If during translation a codec could not handle a given frame the translation
core would return NULL, thus not passing along the "missing" frame. Due to this
there was no frame to apply generic plc to, thus rendering it useless.
This patch makes it so the translation core produces an interpolated slin frame
in the cases where an attempt was made to translate to slin, but failed. This
interpolated frame is then passed along and can be used by the generic plc
algorithms to fill in the frame.
ASTERISK-27814 #close
Change-Id: I133d084da87adef913bf2ecc9c9240e3eaf4f40a
When an externally initiated renegotiation occurred it was
possible for video streams to be incorrectly remapped,
resulting in no video flowing to some receivers.
This change ensures that only the video source sets up
mappings and also that removed streams do not have mappings
set up.
Change-Id: Iab05f2254df3606670774844bb0935f833d3a9b0
Call ooManualProgress/Ringback outside of ast_debug function
when ooh323 debugging is on
ASTERISK-27812 #close
ASTERISK-26893 #close
Reported by: Dimos, Marco Giordani
Change-Id: I5873762e4f05824e7b6e94a19dd4eb56adbbbb79
This change fixes a bug where a REMB collector may be
freed twice, and also tweaks REMB combining such that if
there is no bitrate from anyone (or there are no sources)
we report 0 instead of using an old bitrate.
ASTERISK-27804
Change-Id: Ia9dc9c150043890ee7ff85e9cdec007f1a77fcfd
In DragonFly BSD, added libraries from ports are placed into /usr/local.
Therefore, this directory must be added for the preprocessor, compiler, and
linker.
Beside that, the script ./configure was updated:
* OSARCH list was outdated and not used, removed.
* AC_CANONICAL_BUILD was not used.
* _REENTRANT, this feature test macro is obsolete.
ASTERISK-27820
Change-Id: I186d88d99cfa4de6569888e12ac97bd2f441c422
Fixes a bug on the "confbridge show profile bridge" cli command
that showed "video_mode=no video" when video_mode was set
to "sfu"
ASTERISK-27418 #close
Change-Id: I481e3172c7f872664c7ac7809879d541c9f031e9
Redirect libc allocation functions to use Asterisk functions for
main/ast_expr2f.c and res/ael/ael_lex.c. This will resolve errors
produced by astmm.h when these files are regenerated, though other
issues still remain.
ASTERISK~27813
Change-Id: I7263e9e4217a17bde4ffaa2087a8f8aeb2a8588c
There are twice as many samples in the same number of bytes, so redefine
some of the G.722 format functions in terms of their PCM counterparts.
Change-Id: I6a8c7352624b930a5f2d9e4857f75283fa5dd9f9
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
SendText now accepts new channel variables that can be used
to override the To and From display names and set the Content-Type
of a message. Since you can now set Content-Type, other text/*
content types are now valid.
Change-Id: I648b4574478119f95de09d9f08e9595831b02830
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
clang 6.0 warned about this. Beside that, this change removes the used variable
'desc'.
ASTERISK-27808
Change-Id: Ia26bdcc0a562c058151814511cfcf70ecafa595b
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
This reverts a problem introduced by the fix for ASTERISK_24329.
Now, when an announcement is played while waiting in a queue, music on
hold will not restart from the beginning of the sound file and will
instead pick up where it left off. However, the incorrect behavior in
ASTERISK_24329 is now present again; if an announcement X seconds
long is played when music on hold starts, music on hold will start X
seconds into the file.
ASTERISK-27774 #close
Reported by: lvl
Change-Id: I86b2885ee7063268f9b9747eddb788336ade989b
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
* A side benefit is that the scheduled tasks are not completely blocked
while the CLI command executes.
* Adjusted the "Task Name" column width to have more room for longer
names.
Change-Id: Iec64aa463ee8b10eef90120e00c38b1fb444087e
If it is not defined, it will add MYSQL_PORT definition. After some
research on MySQL/MariaDB development tree, I couldn't find any reference
to MYSQL_PORT definition in include files.
ASTERISK-27782 #close
Change-Id: Ieee56c836fc2e8bd021c456145bba04c6068bb77
It now appends the external IP address on the
o= line of the SDP packet. The decision was made to write
the numeric IP address as opposed to the RFC that states
the FQDN should be used if and when available. We believe
the usage of literal IP address will help avoid
potential problems.
ASTERISK-27614 #close
Change-Id: I84f3360f3606b8c4e8d161edb228799ec0b8a302
