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

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

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

This patch adds support to send in-dialog SIP NOTIFY commands on
chan_pjsip channels, similar to the functionality recently added
for chan_sip (ASTERISK_27461).

This extends res_pjsip_notify to allow for in-dialog messages.

ASTERISK-27697

Change-Id: If7f3151a6d633e414d5dc319d5efc1443c43dd29

* 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

This change allows chan_pjsip to be given an AST_FRAME_RTCP
containing REMB feedback and pass it to res_rtp_asterisk.
Once res_rtp_asterisk receives the frame a REMB RTCP feedback
packet is constructed with the appropriate contents and sent
to the remote endpoint.

ASTERISK-27776

Change-Id: Ic53f821c1560d8924907ad82c4d9c0bc322b38cd

The minimum block word length is actually 4, not 5.

Change-Id: I878542218225aed72c72bdf1b856fc822cd2d649

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

Given the below call scenario:
A -> Ast1 -> B
C <- Ast2 <- B

1) A calls B through Ast1
2) B calls C through Ast2
3) B transfers A to C

When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to
send an INVITE with replaces to Ast2.  Ast2 then leaks a channel ref of
the channel between Ast1 and Ast2.

Channel ref leaks are easily seen in the CLI "core show channels" output.
The leaked channels appear in the output but you can do nothing with them
and they never go away unless you restart Asterisk.

* Properly account for the channel refs when imparting a channel into a
bridge when handling an INVITE with replaces in handle_invite_replaces().
The ast_bridge_impart() function steals a channel ref but the code didn't
account for how many refs were held by the code at the time and which ref
was stolen.

* Eliminated RAII_VAR in handle_invite_replaces().

ASTERISK-27740

Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4

Change-Id: I3811de0014b1ffe96d4a3b49cddd5d4ca02ee5d4

Change-Id: Ib427ffc2c802620eaafb08b1c2a17dddd8fb8eb6

Asterisk requires GNU C extensions.  On some systems certain libraries
may incorrectly push -std=c99 into CFLAGS, thus breaking the build.
This change causes that flag to be stripped so the Asterisk build is not
broken by those libraries.  This change is made for both pkgconfig and
tool based libraries.

ASTERISK-27629 #close

Change-Id: I13389613b194abbac77becf90cd950dc168704db

* Replace all 'else if' statements with 'elif'.
* Use loop to detect versioned lua headers and libraries.

The loop for detecting lua fixes a bug where LUA_INCLUDE would be
appended with the directory of every lua version after the first one is
found.

Change-Id: I3276f9aee955014108345be6092f51c932b43a0f

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