chan_skinny creates a new thread for each new session.  In trying
to be a good cleanup citizen, the threads are joinable and the
unload_module function does a pthread_cancel() and a pthread_join()
on any sessions that are active at that time.  This has an
unintended side effect though. Since you can call pthread_join on a
thread that's already terminated, pthreads keeps the thread's
storage around until you explicitly call pthread_join (or
pthread_detach()).   Since only the module_unload function was
calling pthread_join, and even then only on the ones active at the
tme, the storage for every thread/session ever created sticks
around until asterisk exits.

* A thread can detach itself so the session_destroy() function
  now calls pthread_detach() just before it frees the session
  memory allocation.  The module_unload function still takes care
  of the ones that are still active should the module be unloaded.

ASTERISK-27452
Reported by: Juan Sacco

Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd
(cherry picked from commit 8f5dff54)

ASTERISK-27434

Change-Id: Iaeed89b4fa05d94c5f0ec2d3b7cd6e93d2d5a8f7

* Balanced the session->inv_session refs on answer failure.

Change-Id: I33542d639d37e692cb46550b972a5fcfc3b804b8

The media frame cache gets in the way of finding use after free errors of
media frames.  Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.

* Added the "cache_media_frames" option to asterisk.conf.  Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG.  The cache gets in the way of determining if the frame is
used after free and who freed it.  NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.

To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.

Sample asterisk.conf setting:
[options]
cache_media_frames=no

ASTERISK-27413

Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00

Change-Id: I3f9dd3c31bd582e54a30381500077de2319d8cc3

This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.

Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.

ASTERISK-27395

Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4

This adds menuselect dependencies for modules that use symbols of other
modules.

ASTERISK-27390

Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385

When chan_sip receives a SUBSCRIBE request with no "Expires" header it
processes the request as an unsubscribe.  This is incorrect, per RFC3264
when the "Expires" header is missing a default expiry should be used.

ASTERISK-18140

Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5

This matches the behavior of the other SIP channel driver, chan_pjsip.

ASTERISK-27365

Change-Id: I8f23a51290a58b75816da2999ed1965441dfc5d6

When sip.conf contains 'sipdebug=yes' it is impossible to disable it
using CLI 'sip set debug off'.  This corrects the output of that CLI
command to instruct the user to turn sipdebug off in the configuration
file.

ASTERISK-23462 #close

Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318

The sys/sysmacros.h include file does not exist in BSD systems and
is not required to build this module there.
Since an "#if defined(__NetBSD__) || defined(__FreeBSD__)" section
already exist I moved that include line inside it's #else branch.

ASTERISK-27343 #close

Change-Id: Ibfb64f4e9a0ce8b6eda7a7695cfe57916f175dc1

chan_vpb was trying to use sizeof(*p->play_dtmf), where
p->play_dtmf is defined as char[16], to get the length of the array
but since p->play_dtmf is an actual array, sizeof(*p->play_dtmf)
returns the size of the first array element, which is 1.  gcc7
validly complains because the context in which it's used could
cause an out-of-bounds condition.

Change-Id: If9c4bfdb6b02fa72d39e0c09bf88900663c000ba

Currently privacy requests are only granted if the Privacy header
value is exactly "id" (defined in RFC 3325). It ignores any other
possible value (or a combination there of). This patch reverses the
logic from testing for "id" to grant privacy, to testing for "none" and
granting privacy for any other value. "none" must not be used in
combination with any other value (RFC 3323 section 4.2).

ASTERISK-27284 #close

Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56

Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec

Some endpoints do not like a stream being reused for a new
media stream. The frame/jitterbuffer can rely on underlying
attributes of the media stream in order to order the packets.
When a new stream takes its place without any notice the
buffer can get confused and the media ends up getting dropped.

This change uses the SSRC change to determine that a new source
is reusing an existing stream and then bridge_softmix renegotiates
each participant such that they see a new media stream. This
causes the frame/jitterbuffer to start fresh and work as expected.

ASTERISK-27277

Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07

chan_pjsip_indicate was missing a case for the recently added
AST_CONTROL_STREAM_TOPOLOGY_CHANGED condition and was returning an
error and causing the call to be hung up instead of just ignoring
it.

ASTERISK-27260
Reported by: Daniel Heckl

Change-Id: I4fecbb00a0b8a853da85155065c1a6bddf235e80

Provide a way to get the contents of the the Request URI from the initial SIP
INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}")

ASTERISK-27278
Reported by: David J. Pryke
Tested by: David J. Pryke

Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e

Multicast/Unicast RTP do not use SDP so we need to use a format that
cleanly maps to one of the static RTP payload types. Without this
change, an Originate to a Multicast or Unicast channel without a format
specified would produce no audio on the receiving device.

ASTERISK-21399 #close
Reported by: Tzafrir Cohen

Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3

In handle_request_invite, when processing a pickup, a call
is made to get_sip_pvt_from_replaces to locate the pvt for
the subscription. The pvt is assumed to be valid when zero
is returned indicating no error, and is dereferenced which
can cause a crash if it was not found.

This change checks the not found case and returns -1 which
allows the calling code to fail appropriately.

ASTERISK-27217 #close
Reported-by: Bryan Walters

Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612

If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE
to both parties to set up media path directly between the endpoints.
In this reINVITE msg SDP origin line (o=) contains IP address of endpoint
instead of IP of asterisk. This behavior violates RFC3264, sec 8:
"When issuing an offer that modifies the session,
the "o=" line of the new SDP MUST be identical to that in the
previous SDP, except that the version in the origin field MUST
increment by one from the previous SDP."
This patch assures IP address of Asterisk is always sent in
SDP origin line.

ASTERISK-17540
Reported by:  saghul

Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e

When rtp_keepalive is on for a PJSIP endpoint dialing to another
Asterisk instance also using PJSIP, Asterisk will continue to print
warning messages about not being able to send frames of a certain
type. This suppresses that warning message.

Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67

Create local_tag and remote_tag in CHANNEL info to get tag from From and
To headers of a SIP dialog.

ASTERISK-27220

Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524

If an SDP answer hasn't been sent yet, it's legal to change it.
This is required for PJSIP_DTMF_MODE to work correctly, and can
also have use in the future for updating codecs too.

ASTERISK-27209 #close

Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1

There are other 1xx and 2xx codes than 100 and 200 respectively.

Change-Id: I680db0997343256add1478714f5bf5b5569aee17

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