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Changelog

All Notable changes to apiRTC are documented in this file.

Latest version : apiRTC Latest

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

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 :

• ApiRTC2 keyword is preserved even if ApiRTC is preferred
• Only the library path change : https://cloud.apizee.com/apiRTC/vX.X

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

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Changelog for older apiRTC version