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The minimum block word length is actually 4, not 5.
Change-Id: I878542218225aed72c72bdf1b856fc822cd2d649
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The previous payload specific feedback handling was very single
minded in that it just assumed everything should trigger a video
update. This was changed but the handling of picture loss indication
was not added. The result was that video may not flow. This change
adds it explicitly in.
Change-Id: I1894be02e39ee10a0af841b5a1dca5f0ec7d60b6
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Change-Id: I3811de0014b1ffe96d4a3b49cddd5d4ca02ee5d4
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Fix some timer heap initializations and cancels to try and prevent
crashes and timer heap issues.
Change-Id: I64885d190fa22097d1b55987091375541e57a7ee
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Apparently it is possible for the transport to be destroyed without
triggering the transport callback logic. As a result the transport gets
destroyed and we have a stale pointer in the active_transports container.
* Invoke the transport monitor callback checks when the transport is
destroyed in addition to when it is disconnected and shutdown.
ASTERISK-27688
Change-Id: Ia9b5469fea8f2b3f2d8476fae6b748a4d23e7261
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This change extends the existing AST_FRAME_RTCP frame type to be
able to contain additional RTCP message types, such as feedback
messages. The payload type is contained in the subclass which allows
knowing what is in the frame itself.
The RTCP feedback message type is now handled and REMB[1] messages
are raised with their containing information.
This also fixes a bug where all feedback messages were triggering
video updates instead of just FIR and FUR.
Finally RTCP frames are now passed up through the Asterisk core to
what is handling the channel, mapped appropriately in the case of
bridging, and written to an outgoing stream. Since RTCP frames are
on a per-stream basis this is only done on multistream capable
channels.
[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
ASTERISK-27758
ASTERISK-26366
Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e
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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
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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
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ASTERISK-27741
Change-Id: I0aa59a54735c6d20b95c54db1bd095dbf93e7adf
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* 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
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media"
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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
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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
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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
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ASTERISK-27554
Change-Id: If61c7faab7d2fa1031c056ed6268fe928e2391cf
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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
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ASTERISK-27722
Change-Id: Ie7b8c30d86cb00a54d6ac4e09e6f28f42d2bd52c
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ASTERISK-27719
Change-Id: I230c5f9f316b2e9465c093c13580f72ebbaf67a7
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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