Reduce server side logging

Add flag to enable/disabled custom heartbeat

Use INFO message as default for sending dtmf on call transfer

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

Don't warn about sip.js if sipjsHackViaWs=true

Removed string interpolation for server debug and verbose logs

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.