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CHANGELOG.md

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Changelog

All Notable changes to apiRTC are documented in this file.

Latest version : apiRTC Latest

apizee_version : 5.0

5.0.31 - 2025-11-24

Added

  • Added for react native :
    • Add Simulcast and callStatsMonitoring support for react-native
  • Check maxBitrate property in sanitizeSimulcastEncoding function
  • Bypass device detection if stream was created from MediaStream
  • Fix callSelectedCandidate not sended on Mobile Samsung Browser
  • Improve audiocomposite tests
  • Add test for recording without streams

5.0.30 - 2025-10-08

Fixed

  • Fix for react native :
    • add checkings on restartCall process when "swipe" is used to kill application on Android

5.0.29 - 2025-10-01

Fixed

  • Fix for react native :
    • manage maintained context for UserAgent, Session, and Conversation when "swipe" is used to kill application on Android

5.0.28 - 2025-08-08

Fixed

  • Fix webRTCClient checking in disputableEntryTimeout()
  • Fix webRTCClient checking in publish()
  • Fix ua-parser has() fonction

5.0.27 - 2025-08-05

Fixed

  • Fix Chrome M138 : switch camera need "exact" constraints

5.0.26 - 2025-07-29

Changed

  • Modification for simulcast Modifications :
    • Modify configureSimulcastEncodings function
    • Disable VP9 simulcast on Safari Browser
  • Remove EventTarget heritage for E2EEManager

Fixed

  • Fix error on call restarting with capableCodec set
  • Fix nullable operator replaced
  • Fix Chrome M138 (removes the ability to remove an entire m= section via SDP munging)

5.0.25 - 2025-05-09

Changed

  • Calls restart improvment in sfu mode after WS reconnection
  • UAParser.js upgrade to v2.0.3 - Pro Edition
  • Usage of ClientHints to get browser versions
  • Add handover check for mobile Calls Restart

5.0.24 - 2025-03-06

Fixed

  • Fix meshOnlyEnabled option no more activated

5.0.23 - 2025-02-24

Changed

  • Improvement of pushData error
  • Manage applicationUUID for ionic apps
  • Mark Conversation#getFriendlyName() and Conference#getFriendlyName() as deprecated

Fixed

  • Fix on userData parsing if user is not found

5.0.22 - 2024-10-10

Added

  • End to End encryption using ApzFrame (implementation of SFrame) - (Feature is in Beta for now)
  • Manage streamListChanged events storage when listener is not yet set

Changed

  • Set Entreprise#uploadFile() as deprecated. Check pushMedia() in Conversation or Contact

Fixed

  • Fix for datachannel with arrayBuffer message (Uncaught TypeError)
  • Fix QoS info on resolution and modification on the getStats() logs so that it is only displayed once

5.0.21 - 2024-07-17

Added

  • Several fix on applyAudioProcessor() when applied on parent stream to keep applied effects
  • Added posibility to Contact#takeSnapshot() remotely. Snapshot is transfered to requester using an apiRTC internal datachannel call.
  • Added ApplicationUUID for react Native

Changed

  • Modify createIncomingCallEvent() interface to give a JSON object
  • Add ERROR_ON_SAVE reason on Conversation#sendMessage() error

Fixed

  • Fix sonarQube on Misc#parseRecordingEvent()
  • Fix sonarQube on ApiCCDataClient
  • Fix Datachannel for blob object
  • Fix to avoid UAParser for react-native
  • Fix log on calls
  • Fix error on processDistantCallEstablished() if call is not found

5.0.20 - 2024-02-16

Added

  • Update libJsCloud to 0.1.41
  • Add tryAudioOnlyAfterUserMediaError on UserAgent#register()
  • Add tryAudioOnlyAfterUserMediaError on createStream()
  • Update typescript description file

5.0.19 - 2024-01-17

Added

  • Updated : upgrade to UAParser 2.0.0 (PRO version)
  • Add usage of clientHints to get Android version and Model

5.0.18 - 2023-12-06

Added

  • add bearer token on webRTCClient.takeSnapshot()
  • add remoteMediaStreamId management for remote media control on record - APIRTC-1280

5.0.17 - 2023-09-19

Added

  • emit constraintsChanged event if torch is ON when user switch browser in background (Torch is switched OFF by Android)

Fixed

  • Fixed remote torch if video only call

5.0.16 - 2023-09-07

Fixed

  • Fixed early streamTrackChanged muted event
  • Fixed wrong Silent state on updateMediaProperties function

Added

  • Added support of react-native for Conversation
  • Added support for selectedCandidate with react-native
  • Added support of screenSharing with react-native

5.0.15 - 2023-08-16

Added

  • Added Call#remoteStreamIdUpdated event to be informed of streamId changes in case of replacePublishedStream() on a remote Contact
  • Added Conversation#remoteStreamIdUpdated event to be informed of streamId changes in case of replacePublishedStream() on a remote Contact

Changed

  • Internal WebRTCClient#addMedia now return an object with userMediaStreamIds

5.0.14 - 2023-07-12

Added

  • Conversation#getMedia(). This function manage authorization token to access media file
  • Update libJsCloud to 0.1.40
  • Added Conversation#getContactsArray()
  • Added Stream#constraintsChanged event
  • Added Contact CONTACT_CHANGED_CONSTRAINTS_IDENTIFIER case management
  • Added UserAgent#getStreamWithMediaStreamId()
  • Added Contact#getLocalStreamWithMediaStreamId()

Changed

  • Updated : upgrade to UAParser 1.0.35
  • Move PendingPromise methods in ApiCC_Methods
  • Contact CONTACT_GET_SETTINGS_IDENTIFIER case to add possibility to find stream with mediaStreamId
  • Contact CONTACT_GET_CAPABILITIES_IDENTIFIER case to add possibility to find stream with mediaStreamId
  • Contact CONTACT_APPLY_CONSTRAINTS_IDENTIFIER case to add possibility to find stream with mediaStreamId

Fixed

  • Fixed callId value on stream on Call
  • Fixed typo on newAvailableStream that may lead to subscribe issue
  • Fixed release all calls on a subscribe timeout in case of network disconnection

5.0.13 - 2023-06-07

Added

  • No connection retries on unauthorized applicationUUID
  • Call statistics monitoring for Safari (macOS and iOS) and Edge
  • use only promise version of getStats() for Chrome > 112
  • Deprecation of QoSMonitor (was only for Chrome <= 112) : Calls stats monitoring -> enableCallStatsMonitoring() as to be used in replacement.

Fixed

  • Restore apiCC.getAudioContext(), apiCC.listenToAudioSources(), apiCC.stopAudioSourceListening() for retrocompatibility - API v3

5.0.12 - 2023-05-17

Fixed

  • Keep mute status synchronised on linked streams (Streams with audio or video filters)

5.0.11 - 2023-05-09

Added

  • Build apiRTC-with-sio : socket.io is embedded. This is usefull for Salesforce integration.

Fixed

  • Remove XSS filtering : this has to be managed at application level (remove apiRTC.sanitizeString())

Changed

  • Update CDN link for apiRTC_NR_Worklet

5.0.10 - 2023-04-24

Fixed

  • Fix audio only issue in mesh mode with noiseReduction activated
  • Fix double clic issue on remoteAccess
  • Fix on XSS filtering

Changed

  • RemoteAccess : wait for screensharing before starting connexion to application

5.0.9 - 2023-04-06

Fixed

  • Fix declarations in apiRTC.d.ts
  • Revert XSS checking method to avoid any integration issues
  • add apiRTC.sanitizeString() usage on Contact#sendMessage() and Conversation#sendMessage()

Added

  • Added apiRTC.sanitizeString()

5.0.8 - 2023-04-05

Fixed

  • Fix issue on XSS checking

5.0.7 - 2023-04-04

Fixed

  • Fix declaration / docs errors on Stream
  • Fix no call restart when participant is ejected from conversation
  • Fix no UserAgent events removal on unregister()
  • Fix no keyboard event handling on application elements for remote access

Changed

  • upgrade remoteAccess application v1.0.1

Added

  • Add XSS checking on Contact#sendMessage()
  • Add XSS checking on Conversation#sendMessage()
  • Add handshake with remoteAccess application
  • Manage close from remoteAccess application

5.0.6 - 2023-03-14

Added

  • Add Conversation#replacePublishStream()
  • Add reason on Stream trackStopped event

5.0.5 - 2023-03-07

Added

  • Possibility to remotely control a user computer (Windows 10+ only) - Remote Access feature
  • Add ReceivedRemoteControlInvitation and SentRemoteControlInvitation
  • Add Stream#isScreensharingRemoteAccess()
  • Add reason on stop event for remote Access

Fixed

  • Error on checkstopAudioNodeCtxConditions()
  • Fix remoteAccess pointer depending of video display
  • Fix unsubscribed stream due to CALL_ABORTED (SFU transition with simultaneous replacePublishStream() )

5.0.4 - 2023-02-07

Fixed

  • screenSharing issue on P2P call

5.0.3 - 2023-01-27

Added

  • Possibility to remotely apply enable/disable media

Fixed

  • Blur compatibility test on Firefox
  • Issue on presence update on CCS on leavePresenceGroup and subscribePresenceGroup

5.0.2 - 2023-01-12

Fixed

  • Fixed set resolution for Edge/Chrome
  • Fixed apirtc.d.ts description

5.0.1 - 2023-01-11

Added

  • Added Stream#isAudioEnabled, Stream#isVideoEnabled to reflect MediaStreamTrack.enabled parameter
  • Added Stream#flowStatusChanged event

Changed

  • Changed Stream#isAudioMuted, Stream#isVideoMuted to reflect MediaStreamTrack.muted parameter instead of MediaStreamTrack.enabled parameter : /!\ YOU MAY NOW USE Stream#isAudioEnabled, Stream#isVideoEnabled

Fixed

  • Fixed Stream#isAudioMuted when stream has no audio
  • Fixed Stream#isVideoMuted when stream has no video
  • Fixed cloudConvId for screensharing calls
  • Fixed reconnection with calls without 200OK received

5.0.0 - 2022-12-21

Added

  • Added noise reduction feature
  • Added noiseReduction capability on UserAgent
  • Added event Stream#release
  • Stream#release also apply on children and parent streams by default

Changed

  • Changed : add security in Cloud interaction
  • Updated : upgrade to UAParser 1.0.32
  • Removed : ApiCCCoBrowsingClient
  • Changed : CDN path for blur files

Fixed

  • Remove deprecated usage of ScriptProcessorNode for activeSpeaker detection
  • Code smells from SonarQube

4.7.8 - 2022-11-02

Added

  • Added applicationUUID on connexion
  • Added bearerToken to Contact#fetchProfileInformation
  • Added bearerToken to Enterprise#fetchEnterpriseInformation
  • Added bearerToken to UserAgent#fetchProfileInformation
  • Added JWTApzToken management
  • Added FileInfo on DataChannel#sendData
  • Added captcha_token parameter on UserAgent#register

Changed

  • Changed : no leaveGroup on conversation presence group if done by application / API
  • Changed : no unsubscribe on conversation presence group if done by application / API
  • Changed error on UserAgent#register to add response from Cloud when registering a Cloud User
  • Update libJsCloud to 0.1.39
  • Removed call to cloudApi.getUserPhotoUrl()

4.7.7 - 2022-08-18

Fixed

  • Fixed : muted status for mesh calls
  • Fixed : stop screenSharing detection from browser
  • Cleaning code : remove toggleVideoScreen, switchVideoToScreen, switchScreenToVideo
  • Fixed : filename checking to applyVideoProcessor/background when using a data:image file

4.7.6 - 2022-07-11

Fixed

  • Fixed : image position and sizing for runPostProcessing
  • Fixed : optimize library loading for applyVideoProcessor
  • Fixed : optimize errors management on applyVideoProcessor

Added

  • Added timeout on stream mute/unmute events

4.7.5 - 2022-06-29

Added

  • Added Stream#applyVideoProcessor

Fixed

  • Fixed : no reconnection on server rejected errors (authentication, deactivated enterprise ...)
  • Fixed : audioMuted, videoMuted state for processed videos (blur, background)
  • Fixed : Fired on error when trying to apply video processing on an audio stream

4.7.4 - 2022-06-15

Added

  • Added Promise on unblur() and unBackgroundImage()

Fixed

  • Cleaning code : remove callURLDestRoom, remove DOM displayed elements (setStatus, ... )
  • Fixed : isScreensharing value in streamInfo for screensharing with audio stream
  • Fixed : Blur and background image optimization and error management
  • Fixed : add default values for device infos

4.7.3 - 2022-04-26

Added

  • Added Conversation#callVoIP
  • Added CONVERSATION_WAITING_ROOM_PREFIX into apirtc.d.ts
  • Added a separate apiRTC file with socket.io embedded
  • Added UserAgent#getCPUArchitecture
  • Added UserAgent#getDeviceVendor
  • Added UserAgent#getDeviceModel
  • Added UserAgent#getEngineName
  • Added UserAgent#getEngineVersion
  • Added deviceVendor, deviceModel, cpuArchitecture, engineName and engineVersion to basic userData

Fixed

  • Fixed Promise rejection upon register
  • Fixed UserAgent#getBrowser

4.7.2 - 2022-03-01

Added

  • Added Conversation#sendMessage options.token optionnal parameter
  • Added Stream background image

Changed

  • Fixed apirtc.d.ts file
  • Fixed documentation errors

4.7.1 - 2022-02-16

Added

  • Added Contact#fetchUserData

Changed

  • Desactivated Stream#applyConstraint
  • Desactivated Stream#setCapability
  • Changed sentry events location for mute/unmute to Core
  • Changed Stream#applyConstraints
  • Changed Stream#getSettings
  • Changed Stream#getConstraints
  • Changed Stream#getCapabilities
  • Removed Stream#askRemoteCapabilityAuthorization
  • Removed Stream#acceptRemoteCapabilityRequest
  • Removed Stream#refuseRemoteCapabilityRequest

Fixed

  • Fixed types for messages history

4.7.0 - 2022-01-25

Added

  • Added Stream#enableVideo
  • Added Stream#disableVideo
  • Added Stream#enableAudio
  • Added Stream#disableAudio
  • Added Stream#getAudioFlowStatus
  • Added Stream#getVideoFlowStatus
  • Added Stream#getRemoteAudioFlowStatus
  • Added Stream#getRemoteVideoFlowStatus
  • Added Stream events :
    • audioFlowStatusChanged
    • videoFlowStatusChanged
    • remoteAudioFlowStatusChanged
    • remoteVideoFlowStatusChanged
  • Added Contact#fetchUserData

Changed

  • Deprecated Stream#muteAudio
  • Deprecated Stream#unmuteAudio
  • Deprecated Stream#muteVideo
  • Deprecated Stream#unmuteVideo

4.6.5 - 2022-01-24

Fixed

  • Fixed room mode transition.

4.6.2 - 2022-01-12

Added

  • Added Stream#applyConstraint
  • Added Stream#applyConstraints
  • Added Stream#mergeReality
  • Added Stream#unmergeReality
  • Added Conversation#fetchMessageHistory new "options.token" possibility
  • Added EventCenter#disableStorage

Changed

  • Fixed token option for Conversation#fetchMediaList
  • Deprecated Stream#setCapability
  • Removed all references to Apizee from ApiRTC
  • Removed cookie usage to sessionStorage
  • Forced parameter bandwidth type to number for WebRTC_Client#setVideoBandwidth
  • Forced parameter bandwidth type to number for ApiCCWebRTCClient#setRemoteVideoBandwidth
  • Forced parameter bandwidth type to number for ApiCCWebRTCClient#setOverallIncomingVideoBandwidth
  • Forced parameter bandwidth type to number for ApiCCWebRTCClient#setOverallOutgoingVideoBandwidth
  • Forced parameter bandwidth type to number for ApiCCWebRTCClient#setPerCallIncomingVideoBandwidth
  • Forced parameter bandwidth type to number for ApiCCWebRTCClient#setPerCallOutgoingVideoBandwidth
  • Forced parameter kbps type to number for UserAgent#setTargetBandwidthUsage

4.6.1 - 2021-11-24

Added

  • Added error message for these browsers : Baidu, baidu, UCBrowser, QQ, QQBrowser and QQBrowserLite

Fixed

  • Fixed Stream methods to remotely use capabilities functions

4.6.0 - 2021-11-17

Added

  • Added Stream#blur
  • Added conversation moderation.
  • Added remote possibility on Stream functions : getCapabilities, getSettings, getConstraints, setCapability
  • Added Stream#askRemoteCapabilityAuthorization
  • Added Stream#acceptRemoteCapabilityRequest
  • Added Stream#refuseRemoteCapabilityRequest
  • Added Stream events :
    • streamCapabilities
    • streamSettings
    • streamConstraints
    • remoteCapabilityRequest
    • remoteCapabilityRequestAccepted
    • remoteCapabilityRequestRefused

Fixed

  • Fixed Promise rejection upon register with access token.

Changed

  • Removed sdpSemantics plan-b forcing for SIP calls
  • Removed CloudAPI from apiRTC to the npm package.
  • Removed all Stream helpers to set a specific capability : i.e. setTorch, setZoom, setContrast, etc...
  • Removed Stream#setCapabilities

4.5.6 - 2021-11-05

Fixed

  • Fixed hangup on set local/remote description failure.

4.5.5 - 2021-09-27

Added

  • Added checking on apiRTC loading when loading with NPM

4.5.4 - 2021-09-24

Added

  • Added Cloud events.
  • Added checking on apiRTC loading

4.5.3 - 2021-08-18

Added

  • Added meshOnlyEnabled option for Conversation/Conference (Session#getOrCreateConversation)
  • Added JoinResult for Conversation#join

4.5.2 - 2021-07-27

Fixed

  • Fixed 1-to-1 call with Safari calling without sending any media

4.5.1 - 2021-07-20

Fixed

  • Fixed preCall test to avoid mesh mode
  • Fixed session presenceGroup update on join/leaveSession
  • Fixed Edge version checking for Datachannel

Added

  • Added Conversation#getSubscribedStream
  • Added Laser mode on whiteboard

4.5.0 - 2021-06-04

Fixed

  • Fixed asymmetric 1-to-1 call with Firefox
  • Fixed Stream#createStreamFromUserMedia with enhancedAudioActivated when audio constraint is true
  • Fixed call stats for Chrome 91

Added

  • Added simulcast (Conversation)

  • Added streamId on Conversation events:

    • audioAmplitude
    • error
    • onCallStatsUpdate
    • remoteStreamUpdated
    • slowLink

Changed

  • Changed Conversation audioAmplitude event
  • Deprecated Conversation#getAvailableStreamList
  • PreCall test : added possibility to establish only one call for QoS testing with record

4.4.12 - 2021-07-27

Fixed

  • Fixed 1-to-1 call with Safari calling without sending any media

4.4.11 - 2021-05-05

Fixed

  • Fixed asymmetric 1-to-1 call with Firefox
  • Fixed call stats for chrome 91

4.4.10 - 2021-04-22

Fixed

  • Fixed asymmetric 1-to-1 call with Firefox
  • Added check against null for options argument

Changed

  • Optimisation of blur filter video quality

4.4.9 - 2021-03-19

Fixed

  • escape removeAllowExtmapMixed for Safari v14.0+ (605)

4.4.8 - 2021-02-17

Added

  • Added TextDetection AIE filter

Changed

  • Modification on backgroundSubstraction filters :
    • FPS optimisation when a filter is applied

4.4.7 - 2021-02-12

Fixed

  • Fixed subscription storm.

Added

  • Added STUN/TURN config from Cloud
  • Added logLevel config from Cloud
  • Added Session#getPresenceGroup
  • Added Session#getSubscribedPresenceGroup

Changed

  • Modification on presence :
    • Optimisation of join/leave presence group
    • Optimisation of sub/unsub presence group

4.4.6 - 2021-01-04

Fixed

  • Fixed ConversationCall#replacePublishedStream
  • Fixed callStatsUpdate trigger increase after ConversationCall#replacePublishedStream

Added

  • Added AIE development

Changed

  • Allowed MOS score for 1-to-1 calls

4.4.5 - 2020-11-26

Fixed

  • Fixed Stream#getCapabilities and Stream#setCapability compability with navigators

Added

  • Added helpers to control stream track capabilities

4.4.4 - 2020-11-02

Fixed

  • Fixed call restart after reconnection
  • Fixed missing hangup event on subscribe side in case of unpublish
  • Fixed Stream#startRecord for Firefox
  • Fixed updateSubscribedStream (streamId)
  • Fixed unsubscribeToStream (streamId)

Changed

  • Use RTCRtpSender#setParameters (instead of SDP AS/TIAS field) to limit bitrate

Added

  • Added callStats emiting for 1-to-1 call

4.4.3 - 2020-10-20

Added

  • Added firstMediaDeviceListInitialization on UserAgent#mediaDeviceChanged event

Fixed

  • Fixed call establishment and call clean up after mode transition
  • Fixed cloud request on IE11
  • Fixed instanceId change on republish stream

4.4.2 - 2020-10-13

Added

  • Added distributedBroadcastEnabled option for Conversation#publish
  • Added RessourceLoader to check Js dependancy loading

Fixed

  • Fixed remote logging integration
  • Fixed updateMedia process to add more checkings status

4.4.1 - 2020-09-25

Added

  • Added forceAudioAndVideoMediaElementSeparation option on addStreamInDiv and removeElementFromDiv
  • Added media stream filtration functionality

Fixed

  • Fixed call clean up on unrecoverable error
  • Fixed callHangup on CallDisconnection for P2P calls
  • Fixed setRecvOnly for audio and video when direction is already inactive

4.4.0 - 2020-08-03

Fixed

  • Fixed Conversation hangup event
  • Fixed apiCCId cookie on beforeunload
  • Fixed exports to have exported interface for UMD module
  • Fixed precall recording, recording only the passed test call
  • Fixed Datachannel on Edge > version 79
  • Fixed DataChannel 'opened' event

Changed

  • Changed Conversation#pushData scheduling: only one at a time

Added

  • Allowed multiple connections of same apiCCId in conference.
  • Added alternative method ApiCCUserMediaStream#listenToAudio2 to analyze audio amplitude
  • Added Stream#audioAmplitudeInfo event contained speaking/sound status besides amplitude info
  • Updated Stream#enableAudioAnalysis sound analysis method
  • Added active option for Session#getOrCreateConversation
  • Added NPM package generation support

4.3.23 - 2020-07-06

Fixed

  • Fixed Conversation#unpublish

4.3.22 - 2020-06-19

Fixed

  • Fixed P2P stream replacement created with ua#createStreamFromMediaStream
  • Fixed second stream creation with a second camera on Firefox

4.3.21 - 2020-06-05

Fixed

  • Fixed unordered packets during data channel transfer
  • Fixed camera switching on iOS < 13 (ideal/exact constraints issue)
  • Fixed recording for PreCallTest
  • Fixed updateMedia process to add more checkings on signaling status and add retry management
  • Fixed session control on subscribeToStreams
  • Fixed calls signaling message checking with callId and instanceId
  • Clean stored call candidate after one minute timer
  • Fixed getDisplayMedia detection on ios Cordova
  • Fixed Conversation#join (on failure)
  • Fixed eventlistener error on eventCenter for reactNative
  • Fixed ConversationCall#startRecordingPublishedStream while in mesh mode
  • Fixed ConversationCall#startStreaming while in mesh mode
  • Fixed to remove turns config for Edge version <= 44

Added

  • Added Conference#getInfo

4.3.20 - 2020-05-12

Added

  • Added optional TTL for Consersation#startRecording
  • Added optional TTL for ConsersationCall#startRecordingPublishedStream
  • Added a TTL of 900 seconds for recordings of precall test
  • Added optional sourceType of Conversation#pushData
  • Added possibility to configure VP9 as prefered codec on P2P call
  • Added screenSharingCompliant and qoSStatCompliant parameters on UserAgent#getCapabilities

4.3.19 - 2020-04-27

Fixed

  • clone data on subscribeToStreams
  • Add checking on feature deactivation on sendPresenceGroupManagementCommand
  • conversationSpace definition on register with login or token
  • Fixed screenSharing on Edge in mesh mode

Added

  • Added streamId in hangup event (conversation)
  • Added videoForbidInactive by default

Changed

  • change default value on publish(video + screen) for switch to audio only to 50kbps

4.3.18 - 2020-04-15

Changed

  • Modified control of call life-cycle.
  • Changed 'anonymous' default username by 'guest'
  • userData management on reconnect to accelerate synchronization

Added

  • Extended mediaDeviceChanged event with data that contains devices with updated labels

4.3.17 - 2020-04-02

Fixed

  • Fixed conversationSpace join in case of channel reconnect
  • Fixed presenceGroup join and subscribe in case of channel reconnect

4.3.16 - 2020-03-30

Fixed

  • Fixed publish call restart on reconnection in mesh mode.
  • Fixed calling evalOutgoingVideoBandwithCap() on caller side for P2P calls

4.3.15 - 2020-03-24

Changed

  • Change error format to display all information from WebRTC API on createStream promise
  • start updateMedia() on iceDisconnection only on caller side

Added

  • Added info on mediaDeviceChanged event to inform whether the event is generated after getUserMedia success
  • Added entryAllowed/entryDenied events to the conference

Fixed

  • Fixed unnecessary stopMaxCallRetriesTimer()
  • Fixed Promise reject on Conversation#publish failure

4.3.14 - 2020-03-03

Changed

  • setAudioSourceIdInConstraint() and setVideoSourceIdInConstraint() to use constraints.video.deviceId by default and 'ideal' instead of 'exact'
  • Modified Contact#sendMessage: removed Cloud api call to save message, duplicate entry in database.
  • Modified updatePresence right after being accepted in conference by moderator

Added

  • Added sequential CCS socket message handler in ApiCCSession#handleSequentialMessage()
  • Added call reestablishment (Conversation) after network loss
  • Added QoS videoMinBitrate and videoStartBitrate (Conversation#publish)
  • Added advice on slowLink (Conversation)
  • Added data channel checking in case of page refresh
  • Change createUserMediaErrorEvent() error format to display all information from WebRTC API, add gum_config info
  • Added info on EventCenter logs
  • Added EventCenter in public API
  • Added info concernant added and removed media devices to mediaDeviceChanged event at event.detail.diff property

    master

4.3.13 - 2020-02-11

Changed

  • Remove call restart when "no candidate where retrieved" event after evolution on SFU side (Subscribe iOS issue)
  • Change default value of SFU Connector

Fixed

  • Remove unnessary cleaning on beforeUnload Function
  • Add values checking for PreCallTest on observeStreamPublished() an observeStreamSubscribed()
  • setGetUserMediaConfig() to avoid malformed constraints
  • Remove reinitialization of GumConfig on getMediaDevices

Added

  • Added EventCenter that handles error, userMediaError events. Possibility to request errors remotely.
  • Added updatePresence right after being accepted in conference by moderator

Changed

  • Deprecated WebRTC_Client#setVideoBandwidth

4.3.12 - 2020-01-30

Fixed

  • Fixed UserAgent#createStream signature
  • Update for reactNative support
  • Rollback fetchNetworkInformation() on first version as Version 2 can use too much bandwith for testing
  • Fixed video cap for conferences with 3 participants and more

4.3.11 - 2020-01-24

Added

  • Added option to disable checksum during pushData.
  • Added posibility to configure bandwidthRatingThresholds on fetchNetworkInformation()

Changed

  • Modify fetchNetworkInformation() method for better network testings

Fixed

  • Added check on event values on onRemoteTrackAdded
  • Forced SDP plan-b for SIP outgoing Call on Chrome

4.3.10 - 2020-01-15

Added

  • Added checking on setGetUserMediaConfig() to avoid malformed constraints : Cannot use both optional/mandatory and specific or advanced constraints
  • Added QoS videoForbidInactive option for subscription call.
  • Added instanceId to identify calls on call restart
  • Update user agent parser
  • Keep instanceId for conf calls in P2P mode

Fixed

  • Fixed bitrate call update during multiple 1-to-1 calls.
  • Fixed switching camera issue when recording on Cordova.

4.3.9 - 2019-12-11

Fixed

  • Fixed canceled after success pushData notification.

Changed

  • Modified Conversation#cancelJoin signature.

Added

  • Added automatic call status check.

4.3.8 - 2019-12-10

Changed

  • Added cause in localStreamUpdated and remoteStreamUpdated events (Conversation).
  • Added delay on outgoing video cap during mode transition.

Added

  • Added max duration for calls retries
  • Added delay management on calls retries
  • Added error CALL_ABORTED event on #Conversation and #Call

4.3.7 - 2019-11-28

Fixed

  • Fixed presenceGroup max limitation length to 750

4.3.6 - 2019-11-26

Fixed

  • Fixed screenSharing publish without any audio/video device.

4.3.5 - 2019-11-20

Added

  • Added meshPublish info in data
  • Added possibility to set mimeType and quality argument on Stream#SnapshotOptions

Fixed

  • Fixed video cap for recorded 1-1 call.
  • Fixed Conversation#pushData promise rejection in case of ICE failed.

4.3.4 - 2019-11-04

Added

  • Added call stats for publisher, in Conversation, in mesh mode.
  • Added meshPublish info in callList for reconnectContext
  • Added possibility to configure width and height on takesnapshot (V3 API)
  • Added the mail property into the UserAgent user data.

Changed

  • Modified UserAgent#setOverallOutgoingVideoBandwidth, constraint applied immediately.
  • Modified UserAgent#setPerCallOutgoingVideoBandwidth, constraint applied immediately.
  • Modified pushData - wait for ack to determine success.
  • Removed force sdpSemantics plan-b for Chrome >= 78

Fixed

  • Added asymmetric 1-to-1 recorded call.

4.3.3 - 2019-10-10

Added

  • Added asymmetric 1-to-1 recorded call.
  • Added cloudConversationId for Contact#call options.
  • Added apiRTC.webRTCCompliant and apiRTC.isWebRTCCompliant().
  • Added checking on getSupportedConstraints() method support on device.
  • Added userMediaError on UserAgent.
  • Added apiRTC.qoSStatCompliant and apiRTC.isQoSStatCompliant().
  • Added UserAgent#getCapabilities(), UserAgent#getBrowser(), UserAgent#getBrowserMajorVersion(), UserAgent#getBrowserVersion()
  • Added UserAgent#getOsName(), UserAgent#getOsVersion(), UserAgent#getBrowserInfo(), UserAgent#getBrowserDetails()

Changed

  • Modified CloudApi success condition (in 200-range).

Fixed

  • Force HD resolution instead of fullHD for iOS 13 devices.
  • Fixed enableActiveSpeakerDetecting threshold.
  • Fixed stream#type on audio only subscribe.

4.3.2 - 2019-10-01

Changed

  • Changed default CCS connexion protocol to HTTPS

4.3.1 - 2019-09-09

Added

  • Added apiLibType parameter on connection for conversationSpace checking.
  • Added getCapabilities() on mediaDevice to get supported resolution ...etc... on media devices

Fixed

  • Fixed clearDisputableEntry() issue on second media flow when switching from P2P mode to SFU

From 4.2.33

Fixed

  • Fixed audio addition on video-only 1-to-1 call for Safari.
  • Fixed audio-only and video-only 1-to-1 call for Firefox.

4.3.0 - 2019-09-05

Added

  • Removed the restrictions on the number of simultaneous joined conversations.
  • Added composite activation (Conversation).
  • Added compositeListChanged event (Conversation).
  • Added replayListChanged event (Conversation).
  • Added subscription of composite stream (Conversation).
  • Added subscription of replay streams (Conversation).
  • Added start-before-stop room mode transition.
  • Added start/stop recording (Call).
  • Added Session#error event

4.2.34 - 2019-09-16

Merged into 4.3.1

Fixed

  • Fixed audio-only and video-only 1-to-1 call for Firefox.

4.2.33 - 2019-09-09

Merged into 4.3.1

Fixed

  • Fixed audio addition on video-only 1-to-1 call for Safari.

4.2.32 - 2019-09-05

Fixed

  • Fixed audio addition on video-only 1-to-1 call for Firefox and Safari.

4.2.31 - 2019-08-20

Fixed

  • Fixed subsequent recorded calls.

4.2.30 - 2019-08-19

Fixed

  • Added possibility to take snapshot from an existing div (issue on iOS)

4.2.29 - 2019-08-07

Fixed

  • Fixed audio addition on video only 1-to-1 call for iOS.

4.2.28 - 2019-07-25

Fixed

  • Fixed switch camera after having add a video track to a audio only 1-to-1 call.
  • Fixed screen sharing for Chrome 72/73.

4.2.27 - 2019-07-23

Fixed

  • Fixed switch camera in SFU mode.

4.2.26 - 2019-07-17

Fixed

  • Fixed whiteboard on IE.
  • Fixed idPersistenceTimeout after page reload.
  • Fixed getUserMedia constraints for video-only 1-to-1 call.

4.2.25 - 2019-07-08

Added

  • activateScreenSharing on UserAgent to enable screenStream creation in offline mode
  • Stream#getLabels() to get the labels of devices sources used for the stream

Fixed

  • Fixed video addition on audio 1-to-1 call.
  • Fixed destCallType for video only 1-to-1 call.

4.2.24 - 2019-07-01

Fixed

  • Fixed streamListChanged events during room mode transition.

4.2.23 - 2019-07-01

Added

  • Added user defined context for publish.
  • Added Stream#createDisplayMediaStream() to support constraints parameters for getDisplayMedia() (add screenSharing compatibility on Opera, Vivaldi, Brave, Safari Preview ... )

Changed

  • Added use of ontrack chrome 64+.

Fixed

  • Fixed call restart with channel disconnection.
  • Fixed pointer sharing activation after guest refesh.
  • Fixed audio addition on video only 1-to-1 call (and reverse).

4.2.22 - 2019-06-28

Fixed

  • Fixed room mode transition.

4.2.21 - 2019-06-18

Added

  • Added ccsToken parameter in Session

Fixed

  • Fixed media type selection on incoming call.
  • Fixed SIP number detection

4.2.20 - 2019-06-18

Fixed

  • Masked unwanted streamListChanged events during room mode transition.
  • Issue on media device presence status

4.2.19 - 2019-06-12

Fixed

  • Fixed switch camera on Safari iOS.
  • Add checking on constraints format in setAudioSourceIdInConstraint and setVideoSourceIdInConstraint
  • Fixed missing audio/video tracks from remote after switch camera.

Added

  • Added Token parameter on userAgent#register()

4.2.18 - 2019-06-11

Changed

  • Replaced PUT/DELETE methods by POST method and added _method parameter accordingly, in CloudApi.

Fixed

  • Fixed destCallType set to audio (instead of audioOnly).
  • Fixed audio-only publish call when getUserMedia is done during call establishment.

4.2.17 - 2019-06-04

Added

  • Added audio/video mute info in streamListChanged event (Conversation).
  • Added getUserMedia timeout for Chrome.

Changed

  • Modified fetch-retry parameters for Conference#setTags.
  • Fixed composite recording after room mode transition.

4.2.16 - 2019-05-16

Added

  • Added streaming management on ConversationCall.
  • Added idPersistenceTimeout parameter for UserAgent#register.

Changed

  • UserAgent#setOverallIncomingVideoBandwidth accepts illegal value (0 or less) to disable the limit.
  • UserAgent#setOverallOutgoingVideoBandwidth accepts illegal value (0 or less) to disable the limit.
  • UserAgent#setPerCallIncomingVideoBandwidth accepts illegal value (0 or less) to disable the limit.
  • UserAgent#setPerCallOutgoingVideoBandwidth accepts illegal value (0 or less) to disable the limit.

Fixed

  • Fixed getStats for Firefox 63 and newer.
  • Fixed selected candidate for Chrome based browsers.

4.2.15 - 2019-05-07

Added

  • Added metadata param on cloudApi#saveGroupChatMessage
  • Added metadata to group chat message.
  • Added tags management.

4.2.14 - 2019-04-26

Added

  • Added bandwidth reservation process.
  • Added overwrite option for Conversation#pushData.
  • Added ttl (time to live) option for Conversation#pushData.
  • Added videoStartQuality for conference publish.
  • Added transfer start timeout for Conversation#pushData.
  • Added better pc_config management and add mp2.apizee.com
  • Added possibility to configure width and height on stream#takeSnapshot()
  • Added Conversation#getContactsNumber()

Changed

  • Change screen sharing stream type to STREAM_TYPE_VIDEO.
  • Change conference waiting room implementation.
  • Removed H.264 restriction for Safari 12.1 and higher.
  • Added delay before joining waiting room.

Fixed

  • Added room ID check upon session update.
  • Reject promise on addMedia()
  • ReplacePublishedStream promise take addMedia() result into account
  • Fixed roomJoined parameter value to enable whiteboard restart
  • Fixed subscription to an unmuted (initially muted) feed.
  • Fixed Manage case when mediaDevice are detected on userAgent and mediaDeviceChanged handler is not yet setted
  • Fixed takeSnapshot modif and documentation

4.2.13 - 2019-04-11

Fixed

  • Fixed issue on UserAgent#getUserMediaDevices() on Safari 12.1
  • Fixed issue on call renegotiation on Safari 12.1

4.2.12 - 2019-04-10

Added

  • Added Conference#eject.
  • Added ReceivedConversationJoinRequest export.
  • Added (Conversation/Conference) messageNotDelivered event.

Fixed

  • Fixed call restart.
  • Fixed destCallType value : add checking on setRemoteCallProfile to process on publish or shareScreen message
  • Fixed update adapter.js - issue with publish on Safari
  • Fixed mediaConstraints for createOffer on old chrome version (< 55)

4.2.11 - 2019-04-04

Added

  • Added userAcceptOnIncomingScreenSharingCall parameter on UserAgent#register() to enable invitation management on screenSharing incoming calls
  • Added for screenSharing on acceptCall() : forcing unidirectionnal call
  • Added Contact#incomingScreenSharingCallInvitation
  • Added Session#incomingScreenSharingCallInvitation
  • Added filtering if a second 200OK is received for a call (call forking)
  • Added recordingStopped event in Conversation/Conference.
  • Added Conversation#isRecorded
  • Added Conversation#getRecordingInfo
  • Added ConversationCall#isRecorded
  • Added ConversationCall#getRecordingInfo
  • Added Session#joinConversationSpace()
  • Added Session#leaveConversationSpace()
  • Added SIP info in StreamInfo (Conversation)

Changed

  • Made CloudApi#setCloudURL static.
  • Added additional info in recording related events.
  • Fixed (Conversation) transferEnded event.
  • Fixed CloudApi for headers.
  • Added apiCCId forgetting on authenticated session disconnection.

4.2.10 - 2019-03-21

Fixed

  • Fixed blurLevel issue on takesnapshot()

4.2.9 - 2019-03-20

Fixed

  • Fixed conversationSpace initialisation in case of apiRTC re-init
  • Remove unuseful "call not found" warning

4.2.8 - 2019-03-18

Added

  • Added checkMutingStateForUserMedia to control muting state in P2P conference.
  • Added support of conversationSpace
  • Added screenSharing with getDisplayMedia API for Chrome > 72
  • Added Promise return on stopNewWhiteboardSession()
  • Added control on apiCCId (128 characters maximum)
  • Adding static code analysis

4.2.7 - 2019-02-21

Added

  • Added apiKey format control.
  • Added Conversation#getStreamInfo
  • Added videoOnly parameter on Conversation#PublishOptions and Conversation#SubscribeOptions
  • Added chatbot messaging.
  • Added strip of rtcp-fb parameters on updateSDPcodecs

Fixed

  • Fixed waiting room.
  • Fixed browser detection in checkCandidateTypes

4.2.6 - 2019-02-15

Added

  • Added osName osVersion into joinSession message.
  • Added new configuration of composite recording.

Merged 4.1.38

Added

  • Added control to avoid Datachannel activation on Edge as it is not supported.

4.2.5 - 2019-02-12

Fixed

  • Fixed bug on video-only and screen sharing.

4.2.4 - 2019-02-11

Added

  • Added Conversation#cancelPushData.

Fixed

  • Fixed mute/unmute for remote stream.
  • Fixed checking on Whiteboard join

4.2.3 - 2019-01-30

Added

  • Added facingMode option on UserAgent#createStream

Fixed

  • fix PreCall test
  • fix on takeSnapshot to use the size of live video

Merged 4.1.37

Fixed

  • Fixed issue : only one instance of babel-polyfill is allowed

4.2.2 - 2019-01-15

Modified

  • Changed peer connection configuration for Firefox 53 and newer.

Merged 4.1.36

Fixed

  • Fixed CallType on events when using a screenSharing stream

Merged 4.1.35

Modified

  • Call getMediaDevices() in userMediaSuccess on all browsers (was only done for Safari and Edge)
  • Setting video Bandwidth default values to 3Mo for overallIncoming and overallOutgoing (used for conference)

4.2.1 - 2019-01-14

Modified

  • Replaced renegotiation by unpublish/publish to avoid audio-video desync during switch camera.

4.2.0 - 2018-12-19

Added

  • Added support for multiple conversations.
  • Added setting videoBandWidth after publish (on live).
  • Added pointer sharing on Session.
  • Added possibility to upload logs to Cloud

4.1.34 - 2019-01-15

Modified

  • Changed peer connection configuration for Firefox 53 and newer (TURNS).

4.1.33 - 2019-01-11

Added

  • Force sdpSemantics to 'plan-b' for Chrome >= 72
  • Deactivate user id conversion to numeric by default
  • Add registerInformation#idConversionActivated parameter to keep the possibility to convert user id

4.1.32 - 2018-12-21

Fixed

  • Fixed callConfiguration setting management on meshPublish

4.1.31 - 2018-12-10

Fixed

  • Fixed moderation on guest side (access-ack).
  • Fixed callType for screenSharing call.
  • Fixed page unload detection on Mobile Safari for correct disconnection and apiCCId preservation

4.1.30 - 2018-12-04

Fixed

  • Fixed support for audio-only and video-only user media grab.
  • Fixed call restart after screenSharing extension installation on Chrome
  • Fixed screenSharing extension installation management for Chrome 71

4.1.29 - 2018-11-28

Added

  • Added "Muted" and "NoStream" values to QoS quality scores.

Fixed

  • Fixed resolution parameter for videoQuality model.
  • Fixed Stream recording. Issue when playing recorded files on Windows.

4.1.28 - 2018-11-23

Fixed

  • Fixed camera checking and selection on Safari iOS (2).
  • Added check on startCallStatsMonitoring for Safari : only supported on Chrome and Firefox for now.

4.1.27 - 2018-11-22

Added

  • Added check on WebRTCClient#enableQos() : only supported on Chrome and Firefox for now.
  • Added check on WebRTCClient#enableCallStatsMonitoring() : only supported on Chrome and Firefox for now.
  • Added check on WebRTCClient#enableQualityEvaluating() : only supported on Chrome and Firefox for now.
  • CloudAPI#setCloudURL()

Fixed

  • Fixed WebRTCClient#getMediaDevices() to update UserAgent#getUserMediaDevices().
  • Fixed micro selection on Safari iOS.
  • Fixed camera checking and selection on Safari iOS.

4.1.26 - 2018-11-20

Fixed

  • Fixed call restart and stream protection.
  • Fixed deactivation of active speaker on iOS-Cordova.

4.1.25 - 2018-11-20

Added

  • Added check on CallStatsMonitoring : only supported on Chrome and Firefox for now.
  • Added Conversation#audioAmplitudeUpdate event.

4.1.24 - 2018-11-15

Added

  • Added Stream#addInDiv() Stream#removeFromDiv() helpers to manage adding video in DOM. These methods includes support of Safari autoplay policies constraints to avoid WebRTC audio/video plays issues.
  • Added Conversation#hangup event.

Fixed

  • Switch Camera on iOS.
  • Fixed auto subscribe restart after failure. Use Conversation#streamAdded and Conversation#streamRemoved to update DOM.

4.1.23 - 2018-11-12

Added

  • Added UserAgent#enableCallStatsMonitoring
  • Added UserAgent#enableActiveSpeakerDetecting

Changed

  • addStream() and removeStream() to use addTrack / replaceTrack API by default
  • enhancedAudioActivated : add checking to activate only on Chrome

Fixed

  • Compatibility with Vivaldi Browser v1 and v2
  • Removed entry from userMediaStreamTable after failure

4.1.22 - 2018-10-30

Added

  • Added call auto restart when no candidate is found (Safari 12 without getUserMedia).

Added

  • Added active speaker detecting

Changed

  • Modified PeerConnection event for Safari 12 (ontrack instead of onaddstream).

Fixed

  • Fixed renegotiation for subscriber in mesh conference.

4.1.21 - 2018-10-18

Added

  • Added descriptor for ICE error.
  • Added event for distant ICE error.

4.1.20 - 2018-10-12

Added

  • Added Conference#cancelJoin.
  • Added enabling video quality events on Session
  • Added frame resolution statistics for sent stream

Modified

  • Enabled dataChannel on Safari version 11 and later

4.1.19 - 2018-10-09

Added

  • Added recording (userStream/composite) in Conversation.
  • Added Conference#cancelJoin.

Modified

  • Enabled dataChannel on Safari version 11 and later

Fixed

  • Fixed firing MediaDeviceChanged Event on UserAgent's instantiation
  • Fixed bug of wrong subscribe to replay.

4.1.18 - 2018-10-02

Added

  • checking call state on updateSubscribedStream() to avoid SDP negotiations error

Modified

  • fire "CallEstablished" event on set remoteDescription success

Fixed

  • add control on userData value
  • add error value on createScreensharingStream() promise reject

4.1.17 - 2018-09-26

Fixed

  • Fix for Hangup on iOS Cordova

4.1.16 - 2018-09-25

Added

  • add screenSharing support on Edge (only available on Windows 10 Pro or Enterprise)

Fixed

  • Fixed issue on call establishment with Edge (call peer to peer and publish to conference)
  • Fixed issue with Chrome 55

4.1.15 - 2018-09-20

Modified

  • Remove subscribeToMedia and unsubscribeToMedia from API Doc. These methods are not useful.
  • Improve mediaDevice change detection for Safari and Edge. Add posibility to activate apiRTCMediaDeviceDetectionEnabled on Safari

4.1.14 - 2018-09-12

Added

  • mediaDeviceChanged event support by apiRTC on Chrome/Android as not supported by browser (parameters apiRTCMediaDeviceDetectionEnabled, apiRTCMediaDeviceDetectionDelay)
  • add enhancedAudioActivated parameters on CreateStreamOptions

4.1.13 - 2018-09-10

Fixed

  • Fixed on userData update when user already in contact list

Modified

  • Get active session if no session is passed as options parameter for Conversation#checkAccess

4.1.12 - 2018-09-04

Added

  • Added conference waiting room messages for exchange with izeeconf.
  • Added 'statusChange' event in Invitation (Available for all kinds of invitation)
  • Added Invitation status : Constants.INVITATION_STATUS_CANCELLED
  • Added Invitation status : Constants.INVITATION_STATUS_ENDED
  • Added cancel() on SendInvitation for FileTransferInvitation - sendFile()
  • type parameter on Invitation

Fixed

  • Correction for fileInfo value on Invitation
  • Correction of remoteId on 'hangup' event
  • Correction of callId value on 'userMediaStop' event

Modified

  • Added convId parameter for fetchEnterpriseInformations
  • Added sessionId parameter as value returned by fetchEnterpriseInformations

Deprecated

  • 'expired' event on ReceivedCallInvitation : check 'statusChange' event
  • 'expired' event on ReceivedDataChannelInvitation : check 'statusChange' event
  • 'response' event on SentInvitation : check 'statusChange' event

4.1.11 - 2018-08-22

Fixed

  • Fixed IE11 console issue
  • Fixed subscribe audio only and video only in P2P conference.

Added

  • Added ApiCCWebRTCClient#updateMediaTypeOnCall
  • Added videoOnly option for Conversation#subscribeToStream
  • Added Conversation#updateSubscribedStream

4.1.10 - 2018-08-21

Fixed

  • avoid exception on loadJs if file is requested twice
  • Fixed IE8 / IE11 support (avoid devices detection, update call management)
  • Added audio/video mute support for P2P conference.

4.1.9 - 2018-08-20

Fixed

  • Cleaning whiteboard elements on start() and stop()
  • Whiteboard drawingId on annotation
  • Fixed issue on contact creation when userAgent is not completely instantiated
  • Fixed issue on contactListUpdate when session is not completely instantiated

4.1.8 - 2018-08-11

Added

  • Added UserAgent#fetchGeolocationInformation in DTS file
  • Added Conversation#getStatus
  • Added Stream#getOwner
  • Added the possibility to initialize some userData parameters at register
  • Added streamId in localStreamUpdated event fired by Conversation

Fixed

  • Removed Stream#isLocal in DTS file
  • Added missing return in UserAgent#getUsername and UserAgent#getPhotoUrl
  • Added missing Stream#callId property in DTS file
  • Fixed Stream#streamId property type in DTS file
  • Fixed remoteStreamUpdated event firing order

Modified

  • Removed _contactCache on Conversation
  • Modification of Contacts management on users presence :
    • Optimisation of connectedUsersListUpdate event management
    • Remove updatePresence event process
    • Remove throttle on Contacts list update
    • Avoid creation of Contact for own user

Fixed

  • removed Conversation#contactListUpdate as not filtered by conversation and events contactJoined / contactLeft are more precise
  • add filtering of incomingCall event for conference calls

4.1.7 - 2018-08-07

Added

  • Added (transfer) id for transferBegun, transferProgress and transferEnded events (Conversation)
  • Added Conversation#isPublishedStream
  • Added audio/video update from audio-only and video-only calls in Conversation
  • Added MCUAvailableStreamUpdate event (Core)
  • Added Conversation#checkAccess
  • Added access-ack on moderation granted
  • Added Session#getOrCreateConference
  • Added check for electron application

Modified

  • Added possibility to get a blob instead of dataURI from takeSnapshot
  • Added isRemote property on Stream
  • Speed up data channel transfer to cloud
  • Added wait for CCS ack before closing data channel after transfer
  • Removed call auto-accept when no audio and video present
  • Added convId optional parameter to Enterprise#fetchSiteAgents
  • Added moderator info in Conversation#checkAccess result
  • Modified special receiveData listener in Conference
  • Fixed missing return after reject in CloudApi#cloudRequest
  • Fixed missing values in Conversation#checkAccess result

Fixed

  • Added UserAgent#getDefaultDevices in DTS file
  • Set stream parameter as optional for Conversation#publish in DTS file
  • Added UserAgent#getUserMediaDevices in DTS file

4.1.6 - 2018-07-20

Added

  • Added UserAgent#fetchGeolocationInformation
  • Added CloudApi (public)

Fixed

  • Fixed fileTransfer for file of type string

4.1.5 - 2018-07-17

Fixed

  • Fixed error during data call when audio/video is globally muted.
  • Added clean up of mesh published stream on leaveSession.
  • Fixed Conversation#destroy and Conference#destroy.
  • Fixed error when publishing audio-only stream in P2P conference.

4.1.4 - 2018-07-11

Modified

  • Added class UserData to represent user data of user agent and contact.
  • Fixed dataChannel transfer (onclose not supported on firefox).
  • Modified total hang up order.
  • Added clean up of MCU session on leave.

4.1.3 - 2018-07-06

Added

  • Generation of lib version min.debug

Fixed

  • apiDBActivated is now activated by default (case of connection/deconnection)

4.1.2 - 2018-07-05

Added

  • Added tests for mute/unmute during calls.
  • Added remoteStreamUpdated event in Call.

Modified

  • Added listeners clean up in Call.
  • Added remoteId in userMediaSuccess event.

4.1.1 - 2018-07-05

Added

  • Session#getOnlineContactsArray()
  • PreCallTest() function

Modified

  • apiDBActivated is now activated by default

Fixed

  • createStream with videoInputId false was creating a video stream

4.1.0 - 2018-06-29

Added

  • ReceivedCallInvitation#getCall()
  • Events Contact#incomingCall
  • Events Contact#incomingScreenSharingCall
  • Events Session#incomingScreenSharingCall
  • Events Call#userMediaError
  • Events Call#desktopCapture
  • Call creation when invitation is received
  • Session#getOrCreateContact()
  • Session#getCall()
  • Session#getCalls()
  • Session#getContactsArray()
  • Stream#getLocalStreams()
  • Stream#getStream()
  • Contact#shareScreen()
  • Possibility to configure captureSourceType on Stream#createScreensharingStream

Fixed

  • Fixed call.callConfiguration merge on acceptCall
  • Checking if audioReceived present for QoS statistics.
  • Fixed mute/unmute (state variable duplication)
  • Fixed tryAudioCallAfterUserMediaError
  • Fixed mute state for muted call establishment.

Modified

  • Removed use of MediaStream#clone.
  • Added check for roomId on ApiCCWebRTCClient#joinMCUSession
  • Deprecation of Session#getContact() replaced by Session#getOrCreateContact()
  • Cleaning of Stream#createScreensharingStream
  • Modified localStreamUpdated and remoteStreamUpdated event firing.

4.0.14 - 2018-06-29

Fixed

  • Remove typeof control on inviteInRoom
  • Fixed when a video stream is used to establish call and call is audioOnly : call was using video
  • Fixed SDPManager#setSendOnlyForAudio
  • Fixed SDPManager#setSendOnlyForVideo
  • Fixed SDPManager#setSendOnly
  • Fixed SDPManager#setRecvOnlyForAudio
  • Fixed SDPManager#setRecvOnlyForVideo
  • Fixed SDPManager#setRecvOnly

Modified

  • Modified Call#replacePublishedStreams: resolve the returned promise only when the action is done.

Added

  • Added Call#stopPublishedStreams

4.0.13 - 2018-06-26

Added

  • Added ConversationCall, created by Conversation/Conference on publish.
  • Added ConversationCall#replacePublishedStream.
  • Added Conversation#destroy to release conversation resources.
  • Added Conference#destroy to release conference resources.

Modified

  • Added callbacks parameters for Call#replacePublishedStream.
  • Removed remaining reference of previous UserMediaStream in WebRTC_Client#addMedia.

4.0.12 - 2018-06-25

Added

  • Added pointer sharing roomType.
  • Added nickname in options for anonymous UserAgent#register.

Fixed

  • Whiteboard optimisation :
    • History is now managed by CCS in case of disconnection / reconnection
    • Whiteboard cursor is deactivated by default (invisible))
    • Added capabilities detection on userAgent to deactivate cursor on iOS
    • Not sending cursor information when cursor is set to invisible
    • Avoid redraw when not necessary

4.0.11 - 2018-06-20

Added

  • Added possibility to set userAccept on incoming dataCall on User Agent (RegisterInformation).

Modified

  • Removed 'Conversation.' prefix from Conversation.
  • Added timeout for webRTCClientCreated event firing.
  • Fixed undefined address for turn server.

Fixed

  • Fixed issue on webRTC library usage when using Safari and cordova is defined

4.0.10 - 2018-06-14

Added

  • Added QoS preferences for Conversation#publish
  • Added uuid from CCS on groupChatMessage sending

Fixed

  • Fixed Call#replacePublishedStreams for mute/unmute
  • Fixed updatePresence request to cloud for conference

4.0.9 - 2018-06-07

Added

  • Added Events Conversation#disconnectionWarning
  • Added Events Conversation#error
  • Added UserAgent#enableMeshRoomMode()
  • Added UserAgent#fetchProfileInformation()
  • Added UserAgent#getPhotoUrl()
  • Added events on Stream
    • Added Stream#activeStateChange
  • Added events on Conversation
    • Conversation#localStreamUpdated
    • Conversation#remoteStreamUpdated
    • Conversation#slowLink
  • Added possibility to set turn server address on User Agent (RegisterInformation). This parameter is a global parameter : all established calls will use this turn server.
  • Added possibility to set turn server address on Conversation#publish() (PublishOptions) . This parameter is only set for the published call.
  • Added possibility to set turn server address on Conversation#subscribeToStream() (SubscribeOptions) . This parameter is only set for the subscribed call.
  • Added possibility to set turn server address on Conversation#pushData() (PushDataOptions) . This parameter is only set for the data call.

Fixed

  • Fixed parameter token update on deconnection/reconnection

4.0.8 - 2018-05-30

Added

  • Added control to avoid Stream constructor usage, please use a static method of class Stream such as : createStreamFromUserMedia, createStreamFromMediaStream, createScreensharingStream
  • Cleaning code on deprecated usage of selectedUserMediaStreamId
  • Added Events Session#reconnecting
  • Added possibility to configure contactDisconnectionDelay (RegisterInformation)

Fixed

  • Fixed Session events list checking to add : 'disconnect' and 'error'

4.0.7 - 2018-05-28

Added

  • Added possibility to configure ccs connection retries number and delay (RegisterInformation)

Fixed

  • Fixed issue on removeGroupDataFromConnectedUsersList when user leave conference

4.0.6 - 2018-05-24

Added

  • Added UserAgent#externalJsLoadingStatus event to be informed on external Js status : loaded, error or retry. This can be an insteresting information to detect network failure.
  • Added UserAgent#ccsConnectionStatus event to be informed on CCS connection status : connected, disconnected, error, retry
  • Added possibility to configure external Js loading retries number and delay (RegisterInformation)
  • Added result to joinMCUSessionAnswer event

Modified

  • Optimize bandwidth test duration

Fixed

  • Fixed issue on stream cleaning when replacePublishedStream is used

4.0.5 - 2018-05-17

Added

  • MediaDevice Manager to improve getting media devices and update when a device is added or removed
  • UserAgent#getUserMediaDevices
  • UserAgent#getDefaultDevices

Fixed

  • Fixed UserAgent#RegisterInformation : remove presence group subscription when joining : this is done using parameter subscribeTo
  • Fixed issue on presenceGroupManagement when disconnect/reconnect
  • Fixed issue on UserAgent#fetchNetworkInformation
  • Fixed issue to use requirejs : conflict with webpack
  • update bandwidthRatingThresholds

Modified

  • Change asynchronous js loader in order to have retries and error management

Removed

  • Removed Presence group subscription to visitor - agent when connecting with apiKey : presence & subscribe group must be unset by default

4.0.4 - 2018-04-26

Modified

  • Stream#startRecord now return a promise
  • Stream#stopRecord now return a promise with Blob of recorded stream
  • Stream#pauseRecord now return a promise
  • Stream#resumeRecord now return a promise

Added

  • Constants.ERROR_STREAM_RECORD : used to identify error on stream recording

4.0.3 - 2018-04-24

Fixed

  • Stream release when a stream is provided on call and conversation
  • Stream muting/unmuting (audio & video) when a stream is provided on call and conversation
  • Fixed screensharing for Firefox : default captureSourceType is now "screen"
  • Fixed parameter restarted on deconnection/reconnection on UserMediaSuccess Event
  • Fixed call restarting with MCU when disconnection
    • restart on bye "disconnect" reason
    • userMediaStream presence and active state checking on restart
  • Fixed usage of plugin functions for iOS/cordova

Modified

  • Check apiKey modification on apiCCId checking when reloading page
  • Update events handlers on CCS connection
  • Modified ApiCCWebRTCClient#updateMediaDeviceOnCall: replaced stream parameter (deprecated) by userMediaStreamId.

Added

  • Added warning when no candidate are retrieved from ice agent on call
  • Added support for WebRTC evolution API on Firefox-59 and Chrome-65 on addStream() and removeStream()
  • Added checking of media device presence before using it on getUserMedia when a deviceId is set
  • Added detection of media device update:
    • Added UserAgent#mediaDeviceChanged event
  • Added management of connection error on register :
    • Cloud CCS connection error and retries
      • Added retries on cloud request
      • Added cloudFetchRetries, cloudFetchRetryDelay configuration parameters on registerInformation
    • CCS connection : authentication error
    • Events on Session
      • Added Session#disconnect
      • Added Session#error
  • Added support for media update (renegotiation).
  • Added UserAgent#createStreamFromMediaStream in order to create a Stream from a mediaStream

Removed

  • Removed Session#disconnectionWarning event
  • Removed UserAgent#getDefaultDevices

4.0.2 - 2018-04-06

Fixed

  • typo : Hangup reason 'User_Refuse_Call' replaced by 'User_Refused_Call'
  • Added mimeType in blob return by stopRecord()

4.0.1 - 2018-04-06

Modified

  • Optimization : Stream#startRecord() : change default recording mimeType to 'video/webm;codecs=vp8' instead of 'video/webm;codecs=vp9'

Added

  • Added posibility to configure mimeType on Stream#startRecord()
  • Added Stream#pauseRecord
  • Added Stream#resumeRecord
  • Added checking of apiKey when getting apiCCId from cookie
  • Support presenceGroup and subscribeToPresenceGroup modifications on disconnect / reconnect

Fixed

  • Fixed tryAudioCall after userMediaError

4.0.0 - 2018-03-23

This version is the merge of ApiRTC and ApiRTC2 : Project name is apiRTC and version is v4.0.0 (former apiRTC version is 3.18.9)

Compatibility with apiRTC2 projects is maintained :

API of ApiRTC v3 is also maintained in order to keep compatibility with existing applications.

Modified

  • Major change with apiRTC2/ApiRTC merging

Fixed

  • apiRTCManager and Session event cleaning on disconnect()
  • clean Conversation event cleaning on leave()

Replaced

  • Typo : persistantDataUpdated event by persistentDataUpdated

Changelog for older apiRTC version