3631 Commits (6b7697ed486fc3a8e5e7a72344437e66bd4ae507)

Author	SHA1	Message	Date
Richard Begg	6b7697ed48	res_pjsip_session: Enable RFC3578 overlap dialing support. ... Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	8 years ago
Sean Bright	fc71c18a9b	thread safety: Don't use getprotobyname() ... POSIX does not require getprotobyname() to be thread safe and some implementations use static memory which causes issues when multiple threads are used. Further, our usage of it today is just to ultimately get IPPROTO_TCP for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48	8 years ago
Joshua Colp	732367e806	Merge "res_pjsip: Symmetric transports"	8 years ago
Joshua Colp	76e64f5589	Merge "RFC sdp: Initial SDP creation"	8 years ago
George Joseph	5013d8f5d3	res_pjsip: Symmetric transports ... A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	8 years ago
Joshua Colp	84f0871cba	Merge "Add rtcp-mux support"	8 years ago
Mark Michelson	10fa49e327	Add rtcp-mux support ... This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5	8 years ago
George Joseph	8470c2bdea	RFC sdp: Initial SDP creation ... * Added additional fields to ast_sdp_options. * Re-organized ast_sdp. * Updated field names to correspond to RFC4566 terminology. * Created allocs/frees for SDP children. * Created getters/setters for SDP children where appropriate. * Added ast_sdp_create_from_state. * Refactored res_sdp_translator_pjmedia for changes. Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48	8 years ago
Joshua Colp	3ed05badb9	core: Add stream topology changing primitives with tests. ... This change adds a few things to facilitate stream topology changing: 1. Control frame types have been added for use by the channel driver to notify the application that the channel wants to change the stream topology or that a stream topology change has been accepted. They are also used by the indicate interface to the channel that the application uses to indicate it wants to do the same. 2. Legacy behavior has been adopted in ast_read() such that if a channel requests a stream topology change it is denied automatically and the current stream topology is preserved if the application is not capable of handling streams. Tests have also been written which confirm the multistream and non-multistream behavior. ASTERISK-26839 Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9	8 years ago
Joshua Colp	fb11f038a3	Merge "stream: Unit tests for stream read and tweaks framework"	8 years ago
George Joseph	0560c32375	stream: Unit tests for stream read and tweaks framework ... * Removed the AST_CHAN_TP_MULTISTREAM tech property. We now rely on read_stream being set to indicate a multi stream channel. * Added ast_channel_is_multistream convenience function. * Fixed issue where stream and default_stream weren't being set on a frame retrieved from the queue. * Now testing for NULL being returned from the driver's read or read_stream callback. * Fixed issue where the dropnondefault code was crashing on a NULL f. * Now enforcing that if either read_stream or write_stream are set when ast_channel_tech_set is called that BOTH are set. * Added the unit tests. ASTERISK-26816 Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2	8 years ago
Mark Michelson	9c55a71798	SDP: Add initial SDP state machine. ... This introduces and documents the various states in the state machine. This also introduces API functions that induce state changes, and places TODO comments telling what needs to be done in addition to what is already there. Those TODOs will be replaced with real code in upcoming changes. Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed	8 years ago
Joshua Colp	ff2b4308d1	bridge_native_rtp: Handle case where channel joins already suspended. ... The bridge_native_rtp module did not properly handle the case where a smart bridge operation occurs while a channel is suspended. In this scenario the module would incorrectly set up local or remote RTP bridging despite the media having to flow through Asterisk. The remote endpoint would see two media streams and experience wonky audio. The module has been changed so that it ensures both channels are not suspended when performing the native RTP bridging and this requirement has been documented in the bridge technology. ASTERISK-26781 Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c	8 years ago
George Joseph	df22d297a6	Merge "channel: Add ast_read_stream function for reading frames from all streams."	8 years ago
Joshua Colp	6ac33bfe3e	Merge "Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix."	8 years ago
zuul	461577b23b	Merge "channel: Add support for writing to a specific stream."	8 years ago
Joshua Colp	c07c6714f2	channel: Add ast_read_stream function for reading frames from all streams. ... This change introduces an ast_read_stream function and callback in the channel technology which allows reading frames from all streams and not just the default streams. The stream number has also been added to frames. This is to allow the case where frames are queued onto the channel instead of being read directly from the driver. This change does impose a restriction on reading though: a chain of frames can only contain frames from the same stream. ASTERISK-26816 Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce	8 years ago
Joshua Colp	6cc890b880	channel: Add support for writing to a specific stream. ... This change adds an ast_write_stream function which allows writing a frame to a specific media stream. It also moves ast_write() to using this underneath by writing media frames provided to it to the default streams of the channel. Existing functionality (such as audiohooks, framehooks, etc) are limited to being applied to the default stream only. Unit tests have also been added which test the behavior of both non-multistream and multistream channels to confirm that the write() and write_stream() callbacks are invoked appropriately. ASTERISK-26793 Change-Id: I4df20d1b65bd4d787fce0b4b478e19d2dfea245c	8 years ago
frahaase	094c26aa68	Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix. ... Adds binaural synthesis to bridge_softmix (via convolution using libfftw3). Binaural synthesis is conducted at 48kHz. For a conference, only one spatial representation is rendered. The default rendering is applied for mono-capable channels. ASTERISK-26292 Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf	8 years ago
zuul	6c22d4b320	Merge "core: Show streams in "core show channel"."	8 years ago
Joshua Colp	10302fa63f	Merge "Add initial SDP state code."	8 years ago
Joshua Colp	f58aefba5b	core: Show streams in "core show channel". ... The "core show channel" CLI command will now output the streams present on the channel with their details. ASTERISK-26811 Change-Id: I9c95b57aa09415005f0677a1949a0feb07e4987a	8 years ago
Joshua Colp	8f248f7a1c	Merge "realtime: Centralize some common realtime backend code"	8 years ago
Mark Michelson	a738772edd	Add initial SDP state code. ... This establishes the basic allocation/destruction of an SDP state object, plus some of the simpler getter methods involved. Subsequent tasks will deal with adding a state machine, creating SDPs from capabilities and options, and merging SDPs into a joint SDP. Change-Id: Ie3757ce186f04b65e9d1883f5aace53f24e53709	8 years ago
Joshua Colp	16b0bb39c1	Merge changes from topic 'sdp_state_beginnings' ... * changes: Add SDP translator and PJMEDIA implementation. Add initial SDP options.	8 years ago
Sean Bright	6e6c96d713	realtime: Centralize some common realtime backend code ... All of the realtime backends create artificial ast_categorys to pass back into the core as query results. These categories have no filename or line number information associated with them and the backends differ slightly on how they create them. So create a couple helper macros to help make things more consistent. Also updated the call sites to remove redundant error messages about memory allocation failure. Note that res_config_ldap sets the category filename to the 'table name' but that is not read by anything in the core, so I've dropped it. Change-Id: I3a1fd91e0c807dea1ce3b643b0a6fe5be9002897	8 years ago
Joshua Colp	fe88f2e5ca	Merge "Binaural synthesis (confbridge): Adds utils/conf_bridge_binaural_hrir_importer"	8 years ago
Mark Michelson	5a130b2e17	Add SDP translator and PJMEDIA implementation. ... This creates the following: * Asterisk's internal representation of an SDP * An API for translating SDPs from one format to another * An implementation of a translator for PJMEDIA Change-Id: Ie2ecd3cbebe76756577be9b133e84d2ee356d46b	8 years ago
Mark Michelson	8af6342555	Add initial SDP options. ... This is step one of adding an SDP API: defining some configurable settings for SDPs. This is based on options that are currently supported in Asterisk. Change-Id: I1ede91aafed403b12a9ccdfb91a88389baa7e5d7	8 years ago
zuul	ab34e46b3a	Merge "stream: Rename creates/destroys to allocs/frees"	8 years ago
Joshua Colp	b1edbc4c83	Merge "res_pjsip_pubsub: Correctly implement persisted subscriptions"	8 years ago
George Joseph	f8f513d363	stream: Rename creates/destroys to allocs/frees ... To be consistent with sdp implementation. Change-Id: I714e300939b4188f58ca66ce9d1e84b287009500	8 years ago
George Joseph	ca7fa7bbd2	Merge "stream: Add stream topology to channel"	8 years ago
George Joseph	4bdf5d329f	res_pjsip_pubsub: Correctly implement persisted subscriptions ... This patch fixes 2 original issues and more that those 2 exposed. * When we send a NOTIFY, and the client either doesn't respond or responds with a non OK, pjproject only calls our pubsub_on_evsub_state callback, no others. Since pubsub_on_evsub_state (which does the sub_tree cleanup) does not expect to be called back without the other callbacks being called first, it just returns leaving the sub_tree orphaned. Now pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE which is what pjproject will set to tell us that it was the transaction that timed out or failed and not the subscription itself timing our or being terminated by the client. If is TSX_STATE, pubsub_on_evsub_state now does the proper cleanup regardless of the state of the subscription. * When a client renews a subscription, we don't update the persisted subscription with the new expires timestamp. This causes subscription_persistence_recreate to prune the subscription if/when asterisk restarts. Now, pubsub_on_rx_refresh calls subscription_persistence_update to apply the new expires timestamp. This exposed other issues however... * When creating a dialog from rdata (which sub_persistence_recreate does from the packet buffer) there must NOT be a tag on the To header (which there will be when a client refreshes a subscription). If there is one, pjsip_dlg_create_uas will fail. To address this, subscription_persistence_update now accepts a flag that indicates that the original packet buffer must not be updated. New subscribes don't set the flag and renews do. This makes sure that when the rdata is recreated on asterisk startup, it's done from the original subscribe packet which won't have the tag on To. * When creating a dialog from rdata, we were setting the dialog's remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq. When the client tried to resubscribe after a restart with the correct cseq, we'd reject the request with an Invalid CSeq error. * The acts of creating a dialog and evsub by themselves when recreating a subscription does NOT restart pjproject's subscription timer. The result was that even if we did correctly recreate the subscription, we never removed it if the client happened to go away or send a non-OK response to a NOTIFY. However, there is no pjproject function exposed to just set the timer on an evsub that wasn't created by an incoming subscribe request. To address this, we create our own timer using ast_sip_schedule_task. This timer is used only for re-establishing subscriptions after a restart. An earlier approach was to add support for setting pjproject's timer (via a pjproject patch) and while that patch is still included here, we don't use that call at the moment. While addressing these issues, additional debugging was added and some existing messages made more useful. A few formatting changes were also made to 'pjsip show scheduled tasks' to make displaying the subscription timers a little more friendly. ASTERISK-26696 ASTERISK-26756 Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e	8 years ago
Dennis Guse	b58de2fab7	Binaural synthesis (confbridge): Adds utils/conf_bridge_binaural_hrir_importer ... Adds the import tool for converting a HRIR database to hrirs.h ASTERISK-26292 Change-Id: I51eb31b54c23ffd9b544bdc6a09d20c112c8a547	8 years ago
George Joseph	bf2f091bbb	stream: Add stream topology to channel ... Adds topology set and get to channel. ASTERISK-26790 Change-Id: Ic379ea82a9486fc79dbd8c4d95c29fa3b46424f4	8 years ago
zuul	09fcfb26fa	Merge "stream: Add media stream topology definition and API"	8 years ago
George Joseph	8b72ec312b	stream: Add media stream topology definition and API ... This change adds the media stream topology definition and API for accessing and using it. Some refactoring of the stream was also done. ASTERISK-26786 Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568	8 years ago
zuul	f9f74f4b75	Merge "manager: Restore Originate failure behavior from Asterisk 11"	8 years ago
Sean Bright	0910773077	manager: Restore Originate failure behavior from Asterisk 11 ... In Asterisk 11, if the 'Originate' AMI command failed to connect the provided Channel while in extension mode, a 'failed' extension would be looked up and run. This was, I believe, unintentionally removed in 51b6c49. This patch restores that behavior. This also adds an enum for the various 'synchronous' modes in an attempt to make them meaningful. ASTERISK-26115 #close Reported by: Nasir Iqbal Change-Id: I8afbd06725e99610e02adb529137d4800c05345d	8 years ago
Joshua Colp	bab4885f1e	stream: Add media stream definition and API with unit tests. ... This change adds the media stream definition and API for accessing and using it. Unit tests have also been written which exercise aspects of the API. ASTERISK-26773 Change-Id: I3dbe54065b55aaa51f467e1a3bafd67fb48cac87	8 years ago
Richard Mudgett	97c308471d	res_agi: Prevent an AGI from eating frames it should not. (Re-do) ... A dialplan intercept routine is equivalent to an interrupt routine. As such, the routine must be done quickly and you do not have access to the media stream. These restrictions are necessary because the media stream is the responsibility of some other code and interfering with or delaying that processing is bad. A possible future dialplan processing architecture change may allow the interception routine to run in a different thread from the main thread handling the media and remove the execution time restriction. * Made res_agi.c:run_agi() running an AGI in an interception routine run in DeadAGI mode. No touchy channel frames. ASTERISK-25951 ASTERISK-26343 ASTERISK-26716 Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43	8 years ago
Richard Mudgett	72e3fc5845	Frame deferral: Revert API refactoring. ... There are several issues with deferring frames that are caused by the refactoring. 1) The code deferring frames mishandles adding a deferred frame to the deferred queue. As a result the deferred queue can only be one frame long. 2) Deferrable frames can come directly from the channel driver as well as the read queue. These frames need to be added to the deferred queue. 3) Whoever is deferring frames is really only doing the __ast_read() to collect deferred frames and doesn't care about the returned frames except to detect a hangup event. When frame deferral is completed we must make the normal frame processing see the hangup as a frame anyway. As such, there is no need to have varying hangup frame deferral methods. We also need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. That fake hangup is to cause the PBX thread to break out of loops to go execute a new dialplan location. 4) To properly deal with deferrable frames from the channel driver as pointed out by (2) above, means that it is possible to process a dialplan interception routine while frames are deferred because of the AST_CONTROL_READ_ACTION control frame. Deferring frames is not implemented as a re-entrant operation so you could have the unsupported case of two sections of code thinking they have control of the media stream. A worse problem is because of the bad implementation of the AMI PlayDTMF action. It can cause two threads to be deferring frames on the same channel at the same time. (ASTERISK_25940) * Rather than fix all these problems simply revert the API refactoring as there is going to be only autoservice and safe_sleep deferring frames anyway. ASTERISK-26343 ASTERISK-26716 #close Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496	8 years ago
George Joseph	6f645a6d4e	Merge "media: Add experimental support for RTCP feedback."	8 years ago
zuul	10631bb209	Merge "PJPROJECT logging: Fix detection of max supported log level."	8 years ago
George Joseph	0ad6d2b3cf	Merge "ari: Implement 'debug all' and request/response logging"	8 years ago
Richard Mudgett	36bdd7c1a0	Add notes about embedded ast_frame structs holding a format ref. ... mod_format.h: Note ast_filestream.fr holds a format ref. translate.h: Note ast_trans_pvt.f holds a format ref. Change-Id: I86bda354d725207b41e08920355d7c31b2d7f749	8 years ago
Richard Mudgett	6f3e8c8e01	PJPROJECT logging: Fix detection of max supported log level. ... The mechanism used for detecting the maximum log level compiled into the linked pjproject did not work. The API call simply stores the requested level into an integer and does no range checking. Asterisk was assuming that there was range checking and limited the new value to the allowable range. To get the actual maximum log level compiled into the linked pjproject we need to get and save off the initial set log level from pjproject. This is the maximum log level supported. * Get and save off the initial log level setting before altering it to the desired level on startup. This has to be done by a macro rather than calling a core function to avoid incorrectly linking pjproject. * Split the initial log level warning messages to warn if the linked pjproject cannot support the requested startup level and if it is too low to get the pjproject buildopts for "pjproject show buildopts". * Adjust the CLI "pjproject set log level" to check the saved max log level and to generate normal output messages instead of a warning message. ASTERISK-26743 #close Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4	8 years ago
George Joseph	6691606723	ari: Implement 'debug all' and request/response logging ... The 'ari set debug' command has been enhanced to accept 'all' as an application name. This allows dumping of all apps even if an app hasn't registered yet. To accomplish this, a new global_debug global variable was added to res/stasis/app.c and new APIs were added to set and query the value. 'ari set debug' now displays requests and responses as well as events. This required refactoring the existing debug code. * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. * In order to print the body of incoming requests even if a request failed, the consumption of the body was moved from the ari stubs to ast_ari_callback in res_ari.c and the moustache templates were then regenerated. The body is now passed to ast_ari_invoke and then on to the handlers. This results in code savings since that template was inserted multiple times into all the stubs. An additional change was made to the ao2_str_container implementation to add partial key searching and a sort function. The existing cli code assumed it was already there when it wasn't so the tab completion was never working. Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf (cherry picked from commit 1d890874f3)	8 years ago
Lorenzo Miniero	1061539b75	media: Add experimental support for RTCP feedback. ... This change adds experimental support for providing RTCP feedback information to codec modules so they can dynamically change themselves based on conditions. ASTERISK-26584 Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857	8 years ago