* netsock2.c: Test the addr->len member first as it may be the only member
initialized in the struct.

* stun.c:ast_stun_handle_packet(): The combinded[] local array could get
used uninitialized by ast_stun_request().  The uninitialized string gets
copied to another location and could overflow the destination memory
buffer.

These valgrind findings were found for ASTERISK_27150 but are not
necessarily a fix for the issue.

Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57

The fix for the issue is broken up into three parts.

This is part three which handles the client side of REGISTER requests.
The registered contact may no longer be valid on the server when the
transport used is reliable and the connection is broken.

* Re-REGISTER our contact if the reliable transport is broken after
registration completes.  We attempt to re-REGISTER immediately to minimize
the time we are unreachable.  Time may have already passed between the
connection being broken and the loss being detected.

* Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's
are still correct if an allocation failure happens.

ASTERISK-27147

Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83

The fix for the issue is broken up into three parts.

This is part two which handles the server side of REGISTER requests when
rewrite_contact is enabled.  Any registered reliable transport contact
becomes invalid when the transport connection becomes disconnected.

* Monitor the rewrite_contact's reliable transport REGISTER contact for
shutdown.  If it is shutdown then the contact must be removed because it
is no longer valid.  Otherwise, when the client attempts to re-REGISTER it
may be blocked because the invalid contact is there.  Also if we try to
send a call to the endpoint using the invalid contact then the endpoint is
not likely to see the request.  The endpoint either won't be listening on
that port for new connections or a NAT/firewall will block it.

* Prune any rewrite_contact's registered reliable transport contacts on
boot.  The reliable transport no longer exists so the contact is invalid.

* Websockets always rewrite the REGISTER contact address and the transport
needs to be monitored for shutdown.

* Made the websocket transport set a unique name since that is what we use
as the ao2 container key.  Otherwise, we would not know which transport we
find when one of them shuts down.  The names are also used for PJPROJECT
debug logging.

* Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state
event.  Now the global keep_alive_interval option, initially idle shutdown
timer, and the server REGISTER contact monitor can work on wetsocket
transports.

* Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction.
Now initially idle websockets will automatically shutdown.

ASTERISK-27147

Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4

The fix for the issue is broken up into three parts.

This is part one which refactors the transport state monitor code to allow
more modules to be able to monitor transports.

* Pull the management of PJPROJECT's transport state callback code from
res_pjsip_transport_management.c into res_pjsip.  Now other modules can
dynamically add and remove themselves from transport monitoring without
worrying about breaking PJPROJECT's callback chain.

* Add the ability for other modules to get a callback whenever a specific
transport is shutdown.

ASTERISK-27147

Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912

* Use monitored instead of the misleading keepalive name.

Change-Id: I9e5bcbb4ab2b82d49bcd0f06dfe85d15e0b552b6

Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c

When handling an incoming SIP MESSAGE, PJSIP
attaches the IP address that the message was
received from to the message in the variable
PJSIP_RECVADDR.  When the IP address is IPv6
the :PORT appended results in an unparseable
mess. By using an additional bit flag on the
pj_sockaddr_print call, the conventional use
of brackets around the address is achieved.

ASTERISK-27193 #close

Change-Id: I12342521f2ce87a5b6e4883d480a3fd957aa9fd9

Asterisk wasn't generating or forwarding RTCP packets when native
bridge was activated.  Also the stats weren't available via
CHANNEL(qos). Now the RTCP stats are always calculated.

ASTERISK-27158 #close

Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b

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

'--with-pjproject-bundled' is now the default when running
./configure. It can be disabled with '--without-pjproject-bundled'.

To make building without an internet connection easier, a new
./configure option '--with-download-cache' was added that sets
the cache for externals (like pjproject, the codecs and the DPMA),
AND the sounds files.  It can also be specified as an environment
variable named "AST_DOWNLOAD_CACHE".  The existing
'--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and
'--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable
remain and if specified, will override '--with-downloads-cache'.

ASTERISK-27189

Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce

A change was made long ago where the session was kept around
until the underlying INVITE session had been destroyed. This
had the side effect of also keeping the underlying media resources
around for this time as well.

This change ensures that when we are told to terminate the
session we immediately release any media sessions associated
with it.

ASTERISK-27110

Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82

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

This change fixes a few locking issues and some video misrouting.

1. When accessing the stream topology of a channel the channel lock
must be held to guarantee the topology remains valid.

2. When a channel was joined to a bridge the bridge specific
implementation for stream mapping was not invoked, causing video
to be misrouted for a brief period of time.

ASTERISK-27182

Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03

joint_cap needs to be released unconditionally as chan->tech->requester
does not steal the reference even on success.

ASTERISK-27180 #close

Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6

Currently, the handling of the msid attribute is not quite right. According to
the spec the msid's between the offer/answer are not dependent upon one another.
Meaning the same msid's given in an offer do not have to be returned in the
answer for a given stream. And they probably shouldn't be (copied/reused) since
this can potentially cause some browser side confusion.

This patch generates new msids when both an offer and answer are sent from
Asterisk. However, Asterisk does reuse the original msid it sent out for a
reinvite. Also audio+video streams are paired together by sharing the same
stream id, but a different track id.

ASTERISK-27179 #close

Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643

* 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

Change-Id: Ib5a19bfd597f63d9021baeb645fc11153b3afa57

* Remove unnecessary CMP_STOP.

* In handle_client_registration() use DEBUG_ATLEAST() to only do work
needed for the debug log message when the debug log message is needed.

* In sip_outbound_registration_state_destroy() check state->registration
for NULL.

Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80

Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da

Most uses of CMP_STOP are superfluous and are only respected when
OBJ_MULTIPLE is used to search the container.

Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8