Otherwise Chrome has been seen using a simulcast layer with a resolution of
240 x 135, which can break implementations not supporting odd resolutions, such
as WebRTC on Android.

Related Chrome issues:
* https://bugs.chromium.org/p/webrtc/issues/detail?id=6651
* https://bugs.chromium.org/p/webrtc/issues/detail?id=7206

Moves part of the XMPP events which are clearly
related to the PeerConnection to RTCEvents.

also fixes lint errors

Some runtimes such as React Native don't have a window.location.protocol
property, so don't check for it, assume removeStream is available.

Make sure there are options to set the extension ID and an enabled flag for
both Chrome and Firefox.

Removed checking for deprecated configuration options: desktopSharing,
chromeExtensionId and minChromeExtVersion.

Add 'replaceTrack' method to API, make all replace track operations local (use same ssrcs so no need for source-add/source-remove)

* M1: device change now uses new flow.  fundamentally "works" but may be corner cases/side effects to other flows. haven't touched ffox yet

* M1a: fixed a bug where the ssrc info of the new stream wasn't being set correctly, causing problems with video mute/unmute

* M1b: fix firefox.  changing camera, video mute/unmute looks good

* M1c: 'replaceStream' processing is not integrated with the same queue as all of the other o/a-related events.

* fix typo, some deprecated code cleanup

* move the rfc4145 fix to traceablepeerconnection::createanswer so both the new and old o/a flows leverage it.  also cleanup some now-unused code and change more functions to arrow functions so we can get rid of 'var self = this' workarounds.

* get rid of separately-mainted local description in JingleSessionPC.  keep TraceablePeerConnection as the source of truth for the local sdp and have JingleSessionPC retrieve it from there.

* change the code in the old modifySources shim to use the _renegotiate method so both old and new flows use that method.

* split out the remote change processing from the local change processing to better set us up for individual remote and local modification queues

* move the setOfferCycle flow off of the old shim and into the new flow.  also removes the new for this this.jingleOfferIq member

* move addRemoteStream (nee addSource) to the new flow.  gets rid of the need for the this.addssrc member.

* move remove remote stream over to the new flow.  gets rid of the need for the this.removessrc member variable

* temporary '_processRemoteChange' method is no longer needed, as all remote changes are processed directly (rather than setting member variables and calling a catch-all method which checks each member to see what needs to be done)

* M2: add 'sdp-consistency' in an attempt to separate out the restoration of the same ssrcs (also requires the changes i've made to sdp-simulcast, currently in my brian/rework branch @90aceaddb1eec4d15026efac82c140579c7af2b3, as well as a commit in jitsi-meet to change the toggle-screenshare flows over to use the new 'replace' chain).  as of this point video mute, change device and toggle screenshare all happen as local operatons.  there is a bug when starting as video muted i need to look into, and i haven't checked firefox yet.

* fix the bug with start-as-video-muted (relies on commit f1d8d3c831 in brian/overual sdp-simulcast branch)

* some code cleanup, marked some TODO list items.  got addStream off of the old shim and onto the new flow

* change removeStream over to the new flow and off of the shim.  shim is now removed!

* fix a bug when muting a stream.  remove some logic from addstream/removestream that i don't think makes sense, and move the source-remove parsing logic into a helper

* move the source-add/source-remove processing into the atomic queued operation since it looks at sdp.

* remove unused code used for old ssrc replacement logic

* get rid of the need for TraceablePeerConnection::addStreamNoSideEffects

* get rid of need for TraceablePeerConnection::removeStreamNoSideEffects

* distinguish XXXStream and XXXStreamToPeerConnection (was XXXStreamNoSideEffects)

* remove shim methods for add/removeRemoteStream

* fix a bug when adding local media for the first time

* replace the old 'dispose' method with the new one

* address feedback comments.  update version of sdp-simulcast.

* update sdp-simulcast version

* add missing semicolons

* move sdpconsistency to es6 style

* restore JitsiConference::removeTrack, restore JitsiLocalTrack::dispose back to how it was (doing o/a if necessary)--we get around this on replace by not disposing the track until after we do the replace)

* change removeTrack and addTrack to leverage replaceStream

* fix removeTrack to return the promise, handle an already-disposed track

* add more documentation

* remove unnecessary semicolon

* update sdp-simulcast version

* address PR feedback

* remove usage of @type in method js docs

* address (more) pr feedback.  upgrades to a new sdp-simulcast version which fixes an issue in smoke tests

* more PR feedback, and clean up the success/error handling done for the work functions.

* tweak the error propagation

RTCBrowserType.supportsSimulcast(). Add TODOs.

PC constrain via configuration.

Lib-jitsi-meet uses Temasys on Internet Explorer and Safari so it does
not make sense to require its adapter.screenshare.js on React Native,
for example, where it fails anyway.

When the Temasys plugin is used the 'muted' property is in the track,
but it's value is undefined.

I added reporting of Promise's reject reason in b10a1fc while merging
jitsi/ss_postinstall_fix since Promise's reject function has such an
argument. Hristo Terezov disagreed with it because reason is undefined
in the use case and found it to be confusing so I'm reverting to what he
intended in the first place.

We can not assume that the audio source is not sending any data based
on the stat until the media connection has not been established.