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

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

This adds the possibility to configure the SIP Via header to plain WS to allow reverse proxying from WSS to WS, internally, to work around a bug in freeswitch where the WSS stack would get deadlocked due to a still unidentified bug in there that has to do with SSL termination

Force using plan-b as default sdpSemantics for chrome

Although Chrome's default is now unified plan, Chrome <-> FreeSWITCH ICE connection fails for some Chrome installations (specially those running on Windows).
FS ICE fails when Chromes's SDP has "a=mid:<index>" (instead of "a=mid:audio").
This fixes Error 1010 and situations where echo test takes too long.
This fixes #6414 regression, once we do the same older version of SIP.js used to do.

Use iceConnectionState to monitor ICE connection status

We now use both peer's connectionstatechange and iceconnectionstatechange to monitor ICE state for audio sessions.
The same way we did with old sip.js version, we leave iceconnectionstate trigger audio actions , such as connect, disconnect, reconnect.
We still listen for 'failed' state for connectionstatechange event, because chrome triggers this (tested on 86+).
This should reduce the audio error 1010 ocurrences, once some browsers (specially Chrome/Android) don't trigger connectionstatechangeevent.
This might reduce problems reported in #10708, which still needs more investigation though.

Remove unwanted debug logging of PDF binaries

Add error boundary to presentation upload modal

Prevent users from sending multiple votes per poll

Correctly Map Websocket error and better monitoring of ICE events

This gives more information about ICE connection, combined with onconnectionstatechange event

Maps WebSocket's 1006 error to BBB's 1002, the same way it was done with old sip.js version
Set user agent's number of reconnection attempts to the same value as older sip.js version

Fix presentation downloadable tooltip label

Properly stops userAgent / peer when audio connection/reconnection fails

Changed the maximum attempts of the UserAgent reconnection (this should be changed when binding audio's websocket to meteor's connection state).
Added a log to monitor WS reconnect attempts.

Prevent 1005 error log when user close/reload bbb's window/tab

When closing/reloading tab with active microphone, audio exits successfully but a wrong log-error (1005) is shown.
We now process closing/reloading tab the same way we do when user hangup the call.

Change default value of iceGatheringTimeout to current's SIP.js default

For some reason (still investigating), using turn/coturn on 443/tcp makes firefox's iceGathering process (during echo test) takes 12+ seconds (tested on webrtc's trickle page with multiple instances).
This was found when testing the current default (15s) on production with a private turn/coturn server on port 443/tcp. For default bbb setup (stun only), echo test still runs fast.
To avoid adding extra delay to iceGathering on this scenario (Firefox + turn on 443/tcp), i am just setting the default value back to the 5s (old default).
So , for those who wants to reduce the 1004 occurrences, increasing the iceGatheringTimeout could help (just be aware this adds delay on the mentioned scenario).

Audio Improvements

The component expects a boolean but isIphone property return an array ["Iphone"] if the user is from an iPhone

They both return an array with ["Iphone"] if you're from an iphone but the component expects bool variables.

Added a default 'MEDIA' option: iceGatheringTimeout. This option allows admin to set a higher ICE gathering timeout, which can help when getting ICE errors during audio negotiation (eg 1004)
Default value set to 15s (current default is 5s).

Sometimes, when user already joined audio session, RTCPeerConnection may
find new ICE candidates, which triggers 'connected' state for peer's
'onconnectionstatechange' event. When this happens we process this
new state the same way when user is not running an audio session, which
makes html5client popup an annoying 'Audio Connected' message.
The audio keeps working fine, but this can make user think that there's a
connection issue, or the audio is reconnecting, while audio is ok.

Dont include extra dot in instructure media url

bbb-webrtc-sfu wait for multiple Kurentos