29 Commits (a02522c140f6d715d71ad61d08c40a94de2238eb)

Author	SHA1	Message	Date
Joshua Colp	bea52b3706	pjsip: Rewrite OPTIONS support with new eyes. ... The OPTIONS support in PJSIP has organically grown, like many things in Asterisk. It has been tweaked, changed, and adapted based on situations run into. Unfortunately this has taken its toll. Configuration file based objects have poor performance and even dynamic ones aren't that great. This change scraps the existing code and starts fresh with new eyes. It leverages all of the APIs made available such as sorcery observers and serializers to provide a better implementation. 1. The state of contacts, AORs, and endpoints relevant to the qualify process is maintained. This state can be updated by external forces (such as a device registering/unregistering) and also the reload process. This state also includes the association between endpoints and AORs. 2. AORs are scheduled and not contacts. This reduces the amount of work spent juggling scheduled items. 3. Manipulation of which AORs are being qualified and the endpoint states all occur within a serializer to reduce the conflict that can occur with multiple threads attempting to modify things. 4. Operations regarding an AOR use a serializer specific to that AOR. 5. AORs and endpoint state act as state compositors. They take input from lower level objects (contacts feed AORs, AORs feed endpoint state) and determine if a sufficient enough change has occurred to be fed further up the chain. 6. Realtime is supported by using observers to know when a contact has been registered. If state does not exist for the associated AOR then it is retrieved and becomes active as appropriate. The end result of all of this is best shown with a configuration file of 3000 endpoints each with an AOR that has a static contact. In the old code it would take over a minute to load and use all 8 of my cores. This new code takes 2-3 seconds and barely touches the CPU even while dealing with all of the OPTIONS requests. ASTERISK-26806 Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082	8 years ago
George Joseph	5947fd148b	AST-2018-005: res_pjsip_transport_management: Move to core ... Since res_pjsip_transport_management provides several attack mitigation features, its functionality moved to res_pjsip and this module has been removed. This way the features will always be available if res_pjsip is loaded. ASTERISK-27618 Reported By: Sandro Gauci Change-Id: I21a2d33d9dda001452ea040d350d7a075f9acf0d	9 years ago
George Joseph	7debdd285c	res_pjsip_pubsub: Prune subs with reliable transports at startup ... In an earlier release, inbound registrations on a reliable transport were pruned on Asterisk restart since the TCP connection would have been torn down and become unusable when Asterisk stopped. This same process is now also applied to inbound subscriptions. Also fixed issues in res_pjsip_registrar where it wasn't handling the monitoring correctly when multiple registrations came in over the same transport. To accomplish this, the pjsip_transport_event feature needed to be refactored to allow multiple monitors (multiple subcriptions or registrations from the same endpoint) to exist on the same transport. Since this changed the API, any external modules that may have used the transport monitor feature (highly unlikey) will need to be changed. ASTERISK-27612 Reported by: Ross Beer Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36	9 years ago
Corey Farrell	82592c3673	res_pjsip: Fix issues that prevented shutdown of modules. ... res_pjsip and res_pjsip_session had circular references, preventing both modules from shutting down. * Move session supplement registration to res_pjsip. * Use create internal functions for use by pjsip_message_filter.c. ASTERISK-27306 Change-Id: Ifbd5c19ec848010111afeab2436f9699da06ba6b	9 years ago
George Joseph	63900374fa	res_pjsip: Filter out non SIP(S) requests ... Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460	9 years ago
Richard Mudgett	ca261d4b70	res_pjsip: PJSIP Transport state monitor refactor. ... The fix for the issue is broken up into three parts. This is part one which refactors the transport state monitor code to allow more modules to be able to monitor transports. * Pull the management of PJPROJECT's transport state callback code from res_pjsip_transport_management.c into res_pjsip. Now other modules can dynamically add and remove themselves from transport monitoring without worrying about breaking PJPROJECT's callback chain. * Add the ability for other modules to get a callback whenever a specific transport is shutdown. ASTERISK-27147 Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912	9 years ago
Mark Michelson	cbc23c31cf	Revert "Update qualifies when AOR configuration changes." ... This reverts commit 6492e91392. The change in question was intended to prevent the need to reload in order to update qualifies on contacts when an AOR changes. However, this ended up causing a deadlock instead. Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e	10 years ago
Mark Michelson	6492e91392	Update qualifies when AOR configuration changes. ... Prior to this change, qualifies would only update in the following cases: * A reload of res_pjsip.so was issued. * A dynamic contact was re-registered after its AOR's qualify_frequency had been changed This does not work well if you are using realtime for your AORs. You can update your database to have a new qualify_frequency, but the permanent contacts on that AOR will not have their qualifies updated. And the dynamic contacts on that AOR will not have their qualifies updated until the next registration, which could be a long time. This change seeks to fix this problem by making it so that whenever AOR configuration is applied, the contacts pertaining to that AOR have their qualifies updated. Additions from this patch: * AOR sorcery objects now have an apply handler that calls into a newly added function in the OPTIONS code. This causes all contacts associated with that AOR to re-schedule qualifies. * When it is time to qualify a contact, the OPTIONS code checks to see if the AOR can still be retrieved. If not, then qualification is canceled on the contact. Alterations from this patch: * The registrar code no longer updates contact's qualify_frequence and qualify_timeout. There is no point to this since those values already get updated when the AOR changes. * Reloading res_pjsip.so no longer calls the OPTIONS initialization function. Reloading res_pjsip.so results in re-loading AORs, which results in re-scheduling qualifies. Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121	10 years ago
Joshua Colp	bb982480d8	pjsip: Support dual stack automatically. ... This change adds support for dual stack automatically. No configuration is required and the IP address and version in the SIP messages and SDP will be automatically changed based on the transport over which the message is being sent. RTP usage has also been changed to listen on both IPv4 and IPv6 simultaneously to allow media to flow, and to allow ICE support on both simultaneously. This also allows failover between IPv6 and IPv4 to work as expected. ASTERISK-26309 #close Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d	10 years ago
George Joseph	9740277713	res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c) ... There are several places that do scheduled tasks or periodic housecleaning, each with its own implementation: * res_pjsip_keepalive has a thread that sends keepalives. * pjsip_distributor has a thread that cleans up expired unidentified requests. * res_pjsip_registrar_expire has a thread that cleans up expired contacts. * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task. * res_pjsip_sdp_rtp also uses ast_sched to send keepalives. There are also places where we should be doing scheduled work but aren't. A good example are the places we have sorcery observers to start registration or qualify. These don't work when changes are made to a backend database without a pjsip reload. We need to check periodically. As a first step to solving these issues, a new ast_sip_sched facility has been created. ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue. When a task is ready to run, ast_sip_task_pusk is called for it. This ensures that the task is executed in a PJLIB registered thread and doesn't hold up the ast_sched thread so it can immediately continue processing the queue. The serializer used by ast_sip_sched is one of your choosing or a random one from the res_pjsip pool if you don't choose one. Another feature is the ability to automatically clean up the task_data when the task expires (if ever). If it's an ao2 object, it will be dereferenced, if it's a malloc'd object it will be freed. This is selectable when the task is scheduled. Even if you choose to not auto dereference an ao2 task data object, the scheduler itself maintains a reference to it while the task is under it's control. This prevents the data from disappearing out from under the task. There are two scheduling models. AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at the specific interval. That is, every "interval" milliseconds, regardless of how long the task takes. If the task takes longer than the interval, it will be scheduled at the next available multiple of interval. For exmaple: If the task has an interval of 60 secs and the task takes 70 secs (it better not), the next invocation will happen at 120 seconds. AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should start "interval" milliseconds after the current invocation has finished. Also, the same ast_sched facility for fixed or variable intervals exists. The task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time. One res_pjsip.h housekeeping change was made. The pjsip header files were added to the top. There have been a few cases lately where I've needed res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because I didn't add the pjsip header files to my source even though I never referenced any pjsip calls. Finally, a few new convenience APIs were added to astobj2 to make things a little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros were also copied from res_phoneprov because I got tired of having to duplicate the same hash, sort and compare functions over and over again. The AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for aor_container_alloc into your source. This facility can be used immediately for the situations where we already have a thread that wakes up periodically or do some scheduled work. For the registration and qualify issues, additional sorcery and schema changes would need to be made so that we can easily detect changed objects on a periodic basis without having to pull the entire database back to check. I'm thinking of a last-updated timestamp on the rows but more on this later. Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c	11 years ago
Mark Michelson	0b4bb19e0b	AST-2016-004: Fix crash on REGISTER with long URI. ... Due to some ignored return values, Asterisk could crash if processing an incoming REGISTER whose contact URI was above a certain length. ASTERISK-25707 #close Reported by George Joseph Patches: 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch AST-2016-004 Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55	11 years ago
George Joseph	99fb87ae13	res_pjsip: Fix SEGV on pending-qualify contacts ... Permanent contacts that hadn't been qualified yet were missing their contact_status entries causing SEGVs when running CLI commands. This patch makes sure that contact_statuses are created for both dynamic and permanent contacts when they are created. It also adds checks in the CLI code to make sure there's a contact_status, just in case. ASTERISK-25018 #close Reported-by: Ivan Poddubny Tested-by: Ivan Poddubny Tested-by: George Joseph Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029	11 years ago
Kevin Harwell	5737650a67	res_pjsip: add CLI command to show global and system configuration ... Added a new CLI command for res_pjsip that shows both global and system configuration settings: pjsip show settings ASTERISK-24918 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/4597/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	bd029688cd	res_pjsip: Move internal init/destroy prototypes to private header file. ... Done as a separate commit from a finding in https://reviewboard.asterisk.org/r/4467/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	9e3d316dd1	res_pjsip: make it unloadable (take 2) ... Due to the original patch causing memory corruptions it was removed until the problem could be resolved. This patch is the original patch plus some added locking around stasis router subcription that was needed to avoid the memory corruption. Description of the original problem and patch (still applicable): The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4363/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	34c220203f	REVERTING res_pjsip: make it unloadable ... Due to the original patch causing memory corruptions the patch is being removed until the problem can be resolved. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	77a036bf3f	res_pjsip: make it unloadable ... The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4311/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	cc40bf5317	res_pjsip: Enable PJSIP DNS client support. ... This change enables DNS client support within PJSIP. System nameservers are automatically discovered using res_init or res_ninit. If this fails then PJSIP will resort to using gethostbyname for resolution. By enabling this support we gain SRV support, failover, and weight support. (closes issue ASTERISK-23435) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3343/ ........ Merged revisions 410795 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
George Joseph	a94c8562fd	sorcery: Create sorcery instance registry. ... In order to retrieve an arbitrary sorcery instance from a dialplan function (or any place else) there needs to be a registry of sorcery instances. ast_sorcery_init now creates a hashtab as a registry. ast_sorcery_open now checks the hashtab for an existing sorcery instance matching the caller's module name. If it finds one, it bumps the refcount and returns it. If not, it creates a new sorcery instance, adds it to the hashtab, then returns it. ast_sorcery_retrieve_by_module_name is a new function that does a hashtab lookup by module name. It can be called by the future dialplan function. res_pjsip/config_system needed a small change to share the main res_pjsip sorcery instance. tests/test_sorcery was updated to include a test for the registry. (closes issue ASTERISK-22537) Review: http://reviewboard.asterisk.org/r/3184/ ........ Merged revisions 408518 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b5ca213e34	res_pjsip: Updates and adds more PJSIP CLI commands. ... * Adds identify, transport, and registration support to the PJSIP CLI. * Creates three additional callbacks, one for an iterator, one for a comparator, and one for a container. This eliminates the link dependency from higher level modules to lower level ones. * Eliminates duplicate sorting in PJSIP CLI commands. * Cleans up PJSIP CLI output formatting. * Pushes CLI command registration down to the implementing source file. * Adds several ast_sip_destroy_sorcery functions to complement existing ast_sip_sorcery_initialize functions. The destroy functions unregister PJSIP CLI commands and PJSIP CLI formatters. Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/3104/ ........ Merged revisions 407568 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	01a537d515	res_pjsip,compat: INFINITY and NAN undefined ... On some systems the values for INFINITY and NAN are not defined thus causing a build error on those systems. Added definitions for those if they had not previously been defined. (closes issue ASTERISK-23056) Reported by: capouch Patches: inf-nan-patch.txt uploaded by capouch (license 6564) ........ Merged revisions 406788 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b172d369c4	res_pjsip: Add PJSIP CLI commands ... Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. (issue ASTERISK-22610) patches: pjsip_cli_v2.patch uploaded by george.joseph (License 6322) ........ Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	c602b086ed	res_pjsip_messaging: send message to a default outbound endpoint ... In some cases messages need to be sent to a direct URI (sip:<ip address>). This patch adds in that support by using a default outbound endpoint. When sending messages, if no endpoint can be found then the default one is used. To facilitate this a new default_outbound_endpoint option was added to the globals section for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/ ........ Merged revisions 403680 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	05cbf8df9b	res_pjsip: AMI commands and events. ... Created the following AMI commands and corresponding events for res_pjsip: PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few select attributes on each. Events: EndpointList - for each endpoint a few attributes. EndpointlistComplete - after all endpoints have been listed. PJSIPShowEndpoint - Provides a detail list of attributes for a specified endpoint. Events: EndpointDetail - attributes on an endpoint. AorDetail - raised for each AOR on an endpoint. AuthDetail - raised for each associated inbound and outbound auth TransportDetail - transport attributes. IdentifyDetail - attributes for the identify object associated with the endpoint. EndpointDetailComplete - last event raised after all detail events. PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound registrations. Events: InboundRegistrationDetail - inbound registration attributes for each registration. InboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound registrations. Events: OutboundRegistrationDetail - outbound registration attributes for each registration. OutboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. (issue ASTERISK-22609) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	2de42c2a25	Multiple revisions 399887,400138,400178,400180-400181 ... ........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	dfa5be76d4	Expose res_pjsip threadpool options ... Expose initial size, automatic increment, maximum size, and idle timeout as configurable parameters for the res_pjsip thread pool. Review: https://reviewboard.asterisk.org/r/2704/ (closes issue ASTERISK-22143) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	41cd06e03f	Add CLI/AMI commands to force chan_pjsip actions ... For chan_pjsip, this introduces CLI/AMI remote unregistration commands, reworks CLI syntax for sending NOTIFYs, adds AMI qualification support, and adds documentation for PJSIPNotify. This also fixes two refcounting bugs in the outbound registration code. Review: https://reviewboard.asterisk.org/r/2695/ (closes issue ASTERISK-21939) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	f6c7e6355e	Fix remnants of the pjsip renaming ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	735b30ad71	The large GULP->PJSIP renaming effort. ... The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*" * All files that were "res_sip*" are now "res_pjsip*" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago