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
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
This change brings in PJSIP 2.8, removes all the patches
that were merged upstream, and makes a minor change to
support a breaking change that was done.
ASTERISK-28059
Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189
If Asterisk is built using a DESTDIR this will cause the bundled jansson
to be installed to an unexpected location and we will fail to find it.
Change-Id: Id033e2813261e0d45232383d44c6391122169548
The authors of PJProject undef s_addr because of some issue in Microsoft
Windows. However in Oracle Solaris, s_addr is not a structure member, but
defined to map to the real structure member.
Updates the patch from ASTERISK_20366
ASTERISK-27997
Change-Id: I8223026d4d54e2a46521085fcc94bfa6ebe35b11
The script configure from Teluu expects shared libraries (.so) in a subfolder
called 'lib', when --with-xyz=PATH is specified. However for OpenSSL, the
default location is the root of the source folder = PATH. Furthermore, Asterisk
supports both, 'lib' and root. For consistency and because Asterisk is using
(only) OpenSSL in PJProject, it is enhanced to support both locations, just
like Asterisk.
ASTERISK-27995
Change-Id: I8eb916a88b6b8c22e29bb40bee8faaca6c73406f
The tdata containing the response can be shared by both the dialog
object and the tsx object. In order to prevent the race condition
between the tsx retransmission and the dialog sending a response,
clone the tdata before modifying it for the dialog send response.
ASTERISK-27966 #close
Change-Id: Ic381004a3a212fe1d8eca0e707fe09dba4a6ab4e
Changing any Menuselect option in the `Compiler Flags` section causes a
full rebuild of the Asterisk source tree. Every enabled option causes
a #define to be added to buildopts.h, thus breaking ccache caching for
every source file that includes "asterisk.h". In most cases each option
only applies to one or two files. Now we only define those options for
the specific sources which use them, this causes much better cache
matching when working with multiple builds. For example testing code
with an without MALLOC_DEBUG will now use just over half the ccache
size, only main/astmm.o will have two builds cached instead of every
file.
Reorder main/Makefile so _ASTCFLAGS set on specific object files are all
together, sorted by filename. Stop adding -DMALLOC_DEBUG to CFLAGS of
bundled pjproject, this define is no longer used by any header so only
serves to break cache.
The only code change is a slight adjustment to how main/astmm.c is
initialized. Initialization functions always exist so main/asterisk.c
can call them unconditionally. Additionally rename the astmm
initialization functions so they are not exported.
Change-Id: Ie2085237a964f6e1e6fff55ed046e2afff83c027
Update the bundled jansson Makefile to do nothing during Asterisk
install, use a target that is not phony to initiate the jansson make and
install.
Change-Id: I7643cc3d39af9feba8fc0da676b646efc5f8b3bb
Previously, Asterisk did not tell its bundled PJProject about this configure
parameter. Therefore, PJProject used the platform provided OpenSSL always.
ASTERISK-27880
Change-Id: Iea545aec854dd0e2c061c69bb118a76ce56c5dc6
Asterisk patched the pjproject source to avoid crashing when pjproject
sip_msg headers are encountered with NULL vptr's, but the patch also
output error messages for some valid headers which simply did not need
to be added to the message itself, such as hidden route headers.
pjproject has since applied a similar patch to their baseline to avoid
crashes, but their version also avoids the spurious error logging.
Lets use their patch instead.
ASTERISK-27961 #close
Change-Id: I2ddbd82c8da10e0dcc9807a48089d1f3c2d6e389
Asterisk modules that use PJPROJECT services have their compiler
optimization and possibly their symbolic debug options overridden by the
PJPROJECT configure script selected settings.
* We need to filter-out any -O and -g options in PJ_CFLAGS before echoing
out the result so the PJPROJECT_INCLUDE variable does not override the
Asterisk module settings when using bundled PJPROJECT.
NOTE: This patch only has an effect when using bundled PJPROJECT.
ASTERISK-27563
Change-Id: If124169735ecf572ad1535cd43bff94cb44d5b30
Turn off the periodic sending of CRLNCRLN. Default is on (90 seconds),
which conflicts with the global section's keep_alive_interval option in
pjsip.conf.
patches:
pjsip_keep_not_alive.patch submitted by Alexander Traud (License 6520)
ASTERISK-27347
Change-Id: I6a197f56e1830d3b7e5ec70f17025840a290b057
pjproject by default currently will follow media forked during an INVITE
on outbound calls if the To tag is different on a subsequent response as
that on an earlier response. We handle this correctly. There have
been reported cases where the To tag is the same but we still need to
follow the media. The pjproject patch in this commit adds the
capability to sip_inv and also adds the capability to control it at
runtime. The original "different tag" behavior was always controllable
at runtime but we never did anything with it and left it to default to
TRUE.
So, along with the pjproject patch, this commit adds options to both the
system and endpoint objects to control the two behaviors, and a small
logic change to session_inv_on_media_update in res_pjsip_session to
control the behavior at the endpoint level.
The default behavior for "different tags" remains the same at TRUE and
the default for "same tag" is FALSE.
Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6
ASTERISK-27936
Reported-by: Ross Beer
There have been cases that when the transaction timer callback is called
the tsx is already destroyed. This causes a crash. We now check the
tsx state and return if the tsx is already destroyed.
Change-Id: If93acd5e48d9ca5bb553f2405d5afc836842fe1c
Added a new pj_timer_entry_reset function that resets a timer_entry
for re-use.
Changed direct settings of timer_entry fields to use
pj_timer_entry_init and pj_timer_entry_reset.
Fixed issues where timers were being rescheduled incorrectly.
Change-Id: I5b624bfbc5c1429117484b9b24567293002148e6
There have been some crashes in the past where something attempts
to use a pj_atomic after it's already been destroyed. This patch
tries to prevent it by making sure that pj_atomic_destroy sets
its mutex to NULL when it's done. The pj_mutex functions already check
for a NULL mutex and just return PJ_EINVAL.
Teluu also added some checks to the win32 implementation as well.
Change-Id: Id25f70b79fdedf44ead6e6e1763a4417d3b3f825
./configure --with-pjproject-bundled
did not display an explanation, when no download utility like wget, curl, or
fetch was installed beforehand, although an explanation existed in code. This
happened because the code expected the variable DOWNLOAD_TO_STDOUT to be empty.
However, the script ./configure set that variable always.
Change-Id: I64c99b76a03525c69471e5055bf124b36a51bbd4
This replaces AST_INLINE_API allocators in utils.h with real functions
implemented in astmm.c. Associated macro's are also moved from utils.h
to astmm.h.
Remove menuselect conflicts between MALLOC_DEBUG and DEBUG_CHAOS as they
can now be combined.
This has multiple benefits:
* Simplifies asterisk/utils.h by removing inline functions and use of
the logger.
* Removal of these inline functions decreases size of Asterisk and
module binaries by 1% or more.
* Puts memory management functions together with and without
MALLOC_DEBUG enabled, simplifying management of the code.
* Enables DEBUG_CHAOS for ASTMM_REDIRECT and bundled pjproject.
Change-Id: If9df4377f74bdbb627461b27a473123e05525887
This allows asterisk to be compiled with MALLOC_DEBUG to load modules
built without MALLOC_DEBUG. Now pre-compiled third-party modules will
still work regardless of MALLOC_DEBUG being enabled or not.
Change-Id: Ic07ad80b2c2df894db984cf27b16a69383ce0e10
pjproject's fmtp retrieval function failed to catch invalid fmtp attributes.
Because of this Asterisk would crash if given an SDP with an invalid fmtp
attribute.
When retrieving the format this patch now makes sure the fmtp attribute is
available. If not available it now returns an error status.
ASTERISK-27583 #close
Change-Id: I5cebe000ce2d846cae3af33b6d72c416e51caf2f
pjproject's media format parsing algorithm failed to catch invalid values.
Because of this Asterisk would crash if given an SDP with a invalid media
format description.
When parsing the media format description this patch now properly parses the
value and returns an error status if it can't successfully parse/convert the
value.
ASTERISK-27582 #close
Change-Id: I883b3a4ef85b6972397f7b56bf46c5779c55fdd6
When <http://github.com/BelledonneCommunications/bcg729> is installed, PJProject
tries to link that. Support for this bcg729 was added with PJProject 2.7. The
issue happens, because Teluu enabled that new feature on default.
ASTERISK-27584
Reported by: Stuart Henderson
Change-Id: I88b6b18ad777bcfe2d8201187b4b90eec0a172a6
We still need to figure out how a bad header is getting into the
outgoing message but this patch to pjproject prevents attempting
to print that header and causing a crash.
For several users, this crash happens when sending 183 progress
messages.
ASTERISK-26832
Reported by: Ross Beer, Jan Rozhon
Change-Id: Ie5c5a921c890c843587763e7f33f987dfe66bd16
When an older GCC version is called with a too new warning option, GCC exited
with an error and Asterisk was not built. Therefore, the configure script tests
the installed compiler whether it supports that warning option. If not, Asterisk
does not pass it to the installed compiler. However, some compilers (like clang)
do not exit (error) but give just a warning in such a case. Because the compiler
did not exit, Asterisk passed the unknown-warning option.
ASTERISK-27560
Change-Id: Ia9d148e689c173df4e91699113605dab2de36038
When we fail over to a new target we create a new transaction
and it becomes the current INVITE transaction. This does not
prevent the previous transaction from raising state changes
and causing the session to be prematurely disconnected if a
transport error occurs immediately.
This change backports a fix from PJSIP that eliminates the
incorrect state change and reduces when they would be raised
in the first place.
ASTERISK-27408
Change-Id: Id22d087591782eee31311753d11e7eca4b95ef34
The definition in config_site.h and the argument to the
configure script are not necessary to disable WebRTC
support. The correct argument, --disable-libwebrtc, is
already passed.
ASTERISK-26980
Change-Id: I27da2c894f87914956a72710222e17462d8a44bc