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.

* 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* doc(JitsiConference): fix outdated comment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

There's no reason to involve the ChatRoom directly into the generation
of remote stream added event. It merely provides some info carried in
the presence about muted state etc.

Video SSRC has to be generated for the recvonly stream if there are no
video tracks in the PeerConnection.

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