Added unit tests for RTCP video stats. These tests include NACK, REMB,
FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR
tests are currently disabled due to a bug. We expect to receive a
compound packet, but the code sends this out as a single packet, which
the browser accepts, but makes Asterisk upset.

While writing these tests, I noticed an issue with NACK as well. Where
it is handling a received NACK request, it was reading in only the first
8 bits of following packets that were also lost. This has been changed
to the correct value of 16 bits.

Also made a minor fix to the data buffer unit test.

Change-Id: I56107c7411003a247589bbb6086d25c54719901b

Support has been added for receiving a NACK request and handling it.
Now, Asterisk can detect when a NACK request should be sent and knows
how to construct one based on the packets we've received from the remote
end. A buffer has been added that will store out of order packets until
we receive the packet we are expecting. Then, these packets are handled
like normal and frames are queued to the core like normal. Asterisk
knows which packets to request in the NACK request using a vector
which stores the sequence numbers of the packets we are currently missing.

If a missing packet is received, cycle through the buffer until we reach
another packet we have not received yet. If the buffer reaches a certain
size, send a NACK request. If the buffer reaches its max size, queue all
frames to the core and wipe the buffer and vector.

According to RFC3711, the NACK request must be sent out in a compound
packet. All compound packets must start with a sender or receiver
report, so some work was done to refactor the current sender / receiver
code to allow it to be used without having to also include sdes
information and automatically send the report.

Also added additional functionality to ast_data_buffer, along with some
testing.

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

ASTERISK-27810 #close

Change-Id: Idab644b08a1593659c92cda64132ccc203fe991d

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