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 listen only reconnection handling
Added proper error handling; now all errors have proper mapped codes which are funneled through to audio-manager logger and should be easier to gauge types of errors
Fixed botched reconnection error rejection, audio modal shouldnt be stuck anymore when it fails
Remove every tie that listen only bridge had to kurento-extension
Moved bbb-webrtc-sfu utilitaries to properly named folder
Logging improvements to base broker
Added onerror/onstart/onended callback interfaces to base broker
Extracts most of the common bbb-webrtc-sfu WebSocket setup, handshaking and message broker procedures that was scattered among HTML5 components (video, screenshare and listen only) into a base class suitable for inheritance
BigBlueButton already allows mirroring the users own webcam as a global
setting set by administrators. Users have no way of choosing this on
their own.
This patch turns this functionality into a user setting for all webcams.
Every camera menu now gets a “mirror” entry.
The global setting is still used as a default value, keeping the current
behavior as it is to not confuse users.
Instead of sending using rfc4733 standard, we use INFO message for all transfers
INFO message was used in older SIP.js version. Although this is not a standard for sending DTMF tones, this has more reliability (once it sent over TCP)
This might reduce occurrences of 1008
This is the same behavior we used to have on older sip.js version code
By doing this we reduce errors when user try to perform join/hangup during an websocket reconnection
This happens because FreeSWITCH is not able to parse the "From" header when it has multiple occurrences of ':'. So user is not able to join audio.
To fix, we now changed the "callerId" to use the base64 value of the user name, instead of directly using user's input (the callerId format keeps being a triple like this: <user_id>-bbbID-<base64_encoded_name>).
Once this callerIdName is encoded at the same point it is generated, there shouldn't be server side effects for changing this value; except for those places where the callerName is retrieved by splitting this triple (such as the voice talking-indicator, as described below).
Updated the talking-indicator to retrieve the username from User's object, instead of retrieving from the one username generated by splitting the callerId triple.
This problem also happens in versions <= 2.2.26.
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.
