nanoStream Webcaster API

RtcUser

nanoStream Webcaster Public API Class

Kind: global class
Version: 5.16.0

---

Constructors

new RtcUser()

An RtcUser is the main class and entry point of the nanoStream Webcaster API.

Example

var rtcUser = new RtcUser();

---

Functions

rtcUser.checkSupport() ⇒ int

Checks if nanoStream Webcaster is supported by current browser.

Kind: instance method of RtcUser

Example

var supportLevel = RtcUser.checkSupport();
// 0: fully supported, no expected issues
// 1: not supported
// 2: partly supported - an outdated browser version
// 3: partly supported - a critically outdated browser version or a browser with issues, e.g. Edge

rtcUser.signIn(options)

Signs in to the server. After signIn succeeded you will have a session with the Webcaster server until reloading the browser or calling signOut. Parameters in braces "[ ]" are optional.

Kind: instance method of RtcUser
Emits: SignInSuccess, SignInError

Param	Type	Description
options	object	The options object.
options.server	string	The URL to the Webcaster server.
[options.jwt]	string	The JWT access token for authentication. The JWT contains the ingest information, please also pass it in startBroadcast
[options.bintuApiKey]	string	The bintu API key for authentication.
[options.token]	string	The access token for authentication - Deprecated: Please migrate to use JWT for authentication.

Example (sign in with JWT)

// rtcUser: instance of RtcUser

var options = {
    server : 'https://bintu-webrtc.nanocosmos.de/p/webrtcws',
    jwt : 'eyJhbGciOiJSUzI1NiIsInR5cCI6IkpXVCIsImtpZCI6Im5hbm9jb3Ntb3MifQ...'
};

rtcUser.signIn(options);

Example (sign in with bintu API key)

// rtcUser: instance of RtcUser

var options = {
    server : 'https://bintu-webrtc.nanocosmos.de/p/webrtcws',
    bintuApiKey : 'awdegfq3490puerg2w54zj2p0w4h46zphm694i0796...'
};

rtcUser.signIn(options);

rtcUser.signOut()

Signs out from the server. The current session and a broadcast (if running) will be stopped.

Kind: instance method of RtcUser
Emits: SignOutSuccess, SignOutError

Example

// rtcUser: instance of RtcUser
rtcUser.signOut();

rtcUser.isSignedIn() ⇒ boolean

Checks if the RtcUser is connected with the Webcaster server and signed in (see signIn).

Kind: instance method of RtcUser

Example

// rtcUser: instance of RtcUser
if (rtcUser.isSignedIn()) {
    console.log('signed in');
} else {
    console.log('not signed in');
}

rtcUser.setConfig(config)

Sets global configuration for the RtcUser instance. Please note that setting the audio transcoding bitrate is currently done in startBroadcast. Parameters in braces "[ ]" are optional.

Kind: instance method of RtcUser
Emits: SetConfigError

Param	Type	Default	Description
config	object		The config object.
[config.logLevel]	0 | 1 | 2 | 3 | 4		Verbosity of console logging. Allowed values: 0 (minimum verbosity) - 4 (maximum verbosity).
[config.metrics]	object		The metrics configuration object.
[config.metrics.accountId]	string		An account id provided to you.
[config.metrics.accountKey]	string		An account key provided to you.
[config.metrics.eventId]	string		An id of an event a stream is related to.
[config.metrics.statsInterval]	number	10	The interval how often the stats should be collected in seconds. Allowed values: 5 - 10 second(s).
[config.metrics.customField*]	string		A custom field. * can be replaced with 1 - 10 e.g. 'customField3'. Possible from 'customField1' to 'customField10'.
[config.reconnect]	object		The reconnect configuration object.
[config.reconnect.minDelay]	number	1	Minimum delay for a reconnect attempt in seconds. Minimum value: 1.
[config.reconnect.maxDelay]	number	8	Maximum delay for a reconnect attempt in seconds. Minimum value: 1.
[config.reconnect.maxRetries]	number	10	Maximum amount of successive attempts to reconnect a broken webcast before failure. Setting this to 0 will disable reconnection.
[config.bitrates]	object		The codec configuration object.
[config.bitrates.videoSendInitialBitrate]	number	0	The minimum video upstream bitrate in kbps (lower bound, does not work in Firefox). Allowed values: 0 - 34000 kbps.
[config.bitrates.videoSendBitrate]	number	0	The maximum video upstream bitrate in kbps. Allowed values: 0 - 34000 kbps.
[config.bitrates.audioSendBitrate]	number	0	The maximum audio upstream bitrate in kbps (does not work in Firefox). Allowed values: 0 - 510 kbps.

Example

// rtcUser: instance of RtcUser
var config = {
    logLevel: 2,
    metrics: {
        accountId: 'YOUR_ACCOUNT_ID',
        accountKey: 'YOUR_ACCOUNT_KEY'
    },
    bitrates: {
        videoSendInitialBitrate: 500,
        videoSendBitrate: 1000
    },
    reconnect: {
        minDelay: 2,
        maxDelay: 8,
        maxRetries: 10
    }
};
rtcUser.setConfig(config);

rtcUser.setIceServers(iceServers)

deprecated

This method is deprecated.

Sets an array of turn/stun-servers for the peer-to-peer connection. Parameters in braces "[ ]" are optional.

Kind: instance method of RtcUser

Param	Type	Description
iceServers	Array.<object>	The ice servers object.
iceServers[].urls	Array.<string>	An array of URLs.
[iceServers[].username]	string	The username for the ice servers if required.
[iceServers[].credential]	string	The password for the ice servers if required.

Example

// rtcUser: instance of RtcUser
var iceServers = [
    {
        urls: [
            'turn:turn.nanocosmos.de:80?transport=udp'
        ],
        username: 'username',
        credential: 'password'
    }
];
rtcUser.setIceServers(iceServers);

rtcUser.checkServer()

Checks the state of a Webcaster server.

Kind: instance method of RtcUser
Emits: ReceivedServerStats

Example

// rtcUser: instance of RtcUser
rtcUser.checkServer();
rtcUser.on('ReceivedServerStats', function(event) {
     console.log(event.data);
});

rtcUser.enableStats([enable], [interval])

Enables to receive WebRTC stats in a given time interval. Parameters in braces "[ ]" are optional.

Kind: instance method of RtcUser
Emits: ReceivedWebRTCStats

Param	Type	Default	Description
[enable]	boolean	true	A flag to enable WebRTC stats.
[interval]	number	1000	The interval time in milliseconds. The minimum value: 1 second.

Example

// rtcUser: instance of RtcUser
rtcUser.enableStats();
rtcUser.on('ReceivedWebRTCStats', function(event) {
     console.log(JSON.stringify(event.data.results));
});

rtcUser.startBroadcast(config)

Starts a broadcast to an RTMP ingest URL. Note: by default, only audio will be transcoded on server side, H264 codec will be used for video upstream and passed through directly to RTMP. Therefore you will find the configuration for the video upstream bitrate here: setConfig. Parameters in braces "[ ]" are optional.

Kind: instance method of RtcUser
Emits: StartBroadcastSuccess, StartBroadcastError, BroadcastStatus, BroadcastError

Param	Type	Default	Description
config	object		The config object.
config.transcodingTargets	object		The transcoding config object.
config.transcodingTargets.output	string		The RTMP ingest URL.
config.transcodingTargets.streamname	string	null	The RTMP streamname.
[config.transcodingTargets.audiobitrate]	number	0	The RTMP audio transcoded bitrate in bps, eg 64000.
[config.transcodingTargets.rtmpconnectinfo]	object		Data to be send with the RTMP streams "onconnect". Pass flat object with key value pairs, hierarchies are not supported.
[config.encoding]	object		The encoding config object.
[config.encoding.h264Profile]	'cbp'		The H264 profile. Note: the H264 profile should only be set to 'cbp' in desktop Chrome (Windows).
[config.jwt]	string		A JWT containing ingest information can be passed. This will replace usage of transcodingTargets.streamname & transcodingTargets.output. Note that the JWT can be passed in signIn as authorization method.

Example (Start a webcast with a JWT)

// rtcUser: instance of RtcUser

rtcUser.startBroadcast({
           jwt: 'eyJhbGciOiJSUzI1NiIsInR5cCI6IkpXVCIsImtpZCI6Im5hbm9jb3Ntb3MifQ...' 
});

Example (Start a webcast with rtmp url and rtmp stream name)

// rtcUser: instance of RtcUser

var config = {
    transcodingTargets: {
        output: 'rtmp://bintu-stream.nanocosmos.de:1935/live',
        streamname: 'xAuOp-IaIYi',
        audiobitrate: 64000,
        rtmpconnectinfo: {
            'key1': 'value1',
            'key2': 7.5,
            'key3': false
        }
    }
};

rtcUser.startBroadcast(config);

rtcUser.stopBroadcast()

Stops a running broadcast.

Kind: instance method of RtcUser
Emits: StopBroadcastSuccess, BroadcastError

Example

// rtcUser: instance of RtcUser
rtcUser.stopBroadcast();

rtcUser.sendMetaData(handlerName, jsonValues)

Adds live meta data to a broadcast stream.

Kind: instance method of RtcUser
Emits: RtcUser#event:Error

Param	Type	Description
handlerName	'onMetaData' | 'onCuePoint'	Name of the meta data handler. Other types are not supported.
jsonValues	object	The data to be sent. The parameter can contain a maximum object depth of 6.

Example

// rtcUser: instance of RtcUser
rtcUser.sendMetaData('onMetaData', {myString: 'hello', myInteger: 1234});

rtcUser.addScreenCaptureExtension(name)

deprecated

This method is deprecated.

Adds a Screen Capture Extension to the RtcUser for Chrome. Only needed in Chrome version < 72.

Kind: instance method of RtcUser

Param	Type	Description
name	string	The name of the screen capture extension.

Example

// rtcUser: instance of RtcUser
var name = 'nanoScreenCapture';
rtcUser.addScreenCaptureExtension(name);

rtcUser.isScreenCaptureAvailable()

Checks if a Screen Capture Extension was added via addScreenCaptureExtension().

Kind: instance method of RtcUser

Example

// rtcUser: instance of RtcUser
var name = 'nanoScreenCapture';
rtcUser.addScreenCaptureExtension(name);
// wait until api has registered extension:
setTimeout(function() {
    var hasScreenCapture = rtcUser.isScreenCaptureAvailable();
}, 1000);

rtcUser.getDevices()

Gets all connected input video and audio devices.

Kind: instance method of RtcUser
Emits: ReceivedDeviceList, GetDevicesError

Example

// rtcUser: instance of RtcUser
rtcUser.getDevices();

rtcUser.setVideoDevice(config)

deprecated

This method is deprecated.

Sets the input video device with config. Parameters in braces "[ ]" are optional.

Kind: instance method of RtcUser

Param	Type	Description
config	object	The config object.
config.device	boolean | number	The value of the video device, possible values: true (auto device), false (no video), number (index of the video device).
[config.width]	number	The input width (only if device will be set by index).
[config.height]	number	The input height (only if device will be set by index).
[config.framerate]	number	The input framerate (only if device will be set by index).

Example

// rtcUser: instance of RtcUser
var config = {
    device: 0,
    width: 640,
    height: 360,
    framerate: 30
};
rtcUser.setVideoDevice(config);

Example

// rtcUser: instance of RtcUser
var config = {
    device: 0
};
rtcUser.setVideoDevice(config);

Example

// rtcUser: instance of RtcUser
var config = {
    device: true // auto device
};
rtcUser.setVideoDevice(config);

Example

// rtcUser: instance of RtcUser
var config = {
    device: false // no video
};
rtcUser.setVideoDevice(config);

rtcUser.setAudioDevice(config)

deprecated

This method is deprecated.

Sets the input audio device.

Kind: instance method of RtcUser

Param	Type	Description
config	object	The config object.
config.device	boolean | number	The value of the audio device, possible values: true (auto device), false (no audio), number (index of the audio device).

Example

// rtcUser: instance of RtcUser
var config = {
    device: 0
};
rtcUser.setAudioDevice(config);

Example

// rtcUser: instance of RtcUser
var config = {
    device: 0
};
rtcUser.setAudioDevice(config);

Example

// rtcUser: instance of RtcUser
var config = {
    device: true // auto device
};
rtcUser.setAudioDevice(config);

Example

// rtcUser: instance of RtcUser
var config = {
    device: false // no video
};
rtcUser.setAudioDevice(config);

rtcUser.getSelectedVideoDevice()

deprecated

This method is deprecated.

Gets the current input video device.

Kind: instance method of RtcUser

Example

// rtcUser: instance of RtcUser
var device = rtcUser.getSelectedVideoDevice();

rtcUser.getSelectedAudioDevice()

deprecated

This method is deprecated.

Gets the current input audio device.

Kind: instance method of RtcUser

Example

// rtcUser: instance of RtcUser
var device = rtcUser.getSelectedAudioDevice();

rtcUser.getSelectedDevice(config)

Gets the current input video/audio device.

Kind: instance method of RtcUser
Emits: GetSelectedDeviceError

Param	Type	Description
config	object	The config object.
config.kind	'videoinput' | 'audioinput'	The device kind.

Example

// rtcUser: instance of RtcUser
var config = {
    kind: 'videoinput'
};
var device = rtcUser.getSelectedDevice(config);

rtcUser.startPreview(config)

Starts the preview. Parameters in braces "[ ]" are optional.

Kind: instance method of RtcUser
Emits: StartPreviewSuccess, ReceivedDeviceList, StartPreviewError

Param	Type	Default	Description
config	object		The config object.
config.videoDeviceConfig	object		The video config object.
config.videoDeviceConfig.device	boolean | number		The device id to use. Allowed values: 0...n - specific device, true - auto device, false - no video.
[config.videoDeviceConfig.width]	number		The video width. Allowed values: 320 - 3840(4K).
[config.videoDeviceConfig.height]	number		The video height. Allowed values: 240 - 2160(4K).
[config.videoDeviceConfig.framerate]	number		The video framerate. Allowed values: 0 - 60 fps.
videoDeviceConfig.source	'camera' | 'screen'		The video source to be requested.
config.audioDeviceConfig	object		The audio config object.
config.audioDeviceConfig.device	boolean | number		The device id to use. Allowed values: 0...n - specific device, true - auto device, false - no audio.
[config.audioDeviceConfig.autoGainControl]	boolean		Automatic gain control attempts to automatically maintain a steady overall volume level.
[config.audioDeviceConfig.echoCancellation]	boolean		Echo cancellation attempts to prevent echo effects.
[config.audioDeviceConfig.noiseSuppression]	boolean		Noise suppression automatically filters the audio to remove background noise.
config.elementId	string		The id of a video element to pass in the requested stream directly.
[config.useWebView]	boolean	false	A flag to indicate that a WebView is used.

Example (Audio & video.)

// rtcUser: instance of RtcUser
var config = {
    videoDeviceConfig : {
         device: 0, // video is enabled using specific device
         width: 640,
         height: 360,
         framerate: 30,
         source: 'camera'
    },
    audioDeviceConfig : {
         device: 0 // audio is enabled using specific device
    },
    elementId : 'video-local',
    useWebView : true
};
rtcUser.startPreview(config);

Example (Audio & screen capture.)

// rtcUser: instance of RtcUser
var config = {
    videoDeviceConfig : {
         width: 640,
         height: 360,
         framerate: 30,
         source: 'screen'
    },
    audioDeviceConfig : {
         device: 0 // audio is enabled using specific device
    },
    elementId : 'video-local',
    useWebView : true
};
rtcUser.startPreview(config);

Example (Video only.)

// rtcUser: instance of RtcUser
var config = {
    videoDeviceConfig : {
         device: 0, // video is enabled using specific device
         width: 640,
         height: 360,
         framerate: 30,
         source: 'camera'
    },
    audioDeviceConfig : {
         device: false // no audio; video-only preview
    },
    elementId : 'video-local',
    useWebView : false
};
rtcUser.startPreview(config);

Example (Audio only.)

// rtcUser: instance of RtcUser
var config = {
    videoDeviceConfig : {
         device: false // no video; audio-only preview
    },
    audioDeviceConfig : {
         device: 0 // audio is enabled using specific device
    },
    elementId : 'video-local',
    useWebView : false
};
rtcUser.startPreview(config);

Example (Audio only with all preprocessing disabled.)

// rtcUser: instance of RtcUser
var config = {
    videoDeviceConfig : {
         device: false // no video; audio-only preview
    },
    audioDeviceConfig : {
         device: 0, // audio is enabled using specific device
         autoGainControl: false,
         echoCancellation: false,
         noiseSuppression: false
    },
    elementId : 'video-local',
    useWebView : false
};
rtcUser.startPreview(config);

rtcUser.stopPreview()

Stops the preview.

Kind: instance method of RtcUser
Emits: StopPreviewSuccess, StopPreviewError

Example

// rtcUser: instance of RtcUser
rtcUser.stopPreview();

rtcUser.muteVideo(mute)

deprecated

This method is deprecated.

Mutes/unmutes the video.

Kind: instance method of RtcUser

Param	Type	Description
mute	boolean	Mute/unmute.

Example

// rtcUser: instance of RtcUser
var mute = true;
rtcUser.muteVideo(mute);

rtcUser.muteAudio(mute)

deprecated

This method is deprecated.

Mutes/unmutes the audio.

Kind: instance method of RtcUser

Param	Type	Description
mute	boolean	Mute/unmute.

Example

// rtcUser: instance of RtcUser
var mute = true;
rtcUser.muteAudio(mute);

rtcUser.muteDevice(config)

Mutes/unmutes a video/audio device.

Kind: instance method of RtcUser
Emits: MuteDeviceError

Param	Type	Description
config	object	The config object.
config.kind	'videoinput' | 'audioinput'	The device kind to mute.
config.mute	boolean	A flag to mute/unmute the device.

Example

// rtcUser: instance of RtcUser
var config = {
    kind: 'videoinput',
    mute: true
}
rtcUser.muteDevice(config);

rtcUser.injectExternalMediaStream(config)

Mixes tracks (currently only audio) of an external MediaStream into the currently previewed local stream.

Kind: instance method of RtcUser

Param	Type	Description
config	object	Config object with info on what to mix in.
config.stream	MediaStream	the MediaStream containing the track(s) to mix in.
config.tracks	Array.<string>	Array with the types of tracks which should be injected (only 'audio' is supported at the moment).

Example

// rtcUser: instance of RtcUser
// externalStream instance of MediaStream (https://developer.mozilla.org/de/docs/Web/API/MediaStream)
var data = {stream: externalStream, tracks: ['audio']};
rtcUser.injectExternalMediaStream(data);

rtcUser.setMediaStream(config)

Sets an MediaStream that will be used as source for webcasting instead of an internally created stream (by your live cam or screen).

note

If you simply intent to utilize a combination of camera, microphone or screen share sources: Use the startPreview call.

Context:

It must contain one or two MediaStreamTracks, one VideoStreamTrack and / or one AudioStreamTrack.

The MediaStream can be mixed together on application level by combining different sources:

• audio and video devices
• screen share sources
• rendering on HTML canvas
• WebAudio API

Restrictions:

When utilizing this API call, application developers are responsible for the externally created stream.

• Maintaining the provided framerate is critical for end to end latency of the webcast.
• Tab throttling. When the current tab gets out of focus, most browsers will enable tab throttling. Timers, like setInterval, will be updated less frequently. You should check wether the tab left focus, and warn your users accordingly. E.g. with the DOM visibilitychange event
• Video dimensions (width & height) must not change during a webcast.
• Under certain circumstances the browsers encoder will adapt the video resolution, for example when CPU overusage is being detected.

Kind: instance method of RtcUser

Param	Type	Description
config	object	The config object.
config.stream	number	The MediaStream.
[config.width]	number	The video width. Allowed values: 320 - 3840(4K).
[config.height]	number	The video height. Allowed values: 240 - 2160(4K).
[config.framerate]	number	The target video framerate. Allowed values: 0 - 60 fps.

Example

// rtcUser: instance of RtcUser
// pass an MediaStream to the API

var canvasWidth = 1280;
var canvasHeight = 720;
var framerate = 30;

// access your HTML canvas (<canvas id="my-canvas"/> element)
var canvas = document.getElementById('my-canvas');
var context = canvas.getContext('2d');

// create a MediaStream from the canvas
var canvasStream = canvas.captureStream(framerate);

// optionally, add an AudioStreamTrack
var myAudioTrack = ...;
canvasStream.addTrack(myAudioTrack);

// your render loop, the loop calls itself 30 (framerate) times per second
// as an example, we simply draw the string "Hello World"
function renderLoop() {
    context.font = '50px serif';
    context.fillText('Hello world', 50, 90);
    setTimeout(loop, 1000 / framerate); // drawing at 30fps
 };

var config = {
    stream: canvasStream,
    width: canvasWidth,
    height: canvasHeight,
    framerate: framerate
};

// start the render loop
renderLoop();

rtcUser.setMediaStream(config);

---

Events

"SignInSuccess"

SignInSuccess event. The event is fired if sign in succeeded.

Kind: event emitted by RtcUser
Typeof: SignInSuccessEvent

"SignInError"

SignInError event. The event is fired if sign in failed.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"SignOutSuccess"

SignOutSuccess event. The event is fired if sign out succeeded. Session is now destroyed.

Kind: event emitted by RtcUser
Typeof: SuccessEvent

"SignOutError"

SignOutError event. The event is fired if an error occured. Session is destroyed in any case.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"SetConfigError"

SetConfigError event. The event is fired if config failed to be set.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"ReceivedServerStats"

ReceivedServerStats event. The event is fired if server stats is received.

Kind: event emitted by RtcUser
Typeof: ServerStatsEvent

"ReceivedWebRTCStats"

ReceivedWebRTCStats event. The event is fired multiple times between the "start" and "stop" broadcast events. The payload contains WebRTC stats information.

Kind: event emitted by RtcUser
Typeof: WebRTCStatsEvent

"StartBroadcastSuccess"

StartBroadcastSuccess event. The event is fired if a broadcast started successfully.

Kind: event emitted by RtcUser
Typeof: SuccessEvent

"StartBroadcastError"

StartBroadcastError event. The event is fired if a broadcast failed to start.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"BroadcastStatus"

BroadcastStatus event. The event is fired multiple times between the "start" and "stop" broadcast events. The payload contains information about the connection state.

Kind: event emitted by RtcUser
Typeof: StatusEvent

"BroadcastError"

BroadcastError event. The event is fired if a broadcasting error occurred between the "start" and "stop" broadcast events.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"StopBroadcastSuccess"

StopBroadcastSuccess event. The event is fired if a broadcast stopped successfully. It does not contain an event payload.

Kind: event emitted by RtcUser
Typeof: SuccessEvent

"StopBroadcastError"

StopBroadcastError event. The event is fired if a broadcast failed to execute the "stop" call as it is expected. Note: the broadcast is guaranteed to stop before the event is fired.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"ReceivedDeviceList"

ReceivedDeviceList event. The event is fired if a list of connected input video and audio devices is received.

Kind: event emitted by RtcUser
Typeof: DevicesReceivedEvent

"GetDevicesError"

GetDevicesError event. The event is fired if an error occured while getting connected input video and audio devices.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"GetSelectedDeviceError"

GetSelectedDeviceError event. The event is fired if a selected video/audio device getting failed.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"StartPreviewSuccess"

StartPreviewSuccess event. The event is fired if a preview started successfully.

Kind: event emitted by RtcUser
Typeof: StartPreviewSuccessEvent

"StartPreviewError"

StartPreviewError event. The event is fired if a preview failed to start.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"StopPreviewSuccess"

StopPreviewSuccess event. The event is fired if a preview stopped successfully.

Kind: event emitted by RtcUser
Typeof: SuccessEvent

"StopPreviewError"

StopPreviewError event. The event is fired if a preview failed to stop. Note: the preview is guaranteed to stop before the event is fired.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

"MuteDeviceError"

MuteDeviceError event. The event is fired if a video/audio device mute/unmute failed.

Kind: event emitted by RtcUser
Typeof: ErrorEvent

---

Type Definitions

ErrorEvent : object

Kind: global typedef
Properties

Name	Type	Description
data	object	The data object - Deprecated: will be removed in nanoStream Webcaster v.6; please use the "error" object instead.
error	object	The error object.
error.code	ErrorCode	The code of the error.
error.message	string	Human readable message of the error.
error.name	string	The name of the error.
error.text	string	The text of the error - Deprecated: will be removed in nanoStream Webcaster v.6; please use the "error.message" property instead.
error.type	string	The type of the error.
error.userinfo	object	The user info associated with the error.
[error.userinfo.metadata]	object	Metadata with additional state information.
[error.userinfo.metadata.sessionId]	object	Session id (defined only if signed in).
[error.userinfo.metadata.isSignedIn]	object	Wether a session to the server was established successfully.
[error.userinfo.metadata.server]	object	Server url (defined only if a connection to the server was made).
[error.userinfo.metadata.streamname]	object	RTMP stream name (defined only if a broadcast was started).
[error.userinfo.metadata.hasAudio]	object	Wether the current stream has an audio track.
[error.userinfo.metadata.hasVideo]	object	Wether the current stream has an video track.
[error.userinfo.metadata.isMobile]	object	Wether the current device is a mobile device.
[error.userinfo.metadata.browser]	object	Currently used browser.
[error.userinfo.metadata.browserVersion]	object	Currently used browser version.
[error.userinfo.metadata.os]	object	Current operating system.
name	string	The name of the event.

SuccessEvent : object

Kind: global typedef
Properties

Name	Type	Description
[data]	object	The data object. Note: the property is optional, therefore, its presence is not guaranteed.
[data.output]	string	The RTMP ingest URL. Note: the property is optional, therefore, its presence is not guaranteed.
[data.streamname]	string	The RTMP ingest streamname. Note: the property is optional, therefore, its presence is not guaranteed.
[data.stream]	object	The stream object. Note: the property is optional, therefore, its presence is not guaranteed.

StatusEvent : object

Kind: global typedef
Properties

Name	Type	Description
data	object	The data object.
data.message	'signalling' | 'connected' | 'reconnecting' | 'broadcasting' | 'reconnecting_broadcast'	The connection state. Note that 'reconnecting' implies an RTMP reconnect on server side and 'reconnecting_broadcast' means a client side reconnect.
[data.number]	object	Reserverd for internal developer's use. Note: the property is optional, therefore, its presence is not guaranteed - Deprecated: will be removed in nanoStream Webcaster v.6.
[data.rtmp]	object	Reserved for internal developer's use. Note: the property is optional, therefore, its presence is not guaranteed - Deprecated: will be removed in nanoStream Webcaster v.6.
data.state	4 | 5 | 6 | 7	The connection state. Possible values: 4 = 'signalling', 5 = 'connected', 6 = 'broadcasting', 7 = 'reconnecting'.
data.streamname	object	The RTMP ingest streamname.
[name]	object	The name of the event. Note: the property is optional, therefore, its presence is not guaranteed.
[target]	object	Reserved for internal developer's use. Note: the property is optional, therefore, its presence is not guaranteed - Deprecated: will be removed in nanoStream Webcaster v.6.

DevicesReceivedEvent : object

Kind: global typedef
Properties

Name	Type	Description
data	object	The data object.
data.devices	object	The audio and video devices object.
[data.devices.audiodevices]	Array.<object>	An array of connected input audio devices. Note: the property is optional, therefore, its presence is not guaranteed.
data.devices.audiodevices.id	string	The id of an audio device.
data.devices.audiodevices.index	number	The index of an audio device. Note: indices start from 0.
[data.devices.videodevices]	Array.<object>	An array of connected input video devices. Note: the property is optional, therefore, its presence is not guaranteed.
data.devices.videodevices.id	string	The id of a video device.
data.devices.videodevices.index	number	The index of a video device. Note: indices start from 0.
data.updated	boolean	Reserved for internal developer's use - Deprecated: will be removed in nanoStream Webcaster v.6.

StartPreviewSuccessEvent : object

Kind: global typedef
Properties

Name	Type	Description
data	object	The data object.
data.constraints	object	The constraints object.
[data.constraints.audio]	object	The audio constraints object. Note: the property is optional, therefore, its presence is not guaranteed.
data.constraints.audio.deviceId	object	The id of an audio device.
[data.constraints.video]	object	The video constraints object. Note: the property is optional, therefore, its presence is not guaranteed.
data.constraints.video.deviceId	object	The id of a video device.
[data.constraints.video.framerate]	object	The video framerate. Note: the property is optional, therefore, its presence is not guaranteed.
[data.constraints.video.height]	object	The video height. Note: the property is optional, therefore, its presence is not guaranteed.
data.constraints.video.source	string	The video source.
[data.constraints.video.width]	object	The video width. Note: the property is optional, therefore, its presence is not guaranteed.
data.metadata	object	The metadata object.
data.metadata.framerate	number	The video framerate set in metdata.
data.metadata.hasAudio	number	A flag to indicate that audio is present.
data.metadata.hasVideo	number	A flag to indicate that video is present.
data.metadata.height	number	The video height set in metadata.
data.metadata.width	number	The video width set in metadata.
data.stream	object	The stream object.
data.stream.active	boolean	A flag to indicate that a stream is active.
data.stream.id	string	The stream id.

SignInSuccessEvent : object

Kind: global typedef
Properties

Name	Type	Description
data	object	The data object.
data.server	string	The URL of the Webcaster server
data.userId	string	The user id of the Webcaster server.

ServerStatsEvent : object

Kind: global typedef
Properties

Name	Type	Description
data	object	The data object.
data.stats	object	The stats object.
data.stats.active_connections	number	The total number of active connections to the Webcaster server.
data.stats.active_passthroughs	number	The number of active connections with passthrough enabled to the Webcaster server.
data.stats.active_transcoders	number	The number of active connections with transcoding enabled to the Webcaster server.
data.stats.cpu_load	number	Server CPU load.
data.stats.disk_space	number	Server disk space.
data.stats.domainname	string	Server domain name.
data.stats.hostname	string	Server host name.
data.stats.license	object	Server license.
data.stats.license.end	string	The license expiration date.
data.stats.license.expired	boolean	A flag to indicate that the license has expired.
data.stats.license.expiring	boolean	A flag to indicate if the license has an end date or not.
data.stats.license.start	string	The license issue date.
data.stats.license.valid	boolean	A flag to indicate that the license is valid.
data.stats.max_cpu_load	number	Server maximum CPU load.
data.stats.max_transcoders	number	Server maximum number of transcoders.
data.stats.server_version	string	The version of the Webcaster server.
data.stats.transcoder_version	string	The version of the transcoder.
name	string	The name of the event.
target	string	Reserved for internal developer's use - Deprecated: will be removed in nanoStream Webcaster v.6.

WebRTCStatsEvent : object

Kind: global typedef
Properties

Name	Type	Description
data	object	The data object.
data.message	string	The connection state message (expected value: 'broadcasting').
data.stats.results	object	The results object. It contains the stats key/value pairs. All stats in the result object are optional and are present based on the capabilities of the currently used browser.
[data.stats.results.audioBitrate]	number	The audio bitrate in kbps.
[data.stats.results.audioBitrateMin]	number	Minimum 'audioBitrate' during past 10 seconds.
[data.stats.results.audioBitrateMax]	number	Maximum 'audioBitrate' during past 10 seconds.
[data.stats.results.audioBitrateAverage]	number	Average 'audioBitrate' during past 10 seconds.
[data.stats.results.audioCodec]	string	The audio codec.
[data.stats.results.frameHeight]	number	The video frame height in pixels.
[data.stats.results.frameWidth]	number	The video frame width in pixels.
[data.stats.results.framerate]	number	The video frame rate in frames per second.
[data.stats.results.framerateMin]	number	Minimum 'framerate' during past 10 seconds.
[data.stats.results.framerateMax]	number	Maximum 'framerate' during past 10 seconds.
[data.stats.results.framerateAverage]	number	Average 'framerate' during past 10 seconds.
[data.stats.results.videoBitrate]	number	The video bitrate in kbps.
[data.stats.results.videoBitrateMin]	number	Minimum 'videoBitrate' during past 10 seconds.
[data.stats.results.videoBitrateMax]	number	Maximum 'videoBitrate' during past 10 seconds.
[data.stats.results.videoBitrateAverage]	number	Average 'videoBitrate' during past 10 seconds.
[data.stats.results.videoCodec]	string	The video codec.
[data.stats.results.videoSendDelay]	number	An indication for video packet delay, before packets are send to the server.
[data.stats.results.videoSendDelayMax]	number	Maximum 'videoSendDelay' during past 10 seconds.
[data.stats.results.videoSendDelayAverage]	number	Average 'videoSendDelay' during past 10 seconds.
[data.stats.results.packetLoss]	number	Percentage of packets lost during past 10 seconds.
[data.stats.results.roundTripTime]	number	Current time in milliseconds that data takes from the client to the webcaster server and back.
data.stats.state	number	The connection state. The expected value: 6 = 'broadcasting'.
data.stats.streamname	string	The RTMP ingest streamname.
data.stats.text	string	The connection state text. The expected value: 'broadcasting'.
name	string	The name of the event.
target	string	Reserved for internal developer's use - Deprecated: will be removed in nanoStream Webcaster v.6.

ErrorCode : number

The possible client error codes in an ErrorEvent event.

Kind: global typedef
See: ErrorEvent
Properties

Name	Type	Description
1000-1999	GeneralError
1000		A general error returned by any nanoStream Webcaster component (see the error message for more informaiton).
1001		A type is not valid.
1002		A value is not valid.
1003		An enum value is not valid.
1004		An object does not exist.
1005		A function is not defined.
1006		An operation is not allowed to be executed.
2000-2999	RequestError
2000		A general request error - Deprecated: will be removed in nanoStream Webcaster v.6.
3000-3999	WebSocketError
3000		A general WebSocket error (see the error message for more information).
3001		A WebSocket state is wrong (see an error message for more information).
3002		An error occurred when creating a WebSocket.
4000-4999	SessionError
4000		Sign in failed.
4001		A session is not signed in.
5000-5999	BroadcastError
5001		A broadcast failed to start.
5002		A Broadcast failed after it was started (see the error message for more information).
5003		Stopping broadcast caused an error. Either no broadcast is runnung or an active broadcast is currently shutting down.
6000-6999	WebRTCError
6000		A media stream was received, but no video/audio data is coming from a device. The device could be in use from another application.
6001		A media stream was received, but it contains no tracks.
6002		A media stream was not found.
6003		A media stream was not removed.
6004		No devices are available for "getUserMedia".
6005		A general "getUserMedia" error (may include: 'PermissionDenied', 'NotAllowedError', 'NotReadableError', 'NotFoundError', 'OverconstrainedError', 'AbortError', 'GetMediaError', 'NoDeviceError', 'NoStreamError').
6006		Devices failed to get enumerated.
6008		A general "getDisplayMedia" error (may include: 'PermissionDenied', 'NotAllowedError').
101000-109999	ServerError	Server-side errors start with 10xxxx.
101000-101999	GeneralError	Server-side general errors.
101001		An internal server error.
101002		A parameter is required.
101003		A data type is wrong.
102000-102999	AccessError	Server-side access errors.
102001		Access is denied.
103000-103999	AuthenticationError	Server-side authentication errors.
103000		The server could not initialize authentication.
103001		Wrong authentification data. A valid token, or username and password is/are required.
103002		Access to a feature is forbidden (features: 'webrtc').
103003		JWT authentication failed.
103004		JWT contains an invalid stream.
104000-104999	BintuError	Server-side bintu related errors.
104001		A stream option is not allowed (options: 'webrtc', 'transcoding').
104002		A bintu API key is invalid.
105000-105999	TranscoderError	Server-side configuration errors.
106000-106999	SystemMonitorError	Server-side monitoring errors.
106001		An error occured while retrieving system info.
107000-107999	SessionError
107001		A configuration is malformed.
107002		A session is invalid.
108000-108999	ChatError	Deprecated.
109000-109999	RTMPError
109000		The RTMP connection has failed.

---