Remove redundant _channelOpen flag from RTC and rely on the channel's isOpen() getter instead.
This fixes an error throw when the _channelOpen flag was not updated while the channel's been
closed by the bridge.

Delays e2e encryption initialization until any encryption key is set.
Recreates peerconnections in order to enable insertable streams
only if the e2ee is used. This is to avoid bug around insertable streams
which may cause audio issue when the main JavaScript thread is loaded:

https://bugs.chromium.org/p/chromium/issues/detail?id=1103280

This is useful for testing insertable streams related issues.

Co-authored-by: Saúl Ibarra Corretgé <saghul@jitsi.org>

* 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

It's supported in Chrome >= 65.

Use the W3C Permissions API where possible. At the time of this writing this is
supported in Chrome >= 64.

* feat(conference): expose method to select multiple participants

* squash: fix typo

* squash: reduce to one code flow

* squash: default to empty array to mimic previous null default

* squash: fix typos, change docs, default params

* squash: remove defaults

* squash: update comment about array arg

* ref(video-quality): cache max frame height and send on channel open

Currently it is possible to try to change the max receive video
frame height before the data channel is open. In that case an error
will be thrown. This change makes it so that the desired frame height
is saved and sent on channel open, avoiding the thrown error the
max receive video frame height logic is exercised through the
JitsiConference api.

* squash: do the second part of the actual fix

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().

This commit will append "-" + tpc.id to every local 'MSID', 'cname',
'label' and 'mslabel', before feeding the local SDP to the Jingle layer.
It will make stream IDs unique across TraceablePeerConnection instances
and prevent from conflicts in some corner cases.

For example this will fix a problem where if the client drops
the conference without leaving the XMPP MUC gracefully and will join
the conference again without recreating the local tracks it would lead
to the MSID conflict, because the stream is still advertised by
"the ghost" participant.

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

When a mac's default audio output is changed, audio elements
are not being updated with the new default. A default audio
output change in the OS fires a DEVICE_LIST_CHANGED event,
so use that to always update the sinkId.

Also, use wrapAttachMediaStream for newGumFlow to be ensure
the proper sinkId is set on audio elements being coupled
a remote track.

* revert: Reverts be665cbff7.

* ref: Renames "peerjid".

* ref: Refactors the initialization of a peer connection.

* feat: Re-implements the A/B test for the "suspend video" feature.

* squash: Deep copy.

* ref: Renames forceSuspendVideo to abtestSuspendVideo.

- 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.

This is performed by stripping all traces of H.264 from the local SDP before
passing it to `setLocalDescription`.

* 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.

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".

* doc(JitsiConference): deprecate 'isInLastN'

The 'isInLastN' method should not be used for the UI purposes, but
ParticipantConnectionStatus value should be used instead.

* fix(ParticipantConn..Status): speed up INACTIVE transition

Before this change when user's video stops playing, after user is
removed from last N we were waiting 2 seconds, before going to INACTIVE
state. This commit reduces the time to 500ms for such case.

* fix(ParticipantConn...): reduce logging

Reduce logging verbosity.

* fix(ParticipantConn...): handle LastN == 0

When LastN is set to 0 we should not rely on video playback and last N
set for figuring out participant connection status.

* fix(JitsiConference): undefined participants

Fixes a crash when this.participants field is accessed from _init.

Do the same which was done for the selected endpoint. That is if
an endpoint is pinned, before data channels are available then the value
should be cached and sent as soon as they are opened.

* 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.

If it's set before the data channel is open, save it and send it as soon as the
data channel is open.

The first time a lastN was received it was ignored because the old lastN was
null, handle that case properly.

Furthermore, use default function parameters to avoid a conditional.

* 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