OpenSSL is one of those packages that often have alternatives
with later versions.  For instance, CentOS/EL 7 has an
openssl package at version 1.0.2 but there's an openssl11
package from the epel repository that has 1.1.1.  This gets
installed to /usr/include/openssl11 and /usr/lib64/openssl11.
Unfortunately, the existing --with-ssl and --with-crypto
./configure options expect to point to a source tree and
don't work in this situation.  Also unfortunately, the
checks in ./configure don't use pkg-config.

In order to make this work with the existing situation, you'd
have to run...
./configure --with-ssl=/usr/lib64/openssl11 \
    --with-crypto=/usr/lib64/openssl11 \
    CFLAGS=-I/usr/include/openssl11

BUT...  those options don't get passed down to bundled pjproject
so when you run make, you have to include the CFLAGS again
which is a big pain.

Oh...  To make matters worse, although you can specify
PJPROJECT_CONFIGURE_OPTS on the ./configure command line,
they don't get saved so if you do a make clean, which will
force a re-configure of bundled pjproject, those options
don't get used.

So...

* In configure.ac... Since pkg-config is installed by install_prereq
  anyway, we now use it to check for the system openssl >= 1.1.0.
  If that works, great.  If not, we check for the openssl11
  package. If that works, great.  If not, we fall back to just
  checking for any openssl.  If pkg-config isn't installed for some
  reason, or --with-ssl=<dir> or --with-crypto=<dir> were specified
  on the ./configure command line, we fall back to the existing
  logic that uses AST_EXT_LIB_CHECK().

* The whole OpenSSL check process has been moved up before
  THIRD_PARTY_CONFIGURE(), which does the initial pjproject
  bundled configure, is run.  This way the results of the above
  checks, which may result in new include or library directories,
  is included.

* Although not strictly needed for openssl, We now save the value of
  PJPROJECT_CONFIGURE_OPTS in the makeopts file so it can be used
  again if a re-configure is triggered.

ASTERISK-29693

Change-Id: I341ab7603e6b156aa15a66f43675ac5029d5fbde

An issue was found where a particular manufacturer's phones add a
trailing space to the end of the rtpmap attribute when specifying
a payload type that has a "param" after the format name and clock
rate. For example:

a=rtpmap:120 opus/48000/2 \r\n

Because pjmedia_sdp_attr_get_rtpmap currently takes everything after
the second '/' up to the line end as the param, the space is
included in future comparisons, which then fail if the param being
compared to doesn't also have the space.

We now use pj_scan_get() to parse the param part of rtpmap so
trailing whitespace is automatically stripped.

ASTERISK-29654

Change-Id: Ibd0a4e243a69cde7ba9312275b13ab62ab86bc1b

If an SSL socket parent/listener was destroyed during the handshake,
depending on timing, it was possible for the handling callback to
attempt access of it after the fact thus causing a crash.

ASTERISK-29415 #close

Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d

RFC7616 and RFC8760 allow more than one WWW-Authenticate or
Proxy-Authenticate header per realm, each with different digest
algorithms (including new ones like SHA-256 and SHA-512-256).
Thankfully however a UAS can NOT send back multiple Authenticate
headers for the same realm with the same digest algorithm.  The
UAS is also supposed to send the headers in order of preference
with the first one being the most preferred.  We're supposed to
send an Authorization header for the first one we encounter for a
realm that we can support.

The UAS can also send multiple realms, especially when it's a
proxy that has forked the request in which case the proxy will
aggregate all of the Authenticate headers and then send them all
back to the UAC.

It doesn't stop there though... Each realm can require a
different username from the others.  There's also nothing
preventing each digest algorithm from having a unique password
although I'm not sure if that adds any benefit.

So now... For each Authenticate header we encounter, we have to
determine if we support the digest algorithm and, if not, just
skip the header.  We then have to find an auth object that
matches the realm AND the digest algorithm or find a wildcard
object that matches the digest algorithm. If we find one, we add
it to the results vector and read the next Authenticate header.
If the next header is for the same realm AND we already added an
auth object for that realm, we skip the header. Otherwise we
repeat the process for the next header.

In the end, we'll have accumulated a list of credentials we can
pass to pjproject that it can use to add Authentication headers
to a request.

NOTE: Neither we nor pjproject can currently handle digest
algorithms other than MD5.  We don't even have a place for it in
the ast_sip_auth object. For this reason, we just skip processing
any Authenticate header that's not MD5.  When we support the
others, we'll move the check into the loop that searches the
objects.

Changes:

 * Added a new API ast_sip_retrieve_auths_vector() that takes in
   a vector of auth ids (usually supplied on a call to
   ast_sip_create_request_with_auth()) and populates another
   vector with the actual objects.

 * Refactored res_pjsip_outbound_authenticator_digest to handle
   multiple Authenticate headers and set the stage for handling
   additional digest algorithms.

 * Added a pjproject patch that allows them to ignore digest
   algorithms they don't support.  This patch has already been
   merged upstream.

 * Updated documentation for auth objects in the XML and
   in pjsip.conf.sample.

 * Although res_pjsip_authenticator_digest isn't affected
   by this change, some debugging and a testsuite AMI event
   was added to facilitate testing.

Discovered during OpenSIPit 2021.

ASTERISK-29397

Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281

In some cases it was possible for a STUN packet to be destroyed
prematurely or even destroyed partially multiple times.

This patch provided by Teluu fixes the lifetime of these
packets and ensures they aren't partially destroyed multiple
times.

https://github.com/pjsip/pjproject/pull/2709

ASTERISK-29377

Change-Id: Ie842ad24ddf345e01c69a4d333023f05f787abca

If a remote side is broken and sends an SDP that can not be
negotiated the call will be torn down but there is a window
where a second 183 Session Progress or 200 OK that is forked
can be received that also attempts to negotiate SDP. Since
the code marked the SDP negotiation as being done and complete
prior to this it assumes that there is an active local and remote
SDP which it can modify, while in fact there is not as the SDP
did not successfully negotiate. Since there is no local or remote
SDP a crash occurs.

This patch changes the pjmedia_sdp_neg_modify_local_offer2
function to no longer assume that a previous SDP negotiation
was successful.

ASTERISK-29196

Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e

In rewrite_uri asterisk was not making deep copies of strings when
changing the uri. This was in some cases causing garbage in the route
header and in other cases even crashing asterisk when receiving a
message with a record-route header set. Thanks to Ralf Kubis for
pointing out why this happens. A similar problem was found in
res_pjsip_transport_websocket.c. Pjproject needs as well to be patched
to avoid garbage in CANCEL messages.

ASTERISK-29024 #close

Change-Id: Ic5acd7fa2fbda3080f5f36ef12e46804939b198b

PJSIP, UDP transport with external_media_address and session timers
enabled. Connected to SIP server that is not in local net. Asterisk
initiated the connection and is refreshing the session after 150s
(timeout 300s). The 2nd refresh-INVITE triggered by the pjsip timer has
a malformed IP address in its SDP (garbage string). This only happens
when the SDP is modified by the nat-code to replace the local IP address
with the configured external_media_address.
Analysis: the code to modify the SDP (in
res_pjsip_session.c:session_outgoing_nat_hook() and also (redundantly?)
in res_pjsip_sdp_rtp.c:change_outgoing_sdp_stream_media_address()) uses
the tdata->pool to allocate the replacement string. But the same
pjmedia_sdp_stream that was modified for the 1st refresh-INVITE is also
used for the 2nd refresh-INVITE (because it is stored in pjmedia's
pjmedia_sdp_neg structure). The problem is, that at that moment, the
tdata->pool that holds the stringified external_media_address from the
1. refresh-INVITE has long been reused for something else.
Fix by Sauw Ming of pjproject (see
https://github.com/pjsip/pjproject/pull/2476): the local, potentially
modified pjmedia_sdp_stream is cloned in
pjproject/source/pjsip/src/pjmedia/sip_neg.c:process_answer() and the
clone is stored, thereby detaching from the tdata->pool (which is only
released *after* process_answer())

ASTERISK-28973
Reported-by: Michael Neuhauser

Change-Id: I272ac22436076596e06aa51b9fa23fd1c7734a0e

When dealing with a lot of video streams on WebRTC
the resulting SDPs can grow to be quite large. This
effectively doubles the maximum size to allow more
streams to exist.

The res_http_websocket module has also been changed
to use a buffer on the session for reading in packets
to ensure that the stack space usage is not excessive.

Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01

I noticed this while looking at another issue and brought
it up with Teluu. It was possible for an uninitialized timer
to be cancelled, resulting in the invalid timer id of 0
being placed into the timer heap causing issues.

This change is a backport from the pjproject repository
preventing this from happening.

Change-Id: I1ba318b1f153a6dd7458846396e2867282b428e7

This patch makes the usual necessary changes when upgrading to a new
version pjproject. For instance, version number bump, patches removed
from third-party, new *.md5 file added, etc..

This patch also includes a change to the Asterisk pjproject Makefile to
explicitly create the 'source/pjsip-apps/lib' directory. This directory
is no longer there by default so needs to be added so the Asterisk
malloc debug can be built.

This patch also includes some minor changes to Asterisk that were a result
of the upgrade. Specifically, there was a backward incompatibility change
made in 2.10 that modified the "expires header" variable field from a
signed to an unsigned value. This potentially effects comparison. Namely,
those check for a value less than zero. This patch modified a few locations
in the Asterisk code that may have been affected.

Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to
check a minimum version of pjproject at compile time.

ASTERISK-28899 #close

Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81

Pjproject makefiles miss some dependencies which can cause race
conditions when building with parallel make processes. This patch
adds such dependencies correctly.

ASTERISK-28879 #close
Reported-by: Dmitry Wagin <dmitry.wagin@ya.ru>

Change-Id: Ie1b0dc365dafe4a84c5248097fe8d73804043c22

The configure.m4 script for pjproject contains some += syntax, which
is specific to bash, replacing it with string substitutions makes
the script compatible with traditional Bourne shells.

ASTERISK-28866 #close
Reported-by: Christoph Moench-Tegeder <cmt@FreeBSD.org>

Change-Id: I382a78160e028044598b7da83ec7e1ff42b91c05

In practice it has been seen that some users come
close to our maximum ICE candidate count of 32.
In case people have gone over this increases the
count to 64, giving ample room.

ASTERISK-28859

Change-Id: I35cd68948ec0ada86c14eb53092cdaf8b62996cf

ASTERISK-28837

Change-Id: Id057324912a3cfe6f50af372675626bb515907d9

ASTERISK-28758
Reported by: Patrick Wakano
Reported by: Dmitriy Serov

Change-Id: Ifb6b85c559d116739af00bc48d1f547caa85efac

Change-Id: Iebf7687613aa0295ea3c82256460b337f1595be2

When handling ICE negotiations, it's possible that there can be a delay
between STUN binding requests which in turn will cause a delay in ICE
completion, preventing media from flowing. It should be possible to send
media when there is at least one valid pair, preventing this scenario
from occurring.

A change was added to PJPROJECT that adds an optional callback
(on_valid_pair) that will be called when the first valid pair is found
during ICE negotiation. Asterisk uses this to start the DTLS handshake,
allowing media to flow. It will only be called once, either on the first
valid pair, or when ICE negotiation is complete.

ASTERISK-28716

Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867

ASTERISK-26955 #close
Reported by: Peter Sokolov

Change-Id: Ib2803640905a77b65d0cee2d0ed2c7b310d470ac

Issues in pjproject 2.9 caused us to revert some of their changes
as a work around.  This introduced another issue where pjproject
wouldn't build with older gcc versions such as that found on
CentOS 6.  This commit replaces the reverts with the official
fixes for the original issues and allows pjproject to be built
on CentOS 6 again.

ASTERISK-28574
Reported-by: Niklas Larsson

Change-Id: I06f8507bea553d1a01b0b8874197d35b9d47ec4c

We've found a connection re-use regression in pjproject 2.9
introduced by commit
"Close #1019: Support for multiple listeners."
https://trac.pjsip.org/repos/changeset/6002
https://trac.pjsip.org/repos/ticket/1019

Normally, multiple SSL requests should reuse the same connection
if one already exists to the remote server.  When a transport
error occurs, the next request should establish a new connection
and any following requests should use that same one.  With this
patch, when a transport error occurs, every new request creates
a new connection so you can wind up with thousands of open tcp
sockets, possibly exhausting file handles, and increasing memory
usage.

Reverting pjproject commit 6002 (and related 6021) restores the
expected behavior.

We also found a memory leak in SSL processing that was introduced by
commit
"Fixed #2204: Add OpenSSL remote certificate chain info"
https://trac.pjsip.org/repos/changeset/6014
https://trac.pjsip.org/repos/ticket/2204

Apparently the remote certificate chain is continually recreated
causing the leak.

Reverting pjproject commit 6014 (and related 6022) restores the
expected behavior.

Both of these issues have been acknowledged by Teluu.

ASTERISK-28521

Change-Id: I8ae7233c3ac4ec29a3b991f738e655dabcaba9f1

NEC SIP Station interface with authenticated registration only supports cnonce
up to 32 characters.  In Linux, PJSIP would generate 36 character cnonce
which included hyphens.  Teluu developed this patch adding a compile time
setting to default to not include the hyphens.  They felt it best to still
generate the UUID and strip the hyphens.
They have indicated it will be part of PJSIP 2.10.

ASTERISK-28509
Reported-by: Dan Cropp

Change-Id: Ibdfcf845d4f8c0a14df09fd983b11f2d72c5f470

Most SSL/TLS error messages coming from pjproject now have either
the peer address:port or peer hostname, depending on what was
available at the time and code location where the error was
generated.

ASTERISK-28444
Reported by: Bernhard Schmidt

Change-Id: I41770e8a1ea5e96f6e16b236692c4269ce1ba91e

Relies on https://github.com/asterisk/third-party/pull/4

Change-Id: Iec9cad42cb4ae109a86a3d4dae61e8bce4424ce3

This patch fixes three compatibility issues for Darwin compatible builds:

(1) Use BSD compatible command line option for sed

For some versions of BSD sed, the -r command line option is unknown.
Both GNU and BSD sed support the -E command line option for enabling
extended regular expressions; as such, this patch replaces the -r
option with -E.

(2) Look for '_' in pjproject generated symbols

In Darwin comaptible systems, the symbols generated for pjproject may be
prefixed with an '_'. When exporting these to a symbol file, the invocation
to sed has to optionally look for a prefix of said '_' character.

(3) Use -all_load/-noall_load when linking

The flags -whole-archive/-no-whole-archive are not supported by the
linker, and must instead be replaced with -all_load/-noall_load.

Change-Id: I58121756de6a0560a6e49ca9d6bf9566a333cde3

Fixed #2191:
  - Stricter double timer entry scheduling prevention.
  - Integrate group lock in SIP transport, e.g: for add/dec ref,
    for timer scheduling.

ASTERISK-28161
Reported-by: Ross Beer

Change-Id: I2e09aa66de0dda9414d8a8259a649c4d2d96a9f5

When the gcc version is >= 8.2.1, we were already setting the
--fno-partial-inlining flag for Asterisk source files to get around
a gcc bug but we weren't passing the flag down to the bundled
builds of pjproject and jansson.

ASTERISK-28392

Change-Id: I99ede9bc35408ecd096f7d5369e8192d3dc75704

ASTERISK-28161 #close
Reported by: Ross Beer

Change-Id: I65331d554695753005eaa66c1d5d4807fe9009c8

On OpenSuse Leap, libjansson.a is installed in
third-party/jansson/dest/lib64 instead of lib (which is where
the top-level makeopts looks).  This causes a link failure.

* Updated jansson/Makefile to add an explicit --libdir to force
  the installation to third-party/jansson/dest/lib.

ASTERISK-28271
Reported by: David Wilcox

Change-Id: Ibf8af75e5da13562105fcc39ed898c6ef0b5a5f3

Bundled pjproject and jansson must be configured with the host and build
parameters provided to the configure script.
Autotools do not permit to check for the existence of local header files, so
the control of hrirs.h must not be done when cross-compiling.

ASTERISK-28250

Change-Id: If0a76e52a87d4ab82b7d4c72d27d8759ca931880

Fixed #2172: Avoid double reference counter decrements in
timer in the scenario of race condition between
pj_timer_heap_cancel() and pj_timer_heap_poll().

Change-Id: If000e9438c83ac5084b678eb811e902c035bd2d8

In ASTERISK-27095 an issue had been fixed because of which chan_pjsip was not
trying to send UPDATE messages when connected_line_method was set to invite.
However this only solved the issue for incoming INVITES. For outgoing INVITES
(important when transferring calls) the options variable needs to be updated
at a different place.

ASTERISK-28182 #close
Reported-by: nappsoft

Change-Id: I76cc06da4ca76ddd6dce814a8b97cc66b98aaf29

This brings in jansson-2.12, removes all patches that were merged
upstream.  README is created in third-party/jansson/patches to explain
how to add patches but also because the patches folder must exist for
the build process to succeed.

Change-Id: If0f2d541c50997690660c21fb7b03d625a5cdadd

We previously allowed resample and g711 codecs to be built when
TEST_FRAMEWORK was enabled.  This could cause errors if the testsuite
was run without this option enabled.  Switch the build system to allow
those codecs to be built when --enable-dev-mode is used.  This removes a
chance for strange testsuite errors from use of an inadequate pjsua
binary.

Change-Id: Iee8a3613cdb711fa7e7d217c5a775a575907ae22

pack_string crashed on non-NULL strings returned when s->has_error was
true if the string was the result of 's' format without '#', '%' or '+'.

Change-Id: Ic125df691d81ba2cbc413e37bdae657b304d20d0

This change implements a few different generic things which were brought
on by Google Voice SIP.

1.  The concept of flow transports have been introduced.  These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target.  These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity).  When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.

2.  Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.

3.  P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module.  If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.

4.  Configurable outbound extension support has been added to the outbound
registration module.  When set the extension will be placed in the
Supported header.

5.  Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.

6.  Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.

All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.

ASTERISK-27971 #close

Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58

This patch is not in the upstream pjproject and does unsafe things with
the timer->_timer_id and timer->_grp_lock values in pj_timer_entry_reset()
outside of the timer heap lock.  pj_timer_entry_reset() is also called for
timers that are not about to be rescheduled in a few places.

Change-Id: I4fe0b4bc648f7be5903cf4531b94fc87275713c1

Change-Id: I045e420d5e73e60639079246e810da6ae21ae22b

Use json_vsprintf from versions which contain fix for va_copy leak.

Apply fixes from jansson master:
* va_copy leak fix.
* Avoid potential invalid memory read in json_pack.
* Rename variable that shadowed another.

Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539

When writing an RTCP report to json the code attempts to pack the "ssrc" and
"source_ssrc" unsigned integer values as a signed int value type. This of course
means if the ssrc's unsigned value is greater than that which can fit into a
signed integer value it gets converted to a negative number. Subsequently, the
negative value goes out in the json report.

This patch now packs the value as a json_int_t, which is the widest integer type
available on a given system. This should make it so the value no longer
overflows.

Note, this was caught by two failing tests hep/rtcp-receiver/ and
hep/rtcp-sender.

Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0