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
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
Added unit tests for the data buffer API. These tests include creating a
data buffer, putting payloads into the buffer, resizing the buffer, and
the nominal case for data buffer usage, which consists of adding
the max number of payloads to the buffer, checking to see if the correct
payloads are present, then adding more payloads and checking again to
see if the previous payloads were replaced or not.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Id5b599aa15a5e61d0ec080f97cd0c57bd07e6f8f
Adds a data buffer with a configurable size that can store different
kinds of packets (like RTP packets for retransmission). Given a number
it will store a data packet at that position relative to the others.
Given a number it will retrieve the given data packet if it is present.
This is purposely a storage of arbitrary things so it can be used not
just for RTP packets but also Asterisk frames in the future if needed.
The API does not internally use a lock, so it will be up to the user of
the API to properly protect the data buffer.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Iff13c5d4795d52356959fe2a57360cd57dfade07
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
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
Setting optind = 0 is forced to 1 in glibc implementation, but
causes option parsing to be flawed in other implementations, for
example on FreeBSD.
ASTERISK-27773 #close
Change-Id: Ia548e69f8302e9754dbbedb6bc451c0700c66f61
This creates a separate source to 'own' symbols related to options.h and
paths.h. This significantly reduces the number of exports created by
main/asterisk.o. This change is required to eventually be able to
link unmodified Asterisk sources to utilities and/or stand-alone tests.
ASTERISK~26245
Change-Id: I5cf184f4757f9363b80c9e678bdc35c477122380
This patch clears the talking flag from the channel (if already set), and
notifies listeners when that channel is put on hold. Note however, if the
endpoint continues to send audio frames and these are received by the bridge
then that channel will be put back into a "talking" state even though they
are on hold.
ASTERISK-27755 #close
Change-Id: I930e16c4662810f9f02043d69062f88173c5e2ef
AST_EXT_LIB_CHECK has several optional parameters. When an optional parameter
is left empty, [] is used to indicate this. However, this is done in the script
./configure only then, when a further parameter is not empty. For example, when
no extra libraries are needed to test the checked library, parameter 5 is not
mentioned. Except parameter 6 and higher are used, then parameter 5 must be
empty.
However, this general rule was broken
* four times for parameter 5 (extra libs) and
* three times for parameter 4 (header)
as found via the Regular Expression \[\]\). In case of parameter 5, all cases
were changed, because that happened for no reason. In case of parameter 4, an
[] improves readability actually. Therefore for parameter 4, the only case which
did not do it was changed. All this aims to create more consistency: Only do
something different if there is a reason to do so.
Change-Id: I037ef170cf1ad94497151a9ea5071a31c656cafe
The functions behind the flag and the flag itself were removed
from Asterisk 12 as incompatible with the new architecture.
Change-Id: I058493ef7a53ee290fd225bbcbb07bf46b623ccf
George Joseph
Jenkins2
Kevin Harwell
Ross Beer
Ben Ford
Joshua Colp
Guido Falsi
Corey Farrell
Alexander Traud
Ivan Poddubny