Reduce delay for the first reconnection attempt of audio's websocket

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

Added check for default stun.freeswitch.org

Do not trigger reconnect when ICE connection terminates before hanging up process is finished

This could leave users to have your audio reconnected in the main room, while joining a breakout room
Some information can be found in #10528

This reverts commit 0a601359.

Breakouts: do not allow mute via who-is-talking button

Handle chat messages in sync way

Set listenOnly to false when user left voice

The message/event UserLeftVoiceConfToClientEvtMsg is used when user leaves mic and listenonly, but it only sets to false the 'voiceJoined' (which represents the <hasVoiceVoiceJoined> property in BBB's XML API.
We now also set to false the 'listeningOnly' (which represents the <isListeningOnly> property in BBB's XML API). Setting both to false is not a problem, once 'MIC' and 'ListenOnly' states are mutually exclusives
Fixes #10852

Filter 'name' field in user's search

This is a part of #10848, as discussed in https://github.com/bigbluebutton/bigbluebutton/pull/10848#discussion_r525480434

Fix audio modal buttons (MIC/LISTEN) after errors AND prevent Error 1006 when user has two or more occurrences of ':'

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.

Tweak sockjs timeout value

Allow user bigbluebutton to delete recordings

This patch modifies `bbb-record` allowing the user `bigbluebutton` to
delete recordings. The user has all necessary access rights, meaning
that the deletion works without a problem and the check for root does
not protect anything. The users owns the data after all. The current
check just makes things less convenient.

Pulled HTML5 locales on Nov 13, 2020

[html5/webrtc] Fix tryGenerateIceCandidates rejections and put it behind a flag

Sanitize parameters of API's. Fixes #10818

Split error message from session token in bbb-web enter API

Simplified redis debug and placed behind check

Increase private key size for dtls-srtp.pem to 4096

Using this larger value helps reducing ocurrences 1005/1010 errors for Chrome, avoiding an openssl's error which interrupts the dtls handshake (Chrome triggers "DTLS timeout expired" error)
This key size is also the default value used by freeswitch on switch_core_cert.c

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

[html5/audio] Add a configurable hackViaWs to SIP.js. Part of a workaround for #9667