wrtc is an object of class wrtcObject that is automatically created by wrtc_kit.js
It takes care of connecting to local media, different events that occur, and starting calls and group calls.
Users should not edit anything inside wrtc_kit.js nor compress it in any way. Include it unchanged into your projects.
The wrtc object can be controlled using function calls and properties and it generates events.
| wrtc.acceptAudio | request audio in calls |
| wrtc.acceptVideo | request video in calls |
| wrtc.addressName | unique username / SIP address |
| wrtc.app | descriptive application name |
| wrtc.audioDestination | select local audio destination (speaker) |
| wrtc.audioSource | select local audio source (microphone) |
| wrtc.camOnBetweenCalls | keep camera on when no calls |
| wrtc.displayName | user's informal, descriptive name |
| wrtc.getAudio | access local microphone |
| wrtc.getVideo | access local camera |
| wrtc.localVideo | video object used to show local camera image |
| wrtc.maxNrCalls | maximum number of simultaneous calls allowed |
| wrtc.nam | wrtc object name |
| wrtc.objType | object type |
| wrtc.sipAlias | user's SIP alias (secondary SIP account) |
| wrtc.sipAuthName | user's SIP authentication (login) name |
| wrtc.sipHa1 | hash sum of the user's SIP password |
| wrtc.sipPassword | user's SIP password |
| wrtc.videoBandwidth | max transmitted video bandwidth |
| wrtc.videoSource | select local video source (video) |
| wrtc.workgroups | list of workgroups user is member of |
| wrtc.wsServer | IXNX server's address |
Advanced properties that are usually not needed to be changed. In non-basic use cases however they might need to be adjusted.
| wrtc.adv.autoGroup | accept incoming calls during setup of group call automatically |
| wrtc.adv.callTo | call address automatically after page has been loaded |
| wrtc.adv.clientID | unique client ID |
| wrtc.adv.dialDTMF | dial DTMF sequence automaticly at beginning of call |
| wrtc.adv.dtmfInfo | send DTMF both inband and outband on Chrome |
| wrtc.adv.getMedia | access local media at once or at call |
| wrtc.adv.groupBridgeAddr | address of voice bridge used in group call |
| wrtc.adv.groupBridgeDname | display name of voice bridge used in group call |
| wrtc.adv.groupID | unique ID of group call |
| wrtc.adv.logSize | maxsize of internal log |
| wrtc.adv.multiConn | allow multiple connections to same user |
| wrtc.adv.requestBandwidth | max received video bandwidth |
| wrtc.adv.stunServer | URL to STUN server if other than IXNX unit's built-in one |
| wrtc.adv.template | select wrtc template to use |
| wrtc.adv.turnEnabled | enable use of TURN server |
| wrtc.adv.turnPwd | TURN server password, if required |
| wrtc.adv.turnServer | URL to TURN server if other than IXNX unit's built-in one |
| wrtc.adv.videoConstraints | image quality of local camera |
| wrtc.addToGroupCall() | add calls to group call |
| wrtc.call() | call a remote user |
| wrtc.callGroup() | create group call with multiple remote users |
| wrtc.changeCredentials() | change user adress, name, etc |
| wrtc.changeMedia() | change local media source (mic/cam) |
| wrtc.consoleLog() | log entry to internal log |
| wrtc.fetchRESTtoken() | load REST authentication token using HTTP GET |
| wrtc.findCallByPartner() | get call currently in call with address |
| wrtc.forceHttps() | reload page in https |
| wrtc.get() | get read-only parameter |
| wrtc.getCall() | get first matching call object |
| wrtc.getCalls() | get all matching call objects |
| wrtc.getCookie() | get a cookie value |
| wrtc.getList() | get a list from IXNX server |
| wrtc.getScreen() | start screensharing |
| wrtc.getURLvar() | get URL tag value |
| wrtc.getURLvars() | decode preconfigured URL tag values |
| wrtc.hangup() | finish all matching calls |
| wrtc.init() | object initialization function |
| wrtc.inviteGroup() | start group call |
| wrtc.is() | get read-only boolean parameter |
| wrtc.moduleLog() | log entry to internal log |
| wrtc.muteCam() | mute local camera |
| wrtc.muteMic() | mute local microphone |
| wrtc.removeFromGroupcall() | remove call from group call |
| wrtc.saveFile() | save to local file |
| wrtc.sendChatMessage() | send text message |
| wrtc.sendFile() | file transfer |
| wrtc.setAudioDestination() | set local audio destination (speaker) |
| wrtc.setCaptureDevice() | change local media source (mic/cam) |
| wrtc.setList() | store a list on IXNX server |
| wrtc.setRESTtoken() | set REST authentication token |
| wrtc.setSIPPresence() | publish SIP presence status |
| wrtc.setSIPSubscribe() | subscribe to SIP presence status |
| wrtc.setTimeout() | call object after timeout, if object still exists |
| wrtc.stopFile() | stop file transfer |
Advanced functions, for advanced users.
| wrtc.adv.createLoopBackCall() | start local call (for volume measurement) |
| wrtc.adv.getLog() | request log from remote user |
| wrtc.adv.getTokenVal() | extract public value from REST token |
| wrtc.adv.idleDetect() | start/stop user activity monitoring |
| wrtc.adv.newGroupLink() | create group call link |
| wrtc.adv.saveLog() | save log to local file |
| wrtc.adv.sendLog() | send log to remote user |
| wrtc.adv.sendWrtcCmd() | send user command to remote user |
| wrtc.adv.stop() | close web socket and terminate wrtc object (permanent stop) |
| wrtc.adv.stopLocalStream() | stop primary access to unit's camera and microphone |
| wrtc.adv.stopLoopBackCall() | end local call |
| wrtc.adv.volumeMeter() | measure audio volume |
Functions called by wrtc_kit when special events occur.
| wrtc.onAuthenticate() | authentication requested |
| wrtc.onAuthenticateFailed() | IXNX server did not accept you as user |
| wrtc.onChatMessage() | chat message received |
| wrtc.onChatMessageError() | chat messaging failed |
| wrtc.onConnecting() | call in progress of being established |
| wrtc.onDataReceived() | data has been received on data channel |
| wrtc.onDenied() | user denied incoming call |
| wrtc.onDisconnected() | call disconnected |
| wrtc.onEarlyMedia() | media (audio) received during setup of call |
| wrtc.onEndCallObject() | call object about to be destroyed |
| wrtc.onError() | an error has occurred |
| wrtc.onFileProgress() | file transfer in progress |
| wrtc.onFileReceived() | a complete file has been received |
| wrtc.onGotHungUp() | person you called denies or ends call |
| wrtc.onGotMuted() | remote client muted/unmuted its mic or cam |
| wrtc.onHold() | call has been put on hold or retrieved by other end |
| wrtc.onHungUp() | local user ended call |
| wrtc.onIdle() | user become inactive or active again |
| wrtc.onIncomingCall() | someone has called you |
| wrtc.onIncomingGroupCall() | call from other member of group call |
| wrtc.onIncompatibleVersion() | incompatible communication protocols |
| wrtc.onInviteGroupProgress() | group call setup progress status |
| wrtc.onIsTalking() | current talker in group call |
| wrtc.onJoinGroup() | group call invitation has been received |
| wrtc.onList() | list received from IXNX server |
| wrtc.onLocalMedia() | got access to local mic/cam, or failed to do so |
| wrtc.onLoggedIn() | IXNX server accepted you |
| wrtc.onLoggedOut() | IXNX server logged you out |
| wrtc.onMaintenanceMessage() | maintenance message received |
| wrtc.onMultiuserChanged() | connection lists multiple users |
| wrtc.onNeedRESTtoken() | REST authentication token about to expire |
| wrtc.onNewCallObject() | new call object created |
| wrtc.onOnlineList() | list of users connected to IXNX server |
| wrtc.onReconnecting() | call interrupted, trying to reconnect |
| wrtc.onRejected() | IXNX server rejected command |
| wrtc.onRemoteStreamStalled() | no media received from other end of call |
| wrtc.onRinging() | other end's phone is ringing |
| wrtc.onSIPPresence() | presence status of a subscribed person changes |
| wrtc.onSIPRawInfo() | SIP INFO packet received |
| wrtc.onSIPRegistration() | SIP registration results |
| wrtc.onSources() | list of audio and video sources has been populated (Chrome only) |
| wrtc.onStats() | statistics about received media stream (Chrome only) |
| wrtc.onTalking() | call has been established |
| wrtc.onTranscription() | voice to text transcription |
| wrtc.onTransferred() | call has been transferred |
| wrtc.onVisible() | web page becomes visible or hidden |
| wrtc.onVolume() | volume sample |
| wrtc.onWebsocket() | websocket connection to IXNX server opened, closed or failed |
| wrtc.onWrtcCmd() | user command received |
All events are part of the wrtc object - even events that were raised by eg the call object. For such events you need to examine the object supplied as argument to be able to determine eg which call raised the event.