address to ast_sdp_state. There seems no reason to keep it in the sdp_state_capabilities struct since it was only used by the ast_sdp_state.proposed_capabilities instance. * Callbacks are now available to allow the using module some customization of negotiated streams and to complete setting up streams for use. See the typedef doxygen for each callback for what is allowable and when they are called. * Added topology answerer modify callback. * Added topology pre and post apply callbacks. * Added topology offerer modify callback. * Added topology offerer configure callback. * Had to rework the unit tests because I changed how SDP topologies are merged. Replaced several unit tests with new negotiation tests. Change-Id: If07fe6d79fbdce33968a9401d41d908385043a06	9 years ago
Corey Farrell	70d2ccb9da	Core: Add support for systemd socket activation. This change adds support for socket activation of certain SOCK_STREAM listeners in Asterisk: * AMI / AMI over TLS * CLI * HTTP / HTTPS Example systemd units are provided. This support extends to any socket which is initialized using ast_tcptls_server_start, so any unknown modules using this function will support socket activation. Asterisk continues to function as normal if socket activation is not enabled or if systemd development headers are not available during build. ASTERISK-27063 #close Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d	9 years ago
George Joseph	3f5bf287a2	Merge "SDP: Add get/set option calls for RTP sched context per type."	9 years ago
Jenkins2	317234bdc6	Merge "res_pjsip: New endpoint option "notify_early_inuse_ringing""	9 years ago
Jenkins2	d81293a5dd	Merge changes from topic 'sdp_api_adjustments' * changes: SDP: Set the remote c= line in RTP instance. SDP: Add t= line in sdp_create_from_state() stream: Ignore declined streams for some topology calls.	9 years ago
Jenkins2	2f684eb6a5	Merge "stream: Add ast_stream_topology_del_stream() and unit test."	9 years ago
Alexei Gradinari	7a46309d3d	res_pjsip: New endpoint option "notify_early_inuse_ringing" This option was added to control whether to notify dialog-info state 'early' or 'confirmed' on Ringing when already INUSE. The value "yes" is useful for some SIP phones (Cisco SPA) to be able to indicate and pick up ringing devices. ASTERISK-26919 #close Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711	9 years ago
George Joseph	1ac0096512	res_ari: Add "module loaded" check to ari stubs The recent change to make the use of LOAD_DECLINE more consistent caused res_ari to unload itself before declining if the ari.conf file wasn't found. The ari stubs though still tried to use the configuration resulting in segfaults. This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests to see if res_ari is actually loaded and causes the stubs to also decline if it isn't. The macro was then added to the mustache template's "load_module" function. ASTERISK-27026 #close Reported-by: Ronald Raikes Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d	9 years ago
Joshua Colp	1f2ab6e72a	Merge "bridge: Add a deferred queue."	9 years ago
Richard Mudgett	e563a1920e	SDP: Add get/set option calls for RTP sched context per type. Change-Id: I82dc75c63c48904e9e5a49e2205dcc06e88487e4	9 years ago
Richard Mudgett	a95584d079	SDP: Set the remote c= line in RTP instance. Change-Id: I23b646392082deab65bedeb19b12dcbcb9216d0c	9 years ago
Richard Mudgett	06265b8c8a	stream: Add ast_stream_topology_del_stream() and unit test. Change-Id: If07e3c716a2e3ff85ae905c17572ea6ec3cdc1f9	9 years ago
Richard Mudgett	4797a8bb81	stream: Ignore declined streams for some topology calls. * Made ast_format_cap_from_stream_topology() not include any formats from declined streams. * Made ast_stream_topology_get_first_stream_by_type() ignore declined streams to return the first active stream of the type. * Updated unit tests to check these changes have the expected effect. Change-Id: Iabbc6a3e8edf263a25fd3056c3c614407c7897df	9 years ago
Joshua Colp	d6386a8f0c	bridge: Add a deferred queue. This change adds a deferred queue to bridging. If a bridge technology determines that a frame can not be written and should be deferred it can indicate back to bridging to do so. Bridging will then requeue any deferred frames upon a new channel joining the bridge. This change has been leveraged for T.38 request negotiate control frames. Without the deferred queue there is a race condition between the bridge receiving the T.38 request negotiate and the second channel joining and being in the bridge. If the channel is not yet in the bridge then the T.38 negotiation fails. A unit test has also been added that confirms that a T.38 request negotiate control frame is deferred when no other channel is in the bridge and that it is requeued when a new channel joins the bridge. ASTERISK-26923 Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415	9 years ago
Kevin Harwell	9e53c30610	res_pjsip_refer/session: Calls dropped during transfer When doing an attended transfer it's possible for the transferer, after receiving an accepted response from Asterisk, to send a BYE to Asterisk, which can then be processed before Asterisk has time to start and/or complete the transfer process. This of course causes the transfer to not complete successfully, thus dropping the call. This patch makes it so any BYEs received from the transferer, after the REFER, that initiate a session end are deferred until the transfer is complete. This allows the channel that would have otherwise been hung up by Asterisk to remain available throughout the transfer process. ASTERISK-27053 #close Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a	9 years ago
Joshua Colp	861984eac0	res_pjsip: Add support for returning only reachable contacts and use it. This introduces the ability for PJSIP code to specify filtering flags when retrieving PJSIP contacts. The first flag for use causes the query code to only retrieve contacts that are not unreachable. This change has been leveraged by both the Dial() process and the PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt calls to contacts which are not unreachable. ASTERISK-26281 Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c	9 years ago
Jenkins2	452e6315bb	Merge "format: Reintroduce smoother flags"	9 years ago
Joshua Colp	1a24543124	Merge "Confbridge: Add "sfu" video mode to bridge profile options."	9 years ago
Jenkins2	bb2f6234da	Merge "Add primitive SFU support to bridge_softmix."	9 years ago
Joshua Colp	97abf6d475	Merge "res_srtp: Add support for libsrtp2"	9 years ago
Jenkins2	48d047ad5a	Merge "res_pjsip: New endpoint option "refer_blind_progress""	9 years ago
Sean Bright	5c27fe2187	format: Reintroduce smoother flags In review 4843 (ASTERISK-24858), we added a hack that forced a smoother creation when sending signed linear so that the byte order was adjusted during transmission. This was needed because smoother flags were lost during the new format work that was done in Asterisk 13. Rather than rolling that same hack into res_rtp_multicast, re-introduce smoother flags so that formats can dictate their own options. Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16	9 years ago
Mark Michelson	39d14834f8	Confbridge: Add "sfu" video mode to bridge profile options. A previous commit added plumbing to bridge_softmix to allow for an SFU experience with Asterisk. This commit adds an option to app_confbridge that allows for a confbridge to actually make use of the SFU video mode. SFU mode is implemented in a "set it and forget it" kind of way. That is, when the bridge is created, if SFU mode is enabled, then the video mode gets set to SFU and cannot be changed. Future improvements may allow for a hybrid experience (e.g. forward multiple video streams, specifically those of the most recent talkers), but for this addition, no such capability is present. Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020	9 years ago
Mark Michelson	2da869408a	Add primitive SFU support to bridge_softmix. This sets up the "plumbing" in bridge_softmix to be able to accommodate Asterisk asking as an SFU (selective forwarding unit) for conferences. The way this works is that whenever a channel enters or leaves a conference, all participants in the bridge get sent a stream topology change request. The topologies consist of the channels' original topology, along with video destination streams corresponding to each participants' source video streams. So for instance, if Alice, Bob, and Carol are in the conference, and each supplies one video stream, then the topologies for each would look like so: Alice: Audio, Source video(Alice), Destination Video(Bob), Destination video (Carol) Bob: Audio, Source video(Bob) Destination Video(Alice), Destination video (Carol) Carol: Audio, Source video(Carol) Destination Video(Alice), Destination video (Bob) This way, video that arrives from a source video stream can then be copied out to the destination video streams on the other participants' channels. Once the bridge gets told that a topology on a channel has changed, the bridge constructs a map in order to get the video frames routed to the proper destination streams. This is done using the bridge channel's stream_map. This change is bare-bones with regards to SFU support. Some key features are missing at this point: * Stream limits. This commit makes no effort to limit the number of streams on a specific channel. This means that if there were 50 video callers in a conference, bridge_softmix will happily send out topology change requests to every channel in the bridge, requesting 50+ streams. * Configuration. The plumbing has been added to bridge_softmix, but there has been nothing added as of yet to app_confbridge to enable SFU video mode. * Testing. Some functions included here have unit tests. However, the functionality as a whole has only been verified by hand-tracing the code. * Selectivenss. For a "selective" forwarding unit, this does not currently have any means of being selective. * Features. Presumably, someone might wish to only receive video from specific sources. There are no external-facing functions at the moment that allow for users to select who they receive video from. * Efficiency. The current scheme treats all video streams as being unidirectional. We could be re-using a source video stream as a desetnation, too. But to simplify things on this first round, I did it this way. Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d	9 years ago
Sean Bright	1f136fe885	res_srtp: Add support for libsrtp2 ASTERISK-25294 #close Reported by: Tzafrir Cohen ASTERISK-26976 #close Reported by: Alex Change-Id: I789b1c3d1ed31365bbd9339fa58ef36f48833c40	9 years ago
Jenkins2	56b6a71548	Merge "asterisk: Audit locking of channel when manipulating flags."	9 years ago
George Joseph	08edd54c1b	unittests: Add a unit test that causes a SEGV and... ...that can only be run by explicitly calling it with 'test execute category /DO_NOT_RUN/ name RAISE_SEGV' This allows us to more easily test CI and debugging tools that should do certain things when asterisk coredumps. To allow this a new member was added to the ast_test_info structure named 'explicit_only'. If set by a test, the test will be skipped during a 'test execute all' or 'test execute category ...'. Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed	9 years ago

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3745 Commits (1bec535df2e8a7968a810cbef594fa17f7b642bc)