During some performance testing of Asterisk with AGI, ARI, and lots of Local
channels, we noticed that there's quite a hit in performance during channel
creation and releasing to the dialplan (ARI continue). After investigating
the performance spike that occurs during channel creation, we discovered
that we create a lot of channel snapshots that are technically unnecessary.
This includes creating snapshots during:
* AGI execution
* Returning objects for ARI commands
* During some Local channel operations
* During some dialling operations
* During variable setting
* During some bridging operations
And more.
This patch does the following:
- It removes a number of fields from channel snapshots. These fields were
rarely used, were expensive to have on the snapshot, and hurt performance.
This included formats, translation paths, Log Call ID, callgroup, pickup
group, and all channel variables. As a result, AMI Status,
"core show channel", "core show channelvar", and "pjsip show channel" were
modified to either hit the live channel or not show certain pieces of data.
While this is unfortunate, the performance gain from this patch is worth
the loss in behaviour.
- It adds a mechanism to publish a cached snapshot + blob. A large number of
publications were changed to use this, including:
- During Dial begin
- During Variable assignment (if no AMI variables are emitted - if AMI
variables are set, we have to make snapshots when a variable is changed)
- During channel pickup
- When a channel is put on hold/unhold
- When a DTMF digit is begun/ended
- When creating a bridge snapshot
- When an AOC event is raised
- During Local channel optimization/Local bridging
- When endpoint snapshots are generated
- All AGI events
- All ARI responses that return a channel
- Events in the AgentPool, MeetMe, and some in Queue
- Additionally, some extraneous channel snapshots were being made that were
unnecessary. These were removed.
- The result of ast_hashtab_hash_string is now cached in stasis_cache. This
reduces a large number of calls to ast_hashtab_hash_string, which reduced
the amount of time spent in this function in gprof by around 50%.
#ASTERISK-23811 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3568/
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Simply establishing a TCP connection and never sending anything to the
configured HTTP port in http.conf will tie up a HTTP connection. Since
there is a maximum number of open HTTP sessions allowed at a time you can
block legitimate connections.
A similar problem exists if a HTTP request is started but never finished.
* Added http.conf session_inactivity timer option to close HTTP
connections that aren't doing anything. Defaults to 30000 ms.
* Removed the undocumented manager.conf block-sockets option. It
interferes with TCP/TLS inactivity timeouts.
* AMI and SIP TLS connections now have better authentication timeout
protection. Though I didn't remove the bizzare TLS timeout polling code
from chan_sip.
* chan_sip can now handle SSL certificate renegotiations in the middle of
a session. It couldn't do that before because the socket was non-blocking
and the SSL calls were not restarted as documented by the OpenSSL
documentation.
* Fixed an off nominal leak of the ssl struct in
handle_tcptls_connection() if the FILE stream failed to open and the SSL
certificate negotiations failed.
The patch creates a custom FILE stream handler to give the created FILE
streams inactivity timeout and timeout after a specific moment in time
capability. This approach eliminates the need for code using the FILE
stream to be redesigned to deal with the timeouts.
This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of
the SSL_read/SSL_write operations.
ASTERISK-23673 #close
Reported by: Richard Mudgett
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This adds the TCP_NODELAY option to accepted connections on the HTTP
server built into Asterisk. This option disables the Nagle algorithm
which controls queueing of outbound data and in some cases can cause
delays on receipt of response by the client due to how the Nagle
algorithm interacts with TCP delayed ACK. This option is already set on
all non-HTTP AMI connections and this change would cover standard HTTP
requests, manager HTTP connections, and ARI HTTP requests and
websockets in Asterisk 12+ along with any future use of the HTTP
server.
Review: https://reviewboard.asterisk.org/r/3466/
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This patch works around a problem with the HTTP body
being dropped from the response to a specific client
and under specific circumstances:
a) Client request comes from node.js user agent
"Shred" via use of swagger-client library.
b) Asterisk and Client are *not* on the same
host or TCP/IP stack
In testing this problem, it has been determined that
the write of the HTTP body is lost, even if the data
is written using low level write function. The only
solution found is to instruct the TCP stack with the
shutdown function to flush the last write and finish
the transmission. See review for more details.
ASTERISK-23548 #close
(closes issue ASTERISK-23548)
Reported by: Sam Galarneau
Review: https://reviewboard.asterisk.org/r/3402/
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* Fix memory leak in ast_unreal_new_channels(). Made it generate the ;2
uniqueid on a stack variable instead of mallocing it.
* Made send error response to ARI and AMI requests instead of just logging
excessive uniqueid length and allowing truncation. action_originate() and
ari_channels_handle_originate_with_id().
* Fixed minor truncating uniqueid hole when generating the ;2 uniqueid
string length. Created public and internal lengths of uniqueid. The
internal length can handle a max public uniqueid plus an appended ;2.
* free() and ast_free() are NULL tolerant so they don't need a NULL test
before calling.
* Made use better struct initialization format instead of the position
dependent initialization format. Also anything not explicitly initialized
in the struct is initialized to zero by the compiler.
* Made ast_channel_internal_set_fake_ids() use the safer
ast_copy_string() instead of strncpy().
Review: https://reviewboard.asterisk.org/r/3371/
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Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an 'ExtensionStatus' event was raised it included the status as a
numerical value, but did not include a text description of the status.
Added a 'StatusText' field to the event which is a string representation
of the extension status. Also added this to the 'Extension State' command
response.
(closes issue ASTERISK-23154)
Reported by: Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic
number" error on a "core restart gracefully" command if an AMI connection
is established.
* Added ao2_global_obj protection to the sessions global container.
* Fixed the order of unreferencing a session object in session_destroy().
* Removed unnecessary container traversals of the white/black filters
during session_destructor().
(closes issue AST-1242)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3144/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The per console verbose level feature as previously implemented caused a
large performance penalty. The fix required some minor incompatibilities
if the new rasterisk is used to connect to an earlier version. If the new
rasterisk connects to an older Asterisk version then the root console
verbose level is always affected by the "core set verbose" command of the
remote console even though it may appear to only affect the current
console. If an older version of rasterisk connects to the new version
then the "core set verbose" command will have no effect.
* Fixed the verbose performance by not generating a verbose message if
nothing is going to use it and then filtered any generated verbose
messages before actually sending them to the remote consoles.
* Split the "core set debug" and "core set verbose" CLI commands to remove
the per module verbose support that cannot work with the per console
verbose level.
* Added a silent option to the "core set verbose" command.
* Fixed "core set debug off" tab completion.
* Made "core show settings" list the current console verbosity in addition
to the root console verbosity.
* Changed the default verbose level of the 'verbose' setting in the
logger.conf [logfiles] section. The default is now to once again follow
the current root console level. As a result, using the AMI Command action
with "core set verbose" could again set the root console verbose level and
affect the verbose level logged.
(closes issue AST-1252)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3114/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AMI action UserEvent event response would include the action header in its
keyvalue pairs list. Adjusted the start of the header loop to skip over the
action part.
(closes issue ASTERISK-22899)
Reported by: outtolunc
Patches:
svn_manager.c.skip_action.diff.txt uploaded by outtolunc (license 5198)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ARI POST calls only accept parameters via the URL's query string.
While this works, it's atypical for HTTP API's in general, and
specifically frowned upon with RESTful API's.
This patch adds parsing for application/x-www-form-urlencoded request
bodies if they are sent in with the request. Any variables parsed this
way are prepended to the variable list supplied by the query string.
(closes issue ASTERISK-22743)
Review: https://reviewboard.asterisk.org/r/2986/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A common idiom in Asterisk is to due something like:
for (ao2_obj = list_beginning; ao2_obj = next_item; ao2_ref(ao2_obj, -1)) {
...do stuff...
}
This is nice because it automatically takes care of the object references
for you. However, there is a pitfall here. If a break statement is in the
for loop, then the current reference is not cleaned up. In some cases, this
is on purpose, but in others there is a leak. This commit fixes the leak
cases.
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Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.
Review: https://reviewboard.asterisk.org/r/2879
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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
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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/
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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/
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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/
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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/
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In r399887, a minor performance improvement was introduced by not allocating
the manager variable struct if it wasn't used. Unfortunately, when directly
accessing an ast_channel struct, manager assumed that the struct was always
allocated. Since this was no longer the case, things got a bit crashy.
This fixes that problem by simply bypassing appending variables if the manager
channel variable struct isn't there.
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This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.
Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Bridge API DTMF hook matching would not deal with DTMF end events
only. It required a DTMF begin event to start matching the DTMF hooks.
There are many places in Asterisk where code only generates DTMF end
events without the corresponding begin event. One such place is the AMI
action Atxfer.
* Fixed DTMF hook matching if there is a string of DTMF frames in the read
queue. We could potentially miss some of them before.
* Fixed AMI Atxfer action documentation.
(closes issue ASTERISK-22037)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2752/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the module
won't load on Linux.
This patch renames the variables, adding the ast_ prefix so they will
be exported.
Review: https://reviewboard.asterisk.org/r/2737
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.
To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.
In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:
single_topic ----------------> all_topic
^
|
single_topic_cached ----+----> all_topic_cached
|
+----> cache
This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.
Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.
(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.
A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.
(closes issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies manager to allow the allowmultiplelogin setting to be set
on an account by account basis. When set in the general context, it will act
as the default for the defined accounts. Setting it in the account will
override the general setting.
(closes issue ASTERISK-21324)
Reported by: vldmr
patches:
asterisk-manager-per-user-allowmultiplelogin.patch uploaded by vldmr (License 6487)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel
information in the RTCP events. Because Stasis provides a cache, Jaco's
patch was modified to pass the channel uniqueid to the RTP layer as
opposed to a pointer to the channel. This has the following benefits:
(1) It keeps the RTP engine 'clean' of references back to channels
(2) It prevents circular dependencies and other potential ref counting issues
* The RTP engine now allows any RTP implementation to raise RTCP messages.
Potentially, other implementations (such as res_rtp_multicast) could also
raise RTCP information. The engine provides structs to represent RTCP headers
and RTCP SR/RR reports.
* Some general refactoring in res_rtp_asterisk was done to try and tame the
RTCP code. It isn't perfect - that's *way* beyond the scope of this work -
but it does feel marginally better.
* A few random bugs were fixed in the RTCP statistics. (Example: performing an
assignment of a = a is probably not correct)
* We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't
raise an event when we sent a RR report.
Note that this work will be of use to others who want to monitor call quality
or build modules that report call quality statistics. Since the events are now
moving across the Stasis message bus, this is far easier to accomplish. It is
also a first step (though by no means the last step) towards getting Olle's
pinefrog work incorporated.
Again: note that the patch by Jaco Kroon was modified slightly for this work;
however, he did all of the hard work in finding the right places to set the
channel in the RTP engine across the channel drivers. Much thanks goes to Jaco
for his hard work here.
Review: https://reviewboard.asterisk.org/r/2603/
(closes issue ASTERISK-20574)
Reported by: Jaco Kroon
patches:
asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)
(closes issue ASTERISK-21471)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Refactored the AMI events in AOC onto Stasis-Core. The ast_aoc_manager_event
function now publishes a channel snapshot, along with a JSON blob describing
the advice of charge. A "to_ami" handler has also been added that converts
the channel snapshot and AOC event data back into the appropriate data structure
for use with AMI.
(closes issue ASTERISK-21472)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2643/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The originate APIs allow callers to provide a pointer to a channel that will
point to the originated channel if the function call succeeds. This is used by AMI
to provide channel information when the originate is performed synchronously.
Unfortunately, if the originate fails in certain ways, the outbound channel is
already disposed of during the dialing itself. This results in the channel being
improperly dereferenced by the internal originate function in pbx.c.
This patch ref bumps the channel to prevent this from occurring. Callers must now
unlock and unref the channel (which is more in line with general channel management
guidelines anyway).
This only affects manager, as it is the only consumer of this API function that
actually passes in a channel pointer.
Review: https://reviewboard.asterisk.org/r/2617/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
