The pjsip_publishc_init() call was referenced with a misplaced
parentheses.  As a result, outbound publication messages went out with an
expiration of 1 second.

ASTERISK-27298

Change-Id: I93622eabc8ee83e7a22e98c107f921284c605a08

If using a legitimate certificate from a trusted certificate authority,
you don't need to provide CA file.

Change-Id: I8623973b4209b44889243716d7880274caed8a6d

ASTERISK-27289

Change-Id: I7a415948116493050614d9f4fa91ffbe0c21ec4c

Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec

This patch reverts the change by patch 2263 from old reviewboard.
Note that reverting that 2263-patch still preserves the behaviour that
the commit log of the 2263-patch claimed to add. The reason for this is:

The function wait_for_answer is only called from try_calling which
in turn is only called from the main for loop in queue_exec, and
earlier in that loop we already check the things that's removed by
this patch. There's no need to check those things twice each loop
iteration, and I think the proper place to check it is before each
ringing cycle. By checking it in wait_for_answer, you allow the issue
explained in the jira - that the head caller hears announcements while
the agents' sip phones are actively ringing.

Reported-by: Stefan Engström
Tested-by: Stefan Engström
ASTERISK-27216 #close

Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0

Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic
Edition after accepting the audio request but declining the video one.

Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c

When pruning a request to change the topology of a channel be
more intelligent about the resulting topology that is actually
used for SDP renegotiation.

In a case where a stream has not already been negotiated we
don't need to renegotiate and offer a declined stream. This can
occur if something in Asterisk (such as ConfBridge) requests
to add video to a PJSIP channel that has no video codecs configured.
In this case since the stream did not already exist we can safely
remove the stream from the requested topology, resulting in no
renegotiation occurring.

In a case where a renegotiation is requested with a codec that is
not supported we can reuse the formats of the existing stream if
it exists to ensure that the stream continues to flow, instead of
removing it.

Change-Id: I636540798d55922377318fe619c510fb6ed125fb

During a reinvite, if a remote endpoint error occurs and it returns a 500 the
session would end. This patch makes it so the session is not terminated, but
continues as it was.

The reason for this is because some endpoints may send non session terminating
"server errors" like a failed codec negotiation. So in this case instead of
ending the call it can hopefully continue. In the case of a real server error
the session is already "doomed", will be known soon enough and appropriately
ended by Asterisk later.

Change-Id: Ifeedae86b8cb44b92d52c79046522ec5f0aff1d5

Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d

When an INVITE came in with both audio and video streams but there
were no audio codecs defined for the endpoint, we weren't declining
the audio stream.  Since it's usually the first/transport stream,
when the video stream was processed and tried to use the transport,
it was empty and caused a crash.  We now decline the the stream if
there are no matching codecs so when the video stream is processed,
it's now the first/transport stream and processes normally.

Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692

Assertions in the v15+ AST-2017-008 patches found that we were not
handling the case if the incoming SDP did not specify the required SSRC
attributes for bundled to work.

* Be strict on matching SSRC for bundled instances including the parent
instance.  If the SSRC doesn't match then discard the packet.  Bundled has
to tell us in the SDP signaling what SSRC to expect.  Otherwise, we will
not know how to find the bundled instance structure.

Change-Id: I152830bbff71c662408909042068fada39e617f9

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

There was an issue reported where an SDP received on a 183 Session
Progress message caused a crash because the pending streams had
already been processed when the OK was received.  In that case the
pending topology was legitimately NULL.  There was an assert for an
incorrect number of streams in the topology but not one for
topology being NULL.  In any case, if you're not in dev-mode the
asserts don't do anything and since the scenario is legit, the
asserts weren't appropriate anyway.

* Changed several asserts to warning or debug messages and return
codes as appropriate.

ASTERISK-27264
Reported by: Daniel Heckl

Change-Id: I58daaa9d2938fa980857ab3ec41925ab5ff9c848

In PostgreSQL 9.1 the backslash are string literals and not the escape
of characters.

In previous issue ASTERISK_26057 was fixed the use of escape LIKE but the
support for old version of Postgresql than 9.1 was dropped. The sentence
before make was "ESCAPE '\'" but in version before than 9.1  need it to be
as follow "ESCAPE '\\'".

ASTERISK-27283

Change-Id: I96d9ee1ed7693ab17503cb36a9cd72847165f949

When a sip session is refreshed, the stream topology is looped
through, checking each stream for compatible formats. This would
cause a crash if the stream state was AST_STREAM_STATE_REMOVED,
since the formats would never be set for this stream, causing
a NULL value to be returned from ast_stream_get_formats. This
commit adds a check for streams with removed states.

Also removed a stray semicolon.

Change-Id: Ic86f8b65a4a26a60885b28b8b1a0b22e1b471d42

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

When two channels were early bridged in a native_rtp bridge, the RTP description
on one side was not updated when the other side answered.
This patch forbids non-answered channels to enter a native_rtp bridge, and
triggers a bridge reconfiguration when an ANSWER frame is received.

ASTERISK-27257

Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df

Previously, sRTP authentication failures were reported on log level WARNING.
When such failures happen, each RT(C)P packet is affected, spamming the log.
Now, those failures are reported at log level VERBOSE 2. Furthermore, the
amount is further reduced (previously all two seconds, now all three seconds).
Additionally, the new log entry informs whether media (RTP) or statistics (RTCP)
are affected.

ASTERISK-16898 #close

Change-Id: I6c98d46b711f56e08655abeb01c951ab8e8d7fa0

pubsub_on_rx_notify_request wasn't checking for a null
Content-Type header before checking that it was
application/simple-message-summary.

ASTERISK-27279
Reported by: Ross Beer

Change-Id: Iec2a6c4d2e74af37ff779ecc9fd35644c5c4ea52