The code for handling device availability has been disabled for a long time,
plus it's ill named since it represents 2 abstractions: lack of permissions and
lack of devices.

Time for it to rest in the git graveyard.

This simple change helps to prevent `'The conference is has been already left'` error after kick. 
`Conference.leave` will detach chatRoom instance itself.

* feat(recording): frontend logic can support live streaming and recording

Instead of either live streaming or recording, now both can live together.
The changes to facilitate such include the following:
- Remove the old recording.js module which allowed one recording at a time.
- Remove events for live stream url changes as the url is now part of a
  sesssion and not fired independently.
- Availability of sipgw and recording are gone. Instead sessions have a
  failure reason. For sipgw sessions, store that failure and emit it to
  listeners.
- Create a new recordingManager singleton that can start/stop sessions
  and handle updating known state of those sessions. Known state is
  emitted through one event.
- Create a JibriSession model to encapsulate state of a session.

* update comments, use map to store sessions

* always pass in focusmucjid

* try to fix jibrisession docs and remove default null

* feat(recording): show the YouTube live stream URL

- Pass you_tube_broadcast_id to Jibri so it can create a
  YouTube link for the live stream.
- Emit the liveStreamViewURL (the YouTube link) when it
  is received from the participant doing the recording.

* set member.liveStreamViewURL only when truthy

* wip: initial version of the new AnalyticsAdapter.

* ref: Restructures the ICE duration and state change events.

* ref: Restructures the JitsiLocalTrack events.

* ref: Restructures the TTFM events.

* ref: Updates the user feedback event.

* ref: Restructures the _CONNECTION_TIMES_ and TTFM events.

* ref: Restructures the BRIDGE_DOWN and NO_DATA_FROM_SOURCE events.

* ref: Restructures the FOCUS_LEFT event.

* ref: Restructures the DATA_CHANNEL_OPEN event.

* ref: Removes the ICE_FAILED event (it is a duplicate of a state change event).

* ref: Restructures the device list events.

Uses one event per device, since the new format does not allow non-atomic attributes.

* fix: Does not obey "unmute" commands from the focus.

* ref: Restructures the "remotely muted" event.

* ref: Restructures the CONFERENCE_ERROR events.

* ref: Removes the CONNECTION_INTERRUPTED event

We can use ICE_STATE_CHANGED instead.

* ref: Renames isreconnect to isReconnect.

* ref: Removes the CONNECTION_RESTORED event. Use ICE state changes instead.

* ref: Restructures the p2p events.

* ref: Restructures the jingle events.

* ref: Restructures the RTP statistics event.

* ref: Restructures the CONNECTION_FAILED and DISCONNECTED events.

* ref: Restructures the getUserMedia analytics events.

* ref: Cleans up AnalyticsEvents and restructures some of the events.

* fix: Adds error logs to the analytics adapter.

* ref: Refactor Statistics.sendEventToAll

Renames to sendEventAndLog, supports the object-based API, uses the
function where appropriate.

* fix: Addresses PR feedback.

* fix: Addresses Lyubomir's feedback.

* ref: Remove unused functions, adds documentation.

* feat: Adds a Statistics.sendAnalytics shortcut.

* ref: Uses the conference name as the default containerId.

* fix: Adrdesses Lenny's feedback.

* fix: Addresses more feedback.

* fix: Uses 'operational' as the default event type.

* doc: Updates the documentation.

* fix: Fixes adding of permanent properties.

* ref: Uses consistent naming for events' attributes.

Uses "_" as a separator instead of camel case or ".".

* feat: Adds the conference name as a permanent property automatically.

* ref: Don't expose Setting.machineId.

* fix: Adds a "p2p" attribute to jingle events.

* ref: Uses "action" instead of "name".

* ref: Uses underscore in events' attribute names.

* ref: Logs a message to the logger/console

instead of callstats in sendAnalyticsAndLog().

Adds rtc.LOCAL_TRACK_SSRC_UPDATED event to be emitted when ssrc is updated for a local track.
Fixes adding rtc listeners for CREATE_ANSWER_FAILED, CREATE_OFFER_FAILED, SET_LOCAL_DESCRIPTION_FAILED, SET_REMOTE_DESCRIPTION_FAILED.

* Changes initialization of videoSIPGW.

* Adds some errors returned on creating videoSIPGW session.

* Fixes sending videoSIPGW session STATE_CHANGED event.

* Fixes sending jibriIQ, no result status is received.

* Adds VIDEO_SIP_GW_SESSION_STATE_CHANGED to JitsiConferenceEvents.

* Fixing comments.

* ref: Simplifies the logic for handling an incoming jingle session-initiate.

* fix: Don't redundantly log cross region

information under a field name called "label".

* cleanup: Simplifies code. Adds the userAgent as a permanent property

for statistics (so that the client doesn't have to).

* ref: Names the parameter which specifies the name of the event "eventName".

* ref: Extracts event names to AnalyticsEvents.

* ref: Exports and imports constants individually.

* fix: Fixes CONNECTION_TIMES event names.

* ref: Arranges constants alphabetically.

* ref: Adds line breaks.

via JSON messages received via the Chatroom

* feat(quality-slider): expose video constraint method and data channel opening

* squash: rename to data channel opened

Some browsers may not support data channels (hi there, Edge!) and sometimes it
may not be desirable for other reasons.

* feat: multiple, simultaneous RTP stats

Makes it possible to have remote RTP stats running for more than one
peerconnection at a time.

* feat(stats): report RTT all the time

Will report JVB RTT (and end to end) while in P2P mode and vice versa.

* fix(JitsiConferenceEvents): remove CONNECTION_STATS

CONNECTION_STATS event is no longer emitted.

* fix(AvgRTPStatsReported): users with no video

Do not include FPS == 0 in average remote FPS calculation. Report NaN
for local FPS when video muted or no video device. NaN will be reported
for avg remote FPS if no video is received.

* fix(AvgRTPStatsReported): reset total packet loss

* feat(AvgRTPStatsReported): report 'screen' FPS

Will report average FPS for screen videos separately from camera videos,
but only if available (camera video reports NaN FPS when not available).

* fix(AvgRTPStatsReported): end2endRTT

Needs to report JSON with value.

* feat(AVG RTP stats): separate audio and video bitrate

Will report average audio and video bitrates separately.

* doc(JitsiConference): try to improve comment

* fix(AvgRTPStatsReporter): remove confusing reset

There's no a clear reason for doing reset there.

* ref(AvgRTPStatsReporter): rename var

* ref: store SSRCs as a number

Converts all the places where SSRCs where stored as string to use
numbers.

* doc(RTPStatsCollector): getNonNegativeStat

Fixes invalid description about returning NaN.

* ref(JitsiConf...EventManager): simplify for..of

* ref(JitsiRemoteTrack): throw TypeError

Will throw a TypeError when 'ssrc' is not a number.

* fix(RTPStatsCollector): invalid reference

* doc(RTPStatsCollector): getNonNegativeStat private

* ref(RTPStatsCollector): simplify for..of

* fix(SSRCs): check for negative value

Will not accept negative SSRCs, since those are supposed to be unsigned.

Webpack 2 defines module.exports as a read-only property in ES6 modules.

* ref(CallStats): cleanup constructor

Changes CallStats constructor to not take the whole JingleSessionPC as
it only needs an alias and the TraceablePeerConnection instance.

Describes the arguments in JSDoc.

* ref(CallStats): rename var

Everything is callstats c'mon...

* ref(CallStats): remove _checkInitialize

The _checkInitialize was trying to workaround CallStats lib issue
without really checking for any specific type of error on whether or not
it makes sense to retry.

Also it depended on some internal fields of 'callStatsBackend' and was
binding 'initCallback' to the backend instead of CallStats instance
which made no sense (it means it's very likely this functionality was
broken anyway).

It would be hard to fix it in a clean way, because CallStats instance
fields would have to be stored in static variables in order to make
the initCallback work (called from '_checkInitialize').

We also need to have more than one CallStats instance running at the
same time, because of the P2P which makes things even more complex.

* fix(CallStats): do not catch 'sendApplicationLog'

Wrapping 'sendApplicationLog' in 'tryCatch' will result in an endless
loop, because it will be logged on the logger.error which then tries
to send the logs immediately again.

* ref(CallStats): do not use call on static

There's nothing more confusing that seeing 'this' in a static method.
Wow maybe these methods are not really static !?

* ref(stats): fix var name

* feat(stats): report P2P to CallStats

Will create new CallStats fabric for the P2P peerconnection in order to
log peer to peer connections.

Refactoring was required in the statistics and CallStats module to be
able to have more than one CallStats instance. Because each CallStats
fabric reports one peer connection now each CallStats will correspond to
one TraceablePeerConnection. CallStats instances are now stored in a Map
mapped by the TraceablePeerConnection.id field.

In order to be able to execute some global/static CallStats reporting
methods all Statistics instances also need to be stored in a static
field.

CallStats API backend(new callstats()) will be initialized only once for
the values provided in the first call to initializeBackend. It is not
possible to have more that one CallStats backend running at the same
time (at the time of this writing). If we would have a routine for
disposing global "Statistics" module we could try to cleanup static
reference and allow to initialize it with new values (but no such use
case yet).

* ref(CallStats): move initCallback

Since there is no alphabetic order preserved in this file anyway at
least place it closer to it's usage.

* ref(CallStats): remove tryCatch

Temporarily removes tryCatch to make the ES6 conversion easier.

* ref(CallStats): convert to ES6

* style(CallStats): fix indentation

* fix(statistics): use import for CallStats

* ref(CallStats): convert static methods

Some of the methods should not be static, because it only make sense
to call it when there is CallStats instance available.

* ref(CallStats): rename var

* doc(CallStats): remove misplaced comment

* chore(CallStats): remove invalid eslint comment

* fix(CallStats): undefined CallStats namespace

If no CallStats ID namespace option is provided the conference will be
reported without it.

* style(stats|CallStats): remove extra lines

* fix(CallStats): do not log error from tryCatch

GlobalOnErrorHandler calls logger.error anyway.

* fix(CallStats): cleanup tryCatch

If I understand correctly our initial intention with doing tryCatch was
to avoid crashing the whole app in case the CallStats backend would
crash. With this commit the tryCatch is done by wrapping original
backend instance methods or using explicit try catch block where
the method is called only from one place or a value needs to be returned
in case of a crash.

* ref(CallStats): make backend a static var

* fix(CallStats): invalid eslint comments

* ref(CallStats): use for..of

* ref(CallStats): fixes around REST args

* ref(CallStats): rename var

* ref(CallStats): reorder static methods

Also renamed some callbacks

* doc(CallStats): adds some docs

* ref(CallStats): make methods not static

Both 'sendDominantSpeakerEvent' and 'sendScreenSharingEvent' methods are
not really static as they require instance to be called.

* fix(CallStats): invalid key

* fix(CallStats): reduce amount of debug logs

* feat(p2p|CallStats): log hold/resume

Will put CallStats fabric for the JVB connection "on hold" while in p2p
mode.

* doc(CallStats): add FIXME

* doc(JitsiConference,CallStats): typos and renaming

Ignores start audio/video muted requests received by jicofo.

* Adds video sip gw handling.

Parsing presence and availability of the service. Adding possibility to start/stop a video sip gw session.

* Fix comments.

* Adds more jsdoc about create session and renames import to use the filename is imports.

* ref(RTC): store remote tracks in peer TPC

In order to implement P2P <-> JVB connection switching we need to
be able to associate remote tracks with the TraceablePeerConnections.

* feat(ChatRoom): multiple presence handlers

Add support for more than 1 presence handler per tag name.

* feat(JitsiLocalTrack): update stored MSID

* ref(stats): add peer connection arg to BYTE_SENT_STATS

Required to store local SSRCs in TraceablePeerConnection.

* ref: change local SSRCs strategy

* fix: generate recvonly SSRC if 0 video tracks

Video SSRC has to be generated for the recvonly stream if there are no
video tracks in the PeerConnection.

* feat: add "attach" and "detach" methods

* feat(jitsi tracks): improve logging

Adds toString methods and improve log messages around local and remote
tracks.

* ref(modify SSRCs): optimisations + fixes

- adds _doRenegotiate to JingleSessionPC that wraps some of
  the duplicated logic
- fixes problems with attach/detach
- renames methods to reflect what that they really do (operate on
  JitsiTrack rather than streams)

* ref(JingleSessionPC): remove duplication

Extracts common code for the 'modificationQueue' execution

* ref(VideoMuteSdpHack): rename, add docs, fix minor

Renames, adds docs and moves 'modified' flag and media direction
modification.

* ref(TPC): add 'isSimulcastOn'

* ref(MungeLocalSdp): move to RTC module

* ref: move "ufrag" events to the RTC module

* ref(JitsiConference): use promises for mute

* feat(XMPPEvents): add CONNECTION_ESTABLISHED

* feat: add peer to peer

* fix(P2P): deal with everyone's a moderator + fixes

* feat(P2P): implement "backToP2PDelay"

* fix(TPC): crash on FF accessing LD/RD

Firefox return null/undefined for localDescription/remoteDescription
objects if they have not been set yet, while Chrome does return empty
object instead. This commit makes the behaviour consistent by making
sure that at least empty object is returned for all browsers.

* fix(TPC): replace isFirefox with feature

* fix(JSPC): fix renegotiate crash on FF

FF does not allow to call 'createAnswer' in 'have-local-offer' state

* fix(JSPC): fix addIceCandidate crash on FF

* doc(JitsiConference): fix outdated comment

To be squashed with "ref(ChatRoom): remove unnecessary JingleSessionPC dependency"

* style(JitsiConference): rename arg to "jingleSession"

* feat(stats): add 'p2p' to 'transport'

The new p2p field will inform whether the transport comes from the peer
to peer type of connection or not.

* doc(TPC): describe local maps

* fix(P2P): multiplex between JVB and P2P ICE status

Will make sure that when in P2P mode the conference will be updated
with the ICE state coming from P2P and when in the JVB mode will get
the JVB one.

* doc(TPC): fixes docs and adds FIXME

* ref: use 'doesVideoMuteByStreamRemove'

* feat(P2P): stop P2P when ICE enters FAILED

The conference will switch back to the JVB connection when P2P
connection breaks (ICE enters failed state).

* feat(P2P): "connectivity-error" for ICE failed

Will use "connectivity-error" reason element name when ending P2P
session due to ICE failure.

* feat(xmpp): make P2P Stun servers configurable

STUN servers used in the P2P connection can be configured through
"p2pStunServers" option.

* ref(JitsiConference): use 'getActivePeerConnection'

* fix(P2P): re-create 'dtmfManager'

* ref(P2P): deal with ICE "completed" state

* ref(RTC): rename "owner" to "ownerEndpointId"

* fix(MungeLocalSdp): fix directions

* ref(ParticipantConnectionStatus): use for..of and () =>

* remove double 'l'

* ref: fix ESLint errors

* fix(MungeLocalSdp): adopt to new SdpTransformerUtil

* ref(MungeLocalSdp): use for .. of

* ref(SdpTransformUtil): remove "forEachSSRCAttr"

* fix(SDPDiffer): fix invalid "arrayEquals" call

* doc(MungeLocalSdp): add fixme

* fix(P2PEnabledConference): JVB tracks not added

* ref(JitsiConference): doc + rename mute methods

* ref(JitsiConference): adjust log level

* fix(JitsiConference): remove invalid eslint comments

Some mistake during rebase merge

* doc(JitsiConference): add FIXME

* ref(JitsiConferenceEventManager): stick to "tpc"

* ref(JitsiLocalTrack): use Set for "peerConnections"

* ref(JitsiLocalTrack): simplify expression

* ref(MungeLocalSdp): style + doc fixes

* ref: rename MungeLocalSdp to LocalSdpMunger

* ref(ParticipantConn..Status): rename method

* ref(SignalingLayerImpl): use Map and =>

* fix(strophe.jingle.js): minor style fixes + rename

* doc(XMPPEvents): typo

* doc(P2PEnabledConference): typo and style

* ref(P2PEnabledConference): rename methods

* ref(P2PEnabledConference): do not use "window"

* fix(P2PEnabledConference): cleanup deferred task

* ref(TPC): make options the last arg

* ref(TPC): use Map

* ref(TPC): syntax and other fixes...

* doc(ChatRoom): remove comment

* ref: remove P2PEnabledConference

* fix: remove JSUtil.js

* ref(LocalSdpMunger): re-use 'RtxModifier'

Reuses RtxModifier for injecting local RTX SSRCs as part of
the LocalSdpMunger logic.

* doc(LocalSdpMunger): remove confusing FIXME

* fix(TPC): setLocalDescription for FF

* fix(LocalSdpMunger): crash on react-native

* fix(JingleSessionPC): no events when ended

The instance once terminated should not emit connection state events.

* fix(P2P): do not start P2P on react-native

* fix: log meaningful error

Prior to this change you would see something like:
JitsiConference <error>: null

* fix(JingleSessionPC): no IQs once ended

* fix(JingleSessionPC): Jingle error logging

* fix: arguments order

* fix: make audio SSRC consistent

Audio SSRC needs to stay consistent between detach and attach operations
in order to avoid source-remove/source-add.

* fix(P2P): disable P2P on FF

There are problems with going back from P2P to JVB in FireFox. Other
participants will not see FF video. Looks like something related to
detach/attach.

* fix(JitsiConference): attach local tracks

Local tracks should be attached back to the JVB connection only
if the P2P was established.

* ref(JitsiConference): PR review fixes

ref(JitsiConference): else if

ref(JitsiConference): use getter

doc(JitsiConference): add comment

style(JitsiConference): remove extra line

log(JitsiConference): misleading msg

ref(JitsiConference): rename method

* ref(RTC): del _iteratePeerConnections

* ref: move getUfrag to SDPUtil

* fix(LocalSdpMunger): docs and if check

* fix(TPC): docs and typo

* ref(JingleSessionPC): PR review fixes

ref(JingleSessionPC): rename 'candidates'

ref(JitsiConference): remove extra check

ref(JitsiConference): rename isP2PEstablished

ref(JitsiConference): rename field (typo)

* doc(JitsiConferenceEventManager): typo

* ref(JitsiLocalTrack): rename var

* ref(JitsiConference): PR review fixes

ref(JitsiConference): rename var

doc(JitsiConference): add comment

doc(JitsiConference): add comment

doc(JitsiConference): fix comment

ref(JitsiConference): rename listener

ref(JitsiConference): rename var

* doc(RTC): remove duplicated arg description

doc(RTC): fill docs

* doc(SignalingLayerImpl): remove fixed FIXME

* ref(strophe.jingle.js): remove comment and break line

* style(TPC): formatting

doc(TPC): add FIXME

ref(TPC): remove unused code

doc(TPC): add docs

* doc(JingleSessionPC): mark "send" methods private

style(JingleSessionPC): extra lines