Introduce a new property to rtp-engine to make it aware of
the desire for assymetric codecs or not.  If asymmetric codecs
is not allowed, the bridge will compare read/write formats
and shut down the p2p bridge if needed

ASTERISK-26745 #close

Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f

This adds a way to access information passed along with SIP headers in
a REFER message that initiates a transfer. Headers matching a dialplan
variable GET_TRANSFERRER_DATA in the transferrer channel are added to
a HASH object TRANSFER_DATA to be accessed with functions HASHKEY and HASH.

The variable GET_TRANSFERRER_DATA is interpreted to be a prefix for
headers that should be put into the hash. If not set, no headers are
included. If set to a string (perhaps 'X-' in a typical case), all headers
starting this string are added. Empty string matches all headers.

If there are multiple of the same header, only the latest occurrence in
the REFER message is available in the hash.

Obviously, the variable GET_TRANSFERRER_DATA must be inherited by the
referrer channel, and should be set with the '_' or '__' prefix.

I avoided a specific reference to SIP or REFER, as in my mind the mechanism
can be generalized to other channel techs.

ASTERISK-27162

Change-Id: I73d7a1e95981693bc59aa0d5093c074b555f708e

* chan_sip: channel in test_sip_rtpqos_1.
* test_config: config hook, config info and global config holder.
* test_core_format: format in format_attribute_set_without_interface.
* test_stream: unneeded frame duplication.
* test_taskprocessor: task_data.

Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31

Syntax: SIP_HEADERS([prefix])

If the argument is specified, only the headers matching the given prefix
are returned.

The function returns a comma-separated list of SIP header names from an
incoming INVITE message. Multiple headers with the same name are included
in the list only once. The returned list can be iterated over using the
functions POP() and SIP_HEADER().

For example, '${SIP_HEADERS(Co)}' might return the string
'Contact,Content-Length,Content-Type'.

Practical use is rather '${SIP_HEADERS(X-)}' to enumerate optional
extended headers sent by a peer.

ASTERISK-27163

Change-Id: I2076d3893d03a2f82429f393b5b46db6cf68a267

Use -Wno-format-truncation only if supported by compiler.

ASTERISK-27171 #close

Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6

GCC 7 has added capability to produce warnings, this fixes most of those
warnings.  The specific warnings are disabled in a few places:

* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().

ASTERISK-27156 #close

Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88

This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis

ASTERISK-27085 #close

Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612

This change adds VP9 as a known codec and creates a cached
"vp9" media format for use.

Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc

On every reload of chan_iax2 module, MWI subscription was added, which
results in additional taskprocessors being accumulated over time.

This commit fixes it by making sure we check for existing subscription
first.

This was verified with 'core show taskprocessors' CLI command.

ASTERISK-27122 #close

Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9

This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:

  rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
  media_encryption=dtls
  dtls_verify=fingerprint
  dtls_setup=actpass

When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.

Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.

ASTERISK-27119 #close

Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd

BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.

This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.

For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.

ASTERISK-27118

Change-Id: I96c0920b9f9aca7382256484765a239017973c11

This API was not actively maintained, was not added to new modules
(such as res_pjsip), and there exist better alternatives to acquire the
same information, such as the ARI.

Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83

When sip.conf contained tcpenable=yes and autodomain=yes, the TCP domain was
added in any case, because of a local Boolean-negation error of the return value
of ast_sockaddr_cmp. After fixing this error for TCP and TLS, the TLS domain was
still always added with tlsenable=yes, because the domains were not compared
just on the address but also on the port – and TLS is always on a different port
than UDP/TCP.

ASTERISK-27106

Change-Id: I14fe9e319e238320b094016980445ef3a5b3337c

Because of a copy-and-paste error when the struct ast_sockaddr changed,
tlsbindaddr was not added, when sip.conf contained autodomain=yes; see
"show sip domains" on the command-line interface (CLI) of Asterisk.

ASTERISK-27106

Change-Id: I3d0957150017c223136968ef1266f275d0d6695e

When connected_line_method is "invite", we're supposed to determine
if the client can support UPDATE and if it can, send UPDATE instead
of INVITE to avoid the SDP renegotiation.  Not only was pjproject
not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing
that invite_tsx wasn't NULL which isn't always the case.

* Updated chan_pjsip/update_connected_line_information to drop the
  requirement that invite_tsx isn't NULL.
* Submitted patch to pjproject sip_inv.c that sets the
  PJSIP_INV_SUPPORT_UPDATE flag correctly.
* Updated pjsip.conf.sample to clarify what happens when "invite"
  is specified.

ASTERISK-27095

Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560

The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated.  This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.

ASTERISK-27066 #close

Change-Id: Id185b11e84afd9191a2f269e8443019047765e91

The stream topology (list of streams and order) is now stored with the
configured PJSIP endpoints and used during the negotiation process.

Media negotiation state information has been changed to be stored
in a separate object. Two of these objects exist at any one time
on a session. The active media state information is what was previously
negotiated and the pending media state information is what the
media state will become if negotiation succeeds. Streams and other
state information is stored in this object using the index (or
position) of each individual stream for easy lookup.

The ability for a media type handler to specify a callback for
writing has been added as well as the ability to add file
descriptors with a callback which is invoked when data is available
to be read on them. This allows media logic to live outside of
the chan_pjsip module.

Direct media has been changed so that only the first audio and
video stream are directly connected. In the future once the RTP
engine glue API has been updated to know about streams each individual
stream can be directly connected as appropriate.

Media negotiation itself will currently answer all the provided streams
on an offer within configured limits and on an offer will use the
topology created as a result of the disallow/allow codec lines.

If a stream has been removed or declined we will now mark it as such
within the resulting SDP.

Applications can now also request that the stream topology change.
If we are told to do so we will limit any provided formats to the ones
configured on the endpoint and send a re-invite with the new topology.

Two new configuration options have also been added to PJSIP endpoints:

max_audio_streams: determines the maximum number of audio streams to
offer/accept from an endpoint. Defaults to 1.

max_video_streams: determines the maximum number of video streams to
offer/accept from an endpoint. Defaults to 1.

ASTERISK-27076

Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7

The construction of the returned string assumed incorrectly that the
supplied buffer would always be initialized as an empty string.  If it is
not an empty string we could overrun the supplied buffer by the length of
the non-empty buffer string plus one.  It is also theoreticaly possible
for the supplied buffer to be overrun by a string terminator during a read
operation even if the supplied buffer is an empty string.

* Fix the assumption that the supplied buffer would already be an empty
string.  The buffer is not guaranteed to contain an empty string by all
possible callers.

* Fix string terminator buffer overrun potential.

Change-Id: If6a0806806527678c8554b1dcb34fd7808aa95c9

The chan_pjsip module uses a calculation approach for
determining device state. This means that in situations
where we would expect device state to change we need to
tell the core to query. A scenario that was missed is
when early media was signaled.

This change adds the notification for the core to
query device state when we are told that early media
is being provided.

ASTERISK-27039

Change-Id: Iafebfd152894966344ff2e950a3cee9f59a3eb6f

PJSIP support in Asterisk differs from chan_sip in that it
allows media to be sent as-is without transcoding provided
the codecs were negotiated in the SDP. This is allowed
according to the RFC. Support for this differs quite a lot
though and some endpoints do not handle it well.

This change extends the 'asymmetric_rtp_codec' option to
also cover this case. When set to no (the default) the code
behaves as chan_sip does - the best codec is selected and
we will only ever send that, unless we change what we are
sending if the remote side changes. When set to yes we
will send media as-is without transcoding if the codec
has been negotiated in the SDP.

ASTERISK-26996

Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51

This introduces the ability for PJSIP code to specify filtering flags
when retrieving PJSIP contacts. The first flag for use causes the
query code to only retrieve contacts that are not unreachable. This
change has been leveraged by both the Dial() process and the
PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt
calls to contacts which are not unreachable.

ASTERISK-26281

Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c

If we are offered or are offering RTCP-MUX, don't consider RTCP ICE
candidates. This confuses certain browsers (current Firefox for
example) and causes intial audio setup delays.

ASTERISK-26982 #close

Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91

The while(1) loop in skinny_session wasn't checking for EOF so
a packet that was longer than a header but still truncated
would spin the while loop infinitely.  Not only does this
permanently tie up a thread and drive a core to 100% utilization,
the call of ast_log() in such a tight loop eats all available
process memory.

Added poll with timeout to top of read loop

ASTERISK-26940 #close
Reported-by: Sandro Gauci

Change-Id: I2ce65f3c5cb24b4943a9f75b64d545a1e2cd2898

Return cahnnel nativeformats to fix bridge technology selection process.
Same approach as in pjsip module.

ASTERISK-26143
Reported-by: Henning Holtschneider

Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48

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

For outgoing TCP connections, Asterisk uses the first IP address of the
interface instead of the IP address we asked him to bind to.

ASTERISK-26922 #close
Reported-by: Ksenia

Change-Id: I43c71ca89211dbf1838e5bcdb9be8d06d98e54eb

Some equipments may send a re-INVITE containing an SDP in the final ACK
request. If this happens in the context of direct media, the remote end
should be updated with a re-INVITE.
This patch queues an "update RTP peer" frame to trigger the re-INVITE,
instead of the "source change" frame wich was used previously.

ASTERISK-26951

Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6

This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compact_headers=yes via the file pjsip.conf.

ASTERISK-26932 #close

Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689

In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE.  This prevents asterisk from exiting
if a module can't be loaded.  If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.

A new API was added to logger: ast_is_logger_initialized().  This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout.  If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.

Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25

SIP user-agents indicate which protocol extensions are allowed in headers
like Supported and Required. Such protocol extensions are Session Timers
(RFC 4028) for example. Session Timers are supported since Mantis-10665.
Since ASTERISK-21721, not only the first but multiple Supported/Required
headers in a message are parsed. In that change, an existing variable was
re-used within a newly added do-loop. Currently, at the end of that loop,
that variable is an empty string always. Previously, that variable was used
within log output. However, the log output was not changed.

ASTERISK-26915 #close

Change-Id: I09315f31b4d78fb214bb2a9fb6c0f5e143eae990

Added missing channel technology read/write stream callback
initialization.

Change-Id: I829043a327d987e0d964485dd3d27964bebbd623

POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.

Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.

Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48

ASTERISK-26846 #close

Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639

When using a non-dynamic peer address, build_peer() invalidates the
peer address structure by setting the address family to unspecified.
However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup()
will not amend the peer address if the cache is still valid, resulting
in peer connectivity failures.
To fix this, we call ast_dnsmgr_refresh() instead.

ASTERISK-26865

Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082

When querying for PJSIP specific information using the dialplan
function CHANNEL() it is possible that the underlying session
will no longer have a channel associated with it. This is
most likely to occur when the RTCP HEP module attempts to get
the channel name. If this happens then a crash will occur.

This change just adds a check that the channel exists on the
session before querying it.

ASTERISK-26857

Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01

* cli_commands.c Fixed CLI output

ASTERISK-26822 #close

Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01

When receiving a 422 response, the invitestate variable must be reset to
INV_CALLING.

ASTERISK-26841

Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099

This change adds a few things to facilitate stream topology changing:

1. Control frame types have been added for use by the channel driver
to notify the application that the channel wants to change the stream
topology or that a stream topology change has been accepted. They are
also used by the indicate interface to the channel that the application
uses to indicate it wants to do the same.

2. Legacy behavior has been adopted in ast_read() such that if a
channel requests a stream topology change it is denied automatically
and the current stream topology is preserved if the application is
not capable of handling streams.

Tests have also been written which confirm the multistream and
non-multistream behavior.

ASTERISK-26839

Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9

* manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
if a hint does not exist for the requested extension.  Ran into this when
developing a testsuite test.  The AMI event ExtensionStatus came out with
the hint header value containing garbage.  The AMI event PresenceStatus
also had the same issue.

* manager.c:action_extensionstate() no need to completely initialize the
hint[].  Only initialize the first element.

* pbx.c:ast_add_hint() Remove unnecessary assignment.

* chan_sip.c: Eliminate an unneeded hint[] local variable.  We only care
about the return value of ast_get_hint() there.

Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b