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

Change-Id: I3811de0014b1ffe96d4a3b49cddd5d4ca02ee5d4

Change-Id: Ib427ffc2c802620eaafb08b1c2a17dddd8fb8eb6

* 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

Fix some timer heap initializations and cancels to try and prevent
crashes and timer heap issues.

Change-Id: I64885d190fa22097d1b55987091375541e57a7ee

There have been cases that when the transaction timer callback is called
the tsx is already destroyed.  This causes a crash.  We now check the
tsx state and return if the tsx is already destroyed.

Change-Id: If93acd5e48d9ca5bb553f2405d5afc836842fe1c

Added a new pj_timer_entry_reset function that resets a timer_entry
for re-use.

Changed direct settings of timer_entry fields to use
pj_timer_entry_init and pj_timer_entry_reset.

Fixed issues where timers were being rescheduled incorrectly.

Change-Id: I5b624bfbc5c1429117484b9b24567293002148e6

Apparently it is possible for the transport to be destroyed without
triggering the transport callback logic.  As a result the transport gets
destroyed and we have a stale pointer in the active_transports container.

* Invoke the transport monitor callback checks when the transport is
destroyed in addition to when it is disconnected and shutdown.

ASTERISK-27688

Change-Id: Ia9b5469fea8f2b3f2d8476fae6b748a4d23e7261

Added unit tests for the data buffer API. These tests include creating a
data buffer, putting payloads into the buffer, resizing the buffer, and
the nominal case for data buffer usage, which consists of adding
the max number of payloads to the buffer, checking to see if the correct
payloads are present, then adding more payloads and checking again to
see if the previous payloads were replaced or not.

For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements

Change-Id: Id5b599aa15a5e61d0ec080f97cd0c57bd07e6f8f

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

There have been some crashes in the past where something attempts
to use a pj_atomic after it's already been destroyed.  This patch
tries to prevent it by making sure that pj_atomic_destroy sets
its mutex to NULL when it's done.  The pj_mutex functions already check
for a NULL mutex and just return PJ_EINVAL.

Teluu also added some checks to the win32 implementation as well.

Change-Id: Id25f70b79fdedf44ead6e6e1763a4417d3b3f825

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

Change-Id: I2d80bc5edf940fab914cba3d8a0fa0b5eb2a3148

Setting optind = 0 is forced to 1 in glibc implementation, but
causes option parsing to be flawed in other implementations, for
example on FreeBSD.

ASTERISK-27773 #close

Change-Id: Ia548e69f8302e9754dbbedb6bc451c0700c66f61