Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.

No functional changes were made with this commit.

Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c

When using the PSJIP_MEDIA_OFFER dialplan function it was not
overriding an endpoint's configured codecs on refresh unless
they had a shared codec between the two.

This patch makes it so whatever is set using PJSIP_MEDIA_OFFER
is used when creating the SDP for a refresh no matter what.

ASTERISK-28878 #close

Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6

The Streams API becomes the home for the core ACN capabilities.
These include...

 * Parsing and formatting of codec negotation preferences.
 * Resolving pending streams and topologies with those configured
   using configured preferences.
 * Utility functions for creating string representations of
   streams, topologies, and negotiation preferences.

For codec negotiation preferences:
 * Added ast_stream_codec_prefs_parse() which takes a string
   representation of codec negotiation preferences, which
   may come from a pjsip endpoint for example, and populates
   a ast_stream_codec_negotiation_prefs structure.
 * Added ast_stream_codec_prefs_to_str() which does the reverse.
 * Added many functions to parse individual parameter name
   and value strings to their respectrive enum values, and the
   reverse.

For streams:
 * Added ast_stream_create_resolved() which takes a "live" stream
   and resolves it with a configured stream and the negotiation
   preferences to create a new stream.
 * Added ast_stream_to_str() which create a string representation
   of a stream suitable for debug or display purposes.

For topology:
 * Added ast_stream_topology_create_resolved() which takes a "live"
   topology and resolves it, stream by stream, with a configured
   topology stream and the negotiation preferences to create a new
   topology.
 * Added ast_stream_topology_to_str() which create a string
   representation of a topology suitable for debug or display
   purposes.
 * Renamed ast_format_caps_from_topology() to
   ast_stream_topology_get_formats() to be more consistent with
   the existing ast_stream_get_formats().

Additional changes:
 * A new function ast_format_cap_append_names() appends the results
   to the ast_str buffer instead of replacing buffer contents.

Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56

chan_sip handle_response() function, for a 400 response to an INVITE,
calls handle_response_invite() and does not generate ACK.
handle_response_invite() does not recognize 400 response and has no
default response processing for unexpected responses, thus it does not
generate ACK either.
The ACK on response repetition comes from handle_response() mechanism
"We must re-send ACKs to re-transmitted final responses".

According to code history, 400 response specific processing was
introduced with commit
"channels/chan_sip: Add improved support for 4xx error codes"
This commit added support for :
- 400/414/493 in handle_response_subscribe() handle_response_register()
  and handle_response().
- 414/493 only in handle_response_invite().

This fix adds 400 response support in handle_response_invite().

ASTERISK-28957

Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad

A patch made a reference to the PJSIP_SC_NULL enumeration value, which
was added to pjproject 2.8 and above thus making it so Asterisk would
fail to compile with prior versions of pjproject.

This patch removes the reference, and instead initializes the value
to '0'.

ASTERISK-28886 #close

Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7

The change to how setvar works for various channels performed in
ASTERISK~23756 missed some required change in the dahdi channel,
where the variables are actually set while reading configuration.
This change should fix the issue.

ASTERISK-28955

Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274

When fax_gateway_framehook is called and a gateway hasn't already
been started, the framehook gets the t38 state for both the current
channel and the peer.  That call trickles down to the channel
driver which determines the state.  If either channel is hung up
(or in the process of being hung up), the channel driver's tech_pvt
is going to be NULL which, in the case of chan_pjsip, will cause a
segfault.

* Added a hangup check for both the channel and peer channel
  before starting a fax gateway.

* Added a check for NULL tech_pvt to chan_pjsip_queryoption
  so we don't attempt to reference a tech_pvt that's already
  gone.

ASTERISK-28923
Reported by: Yury Kirsanov

Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c

Some places in Asterisk did not treat the formats on a stream
as immutable when they are.

The ast_stream_get_formats function is now const to enforce this
and parts of Asterisk have been updated to take this into account.
Some violations of this were also fixed along the way.

An additional minor tweak is that streams are now allocated with
an empty format capabilities structure removing the need in various
places to check that one is present on the stream.

ASTERISK-28846

Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe

ASTERISK-28372
Reported by: Anton Satskiy

ASTERISK-24428
Reported by: sstream

Change-Id: I2b7432a9bf3b09dc8515297ff955636db7a6224c

ASTERISK-27195
Reported by: Joshua Roys

Change-Id: I6e72ecb874200dec7a3865c7babaf5ac0d3101de

It is possible to configure a TCP/TLS client without having a TCP/TLS
server. In that case, no error or warning was printed but the headers
Contact and Via in SIP REGISTER were "(null)".

ASTERISK-28798

Change-Id: I387ca5cb6a65f1eb675a29c5e41df8ec6c242ab2

If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a
digit begin before media, or rtp has been setup then it's possible the
outgoing channel will hear a constant DTMF tone upon answering.

This happens because when there is no media, or rtp chan_pjsip notifies
the core to initiate inband DTMF. However, upon digit end if media, and
rtp become available then chan_pjsip does not notify the core to stop
inband DTMF. Thus the tone continues playing.

This patch makes it so chan_pjsip only notifies the core to start
inband DTMF in only the required cases. Now if there is no media, or
rtp availabe upon digit begin chan_pjsip does nothing, but tells the
core it handled it.

ASTERISK-28817 #close

Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5

ASTERISK-28803

Change-Id: I15449621b68d0ad4d57b7c337c1167adb15135af

Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2

Do not hang up a PJSIP channel on RTP timeout if that channel is in
a direct-media bridge. Also reset the time of the last received RTP packet when
direct-media ends (wait full rtp_timeout period before checking first time after
audio came back to Asterisk).

ASTERISK-28774
Reported-by: Michael Neuhauser

Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1

Fixes the following compile error:

    chan_vpb.cc:2688:26: error: catching polymorphic type
        ‘class std::exception’ by value

Change-Id: Ic87bc357d72427d77626735c83200fd278a7a649

If the SSRC of a received RTP packet differed from the previous SSRC
an SSRC change control frame would be queued ahead of the media
frame. In the case of audio this would result in the format of the
audio frame not being checked, and if it differed or was not allowed
then it could cause the call to drop due to failure to set up a
translation path.

The chan_pjsip module will now no longer assume the first frame
will be the audio frame and instead goes through the complete list
to find it.

ASTERISK-28759

Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec

If you're for some reason out of RTP ports, chan_sip would previously
responde to an INVITE with a 403, which will fail the call.

Now, it returns a 503, allowing the device/proxy to retry the call on a
different machine.

ASTERISK-28718

Change-Id: I968dcf6c1e30ecddcce397dcda36db727c83ca90

Fixes no-audio issues when the media source is changed and
strictrtp is enabled (default).

If the peer media source changes, the SDP session version also changes.
If it is lower than the one we had stored, chan_sip would ignore it.

This changeset keeps track of the remote media origin identifier,
comparing that as well. If it changes, the session version needn't be
higher for us to accept the SDP.

Common scenario where this would've caused problems: a separate media
gateway that informs the caller about premium rates before handing off
the call to the final destination.

(An alternative fix would be to set ignoresdpversion=yes on the peer.)

ASTERISK-28686

Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee

If chan_pjsip receives an RTP packet whose payload differs from the
channel's native format, and asymmetric_rtp_codec is disabled (the
default), Asterisk will switch the channel's native format to match
that of the incoming packet without regard to the negotiated payloads.

We now check that the received frame is in a format we have negotiated
before switching payloads which results in these packets being dropped
instead of causing the session to terminate.

ASTERISK-28139 #close
Reported by: Paul Brooks

Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3

The no-entry timeout set to 999999 == 16⅔ minutes, change to INT_MAX
to match behavior of "no timeout" defined in comment.

ASTERISK-28702 #close

Change-Id: I4ea015986e061374385dba247b272f7aac60bf11

lws2sws() does not stop trying to handle header continuation lines
even after all headers have been found. This is problematic if the
first character of a SIP message body is a space or tab character, so
we update to recognize the end of the message header.

ASTERISK-28693 #close
Reported by: Frank Matano

Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df

This commit adds support for
[AudioSocket](
https://wiki.asterisk.org/wiki/display/AST/AudioSocket),
a very simple bidirectional audio streaming protocol. There are both
channel and application interfaces.

A description of the protocol can be found on the above referenced
GitHub page.  A short talk about the reasons and implementation can be
found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from
CommCon 2019.

ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.

ASTERISK-28484 #close

Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5

The change to add setting hangupsource to sig_pri_queue_hangup()
made in https://gerrit.asterisk.org/c/asterisk/+/12857 casued
deadlocks when a hangup request was received from the core at the
same time a hanguprequest was received from the remote end via the
D channel.

Although the PRI's channel private structure was being unlocked
before setting the hangupsource, the PRI's own lock was still being
held during the process.  If channel actions were also coming from
the core, a deadlock on the PRI could result.  This deadlock could
then escalate to the entire DAHDI subsystem via DAHDI's global
interface list lock, especially if someone used the PRI CLI commands.

Fix:

* We now unlock the PRI as well as the PRI's channel private
  structure before setting the hangupsource, then relock both
  afterwards.

ASTERISK-28605
Reported by: Dirk Wendland

Change-Id: Id74aaa5d4e3746063dbe9deed188eb65193cb9c9

chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to
213.150.203.60:1492 returned -2: Interrupted system call

returned -2 implies this wasn't actually an OS error, so errno makes no
sense either.  Internal error was already logged higher up, and -2
generally means that either there isn't a valid connection available, or
the pipe notification failed, and that is already correctly logged.

ASTERISK-28651 #close

Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

The SIP transaction state was reset when emitting an UPDATE or a re-INVITE
related to a COLP, preventing RTP packets to be emitted.

ASTERISK-28647

Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b

During capabilities selection (joint capabilities of us and peer,
configured capability for this peer, or general configured
capabilities), if sip_new() does not keep framing information,
then directmedia activation will fail for any framing different
from default framing.

ASTERISK-28637

Change-Id: I99257502788653c2816fc991cac7946453082466

If the name of a peer is known and a SIP request is sent using that
peer's name, the address of the peer will change even if the request
fails the authentication challenge. This means that an endpoint can
be altered and even rendered unusuable, even if it was in a working
state previously. This can only occur when the nat option is set to the
default, or auto_force_rport.

This change checks the result of authentication first to ensure it is
successful before setting the address and the nat option.

ASTERISK-28589 #close

Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df

Upon a short PRI disconnection, libpri may maintain Q.921 layer 'up' and
may thus not send PRI_EVENT_DCHAN_DOWN / PRI_EVENT_DCHAN_UP events.
If pri_event_alarm() clears DCHAN_UP status bit upon alarm detection
and no Q.921 reconnection sequence occurs, chan_dahdi will keep
seeing span status "Down" at the end of alarm.

This patch modifies pri_event_alarm() in order to keep DCHAN_UP bit
unchanged. libpri will send a PRI_EVENT_DCHAN_DOWN event if it detects
a disconnection of Q.921 layer and this will clear DCHAN_UP if required.

ASTERISK-28615

Change-Id: Ibe27df4971fd4c82cc6850020bce4a8b2692c996

This patch fixes several issues reported by the lgtm code analysis tool:

https://lgtm.com/projects/g/asterisk/asterisk

Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:

* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards

Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25

* Pass caller information to frame allocation functions.
* Disable caching as it interfers with MALLOC_DEBUG reporting.
* Stop using ast_calloc_cache.

Change-Id: Id343cd80a3db941d2daefde2a060750fea8cd260

During execution "pjsip show channelstats" cli command by an
external module asterisk crashed. It seems this is a separate
thread running to fetch and print rtp stats. The crash happened on
the ao2_lock method, just before it going to read the rtp stats on
a rtp instance. According to gdb backtrace log, it seems the
session media was already cleaned up at that moment.

ASTERISK-28578

Change-Id: I3e05980dd4694577be6d39be2c21a5736bae3c6f

ASTERISK-28086 #close

Change-Id: Ib3baadc89b9f0477a6f25a63861433812368c5ea

Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows
the on-hold behavior to be controlled on a per-call basis

ASTERISK-28542 #close

Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8

This change adds H.265/HEVC as a known codec and creates a cached
"h265" media format for use.

Note that RFC 7798 section 7.2 also describes additional SDP
parameters. Handling of these is not yet supported.

ASTERISK-28512

Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2

On FreeBSD using the clang/llvm compiler build fails to build due
to the switch statement argument being a non integer type expression.
Switch to an if/else if/else construct to sidestep the issue.

ASTERISK-28536 #close

Change-Id: Idf4a82cc1e94580a2d017fe9e351c226f23e20c8

When fax detection occurs on an outbound PJSIP channel the
redirect operation will result in a masquerade occurring and
the underlying channel on the session changing. The code
incorrectly relocked the new channel instead of the old
channel when returning. This resulted in the new channel
being locked indefinitely. The code now always acts on the
expected channel.

ASTERISK-28538

Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3

When modifying an already defined variable in some channel drivers they
add a new variable with the same name to the list, but that value is
never used, only the first one found.

Introduce ast_variable_list_replace() and use it where appropriate.

ASTERISK-23756 #close
Patches:
  setvar-multiplie.patch submitted by Michael Goryainov

Change-Id: Ie1897a96c82b8945e752733612ee963686f32839

When the remote ISDN party ends an ISDN call on a PRI link
(DISCONNECT), CHANNEL(hangupsource) information is not available.

chan_dahdi already contains an ast_set_hangupsource() in
__dahdi_exception() function but it seems that ISDN message processing
does not use this part of code.

Two other channel modules associate ast_queue_hangup() and
ast_set_hangupsource() functions calls:
- chan_pjsip in chan_pjsip_session_end() function,
- chan_sip in sip_queue_hangup_cause() function.
chan_iax2 separates them, in iax2_queue_hangup()/iax2_destroy() and
set_hangup_source_and_cause().

Thus, I propose to add ast_set_hangupsource() beside
ast_queue_hangup() in sig_pri_queue_hangup(), like chan_pjsip and
chan_sip already do.

ASTERISK-28525

Change-Id: I0f588a4bcf15ccd0648fd69830d1b801c3f21b7c

The links in the deprecation notice were the shortened
variety but it makes better sense to show the unshortened
links as they're more descriptive.

I.E.
wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
rather than
wiki.asterisk.org/wiki/x/tAHOAQ

Change-Id: If2da5d5243e2d4a6f193b15691d23e7e5a7c57a9