Change-Id: Ie29760c49c25d7022ba2124698283181a0dd5d08

Change-Id: I7707c9d872c476d897ff459008652b35142a35e1

Change-Id: I74431b385da333f2c5f5a6d7c55e70b69a4f05d2

Change-Id: I6d9edd34d8b2474222c86f44e379ead61e57a54f

If you use ast_request to create a PJSIP channel but then hang it
up without causing a transaction to be sent, the session will
never be destroyed.  This is due ot the fact that it's pjproject
that triggers the session cleanup when the transaction ends.
app_chanisavail was doing this to get more granular channel state
and it's also possible for this to happen via ARI.

* ast_sip_session_terminate was modified to explicitly call the
  cleanup tasks and unreference session if the invite state is NULL
  AND invite_tsx is NULL (meaning we never sent a transaction).

* chan_pjsip/hangup was modified to bump session before it calls
  ast_sip_session_terminate to insure that session stays valid
  while it does its own cleanup.

* Added test events to session_destructor for a future testsuite
  test.

ASTERISK-26908 #close
Reported-by: Richard Mudgett

Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9

This change extends the ast_request functionality by adding another
function and callback to create an outgoing channel with a requested
stream topology. Fallback is provided by either converting the
requested stream topology into a format capabilities structure if
the channel driver does not support streams or by converting the
requested format capabilities into a stream topology if the channel
driver does support streams.

The Dial application has also been updated to request an outgoing
channel with the stream topology of the calling channel.

ASTERISK-26959

Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6

It's possible for a name in a party id structure to be marked as valid, but the
name string itself be NULL (for instance this is possible to do by using the
dialplan CALLERID function). There were a couple of places where the name was
validated, but the string itself was not checked before passing it to functions
like 'strlen'. This of course caused a crashed.

This patch adds in a NULL check before attempting to pass it into a function
that is not NULL tolerant.

ASTERISK-25823 #close

Change-Id: Iaa6ffe9d92f598fe9e3c8ae373fadbe3dfbf1d4a

Added an pre-defined integer vector declaration. This makes integer vectors
easier to declare and pass around. Also, added the ability to default a vector
up to a given size with a default value. Lastly, added functionality that
returns the "nth" index of a matching value.

Also, updated a unit test to test these changes.

Change-Id: Iaf4b51b2540eda57cb43f67aa59cf1d96cdbcaa5

Interpolated frames are frames which contain a number of
samples but have no actual data. Audiohooks did not
handle this case when translating an incoming frame into
signed linear. It assumed that a frame would always contain
media when it may not. If this occurs audiohooks will now
immediately return and not act on the frame.

As well for users of ast_trans_frameout the function has
been changed to be a bit more sane and ensure that the data
pointer on a frame is set to NULL if no data is actually
on the frame. This allows the various spots in Asterisk that
check for an interpolated frame based on the presence of a
data pointer to work as expected.

ASTERISK-26926

Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b

Without the disable, pjproject tries to build it's internal
webrtc implementation which requires sse2.  This fails on
platforms without sse2.

ASTERISK-26930 #close
Reported-by: abelbeck

Change-Id: I07231f9160c35cfa42b194d3aad4e7d51fd9a410

Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in
the format modules. Neither of these functions will ever return a value
less than 0, which we were checking for in some cases.

I've introduced a fair amount of duplication in the format modules, but
I plan to change how format modules work internally in a subsequent
patch set, so this is simply a stop-gap.

Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872

This change adds an Alembic migration which adds a
ps_resource_list table that can contain resource_list
RLS configuration objects.

ASTERISK-26929

Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05

This change adds a T.38 format which can be used in a stream
topology to specify that a UDPTL stream needs to be created.
The SDP API has been changed to understand T.38 and create
the UDPTL session, add the attributes, and parse the attributes.

This change does not change the boundary of the T.38 state
machine. It is still up to the channel driver to implement and
act on it (such as queueing control frames or reacting to them).

ASTERISK-26949

Change-Id: If28956762ccb8ead562ac6c03d162d3d6014f2c7

The gist of this work ensures that when a remote SDP is received, it is
merged properly with the local capabilities. The remote SDP is converted
into a stream topology. That topology is then merged with the current
local topology on the SDP state. That new merged topology is then used
to create an SDP. Finally, adjustments are made to RTP instances based
on knowledge gained from the remote SDP.

There are also a battery of tests in this commit that ensure that some
basic SDP merges work as expected.

While this may not sound like a big change, it has the property that it
caused lots of ancillary changes.

* The remote SDP is no longer stored on the SDP state. Biggest reason:
  there's no need for it. The remote SDP is used at the time it is being
  set and nowhere else.

* Some new SDP APIs were added in order to find attributes and convert
  generic SDP attributes into rtpmap structures.

* Writing tests made me realize that retrieving a value from an SDP
  options structure, the SDP options needs to be made const.

* The SDP state machine was essentially gutted by a previous commit.
  Initially, I attempted to reinstate it, but I found that as it had
  been defined, it was not all that useful. What was more useful was
  knowing the role we play in SDP negotiation, so the SDP state machine
  has been transformed into an indicator of role.

* Rather than storing separate local and joint stream state
  capabilities, it makes more sense to keep track of current stream
  state and update it as things change.

Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d

* Initialize hepv3_runtime_data.sockfd to -1 so that our ao2 destructor
  does not close fd 0

* Add logging output when the required option - capture_address - is not
  specified.

* Remove a no longer relevant #define and correct related documentation

* Pass appropriate flags to aco_option_register so that capture_address
  cannot be the empty string.

ASTERISK-26953 #close

Change-Id: Ief08441bc6596d6f1718fa810e54a5048124f076

The primary win of switching to eventfd when possible is that it only
uses a single file descriptor while pipe() will use two. This means for
each bridge channel we're reducing the number of required file
descriptors by 1, and - if you're using timerfd - we also now have 1
less file descriptor per Asterisk channel.

The API is not ideal (passing int arrays), but this is the cleanest
approach I could come up with to maintain API/ABI.

I've also removed what I believe to be an erroneous code block that
checked the non-blocking flag on the pipe ends for each read. If the
file descriptor is 'losing' its non-blocking mode, it is because of a
bug somewhere else in our code.

In my testing I haven't seen any measurable difference in performance.

Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d

If ICE is enabled and a STUN server does not respond then we will block
until we give up on the STUN response.  This will take nine seconds.  In
the mean time the peer that sent the INVITE will send retransmissions.

* Restructure res_pjsip_session.c:new_invite() to send a 100 Trying out
earlier to prevent these retransmissions.

ASTERISK-26890

Change-Id: Ie3fc611e53a0eff6586ad55e4aacad81cf6319a8

* Restructure ast_sip_session_alloc() to need less cleanup on off nominal
error paths.

* Made ast_sip_session_alloc() and ast_sip_session_create_outgoing() avoid
unnecessary ref manipulation to return a session.  This is faster than
calling a function.  That function may do logging of the ref changes with
REF_DEBUG enabled.

Change-Id: I2a0affc4be51013d3f0485782c96b8fee3ddb00a

Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core
to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE
is passed to these functions, the calling thread will be blocked until
the newly created channel has been hung up.

After this patch, we run the dial on the current thread rather than
spawning a new one. The only in-tree code that passes
AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced
thread usage if you are using .call files.

Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913

Occasionally a crash happens when processing the RTCP DTLS timeout
handler.  The RTCP DTLS timeout timer could be left running if we have not
completed the DTLS handshake before we place the call on hold or we
attempt direct media.

* Made ast_rtp_prop_set() stop the RTCP DTLS timer when disabling RTCP.

* Made some sanity tweaks to ast_rtp_prop_set() when switching from
standard RTCP mode to RTCP multiplexed mode.

ASTERISK-26692 #close

Change-Id: If6c64c79129961acfa4b3d63a864e8f6b664acc0