We add the capScreenshareBitrate value to the permanent properties so
that it's included with every event that we send to the analytics
backend.

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.

The client will reject P2P session-initiate if the conditions which
allow for P2P session to start are not met(there more or less than 2
human participants in the call).

If DTLS fingerprint remains unchanged after ice-restart there will be no
media flow even thought the ICE connection will be re-established.
That's because JVB has all layers re-initialized from scratch, but
the WebRTC stack will keep the current state of the DTLS layer.

Reset the DTLS layer by setting an all zeros fingerprint on the hacky
offer/answer cycle which is currently done in order to reset the data
channels and the sequence number counters on the RTP receivers.

Also do not execute any peerconnection callbacks like
'oniceconnectionstatechange', 'onicecandidate' etc. after close.

Make the client send ICE failed notification to let Jicofo know
that the ICE connection with the bridge has failed. It can then
re-create the channels and send new transport in the
'transport-replace' message to give it another try.

Also adds a bridge session ID so that Jicofo can discard outdated
requests. When a bridge dies multiple clients may send ICE failed
notifications at the same time. Jicofo will detect bridge failure
when processing the first one and it must not act on the following
requests for the broken bridge session which is no longer current.

A bridge session ID will be sent in new 'bridge-session' XML element
which is part of either 'session-initiate' or 'transport-replace'
message. The server's region is moved to this element as well as
it's part of a bridge session.

Reset the state of remote MediaStreams in order to fix sequence number
synchronization problem. When a conference is moved to another bridge or
if channels are re-allocated the bridge looses the most recent sequence
number seen by the client which makes Chrome drop all video packets
until the JVB catches up with the sequence numbers (if at all).

Changes the behavior to actively open new WebRTC Data Channel instead
of waiting for the JVB to do it.

Adds ICE_RESTART_SUCCESS event used to re-initialize the data channel in
case of ICE restart where the conference could have been moved to
another bridge.

Keeps it consistent with the previous implementation.

It's 2018 after all...

The previous check looks like it was meant to check if it's
the initiator.

The options are coming from the config anyway.

the docs denote this behavior, but it appears to have been removed at
some point

* fix(migration): reject the data channel to restart it

Starting on chrome 63, an error occurs when renegotiating
with the new bridge on bridge migration when the SDP sections
do not match, which occurs when removing the data channel
section. Not doing the double renegotiate prevents the data
channel from being reopened. The way around this, while still
avoing the error, is to reject the data channel section instead
of removing it to trigger a new data channel.

* squash: reword comments

* Support layer suspension

Add support for a message which notifies the endpoint whether or not it
is selected (meaning its HD stream is in use).  If it is not
selected and enableLayerSuspension is set to true then it will impose a
bandwidth limit in the SDP to suspend sending the higher layers.

* only add the IS_SELECTED_CHANGED listener in JingleSessionPC if layer
suspension is enabled

this prevents doing a local o/a when we don't need it

When switching bridges, removing the "data" section from the SDP
and doing a double offer-answer results in an error starting on
Chrome 63. So on 63 and up, do a single off-answer without
removing the "data" section.

This PR exposes the functionality that's already built-in the
TraceablePeerConnection that has it gather stats from the wrapped
PeerConnection.

* 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().

* 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.

* 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.

Do not log whole objects on Jingle error, as these may cause trouble on
iOS react-native.

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

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.

The 'setMutedInProgress' flag is unreliable and it can lead to weird
situations when mute operation is in progress, but the removal from
JingleSessionPC is waiting on the queue. In such case LocalSdpMunger
will fake SDP even if the track is in the PeerConnection which may
confuse some parts of the SDP transformation chain.

For example RtxModifier will not attempt to inject Rtx group description
if it sees that the SSRC are already there. But it can happen that even
though the SSRCs are there they do not contain valid track ID, because
the track has changed.

The LocalSdpMunger is supposed to fake the SDP only when the video
MediaStream is not in the PeerConnection. So this commit implements just
that: a way to tell if JitsiLocalTrack's stream is currently in
the PeerConnection (added through TraceablePeerConnection).

Moves the call to 'setLocalDescription' from 'sendSessionInitiate' to
'invite' as it was easy to overlook.

In case when the answer is being set for the first time, full sRD/sLD
cycle is required to have the local description updated and SSRCs
synchronized correctly. Otherwise SSRCs for streams added after invite,
but before the answer was accepted will not be detected. The reason for
that is that renegotiate can not be called when adding tracks and they
will not be reflected in the local SDP.

The 'ondatachannel' even listener must be set as soon as
the TraceablePeerConnection is created and before the offer is accepted.
We've observed situations where the ondatachannel event was missed,
because the listener is bound too late from the JingleSessionPC
'acceptOffer' success callback.

* add support for signaling simulcast for unified plan

1) munge the incoming offer to advertise support for receiving simulcast in unified plan
2) add 3 simulcast encodings via rtp sender
3) inject an ssrc-group line, even when using rids for simulcast (jicofo will rely on this)

* add rid information to the session-accept jingle

* address lenny's feedback

* make firefox simulcast configurable

* fix missing comma

Due to it being ran asyncronously, it's possible a replaceTrack operation is
scheduled while there is a valid PC, but by the time it runs was closed.

Accommodate to this situation by pretending everything is just peachy.

Also add a preferH264 option for P2P.

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

* ref(JSPC): simplify SSRC owner in P2P

There's no need for any extra extensions for SSRCs owner signalling in
P2P, because it's always the remote peer who owns them.

This also fixes a problem where no SSRC owner was added for 'source-add'
in P2P (for JVB conference Jicofo adds that).

* fix(TPC): always advertise 'sendrecv'

Our media direction is only ever updated on the remote side with
the initial offer (or answer). Because of that we want to advertise
'sendrecv' even if we start with no video (or audio) track.

It is OK to adjust this direction in the localDescription getter,
because it's adjusted again in the setter to the correct value based on
local tracks, so the SDP transformation chain still works fine.

This code would crash on react-native if 'sdpMid' is not defined at all.
Still works as expected with an empty string though.

Adds session converted to string to the error object.

Do not log errors for 'item-not-found', after jingle session was
terminated.

When the lastN is set to 0 the P2P session will notify the remote peer
to stop sending video.

Creating SDP instance was unnecessary.