* 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.
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.
Handles private messages received by speakerstats component.
Updates speakers stats with values received by the spealerstats component, all the logic is activate only in case we discover speakerstats component address from disco-info and the incoming message is coming from that component.
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.
core: refactor initialization not to return a Promise
There is nothing asynchronous about the initialization process (anymore), thus
turn it into a synchronous method.
In addition, WebRTC support is absolute, it cannot change from not being
supported to being supported (as it plreviously could, thanks to Temasys) so get
rid of the ancillary logic to support that.
Last, introduce a way to check if WebRTC is supported in the current
environment: JitsiMeetJS.isWebRtcSupported().
ref(recording): change implementation to match VideoSIPGW (#769)
* ref(recording): change implementation to match VideoSIPGW
VideoSIPGW takes in a chat room and uses instance variables
on the chat room. RecordingManager has been changed to
mirror this approach because of the case where jitsi is
deployed on a domain requiring authentication. In that
case, the initial chat room is created and fails, a
new chat room (connection) is created for authentication,
authentication is put onto the failed chat room, and the
failed chat room is used. As such, recordingManager
should not be tied to chat room creation itself but
rather tied to the first chat room.
* squash: use git mv to detect capitalization change
* Support layer suspension
Add support for a message which notifies the endpoint whether or not it
is selected (meaning its HD stream is in use). If it is not
selected and enableLayerSuspension is set to true then it will impose a
bandwidth limit in the SDP to suspend sending the higher layers.
* only add the IS_SELECTED_CHANGED listener in JingleSessionPC if layer
suspension is enabled
this prevents doing a local o/a when we don't need it
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
Adds new parameter details for the connection failed event. (#697)
* Adds new parameter details for the connection failed event.
Adds suspend and shard_changed details.
* Updates the used strophejs version in dependencies.
* Updates the used strophejs version in dependencies (package-lock file).
* Changes to always report suspend time, reports 0 on no suspend detected.
* Updates the used strophejs version in dependencies (1.2.14-1).
* Changes details field name.
* 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.
* doc(JitsiConference): deprecate 'isInLastN'
The 'isInLastN' method should not be used for the UI purposes, but
ParticipantConnectionStatus value should be used instead.
* fix(ParticipantConn..Status): speed up INACTIVE transition
Before this change when user's video stops playing, after user is
removed from last N we were waiting 2 seconds, before going to INACTIVE
state. This commit reduces the time to 500ms for such case.
* fix(ParticipantConn...): reduce logging
Reduce logging verbosity.
* fix(ParticipantConn...): handle LastN == 0
When LastN is set to 0 we should not rely on video playback and last N
set for figuring out participant connection status.
* fix(JitsiConference): undefined participants
Fixes a crash when this.participants field is accessed from _init.
* 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.
* 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
Hristo Terezov
Saúl Ibarra Corretgé
George Politis
Дамян Минков
paweldomas
bgrozev
virtuacoplenny
bbaldino
Guénaël Muller
nikvaessen
Lyubo Marinov