* fix(SDP) Do not filter out SSRC 'cname' attributes.
Chrome doesn't generate the a=ssrc line with 'msid' attribute when the track is removed from pc and a renegotiation is triggered. As a result a source-remove is sent to the p2p peer when it shouldn't be causing issues on the receiver when the same source is added back to the conference. Fixes toggleCamera on mobile browser not working as expected.

* fix(SDP) Include the trackId in the signaled msid for the source.
Without the trackID in a=ssrc lines in SDP, older versions of Chrome (108) reject the remote description resulting in endpoint getting kicked out of the conference.

* squash: Fix unit test

The constructor for RTCSessionDescription has been deprecated - https://developer.mozilla.org/en-US/docs/Web/API/RTCSessionDescription/RTCSessionDescription.

* ref(SDPDiffer) Convert to ES6 class.
Make it work directly with unified plan SDP that has multiple m-lines and add more unit tests.

* ref(xmpp) translate unified-plan SDP->Jingle directly.
Without having to run it through the SDPInterop.toPlanB cycle.

* ref(SDP) Always generate the MSID for signaling it to Jicofo.

* fix(SDPDiffer) Check explicitly for ssrc changes

* fix(SDP): Fix comments and cleanup.
Remove LOCAL_TRACK_SSRC_UPDATED event as the application ignores the event and no additional action needs to be taken when that event is fired.

* ref(SDP) Add a helper function for parsing the 'a=ssrc-group' line.

* squash: Address review comments

Chrome 122 does not generate a 'msid' attribute in the local SDP when a local source is added after the user joins muted. The SSRC doesn't get signaled to remote p2p peer when this happens. Also, when the direction of the transceiver changes from 'sendonly' to 'sendrecv', Chrome is generating a new set of SSRCs resulting the client sending out a source-remove followed by source-add. Fixes various p2p related bugs that have surfaced in p2p calls where both the endpoints are running Chromium 122.

* fix(RTC): Remove support for plan-b SDP.
Since unified plan mode becomes the default mode, so does multi-stream mode.

* squash: Remove plan-b unit tests and fix comments

* squash: address review comments

Add SVC and Simulcast support for AV1 and VP9.

The default mode is the full SVC mode for VP9. It works as expected with the current version of JVB.
AV1 in the client is only supported when the bridge negotiates the Dependency Descriptor extension headers which are needed for reading the layer information for AV1 and H.264 video frames.

* feat: Enable H.264 simulcast support.
Multi-encoding simulcast for H.264 is supported now because of the DD header ext support added on the bridge side.

* fix(video-quality): Add default bitrates for all codecs.
Expect the videoQuality settings in the new format, deprecated configs are still supported.
videoQuality: {
    AV1: {
        maxBitratesVideo: {
            low: 100000,
            standard: 300000,
            high: 1000000,
            ssHigh: 1500000
        },
        useScalabilityModeAPI: {
            enabled: true,
            useSimulcast: true, //(defaults to SVC),
            useKSVC: true //(defaults to L3T3_KEY)
        },
    },
    H264: {
        maxBitratesVideo: {
            low: 200000,
            standard: 500000,
            high: 1500000,
            ssHigh: 2500000
        }
    },
    VP8: {
        maxBitratesVideo: {
            low: 200000,
            standard: 500000,
            high: 1500000,
            ssHigh: 2500000
        }
    },
    VP9: {
        maxBitratesVideo: {
            low: 100000,
            standard: 300000,
            high: 1200000,
            ssHigh: 2500000
        },
        useScalabilityModeAPI: {
            enabled: true,
            useSimulcast: true, //(defaults to SVC),
            useKSVC: true //(defaults to L3T3_KEY)
        },
    }
}

* Disable encodings when requested height for screenshare is 0.

* feat: add unit tests for scalability modes.

* ref: Move all encoding configuration calculation to TPCUtils.

* Add more unit tests for VP9 K-SVC mode.

* Make codec name in settings case insensitive.
Also read the deprecated max bitrates correctly, add unit tests to test it.

* ref(TPC): isSimulcastOn -> isSpatialScalabilityOn.
It makes more sense to call it spatial scalability than simulcast now that full SVC support is available.

* fix: Negotiate AV1 DD header exts only for AV1 and H.264.

* fix: Rename setting useL3T3Mode->useKSVC and adjust the defaults.

* feat(signaling) Only include 'msid' in the source description.
'cname', 'label' and 'mslabel' attributes that are included in the source description of the signaling messages are ignored by Jicofo. Only 'msid' is used for source validation.
Therefore, drop those attributes and make sure 'msid' is always included and in the format '<endpoint_id>-<mediaType>-<trackIndex>-<tpcId>'.

* fix(p2p) Enable p2p for Firefox.
Set the senders attribute correctly in the outgoing jingle message. Ignore the media direction from the session, only look for the direction in the media desccription. This fixes an issue where Firefox rejects an answer because the direction is not set correctly.

* Remove support for legacy endpoint based signaling and make source-name signaling as the only signaling mode. The legacy screensharing mode will no longer be supported. The client will still be able to process remote presence sent in the old format to support interop with very old mobile clients and jigasi that do not support source-name signaling.
* remove code related to presenter mode.

Adds a helper function getSourceIndexFromSourceName to
have track index parsing in one place.

In _processLocalSSRCsMap there was an assumption that a track
index can be a single digit only.

Use "last index of" or search from the end of the collection to
make parsing a bit more universal (does not matter as much for
Jitsi endpoint IDs, but if fixes issues if someone has
a customization there).

* feat: Add videoType to sources signaled in jingle.

Reusing(removing it and adding it again) an SSRC in the
remote SDP is leading to the issue where remote video is received but
not rendered. That's why we shouldn't remove the receive only SSRCs
from the SDP. This way the sender won't send any source-remove and
source-add for the same SSRCs.

When source-name signaling is enabbled on plan-b clients, the sources to MSID map needs to be cleared after every transformation since browser in plan-b mode produces a new set of SSRCs and trackID when a track is removed and new track is added to the peerconnection. This fixes an issue where the track index in the source-name gets incremented after a track replace operation.

Do this even if the browser provides a non '-' string for streamId since we rely on this to generate the source name.

* fix(multi-stream-support) Support muting of desktop track.

* fix(multi-stream) Always initiate responder renegotiation even for p2p.
This is needed since the new m-line is always added to the remote description.

* fix(multi-stream) Support multi-stream only on Unfied plan endpoints.

* fix(multi-stream) Advertise source-name signaling to Jicofo when supported.

* fix(JitsiLocalTrack): Remove unnecessary reject.

* fix: add the correct source name attribute for the second video track.

* squash: Address review comments.

* feat(multi-stream-support) Add the support for multiple local video streams.
This feature is behind 'sendMultipleVideoStreams' config.js flag and is currently supported only on clients running in Unified plan mode.

* squash: Use the track index from source name to find the matching transceiver on unmute.

...to advertise track's muted state and the video type.

For now, if the source name signaling flag is enabled, both legacy and
new <SourceInfo> element will be used at the same time. This is to be
able to interoperate with legacy clients and start testing the new
format at the same time. Whenever possible <SourceInfo> will be used
as main source of truth with the fallback to legacy <audiomuted/>,
<videomuted/> and <videotype/> elements.

This is the first step in adding support for multiple
streams per endpoint. Each source(stream?) needs to have
an identifier. For now always generate the 0 index name
until the machinery for sending more than 1 stream is put
in place.

* send only MSID attribute

* add feature flag for source name signaling

* log a msg if source name signaling is enabled

... is stopped. If there's any sRD/sLD cycle happening
while the screen sharing stops, the local track
returns muted and it was injecting fake video SSRCs
which results in invalid SSRC description.

At the time when it happens there's error log in jicofo:
"Error adding SSRCs from X"
and the client will get bad-request error in response
to the source-add request.

In unified plan, re-use of m-line (i.e., adding an SSRC, removing it and then adding it back) causes the browser to not render the media on Chrome and Safari. The WebRTC spec is not clear as to how browsers have to behave, this doesn't cause any issues on Firefox. As a workaround, only change the media direction and leave the ssrc in the remote desc. This automatically triggers a 'removetrack' event on the associated MediaStream and the track can be removed from the UI.

Avoid configuring the encodings on chromium immediately after replace track since the encoding params are read-only until the renegotation is done.

This is needed only for JVB connections. Add unit tests for LocalSdpMunger.

If the msid attribute is missing, then remove the ssrc from the transformed description so that a source-remove is signaled to Jicofo. This happens when the direction of the transceiver (or m-line) is set to 'inactive' or 'recvonly' on Firefox. Not signaling these source updates creates issues with remote track handling on the other endpoints in the call.

Get the correct media type when generating the source identifier.

When a msid attribute is missing in the 'a=ssrc' line, use the local endpoint id as an indentifier. Move this generation logic to LocalSdpMunger. Also suppress notifying Jicfo of a ssrc change on Firefox when the change is a result of the media being suspended on the jvb connection.

Get rid of noisy logs related to SDP transformations which are redundant. Fix formatting and add missing information.

SDP utility classes are spread across RTC and XMPP directories now, moving these class files to a 'sdp' directory.

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.

Error handling was added to ensure some munging helpers
were receiving RTCSessionDescriptions. Temasys is doing
something else and is passing its own object into
the mungers. So, let temasys do that.

* ref(sdp): lower logs about missing local video to warning

* ref(sdp): earlier error handling for missing description and mlines

* ref(sdp): rename maybeMungeLocalSdp to be more specific

Firefox has deprecated modifying a description's sdp through
re-assignment. Safari (without temasys) does not allow it.
As such, wherever the sdp has to be modified instead create
a new description with the modified sdp.

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).

Random coding style-related issues that bothered me while reading the
source code touched by "Fix local video disappearing from local
thumbnail" (https://github.com/jitsi/lib-jitsi-meet/pull/571). These
include naming, formatting, comments, typos, duplication.

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.

* 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.

* 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