If the current browser does not support given stat no error should be
printed.

The peer count excludes the current user, so it should be 1 not 2.

AvgRTPStatsReporter will calculate arithmetic means of 'n' samples
and submit the values to the analytics module. The 'n' value is
configurable through 'avgRtpStatsN' conference config option. When set
to non-positive value the AvgRTPStatsReporter will be disabled.

The following values are reported:
- average upload bitrate => 'stat.avg.bitrate.upload'
- average download bitrate => 'stat.avg.bitrate.download'
- average upload bandwidth => 'stat.avg.bandwidth.upload'
- average download bandwidth => 'stat.avg.bandwidth.download'
- average total packet loss => 'stat.avg.packetloss.total'
- average upload packet loss => 'stat.avg.packetloss.upload'
- average download packet loss => 'stat.avg.packetloss.download'
- average FPS for remote videos => 'stat.avg.framerate.remote'
- average FPS for local video => 'stat.avg.framerate.local'
- average connection quality as defined by
  the ConnectionQuality module => 'stat.avg.cq'

If the conference runs in P2P mode 'p2p.' prefix will be added to
the event's name. Any pending calculations are wiped out on every switch
between P2P and JVB modes and samples have to be collected from
the start.

We ignore stats that were remotely calculated and take into account only local one.

Currently, when a participant joins hidden/video muted, we add a
recvonly SSRC so that the client sends RTCP reports with that SSRC. This
SSRC doesn't have neither simulcast nor RTX enabled. When the user
decides to enable his video camera, we "enhance" this SSRC with params
for RTX and simulcast. However, Chrome does not take into account these
new params because of
https://bugs.chromium.org/p/webrtc/issues/detail?id=7555. By not adding
a recvonly SSRC, we activate the default behavior in Chrome which is to
send RTCP reports with SSRC 1. So, when the user decides to enable his
video camera, instead of updating/enhancing the existing SSRC, we
initalize the primary SSRC with the correct params.

Internet Explorer is throwing an error when doing a for...of over
Sets. This is because IE needs a Set polyfill, which may be applied
after lib-jitsi-meet has loaded. However, lib-jitsi-meet on load
creates two Sets, one for CallStats.fabrics and another for
Statistics.instances. When the polyfill is then applied, the two
existing Sets do not get new methods. The fix is to lazily
initialize the Sets to allow for polyfills to be applied.

For muted tracks, it is possible for undefined to get pushed
into the track's internal containers reference. The fix is
to immediately set the internal container variable in .attach()
to the passed in container value.

Sending the bare jid, speeds call setup as otherwise jigasi needs to do a feature discovery finding the conference component address and use it to construct the room address to join.

Avoids having to do browser checks twice.

Fixes: #423

Ensures that the simulcast ssrc-group appears after any other
ssrc-groups in the SDP so that simulcast is properly activated.

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

Splits the participant connection status detection logic into two
separate flows one for P2P and one for JVB.

Fixes problem where user outside of last N would be reported as INACTIVE
when in P2P mode.

Added some improvements to the detection logic by adding more priority
for the video tracks frozen detection (if supported) over the last N.
When the browser supports this detection and it's playing video we
should not check the last N, because it's less reliable.

* fix(JingleSessionPC): do not log whole object
* fix(JitsiConference): fix a crash

The code will crash if there's neither P2P nor JVB session yet.

ICE connection state analytics for the JVB connection should not be
mixed up with the ones for the P2P connection.

Will emit BEFORE_STATISTICS_DISPOSED event, before the last CallStats
fabric is terminated to make sure that the final logs batch is reported
correctly.

Also adds a check for CallStats instance to queue events when
the backend has been initialized, but there is no CallStats instance
available to report on.

* 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

When muted track was added to conference "mute" operation was
executed again which was not executing anything because the state
of the track was already muted.

If the platform supports it, it calls MediaStreamTrack._switchCamera in order to
switch the camera facing mode at a low level. It's currently only implemented
for React Native.

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

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.

Renames '_applyLocalSSRCMap' and improves description to remove
confusion with SSRC manipulation.

* ref(JSPC): rollback doRenegotiate

Modify the code to the old strategy without shared 'doRenegotiate'
method.

* ref(JSPC): add/remove duplication

Removes duplication between add and remove remote stream methods.

* fix(JSPC): remove Timeout

Removes wait timeout for remote stream added/removed. It should not be
necessary given that the task executes on the modification queue and
the initial offer/answer should execute before
'source-add'/'source-remove'. Jingle 'session-invite'/'session-accept'
is sent, before any other notifications.

* feat(JSPC): verify SSRC changes

Will print an error if there was change to local SSRCs where it should
not happen (video mute on browsers where video stream is disposed on
mute).

* fix(JSPC): not always renegotiate

When adding/removing tracks as mute/unmute it only makes sense to
renegotiate if the initial offer/answer cycle has already been executed.

* ref(JSPC): remove duplication

Removes duplication around add/remove as unmute/mute.

* ref(JingleSessionPC): add local tracks with offer/answer

Refactor the current code to add local tracks together with initial
offer/answer to make this atomic/single operation on
the modificationQueue.

This is required to be able to get rid of 'delayedIceCandidates' list
and to execute 'addIceCandidate' task on the queue. If local track
addition is not bundled with initial offer it will often happen that ICE
candidates are queued, before the offer/answer which is queued only once
the local tracks task is done.

* ref(JSPC): use queue for ICE candidates

Refactors the code to get rid of 'delayedIceCandidates' buffer and
use the modification queue to synchronise with the initial offer/answer.

* doc(JingleSessionPC): improve docs

#416

* ref(p2p): use SDP 'inactive' for JVB

This commit gets rid of detach/attach logic and
replaces it with SDP media direction modification.
Now after switching to P2P instead of detaching local
tracks from the JVB peerconnection only the media direction
will be changed to 'inactive'. This will suspend media
transfer on the JVB connection.

It's meant to mitigate Chrome issue where it would randomly
choose to stop sending audio stream, after we modify it's
local SSRC.

* ref(SdpConsistency): cleanup 'recvonly'

Checking on 'recvonly' is no longer reliable way for detecting
whether video media is recvonly, because it can be set to
'inactive' at any point.

Added a check to make sure that the recvonly SSRC is not generated
more than once from '_createOfferOrAnswer'.

* log TPC instance

Logs the TraceablePeerConnection from LocalSdpMunger and
SdpConsistency as part of log messages. Given that there can be
2 TPCs at the same time it was hard to tell for which connection
a log message was printed.

* fix(SdpConsistency): depend on 'recvonly' direction

When createAnswer/createOffer returns SDP it will contain 'recvonly'
direction to reflect current media stream state even though we set
it to 'sendrecv' previously. Using 'hasAnyTracksOfType(VIDEO)' can
not be used to reliably detect 'recvonly' direction when doing video
mute, because muted video track is still in the TPC's map.

* doc(LocalSdpMunger): update description

LocalSdpMunger is currently used only for muted local video tracks.

* fix(P2P): enable P2P on Firefox

After 'detach/attach' is gone it seems that P2P works fine with Firefox.

* ref(TPC): simplify expression

* ref(SdpConsistency): do not pass whole TPC

There's no need to pass whole TPC instance if the only thing needed is
a logging prefix.

The setting of currentBrowser in RTCBrowserType was
accidentally removed.

* 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.

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.

Temasys plugin does not allow to bind PeerConnection's
methods.

* 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

Logic broken while refactoring (in b8a2a7baaf [eslint] no-unneeded-ternary).