Both debounce or throttle could be used to prevent a spammer from
annoying the moderators.
With debounce, a spammer won't even be noticed by the mods, but
there won't be new notifications for legit guests during the spam.
With throttle, a spammer could still annoy the mods, we would only make
the interval between notifications bigger.
It is a tradeoff. An ideal solution would be preventing spamming from
the same user, but probably unnecessarily complex.
When managing Etherpad's pads, Meteor makes API calls to initiate the closed captions
and shared notes modules. The pad id was being mapped to a shorter id than the meeting
id because of a Etherpad lenght limitation.
Changed to something less guessable.
One of the main issues in 2.2.x problem reports currently in the wild in social media and in the bbb admin groups seems to be that people turn a couple of webcams on and the clients or the server already can't handle it any more. In most cases, enabling video-pagination for mobile devices and also cameraQualityThresholds already "solves" the problem and makes it possible for much more students to attend online-class as well as raise acceptance of using BBB instead of other commercial services.
Even after promoting these new settings for weeks, many BBB operators still don't know of them and are surprised and happy once they enable it.
This change contians rather high values so that admins see that these features exist, but typical use cases which might not want video-pagination enabled (typical 28people school class) are still not "annoyed".
* add param to force echo test when user joins audio after init
* fix UI stuck on connecting when userdata-bbb_auto_join_audio=false
* fix conditions for joinFullAudioImmediately and joinFullAudioEchoTest | remove old format
* remove extra param in getItem
* recover audioLocked | only set getEchoTest if doesnt exist
As explained in #11143, disabling audio filters is desired in some scenarios.
This basically adds an option for user to disable default constraints.
When user doesn't change this value in Settings > Application, the default
value for each audio constraints is retrieved from settings.yml.
When user changes this value in Settings > Application, audio
filters (AGC, Noise Supression and Echo Cancellation) are all set to
true/false, according to the value selected in the Settings GUI.
To start it simple, we decided to not to add a different setting in the GUI for
each audio contraint. This may be added in the future, though (perhaps in a
dedicated Audio Settings tab)
This is related to #4873
Changes to the current (v2.2) default configuration of the guest feature.
The ideal is to keep the simplified guest feature as default (`authenticatedGuests=false`)
but we also need to be in sync with Greenlight settings to make this happen.
Greenlight will have to re-add the `guest=true` param on user's join API call when ASK_MODERATOR
is set as guest's policy.
As explained in #11143, disabling audio filters is desired in some scenarios.
This basically adds an option for user to disable default constraints.
When user doesn't change this value in Settings > Application, the default
value for each audio constraints is retrieved from settings.yml.
When user changes this value in Settings > Application, audio
filters (AGC, Noise Supression and Echo Cancellation) are all set to
true/false, according to the value selected in the Settings GUI.
To start it simple, we decided to not to add a different setting in the GUI for
each audio contraint. This may be added in the future, though (perhaps in a
dedicated Audio Settings tab)
This is related to #4873
In some scenarios, there's no need for the browser to apply such audio filters. For example, when user's microphone already does audio filtering (echo cancellation, noise supression ...).
This commit doens't change the current behavior (filters still follow browser's default config): admins need to uncomment/set these values if disabling/enabling specific filters if desired.
This is related to #4873
Some browsers seem to (occasionally) not return the getUserMedia promise call and the
user gets stuck in this state unable to share her/his webcam.
Since enumerateDevices still works even on a gUM rejection this includes a racing
timeout that skips gUM. Configured at settings `gUMTimeout`.
Reproduced with Windows 10 Chrome 87.
Audio client logs already cover audio session progress the way we need.
This avoids keepAlive and other unnecessary messages to be logged in browser's console.
If setting is not present, default value is set to false.
This was added as an option (websocketKeepAliveInterval), which is the interval to send keep alive messages.
Setting websocketKeepAliveInterval to 0 disables the keep alive, producing the same old behavior.
This helps avoid websocket disconnection due to socket inactivity, preventing it to unnecessarily reconnect.
Also, sometimes reconnect fails and error 1005 is triggered.
Fixes problems reported in #10985.
Also reduces occurrences of error 1005.
Added new SFU broker for screen sharing
Removed kurento-extension entirely
Added inbound and outbound reconnection procedures
Improve UI responsiveness when sharing
Add reconnection UI states
Redo error handling
Refactor actions-bar screen share components. Make it smarter with less prop drilling and less re-rendering. Also more readable. Still work to do in that I think
Add a connection retry procedure for screen presenters when they are sharing; try a configurable amount of times when failure is triggered, with configurable min and max reconn timeouts and timeout increase factor
Make local preview attachment smarter
ADD PARTIAL SUPPORT FOR AUDIO SHARING VIA SCREEN SHARING WITH GET DISPLAY MEDIA, RECORDING STILL NOT SUPPORTED!!!
Fixed two occurrences where the tryGenerateIceCandidates workaround rejected without an error, which borked the callers error handling
Also put it behind a config flag. This workaround used to be important when Kurento didnt infer prflx candidates properly, but that`s no longer the case. With the flag, we can disable the workaround to see if there`s any visible regression and hopefully remove it down the road
This adds the possibility to configure the SIP Via header to plain WS to allow reverse proxying from WSS to WS, internally, to work around a bug in freeswitch where the WSS stack would get deadlocked due to a still unidentified bug in there that has to do with SSL termination
For some reason (still investigating), using turn/coturn on 443/tcp makes firefox's iceGathering process (during echo test) takes 12+ seconds (tested on webrtc's trickle page with multiple instances).
This was found when testing the current default (15s) on production with a private turn/coturn server on port 443/tcp. For default bbb setup (stun only), echo test still runs fast.
To avoid adding extra delay to iceGathering on this scenario (Firefox + turn on 443/tcp), i am just setting the default value back to the 5s (old default).
So , for those who wants to reduce the 1004 occurrences, increasing the iceGatheringTimeout could help (just be aware this adds delay on the mentioned scenario).
Added a default 'MEDIA' option: iceGatheringTimeout. This option allows admin to set a higher ICE gathering timeout, which can help when getting ICE errors during audio negotiation (eg 1004)
Default value set to 15s (current default is 5s).
This considerably changes the way we process audio signaling and start audio elements in user's browser.
We now avoid using AudioContext element for both microphone and listenonly calls, once it is unstable for some iOS devices (cracky audio, user stops hearing audio after a while).
Increased default value for listenOnlyCallTimeout: this avoids activating FreeSWITCH's fallback when ICE negotiation takes longer than 15sec (tested on DO).
Increased listenonly logs.
This fixes#8133#10388
This patch switchessing a somewhat more modern default for the camera
profiles by requesting 720p for the `high` and `hd` profiles like this.
This was discussed on the developers list at:
- https://groups.google.com/g/bigbluebutton-dev/c/PL3kXV9pZmo
This patch adds a short comment block to the HTML5 client's
configuration file explaining the options available for the camera
profiles. Having this in the configuration file helps as a quick
reference and means that administrators do not need to find external
sources of information.
Refactored STUN/TURN fetch to be done only once, when successful, per session and cache it in mem to avoid too many reqs. Current way is a bit dumb, this should increase reliability a bit more. The caching is configurable so folks who want to use very short lived TURN credentials can disable it
Add a fallback STUN config option to be used when the default STUN/TURN fetch fails
Clean the safari/no candidate generation pre flight check from 3rd party STUNs
Fix deadlock in audio join when STUN/TURN fetch failed
This adds support for screensharing on Safari 12.1+ or whichever has getDisplayMedia enabled
To be conservative and backwards compatible, I`ve maintained the current gUM-based code available as a fallback for those still using older versions of FF or Chrome
Made screenshare constraints configurable. The constraints config will be piped directly to gDM, so it`s just a regular gDM constraint dictionary
We need to increase the length of the voice conference. If we have lots of meetings running,
there is a high chance of collision.
Need corresponding changes to FreeSWITCH dialplan.
In bbb_echo_test.xml, change to `expression="^echo(\d{5,11})$"`.
In bbb_conference.xml, change to `expression="^(\d{5,11})$"`.
