The stream topology has no lock of its own resulting in
another lock protecting it in some way (for example the
channel lock). If multiple channels are being juggled at
the same time this can be problematic. This change makes
the topology a reference counted object instead which
guarantees it will remain valid even without the channel
lock being held.
Change-Id: I4f4d3dd856a033ed55fe218c3a4fab364afedb03
How it works today:
media_cache tries to parse out the extension of the media file to be played
from the URI provided to Asterisk while caching the file.
What's expected:
Better will be to have Asterisk get extension from other ways too. One of the
common ways is to get the type of content from the CONTENT-TYPE header in the
HTTP response for fetching the media file using the URI provided.
Steps to Reproduce:
Provide a URL of the form: http://host/media/1234 to Asterisk for media
playback. It fails to play and logs show the following error line:
[Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c:
File http://host/media/1234 does not exist in any format
Scenario this issue is blocking:
In the case where the media files are stored in some cloud object store,
following can block the media being played via Asterisk:
Cloud storage generally needs authenticated access to the storage. The way
to do that is by using signed URIs. With the signed URIs there's no way to
preserve the name of the file.
In most cases Cloud storage returns a key to access the object and preserving
file name is also not a thing there
ASTERISK-27286
Reporter: Gaurav Khurana
Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89
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
This change adds the ability for multiple REMB reports in
bridge_softmix to be combined according to a configured
behavior into a single report. This single report is sent
back to the sender of video, which adjusts the encoding bitrate
to be at or below the bitrate of the report. The available
behaviors are: lowest, highest, and average. Lowest uses the
lowest received bitrate. Highest uses the highest received
bitrate. Average goes through the received bitrates adding
them to the previous average and creates a new average.
Other behaviors can be added in the future and the existing
average one may be adjusted, but this provides the foundation
to do so.
Support for configuring which behavior to use has been
added to app_confbridge.
ASTERISK-27804
Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66
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
Similar to pjproject's PJ_ASSERT_RETURN macro, this one will do the
following...
If the assert passes... NoOp
If the assert fails and AST_DEVMODE is defined, execute ast_assert()
then, if DO_CRASH isn't set, return from the calling function with
the supplied value.
If the assert fails and AST_DEVMODE is not defined, return from the
calling function with the supplied value.
The macro will execute a return without a value if one isn't suppled.
Change-Id: I0003844affeab550d5ff5bca7aa7cf8a559b873e
Core bridging and, more specifically, bridge_softmix have been
enhanced to relay received frames of type TEXT or TEXT_DATA to all
participants in a softmix bridge. res_pjsip_messaging and
chan_pjsip have been enhanced to take advantage of this so when
res_pjsip_messaging receives an in-dialog MESSAGE message from a
user in a conference call, it's relayed to all other participants
in the call.
res_pjsip_messaging already queues TEXT frames to the channel when
it receives an in-dialog MESSAGE from an endpoint and chan_pjsip
will send an MESSAGE when it gets a TEXT frame. On a normal
point-to-point call, the frames are forwarded between the two
correctly. bridge_softmix was not though so messages weren't
getting forwarded to conference bridge participants. Even if they
were, the bridging code had no way to tell the participants who
sent the message so it would look like it came from the bridge
itself.
* The TEXT frame type doesn't allow storage of any meta data, such
as sender, on the frame so a new TEXT_DATA frame type was added that
uses the new ast_msg_data structure as its payload. A channel
driver can queue a frame of that type when it receives a message
from outside. A channel driver can use it for sending messages
by implementing the new send_text_data channel tech callback and
setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech
properties. If set, the bridging/channel core will use it instead
of the original send_text callback and it will get the ast_msg_data
structure. Channel drivers aren't required to implement this. Even
if a TEXT_DATA enabled driver uses it for incoming messages, an
outgoing channel driver that doesn't will still have it's send_text
callback called with only the message text just as before.
* res_pjsip_messaging now creates a TEXT_DATA frame for incoming
in-dialog messages and sets the "from" to the display name in the
"From" header, or if that's empty, the caller id name from the
channel. This allows the chat client user to set a friendly name
for the chat.
* bridge_softmix now forwards TEXT and TEXT_DATA frames to all
participants (except the sender).
* A new function "ast_sendtext_data" was added to channel which
takes an ast_msg_data structure and calls a channel's
send_text_data callback, or if that's not defined, the original
send_text callback.
* bridge_channel now calls ast_sendtext_data for TEXT_DATA frame
types and ast_sendtext for TEXT frame types.
* chan_pjsip now uses the "from" name in the ast_msg_data structure
(if it exists) to set the "From" header display name on outgoing text
messages.
Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489
Adds the ability to receive and handle incoming NACK requests if
retransmissions are enabled. If retransmissions are enabled, a data
buffer is allocated that stores packets being sent. If a NACK request
is received, the packet requested for retransmission is sent if it is
still in the buffer. In the same request, if any of the following 16
packets are marked as not received, those will be sent as well if
available, as outlined in RFC4585.
Also changes RTCP RR and SR to use media source SSRC instead of packet
source SSRC when determining which instance to use for RTCP reports.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
ASTERISK-27806 #close
Change-Id: I7f7f124af3b9d5d2fd9cffc6ba8cb48a6fff06ec
Use of extended stringfields is a temporary mechanism to avoid ABI
breakage in released branches without resorting to more inconvienient
methods.
* Collect existing extended stringfields into the parent stringfield
section of the struct.
Change-Id: I8d46d037801b4518837c3ea4b6df95ceadc9436b
When a scheduled task is created you can pass in the
AST_SIP_SCHED_TASK_TRACK flag. This new flag causes scheduling events to
be logged.
Change-Id: I91967eb3d5a220915ce86881a28af772f9a7f56b
ast_sip_push_task_synchronous() did not necessarily execute the passed in
task under the specified serializer. If the current thread is any
registered pjsip thread then it would execute the task immediately instead
of under the specified serializer. Reentrancy issues could result if the
task does not execute with the right serializer.
The original reason ast_sip_push_task_synchronous() checked to see if the
current thread was a registered pjsip thread was because of a deadlock
with masquerades and the channel technology's fixup callback
(ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356)
involving call pickups avoided the original deadlock situation entirely.
The PJSIP channel technology's fixup callback no longer needed to call
ast_sip_push_task_synchronous().
However, there are a few places where this unexpected behavior is still
required to avoid deadlocks. The pjsip monitor thread executes callbacks
that do calls to ast_sip_push_task_synchronous() that would deadlock if
the task were actually pushed to the specified serializer. I ran into one
dealing with the pubsub subscriptions where an ao2 destructor called
ast_sip_push_task_synchronous().
* Split ast_sip_push_task_synchronous() into
ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer().
ast_sip_push_task_wait_servant() has the old behavior of
ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has
the new behavior where the task is always executed by the specified
serializer or a picked serializer if one is not passed in. Both functions
behave the same if the current thread is not a SIP servant.
* Redirected ast_sip_push_task_synchronous() to
ast_sip_push_task_wait_servant() to preserve API for released branches.
ASTERISK_26806
Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3
* Fix the periodic interval wander because it may take significant time
between the sched thread queueing the task in the serializer and the
serializer actually executing the task. The time it takes to actually
execute the task was already taken into account.
* Pass a schtd ref to the serializer when we queue a scheduled task on
the serializer. We don't want it going away on us while it is in the
serializer queue.
* Skip the scheduled task if the task was canceled between queueing the
task to the serializer and the serializer actually executing the task.
* Reorder struct ast_sip_sched_task to avoid unnecessary padding. Removed
task_id and added next_periodic.
* Hold a ref to the passed in serializer so the serializer cannot go away
on the scheduled task.
ASTERISK_26806
Change-Id: I6c8046b75f6953792c8c30e55b836a4291143f24
* Removed several invalid uses of OBJ_NOLOCK. These uses resulted in the
'tasks' container being accessed without a lock in a multi-threaded
environment. A recipe for crashes.
* Removed needlessly obtaining schtd object references. If the caller
providing you a pointer to an object doesn't have a valid reference then
you cannot safely get one from it.
* Getting a ref to 'tasks' when you aren't copying the pointer into
another location is useless. The 'tasks' container pointer is global.
* Removed many unnecessary uses of RAII_VAR.
* Make ast_sip_schedule_task() name parameter const.
ASTERISK_26806
Change-Id: I5c62488e651314e2a1dbc01f5b078a15512d73db
There is a problem when an INVITE-with-Replaces transfer targets a channel
in a ConfBridge. The transfer will unconditionally swap out the
ConfBridge channel. Unfortunately, the ConfBridge state will not be aware
of this change. Unexpected behavior will happen as a result since
ConfBridge channels currently can only be replaced by a masquerade and not
normal bridge channel moves.
* We just need to pretend that the channel isn't in a bridge (like other
transfer methods already do) so the transfer channel will masquerade into
the ConfBridge channel.
Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82
The previous payload specific feedback handling was very single
minded in that it just assumed everything should trigger a video
update. This was changed but the handling of picture loss indication
was not added. The result was that video may not flow. This change
adds it explicitly in.
Change-Id: I1894be02e39ee10a0af841b5a1dca5f0ec7d60b6
This change adds a configuration option to app_confbridge which can be
used to set the interval at which we will send a combined REMB (remote
estimated maximum bitrate) frame to sources of video. The bridging API
has also been extended slightly to allow setting this so bridge_softmix
can use it.
ASTERISK-27786
Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82
Adds a data buffer with a configurable size that can store different
kinds of packets (like RTP packets for retransmission). Given a number
it will store a data packet at that position relative to the others.
Given a number it will retrieve the given data packet if it is present.
This is purposely a storage of arbitrary things so it can be used not
just for RTP packets but also Asterisk frames in the future if needed.
The API does not internally use a lock, so it will be up to the user of
the API to properly protect the data buffer.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Iff13c5d4795d52356959fe2a57360cd57dfade07
This change extends the existing AST_FRAME_RTCP frame type to be
able to contain additional RTCP message types, such as feedback
messages. The payload type is contained in the subclass which allows
knowing what is in the frame itself.
The RTCP feedback message type is now handled and REMB[1] messages
are raised with their containing information.
This also fixes a bug where all feedback messages were triggering
video updates instead of just FIR and FUR.
Finally RTCP frames are now passed up through the Asterisk core to
what is handling the channel, mapped appropriately in the case of
bridging, and written to an outgoing stream. Since RTCP frames are
on a per-stream basis this is only done on multistream capable
channels.
[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
ASTERISK-27758
ASTERISK-26366
Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e
This creates a separate source to 'own' symbols related to options.h and
paths.h. This significantly reduces the number of exports created by
main/asterisk.o. This change is required to eventually be able to
link unmodified Asterisk sources to utilities and/or stand-alone tests.
ASTERISK~26245
Change-Id: I5cf184f4757f9363b80c9e678bdc35c477122380
* 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
If the two formats on a channel are equal, we don't transcode and since
the generic plc needs slin to work, it doesn't get invoked.
* A new configuration option "genericplc_on_equal_codecs" was added
to the "plc" section of codecs.conf to allow generic packet loss
concealment even if no transcoding was originally needed.
Transcoding via SLIN is forced in this case.
ASTERISK-27743
Change-Id: I0577026a179dea34232e63123254b4e0508378f4
This change adds a property to RTP instances to indicate that
REMB support is enabled and that sending/receiving should be
passed through.
This also enables it on video RTP instances in PJSIP if
WebRTC support is enabled.
Finally the goog-remb extension is added to the SDP using
the rtcp-fb attribute to indicate our support for it.
Details about REMB can be found on the draft document for it:
https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
Change-Id: I1902dda1c0882bd1a0d71b2f120684b44b97e789
Asterisk does not need the development package of libltdl, because it does not
use any symbol of -lltdl directly. Instead, it uses the runtime package via the
shared library -lodbc. On the supported platforms, that shared library declares
its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have
failed.
ASTERISK-27745
Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba
This replaces AST_INLINE_API allocators in utils.h with real functions
implemented in astmm.c. Associated macro's are also moved from utils.h
to astmm.h.
Remove menuselect conflicts between MALLOC_DEBUG and DEBUG_CHAOS as they
can now be combined.
This has multiple benefits:
* Simplifies asterisk/utils.h by removing inline functions and use of
the logger.
* Removal of these inline functions decreases size of Asterisk and
module binaries by 1% or more.
* Puts memory management functions together with and without
MALLOC_DEBUG enabled, simplifying management of the code.
* Enables DEBUG_CHAOS for ASTMM_REDIRECT and bundled pjproject.
Change-Id: If9df4377f74bdbb627461b27a473123e05525887
* acl (named_acl.c)
* cdr
* cel
* ccss
* dnsmgr
* dsp
* enum
* extconfig (config.c)
* features
* http
* indications
* logger
* manager
* plc
* sounds
* udptl
These modules are now loaded at appropriate time by the module loader.
Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so
the module loader will abort startup on failure of these modules.
Some of these modules are still initialized or shutdown from outside the
module loader. logger.c is initialized very early and shutdown very
late, manager.c is initialized by the module loader but is shutdown by
the Asterisk core (too much uses it without holding references).
Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f
GCC documentation states that when __attribute__((malloc)) is used it
should not return storage which contains any valid pointers. It
specifically mentions that realloc functions should not have the malloc
attribute, but this also means that complex initializers which could
contain initialized pointers should not use this attribute.
Change-Id: If507f33ffb3ca3b83b702196eb0e8215d27fc7d2
This causes MALLOC_DEBUG reporting to be slightly different, calls which
cause additional memory pools to be allocated now report the callers
location rather than the location which originally allocated the
string field structure. This reduces storage needed by string fields
and allows MALLOC_DEBUG to identify the source of additional allocations
rather than obscuring it by reporting the original allocation caller.
Change-Id: Idd18e6639a87ab862079b580c114d90361412289
When built-in components of Asterisk fail to start they cause the
Asterisk startup to abort. In these cases only the most critical
cleanup should be performed - closing databases and terminating
proceses. These cleanups are registered using ast_register_atexit, all
other cleanups should not be run during startup abort.
The main reason for this change is that these cleanup procedures are
untestable from the partially initialized states, if they fail it could
prevent us from ever running the critical cleanup with ast_run_atexits.
Create separate initialization for dns_core.c to be run unconditionally
during startup instead of being initialized by the first dns resolver to
be registered. This ensures that 'sched' is initialized before it can be
potentially used.
Replace ast_register_atexit with ast_register_cleanup in media_cache.c.
There is no reason for this cleanup to happen unconditionally.
Change-Id: Iecc2df98008b21509925ff16740bd5fa29527db3
In NetBSD, PortAudio 1 is still the default version. PortAudio 2 can be
installed side by side but gets placed in a 'portaudio2' subdirectory. To
find PortAudio 2 even in a subdirectory, the tool pkg-config is queried via
AST_PKG_CONFIG_CHECK. For those platforms, which do not list PowerAudio 2
via pkg-config, the previous check remains and is executed thereafter.
ASTERISK-27721
Change-Id: I4175500126909ad1b181fff8e11bb4a3a6ae4fa9
astosp.h is leftover from when logic was split between app_osplookup and
res_osp. All logic was moved into app_osplookup by 109737eb1c in 2006,
but astosp.h remained. This moves the remaining defines into
app_osplookup and deletes astosp.h.
Change-Id: I0a6c4debd7c9543b608520b1765abfa4fab7b2fd
This allows asterisk to be compiled with MALLOC_DEBUG to load modules
built without MALLOC_DEBUG. Now pre-compiled third-party modules will
still work regardless of MALLOC_DEBUG being enabled or not.
Change-Id: Ic07ad80b2c2df894db984cf27b16a69383ce0e10
The pool cache gets in the way of finding use after free errors of memory
pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a
pool is released because it gets put into the cache instead of being
freed.
* Added the "cache_pools" option to pjproject.conf. Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the pool
contents are used after free and who freed it.
To disable the pool caching simply disable the cache_pools option in
pjproject.conf and restart Asterisk.
Sample pjproject.conf setting:
[startup]
cache_pools=no
* Made current users of the caching pool factory initialization and
destruction calls call common routines to create and destroy cached pools.
ASTERISK-27704
Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828
A couple of additional properties are needed in rtp_engine to enable
support for packet retransmission: AST_RTP_PROPERTY_RETRANS_RECV and
AST_RTP_PROPERTY_RETRANS_SEND. These will both be enabled automatically
if an endpoint has the webrtc option enabled. While this adds no
functionality currently, it will serve as a building block for future
changes for RTP retransmission support.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc
The dsp_talking_threshold does not represent time in milliseconds. It
represents the average magnitude per sample in the audio packets. This is
what the DSP uses to determine if a packet is silence or talking/noise.
Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443
Create ast_atomic macro's to provide a consistent interface to the
common functionality of __atomic and __sync built-in functions.
ASTERISK-27619
Change-Id: Ieba3f81832a0e25c5725ea067e5d6f742d33eb5b
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
This addresses all performance issues with 'module load' completion. In
addition to using ast_cli_completion_add we stop using libedit's
filename_completion_function, instead using ast_file_read_dir. This
ensures all results are produced from a single call to opendir.
Change-Id: I8bf51ffaa7ef1606f3bd1b5bb13f1905d72c6134
Because of a copy-and-paste error, the Asterisk project was using __typeof
instead of typeof. It works because typeof, __typeof, and __typeof__ are
supported by GCC, but here the escaped variant was not intended. Therefore,
for consistence, we change this to typeof.
Change-Id: I2a962c3e596e882f691a19345445b14571a5f07c
This change causes the configure script to throw an error if neither
__sync nor __atomic builtin functions are available.
ASTERISK-27619
Change-Id: Ie01a281e0f5c41dfeeb5f250c1ccea8752f56ef9
Add a check to configure.ac for __atomic_fetch_add support. If found
use the __atomic built-in operators for ast_atomic_dec_and_test and
ast_atomic_fetchadd_int.
ASTERISK~27619
Change-Id: I65b4feb02bae368904ed0fb03f585c05f50a690e
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
* Copy more than one character at a time when there is nothing to
substitute.
* Fix off by one error if a '}' or ']' is missing.
* Eliminated the requirement that the "used" parameter had to point to a
variable. The current callers were always declaring a variable to meet
the requirement and discarding the value put into that variable. Now it
can be NULL.
* In ast_str_substitute_variables_full() fixed using the bogus channel to
evaluate a function. We were not using the bogus channel we just created
to help evaluate a subexpression.
Change-Id: Ia83d99f4f16abe47f329eb39b6ff2013ae7c9854
The type=identify endpoint identification method can match by IP address
and by SIP header. However, the SIP header matching has limited
usefulness because you cannot specify the SIP header matching priority
relative to the IP address matching. All the matching happens at the same
priority and the order of evaluating the identify sections is
indeterminate. e.g., If you had two type=identify sections where one
matches by IP address for endpoint alice and the other matches by SIP
header for endpoint bob then you couldn't predict which endpoint is
matched when a request comes in that matches both.
* Extract the SIP header matching criteria into its own "header" endpoint
identification method so the user can specify the relative priority of the
SIP header and the IP address matching criteria in the global
endpoint_identifier_order option. The "ip" endpoint identification method
now only matches by IP address.
ASTERISK-27491
Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095
* 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
ast_vector_string_split:
This function will add items to an ast_vector_string by splitting values
of a string buffer. Items are appended to the vector in the order they
are found.
ast_vector_const_string:
A vector of 'const char *'.
Change-Id: I1bf02a1efeb2baeea11c59c557d39dd1197494d7
This function returns NULL if the module in question is not running. I
did not change ast_module_ref as most callers do not check the result
and they always call ast_module_unref.
Make use of this function when running registered items from:
* app_stack API's
* bridge technologies
* CLI commands
* File formats
* Manager Actions
* RTP engines
* Sorcery Wizards
* Timing Interfaces
* Translators
* AGI Commands
* Fax Technologies
ASTERISK-20346 #close
Change-Id: Ia16fd28e188b2fc0b9d18b8a5d9cacc31df73fcc
Add a reference to the calling module when it is active to protect
access to datastore->info. Remove module references done by
func_periodic_hook as the datastore now handles it.
ASTERISK-25128 #close
Change-Id: I8357a3711e77591d0d1dd8ab4211a7eedd782c89
This is the old ASTOBJ macro's which are no longer used except by the
deprecated netsock.c. Move it to the chan_iax2 include folder so it
does not get used elsewhere.
Change-Id: I7e4ae96678b36b9f41d3cae14b167f110eb5d349
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
This moves netsock.c / netsock.h to the chan_iax2 module. netsock.h has
been marked deprecated since 13.0.0, chan_iax2 is the only remaining
user.
Change-Id: I28c6578043bac18de5ea608e136acec4f83d5dd3
ACO uses regex in many situations where it is completely unneeded. In
some cases this doubles the total processing performed by
aco_process_config.
* Create ACO_IGNORE category type for use in place of skip_category
regex source string.
* Create additional aco_category_op values to allow specifying category
filter using either a single plain string or a NULL terminated array
of plain strings.
* Create ACO_PREFIX to allow matching option names to case insensitive
prefixes.
Change-Id: I66a920dcd8e2b0301f73f968016440a985e72821
This is needed for future changes which will require being able to
process the load priority out of order.
Change-Id: Ia23421197f09789940510b03ebbbf3bf24d51bea
We should not do flood detection on video RTP streams. Video RTP streams
are very bursty by nature. They send out a burst of packets to update the
video frame then wait for the next video frame update. Really only audio
streams can be checked for flooding. The others are either bursty or
don't have a set rate.
* Added code to selectively disable packet flood detection for video RTP
streams.
ASTERISK-27440
Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70
This patch does three things associated with the initial incoming INVITE
request URI.
1) Add access to the full initial incoming INVITE request URI.
2) We were not setting DNID on incoming PJSIP channels. The DNID is the
user portion of the initial incoming INVITE Request-URI. The value is
accessed by reading CALLERID(dnid).
3) Fix CHANNEL(pjsip,target_uri) documentation.
* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).
* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.
* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.
* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.
ASTERISK-27478
Change-Id: I512e60d1f162395c946451becb37af3333337b33
A couple of places were setting the status to "UNKNOWN" when qualifies were
being disabled. Instead this should be set to the "CREATED" status that
represents when a contact is given (uri available), but the qualify frequency
is set to zero so we don't know the status.
This patch updates the relevant places with "CREATED". It also updates the
"CREATED" status description (value shown in CLI/AMI/ARI output) to a value
of "NonQualified"/"NonQual" as this description is hopefully less confusing.
ASTERISK-27467
Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89
Using the LIKE operator requires a full table scan of 'astdb', whereas a
comparison operation is able to use the primary key index.
This patch adds a new function to the AstDB API for quick prefix matches
and updates res_sorcery_astdb to utilize it. This showed substantial
performance improvement in my test environment.
Related to ASTERISK~26806, but does not completely resolve it.
Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1
There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.
Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value. This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object(). i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.
Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
Previous commits maintained compatibility with older remote console
clients as well as maintaining all API's.
Remove the following compatibility code:
* ast_cli_generatornummatches.
* Remote command "_command nummatches".
* Sorting / duplicate removal by remote console.
Change-Id: I59e6ce94fa57ae564888442049695f7e46746437
Some completion generators are very inefficent due to the way CLI
requests matches one at a time. ast_cli_completion_add can be called
multiple times during one invokation of a CLI generator to add all
results without having to reinitialize the search state for each match.
Change-Id: I73d26d270bbbe1e3e6390799cfc1b639e39cceec
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
This adds the printf attribute and changes 'fmt' from 'const void *' to
'const char *'. This resolves a warning from some compiler for
vsnprintf needing a literal string for format.
Change-Id: I71c33a8262590042ee451e1146760c10bb22fb78
Previously for PJSIP the local address of WebSocket connections
was set to the remote address. For logging purposes this is
not particularly useful.
The WebSocket API has been extended to allow the local
address to be queried and this is used in PJSIP to set the
local address to the correct value.
The PJSIP HEP support has also been tweaked so that reliable
transports always use the local address on the transport
and do not try to (wrongly) guess. As they are connection
based it is impossible for the source to be anything else.
ASTERISK-26758
ASTERISK-27363
Change-Id: Icd305fd038ad755e2682ab2786e381f6bf29e8ca
Some consumers of the sorcery API use ast_sorcery_retrieve_by_regex
only so that they can anchor the potential match as a prefix and not
because they truly need regular expressions.
Rather than using regular expressions for simple prefix lookups, add
a new operation - ast_sorcery_retrieve_by_prefix - that does them.
Change-Id: I56f4e20ba1154bd52281f995c27a429a854f6a79
This is a rewrite of ast_cli_completion_matches using a vector to build
the list. The original function calls the vector version, NULL
terminates the vector and extracts the elements array.
One change in behavior the results are now sorted and deduplicated. This
will solve bugs where some duplicate checking was done before the list
was sorted.
Change-Id: Iede20c5b4d965fa5ec71fda136ce9425eeb69519
* AST_VECTOR_STEAL_ELEMENTS - steal the array of elements for use
with non-vector code.
* struct ast_vector_string - a vector of 'char *'.
Change-Id: I104d1b204be03fccf67e02a195596adcb5ab1e42
The media frame cache gets in the way of finding use after free errors of
media frames. Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.
* Added the "cache_media_frames" option to asterisk.conf. Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the frame is
used after free and who freed it. NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.
To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.
Sample asterisk.conf setting:
[options]
cache_media_frames=no
ASTERISK-27413
Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00
This change causes the configure script to fail if the C compiler does
not support both function attributes constructor and destructor. These
were already required as modules cannot function without these attributes
and Asterisk requires modules.
This also has AST_GCC_ATTRIBUTE set a variable
ax_cv_have_func_attribute_$1. This is the same variable name used by
autoconf-archive's AX_GCC_FUNC_ATTRIBUTE, used for the same purpose.
Change-Id: Id68e8a1447f2a6d707c54b56350e7bfdb33fb663
This adds FD tracking for the following functions:
* eventfd
* timerfd_create
* socketpair
* accept
ASTERISK-27404
Change-Id: Id6848fe904ade2d34eb39d2a20bd6b223e1111fc
When (v)asprintf() fails, the state of the allocated buffer is undefined.
The library had better not leave an allocated buffer as a result or no one
will know to free it. The most likely way it can return failure is for an
allocation failure. If the printf conversion fails then you actually have
a threading problem which is much worse because another thread modified
the parameter values.
* Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL
on failure. That is much more useful than either an uninitialized pointer
or a pointer that has already been freed. Many uses won't have to check
for failure to ensure that the buffer won't be double freed or prevent an
attempt to free an uninitialized pointer.
* stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by
ast_asprintf().
* ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to
the wrong thing which is now not needed even if assigning to the right
thing.
Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23
This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.
Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.
ASTERISK-27395
Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4
This check is being added to make it easier for end-users of third party
open source Opus modules. This was removed by ASTERISK-26426 but only
the module needed to be removed.
Change-Id: I62b9cd0c4fa8a77596ab0e042948a643a1152677
Replace 'needsreload' argument with a 'type' argument to specify which
type of modules you want completion. This provides more accurate CLI
completion for load and unload commands.
* 'module unload' now excludes modules that have active references or are
not running.
* 'module load' now excludes modules that are already running.
* 'core set debug [atleast] <level> [module]' shows running modules only.
ASTERISK-27378
Change-Id: Iea3e00054461484196c46f688f02635cc886bad1
In WebRTC streams (or media tracks in their world) can be grouped
together using the mslabel. This informs the browser that each
should be synchronized with each other.
This change extends the stream API so this information can
be stored with streams. The PJSIP support has been extended
to use the mslabel to determine grouped streams and store
this association on the streams. Finally when creating the
SDP the group information is used to cause each media stream
to use the same mslabel.
ASTERISK-27379
Change-Id: Id6299aa031efe46254edbdc7973c534d54d641ad
When the identify_by option on an endpoint is set to ip it will
only be identified using the res_pjsip_endpoint_identifier_ip module.
This ensures that it is not mistakenly matched using the username of
the From header. To ensure behavior has not changed the default has
been changed to "username,ip" for the identify_by option.
ASTERISK-27206
Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd
The configure option to disable XML documentation does not currently
work. This patch makes it effective, but also causes an ABI change by
removing the ast_xmldoc_* symbols. Disabling xmldoc also prevents docs
from being automatically generated, but they can still be manually
generated with 'make doc/core-en_US.xml'.
ASTERISK-26639
Change-Id: Ifac562340c09f80c83e0203de098fcac93bf8c44
This makes the 'bt' parameter unconditional for ast_store_lock_info and
ast_remove_lock_info. The 'bt' parameter is unused when HAVE_BKTR is
undefined.
Change-Id: Ieced0e920928b735a39c3b5952b806c473d67453
Beanstalkd is a simple to use job queue. It provides a means to
create multiple job queues called "tubes". Each tube can store
multiple jobs, with varying priorities with the queue. Queue
processing is available via a simple TCP socket or via well defined
libraries, avaialble at
https://github.com/kr/beanstalkd/wiki/client-libraries
This module is based upon the beanstalk-client library, available
for download at: https://github.com/deepfryed/beanstalk-client
Change-Id: I5fe4089a34ab3b39230786d9bbfddafa56715f48
Add bridge_features structure to bridge creation. Specifically, this
implements mute and DTMF suppression, but others should be able to be
easily added to the same structure.
ASTERISK-27322 #close
Reported by: Darren Sessions
Sponsored by: AVOXI
Change-Id: Id4002adfb65c9a8027ee9e1a5f477e0f01cf9d61
* Store weak proxy objects in instances container.
* Remove special unreference function and replace with macro that calls
ao2_cleanup.
* Add REF_DEBUG information to ast_sorcery_open.
Change-Id: I5a150a4e13cee319d46b5a4654f95a4623a978f8
This function finds a weak proxy in an ao2_container and returns the
real object associated with it.
Change-Id: I9da822049747275f5961b5c0a7f14e87157d65d8
Copy the list of weakproxy callbacks to temporary memory so they can be
run without holding the weakproxy lock.
Change-Id: Ib167622a8a0f873fd73938f7611b2a5914308047
When "rewrite_contact" is enabled, the "max_contacts" count option can
block re-registrations because the source port from the endpoint can be
random. When the re-registration is blocked, the endpoint may give up
re-registering and require manual intervention.
* The "remove_existing" option now allows a registration to succeed by
displacing any existing contacts that now exceed the "max_contacts" count.
Any removed contacts are the next to expire. The behaviour change is
beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
than one. The removed contact is likely the old contact created by
"rewrite_contact" that the device is refreshing.
ASTERISK-27192
Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b
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
This provides better information to REF_DEBUG log for troubleshooting
when the system is unable to unload res_pjsip.so during shutdown due to
module references.
ASTERISK-27306
Change-Id: I63197ad33d1aebe60d12e0a6561718bdc54e4612
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
Somewhere along the way we lost the ability to debug individual
source files. For modules, this wasn't a big deal but all the
source files in ./main are in the one "core" module so debugging
individual core capabilities was almost impossible.
* Added a test to DEBUG_ATLEAST that also checks __FILE__ instead
of just module name. Any source file will work even if it's in
a module subdirectory.
Change-Id: Icc0af41837f3b1679dec7af21fa32cd1f7469f6e
Some endpoints do not like a stream being reused for a new
media stream. The frame/jitterbuffer can rely on underlying
attributes of the media stream in order to order the packets.
When a new stream takes its place without any notice the
buffer can get confused and the media ends up getting dropped.
This change uses the SSRC change to determine that a new source
is reusing an existing stream and then bridge_softmix renegotiates
each participant such that they see a new media stream. This
causes the frame/jitterbuffer to start fresh and work as expected.
ASTERISK-27277
Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07
When two channels were early bridged in a native_rtp bridge, the RTP description
on one side was not updated when the other side answered.
This patch forbids non-answered channels to enter a native_rtp bridge, and
triggers a bridge reconfiguration when an ANSWER frame is received.
ASTERISK-27257
Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df
A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to
receive unsolicited MWI NOTIFY requests and make them available to
other modules via the stasis message bus.
res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request"
that parses a simple-message-summary body and, if
endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state
with the voice-message counts from the message.
Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c
If an error occurs during a bridge impart it's possible that
the "bridge_after" callback might try to run before
control_swap_channel_in_bridge has been signalled to continue.
Since control_swap_channel_in_bridge is holding the control lock
and the callback needs it, a deadlock will occur.
* control_swap_channel_in_bridge now only holds the control
lock while it's actually modifying the control structure and
releases it while the bridge impart is running.
* bridge_after_cb is now tolerant of impart failures.
Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3
In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was
confusion about whether the transport_state->localnet ACL has ALLOW or
DENY semantics.
For the record: the localnet has DENY semantics, meaning that "not in
the list" means ALLOW, and the local nets are in the list.
Therefore, checks like this look wrong, but are right:
/* See if where we are sending this request is local or not, and if
not that we can get a Contact URI to modify */
if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
ast_debug(5, "Request is being sent to local address, "
"skipping NAT manipulation\n");
(In the list == localnet == DENY == skip NAT manipulation.)
And conversely, other checks that looked right, were wrong.
This change adds two macro's to reduce the confusion and uses those
instead:
ast_sip_transport_is_nonlocal(transport_state, addr)
ast_sip_transport_is_local(transport_state, addr)
ASTERISK-27248 #close
Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934
An admin can configure app_minivm with an externnotify program to be run
when a voicemail is received. The app_minivm application MinivmNotify
uses ast_safe_system() for this purpose which is vulnerable to command
injection since the Caller-ID name and number values given to externnotify
can come from an external untrusted source.
* Add ast_safe_execvp() function. This gives modules the ability to run
external commands with greater safety compared to ast_safe_system().
Specifically when some parameters are filled by untrusted sources the new
function does not allow malicious input to break argument encoding. This
may be of particular concern where CALLERID(name) or CALLERID(num) may be
used as a parameter to a script run by ast_safe_system() which could
potentially allow arbitrary command execution.
* Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
instead of ast_safe_system() to avoid command injection.
* Document code injection potential from untrusted data sources for other
shell commands that are under user control.
ASTERISK-27103
Change-Id: I7552472247a84cde24e1358aaf64af160107aef1
If an SDP answer hasn't been sent yet, it's legal to change it.
This is required for PJSIP_DTMF_MODE to work correctly, and can
also have use in the future for updating codecs too.
ASTERISK-27209 #close
Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1
* Fix deadlock in
bridge_softmix.c:softmix_bridge_stream_topology_changed() between
bridge_channel and channel locks.
* The new bridge technology topology change callbacks must be called with
the bridge locked. The callback references the bridge channel list, the
bridge technology could change, and the bridge stream mapping is updated.
ASTERISK-27212
Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be
* 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
The fix for the issue is broken up into three parts.
This is part two which handles the server side of REGISTER requests when
rewrite_contact is enabled. Any registered reliable transport contact
becomes invalid when the transport connection becomes disconnected.
* Monitor the rewrite_contact's reliable transport REGISTER contact for
shutdown. If it is shutdown then the contact must be removed because it
is no longer valid. Otherwise, when the client attempts to re-REGISTER it
may be blocked because the invalid contact is there. Also if we try to
send a call to the endpoint using the invalid contact then the endpoint is
not likely to see the request. The endpoint either won't be listening on
that port for new connections or a NAT/firewall will block it.
* Prune any rewrite_contact's registered reliable transport contacts on
boot. The reliable transport no longer exists so the contact is invalid.
* Websockets always rewrite the REGISTER contact address and the transport
needs to be monitored for shutdown.
* Made the websocket transport set a unique name since that is what we use
as the ao2 container key. Otherwise, we would not know which transport we
find when one of them shuts down. The names are also used for PJPROJECT
debug logging.
* Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state
event. Now the global keep_alive_interval option, initially idle shutdown
timer, and the server REGISTER contact monitor can work on wetsocket
transports.
* Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction.
Now initially idle websockets will automatically shutdown.
ASTERISK-27147
Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4
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
Introduce a new property to rtp-engine to make it aware of
the desire for assymetric codecs or not. If asymmetric codecs
is not allowed, the bridge will compare read/write formats
and shut down the p2p bridge if needed
ASTERISK-26745 #close
Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f
Currently, the handling of the msid attribute is not quite right. According to
the spec the msid's between the offer/answer are not dependent upon one another.
Meaning the same msid's given in an offer do not have to be returned in the
answer for a given stream. And they probably shouldn't be (copied/reused) since
this can potentially cause some browser side confusion.
This patch generates new msids when both an offer and answer are sent from
Asterisk. However, Asterisk does reuse the original msid it sent out for a
reinvite. Also audio+video streams are paired together by sharing the same
stream id, but a different track id.
ASTERISK-27179 #close
Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643
The "external_media_address" option on transports is now
resolved using dnsmgr. This allows it to be automatically
refreshed regularly if refreshes are enabled in dnsmgr.
If the system is using a dynamic IP address a dynamic DNS
hostname can be provided to keep the IP address up to
date.
Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2
This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis
ASTERISK-27085 #close
Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
AMI goes from 3.2.0 to 4.0.0
ARI goes from 2.0.0 to 3.0.0
Copied UPGRADE.txt -> UPGRADE-15.txt
Created new UPGRADE.txt
Removed a log file that was accidentally checked in a while ago
Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7
This change does a few things to improve packet loss and renegotiation:
1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.
2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.
3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.
4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.
5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.
ASTERISK-27143
Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45
This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:
rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.
Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.
ASTERISK-27119 #close
Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd
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
BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.
This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.
For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.
ASTERISK-27118
Change-Id: I96c0920b9f9aca7382256484765a239017973c11
This adds a parameter to ast_waitfordigit_full which can be used to only
stop waiting when certain expected digits are received. Any unexpected
DTMF digits are simply ignored.
This also creates a new dialplan application WaitDigit.
ASTERISK-27129 #close
Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9
This change fixes a few things uncovered during SFU testing.
1. Unreal channels incorrectly forwarded video frames when
no video stream was present on them. This caused a crash when
they were read as the core requires a stream to exist for the
underlying media type. The Unreal channel will now ensure a
stream exists for the media type before forwarding the frame
and if no stream exists then the frame is dropped.
2. Mapping of frames during bridging from the stream number of
the underlying channel to the stream number of the bridge was
done in the wrong location. This resulted in the frame getting
dropped. This mapping now occurs on reading of the frame from
the channel.
3. Bridging was using the wrong ast_read function resulting in
it living in a non-multistream world.
4. In bridge_softmix when adding new streams to existing channels
the wrong stream topology was copied resulting in no streams
being added.
Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8
This API was not actively maintained, was not added to new modules
(such as res_pjsip), and there exist better alternatives to acquire the
same information, such as the ARI.
Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83
The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated. This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.
ASTERISK-27066 #close
Change-Id: Id185b11e84afd9191a2f269e8443019047765e91
The stream topology (list of streams and order) is now stored with the
configured PJSIP endpoints and used during the negotiation process.
Media negotiation state information has been changed to be stored
in a separate object. Two of these objects exist at any one time
on a session. The active media state information is what was previously
negotiated and the pending media state information is what the
media state will become if negotiation succeeds. Streams and other
state information is stored in this object using the index (or
position) of each individual stream for easy lookup.
The ability for a media type handler to specify a callback for
writing has been added as well as the ability to add file
descriptors with a callback which is invoked when data is available
to be read on them. This allows media logic to live outside of
the chan_pjsip module.
Direct media has been changed so that only the first audio and
video stream are directly connected. In the future once the RTP
engine glue API has been updated to know about streams each individual
stream can be directly connected as appropriate.
Media negotiation itself will currently answer all the provided streams
on an offer within configured limits and on an offer will use the
topology created as a result of the disallow/allow codec lines.
If a stream has been removed or declined we will now mark it as such
within the resulting SDP.
Applications can now also request that the stream topology change.
If we are told to do so we will limit any provided formats to the ones
configured on the endpoint and send a re-invite with the new topology.
Two new configuration options have also been added to PJSIP endpoints:
max_audio_streams: determines the maximum number of audio streams to
offer/accept from an endpoint. Defaults to 1.
max_video_streams: determines the maximum number of video streams to
offer/accept from an endpoint. Defaults to 1.
ASTERISK-27076
Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7
There have been reports of deadlocks caused by an attempt to send a frame
to a channel's rtp instance after the channel has left the native bridge
and been destroyed. This patch effectively causes the bridge channel to
keep a reference to the glue and both the audio and video rtp instances
so what gets started will get stopped.
ASTERISK-26978 #close
Reported-by: Ross Beer
Change-Id: I9e1ac49fa4af68d64826ccccd152593cf8cdb21a
In an earlier version of Asterisk a local channel [un]lock all functions were
added in order to keep a crash from occurring when a channel hung up too early
during an attended transfer. Unfortunately, when a transfer failure occurs and
depending on the timing, the local channels sometime do not get properly
unlocked and deref'ed after being locked and ref'ed. This happens because the
underlying local channel structure gets NULLed out before unlocking.
This patch reworks those [un]lock functions and makes sure the values that get
locked and ref'ed later get unlocked and deref'ed.
ASTERISK-27074 #close
Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09
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
This change adds support for socket activation of certain SOCK_STREAM
listeners in Asterisk:
* AMI / AMI over TLS
* CLI
* HTTP / HTTPS
Example systemd units are provided. This support extends to any socket
which is initialized using ast_tcptls_server_start, so any unknown
modules using this function will support socket activation.
Asterisk continues to function as normal if socket activation is not
enabled or if systemd development headers are not available during
build.
ASTERISK-27063 #close
Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d
* 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.
This option was added to control whether to notify dialog-info state
'early' or 'confirmed' on Ringing when already INUSE.
The value "yes" is useful for some SIP phones (Cisco SPA)
to be able to indicate and pick up ringing devices.
ASTERISK-26919 #close
Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711
The recent change to make the use of LOAD_DECLINE more consistent
caused res_ari to unload itself before declining if the ari.conf
file wasn't found. The ari stubs though still tried to use the
configuration resulting in segfaults.
This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests
to see if res_ari is actually loaded and causes the stubs to also
decline if it isn't. The macro was then added to the mustache
template's "load_module" function.
ASTERISK-27026 #close
Reported-by: Ronald Raikes
Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d
* 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
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
When doing an attended transfer it's possible for the transferer, after
receiving an accepted response from Asterisk, to send a BYE to Asterisk,
which can then be processed before Asterisk has time to start and/or
complete the transfer process. This of course causes the transfer to not
complete successfully, thus dropping the call.
This patch makes it so any BYEs received from the transferer, after the REFER,
that initiate a session end are deferred until the transfer is complete. This
allows the channel that would have otherwise been hung up by Asterisk to
remain available throughout the transfer process.
ASTERISK-27053 #close
Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a
This introduces the ability for PJSIP code to specify filtering flags
when retrieving PJSIP contacts. The first flag for use causes the
query code to only retrieve contacts that are not unreachable. This
change has been leveraged by both the Dial() process and the
PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt
calls to contacts which are not unreachable.
ASTERISK-26281
Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c
In review 4843 (ASTERISK-24858), we added a hack that forced a smoother
creation when sending signed linear so that the byte order was adjusted
during transmission. This was needed because smoother flags were lost
during the new format work that was done in Asterisk 13.
Rather than rolling that same hack into res_rtp_multicast, re-introduce
smoother flags so that formats can dictate their own options.
Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16
A previous commit added plumbing to bridge_softmix to allow for an SFU
experience with Asterisk. This commit adds an option to app_confbridge
that allows for a confbridge to actually make use of the SFU video mode.
SFU mode is implemented in a "set it and forget it" kind of way. That
is, when the bridge is created, if SFU mode is enabled, then the video
mode gets set to SFU and cannot be changed. Future improvements may
allow for a hybrid experience (e.g. forward multiple video streams,
specifically those of the most recent talkers), but for this addition,
no such capability is present.
Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020
This sets up the "plumbing" in bridge_softmix to
be able to accommodate Asterisk asking as an SFU
(selective forwarding unit) for conferences.
The way this works is that whenever a channel enters or leaves a
conference, all participants in the bridge get sent a stream topology
change request. The topologies consist of the channels' original
topology, along with video destination streams corresponding to each
participants' source video streams. So for instance, if Alice, Bob, and
Carol are in the conference, and each supplies one video stream, then
the topologies for each would look like so:
Alice:
Audio,
Source video(Alice),
Destination Video(Bob),
Destination video (Carol)
Bob:
Audio,
Source video(Bob)
Destination Video(Alice),
Destination video (Carol)
Carol:
Audio,
Source video(Carol)
Destination Video(Alice),
Destination video (Bob)
This way, video that arrives from a source video stream can then be
copied out to the destination video streams on the other participants'
channels.
Once the bridge gets told that a topology on a channel has changed, the
bridge constructs a map in order to get the video frames routed to the
proper destination streams. This is done using the bridge channel's
stream_map.
This change is bare-bones with regards to SFU support. Some key features
are missing at this point:
* Stream limits. This commit makes no effort to limit the number of
streams on a specific channel. This means that if there were 50 video
callers in a conference, bridge_softmix will happily send out topology
change requests to every channel in the bridge, requesting 50+
streams.
* Configuration. The plumbing has been added to bridge_softmix, but
there has been nothing added as of yet to app_confbridge to enable SFU
video mode.
* Testing. Some functions included here have unit tests.
However, the functionality as a whole has only been verified by
hand-tracing the code.
* Selectivenss. For a "selective" forwarding unit, this does not
currently have any means of being selective.
* Features. Presumably, someone might wish to only receive video from
specific sources. There are no external-facing functions at the moment
that allow for users to select who they receive video from.
* Efficiency. The current scheme treats all video streams as being
unidirectional. We could be re-using a source video stream as a
desetnation, too. But to simplify things on this first round, I did it
this way.
Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d
...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
When manipulating flags on a channel the channel has to be
locked to guarantee that nothing else is also manipulating
the flags. This change introduces locking where necessary to
guarantee this. It also adds helper functions that manipulate
channel flags and lock to reduce repeated code.
ASTERISK-26789
Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10
* 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.
This option was added to turn off notifying the progress details
on Blind Transfer. If this option is not set then the chan_pjsip
will send NOTIFY "200 OK" immediately after "202 Accepted".
Some SIP phones like Mitel/Aastra or Snom keep the line busy until
receive "200 OK".
ASTERISK-26333 #close
Change-Id: Id606fbff2e02e967c02138457badc399144720f2
When we optionally set the interface_address we are forcing the media to
go out a specific interface address. This allows us to optionally have
the media go out the interface that SIP signalling came in on or if we are
configured to have the media always go out a specific address.
Change-Id: I160d9fac322a075bd2557b430632544178196189
* 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
ast_stream_clone() cannot copy the opaque user data stored on a stream.
We don't know how to clone the data so it isn't copied into the clone.
Change-Id: Ia51321bf38ecbfdcc53787ca77ea5fd2cabdf367
When using the Bridge AMI action on the same channel multiple times
it was possible for the channel to return to the wrong location in
the dialplan if the other party hung up. This happened because the
priority of the channel was not preserved across each action
invocation and it would fail to move on to the next priority in
other cases.
This change makes it so that the priority of a channel is preserved
when taking control of it from another thread and it is incremented
as appropriate such that the priority reflects where the channel
should next be executed in the dialplan, not where it may or may not
currently be.
The Bridge AMI action was also changed to ensure that it too
starts the channels at the next location in the dialplan.
ASTERISK-24529
Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a
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
The telephone_event option was used as a flag and a bit mapped value in
different places when it is a boolean. It is also inadequate to configure
the DTMF operation of the RTP instance created for the stream.
Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b
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
If you use ast_request to create a PJSIP channel but then hang it
up without causing a transaction to be sent, the session will
never be destroyed. This is due ot the fact that it's pjproject
that triggers the session cleanup when the transaction ends.
app_chanisavail was doing this to get more granular channel state
and it's also possible for this to happen via ARI.
* ast_sip_session_terminate was modified to explicitly call the
cleanup tasks and unreference session if the invite state is NULL
AND invite_tsx is NULL (meaning we never sent a transaction).
* chan_pjsip/hangup was modified to bump session before it calls
ast_sip_session_terminate to insure that session stays valid
while it does its own cleanup.
* Added test events to session_destructor for a future testsuite
test.
ASTERISK-26908 #close
Reported-by: Richard Mudgett
Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9
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
The primary win of switching to eventfd when possible is that it only
uses a single file descriptor while pipe() will use two. This means for
each bridge channel we're reducing the number of required file
descriptors by 1, and - if you're using timerfd - we also now have 1
less file descriptor per Asterisk channel.
The API is not ideal (passing int arrays), but this is the cleanest
approach I could come up with to maintain API/ABI.
I've also removed what I believe to be an erroneous code block that
checked the non-blocking flag on the pipe ends for each read. If the
file descriptor is 'losing' its non-blocking mode, it is because of a
bug somewhere else in our code.
In my testing I haven't seen any measurable difference in performance.
Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d
This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compact_headers=yes via the file pjsip.conf.
ASTERISK-26932 #close
Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689
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
On 2's compliment machines abs(INT_MIN) behavior is undefined and
results in a negative value still being returnd. This results in
negative hash codes that can result in crashes.
ASTERISK-26528 #close
Change-Id: Idff550145ca2133792a61a2e212b4a3e82c6517b
0035-r5572-svn-backport-dialog-transaction-deadlock.patch
0036-r5573-svn-backport-ua-pjsua-transaction-deadlock.patch
0037-r5576-svn-backport-session-timer-crash.patch
Also removed the progress bar from wget download to stdout.
ASTERISK-26905 #close
Reported-by: Ross Beer
Change-Id: I268fb3cf71a3bb24283ff0d24bd8b03239d81256
* Report failures if configure finds a required header is missing.
* Deduplicate includes between asterisk.h, astmm.h and compat.h.
* Unconditionally include headers in compat.h if required elsewhere.
Change-Id: Ie67d0185ca71fbfb81c9bdfaebe46a49e3c56dc5
The ast_json structure is used in many Asterisk headers and is often the
only part of json.h used. This adds a forward declaration to asterisk.h
and removes the include of json.h from many headers. The declaration
has been left in endpoints.h and stasis.h to avoid problems with source
files that use ast_json functions without directly including json.h.
ari.h continues to include json.h as it uses enum
ast_json_encoding_format.
Change-Id: Id766aabce6bed56626d27e8d29f559b5e687b769
This change cleans up state management for media streams by moving
RTP instances into their own session structure and adding additional
details that are not relevant to the core (such as connection address).
These can live either in the local capabilities or joint capabilities.
The ability to set explicit connection address information for
the purposes of direct media and NAT has also been added at the
global and stream specific level.
ASTERISK-26900
Change-Id: If7e5307239a9534420732de11c451a2705b6b681
The ao2_global_obj_release() function holds an exclusive lock on the
global object while it is being dereferenced. Any destructors that
run during this time that call ao2_global_obj_ref() will deadlock
because a read lock is required.
Instead, we make the global object inaccessible inside of the write
lock and only dereference it once we have released the lock. This
allows the affected destructors to fail gracefully.
While this doesn't completely solve the referenced issue (the error
message about not being able to create an IQ continues to be shown)
it does solve the backtrace spew that accompanied it.
ASTERISK-21009 #close
Reported by: Marcello Ceschia
Change-Id: Idf40ae136b5070dba22cb576ea8414fbc9939385
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
Updated the AMI version for the following reason (see CHANGES for more details):
The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now
contains a new optional parameter, 'MatchHeader'.
Change-Id: Ie206913ef1dcfa6a2ebe3282da2387e52d6f05b9
Jenkins2
Joshua Colp
Sean Bright
Gaurav Khurana
George Joseph
Richard Mudgett
Alexander Traud
Ben Ford
Kevin Harwell
Corey Farrell
Nir Simionovich
Thomas Sevestre
Seán C McCord
Jean Aunis
Walter Doekes
Torrey Searle
Matthew Fredrickson
Mark Michelson
Alexei Gradinari
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