| Age | Commit message (Collapse) | Author |
|---|
|
|
|
Add some #if defined checks which allow building against LibreSSL.
These patchess come from OpenBSD ports:
https://cvsweb.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/patches/
ASTERISK-27043 #close
Reported by: OpenBSD ports
Change-Id: I2f6c08a5840b85ad4d2b75370b947ddde7a9a572
|
|
FreeBSD does not include a crypt.h include file. Definitions for
crypt() and crypt_r() are in unistd.h
ASTERISK-27042 #close
Change-Id: Ib307ee5e384870c6af50efa89fb73722dd0c3a7e
|
|
|
|
|
|
|
|
|
|
ASTERISK-26419 introduced a bug when calling ast_audiohook_write_list in
ast_write. It would free the frame given to ast_write if the frame returned
by ast_audiohook_write_list was different than the given one. The frame give
to ast_write should never be freed within that function. It is the caller's
resposibility to free the frame after writing (or when it its done with it).
By freeing it within ast_write this of course led to some memory corruption
problems.
This patch makes it so the frame given to ast_write is no longer freed within
the function. The frame returned by ast_audiohook_write_list is now subsequently
used in ast_write and is freed later. It is freed either after translate if the
frame returned by translate is different, or near the end of ast_write prior to
function exit.
ASTERISK-26973 #close
Change-Id: Ic9085ba5f555eeed12f6e565a638c3649695988b
|
|
Before this patch, when a user hung up during a Background, we would
stuff 0xff into a char and attempt a dialplan lookup of it. This caused
problems for some realtime engines which interpreted the value as the
beginning of an invalid UTF-8 sequence.
ASTERISK-19291 #close
Reported by: Andrew Nowrot
Change-Id: I8ca6da93252d61c76ebdb46a4aa65e73ca985358
|
|
ASTERISK-27025
Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed
|
|
In review 4843 (ASTERISK-24858), we added a hack that forced a smoother
creation when sending signed linear so that the byte order was adjusted
during transmission. This was needed because smoother flags were lost
during the new format work that was done in Asterisk 13.
Rather than rolling that same hack into res_rtp_multicast, re-introduce
smoother flags so that formats can dictate their own options.
Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16
|
|
A previous commit added plumbing to bridge_softmix to allow for an SFU
experience with Asterisk. This commit adds an option to app_confbridge
that allows for a confbridge to actually make use of the SFU video mode.
SFU mode is implemented in a "set it and forget it" kind of way. That
is, when the bridge is created, if SFU mode is enabled, then the video
mode gets set to SFU and cannot be changed. Future improvements may
allow for a hybrid experience (e.g. forward multiple video streams,
specifically those of the most recent talkers), but for this addition,
no such capability is present.
Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020
|
|
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
|
|
During the channel flag audit an incorrect change was
done. The flag should be cleared on the second channel.
ASTERISK-26469
Change-Id: I770c5a389550a2fb5a6ade942fccbb2e1d9199c8
|
|
|
|
...that can only be run by explicitly calling it with
'test execute category /DO_NOT_RUN/ name RAISE_SEGV'
This allows us to more easily test CI and debugging tools that
should do certain things when asterisk coredumps.
To allow this a new member was added to the ast_test_info
structure named 'explicit_only'. If set by a test, the test
will be skipped during a 'test execute all' or
'test execute category ...'.
Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed
|
|
Added functions that convert a string to an unsigned integer or unsigned long.
A couple of unit test were also created to test the routines. The reasons for
adding these conversion utilities (and hopefully eventually more) are as
follows:
* Conversion routines are functionally contained with consistent and
better error checking
* The function names offer a better description of what is happening
* It encourages code reuse for easier bug fixing at a single source
* It's simpler to use
* It's unit testable
For instance, currently in a lot of places when converting to an integer or
similar the "sscanf" function is used. When using "sscanf" it may not be
immediately clear what's happening as it lacks semantic naming. Limited error
checking is usually done as well. For example, most of the time a check is done
to make sure the value converted, but does not check for overflows or negative
valued conversions when converting unsigned numbers.
Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the
built in error handling that "strtoul" has. For instance "strtoul" contains
overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly
more complex in its use. And maybe a bit controversial, but it may be ("big if")
potentially slower than "strtoul" in some cases.
Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb
|
|
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
|
|
* changes:
SDP: Make process possible multiple fmtp attributes per rtpmap.
SDP: Explicitly stop a RTP instance before destoying it.
SDP: Rework merge_capabilities().
SDP: Update ast_get_topology_from_sdp() to keep RTP map.
|
|
|
|
|
|
|
|
|
|
Change-Id: Ie7511008d82b59590e0eb520a21b5e1da4bd7349
|
|
The sdp_state.remote_capabilities was only used inside merge_sdps() and
subsequent calls to merge_sdps() by re-INVITE's would leak them.
Change-Id: I0ceb7838ea044cc913e8ad4a255c39c9740ae0ce
|
|
When we optionally set the interface_address we are forcing the media to
go out a specific interface address. This allows us to optionally have
the media go out the interface that SIP signalling came in on or if we are
configured to have the media always go out a specific address.
Change-Id: I160d9fac322a075bd2557b430632544178196189
|
|
* Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream()
handle generating disabled/declined streams.
* Added /main/sdp/sdp_merge_asymmetric unit test. It currently does not
check the offerer side negotiated SDP because that isn't the purpose of
this patch and there is much to be done to handle declined/dummy streams.
* Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and
/main/sdp/sdp_merge_crisscross unit tests.
Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31
|
|
* Tried to give better variable names.
* Made our SDP answer use the offer's RTP payload types as the SDP RFC
says we SHOULD.
* Updating the local topology now takes the stream format caps. We are
likely preparing to send an offer.
Change-Id: I34d3be8e3036402a8575ffcae3eebc5ce348d7c0
|
|
* Add failure exits to ast_get_topology_from_sdp().
Change-Id: I4cc85c1ede8d712766ed20f544dbcef04c8c1049
|
|
This change uses the functions provided by OpenSSL to query
and better construct error messages for situations where
the connection encounters a problem.
ASTERISK-26606
Change-Id: I7ae40ce88c0dc4e185c4df1ceb3a6ccc198f075b
|
|
It is possible to initialize a valid config without a capath
or cafile definition. This will cause a crash on a reload.
This fix ensures capath is always allocated.
ASTERISK-26983 #close
Change-Id: I63ff715d9d9023427543a5b8a4ba7b0d82533c12
|
|
|
|
All log messages go to a queue serviced by a single thread
which does all the IO. This setting controls how big that
queue can get (and therefore how much memory is allocated)
before new messages are discarded. The default is 1000.
Should something go bezerk and log tons of messages in a tight
loop, this will prevent memory escalation.
When the limit is reached, a WARNING is logged to that effect
and messages are discarded until the queue is empty again. At
that time another WARNING will be logged with the count of
discarded messages. There's no "low water mark" for this queue
because the logger thread empties the entire queue and processes it
in 1 batch before going back and waiting on the queue again.
Implementing a low water mark would mean additional locking as
the thread processes each message and it's not worth it.
A "test" was added to test_logger.c but since the outcome is
non-deterministic, it's really just a cli command, not a unit
test.
Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1
|
|
|
|
|
|
ast_stream_clone() cannot copy the opaque user data stored on a stream.
We don't know how to clone the data so it isn't copied into the clone.
Change-Id: Ia51321bf38ecbfdcc53787ca77ea5fd2cabdf367
|
|
Change-Id: I532052bd7cd95a4b3565485fc01e2a1ea07ee647
|
|
menu_template_handler wasn't properly accounting for the fact that
it might be called both during a load/reload (which isn't really
valid but not prevented) and by a dialplan function. In both cases
it was attempting to use the "pending" config which wasn't valid in
the latter case. aco_process_config is also partly to blame because
it wasn't properly cleaning "pending" up when a reload was done and
no changes were made. Both of these contributed to a crash if
CONFBRIDGE(menu,template) was called in a dialplan after a reload.
* aco_process_config now sets info->internal->pending to NULL
after it unrefs it although this isn't strictly necessary in the
context of this fix.
* menu_template_handler now uses the "current" config and silently
ignores any attempt to be called as a result of someone uses the
"template" parameter in the conf file.
Luckily there's no other place in the codebase where
aco_pending_config is used outside of aco_process_config.
ASTERISK-25506 #close
Reported-by: Frederic LE FOLL
Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7
|
|
|
|
When using the Bridge AMI action on the same channel multiple times
it was possible for the channel to return to the wrong location in
the dialplan if the other party hung up. This happened because the
priority of the channel was not preserved across each action
invocation and it would fail to move on to the next priority in
other cases.
This change makes it so that the priority of a channel is preserved
when taking control of it from another thread and it is incremented
as appropriate such that the priority reflects where the channel
should next be executed in the dialplan, not where it may or may not
currently be.
The Bridge AMI action was also changed to ensure that it too
starts the channels at the next location in the dialplan.
ASTERISK-24529
Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a
|
|
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
|
|
The telephone_event option was used as a flag and a bit mapped value in
different places when it is a boolean. It is also inadequate to configure
the DTMF operation of the RTP instance created for the stream.
Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b
|
|
|
|
cap."
|
|
|
|
|
|
|
|
Change-Id: I473a174b869728604b37c60853896b0c458bc504
|
|
Change-Id: Ie29760c49c25d7022ba2124698283181a0dd5d08
|
|
Change-Id: I7707c9d872c476d897ff459008652b35142a35e1
|
