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Added an pre-defined integer vector declaration. This makes integer vectors
easier to declare and pass around. Also, added the ability to default a vector
up to a given size with a default value. Lastly, added functionality that
returns the "nth" index of a matching value.
Also, updated a unit test to test these changes.
Change-Id: Iaf4b51b2540eda57cb43f67aa59cf1d96cdbcaa5
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Without the disable, pjproject tries to build it's internal
webrtc implementation which requires sse2. This fails on
platforms without sse2.
ASTERISK-26930 #close
Reported-by: abelbeck
Change-Id: I07231f9160c35cfa42b194d3aad4e7d51fd9a410
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retransmissions."
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Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in
the format modules. Neither of these functions will ever return a value
less than 0, which we were checking for in some cases.
I've introduced a fair amount of duplication in the format modules, but
I plan to change how format modules work internally in a subsequent
patch set, so this is simply a stop-gap.
Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872
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This change adds an Alembic migration which adds a
ps_resource_list table that can contain resource_list
RLS configuration objects.
ASTERISK-26929
Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05
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This change adds a T.38 format which can be used in a stream
topology to specify that a UDPTL stream needs to be created.
The SDP API has been changed to understand T.38 and create
the UDPTL session, add the attributes, and parse the attributes.
This change does not change the boundary of the T.38 state
machine. It is still up to the channel driver to implement and
act on it (such as queueing control frames or reacting to them).
ASTERISK-26949
Change-Id: If28956762ccb8ead562ac6c03d162d3d6014f2c7
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The gist of this work ensures that when a remote SDP is received, it is
merged properly with the local capabilities. The remote SDP is converted
into a stream topology. That topology is then merged with the current
local topology on the SDP state. That new merged topology is then used
to create an SDP. Finally, adjustments are made to RTP instances based
on knowledge gained from the remote SDP.
There are also a battery of tests in this commit that ensure that some
basic SDP merges work as expected.
While this may not sound like a big change, it has the property that it
caused lots of ancillary changes.
* The remote SDP is no longer stored on the SDP state. Biggest reason:
there's no need for it. The remote SDP is used at the time it is being
set and nowhere else.
* Some new SDP APIs were added in order to find attributes and convert
generic SDP attributes into rtpmap structures.
* Writing tests made me realize that retrieving a value from an SDP
options structure, the SDP options needs to be made const.
* The SDP state machine was essentially gutted by a previous commit.
Initially, I attempted to reinstate it, but I found that as it had
been defined, it was not all that useful. What was more useful was
knowing the role we play in SDP negotiation, so the SDP state machine
has been transformed into an indicator of role.
* Rather than storing separate local and joint stream state
capabilities, it makes more sense to keep track of current stream
state and update it as things change.
Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d
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* Initialize hepv3_runtime_data.sockfd to -1 so that our ao2 destructor
does not close fd 0
* Add logging output when the required option - capture_address - is not
specified.
* Remove a no longer relevant #define and correct related documentation
* Pass appropriate flags to aco_option_register so that capture_address
cannot be the empty string.
ASTERISK-26953 #close
Change-Id: Ief08441bc6596d6f1718fa810e54a5048124f076
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The primary win of switching to eventfd when possible is that it only
uses a single file descriptor while pipe() will use two. This means for
each bridge channel we're reducing the number of required file
descriptors by 1, and - if you're using timerfd - we also now have 1
less file descriptor per Asterisk channel.
The API is not ideal (passing int arrays), but this is the cleanest
approach I could come up with to maintain API/ABI.
I've also removed what I believe to be an erroneous code block that
checked the non-blocking flag on the pipe ends for each read. If the
file descriptor is 'losing' its non-blocking mode, it is because of a
bug somewhere else in our code.
In my testing I haven't seen any measurable difference in performance.
Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d
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If ICE is enabled and a STUN server does not respond then we will block
until we give up on the STUN response. This will take nine seconds. In
the mean time the peer that sent the INVITE will send retransmissions.
* Restructure res_pjsip_session.c:new_invite() to send a 100 Trying out
earlier to prevent these retransmissions.
ASTERISK-26890
Change-Id: Ie3fc611e53a0eff6586ad55e4aacad81cf6319a8
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* Restructure ast_sip_session_alloc() to need less cleanup on off nominal
error paths.
* Made ast_sip_session_alloc() and ast_sip_session_create_outgoing() avoid
unnecessary ref manipulation to return a session. This is faster than
calling a function. That function may do logging of the ref changes with
REF_DEBUG enabled.
Change-Id: I2a0affc4be51013d3f0485782c96b8fee3ddb00a
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Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core
to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE
is passed to these functions, the calling thread will be blocked until
the newly created channel has been hung up.
After this patch, we run the dial on the current thread rather than
spawning a new one. The only in-tree code that passes
AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced
thread usage if you are using .call files.
Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913
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Occasionally a crash happens when processing the RTCP DTLS timeout
handler. The RTCP DTLS timeout timer could be left running if we have not
completed the DTLS handshake before we place the call on hold or we
attempt direct media.
* Made ast_rtp_prop_set() stop the RTCP DTLS timer when disabling RTCP.
* Made some sanity tweaks to ast_rtp_prop_set() when switching from
standard RTCP mode to RTCP multiplexed mode.
ASTERISK-26692 #close
Change-Id: If6c64c79129961acfa4b3d63a864e8f6b664acc0
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The struct ast_rtp_instance has historically been indirectly protected
from reentrancy issues by the channel lock because early channel drivers
held the lock for really long times. Holding the channel lock for such a
long time has caused many deadlock problems in the past. Along comes
chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock
because sometimes there may not be an associated channel created yet or
the channel pointer isn't available.
In the case of ASTERISK-26835 a pjsip serializer thread was processing a
message's SDP body while another thread was reading a RTP packet from the
socket. Both threads wound up changing the rtp->rtcp->local_addr_str
string and interfering with each other. The classic reentrancy problem
resulted in a crash.
In the case of ASTERISK-26853 a pjsip serializer thread was processing a
message's SDP body while another thread was reading a RTP packet from the
socket. Both threads wound up processing ICE candidates in PJPROJECT and
interfering with each other. The classic reentrancy problem resulted in a
crash.
* rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP
instance struct.
* rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP
instance struct for the API call.
* res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy
problem with rtp->rtcp->local_addr_str in the scheduler thread running
ast_rtcp_write().
* res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in
bridge_p2p_rtp_write() because there are two RTP instance structs
involved.
* res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler
callbacks. We cannot hold the instance lock when trying to stop a
scheduler callback.
* res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the
struct ast_rtp_instance ao2 object lock instead. The lock was used to
synchronize two threads to prevent a race condition between starting and
stopping a timeout timer. The race condition is no longer present between
dtls_perform_handshake() and __rtp_recvfrom() because the instance lock
prevents these functions from overlapping each other with regards to the
timeout timer.
* res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct
ast_rtp_instance ao2 object lock instead. The lock was used to
synchronize two threads using a condition signal to know when TURN
negotiations complete.
* res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN
ioqueue_worker_thread(). We cannot hold the instance lock when trying to
create or shut down the worker thread without a risk of deadlock.
This patch exposed a race condition between a PJSIP serializer thread
setting up an ICE session in ice_create() and another thread reading RTP
packets.
* res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we
have re-locked the RTP instance to prevent the other thread from trying to
process ICE packets on an incomplete ICE session setup.
A similar race condition is between a PJSIP serializer thread resetting up
an ICE session in ice_create() and the timer_worker_thread() processing
the completion of the previous ICE session.
* res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an
uninitialized/null remote_address after calling
update_address_with_ice_candidate().
* res_rtp_asterisk.c: Eliminate the chance of ice_reset_session()
destroying and setting the rtp->ice pointer to NULL while other threads
are using it by adding an ao2 wrapper around the PJPROJECT ice pointer.
Now when we have to unlock the RTP instance object to call a PJPROJECT ICE
function we will hold a ref to the wrapper. Also added some rtp->ice NULL
checks after we relock the RTP instance and have to do something with the
ICE structure.
ASTERISK-26835 #close
ASTERISK-26853 #close
Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4
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Change-Id: Id078a1df07a771808775e1053cdfe1d99c8fb172
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When opening a PCM wave file for reading, we aren't tracking the
frequency of the opened file, so we treat 16khz files as 8khz and do
half reads.
This patch also cleans up some of the data types and an unnecessarily
complex `if` expression.
ASTERISK-26613 #close
Reported by: Vitaly K
Change-Id: I05f8b263058dc573ea8ffe0c62e7964506e11815
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The periodic doc job does a make ari-stubs and checks that
there are no changes before generating the docs. Since I changed
the mustache template (and the generated code directly) recently
and forgot to regenerate the stubs, the doc job thinks they're out
of date.
Change-Id: I94b97035311eccf52b0101b8590223265a7881d4
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On filestream close, we need to clear out the ogg & vorbis data
structures to prevent a memory leak.
ASTERISK-26169 #close
Reported by: Ivan Myalkin
Change-Id: Iee94c5a5d5bdafbf8b181c5c064d15d90ace8274
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This reverts commit 7819f95791fe0ca0e0cdc417e2687a5900444053.
Change-Id: Ib91a7e6c9856f5f41329e42f40ba2394fee861a4
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res_stun_monitor will fail to load if DNS resolution of the STUN server
fails. Instead, we continue without the STUN server being resolved and
we will re-attempt the resolution on the STUN refresh interval.
ASTERISK-21856 #close
Reported by: Jeremy Kister
Change-Id: I6334c54a1cc798f8a836b4b47948e0bb4ef59254
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Sun's Au file format has a minimum data offset 24 bytes, but this
offset is encoded in each .au file. Instead of assuming the minimum,
read the actual value and store it for later use.
ASTERISK-20984 #close
Reported by: Roman S.
Patches:
asterisk-1.8.20.0-au-clicks-2.diff (license #6474) patch
uploaded by Roman S.
Change-Id: I524022fb19ff2fd5af2cc2d669d27a780ab2057c
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Change-Id: Iac40ecb20e10513d67bf0eaf61807f306067b258
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This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compact_headers=yes via the file pjsip.conf.
ASTERISK-26932 #close
Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689
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Change-Id: If99e3b4fc2d7e86fc3e61182aa6c835b407ed49e
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In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
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When a T.38 happens immediatly after call establishment, the control
frame can be lost because the other leg is not yet in the bridge.
This patch detects this case an makes sure T.38 negotation happens
when the 2nd leg is being made compatible with the negotating
first leg
ASTERISK-26923 #close
Change-Id: If334125ee61ed63550d242fc9efe7987e37e1d94
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