517 Commits (c98c1b3f744fba671ff23a212bb970cc2d38b203)

Author	SHA1	Message	Date
Corey Farrell	b5914d90ac	Fix GCC 8 build issues. ... This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	8 years ago
Richard Mudgett	661fec4b59	core: Remove unused/incomplete SDP modules. ... Change-Id: Icc28fbdc46f58e54a21554e6fe8b078f841b1f86	8 years ago
George Joseph	f79a372941	streams: Add string metadata capability ... Replaces the never used opaque data array. Updated stream tests to include get/set metadata and stream clone with metadata. Added stream metadata dump to "core show channel" Change-Id: Id7473aa4b374d7ab53046c20e321037ba9a56863	8 years ago
Ben Ford	879743ab8f	test_data_buffer.c: Add unit tests for data buffer API. ... Added unit tests for the data buffer API. These tests include creating a data buffer, putting payloads into the buffer, resizing the buffer, and the nominal case for data buffer usage, which consists of adding the max number of payloads to the buffer, checking to see if the correct payloads are present, then adding more payloads and checking again to see if the previous payloads were replaced or not. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements Change-Id: Id5b599aa15a5e61d0ec080f97cd0c57bd07e6f8f	8 years ago
Corey Farrell	de77cf8698	core: Remove dead symbols from asterisk.exports.in. ... * dahdi_chan_name * dahdi_chan_name_len * dahdi_chan_mode * __manager_event * dialed_interface_info Added comment about __progname and environ being needed for FreeBSD to prevent accidental removal in the future. Change-Id: I3ae026bc541cd9cb572be2ffa95fc359547642b5	8 years ago
Jenkins2	a231e1d155	Merge "Remove redundant module checks and references."	8 years ago
Corey Farrell	527cf5a570	Remove redundant module checks and references. ... This removes references that are no longer needed due to automatic references created by module dependencies. In addition this removes most calls to ast_module_check as they were checking modules which are listed as dependencies. Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e	8 years ago
Corey Farrell	679fa5fb34	Add missing OPTIONAL_API and ARI dependences. ... I've audited all modules that include any header which includes asterisk/optional_api.h. All modules which use OPTIONAL_API now declare those dependencies in AST_MODULE_INFO using requires or optional_modules as appropriate. In addition ARI dependency declarations have been reworked. Instead of declaring additional required modules in res/ari/resource_*.c we now add them to an optional array "requiresModules" in api-docs for each module. This allows the AST_MODULE_INFO dependencies to include those missing modules. Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606	8 years ago
Corey Farrell	9cfdb81e91	loader: Add dependency fields to module structures. ... * Declare 'requires' and 'enhances' text fields on module info structure. * Rename 'nonoptreq' to 'optional_modules'. * Update doxygen comments. Still need to investigate dependencies among modules I cannot compile. Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf	8 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Corey Farrell	1b80ffa495	Fix Common Typo's. ... Fix instances of: * Retreive * Recieve * other then * different then * Repeated words ("the the", "an an", "and and", etc). * othterwise, teh ASTERISK-24198 #close Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31	8 years ago
Corey Farrell	bf2d35931d	aco: Minimize use of regex. ... Remove nearly all use of regex from ACO users. Still remaining: * app_confbridge has a legitamate use of option name regex. * ast_sorcery_object_fields_register is implemented with regex, all callers use simple prefix based regex. I haven't decided the best way to fix this in both 13/15 and master. Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b	8 years ago
George Joseph	93dba6f396	Merge "CLI: Remove calls to ast_cli_generator."	8 years ago
Joshua Colp	509e713333	Merge "Loader: Remove unneeded load_pri declarations."	8 years ago
Corey Farrell	a02cbc2ef3	CLI: Remove calls to ast_cli_generator. ... The ability to add to localized storage cannot be supported by ast_cli_generator. The only calls to ast_cli_generator should be by functions that need to proxy the CLI generator, for example 'cli check permissions' or 'core show help'. * ast_cli_generatornummatches now retrieves the vector of matches and reports the number of elements (not including 'best' match). * test_substitution retrieves and iterates the vector. Change-Id: I8cd6b93905363cf7a33a2d2b0e2a8f8446d9f248	8 years ago
Joshua Colp	86e894c074	Merge "menuselect: Remove ineffective weak attribute detection."	8 years ago
Corey Farrell	58fa3885cc	Loader: Remove unneeded load_pri declarations. ... Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT. Change-Id: I0123258eafce324249433a69df15a85cc16e509f	8 years ago
Corey Farrell	75cb403775	tests: Fix warnings found on Mac. ... test_pbx used raise without explicitly including signal.h. On Mac for some reason nothing else includes it. test_logger checked if an unsigned int was negative. Switch the variable to 'int' so that error check can be effective. Change-Id: Ie1db5dd1818ac25cc2ae41b644f848b5865b1362	8 years ago
Corey Farrell	b9f4bb5988	menuselect: Remove ineffective weak attribute detection. ... menuselect detects compiler support for multiple styles of weak functions. This is a remnant from 2013 when OPTIONAL_API required weak functions. It is no longer correct for menuselect to switch dependencies from optional to required based on lack of weak function support. Note an issue remains - dependencies should switch from optional to required based on OPTIONAL_API being enabled or disabled. I don't think this is possible. menuselect needs to know at startup if OPTIONAL_API is enabled or disabled, so the only way to fix this is to remove OPTIONAL_API from menuselect and create a configure option. I've left the code that switches in place but it's preprocessed out. Additionally removed: - WEAKREF variable from Asterisk makeopts.in. - Related disabled code from test_utils. - Pointless AC_REVISION call from menuselect/configure.ac. Change-Id: Ifa702e5f98eb45f338b2f131a93354632a8fb389	8 years ago
Corey Farrell	24b9751aaa	test_sorcery_memory_cache_thrash: Handle error from vector append. ... Cleanup resources when we fail to append the vector and report test failure. Change-Id: I6eb41586fd11dee8c0dfe35e91cb465a4cab7298	8 years ago
Corey Farrell	606ae3484a	Add missing menuselect dependencies. ... This adds menuselect dependencies for modules that use symbols of other modules. ASTERISK-27390 Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385	8 years ago
Corey Farrell	a9e9608982	test_config: Fix failure and segfault when config_hook is run twice. ... On second run the config_hook test was unexpectedly failing to load test_config.conf because it was still unmodified since the last load. This is fixed by not passing CONFIG_FLAG_FILEUNCHANGED for the initial loads, only using it when we are tested that a reload of unmodified files do not initiate the hook. ASTERISK-25960 Change-Id: Ifd679509a23ed163e5cc647490bf7df4ae3cd856	8 years ago
Corey Farrell	3bd00c4a7e	vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. ... Use temporary variable to prevent multiple evaluations of elem argument. This resolves a memory leak in res_pjproject startup. ASTERISK-27317 #close Change-Id: Ib960d7f5576f9e1a3c478ecb48995582a574e06d	8 years ago
Richard Mudgett	9c70c88369	channel: Fix topology API locking. ... * ast_channel_request_stream_topology_change() must not be called with any channel locks held. * ast_channel_stream_topology_changed() must be called with only the passed channel lock held. ASTERISK-27212 Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691	8 years ago
Kevin Harwell	e4e2e53c8a	manager: hook event is not being raised ... When the iostream code went in it introduced a conditional that made it so the hook event was not being raised even if a hook is present. This patch adds a check to see if a hook is present in astman_append. If so then call into the send_string function, which in turn raises the even for specified hook. Also updated the ami hooks unit test, so the test could be automated. ASTERISK-27200 #close Change-Id: Iff37f02f9708195d8f23e68f959d6eab720e1e36	8 years ago
Corey Farrell	7f8f3ca4dd	Correct some leaks in unit tests. ... * chan_sip: channel in test_sip_rtpqos_1. * test_config: config hook, config info and global config holder. * test_core_format: format in format_attribute_set_without_interface. * test_stream: unneeded frame duplication. * test_taskprocessor: task_data. Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31	8 years ago
Joshua Colp	3dff2711a3	Merge "SDP: Create declined m= SDP lines using remote SDP if applicable."	8 years ago
Joshua Colp	8412cc1e07	Merge "SDP: Rework SDP offer/answer model and update capabilities merges."	8 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	8 years ago
Richard Mudgett	d2fbbdd692	SDP: Create declined m= SDP lines using remote SDP if applicable. ... * Update SDP unit tests to test negotiating with declined streams. Generation of declined m= lines created and responded tested. Change-Id: I5cb99f5010994ab0c7d9cf2d395eca23fab37b98	8 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	8 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.	8 years ago
Jenkins2	2f684eb6a5	Merge "stream: Add ast_stream_topology_del_stream() and unit test."	8 years ago
Richard Mudgett	06265b8c8a	stream: Add ast_stream_topology_del_stream() and unit test. ... Change-Id: If07e3c716a2e3ff85ae905c17572ea6ec3cdc1f9	8 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	8 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	8 years ago
George Joseph	80206cdc65	test_json: Fix test names with reserved words ... Some of the test names were actually reserved words (true, false, int, null, string, bool). When the jenkins test results analyzer does its thing it tries to create a map using the test names as keys and fails because they're reserved words. Added "type_" to those test names. Change-Id: I90d809f46969c78a1c605b736ff0635196a2cf1b	8 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
Kevin Harwell	51375686f7	core/conversions: Added string to unsigned integer and long conversions ... Added functions that convert a string to an unsigned integer or unsigned long. A couple of unit test were also created to test the routines. The reasons for adding these conversion utilities (and hopefully eventually more) are as follows: * Conversion routines are functionally contained with consistent and better error checking * The function names offer a better description of what is happening * It encourages code reuse for easier bug fixing at a single source * It's simpler to use * It's unit testable For instance, currently in a lot of places when converting to an integer or similar the "sscanf" function is used. When using "sscanf" it may not be immediately clear what's happening as it lacks semantic naming. Limited error checking is usually done as well. For example, most of the time a check is done to make sure the value converted, but does not check for overflows or negative valued conversions when converting unsigned numbers. Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the built in error handling that "strtoul" has. For instance "strtoul" contains overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly more complex in its use. And maybe a bit controversial, but it may be ("big if") potentially slower than "strtoul" in some cases. Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb	9 years ago
George Joseph	ce4d8dac91	Merge changes from topic 'sdp_api_adjustments' ... * changes: SDP: Make process possible multiple fmtp attributes per rtpmap. SDP: Explicitly stop a RTP instance before destoying it. SDP: Rework merge_capabilities(). SDP: Update ast_get_topology_from_sdp() to keep RTP map.	9 years ago
Richard Mudgett	b8659be9b0	SDP: Make process possible multiple fmtp attributes per rtpmap. ... Change-Id: Ie7511008d82b59590e0eb520a21b5e1da4bd7349	9 years ago
Richard Mudgett	367042bd3e	SDP: Explicitly stop a RTP instance before destoying it. ... * Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream() handle generating disabled/declined streams. * Added /main/sdp/sdp_merge_asymmetric unit test. It currently does not check the offerer side negotiated SDP because that isn't the purpose of this patch and there is much to be done to handle declined/dummy streams. * Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and /main/sdp/sdp_merge_crisscross unit tests. Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31	9 years ago
Richard Mudgett	ae7689f093	SDP: Update ast_get_topology_from_sdp() to keep RTP map. ... * Add failure exits to ast_get_topology_from_sdp(). Change-Id: I4cc85c1ede8d712766ed20f544dbcef04c8c1049	9 years ago
George Joseph	201346fb7d	logger: Added logger_queue_limit to the configuration options. ... All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	9 years ago
Kevin Harwell	7b0e3b92fd	bridge_simple: Added support for streams ... This patch is the first cut at adding stream support to the bridging framework. Changes were made to the framework that allows mapping of stream topologies to a bridge's supported media types. The first channel to enter a bridge initially defines the media types for a bridge (i.e. a one to one mapping is created between the bridge and the first channel). Subsequently added channels merge their media types into the bridge's adding to it when necessary. This allows channels with different sized topologies to map correctly to each other according to media type. The bridge drops any frame that does not have a matching index into a given write stream. For now though, bridge_simple will align its two channels according to size or first to join. Once both channels join the bridge the one with the most streams will indicate to the other channel to update its streams to be the same as that of the other. If both channels have the same number of streams then the first channel to join is chosen as the stream base. A topology change source was also added to a channel when a stream toplogy change request is made. This allows subsystems to know whether or not they initiated a change request. Thus avoiding potential recursive situations. ASTERISK-26966 #close Change-Id: I1eb5987921dd80c3cdcf52accc136393ca2d4163	9 years ago
Jenkins2	94b97e0835	Merge "SDP API: Add SSRC-level attributes"	9 years ago
Jenkins2	528e238447	Merge "channel: Add ability to request an outgoing channel with stream topology."	9 years ago
Jenkins2	49b2d1bde5	Merge "vector: defaults and indexes"	9 years ago
Mark Michelson	d6535c0080	SDP API: Add SSRC-level attributes ... RFC 5576 defines how SSRC-level attributes may be added to SDP media descriptions. In general, this is useful for grouping related SSRCes, indicating SSRC-level format attributes, and resolving collisions in RTP SSRC values. These attributes are used widely by browsers during WebRTC communications, including attributes defined by documents outside of RFC 5576. This commit introduces the addition of SSRC-level attributes into SDPs generated by Asterisk. Since Asterisk does not tend to use multiple SSRCs on a media stream, the initial support is minimal. Asterisk includes an SSRC-level CNAME attribute if configured to do so. This at least gives browsers (and possibly others) the ability to resolve SSRC collisions at offer-answer time. In order to facilitate this, the RTP engine API has been enhanced to be able to retrieve the SSRC and CNAME on a given RTP instance. res_rtp_asterisk currently does not provide meaningful CNAME values in its RTCP SDES items, and therefore it currently will always return an empty string as the CNAME value. A task in the near future will result in res_rtp_asterisk generating more meaningful CNAMEs. Change-Id: I29e7f23e7db77524f82a3b6e8531b1195ff57789	9 years ago
Joshua Colp	2b22c3c84b	channel: Add ability to request an outgoing channel with stream topology. ... This change extends the ast_request functionality by adding another function and callback to create an outgoing channel with a requested stream topology. Fallback is provided by either converting the requested stream topology into a format capabilities structure if the channel driver does not support streams or by converting the requested format capabilities into a stream topology if the channel driver does support streams. The Dial application has also been updated to request an outgoing channel with the stream topology of the calling channel. ASTERISK-26959 Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6	9 years ago