Add API to pass internet online/offline status to LJM
which allows it to make more informed decisions about
reconnects.

- Use getDisplayMedia on browsers where it is supported.
- desktopSharingChromeDisabled and desktopSharingFirefoxDisabled will no longer be supported.

Enable/disable this feature using 'capScreenshareBitrate' testing flag in config.js and send analytics
events to Amplitude to indicate which type of screensharing is enabled.

* feat(screenshare): support remote wireless screenshare

- ProxyConnectionService is exposed and meant to be the
  facade for creating and updating a peer connection with
  another peer, outside of the muc.
- ProxyConnectionPC wraps JingleSessionPC so the peer
  connection handling can be reused.

* attempt to make more configurable

* feat: Adds the user's region to presence. Fixes typos.

* feat: Measures end-to-end RTT with ping over the data channel.

1. The example was using the Promise return value of JitsiMeetJS.init
which is no longer possible/correct after commit "core: refactor
initialization not to return a Promise".

2. We went back and forth with the value returned by JitsiMeetJS.init:
we initially didn't return a value, then we started returning a Promise,
and now we're not returning a value. Whether we'll go back to returning
a value is up in the air. Anyway, the return value is practically
determined by the last in a chain of function calls: JitsiMeetJS, RTC,
RTCUtil. Since the chain is not really documented, it will not hurt much
to make it easier to refactor the chain by "composing" the functions.

There is nothing asynchronous about the initialization process (anymore), thus
turn it into a synchronous method.

In addition, WebRTC support is absolute, it cannot change from not being
supported to being supported (as it plreviously could, thanks to Temasys) so get
rid of the ancillary logic to support that.

Last, introduce a way to check if WebRTC is supported in the current
environment: JitsiMeetJS.isWebRtcSupported().

* 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

* Switches camera id to mandatory when using old gum flow.

When it fails we retry with different resolutions, and if that doesn't work we remove device id and let gum to decide which device to use.

* Fixes comments.

* log: Logs when analytics are disposed, drops events early.

* fix: Adds a missing parameter to a function call.

In order to include browser capabilities in RTCBrowserType I needed to
refactor RTCBrowserType.

* 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().

The error handling block for unsupported resolution can be
skipped because native gum should be doing automatic retries
at lower resolutions and leaving in the handling will only
cause an infinite loop. Analytics for the error still get
sent in the error handling directly below the retry logic.

* 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.

- Add a new browser check so adapter shim usage can be gated.
- Get track resolution for stats from the track itself to account
  for browser resolution fallback logic. Do this only if
  we can be sure adapter has shimmed it in.
- Create a new getUserMediaFlow, with RTC being the orchestration
  for various RTCUtils calls.
- Remove connection quality stat "resolution" which was being
  emitted but not used but listeners.

ESLint 4.8.0 discovers a lot of error related to formatting. While I
tried to fix as many of them as possible, a portion of them actually go
against our coding style. In such a case, I've disabled the indent rule
which effectively leaves it as it was before ESLint 4.8.0.

Because the implementation of LocalStorage in jitsi-meet at the time
of this writing is asynchronous with respect to loading and saving,
delay the generation of machineId in order to allow LocalStorage to
load any previously saved value.

Fixes crash on undefined var in case get user media was not called,
before the conference started.

* ref: Moves the deployment info variables to config.js

instead of using globals.

* feat: If configured, adds the user's region to presence.

* fix: Guards against accessing undefined properties,

and uses the crossRegion variable from the config.

* style: Fixes formatting.

Firefox's getUserMedia can take longer than 500ms to complete. This
causes the permission prompt timeout to fire even if the user has
given permission to audio and video. There may not be a reliable
way right now to check if the user has given permission to both
audio and video so dumbly increase the timeout instead.

Reorders JitsiTrackEvents.TRACK_AUDIO_LEVEL_CHANGED event arguments by
putting TraceablePeerConnection at the end. This way it's easier to
treat it as "library internal".

* move all local deployment properties into  window.jitsiAnalyticsPermanentProperties property
no longer need to set jitsiRegionInfo from external_connect, now set from Jitsi meet local.html, can be customized by deployment

* change to using shorter lines by extracting longer property name into a shorter local variable name

* changed to using more generic variable name jitsiDeploymentInfo after discussion with the team
added comment describing source of this variable

* 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

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

For audio previewing, some browsers cannot have multiple local
audio streams at once. For audio output previewing, local
stats need to be collected to detect changes in volume level.

Pull Request #435

* Adds peer connection statuses (active/inactive/interrupted/restoring).

The peer connection used to be check with is connection active. Adding new states like active, inactive, when the connection was intentionally stopped by the bridge, interrupted due to network problem and restoring, which is the state of a peer which was inactive as being out of lastN and is now entering lastM and eventually will become active.

* Adds timers to track restoring status.

If restoring too long we set the connection status to interrupted. We save timestamps for all participants entering lastN and set the status for the connection to restoring for 5 seconds, or till it goes into active or exits lastN.

* Adds initial connectionStatus of newly created participant - active.

* Fixes some jsdocs.

* Uses symbols for strings and use of map, fixing comments on PR.

* Removes symbols.

* 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.