Commit Graph

115 Commits

Author SHA1 Message Date
George Joseph
34f3fe9552 app_confbridge: Use the SDP 'label' attribute to correlate users
Previously, the msid "label" attribute was used to correlate
participant info but because streams could be reused, the msid
wasn't being updated correctly when someone left the bridge and
another joined.

Now, instead of looking for the msid attribute on a channel's streams,
app_confbridge sets an "SDP:LABEL" attribute on the stream which
res_pjsip_sdp_rtp looks for.  If it finds it, it adds a "label"
attribute to the current sdp.

Change-Id: I6cbaa87fb59a2e0688d956e72d2d09e4ac20d5a5
2018-07-13 11:33:30 -05:00
George Joseph
5f12e2bd07 app_confbridge: Move participant info code to confbridge_manager.
With the participant info code in app_confbridge, we were still
in the process of adding the channel to the bridge when trying to send
an in-dialog MESSAGE.  This caused 2 threads to grab the channel
blocking flag at the same time.  To mitigate this, the participant
info code was moved to confbridge_manager so it runs after all
channel/bridge actions have finished.

Change-Id: I228806ac153074f45e0b35d5236166e92e132abd
2018-06-26 15:18:35 -06:00
George Joseph
e7a7506f9c app_confbridge: Enable sending events to participants
ConfBridge can now send events to participants via in-dialog MESSAGEs.
All current Confbridge events are supported, such as ConfbridgeJoin,
ConfbridgeLeave, etc.  In addition to those events, a new event
ConfbridgeWelcome has been added that will send a list of all
current participants to a new participant.

For all but the ConfbridgeWelcome event, the JSON message contains
information about the bridge, such as its id and name, and information
about the channel that triggered the event such as channel name,
callerid info, mute status, and the MSID labels for their audio and
video tracks. You can use the labels to correlate callerid and mute
status to specific video elements in a webrtc client.

To control this behavior, the following options have been added to
confbridge.conf:

bridge_profile/enable_events:  This must be enabled on any bridge where
events are desired.

user_profile/send_events:  This must be set for a user profile to send
events.  Different user profiles connected to the same bridge can have
different settings.  This allows admins to get events but not normal
users for instance.

user_profile/echo_events:  In some cases, you might not want the user
triggering the event to get the event sent back to them.  To prevent it,
set this to false.

A change was also made to res_pjsip_sdp_rtp to save the generated msid
to the stream so it can be re-used.  This allows participant A's video
stream to appear as the same label to all other participants.

Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e
2018-06-13 09:12:18 -06:00
William McCall
a7f4121238 app_confbridge: Add talking indicator for ConfBridgeList AMI response
When an AMI client connects, it cannot determine if a user was talking
prior to a transition in the user speaking state (which would generate
a ConfbridgeTalking event). This patch causes app_confbridge to track the
talking state and make this state available via ConfBridgeList.

ASTERISK-27877 #close

Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6
2018-06-01 07:46:30 -06:00
Chris-Savinovich
b437656c2e "confbridge show profile bridge" does not output "sfu" when video_mode is sfu
Fixes a bug on the "confbridge show profile bridge" cli command
that showed "video_mode=no video" when video_mode was set
to "sfu"

ASTERISK-27418  #close

Change-Id: I481e3172c7f872664c7ac7809879d541c9f031e9
2018-04-18 16:10:54 -06:00
Joshua Colp
8de3fa2b56 bridge_softmix / app_confbridge: Add support for REMB combining.
This change adds the ability for multiple REMB reports in
bridge_softmix to be combined according to a configured
behavior into a single report. This single report is sent
back to the sender of video, which adjusts the encoding bitrate
to be at or below the bitrate of the report. The available
behaviors are: lowest, highest, and average. Lowest uses the
lowest received bitrate. Highest uses the highest received
bitrate. Average goes through the received bitrates adding
them to the previous average and creates a new average.

Other behaviors can be added in the future and the existing
average one may be adjusted, but this provides the foundation
to do so.

Support for configuring which behavior to use has been
added to app_confbridge.

ASTERISK-27804

Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66
2018-04-17 11:25:17 -06:00
Joshua Colp
0f6431e8e4 app_confbridge / bridge_softmix: Add ability to configure REMB interval.
This change adds a configuration option to app_confbridge which can be
used to set the interval at which we will send a combined REMB (remote
estimated maximum bitrate) frame to sources of video. The bridging API
has also been extended slightly to allow setting this so bridge_softmix
can use it.

ASTERISK-27786

Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82
2018-04-03 08:13:11 -06:00
Richard Mudgett
f4b161440b app_confbridge: Add the Muted header to ConfbridgeJoin AMI event.
ASTERISK-27651

Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34
2018-02-05 13:47:30 -06:00
Richard Mudgett
b9024197ab app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs.
The dsp_talking_threshold does not represent time in milliseconds.  It
represents the average magnitude per sample in the audio packets.  This is
what the DSP uses to determine if a packet is silence or talking/noise.

Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443
2018-01-31 13:13:27 -06:00
Corey Farrell
bf2d35931d aco: Minimize use of regex.
Remove nearly all use of regex from ACO users.  Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
  callers use simple prefix based regex.  I haven't decided the best
  way to fix this in both 13/15 and master.

Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b
2017-12-15 10:14:31 -05:00
Joshua Colp
6fd3db51e8 app_confbridge: Only create a channel that records audio.
This change makes it so that the conference recorder channel
that is created only contains audio formats and an audio stream.
This is because the underlying application used by ConfBridge to
record, MixMonitor, only allows recording audio.

Having additional streams (and in particular a video stream) can
result in clients needlessly renegotiating to add a video stream
that will never receive video.

Change-Id: I89d38aedc9205eca7741d5435e73e73bb9de97a0
2017-09-19 07:56:00 -05:00
Joshua Colp
c7edd0b493 Merge "app_confbridge: Document sfu video_mode value." 2017-08-23 13:05:35 -05:00
Richard Mudgett
5bbf7b2aad app_confbridge: Document sfu video_mode value.
Change-Id: I26e17df2c93f3933b23f78070603adbcc84ba204
2017-08-22 11:23:45 -05:00
Richard Mudgett
f96536b1ea confbridge.h: Fix doxygen comments.
Change-Id: I16133166a85fdb557c66ffcbfe8128d0b4725b0e
2017-08-22 11:21:13 -05:00
Joshua Colp
680c491a62 bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
This change does a few things to improve packet loss and renegotiation:

1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.

2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.

3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.

4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.

5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.

ASTERISK-27143

Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45
2017-07-19 13:23:26 +00:00
Mark Michelson
39d14834f8 Confbridge: Add "sfu" video mode to bridge profile options.
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
2017-05-30 10:24:20 -05:00
George Joseph
0001834157 app_confbridge: Fix reference to cfg in menu_template_handler
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
2017-05-04 20:13:55 -05:00
Richard Mudgett
57656e2b5b app_confbridge: Fix ConfbridgeTalking AMI event description.
Thanks to Chris Howard for pointing this out on the wiki.

Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705
2017-03-16 15:54:10 -06:00
frahaase
5b1796f59d Binaural synthesis (confbridge): DTMF conference management.
DTMF configuration options for the binaural softmix bridge:
toggle binaural rendering (per channel).

ASTERISK-26292

Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8
2017-02-24 15:13:56 -06:00
Sean Bright
3f94373778 cli: Fix various CLI documentation and completion issues
* app_minivm: Use built-in completion facilities to complete optional
arguments.

* app_voicemail: Use built-in completion facilities to complete
optional arguments.

* app_confbridge: Add missing colons after 'Usage' text.

* chan_alsa: Use built-in completion facilities to complete optional
arguments.

* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'

* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'

* func_odbc: Correct completions for 'odbc read' and 'odbc write'

* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.

* main/bridge: Correct completions for 'bridge kick.'

* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.

* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'

* main/pbx_app: Remove redundant completions for 'core show
applications.'

* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'

* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.

Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca
2017-02-13 11:33:15 -05:00
Corey Farrell
a6e5bae3ef Remove ASTERISK_REGISTER_FILE.
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
2016-10-27 09:53:55 -04:00
frahaase
dce31f90ba Binaural synthesis (confbridge): On/off setting for binaural synthesis.
Adds setting to confbridge.conf (binaural_active) that determines if binaural
synthesis can be available in bridge_softmix.

ASTERISK-26292

Change-Id: I59dfcb8e55fe1df4ef32045882fea5bb58fc71db
2016-10-17 18:58:14 +00:00
Mark Michelson
ac02bbd9a0 ConfBridge: Make some announcements asynchronous.
Confbridge announcements tend to block a channel while they are being
played. In some circumstances, this is warranted since you want that
particular channel not to hear the announcement (Example: "John Doe has
entered the conference"). For others it makes less sense.

This change first introduces methods for playing sounds asynchronously
into the conference. This is very similar to how synchronous sounds are
played, except the channel initiating the playback does not wait for the
sound to complete before moving on.

Asynchronous announcements are used for two circumstances:
* Sounds played for a user after they have left the bridge
* Sounds that play first to a single user and then the rest of the
  conference (if the channel and conference use the same language)

ASTERISK-26289 #close
Reported by Mark Michelson

Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a
2016-09-07 09:12:41 -05:00
Mark Michelson
ded22c712a ConfBridge: Rework announcer channel methodology
NOTE: This patch was submitted earlier and reverted because of a failing
test. The test has been patched so that it adjusts for the changes here,
so this is being resubmitted for review.

One feature that confbridge has is the ability to play sounds to all
participants in the conference. Prior to this commit, the algorithm for
this was as follows:

* Grab the playback lock
* Push the conference announcer channel into the bridge
* Play back the sound
* Pull the conference announcer channel from the bridge
* Release the playback lock

The issue here is that the act of adding the playback channel to the
bridge and removing it for each announcement is expensive. Amongst the
expenses:

* The announcer channel is imparted into the bridge, meaning a new
  thread is spun up for each playback.
* When the announcer is added or removed from the bridge, it results
  in the BRIDGEPEER channel variable being set on all channels in the
  bridge. This requires keeping the bridge locked and locking each
  individual channel in order to set it.
* There's also just the general overhead of adding the channel and
  removing it from the bridge. The bridge potentially has to reconfigure
  every single time

With this commit, the paradigm for playing back announcements has
shifted.

* The announcer channel is now added to the bridge when the conference
  is allocated, and it is hung up when the conference is destroyed.
* A taskprocessor is used to queue playbacks onto the announcer channel.
  This keeps the behavior from before where playbacks do not overlap.
* The announcer channel is no longer placed into the bridge as
  departable. Since we are not constantly removing the channel from
  the bridge, it is safe to add the channel using an independent thread
  and simply hang the channel up when it is time for the conference to
  be destroyed.

The use of the taskprocessor for playbacks opens up the interesting
possibility of having asynchronous announcements played. In this commit,
however, the behavior is still exactly the same as it previously was.

ASTERISK-26289
Reported by Mark Michelson

Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0
2016-08-23 13:03:05 -05:00
Joshua Colp
065d810d3f Revert "ConfBridge: Rework announcer channel methodology"
This reverts commit 5aa8773052.

Change-Id: I9ab45776e54a54ecf1bac9ae62d976dec30ef491
2016-08-23 05:54:02 -05:00
Mark Michelson
5aa8773052 ConfBridge: Rework announcer channel methodology
One feature that confbridge has is the ability to play sounds to all
participants in the conference. Prior to this commit, the algorithm for
this was as follows:

* Grab the playback lock
* Push the conference announcer channel into the bridge
* Play back the sound
* Pull the conference announcer channel from the bridge
* Release the playback lock

The issue here is that the act of adding the playback channel to the
bridge and removing it for each announcement is expensive. Amongst the
expenses:

* The announcer channel is imparted into the bridge, meaning a new
  thread is spun up for each playback.
* When the announcer is added or removed from the bridge, it results
  in the BRIDGEPEER channel variable being set on all channels in the
  bridge. This requires keeping the bridge locked and locking each
  individual channel in order to set it.
* There's also just the general overhead of adding the channel and
  removing it from the bridge. The bridge potentially has to reconfigure
  every single time

With this commit, the paradigm for playing back announcements has
shifted.

* The announcer channel is now added to the bridge when the conference
  is allocated, and it is hung up when the conference is destroyed.
* A taskprocessor is used to queue playbacks onto the announcer channel.
  This keeps the behavior from before where playbacks do not overlap.
* The announcer channel is no longer placed into the bridge as
  departable. Since we are not constantly removing the channel from
  the bridge, it is safe to add the channel using an independent thread
  and simply hang the channel up when it is time for the conference to
  be destroyed.

The use of the taskprocessor for playbacks opens up the interesting
possibility of having asynchronous announcements played. In this commit,
however, the behavior is still exactly the same as it previously was.

ASTERISK-26289
Reported by Mark Michelson

Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5
2016-08-18 09:51:24 -05:00
Jaco Kroon
8923c9ac96 app_confbridge: Add a regcontext option for confbridge bridge profiles.
This patch allows for having app_confbridge register the name of the
conference as an extension into a specific context, similar to
regcontext for chan_sip.  This variant is not quite as involved as the
one in chan_sip and doesn't allow for multiple contexts or custom
extensions, you can only specify the context and the conference name
will always be used as the extension to register.

ASTERISK-25989 #close

Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f
2016-05-09 08:18:56 -05:00
Richard Mudgett
a63656b419 Bridge system: Fix memory leaks and double frees on impart failure.
You cannot reference the passed in features struct after calling
ast_bridge_impart().  Even if the call fails.

Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21
2016-04-22 15:45:47 -05:00
Richard Mudgett
1bc54aee80 app_confbridge: Only use b_profile options from the conference.
A user cannot set new bridge options after the conference is created by
the first user.  Attempting to do so is documented as undefined behavior.

This patch ensures that the bridge profile options used are from the
conference and not what a subsequent user may have tried to set.

Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266
2016-02-05 11:57:48 -06:00
Mark Michelson
ed13732188 Confbridge: Add a user timeout option
This option adds the ability to specify a timeout, in seconds, for a
participant in a ConfBridge. When the user's timeout has been reached,
the user is ejected from the conference with the CONFBRIDGE_RESULT
channel variable set to "TIMEOUT".

The rationale for this change is that there have been times where we
have seen channels get "stuck" in ConfBridge because a network issue
results in a SIP BYE not being received by Asterisk. While these
channels can be hung up manually via CLI/AMI/ARI, adding some sort of
automatic cleanup of the channels is a nice feature to have.

ASTERISK-25549 #close
Reported by Mark Michelson

Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98
2015-11-16 14:13:13 -06:00
Kevin Harwell
9f65ea482e app_confbridge: Default the template option to a compatible default profile.
Confbridge dynamic profiles did not have a default profile unless you
explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a
template was not set prior to the bridge being created then some
options were left with no default values set. This patch makes it so
the default templates are set to the default bridge and user profiles.

ASTERISK-24749 #close
Reported by: philippebolduc

Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a
2015-04-24 12:20:45 -05:00
Matt Jordan
4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro
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
2015-04-13 03:48:57 -04:00
Corey Farrell
c08fd275bf Logger: Convert 'struct ast_callid' to unsigned int.
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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-13 01:12:35 +00:00
Richard Mudgett
94eebd5ba5 app_confbridge: Repeatedly starting and stopping recording ref leaks the recording channel.
Starting and stopping conference recording more than once causes the
recording channels to be leaked.  For v13 the channels also show up in the
CLI "core show channels" output.

* Reworked and simplified the recording channel code to use
ast_bridge_impart() instead of managing the recording thread in the
ConfBridge code.  The recording channel's ref handling easily falls into
place and other off nominal code paths get handled better as a result.

ASTERISK-24719 #close
Reported by: John Bigelow

Review: https://reviewboard.asterisk.org/r/4368/
Review: https://reviewboard.asterisk.org/r/4369/
........

Merged revisions 431135 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 431160 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-27 17:48:18 +00:00
Matthew Jordan
aa8fd7d1b9 app_confbridge: Restore user's menu name to CLI output of 'confbridge list'
When issuing a 'confbridge list XXXX' CLI command, the resulting output no
longer displays the menu associated with a ConfBridge participant.

The issue was caused by ASTERISK-22760. When that patch was done, it removed
the copying of the menu name associated with the user from the actual user
profile.

This patch fixes the issue by copying the menu name over to the user profile
when the menu hooks are applied to the user. Since that function now does a
little bit more than just apply the hooks, the name of the function has been
changed to cover the copying of the menu name over as well.

In addition, there is a disparity between the menu name length as it is stored
on the conf_menu structure and the confbridge_user structure; this patch makes
the lengths match so that a strcpy can be used.

Review: https://reviewboard.asterisk.org/r/4372/

ASTERISK-24723 #close
Reported by: Steve Pitts
........

Merged revisions 431134 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-27 17:16:54 +00:00
Richard Mudgett
b69b0d12ee app_confbridge: Shorten CBRec channel names to CBRec/<conf_name>-<seq-num>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23 20:17:20 +00:00
Richard Mudgett
c780223507 app_confbridge: Make CBRec channel names more unique.
Channel names should be different from other channels in the system while
the channel exists.

* Use a sequence number for CBRec channels instead of a random number
because the same random number could be picked again for the next CBRec
channel.
........

Merged revisions 431052 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23 20:14:26 +00:00
Matthew Jordan
2afeadcc84 app_confbridge: Fix build error caused by XML validation errors
Summaries can't contain XML nodes, as they are defined to contain only text
data.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-22 12:16:36 +00:00
Matthew Jordan
b79a4a464f app_confbridge: Add the ability to pass options/command to MixMonitor
This patch adds the ability to pass options and a command to MixMontor when
recording a conference using ConfBridge.

New options are -

* record_options: Options to MixMontor, eg: m(), W() etc.
* record_command: The command to execute when recording is over.
* record_file_timestamp: Append the start time to the file name.

These options can also be used with the CONFBRIDGE function, e.g.,
Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME}))

Review: https://reviewboard.asterisk.org/r/4023

ASTERISK-24351 #close
Reported by: Gareth Palmer
patches:
  record_command-428838.patch uploaded by Gareth Palmer (License 5169)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-22 02:35:05 +00:00
Matthew Jordan
948af7fd79 apps/app_confbridge: Ensure 'normal' users hear message when last marked leaves
When r428077 was made for ASTERISK-24522, it failed to take into account users
who are neither wait_marked nor end_marked. These users are *also* supposed to
hear the 'leader has left the conference' message. Granted, this behaviour is
a bit odd; however, that is how it used to work... and behaviour changes are
not good.

This patch ensures that if there are any 'normal' users present when the last
marked user leaves the conference, the message will still be played to them.

Note that this regression was caught by the Asterisk Test Suite's
confbridge_nominal test, which has a quirky combination of users.
........

Merged revisions 428113 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 428114 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 428115 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-17 15:27:33 +00:00
Matthew Jordan
fc2279afea app_confbridge: Don't play leader leaving prompt if no one will hear it
Consider the following:
- A marked user in a conference
- One or more end_marked only users in the conference

When the marked users leaves, we will be in the conf_state_multi_marked state.
This currently will traverse the users, kicking out any who have the end_marked
flags. When they are kicked, a full ast_bridge_remove is immediately called on
the channels. At this time, we also unilaterally set the need_prompt flag.

When the need_prompt flag is set, we then playback a sound to the bridge
informing everyone that the leader has left; however, no one is left in the
bridge. This causes some odd behaviour for the end_marked users - they are
stuck waiting for the bridge to be unlocked. This results in them waiting for
5 or 6 seconds of dead air before hearing that they've been kicked.

Unfortunately, we do have to keep the bridge locked while we're playing back
the 'leader-has-left' prompt. If there are any wait_marked users in the
conference, this behaviour can't be easily changed - but we do make the case
of the end_marked users better with this patch.

Review: https://reviewboard.asterisk.org/r/4184/

ASTERISK-24522 #close
Reported by: Matt Jordan
........

Merged revisions 428077 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 428078 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 428079 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-17 03:08:11 +00:00
Joshua Colp
737b811749 app_confbridge: Play "leader has left" sound even when musiconhold is enabled.
Currently if the leader of a conference bridge leaves any participant
that has musiconhold enabled will not hear the "leader has left" sound.
This is because musiconhold is started and THEN the sound is played.

This change makes it so that the sound is played and THEN musiconhold
is started. This provides a better experience for users as they may not
have known previously why they went back to musiconhold.

Review: https://reviewboard.asterisk.org/r/4177/
........

Merged revisions 427844 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 427845 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 427846 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-14 14:56:53 +00:00
George Joseph
7c1a22fba7 confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events
Currently there's no way to tell if a user is an admin or not when receiving
the join, leave, mute, unmute and talking events.  This patch adds that
capability.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3950/
........

Merged revisions 422176 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 422177 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-27 17:30:51 +00:00
Matthew Jordan
a2c912e997 media formats: re-architect handling of media for performance improvements
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
2014-07-20 22:06:33 +00:00
Corey Farrell
db2ee74883 app_confbridge: Correct verification of conference name length
Conference names were not checked for maximum length, allowing unexpected
behaviour.  This change adds checking to ensure the maximum length is not
exceeded.  The maximum length is also changed from 32 to AST_MAX_EXTENSION.

ASTERISK-23035 #close
Reported by: Iñaki Cívico
Tested by: Iñaki Cívico
Patches:
    confbridge-enforce_max-1.8.patch uploaded by coreyfarrell (license 5909)
    confbridge-enforce_max-11up.patch uploaded by coreyfarrell (license 5909)
........

Merged revisions 415060 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 415066 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 415078 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-04 07:27:21 +00:00
Kinsey Moore
abd3e4040b Allow Asterisk to compile under GCC 4.10
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.
........

Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09 22:49:26 +00:00
Richard Mudgett
158bd5dd74 app_confbridge: Fix confbridge.conf dsp_talking_threshold option setting wrong parameter.
Fixed copy pasta error.

ASTERISK-23545 #close
Reported by: John Knott
........

Merged revisions 411944 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 411945 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-08 18:10:43 +00:00
Corey Farrell
fbe0dfaf44 Fix dialplan function NULL channel safety issues
(closes issue ASTERISK-23391)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3386/
........

Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-27 19:21:44 +00:00
Richard Mudgett
1900bae7b6 app_confbridge: Add missing destructor call to announcer channel destructor.
........

Merged revisions 410671 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-17 16:48:55 +00:00
Richard Mudgett
de3dc17cc5 app_confbridge: Make explicitly stop MOH if a user is kicked or hangs up while MOH is playing.
When MOH is playing to a user in a conference and the user is kicked or
hangs up from the conference then the AMI MusicOnHoldStop events didn't
happen.  (Asterisk v11 AMI event: MusicOnHold, state:Stop)

(closes issue ASTERISK-23311)
Reported by: Benjamin Keith Ford

Review: https://reviewboard.asterisk.org/r/3306/
........

Merged revisions 410490 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 410491 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-12 18:47:10 +00:00