If you use ast_request to create a PJSIP channel but then hang it
up without causing a transaction to be sent, the session will
never be destroyed. This is due ot the fact that it's pjproject
that triggers the session cleanup when the transaction ends.
app_chanisavail was doing this to get more granular channel state
and it's also possible for this to happen via ARI.
* ast_sip_session_terminate was modified to explicitly call the
cleanup tasks and unreference session if the invite state is NULL
AND invite_tsx is NULL (meaning we never sent a transaction).
* chan_pjsip/hangup was modified to bump session before it calls
ast_sip_session_terminate to insure that session stays valid
while it does its own cleanup.
* Added test events to session_destructor for a future testsuite
test.
ASTERISK-26908 #close
Reported-by: Richard Mudgett
Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9
This change extends the ast_request functionality by adding another
function and callback to create an outgoing channel with a requested
stream topology. Fallback is provided by either converting the
requested stream topology into a format capabilities structure if
the channel driver does not support streams or by converting the
requested format capabilities into a stream topology if the channel
driver does support streams.
The Dial application has also been updated to request an outgoing
channel with the stream topology of the calling channel.
ASTERISK-26959
Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6
It's possible for a name in a party id structure to be marked as valid, but the
name string itself be NULL (for instance this is possible to do by using the
dialplan CALLERID function). There were a couple of places where the name was
validated, but the string itself was not checked before passing it to functions
like 'strlen'. This of course caused a crashed.
This patch adds in a NULL check before attempting to pass it into a function
that is not NULL tolerant.
ASTERISK-25823 #close
Change-Id: Iaa6ffe9d92f598fe9e3c8ae373fadbe3dfbf1d4a
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
Interpolated frames are frames which contain a number of
samples but have no actual data. Audiohooks did not
handle this case when translating an incoming frame into
signed linear. It assumed that a frame would always contain
media when it may not. If this occurs audiohooks will now
immediately return and not act on the frame.
As well for users of ast_trans_frameout the function has
been changed to be a bit more sane and ensure that the data
pointer on a frame is set to NULL if no data is actually
on the frame. This allows the various spots in Asterisk that
check for an interpolated frame based on the presence of a
data pointer to work as expected.
ASTERISK-26926
Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b
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
For outgoing TCP connections, Asterisk uses the first IP address of the
interface instead of the IP address we asked him to bind to.
ASTERISK-26922 #close
Reported-by: Ksenia
Change-Id: I43c71ca89211dbf1838e5bcdb9be8d06d98e54eb
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
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
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
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
* 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
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
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
* 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
Some equipments may send a re-INVITE containing an SDP in the final ACK
request. If this happens in the context of direct media, the remote end
should be updated with a re-INVITE.
This patch queues an "update RTP peer" frame to trigger the re-INVITE,
instead of the "source change" frame wich was used previously.
ASTERISK-26951
Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6
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
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
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
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
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
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
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
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
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
