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
