feat(conference) Implement audio/video mute disable when sender limit is reached.
Jicofo sends a presence when the audio/video sender limit is reaced in the conference. The client can then proceed to disable the audio and video mute buttons when this occurs.
They are companion rooms created in a separate MUC. The room relationship is
maintained by a Prosody plugin.
All signalling happens through the breakout rooms MUC component.
Co-authored-by: Saúl Ibarra Corretgé <saghul@jitsi.org>
feat: add facial-expressions in speaker stats (#1724)
* feat: facial expression in speaker stats
* feat: send xmpp message with facial expression to server
* fix: rebase conflicts
* feat: facial expression in speaker stats
* feat: facial expression in speaker stats
* fchore(facial-expressions): remove the send facial expression call from update facial expression function
* feat(facial-expressions): store expresions as a timeline
* refactor(facial-expressions): store expressions by counting them in a map
* feat(facial-expressions): camera time tracker
* add(facial-expression): increase facial expression with duration parameter from payload
* add(facial-expressions): the disgusted expression
* refactor(facial-expressions): remove camera time tracker
* refactor(facial-expressions): change data channel message handel position for facial expressions and renamed some types
* fix(facial-expressions): move facial expression endpoint message handler from statistics.js to ConnectionQuality.js
* fix(facial-expressions): remove unused type
* feat: Audio/Video moderation.
* squash: Fix docs.
* squash: Adds some warning logs when execution is rejected.
* squash: Changes a field name in the message for adding jid to whitelist.
* squash: Send to participants only message about approval.
Skips sending the whole list.
* squash: Fixes tests.
* squash: Adds more logs.
* feat: Separates enable/disable by media type.
Adds actor to the messages to inform who enabled it.
* squash: Fixes log line.
* squash: Fixes comments.
* squash: Fixes log messages.
* squash: Fixes comments.
feat(stats): Add a new bridge message "EndpointStats" for stats.
Use the new Colibri message "EndpointStats" for broadcasting the local stats. The bridge then will be able to filter the endpoint stats and send them only to the interested parties instead of broadcasting it to all the endpoints in the call.
Refactors the way send/receive video constraints are managed and puts the high level
logic in a separate module. See QualityController.js for high level overview on how
the constraints are managed now.
Adds events to JitsiConference fired whenever it starts new jvb/p2p session.
Adds event to JingleSessionPC.js when remote party signals receive max frame height.
Also adds signaling of the local recv preference for the p2p mode(only existed for JVB).
* Initial impl of lobby rooms.
* Fixes tests to check the new fulljid added to MUC_MEMBER_JOINED.
* Updates few of the comments, renaming some functions.
* Renames disableLobby ChatRoom option to enableLobby.
* Fixes a comment.
* Moves setMembersOnly method to ChatRoom.
* Fixes counting members, to exclude jicofo.
* Moves setLobbyRoomJid earlier and renames a method.
Rename _maybeEnableDisable to maybeJoinLeaveLobbyRoom.
* Drops using custom roomconfig lobbypassword field and reuse room lock.
* Handles destroying the lobby room.
* Handles clear lobby room on destroy for moderators.
We do not try to leave the lobby room as it is server-side destroyed and we handle that. The only case of leaving a lobby room is when request to join room is being approved.
* Join main room if lobby is disabled while waiting.
* Adds MEMBERS_ONLY_CHANGED conference event.
* fix: Make sure we leave lobby if main room is joined.
* fix: Setting password when joining locked room.
* fix: Fixes case where we enable lobby for already locked room.
* fix: Fixes case where we enable lobby and then lock room.
* fix: Fixes lint.
* ref: Removes shared password for lobby.
This functionality is handled by the lock room password and handled there.
Removes duplication and unnecessary complicated API for lobby room.
* fix: Fixes comments.
* Updates kicked event to inform local and remote kicks and who it affects
* Parses actor from MuteIQ and adds participant info to mute track event.
* Fixes emitting both events.
* Fixes comments.
The code for handling device availability has been disabled for a long time,
plus it's ill named since it represents 2 abstractions: lack of permissions and
lack of devices.
Time for it to rest in the git graveyard.
Exports additional statistics through ConnectionQuality (#813)
* feat: Read the server region from Jingle and broadcast it with statistics.
* feat: Adds the bridge count to local "statistics", refactors conference-properties.
* fix: Emits the conference properties with the event, small fixes.
* ref: Orders the imports alphabetically.
As described by @virtuacoplenny:
[T]he ordering is based on import path, not import name, with different
file depths being grouped together, node modules being grouped together
at the top.
* fix: Keeps JitsiConference#properties always defined.
* fix: Does not fire an event when the argument is undefined.
feat(recording): frontend logic can support live streaming and recording (#741)
* feat(recording): frontend logic can support live streaming and recording
Instead of either live streaming or recording, now both can live together.
The changes to facilitate such include the following:
- Remove the old recording.js module which allowed one recording at a time.
- Remove events for live stream url changes as the url is now part of a
sesssion and not fired independently.
- Availability of sipgw and recording are gone. Instead sessions have a
failure reason. For sipgw sessions, store that failure and emit it to
listeners.
- Create a new recordingManager singleton that can start/stop sessions
and handle updating known state of those sessions. Known state is
emitted through one event.
- Create a JibriSession model to encapsulate state of a session.
* update comments, use map to store sessions
* always pass in focusmucjid
* try to fix jibrisession docs and remove default null
feat(recording): show the YouTube live stream URL (#736)
* feat(recording): show the YouTube live stream URL
- Pass you_tube_broadcast_id to Jibri so it can create a
YouTube link for the live stream.
- Emit the liveStreamViewURL (the YouTube link) when it
is received from the participant doing the recording.
* set member.liveStreamViewURL only when truthy
Will emit CONNECTION_ESTABLISHED event to fill up the gap where the lib
users are not able to tell whether or not the connection has been
established if there were no interrupted/restored events emitted.
* 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().
* Changes initialization of videoSIPGW.
* Adds some errors returned on creating videoSIPGW session.
* Fixes sending videoSIPGW session STATE_CHANGED event.
* Fixes sending jibriIQ, no result status is received.
* Adds VIDEO_SIP_GW_SESSION_STATE_CHANGED to JitsiConferenceEvents.
* Fixing comments.
* feat: multiple, simultaneous RTP stats
Makes it possible to have remote RTP stats running for more than one
peerconnection at a time.
* feat(stats): report RTT all the time
Will report JVB RTT (and end to end) while in P2P mode and vice versa.
* fix(JitsiConferenceEvents): remove CONNECTION_STATS
CONNECTION_STATS event is no longer emitted.
* fix(AvgRTPStatsReported): users with no video
Do not include FPS == 0 in average remote FPS calculation. Report NaN
for local FPS when video muted or no video device. NaN will be reported
for avg remote FPS if no video is received.
* fix(AvgRTPStatsReported): reset total packet loss
* feat(AvgRTPStatsReported): report 'screen' FPS
Will report average FPS for screen videos separately from camera videos,
but only if available (camera video reports NaN FPS when not available).
* fix(AvgRTPStatsReported): end2endRTT
Needs to report JSON with value.
* feat(AVG RTP stats): separate audio and video bitrate
Will report average audio and video bitrates separately.
* doc(JitsiConference): try to improve comment
* fix(AvgRTPStatsReporter): remove confusing reset
There's no a clear reason for doing reset there.
* ref(AvgRTPStatsReporter): rename var
* 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.
* 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.
* 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
Currently if RTCPeerConnection is closed (an event is delivered) and we haven't called close method ourselves, this means browser has closed it due to suspending the PC.
As I see no hidden meaning in the current non-preservation of
alphabetical order, I prefer to facilitate scanning (in the fashion of
searching in a dictionary).
Jaya Allamsetty
Werner Fleischer
Gabriel Borlea
Дамян Минков
robertpin
Jade Guiton
Andrei Gavrilescu
Paweł Domas
theunafraid
Andrei Gavrilescu
Saúl Ibarra Corretgé
bgrozev
hristoterezov
virtuacoplenny
Guénaël Muller
Lyubo Marinov
Lyubomir Marinov
Sam Whited