You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:

[ice_host_candidates]
192.168.1.1 = 1.2.3.4,include_local_address

This causes both 192.168.1.1 and 1.2.3.4 to be advertized.

Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db

This change fixes two bugs which both resulted in the packet loss
count exceeding 65,000.

The first issue is that the sequence number check to determine if
cycling had occurred was using the wrong variable resulting in the
check never seeing that cycling has occurred, throwing off the
packet loss calculation. It now uses the correct variable.

The second issue is that the packet loss calculation assumed that
the received number of packets in an interval could never exceed
the expected number. In practice this isn't true due to delayed
or retransmitted packets. The expected will now be updated to
the received number if the received exceeds it.

ASTERISK-28379

Change-Id: If888ebc194ab69ac3194113a808c414b014ce0f6

When multiple endpoints try to register close together using the same
AOR with qualify_frequency set, one contact would qualify immediately
while the other contacts would have to wait out the duration of the
timer before being able to qualify. Changing the conditional to check
the contact container count for a non-zero value allows all contacts to
qualify immediately.

Change-Id: I79478118ee7e0d6e76af7c354d66684220db9415

When we use early bridge with create and dial from stasis using Local channel
and the dialplan does not any entry the it is returned from core_local.c with
No such extension .

In such case asterisk locks up till the channel is not hangup with the error
Exceptionally long voice queue length

* Found that in such case app_control_dial fails on ast_call method and
  return -1
* Since it is called from stasis_app_send_command_async and return -1 does
  not cause resources to be freed and since no PBX exist it is not able to
  read from channel causing exceptionally long queue
* After putting this code found that the channel was releasing immediately
  and resources were freed.

ASTERISK-28399
Reported by: Abhay Gupta
Tested by: Abhay Gupta

Change-Id: I0a55c923fc6995559f808d63b9488762b4489318

Updated ast_sip_create_rdata_with_contact and registrar_find_contact
to check the return from pjsip_parse_uri before attempting to
use the uri returned.

ASTERISK-28402
Reported-by: Ross Beer

Change-Id: I9810b3b163c45ed5a56ec743586e5ce107f13ba7

The dial bridge is meant to hold channels which have been created
and dialed in stasis. It handles the frames coming from them and raises
the appropriate events.

It was possible for the code to mistakenly place calls which came
from the dialplan into the dial bridge if they were not in an
answered state. These channels are not outgoing channels and
should not be placed into the dial bridge.

The code now checks to ensure that only stasis created channels are
placed into the dial bridge by checking that a PBX does not exist
on the channel.

ASTERISK-27756

Change-Id: Ideee69ff06c9a0b31f7ed61165f5c055f51d21b6

The transport-cc draft is a mechanism by which additional information
about packet reception can be provided to the sender of packets so
they can do sender side bandwidth estimation. This is accomplished
by having a transport specific sequence number and an RTCP feedback
message. This change implements this in the receiver direction.

For each received RTP packet where transport-cc is negotiated we store
the time at which the RTP packet was received and its sequence number.
At a 1 second interval we go through all packets in that period of time
and use the stored time of each in comparison to its preceding packet to
calculate its delta. This delta information is placed in the RTCP
feedback message, along with indicators for any packets which were not
received.

The browser then uses this information to better estimate available
bandwidth and adjust accordingly. This may result in it lowering the
available send bandwidth or adjusting how "bursty" it can be.

ASTERISK-28400

Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc

There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:

main/mwi.h
main/mwi.c

Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.

Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0

...and return the frame unaltered if bitrate can't be determined.

Change-Id: Ib2175ab84f85a3d7060d31625f5a2c7fbcc2ba4c

Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.

res_pjsip_refer:  Configures PJSIP norefersub capability accordingly.

Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.

This is useful for Cisco switches that do not follow RFC4488.

ASTERISK-28375 #close
Reported-by: Dan Cropp

Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9

Suggested by abelbeck on the issue tracker.

ASTERISK~28384
Reported by: abelbeck

Change-Id: Icee0fff2b58dbfaa80f2b68270fe69dfb0463fc0

An earlier contributor apparently forgot to run 'make ari-stubs'
before committing after making ARI model changes.

Change-Id: I7813e5638e2821d11f4b968dc2aeab4f725190a6

Reset the internal counter that the AEL2 compiler uses for unique label
names before compiling. This keeps dialplan labels consistent across
reloads assuming the AEL2 has not changed.

ASTERISK-17799 #close
Reported by: Kirill Katsnelson

Change-Id: I30b3cc887d1ee0644d3f341e2fef16f525d7fae5

In AEL2, if a 'for' statement contains macro* calls, like:

    for (&iterator(${TRY},A); "${A}" != ""; &iterate(A)) {

The AEL2 parser will translate these into calls to the deprecated Macro
dialplan application and use the antiquated pipe delimiter.

Instead, convert these into calls to the Gosub dialplan application and
use commas as argument separators.

ASTERISK-18593 #close
Reported by: Luke-Jr

* 'macro' in this context means AEL2 macros, not the 'Macro' application

Change-Id: I3d73716033b8e3e42e0209d355bf5f10c97045fc

When generating the regular expression that matches against existing
extensions, we need to escape literal characters that can also be
regular expression metacharacters. This was already being done for '*'
but we need to do the same for '+'.

In passing, remove some unreachable code - strcmp() is already run
immediately when entering extension_matches().

ASTERISK-14939 #close
Reported by: klaus3000

Change-Id: I8d2cccb3479168fba1b0a6704c52198b396468f1

Add missing ao2_ref(transport_state, -1) while iterate on a transport_states
container.

Change-Id: I40e35b5a339121300c80075c30db47201a6c374e

REMB's exponent is 6-bits (0..63) and has a mantissa of 18-bits. We were using
an unsigned integer to represent the bitrate. However, that type is not large
enough to hold all potential bitrate values. If the bitrate is large enough
bits were being shifted off the "front" of the mantissa, which caused the
wrong value to be sent to the browser.

This patch makes it so it now uses a float type to hold the bitrate. Using a
float allows for all bitrate values to be correctly represented.

ASTERISK-28255

Change-Id: Ice00fdd16693b16b41230664be5d9f0e465b239e

It looks like we're not properly calculating jitter values on received
video streams.  This patch enables the code that does jitter calculations
for those streams.

Change-Id: Iaac985808829c8f034db8c57318789c4c8c11392

It was difficult to check the channel's current application and
parameters using ARI for current channels. Added app_name, app_data
items to show the current application information.

ASTERISK-28343

Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c

If Realtime @ variable value is NULL or empty or contains only whitespaces
then when we try to retrieve it using PJSIP_ENDPOINT we get WARNING
pjsip_endpoint_function_read: Unknown property @my_var for PJSIP endpoint.
And the variable is missing in the result of CLI pjsip show endpoint.

This patch keeps empty sorcery extended field.

ASTERISK-28341 #close

Change-Id: I221fccc04cbfa2be17ce971f64ae0e74e465eea0

Because StasisEnd's timestamp created it's own timestamp, it makes
wrong timestamp, compare to other channel event(ChannelDestroyed).
Fixed to getting a timestamp from the Channel's timestamp.

ASTERISK-28333

Change-Id: I5eb8380fc472f1100535a6bc4983c64767026116

As part of an earlier voicemail refactor, ast_delete_mwi_state_full
was modified to remove the pool topic for a mailbox when the state
was deleted.  This was an attempt to prevent stale topics from
accumulating when app_voicemail was reloaded and a mailbox went
away.  Unfortunately because of the fact that when app_voicemail
reloads, ALL mailboxes are deleted then only current ones recreated,
topics were being removed from the pool that still had subscribers
on them, then recreated as new topics of the same name.  So now
modules like res_pjsip_mwi are listening on a topic that will
never receive any messages because app_voicemail is publishing on
a different topic that happens to have the same name.  The solutiuon
to this is not easy and given that accumulating topics for
deleted mailboxes is less evil that not sending NOTIFYs...

* Removed the call to stasis_topic_pool_delete_topic in
  ast_delete_mwi_state_full.

Also:

* Fixed a topic reference leak in res_pjsip_mwi
  mwi_stasis_subscription_alloc.

* Added some debugging to mwi_stasis_subscription_alloc,
  stasis_topic_create, and topic_dtor.

* Fixed a topic reference leak in an error path in
  internal_stasis_subscribe.

ASTERISK-28306
Reported-by: Jared Hull

Change-Id: Id7da0990b3ac4be4b58491536b35f41291247b27

Added ARI resource for channel statistics.
GET /ari/channels/{channelId}/rtp_statistics : It returns given
channel's rtp statistics detail.

ASTERISK-28320

Change-Id: I4343eec070438cec13f2a4f22e7fd9e574381376

Change-Id: I1e4d37415f3034abe36496dc30209c2303e6af5c

Changed to requirement to having timestamp for all of ARI events.
The below ARI events were changed to having timestamp.
PlaybackStarted, PlaybackContinuing, PlaybackFinished,
RecordingStarted, RecordingFinished, RecordingFailed,
ApplicationReplaced, ApplicationMoveFailed

ASTERISK-28326

Change-Id: I382c2fef58f5fe107e1074869a6d05310accb41f

Topic names now follow: <subsystem>:<functionality>[/<object>]

This ensures that they are all unique, and also provides better
insight in to what each topic is for.

Subscriber ids now also use the main topic name they are
subscribed to and an incrementing integer as their identifier to
make it easier to understand what the subscription is primarily
responsible for.

Both the CLI commands for listing topic and subscription statistics
now sort to make it a bit easier to see what is going on.

Subscriptions will now show all topics that they are receiving messages
from, not just the main topic they were subscribed to.

ASTERISK-28335

Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d

Currently, when the Asterisk calculates rtp statistics, it uses
sample_count as a unsigned integer parameter. This would be fine
for most of cases, but in case of large enough number of sample_count,
this might be causing the divide by zero error.

ASTERISK-28321

Change-Id: If7e0629abaceddd2166eb012456c53033ea26249

Change-Id: I481fabd8eaf2e4e7ffb5c8285b294742826e7d12

chan_sip will always ignore 183 responses that do not contain SDP
however, chan_pjsip will currently always translate it into a
183 with SDP.  This new flag allows chan_pjsip to have the same
behavior as chan_sip.

ASTERISK-28322 #close

Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a

strtok() uses a static buffer, making it not thread safe.

Also add a #define to cause a compile failure if strtok is used.

Change-Id: Icce265153e1e65adafa8849334438ab6d190e541

Added the ability to move between Stasis applications within Stasis.
This can be done by calling 'move' in an application, providing (at
minimum) the channel's id and the application to switch to. If the
application is not registered or active, nothing will happen and the
channel will remain in the current application, and an event will be
triggered to let the application know that the move failed. The event
name is "ApplicationMoveFailed", and provides the "destination" that the
channel was attempting to move to, as well as the usual channel
information. Optionally, a list of arguments can be passed to the
function call for the receiving application. A full example of a 'move'
call would look like this:

client.channels.move(channelId, app, appArgs)

The control object used to control the channel in Stasis can now switch
which application it belongs to, rather than belonging to one Stasis
application for its lifetime. This allows us to use the same control
object instead of having to tear down the current one and create
another.

ASTERISK-28267 #close

Change-Id: I43d12b10045a98a8d42541889b85695be26f288a

This small feature will help to checking the bridge's status to
figure out which bridge is in old/zombie or not. Also added
detail items for the 'bridge show *' cli to provide more detail
info. And added creation item to the ARI as well.

ASTERISK-28279

Change-Id: I460238c488eca4d216b9176576211cb03286e040

PJSIP assumes that these header names are not allocated, and does not
clone the name strings when reusing headers.

Block unload of res_pjsip_diversion until shutdown to ensure static
memory stays valid.

ASTERISK-28312 #close

Change-Id: Ibd6ea55ec4a604bbd43ac07f8d0b54da2c39b8b9

Rather than calling ast_odbc_find_table() in the prepare callback, call
it beforehand and pass it in to the callback to avoid the need for a
second connection.

ASTERISK-28166 #close

Change-Id: I6f8a0b9990d636fd6bc1a92ed70f7050d2436202

apply_negotiated_sdp_stream was returning a "1" when no joint
capabilities were found on an outgoing call instead of a "-1".
This indicated to res_pjsip_session that the handler DID handle
the sdp when in fact it didn't.  Without the appropriate setup,
a subsequent media frame coming in would have an invalid stream_num
and cause a seg fault when the stream was attempted to be retrieved.

apply_negotiated_sdp_stream now returns the correct "-1" and any
media is now discarded before it reaches the core stream processing.

ASTERISK-28260
Reported by: Sotiris Ganouris

Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f

This reverts commit d524ad52.

Reason for revert: This causes Contact and Via headers to have the wrong
transport address.

ASTERISK-28309 #close

Change-Id: Ibba4d6176f68e39279fcd9a545f81d56e747bed8

If both send_registrations and send_auth are both set to yes,
outbound_auth/username must be set or we crash.

ASTERISK-27992 #close

Change-Id: I6418d56de1ae53f80393b314c2584048fbf7f11d

When a contact was removed by the registrar it did not always check to see if
the circumstances involved a monitored reliable transport. For instance, if the
'remove_existing' option was set to 'true' then when existing contacts were
removed due to 'max_contacts' being reached, those existing contacts being
removed did not unregister the transport monitor.

Also, it was possible to add more than one monitor on a reliable transport for
a given aor and contact.

This patch makes it so all contact removals done by the registrar also remove
any associated transport monitors if necessary. It also makes it so duplicate
monitors cannot be added for a given transport.

ASTERISK-28213

Change-Id: I94b06f9026ed177d6adfd538317c784a42c1b17a

This module allows presence subscriptions to voicemail boxes.  This
allows common BLF keys to act as voicemail waiting indicators.

ASTERISK-28301

Change-Id: I62a246c24f3d7d432e33e22d7a4a57c15c292fdd

Delivery timeval in the smoother object will fall behind while a DTMF is
being generated.  This can eventually lead to invalid rtp timestamps.
To prevent this from happening the smoother needs to be reset after every
DTMF to keep the timing up to date.

ASTERISK-28303 #close

Change-Id: Iaba3f7b428ebd72a4caa90e13b829ab4f088310f