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When two channels were early bridged in a native_rtp bridge, the RTP description
on one side was not updated when the other side answered.
This patch forbids non-answered channels to enter a native_rtp bridge, and
triggers a bridge reconfiguration when an ANSWER frame is received.
ASTERISK-27257
Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df
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* Add protection checks when mapping streams to the bridge. The channel
and bridge may be in the process of updating the stream mapping when a
media frame comes in so we may not be able to map the frame at the time.
* We need to map the streams to the bridge's stream numbers right before
they are written into the bridge. That way we don't have to keep
locking/unlocking the bridge and we won't have any synchronization
problems before the frames actually go into the bridge.
* Protect the deferred queue with the bridge_channel lock.
ASTERISK-27212
Change-Id: Id6860dd61b594b90c8395f6e2c0150219094c21a
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* Fix deadlock in
bridge_softmix.c:softmix_bridge_stream_topology_changed() between
bridge_channel and channel locks.
* The new bridge technology topology change callbacks must be called with
the bridge locked. The callback references the bridge channel list, the
bridge technology could change, and the bridge stream mapping is updated.
ASTERISK-27212
Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be
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This change fixes a few locking issues and some video misrouting.
1. When accessing the stream topology of a channel the channel lock
must be held to guarantee the topology remains valid.
2. When a channel was joined to a bridge the bridge specific
implementation for stream mapping was not invoked, causing video
to be misrouted for a brief period of time.
ASTERISK-27182
Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03
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This change fixes a few things uncovered during SFU testing.
1. Unreal channels incorrectly forwarded video frames when
no video stream was present on them. This caused a crash when
they were read as the core requires a stream to exist for the
underlying media type. The Unreal channel will now ensure a
stream exists for the media type before forwarding the frame
and if no stream exists then the frame is dropped.
2. Mapping of frames during bridging from the stream number of
the underlying channel to the stream number of the bridge was
done in the wrong location. This resulted in the frame getting
dropped. This mapping now occurs on reading of the frame from
the channel.
3. Bridging was using the wrong ast_read function resulting in
it living in a non-multistream world.
4. In bridge_softmix when adding new streams to existing channels
the wrong stream topology was copied resulting in no streams
being added.
Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8
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This change adds a deferred queue to bridging. If a bridge
technology determines that a frame can not be written and
should be deferred it can indicate back to bridging to do so.
Bridging will then requeue any deferred frames upon a new
channel joining the bridge.
This change has been leveraged for T.38 request negotiate
control frames. Without the deferred queue there is a race
condition between the bridge receiving the T.38 request
negotiate and the second channel joining and being in the
bridge. If the channel is not yet in the bridge then the T.38
negotiation fails.
A unit test has also been added that confirms that a T.38
request negotiate control frame is deferred when no other
channel is in the bridge and that it is requeued when a new
channel joins the bridge.
ASTERISK-26923
Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415
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This sets up the "plumbing" in bridge_softmix to
be able to accommodate Asterisk asking as an SFU
(selective forwarding unit) for conferences.
The way this works is that whenever a channel enters or leaves a
conference, all participants in the bridge get sent a stream topology
change request. The topologies consist of the channels' original
topology, along with video destination streams corresponding to each
participants' source video streams. So for instance, if Alice, Bob, and
Carol are in the conference, and each supplies one video stream, then
the topologies for each would look like so:
Alice:
Audio,
Source video(Alice),
Destination Video(Bob),
Destination video (Carol)
Bob:
Audio,
Source video(Bob)
Destination Video(Alice),
Destination video (Carol)
Carol:
Audio,
Source video(Carol)
Destination Video(Alice),
Destination video (Bob)
This way, video that arrives from a source video stream can then be
copied out to the destination video streams on the other participants'
channels.
Once the bridge gets told that a topology on a channel has changed, the
bridge constructs a map in order to get the video frames routed to the
proper destination streams. This is done using the bridge channel's
stream_map.
This change is bare-bones with regards to SFU support. Some key features
are missing at this point:
* Stream limits. This commit makes no effort to limit the number of
streams on a specific channel. This means that if there were 50 video
callers in a conference, bridge_softmix will happily send out topology
change requests to every channel in the bridge, requesting 50+
streams.
* Configuration. The plumbing has been added to bridge_softmix, but
there has been nothing added as of yet to app_confbridge to enable SFU
video mode.
* Testing. Some functions included here have unit tests.
However, the functionality as a whole has only been verified by
hand-tracing the code.
* Selectivenss. For a "selective" forwarding unit, this does not
currently have any means of being selective.
* Features. Presumably, someone might wish to only receive video from
specific sources. There are no external-facing functions at the moment
that allow for users to select who they receive video from.
* Efficiency. The current scheme treats all video streams as being
unidirectional. We could be re-using a source video stream as a
desetnation, too. But to simplify things on this first round, I did it
this way.
Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d
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When manipulating flags on a channel the channel has to be
locked to guarantee that nothing else is also manipulating
the flags. This change introduces locking where necessary to
guarantee this. It also adds helper functions that manipulate
channel flags and lock to reduce repeated code.
ASTERISK-26789
Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10
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This patch is the first cut at adding stream support to the bridging framework.
Changes were made to the framework that allows mapping of stream topologies to
a bridge's supported media types.
The first channel to enter a bridge initially defines the media types for a
bridge (i.e. a one to one mapping is created between the bridge and the first
channel). Subsequently added channels merge their media types into the bridge's
adding to it when necessary. This allows channels with different sized
topologies to map correctly to each other according to media type. The bridge
drops any frame that does not have a matching index into a given write stream.
For now though, bridge_simple will align its two channels according to size or
first to join. Once both channels join the bridge the one with the most streams
will indicate to the other channel to update its streams to be the same as that
of the other. If both channels have the same number of streams then the first
channel to join is chosen as the stream base.
A topology change source was also added to a channel when a stream toplogy
change request is made. This allows subsystems to know whether or not they
initiated a change request. Thus avoiding potential recursive situations.
ASTERISK-26966 #close
Change-Id: I1eb5987921dd80c3cdcf52accc136393ca2d4163
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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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When a channel is made the video source, the bridge holds a reference to
it. Whenever the video source changes, that reference is released.
However, a ref leak does occur if the channel leaves the bridge (such as
being hung up) while it is the video source, as the bridge never
releases the ref in such a case.
This patch adds a line to the bridge_channel_internal_join routine such
that, when a channel finishes its time in the bridge, it notifies the
bridge via ast_bridge_remove_video_src that if it is a video source its
reference should be released.
ASTERISK-26555 #close
Change-Id: I3a2f5238a9d2fc49c591f0e65199d782ab0be76a
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ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
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Found as a result of the testsuite tests/callparking test crashing.
Several calls to ast_get_chan_featuremap_config() and
ast_get_chan_features_xfer_config() did not lock the channel before
calling so the channel's datastore list was accessed without the lock's
protection. Apparently another thread deleted a datastore on the
channel's list while the crashing thread was walking the list. Crash at
0xdeaddead due to MALLOC_DEBUG's memory filler value as a result.
* Add missing channel locks to calls that were not already protected
as the doxygen for those calls indicates.
Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1
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Added a new channel variable FORWARDERNAME which indicates which
channel was responsible for a forwarding requests received on dial attempt.
Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.
ASTERISK-26059 #close
Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2
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Dial events up to this point have come in two flavors
* A Dial event with no status to indicate that dialing has begun
* A Dial event with a status to indicate that dialing has ended
With this change, Dial events have been expanded to also give
intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS".
This is especially useful for ARI dialing, as it gives the application
writer the opportunity to place a channel into an early bridge when
early media is detected.
AMI handles these in-progress dial events by sending a new event called
"DialState" that simply indicates that dial state has changed but has
not ended. ARI never distinguished between DialBegin and DialEnd, so no
change was made to the event itself.
Another change here relates to dial forwards. A forward-related event
was previously only sent when a channel was successfully able to forward
a call to a new channel. With this set of changes, if forwarding is
blocked, we send a Dial event with a forwarding destination but no
forwarding channel, since we were prevented from creating one. This is
again useful for ARI since application writers can now handle call
forward attempts from within their own application.
ASTERISK-25925 #close
Reported by Mark Michelson
Change-Id: I42cbec7730d84640a434d143a0d172a740995543
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An earlier patch blocked the ast_bridge_impart() call until the channel
either entered the target bridge or it failed. Unfortuantely, if the
target bridge is stasis and the imprted channel is not a stasis channel,
stasis bounces the channel out of the bridge to come back into the bridge
as a proper stasis channel. When the channel is bounced out, that
released the block on ast_bridge_impart() to continue. If the impart was
a result of a transfer, then it became a race to see if the swap channel
would get hung up before the imparted channel could come back into the
stasis bridge. If the imparted channel won then everything is fine. If
the swap channel gets hung up first then the transfer will fail because
the swap channel is leaving the bridge.
* Allow a chain of ast_bridge_impart()'s to happen before any are
unblocked to prevent the race condition described above. When the channel
finally joins the bridge or completely fails to join the bridge then the
ast_bridge_impart() instances are unblocked.
ASTERISK-25947
Reported by: Richard Mudgett
ASTERISK-24649
Reported by: John Bigelow
ASTERISK-24782
Reported by: John Bigelow
Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1
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We have to setup the channel roles after the bridge class push is called
because the bridge class push callback may have set roles on the incoming
channel. Since we have already partially pushed the channel into the
bridge and reversing what we have already done could be problematic, the
only thing we can do is press on to complete pushing the channel into the
bridge.
* Ignore any channel role setup errors after pushing the channel into a
bridge. The channel may behave incorrectly in the bridge but we can no
longer abort the push at this time.
Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00
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The channel is now going to get T.38 terminated when it leaves the
bridging system and the bridged peers are going to get T.38 terminated as
well.
ASTERISK-25582
Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7
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Local channel optimization could cause DTMF digits to be duplicated.
Pending DTMF end events would be posted to a bridge when the local channel
optimizes out and is replaced by the channel further down the chain. When
the real digit ends, the channel would get another DTMF end posted to the
bridge.
A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B
1) LocalA has the /n flag to prevent optimization.
2) B is sending DTMF to A through the local channel chain.
3) When LocalB optimizes out it can move B to the position of LocalB;1
4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would
settle an owed DTMF end to the bridge toward LocalA;2.
5) When B finally ends its DTMF it sends the DTMF end down the chain.
6) Without this patch, A would hear the DTMF digit end when LocalB
optimizes out and when B ends the original digit.
ASTERISK-25582
Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251
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Currently if a channel is transferred out of a bridge, the BRIDGEPEER
variable (also BRIDGEPVTCALLID) remain set even once the channel is
out of the bridge. This patch removes these variables when leaving
the bridge.
ASTERISK-25600 #close
Reported by: Mark Michelson
Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da
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When executing an action in a bridge it is possible for the
channel to be hung up without the bridge becoming aware of it.
This is most easily reproducible by hanging up when the bridge
is streaming DTMF due to a feature timeout. This change makes
it so after action execution the channel is checked to determine
if it has been hung up and if it has it is kicked from the bridge.
ASTERISK-25341 #close
Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062
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During an attended transfer a thread is started that handles imparting the
bridge channel. From the start of the thread to when the bridge channel is
ready exists a gap that can potentially cause problems (for instance, the
channel being swapped is hung up before the replacement channel enters the
bridge thus stopping the transfer). This patch adds a condition that waits
for the impart thread to get to a point of acceptable readiness before
allowing the initiating thread to continue.
ASTERISK-24782
Reported by: John Bigelow
Change-Id: I08fe33a2560da924e676df55b181e46fca604577
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When performing a blonde transfer the code uses the old masquerade
mechanism to move a channel around. As a result of this certain information,
such as connected line, is moved between the channels involved. Upon
completion of the move a frame is queued which is supposed to update the
connected line information on the channel. This does not occur as the
code considers it a redundant update since the masquerade operation
updated the channel (but did not inform it of the new connected line
information). The code also does not queue a connected line update
to be handled by the thread handling the channel. Without this any
other channel that may be loosely involved does not know it is
talking to a different caller.
This change does the following to resolve this:
1. The indicated connected line information is cleared upon
completion of the masquerade operation when doing a blonde transfer.
This prevents the connected line update from being considered
redundant.
2. A connected line update frame is now queued upon the completion
of the masquerade operation so any other channel loosely involved
knows that there is a different caller.
ASTERISK-25157 #close
Reported by: Joshua Colp
Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20
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Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
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For some applications - such as SLA - a phone pressing hold should not behave
in the fashion that the Asterisk core would like it to. Instead, the hold
action has some application specific behaviour associated with it - such as
disconnecting the channel that initiated the hold; only playing MoH to channels
in the bridge if the channels are of a particular type, etc.
One way of accomplishing this is to use a framehook to intercept the
hold/unhold frames, raise an event, and eat the frame. Tasty. This patch
accomplishes that using a new dialplan function, HOLD_INTERCEPT.
In addition, some general cleanup of raising hold/unhold Stasis messages was
done, including removing some RAII_VAR usage.
Review: https://reviewboard.asterisk.org/r/4549/
ASTERISK-24922 #close
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Attempting to execute DTMF in a confbridge while file playback (prompt,
announcement, etc) is occurring is not allowed. You have to wait until
the sound file has completed before entering DTMF. This patch fixes it
so that app_confbridge now monitors for dtmf key presses during menu
driven file playback. If a key is pressed playback stops and it executes
the matched menu option.
ASTERISK-24864 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4510/
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Switch logger callid's from AO2 objects to simple integers.
This helps in two ways. Copying integers is faster than
referencing AO2 objects, so this will result in a small
reduction in logger overhead. This also erases the possibility
of an infinate loop caused by an invalid callid in
threadstorage.
ASTERISK-24833 #comment Committed callid conversion to trunk.
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4466/
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direct media.
This change fixes two issues:
1. During a swap operation bridging added the new channel before having the swap channel
leave. This was not handled in bridge_native_rtp and could result in a channel not getting
reinvited back to Asterisk. After this change the swap channel will leave first and the
new channel will then join.
2. If a re-invite was received after a session had been established any upstream elements
(such as bridge_native_rtp) were not notified that they may want to re-evaluate things.
After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs
and upstream can react.
AST-1524 #close
Review: https://reviewboard.asterisk.org/r/4378/
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This patch addresses compilation errors on OS X. It's been a while, so
there's quite a few things.
* Fixed __attribute__ decls in route.h to be portable.
* Fixed htonll and ntohll to work when they are defined as macros.
* Replaced sem_t usage with our ast_sem wrapper.
* Added ast_sem_timedwait to our ast_sem wrapper.
* Fixed some GCC 4.9 warnings using sig*set() functions.
* Fixed some format strings for portability.
* Fixed compilation issues with res_timing_kqueue (although tests still fail
on OS X).
* Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue
on OS X).
ASTERISK-24539 #close
Reported by: George Joseph
ASTERISK-24544 #close
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/4327/
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When code imparts a channel into a bridge to swap with another channel, a
ref needs to be held on the swap channel to ensure that it cannot
dissapear before finding it in the bridge.
* The ast_bridge_join() swap channel parameter now always steals a ref for
the swap channel. This is the only change to the bridge framework's
public API semantics.
* bridge_channel_internal_join() now requires the bridge_channel->swap
channel to pass in a ref.
ASTERISK-24649
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4354/
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Any partially collected DTMF digits for a DTMF hook need to be pushed into
the bridge when a channel leaves the bridging system as if there were a
timeout.
Review: https://reviewboard.asterisk.org/r/4199/
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matching digits.
* Made collecting DTMF digits for the DTMF feature hooks pass frames from
the bridge.
* Made collecting DTMF digits possible by other bridge hooks if there is a
need.
ASTERISK-24447 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4123/
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Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.
A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade. With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer. Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.
* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.
* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.
* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.
ASTERISK-24356 #close
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/4034/
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On a SIP reinvite that changes media strams, the PJSIP channel driver was
flooding the log with "Asked to transmit frame type %s, while native
formats is %s" warnings.
* Fixes PJSIP not setting up translation paths when the formats change on
a reinvite. AFS-63 was effectively reintroduced because of the media
formats work. res_pjsip_sdp_rtp.c:set_caps()
* Improved the unexpected frame format WARNING message to include more
information.
* Added protective locking while altering formats on a channel. Reworked
set_format() to simplify and protect the formats under manipulation.
* Restored some code that got lost in the media_formats work.
(channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps())
AFS-137 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3906/
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If a manager or CLI user attached a mixmonitor to a call running a dynamic
bridge feature while in a bridge, the feature would be interrupted and the
channel would be forcibly kicked out of the bridge (usually ending the call
during a simple 1 to 1 call). This would also occur during any similar action
that could set the unbridge soft hangup flag, so the fix for this was to
remove unbridge from the soft hangup flags and make it a separate thing all
together.
ASTERISK-24027 #close
Reported by: mjordan
Review: https://reviewboard.asterisk.org/r/3900/
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Say you wanted to include variables in an application map and have those
variables substituted and passed along to the application being executed;
currently this does not happen.
This patch adds this ability to pass channel variable values to an
application before being executed.
ASTERISK-22608 #close
Reported by: Michael L. Young
patches:
features_substitute_arguments_v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3819/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
1. Asterisk was limited in how many formats it could handle.
2. Formats, being a bit field, could not include any attribute information.
A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.
Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.
Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.
Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.
Generally, the new philosophy for handling formats is as follows:
* The ast_format structure is reference counted. This removed a large amount
of the memory allocations and copying that was done in prior versions.
* In order to prevent race conditions while keeping things performant, the
ast_format structure is immutable by convention and lock-free. Violate this
tenet at your peril!
* Because formats are reference counted, codecs are also reference counted.
The Asterisk core generally provides built-in codecs and caches the
ast_format structures created to represent them. Generally, to prevent
inordinate amounts of module reference bumping, codecs and formats can be
added at run-time but cannot be removed.
* All compatibility with the bit field representation of codecs/formats has
been moved to a compatibility API. The primary user of this representation
is chan_iax2, which must continue to maintain its bit-field usage of formats
for interoperability concerns.
* When a format is negotiated with attributes, or when a format cannot be
represented by one of the cached formats, a new format object is created or
cloned from an existing format. That format may have the same codec
underlying it, but is a different format than a version of the format with
different attributes or without attributes.
* While formats are reference counted objects, the reference count maintained
on the format should be manipulated with care. Formats are generally cached
and will persist for the lifetime of Asterisk and do not explicitly need
to have their lifetime modified. An exception to this is when the user of a
format does not know where the format came from *and* the user may outlive
the provider of the format. This occurs, for example, when a format is read
from a channel: the channel may have a format with attributes (hence,
non-cached) and the user of the format may last longer than the channel (if
the reference to the channel is released prior to the format's reference).
For more information on this work, see the API design notes:
https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.
There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).
Reviews:
https://reviewboard.asterisk.org/r/3814
https://reviewboard.asterisk.org/r/3808
https://reviewboard.asterisk.org/r/3805
https://reviewboard.asterisk.org/r/3803
https://reviewboard.asterisk.org/r/3801
https://reviewboard.asterisk.org/r/3798
https://reviewboard.asterisk.org/r/3800
https://reviewboard.asterisk.org/r/3794
https://reviewboard.asterisk.org/r/3793
https://reviewboard.asterisk.org/r/3792
https://reviewboard.asterisk.org/r/3791
https://reviewboard.asterisk.org/r/3790
https://reviewboard.asterisk.org/r/3789
https://reviewboard.asterisk.org/r/3788
https://reviewboard.asterisk.org/r/3787
https://reviewboard.asterisk.org/r/3786
https://reviewboard.asterisk.org/r/3784
https://reviewboard.asterisk.org/r/3783
https://reviewboard.asterisk.org/r/3778
https://reviewboard.asterisk.org/r/3774
https://reviewboard.asterisk.org/r/3775
https://reviewboard.asterisk.org/r/3772
https://reviewboard.asterisk.org/r/3761
https://reviewboard.asterisk.org/r/3754
https://reviewboard.asterisk.org/r/3753
https://reviewboard.asterisk.org/r/3751
https://reviewboard.asterisk.org/r/3750
https://reviewboard.asterisk.org/r/3748
https://reviewboard.asterisk.org/r/3747
https://reviewboard.asterisk.org/r/3746
https://reviewboard.asterisk.org/r/3742
https://reviewboard.asterisk.org/r/3740
https://reviewboard.asterisk.org/r/3739
https://reviewboard.asterisk.org/r/3738
https://reviewboard.asterisk.org/r/3737
https://reviewboard.asterisk.org/r/3736
https://reviewboard.asterisk.org/r/3734
https://reviewboard.asterisk.org/r/3722
https://reviewboard.asterisk.org/r/3713
https://reviewboard.asterisk.org/r/3703
https://reviewboard.asterisk.org/r/3689
https://reviewboard.asterisk.org/r/3687
https://reviewboard.asterisk.org/r/3674
https://reviewboard.asterisk.org/r/3671
https://reviewboard.asterisk.org/r/3667
https://reviewboard.asterisk.org/r/3665
https://reviewboard.asterisk.org/r/3625
https://reviewboard.asterisk.org/r/3602
https://reviewboard.asterisk.org/r/3519
https://reviewboard.asterisk.org/r/3518
https://reviewboard.asterisk.org/r/3516
https://reviewboard.asterisk.org/r/3515
https://reviewboard.asterisk.org/r/3512
https://reviewboard.asterisk.org/r/3506
https://reviewboard.asterisk.org/r/3413
https://reviewboard.asterisk.org/r/3410
https://reviewboard.asterisk.org/r/3387
https://reviewboard.asterisk.org/r/3388
https://reviewboard.asterisk.org/r/3389
https://reviewboard.asterisk.org/r/3390
https://reviewboard.asterisk.org/r/3321
https://reviewboard.asterisk.org/r/3320
https://reviewboard.asterisk.org/r/3319
https://reviewboard.asterisk.org/r/3318
https://reviewboard.asterisk.org/r/3266
https://reviewboard.asterisk.org/r/3265
https://reviewboard.asterisk.org/r/3234
https://reviewboard.asterisk.org/r/3178
ASTERISK-23114 #close
Reported by: mjordan
media_formats_translation_core.diff uploaded by kharwell (License 6464)
rb3506.diff uploaded by mjordan (License 6283)
media_format_app_file.diff uploaded by kharwell (License 6464)
misc-2.diff uploaded by file (License 5000)
chan_mild-3.diff uploaded by file (License 5000)
chan_obscure.diff uploaded by file (License 5000)
jingle.diff uploaded by file (License 5000)
funcs.diff uploaded by file (License 5000)
formats.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
bridges.diff uploaded by file (License 5000)
mf-codecs-2.diff uploaded by file (License 5000)
mf-app_fax.diff uploaded by file (License 5000)
mf-apps-3.diff uploaded by file (License 5000)
media-formats-3.diff uploaded by file (License 5000)
ASTERISK-23715
rb3713.patch uploaded by coreyfarrell (License 5909)
rb3689.patch uploaded by mjordan (License 6283)
ASTERISK-23957
rb3722.patch uploaded by mjordan (License 6283)
mf-attributes-3.diff uploaded by file (License 5000)
ASTERISK-23958
Tested by: jrose
rb3822.patch uploaded by coreyfarrell (License 5909)
rb3800.patch uploaded by jrose (License 6182)
chan_sip.diff uploaded by mjordan (License 6283)
rb3747.patch uploaded by jrose (License 6182)
ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
sip_cleanup.diff uploaded by opticron (License 6273)
chan_sip_caps.diff uploaded by mjordan (License 6283)
rb3751.patch uploaded by coreyfarrell (License 5909)
chan_sip-3.diff uploaded by file (License 5000)
ASTERISK-23960 #close
Tested by: opticron
direct_media.diff uploaded by opticron (License 6273)
pjsip-direct-media.diff uploaded by file (License 5000)
format_cap_remove.diff uploaded by opticron (License 6273)
media_format_fixes.diff uploaded by opticron (License 6273)
chan_pjsip-2.diff uploaded by file (License 5000)
ASTERISK-23966 #close
Tested by: rmudgett
rb3803.patch uploaded by rmudgetti (License 5621)
chan_dahdi.diff uploaded by file (License 5000)
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
rb3814.patch uploaded by rmudgett (License 5621)
moh_cleanup.diff uploaded by opticron (License 6273)
bridge_leak.diff uploaded by opticron (License 6273)
translate.diff uploaded by file (License 5000)
rb3795.patch uploaded by rmudgett (License 5621)
tls_fix.diff uploaded by mjordan (License 6283)
fax-mf-fix-2.diff uploaded by file (License 5000)
rtp_transfer_stuff uploaded by mjordan (License 6283)
rb3787.patch uploaded by rmudgett (License 5621)
media-formats-explicit-translate-format-3.diff uploaded by file (License 5000)
format_cache_case_fix.diff uploaded by opticron (License 6273)
rb3774.patch uploaded by rmudgett (License 5621)
rb3775.patch uploaded by rmudgett (License 5621)
rtp_engine_fix.diff uploaded by opticron (License 6273)
rtp_crash_fix.diff uploaded by opticron (License 6273)
rb3753.patch uploaded by mjordan (License 6283)
rb3750.patch uploaded by mjordan (License 6283)
rb3748.patch uploaded by rmudgett (License 5621)
media_format_fixes.diff uploaded by opticron (License 6273)
rb3740.patch uploaded by mjordan (License 6283)
rb3739.patch uploaded by mjordan (License 6283)
rb3734.patch uploaded by mjordan (License 6283)
rb3689.patch uploaded by mjordan (License 6283)
rb3674.patch uploaded by coreyfarrell (License 5909)
rb3671.patch uploaded by coreyfarrell (License 5909)
rb3667.patch uploaded by coreyfarrell (License 5909)
rb3665.patch uploaded by mjordan (License 6283)
rb3625.patch uploaded by coreyfarrell (License 5909)
rb3602.patch uploaded by coreyfarrell (License 5909)
format_compatibility-2.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Ensure that three-way transfers can be reported even if featuremap is
non-NULL.
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This fixes two reference leaks that would occur when TEST_FRAMEWORK was
enabled and features were successfully executed.
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* Create a Stasis bridge sub-class to propagate linkedids and
accountcodes.
* Fixed the basic bridge sub-class to update peeraccount codes when the
number of channels in the bridge drops back down to two parties.
* Refactored ast_bridge_channel_update_accountcodes() to handle channels
joining/leaving the bridge.
* Fixed the basic bridge sub-class to not call the base bridge class pull
method twice.
AFS-105 #close
ASTERISK-23852 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3720/
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This patch allows the current owner of a channel to define various
feature hooks to be made available once the channel has entered a
bridge. This includes any hooks that are setup on the
ast_bridge_features struct such as DTMF hooks, bridge event hooks
(join, leave, etc.), and interval hooks.
Review: https://reviewboard.asterisk.org/r/3649/
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During some performance testing of Asterisk with AGI, ARI, and lots of Local
channels, we noticed that there's quite a hit in performance during channel
creation and releasing to the dialplan (ARI continue). After investigating
the performance spike that occurs during channel creation, we discovered
that we create a lot of channel snapshots that are technically unnecessary.
This includes creating snapshots during:
* AGI execution
* Returning objects for ARI commands
* During some Local channel operations
* During some dialling operations
* During variable setting
* During some bridging operations
And more.
This patch does the following:
- It removes a number of fields from channel snapshots. These fields were
rarely used, were expensive to have on the snapshot, and hurt performance.
This included formats, translation paths, Log Call ID, callgroup, pickup
group, and all channel variables. As a result, AMI Status,
"core show channel", "core show channelvar", and "pjsip show channel" were
modified to either hit the live channel or not show certain pieces of data.
While this is unfortunate, the performance gain from this patch is worth
the loss in behaviour.
- It adds a mechanism to publish a cached snapshot + blob. A large number of
publications were changed to use this, including:
- During Dial begin
- During Variable assignment (if no AMI variables are emitted - if AMI
variables are set, we have to make snapshots when a variable is changed)
- During channel pickup
- When a channel is put on hold/unhold
- When a DTMF digit is begun/ended
- When creating a bridge snapshot
- When an AOC event is raised
- During Local channel optimization/Local bridging
- When endpoint snapshots are generated
- All AGI events
- All ARI responses that return a channel
- Events in the AgentPool, MeetMe, and some in Queue
- Additionally, some extraneous channel snapshots were being made that were
unnecessary. These were removed.
- The result of ast_hashtab_hash_string is now cached in stasis_cache. This
reduces a large number of calls to ast_hashtab_hash_string, which reduced
the amount of time spent in this function in gprof by around 50%.
#ASTERISK-23811 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3568/
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This patch is a re-do of r414122.
When r414122 was merged, a major problem with it was uncovered. UNBRIDGE soft
hangup flags have a catastrophic effect on the pbx core if they leak out from
the bridge layer: the channel gets hung up. With the number of threads
involved in a blind transfer, and with the initial patch, it was likely that
this would occur. This caused a large number of test failures
This patch is nearly identical with the one proposed in r414122, save for the
following changes:
- We explicitly clear the UNBRIDGE flag when setting an after goto on a
channel in a bridge
- Defensively, if we encounter an UNBRIDGE flag in the pbx core, we handle it
https://reviewboard.asterisk.org/r/3585/
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The Test Suite caught a few problems, undoing until those are resolved
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch fixes issues with direct media bridges that occur after a blind
transfer. These issues were caught by the (currently failing)
pjsip/transfers/blind_transfer/caller_direct_media test.
The test currently fails primarily for two reasons:
(1) When Bob and Charlie (the transfer target and the transfer destination)
enter a bridge together, the framehook remains on the transfer target
channel until both channels are in the bridge. As it consumes voice frames,
the initial bridge type is a simple bridge. The framehook is removed when
both channels are in the bridge; however, this does not currently cause the
bridging framework to re-evaluate the bridge. This patch adds a
AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a
framehook is removed so the bridge can re-evaluate itself.
(2) When a channel leaves a native RTP bridge, it may be leaving due to being
hung up. Sending a re-INVITE to a channel that is about to be hung up is
not nice - in fact, there's a good chance we'll send the BYE request before
the channel has had a chance to send back a 200 OK. To be somewhat nicer,
this patch adds a function to channel.h that allows the bridging framework
to query for exactly why a channel is leaving a bridge via the channel's
soft hangup flags. This allows it to only send the re-INVITE if there's a
chance the channel will survive the native bridging experience.
Review: https://reviewboard.asterisk.org/r/3535/
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This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
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Playing back a file to a channel in an ARI bridge would attempt to wait until
the playback concluded before returning. The method used involved signaling the
waiting thread in the ARI custom playback function.
The problem with this is that there were some corner cases that were not accounted for:
* If a bridge channel could not be found, then we never would attempt the playback but
would still attempt to wait for the playback to complete.
* If the bridge playfile action failed to queue, we would still attempt to wait for the
playback to complete.
* If the bridge playfile action were queued but some circumstance caused the playback
not to occur (the bridge dies, the channel is removed from the bridge), then we would
never be notified.
The solution to this is to move the waiting logic into the bridge code. A new bridge
API function is added to queue a synchronous action on a bridge. The waiting thread
is notified when the queued frame has been freed, either due to an error occurring
or due to successful playback. As a failsafe, the waiting thread has a 10 minute
timeout just in case there is a frame leak somewhere.
Review: https://reviewboard.asterisk.org/r/3338
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Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
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Original commit message by mmichelson (asterisk 12 r403311):
"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."
The above was initially committed and then reverted at r403398. The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed. Fixed by unreffing the channels.
Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel. Fixed by
unlocking "other->chan"
(closes issue ASTERISK-22709)
Reported by: John Bigelow
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