Change-Id: Ib3111b151d37cbda40768cf2a8a9c6cf6c5c7cbd

Copied UPGRADE.txt -> UPGRADE-16.txt
Created new UPGRADE.txt

Updated ARI stubs version to 17.

Change-Id: I4210e53f8022a2a68c7653595bdd13fbebac41ee

ARI goes from 3.0.0 to 4.0.0

Change-Id: I0649fa34926dc4fc89a166f1d2e3bbd965ef9ebe

Previously, Asterisk did not tell its bundled PJProject about this configure
parameter. Therefore, PJProject used the platform provided OpenSSL always.

ASTERISK-27880

Change-Id: Iea545aec854dd0e2c061c69bb118a76ce56c5dc6

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

A problem I've seen countless times is a global or system section
for PJSIP not getting applied. This is inevitably the result of
the "type=" line missing. This change alleviates that problem.

The ability to specify an explicit section name has been
added to res_sorcery_config. If the configured section
name matches this and there are no unknown things configured
the section is taken as being for the given type.

Both the PJSIP "global" and "system" types now support this
so you can just name your section "global" or "system" and it
will be matched and used, even without a "type=" line.

ASTERISK-27972

Change-Id: Ie22723663c1ddd24f869af8c9b4c1b59e2476893

I have removed the STATIC_BUILD option immediately as it has not
been maintained in many years and is non-functional.

ASTERISK-27965

Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7

Fixes issue where error msg
"Use of before/init after destruction"
was being printed on disabled messages
in dev mode.  With this
fix if message is disabled
a warning will print.

ASTERISK-25548
Change-Id: Ie0d866d1cbc60c16dbef08bc65e99505c3c1adfa

SRTP SDES key lifetime support was added in ASTERISK_17899.

In that addition, the minimum key lifetime to be accepted was
set at the 10 hours @ 20ms/packet = 1800000 packets.

The firmware in the obi1xx ATA uses a hardcoded lifetime of
2^20 packets.

Lower the limit to 2^20 to support a wider field of clients.

ASTERISK-27967 #close

Change-Id: I81a0703c595a0c9101dfdf02300149a3cc39bf94

Change-Id: I5e3fb6adfa6cbf694c0deecf02e3879297b0c12e

Change-Id: Icf2e67818b2155a158d2390b138613e1f653ea92

Asterisk patched the pjproject source to avoid crashing when pjproject
sip_msg headers are encountered with NULL vptr's, but the patch also
output error messages for some valid headers which simply did not need
to be added to the message itself, such as hidden route headers.

pjproject has since applied a similar patch to their baseline to avoid
crashes, but their version also avoids the spurious error logging.

Lets use their patch instead.

ASTERISK-27961 #close

Change-Id: I2ddbd82c8da10e0dcc9807a48089d1f3c2d6e389

Change-Id: I0ff1730ef4a4f0ac9f18ccc9bc0dfe7a782f57a8

Change-Id: Ibc151f63dcec4db847915c2f3cbe5b467dd59574

Change-Id: I56399aa384468f45494c2c3650420563a0b6efe1

Change-Id: I2fe55c38607eaec2fbf69ef23a5019e0c443a64b

* Merge the preload and load stages, use load ordering to try preload's
  first.  This fixes an issue where `preload=res_config_curl` would fail
  unless res_curl and func_curl were also preloaded.  Now it is only
  required that those modules be loaded during startup: autoload or
  regular load is good enough.
* The configuration option `require` and `preload-require` were only
  effective if the modules failed to load.  These options will now abort
  Asterisk startup if required modules fail to reach the 'Running'
  state.
* Missing or invalid 'module.conf' did not prevent startup.  Asterisk
  doesn't do anything without modules so this a fatal error.

Change-Id: Ie4176699133f0e3a823b43f90c3348677e43a5f3

Change-Id: I9cae8bb3d1a2cea335d3ccd88d471832549666fd

Asterisk modules that use PJPROJECT services have their compiler
optimization and possibly their symbolic debug options overridden by the
PJPROJECT configure script selected settings.

* We need to filter-out any -O and -g options in PJ_CFLAGS before echoing
out the result so the PJPROJECT_INCLUDE variable does not override the
Asterisk module settings when using bundled PJPROJECT.

NOTE: This patch only has an effect when using bundled PJPROJECT.

ASTERISK-27563

Change-Id: If124169735ecf572ad1535cd43bff94cb44d5b30

Change-Id: Icebc0d013896f3b2a7214945cac60647435c1651

Change-Id: I6dc8ab1679b3505c6dde1d47e1b9276df47814f8

Change-Id: I5b581d0a0d1d1bb9b38961d40b112fb448355037

Change-Id: I6f31a130b3ba0187149aaaa2ce94195a79e0f6a6

Change-Id: I3a4b8a4a9c488ddabf9daf651dc1334222056f38

When `--without-libcurl` is used PBX_CURL is never set.  Set default
value 0 so the proper value is passed to menuselect.

Change-Id: I03e2842a00899cbca2dbde52bb1f6636d54bae1e

Previously, the msid "label" attribute was used to correlate
participant info but because streams could be reused, the msid
wasn't being updated correctly when someone left the bridge and
another joined.

Now, instead of looking for the msid attribute on a channel's streams,
app_confbridge sets an "SDP:LABEL" attribute on the stream which
res_pjsip_sdp_rtp looks for.  If it finds it, it adds a "label"
attribute to the current sdp.

Change-Id: I6cbaa87fb59a2e0688d956e72d2d09e4ac20d5a5