When listenOnlyMode=false, skipCheck=true and skipCheckOnJoin=true, the
audio tries to start a session more than one time, causing it to fail
at the first one (and reconnect after that).
Now we check if user is already connecting before trying to start a new
audio session.
Added some info in settings.yml for the options related to this commit
Closes#12190
When setting skipCheckOnJoin to true, an extra audio channel is created in
FreeSWITCH, after user accepts the echo test. The extra channel is removed
when user leaves the room, but this still may affect performance.
When joining/returning breakouts, audio would always connect
with full audio. This can lead to a performance problem, once
all listenonly users would join full audio, increasing the
number of streams in FreeSWITCH.
We now have a consistent behavior, which is:
1 - The choice made by the user in the main room is predominant:
if mic is active in main room, user will automatically
join mic in breakout room. When returning from breakout
room, user will also join with mic again.
2 - Changes made in breakout room won't have effect when
returning to the main room. This means if user, for example,
change from listenonly to mic in breakout room, the returning
will consider the option choosen previously (listenonly) and
listenonly will be active again in the main room.
3 - If user didn't join audio in the main room, the audio modal
will be prompted when joining the breakout room (this is
a special case of (1))
The following is some technicall information:
InputStreamLiveSelector (component.jsx) now calls
'handleLeaveAudio' function, which is the default
function when user leaves audio (also used when
dynamic devices are inactive).
We now store information about user's choice (mic or listenonly)
using local storage, instead of the previous cookie method (this
was triggering some warnings in browser's console).
Also did a small refactoring to match eslint rules.
Fixes#11662.
Firefox doesn't create a device called 'default' and we were trying
to set this when user is joining the room. We don't do this anymore, letting
devices to be changed when there's some user request.
Moved outputDeviceId inputDeviceId information to be managed in bridge
(just like we do with inputDeviceId), we don't store this duplicated
information in audio container anymore.
Fixed the eslint warning in "playAlertSound(url) { ..."
We are safe to let users try to change input/output devices because the
device list is retrieved from enumerateDevices.
When listen only fallbacks from Kurento to FreeSWITCH, we must guarantee
the muted alert won't be created, speciallly because listen-only's fallback
uses a flow similar to microphone's. Client currently crashes when this
happens: this commit fixes this peoblem.
Allow listenonly users to change output devices
Fixed dynamic audio device change for firefox
Fixed shortcuts for audio join/leave
Show (with a bold font) the current selected device
[performance] Prevent calling mediaDevices.enumerateDevices every time we render
the selector. This adds a delay (~200ms, on my chrome setup) to render this component
[performance] Do not call enumerateDevices to search for new devices, instead we listen on mediaDevices.deviceChange event
Small refactoring and fixed a few errors that were being throw in browser's console
Fixed device selection when this is done in audio-settings modal
Fallback to default device when current device is removed
Truncate device name length
Renamed "Input","Output" labels to "Microphone","Speakers", respectively
Update eslint rule for accessKey
Currently this information is lost everytime breakout-room component is
unmounted, causing the panel to shows wrong information during next renders
Fixes#11333
After ending the notification playback, we set the ".src" property to null, which immediately stop the internal player of mobile browser (tested on Chrome for Android - device list is on #11458).
For the specific list of devices, this prevents the internal buffer error "-61" described in #11458.
Fixes#11458.
* updating old tests + collecting more snapshots [WIP]
* updates old test suites and collects more visual regressions screenshots
* remove snapshots and their collection temporary
* run tests from packages.json
* update test execution command/export constants from .env to core/constants.js
* update tests/puppeteer/README.md file
* update LOOP_INTERVAL variable call and assign timeouts to the webcam share spec
* redefine waitForSelector func in page.js, update chat test suite specs and add poll chat message test spec
* Merge remote-tracking branch 'upstream/develop' into updating-old-tests-visual-with-visual-regressions
* update webcam test specs collecting videoPreviewTimeout and use it to wait for videoPreview selector
* update custom parameters test suite
* update breakout test suite
* update webcam layout test suite
* update multiusers test suite
* update notifications test suite
* update presentation test suite
* whiteboard test suite
* screenshare test suite
* update sharednotes test suite
* user ELEMENT_WAIT_TIME variable from timeouts constants.js
* list TEST CONSTANTS by category
* add poll test suite and assigns the right unassigned timeouts
* set test pages to headless
After audio reconnection, a muted user would have it's microphone unmuted by default, unless muteOnStart is set to true. This fix this problem.
Fixes#9016
* 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
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
When user joins audio and for some reason an error (such as 1001, 1002,...), happens, the user is not able to click "Mic" and "Listen Only Buttons"; except if the audio window is closed and oppened again.
When getting disconnected with 1001 ("websocket closed unexpectedly" error) we were creating a new SIP session, therefore a new FreeSWITCH channel.
While reconnecting the socket, instead of closing the SIP session, we keep it alive during reconnection (audio should keep working in the meantime). When reconnected we keep using this same session (avoiding the creation of an extra one).
We also better handle WebSocket error codes from SIP.js.
FF immediately closes websocket when unloading page, so we now to stop user agent when 'beforeunload' event is triggered, to avoid leaving open sessions in FreeSWITCH when user leaves page.
When refusing ("thumbs down" button) echo test, user is able to select a different input device. This should work fine for chrome, firefox and safari (once user grants permission when asked by html5client).
For output devices, we depend on setSinkId function, which is enabled by default on current chrome release (2020) but not in Firefox (user needs to enable "setSinkId in about:config page). This implementation is listed as (?) in MDN.
In other words, output device selection should work out of the box for chrome, only.
When selecting an outputDevice, all alert sounds (hangup, screenshare , polling, etc) also goes to the same output device.
This solves #10592
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
