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Changelog

All Notable changes to apiRTC are documented in this file.

Latest version : apiRTC Latest

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