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
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
...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
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
* 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.
* 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
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
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
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
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
Added an pre-defined integer vector declaration. This makes integer vectors
easier to declare and pass around. Also, added the ability to default a vector
up to a given size with a default value. Lastly, added functionality that
returns the "nth" index of a matching value.
Also, updated a unit test to test these changes.
Change-Id: Iaf4b51b2540eda57cb43f67aa59cf1d96cdbcaa5
This change adds a T.38 format which can be used in a stream
topology to specify that a UDPTL stream needs to be created.
The SDP API has been changed to understand T.38 and create
the UDPTL session, add the attributes, and parse the attributes.
This change does not change the boundary of the T.38 state
machine. It is still up to the channel driver to implement and
act on it (such as queueing control frames or reacting to them).
ASTERISK-26949
Change-Id: If28956762ccb8ead562ac6c03d162d3d6014f2c7
The gist of this work ensures that when a remote SDP is received, it is
merged properly with the local capabilities. The remote SDP is converted
into a stream topology. That topology is then merged with the current
local topology on the SDP state. That new merged topology is then used
to create an SDP. Finally, adjustments are made to RTP instances based
on knowledge gained from the remote SDP.
There are also a battery of tests in this commit that ensure that some
basic SDP merges work as expected.
While this may not sound like a big change, it has the property that it
caused lots of ancillary changes.
* The remote SDP is no longer stored on the SDP state. Biggest reason:
there's no need for it. The remote SDP is used at the time it is being
set and nowhere else.
* Some new SDP APIs were added in order to find attributes and convert
generic SDP attributes into rtpmap structures.
* Writing tests made me realize that retrieving a value from an SDP
options structure, the SDP options needs to be made const.
* The SDP state machine was essentially gutted by a previous commit.
Initially, I attempted to reinstate it, but I found that as it had
been defined, it was not all that useful. What was more useful was
knowing the role we play in SDP negotiation, so the SDP state machine
has been transformed into an indicator of role.
* Rather than storing separate local and joint stream state
capabilities, it makes more sense to keep track of current stream
state and update it as things change.
Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d
In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
This change removes the old epoll support which has not been used or
maintained in quite some time.
The fixed number of file descriptors on a channel has also been removed.
File descriptors are now contained in a growable vector. This can be
used like before by specifying a specific position to store a file
descriptor at or using a new API call, ast_channel_fd_add, which adds
a file descriptor to the channel and returns its position.
Tests have been added which cover the growing behavior of the vector
and the new API call.
ASTERISK-26885
Change-Id: I1a754b506c009b83dfdeeb08c2d2815db30ef928
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
* 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
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
This change adds unit tests cover the following:
1. That retrieving the first media stream of a specific media
type from a stream topology retrieves the expected media
stream.
2. That setting the native formats of a channel which does
not support streams results in the creation of streams on
its behalf according to the formats of the channel.
3. That setting a stream topology on a channel which supports
streams sets the topology to the provided one.
ASTERISK-26790
Change-Id: Ic53176dd3e4532e8c3e97d9e22f8a4b66a2bb755
This change adds unit tests for the various API calls relating
to stream topologies. This includes creation, destruction,
inspection, and manipulation.
Through this a few bugs were uncovered in the implementation:
1. Creating a topology using a format capabilities would fail as
the code considered a return value of 0 from the append stream
function to indicate an error which is incorrect.
2. Not all functions which placed a stream into a topology
set the position on the stream itself.
3. Appending a stream would cause a frack if the position
provided was the last one. This occurred because the existing
stream was queried but the index was outside of what the
vector was currently at for size.
ASTERISK-26786
Change-Id: Id5590e87c8a605deea1a89e53169a9c011d66fa0
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
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)
Fix order of parameters in calls to VM_API_INT_VERIFY and
VM_API_STRING_VERIFY
ASTERISK-26739 #close
Change-Id: I30dc6b36893aadad6012be3f16f93aa5720870d6
Note: status: builds. Not tested any further.
Feeding LISTFILTER an empty variable results in an invalid ERROR message.
Earlier changes made the message useless because we can no longer tell if
the variable is empty or does not exist. It is valid to try to remove a
value from an empty list just as it is valid to try to remove a value that
is not in a non-empty list.
* Removed the outdated ERROR message.
* Added more test cases to the LISTFILTER unit test.
Change-Id: Ided9040e6359c44a335ef54e02ef5950a1863134
Some (voicemail-related) tests API symlinks beep.gsm and other files
from ast_config_AST_VAR_DIR. It should use ast_config_AST_DATA_DIR.
ASTERISK-26740 #close
Change-Id: Id49c56fb9e16df64b1a2b829693ca7601252df89
Fix the tests for DNS to use older style nameser.h as
in ASTERISK-26608.
Tested on: OpenBSD 6.0, Debian 8
ASTERISK-26647 #close
Change-Id: I285913c44202537c04b3ed09c015efa6e5f9052d
One of the code paths in __ast_file_read_dirs will only get executed if
the OS doesn't support dirent->d_type OR if the filesystem the
particular file is on doesn't support it. So, while standard Linux
systems support the field, some filesystems like XFS do not. In this
case, we need to call stat() to determine whether the directory entry
is a file or directory so we append the filename to the supplied
directory path and call stat. We forgot to truncate path back to just
the directory afterwards though so we were passing a complete file name
to the callback in the dir_name parameter instead of just the directory
name.
The logic has been re-written to only create a full_path if we need to
call stat() or if we need to descend into another directory.
Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba
The readdir_r function has been deprecated and should no longer be used. This
patch removes the readdir_r dependency (replaced it with readdir) and also moves
the directory search code to a more centralized spot (file.c)
Also removed a strict dependency on the dirent structure's d_type field as it
is not portable. The code now checks to see if the value is available. If so,
it tries to use it, but defaults back to using the stats function if necessary.
Lastly, for most implementations of readdir it *should* be thread-safe to make
concurrent calls to it as long as different directory streams are specified.
glibc falls into this category. However, since it is possible that there exist
some implementations that are not safe, locking has been added for those other
than glibc.
ASTERISK-26412
ASTERISK-26509 #close
Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
The test uses 4 threads to grow, count, lookup and shrink 15K objects
in a container. If there's only 1 execution engine available, the test
will complete in <50ms. If each threads gets its own execution engine,
the test may timeout after 60 seconds because the count thread does a
locked ao2_callback on the whole container in a tight loop with only
a sched_yield to give up time. The lock contention makes the test
execution times wildly variable and mostly timeout. 2 execution
engines are OK, 3 results in about 33% failure rate and >=4 causes
a 80% failure rate.
To fix, the sched_yield was changed to a usleep(500).
Also, the number of buckets specified for the container was an even
number so that was changed to the next prime number greater than
(MAX_HASH_ENTRIES / 100). That's 151 currently.
Change-Id: I50cd2344161ea61bfe4b96d2a29a6ccf88385c77
This patch adds the Asterisk EID field to all outgoing ARI events.
Because this field should be added to all events as they are
transmitted, it is appended to the JSON message just prior to it being
handed off to the application message handler. This makes it somewhat
resilient to both new events being added to ARI, as well as other
potential event transport mechanisms.
ASTERISK-26470 #close
Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d
* Updated unit test as ast_json_name_number() is now NULL tolerant.
ASTERISK-26466 #close
Reported by: Richard Mudgett
Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6
updated the uri handler to include the url prefix of the http server
this enables res_ari to add it to the uris when generating docs
Change-Id: I279335a2625261a8492206c37219698f42591c2e
(cherry picked from commit 6f448f32fe)
If the SQL UPDATE statement changes nothing then SQLRowCount returns 0.
This value should be treated as success.
But the function sorcery_realtime_update treats it as failed.
This bug was found using stress tests on PJSIP.
If there are 2 consecutive SIP REGISTER requests with the same contact data
during 1 second then res_pjsip_registrar adds contact location on 1st request
and tries to update contact location on 2nd.
The update fails and res_pjsip_registrar even removes correct contact location.
The test "object_update_uncreated" was removed from test_sorcery_realtime.c
because it's now a valid situation.
This patch also adds missing debug of extra SQL parameter.
ASTERISK-26172 #close
Change-Id: I05a7f3051455336c9dda29efc229decf86071303
Since the file was missing the depends on pjproject, it wasn't
picking up the pjproject related include path. If there was no
system installed pjproject and pjproject-bundled was used, a compile
would fail because pjsip.h wasn't found.
ASTERISK-26139 #close
Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757
CEL wrongly assumed that a channel would only have a single dial
event on it. This is incorrect. Particularly in a queue each
call attempt to a member will result in a dial event, adding
a new dial status in CEL without removing the old one. This
would cause the container to grow with only one dial status
being removed when the channel went away. The other dial status
entries would remain leaking memory.
This change fixes the memory leak by ensuring that only one dial
status will only ever exist for each channel.
The behavior during the scenario where multiple events are received
has also been improved. For failure cases the first failure will
be the dial status. If an answer dial status is received, though,
it will take priority and the dial status for the channel will be
answer.
Memory usage has also been decreased by storing the minimal
amount of information and the code has been cleaned up slightly.
ASTERISK-25262 #close
Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe
The retrieve_cache_control_directives test has been failing occasionally
in Jenkins. The apparent failure occurs when attempting to validate the
expiration of the retrieved file.
After reproducing, the problem was pretty clear. At the beginning of the
test, the current time is retrieved. The seconds value of this timestamp
is X. When the file is retrieved, res_http_media_cache calculates the
expiration and in doing so retrieves the current time. In most cases,
since the test executes quickly, it will also retrieve a timestamp with
X seconds. However, if the test starts very near to when the timestamp
seconds are set to increment, res_http_media_cache may retrieve a
timestamp with X+1 seconds instead.
The test attempted to account for this by allowing a tolerance of 1
second when validating the expiration. However, the problem was that the
comparisons being used in the validation used > and < operations. This
meant that values that fell within the tolerance (because they equaled
the upper bound of the tolerance) would fail.
The solution is to use >= and <= operators in the expiration validation.
However, I estimated that while the one second tolerance should be
fine on most machines, it would still be possible on a very slow machine
to end up falling outside the one second tolerance. So I have also
relaxed the tolerance of expiration validation to be three seconds
instead.
The final change here is to add a debug message when validating
expiration so that we can see what values are being compared.
ASTERISK-25959 #close
Reported by Joshua Colp
Change-Id: Ic1a0e10722c1c5d276d5a4d6a67136d6ec26c247
