* Added video mute participant
* Trigger mute event
* Optimized mute type checks
* Fixed event name
* Fixed some linter issues
* Fixed more linter issues
* And even more linter issues fixed
* And more linter fixes
* Added media type to analytics event
fix(ice-restart): Force client reloads when call is migrated.
Force the client to reload when the bridge that is handling the media goes down.
This mitigates issues seen on the bridge because of a client re-joining the call with the same endpointId, BWE issues, etc.
This behavior is configurable through 'enableForcedReload' setting in config.js.
feat: Skips using disco-info for features. (#1450)
* fix: Drops unused parameter of join and sendPresence.
* feat: Skips using disco-info for features.
Uses presence to publish features added externally.
Recognizes jigasi participants using a specific presence extension used by jigasi.
* squash: Fixes tests.
* squash: Adds e2ee to the features in presence.
* squash: Fix function name and docs.
* squash: Drops detecting jigasi from initiator.
Using the newly added presence features.
* squash: Drops unused var.
* squash: Fix comments.
* squash: Adds a constant and for E2EEE feature.
fix: Maybe start p2p only when we have received all presences.
There are cases where we start p2p on the first participant received presence, where there are more presences coming. This starts the process of p2p resulting and few iq errors while terminating it quickly after starting it.
On a normal joining as third participant we see around 38 send or received iq messages, in the problem scenario we see 60.
* 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.
Fix: Old JitsiConference instance remains listening on incoming calls… (#858)
* Fix: Old JitsiConference instance remains listening on incoming calls after leave - rejoin to same room
* Remove all listeners added by setupXMPPListeners
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.
When the main domain advertises component with type 'speakerstats' every participant will send message to the component with the name of the room where this happen.
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.
This simple change helps to prevent `'The conference is has been already left'` error after kick.
`Conference.leave` will detach chatRoom instance itself.
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
* 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().
Reports ssrc to callstats when screen sharing is started. (#657)
Adds rtc.LOCAL_TRACK_SSRC_UPDATED event to be emitted when ssrc is updated for a local track.
Fixes adding rtc listeners for CREATE_ANSWER_FAILED, CREATE_OFFER_FAILED, SET_LOCAL_DESCRIPTION_FAILED, SET_REMOTE_DESCRIPTION_FAILED.
* 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.
* 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.
* 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
* ref: store SSRCs as a number
Converts all the places where SSRCs where stored as string to use
numbers.
* doc(RTPStatsCollector): getNonNegativeStat
Fixes invalid description about returning NaN.
* ref(JitsiConf...EventManager): simplify for..of
* ref(JitsiRemoteTrack): throw TypeError
Will throw a TypeError when 'ssrc' is not a number.
* fix(RTPStatsCollector): invalid reference
* doc(RTPStatsCollector): getNonNegativeStat private
* ref(RTPStatsCollector): simplify for..of
* fix(SSRCs): check for negative value
Will not accept negative SSRCs, since those are supposed to be unsigned.
* ref(CallStats): cleanup constructor
Changes CallStats constructor to not take the whole JingleSessionPC as
it only needs an alias and the TraceablePeerConnection instance.
Describes the arguments in JSDoc.
* ref(CallStats): rename var
Everything is callstats c'mon...
* ref(CallStats): remove _checkInitialize
The _checkInitialize was trying to workaround CallStats lib issue
without really checking for any specific type of error on whether or not
it makes sense to retry.
Also it depended on some internal fields of 'callStatsBackend' and was
binding 'initCallback' to the backend instead of CallStats instance
which made no sense (it means it's very likely this functionality was
broken anyway).
It would be hard to fix it in a clean way, because CallStats instance
fields would have to be stored in static variables in order to make
the initCallback work (called from '_checkInitialize').
We also need to have more than one CallStats instance running at the
same time, because of the P2P which makes things even more complex.
* fix(CallStats): do not catch 'sendApplicationLog'
Wrapping 'sendApplicationLog' in 'tryCatch' will result in an endless
loop, because it will be logged on the logger.error which then tries
to send the logs immediately again.
* ref(CallStats): do not use call on static
There's nothing more confusing that seeing 'this' in a static method.
Wow maybe these methods are not really static !?
* ref(stats): fix var name
* feat(stats): report P2P to CallStats
Will create new CallStats fabric for the P2P peerconnection in order to
log peer to peer connections.
Refactoring was required in the statistics and CallStats module to be
able to have more than one CallStats instance. Because each CallStats
fabric reports one peer connection now each CallStats will correspond to
one TraceablePeerConnection. CallStats instances are now stored in a Map
mapped by the TraceablePeerConnection.id field.
In order to be able to execute some global/static CallStats reporting
methods all Statistics instances also need to be stored in a static
field.
CallStats API backend(new callstats()) will be initialized only once for
the values provided in the first call to initializeBackend. It is not
possible to have more that one CallStats backend running at the same
time (at the time of this writing). If we would have a routine for
disposing global "Statistics" module we could try to cleanup static
reference and allow to initialize it with new values (but no such use
case yet).
* ref(CallStats): move initCallback
Since there is no alphabetic order preserved in this file anyway at
least place it closer to it's usage.
* ref(CallStats): remove tryCatch
Temporarily removes tryCatch to make the ES6 conversion easier.
* ref(CallStats): convert to ES6
* style(CallStats): fix indentation
* fix(statistics): use import for CallStats
* ref(CallStats): convert static methods
Some of the methods should not be static, because it only make sense
to call it when there is CallStats instance available.
* ref(CallStats): rename var
* doc(CallStats): remove misplaced comment
* chore(CallStats): remove invalid eslint comment
* fix(CallStats): undefined CallStats namespace
If no CallStats ID namespace option is provided the conference will be
reported without it.
* style(stats|CallStats): remove extra lines
* fix(CallStats): do not log error from tryCatch
GlobalOnErrorHandler calls logger.error anyway.
* fix(CallStats): cleanup tryCatch
If I understand correctly our initial intention with doing tryCatch was
to avoid crashing the whole app in case the CallStats backend would
crash. With this commit the tryCatch is done by wrapping original
backend instance methods or using explicit try catch block where
the method is called only from one place or a value needs to be returned
in case of a crash.
* ref(CallStats): make backend a static var
* fix(CallStats): invalid eslint comments
* ref(CallStats): use for..of
* ref(CallStats): fixes around REST args
* ref(CallStats): rename var
* ref(CallStats): reorder static methods
Also renamed some callbacks
* doc(CallStats): adds some docs
* ref(CallStats): make methods not static
Both 'sendDominantSpeakerEvent' and 'sendScreenSharingEvent' methods are
not really static as they require instance to be called.
* fix(CallStats): invalid key
* fix(CallStats): reduce amount of debug logs
* feat(p2p|CallStats): log hold/resume
Will put CallStats fabric for the JVB connection "on hold" while in p2p
mode.
* doc(CallStats): add FIXME
* doc(JitsiConference,CallStats): typos and renaming
* 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
Jaya Allamsetty
Steffen Kolmer
Дамян Минков
Andrei Gavrilescu
Jaya Allamsetty
Saúl Ibarra Corretgé
paweldomas
theunafraid
Bettenbuk Zoltan
Hristo Terezov
Gábor Takaró
Andrey Davydov
virtuacoplenny
Guénaël Muller
bgrozev
nikvaessen
Iñaki Baz Castillo
Lyubo Marinov