This change adds a property to RTP instances to indicate that
REMB support is enabled and that sending/receiving should be
passed through.
This also enables it on video RTP instances in PJSIP if
WebRTC support is enabled.
Finally the goog-remb extension is added to the SDP using
the rtcp-fb attribute to indicate our support for it.
Details about REMB can be found on the draft document for it:
https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
Change-Id: I1902dda1c0882bd1a0d71b2f120684b44b97e789
Asterisk does not need the development package of libltdl, because it does not
use any symbol of -lltdl directly. Instead, it uses the runtime package via the
shared library -lodbc. On the supported platforms, that shared library declares
its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have
failed.
ASTERISK-27745
Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba
* acl (named_acl.c)
* cdr
* cel
* ccss
* dnsmgr
* dsp
* enum
* extconfig (config.c)
* features
* http
* indications
* logger
* manager
* plc
* sounds
* udptl
These modules are now loaded at appropriate time by the module loader.
Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so
the module loader will abort startup on failure of these modules.
Some of these modules are still initialized or shutdown from outside the
module loader. logger.c is initialized very early and shutdown very
late, manager.c is initialized by the module loader but is shutdown by
the Asterisk core (too much uses it without holding references).
Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f
Since ASTERISK-27253, no symbols from the header srtp2/crypto_types.h are used
anymore. Therefore, its include statement can be removed. This allows to compile
Asterisk on platforms which do not offer this private header, like openSUSE.
ASTERISK-27733
Change-Id: I25c5cb8fa966043d1506ebef449e5a724412b4b6
The menuselect comment was updated to deprecate these modules but the
AST_MODULE_INFO block at the end of file was missed.
ASTERISK-27671
Change-Id: I63070b5c4d4f08af010c6034acd4793c1bcef839
Checking option_debug directly is incorrect as it ignores file/module
specific debug settings. This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.
Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0
In handle_negotiated_sdp(), use session->active_media_state when
session->pending_media_state is empty. The 200's SDP should be fed into
handle_negotiated_sdp_session_media() together with the already negotiated
state, which is now in session->active_media_state instead. Only if both
the session's pending and active media are empty should
handle_negotiated_sdp() abort.
ASTERISK-27441
Change-Id: If0d5150ffe6f38d8a854831fef37942258d4629c
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
The pool cache gets in the way of finding use after free errors of memory
pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a
pool is released because it gets put into the cache instead of being
freed.
* Added the "cache_pools" option to pjproject.conf. Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the pool
contents are used after free and who freed it.
To disable the pool caching simply disable the cache_pools option in
pjproject.conf and restart Asterisk.
Sample pjproject.conf setting:
[startup]
cache_pools=no
* Made current users of the caching pool factory initialization and
destruction calls call common routines to create and destroy cached pools.
ASTERISK-27704
Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828
A couple of additional properties are needed in rtp_engine to enable
support for packet retransmission: AST_RTP_PROPERTY_RETRANS_RECV and
AST_RTP_PROPERTY_RETRANS_SEND. These will both be enabled automatically
if an endpoint has the webrtc option enabled. While this adds no
functionality currently, it will serve as a building block for future
changes for RTP retransmission support.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc
The transferrer's session channel was destroyed by the transferrer's
serializer thread in a race condition with the transfer target's
serializer thread during an attended transfer. The transfer target's
serializer was attempting to clean up a deferred end status on behalf of
the transferrer's channel when it should have passed the action to the
transferrer's serializer. When the transfer target's serializer lost the
race then both threads wind up trying to end the transferrer's session.
* Push the ast_sip_session_end_if_deferred() call onto the transferrer's
serializer to avoid a race condition that results in a crash. The
session_end() function that could be called by
ast_sip_session_end_if_deferred() really must be executed by the
transferrer's serializer to avoid this kind of crash.
ASTERISK-27568
Change-Id: Iacda724e7cb24d7520e49b2fd7e504aa398d7238
In ast_websocket_read() we were not adequately checking that the
payload_len was non-zero before passing it to ws_safe_read(). Calling
ws_safe_read with a len argument of 0 will result in a busy loop until
the underlying socket is closed.
ASTERISK-27658 #close
Change-Id: I9d59f83bc563f711df1a6197c57de473f6b0663a
Since res_pjsip_transport_management provides several attack
mitigation features, its functionality moved to res_pjsip and
this module has been removed. This way the features will always
be available if res_pjsip is loaded.
ASTERISK-27618
Reported By: Sandro Gauci
Change-Id: I21a2d33d9dda001452ea040d350d7a075f9acf0d
pjsip_distributor:
authenticate() creates a tdata and uses it to send a challenge or
failure response. When pjsip_endpt_send_response2() succeeds, it
automatically decrements the tdata ref count but when it fails, it
doesn't. Since we weren't checking for a return status, we weren't
decrementing the count ourselves on error and were therefore leaking
tdatas.
res_pjsip_session:
session_reinvite_on_rx_request wasn't decrementing the ref count
if an error happened while sending a 491 response.
pre_session_setup wasn't decrementing the ref count if
while sending an error after a pjsip_inv_verify_request failure.
res_pjsip:
ast_sip_send_response wasn't decrementing the ref count on error.
ASTERISK-27618
Reported By: Sandro Gauci
Change-Id: Iab33a6c7b6fba96148ed465b690ba8534ac961bf
It was discovered that there are some corner cases where a pjsip tsx
might have no last_tx so calling ast_sip_failover_request with
a NULL last_tx as its tdata would cause a crash.
ASTERISK-27618
Reported By: Sandro Gauci
Change-Id: Ic2b63f6d4ae617c4c19dcdec2a7a6156b54fd15b
When receiving a SUBSCRIBE request the Accept headers from it are
stored locally. This operation has a fixed limit of 32 Accept headers
but this limit was not enforced. As a result it was possible for
memory outside of the allocated space to get written to resulting
in a crash.
This change enforces the limit so only 32 Accept headers are
processed.
ASTERISK-27640
Reported By: Sandro Gauci
Change-Id: I99a814b10b554b13a6021ccf41111e5bc95e7301
If the ICE role is not set right away, we might have a role conflict
that stays undetected and ICE finishing with successful tests and no
candidate nominated. This was introduced by ASTERISK-27088.
To avoid this, we set the role as soon as before but only if the ICE
state permits it: still checking and not yet nominating candidates or
completed.
ASTERISK-27646
Change-Id: I5dbc69ad63cacbb067922850fbb113d479bd729c
* Prefer strcasecmp() over stricmp()
* Use a list with no lock since we never actually lock
* Minor cleanups to error messages
Change-Id: I8446f44795ee8f3072e1c1f9193c6912dfc0c42b
There is a dedicated slot in the pjsip_sip_uri for the 'user'
parameter, so use that instead of adding to the list of generic URI
parameters.
Change-Id: I0a0ce8a60ecee27489735bf56fd707719d8c2ed6
* app_fax (replaced by res_fax).
* res_config_sqlite (replaced by res_config_sqlite3).
* res_monitor (replaced by app_mixmonitor).
This is related to ASTERISK~23657 but does not resolve that ticket.
Resolving that ticket would require complete removal of res_monitor.
ASTERISK-27671 #close
Change-Id: I16a3edd61fc1abd4a7b2e9357693ed663f62dd49
res_pjsip_endpoint_identifier_user.c:
* Fix copy/paste error in find_endpoint(). We were using a constant
"anonymous" string instead of the passed in endpoint_name when checking
the transport domain for an endpoint match.
* Eliminate RAII_VAR in find_endpoint().
* Remove always true check in find_transport_state_in_use().
* Remove useless CMD_STOP in find_transport_state_in_use().
res_pjsip_endpoint_identifier_anonymous.c:
* Eliminate RAII_VAR in anonymous_identify().
* Remove always true check in find_transport_state_in_use().
* Remove useless CMD_STOP in find_transport_state_in_use().
Change-Id: I86924c31db5bd225ca0c1219c761b668c6f91189
pjproject does not have a function to reverse pjsip_inv_usage_init.
This means we need to ignore any calls to the functions once shutdown is
final.
ASTERISK-27571 #close
Change-Id: Ia550fcba563e2328f03162d79fb185f16b7c9b9d
In an earlier release, inbound registrations on a reliable transport
were pruned on Asterisk restart since the TCP connection would have
been torn down and become unusable when Asterisk stopped. This same
process is now also applied to inbound subscriptions.
Also fixed issues in res_pjsip_registrar where it wasn't handling the
monitoring correctly when multiple registrations came in over the same
transport.
To accomplish this, the pjsip_transport_event feature needed to
be refactored to allow multiple monitors (multiple subcriptions or
registrations from the same endpoint) to exist on the same transport.
Since this changed the API, any external modules that may have used the
transport monitor feature (highly unlikey) will need to be changed.
ASTERISK-27612
Reported by: Ross Beer
Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36
This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
I've audited all modules that include any header which includes
asterisk/optional_api.h. All modules which use OPTIONAL_API now declare
those dependencies in AST_MODULE_INFO using requires or optional_modules
as appropriate.
In addition ARI dependency declarations have been reworked. Instead of
declaring additional required modules in res/ari/resource_*.c we now add
them to an optional array "requiresModules" in api-docs for each module.
This allows the AST_MODULE_INFO dependencies to include those missing
modules.
Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606
The type=identify endpoint identification method can match by IP address
and by SIP header. However, the SIP header matching has limited
usefulness because you cannot specify the SIP header matching priority
relative to the IP address matching. All the matching happens at the same
priority and the order of evaluating the identify sections is
indeterminate. e.g., If you had two type=identify sections where one
matches by IP address for endpoint alice and the other matches by SIP
header for endpoint bob then you couldn't predict which endpoint is
matched when a request comes in that matches both.
* Extract the SIP header matching criteria into its own "header" endpoint
identification method so the user can specify the relative priority of the
SIP header and the IP address matching criteria in the global
endpoint_identifier_order option. The "ip" endpoint identification method
now only matches by IP address.
ASTERISK-27491
Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095
* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
We did this for TCP transports already but I'm not sure why we
didn't do it for TLS transports.
ASTERISK_27474 #not_final_fix
Change-Id: I5b1ef4b882f7b859e718236686b7898751dbb262
If any component of ast_config_AST_RECORDING_DIR is a symbolic link we
would incorrectly assume the ARI user was trying to escape the recording
path. Create additional check to check the recording directory's
realpath, only deny access if both do not match.
This is needed by the testsuite when run by 'run-local'.
Change-Id: I9145e841865edadcb5f75cead3471ad06bbb56c0
The requirement that "ip" must be in the endpoint identify_by list to
allow the type=identify method to identify the endpoint is not necessary.
The "ip" identifier method can match one and only one endpoint. To even
work, the "ip" identifier method configuration must explicitly specify the
identified endpoint. Therefore, why bother configuring the type=identify
identifier in the first place? The requirement only adds the potential
for configuration errors for no benefit. Even worse, those configuration
errors cannot be detected when the configuration loads. The requirement
was introduced with the ASTERISK_27206 patch.
* Remove the code change that enforces the requiremnt. Listing the "ip"
method in the identify_by value is simply documentation.
Change-Id: Ia057f92a33fb5d9f51dc5d5692e3d5ee1a6f2c11
* Extracted sip_endpoint_identifier_type2str() and
sip_endpoint_identifier_str2type() to simplify the calling functions.
* Fixed pjsip_configuration.c:ident_to_str() building the endpoint's
identify_by value string.
Change-Id: Ide876768a8d5d828b12052e2a75008b0563fc509
The AMI PJSIPShowEndpoint action could only list one IdentifyDetail AMI
event per endpoint. However, there is no reason that multiple
type=identify sections cannot identify the same endpoint.
* Reworked format_ami_endpoint_identify() to generate as many
IdentifyDetail AMI events as there are matching identifiers.
Change-Id: Ie146792aef72d78e05416ab5b27bc552a30399db
Some WebRTC clients can't handle renegotiation with the addition of
streams that include an offer to bundle. They instead expect the
newly added streams to already be bundled. This change does such a thing
if WebRTC support is enabled on an endpoint.
ASTERISK-27566
Change-Id: I7fe9b7ac35a2798627d9c2c8369129f407af6461
In addition to being a micro-optimization (RAII_VAR has overhead), this
change improves output of REF_DEBUG. Unfortunately when RAII_VAR calls
ao2_cleanup it does so from a generated _dtor_varname function. For
example this caused _dtor_app to release a reference instead of
__stasis_app_unregister.
Change-Id: I4ce67120583a446babf9adeec678b71d37fcd9e5
If the dial bridge has been created it must be released by calling
ast_bridge_destroy, simply releasing the ao2 reference is not enough.
Also move stasis_app_control_shutdown earlier in unload to ensure the
bridge cannot be created or grabbed after the app_bridges container is
released.
Change-Id: I372302de94ca63876069e2585a049c5060e5e767
Instead of searching for bridge_id provided in an argument this function
always searched for BRIDGE_ALL first. Rewrite this function to work
like the similar functions for channel and endpoint functions.
Change-Id: Ib5caca69e11727c5c8a7284a1d00621f40f1e60a
Jenkins2
Joshua Colp
Alexander Traud
Ross Beer
Corey Farrell
Richard Mudgett
lvl
Ben Ford
George Joseph
Kevin Harwell
Sean Bright
Thomas Guebels
Joshua Elson
Sungtae Kim