Developers Guide

Let's start building video communication stuff!

UserAgent

The UserAgent is an entry point to ApiRTC CPaaS. This is the first object to instantiate when implementing a front-end application. It represents the local user that will participate in the conversation.

The UserAgent can be either anonymous or identified.

Identification is done through a JWT retrieved from an authentication service.

• Read the Authentication page for more details on how to authenticate.

• Read the UserAgent reference page.

MediaDevices

UserAgent's mediaDeviceChanged event can be listened to in order to get notified of the list of devices available to the browser:

Here is what the mediaDevices object looks like:

userAgent.on("mediaDeviceChanged", () => {
  const mediaDevices = this.userAgent.getUserMediaDevices();
  // handle new set of mediaDevices
});

This is useful to propose a list of available media devices to the user.

Session

A Session instance represents a connection to the ApiRTC CPaaS. A Session is configured by an API key and a declared UserAgent.

A session handles all the interactions of participants, including video/audio streams and data exchanges for one Enterprise identified by its API key.

A Session object is to get through the UserAgent.register(options) method. Some options (of type RegisterInformation) controls the authentication mechanisms.

• Read the Authentication page for more details on how to authenticate.

• See the Session reference page.

Contacts

Contacts are the participants to a Session. They can be authenticated or anonymous.

session.getContacts().foreach({username, contactObject} => {
    console.log(contactObject.getId());
});

• See the Contact reference page

Data exchange

Data can be exchanged across Contacts by using the Contact.sendData(object) method:

contact.sendData({aProperty:'aValue'})
  .then(() => {
    console.log("message sent")
  }).catch((error: any) => {
    console.error("sendData", error)
  })

To receive the data, listen on the Session's contactData event:

session.on('contactData', contactDataEvent => {
  console.log('received data from sender', contactDataEvent.sender, contactDataEvent.content)
})

Presence group

Each Contact getting into a Session can join presence groups and segment all the connected users into subcategories.

For example: an employee can get into a Session, and join the "Operator" and "Available" groups, while a customer will join the "Customer" group.

To make a user connect within some group, set the RegisterInformation.groups in the UserAgent.register(options) options, or user the Session.joinGroupmethod.

Joining a group as a participant will activate session's contactListUpdate event listening on this group. Alternatively, you can subscribe to the group's events without joining it with the Session.subscribeToGroup method.

If the current participant doesn't subscribe to or join a group, they will not receive event regarding group changes.

The data object associated to Session.contactListUpdate event has joinedGroup and leftGroup properties to carry information on which Contact joined of left which group:

session.on('contactListUpdate', (updatedContacts: any) => {
  for (const group of Object.keys(updatedContacts.joinedGroup)) {
    for (const contact of updatedContacts.joinedGroup[group]) {
      // ...
    }
  }
  for (const group of Object.keys(updatedContacts.leftGroup)) {
    for (const contact of updatedContacts.leftGroup[group]) {
      // ...
    }
  }
})

UserData

UserData is a class that holds a data object to store values associated to a UserAgent. Make sure to call UserData.setProp(key, value) to set up a property.

Once connected to Session, call userData.setToSession() to make other connected peers notified of UserData properties change through the Session.contactListUpdate event.

For that purpose, the data object associated to Session.contactListUpdate event has a userDataChanged property which is an array of Contacts for which UserData has changed.

session.on('contactListUpdate', updatedContacts => {
  for (const contact of updatedContacts.userDataChanged) {
    // ...
  }
}

Conversation

A conversation is the way to gather participants to exchange medias. It can be text message, audio/video streams, files...

Whenever there are 2 participants or more, a conversation takes place.

getOrCreateConversation

To get a Conversation instance, the Session's method getOrCreateConversation(name, options) should be used.

The name is of string type without any constraint.

conversation = session.getOrCreateConversation(name, {
  meshModeEnabled: false,
  meshOnlyEnabled: false,
  moderationEnabled: false,
  moderator: false
});

Options

key	description
meshModeEnabled	enables the mesh mode (otherwise SFU star topology is used by default). (default: false)
meshOnlyEnabled	forces the mesh mode for the whole Conversation.
moderationEnabled	enables moderation on the Conversation. This option may change the behavior of the joining process depending on the moderator option value. (default: false)
moderator	adds the local user to the list of moderators for the conversation. (default: false)

Every participant must enable the moderationEnabled option to have consistent moderation apply throughout the Conversation.

Precision on the Mesh Mode

The mesh mode enables a peer-to-peer connection across participants, without going through a stream routing server (called SFU for Selective Forwarding Unit).

Mesh mode multiplies the stream sent by each participant. As upload bandwidth is often lower than download bandwith, network connection can become shaky as the number of participants grows.

If meshModeEnabled is true when setting the conversation mode, the stream exchanges will remain in P2P until:

• the number of participant goes over 4,

• or too many packet loss is detected for one participant.

Then the conversation will automatically switch to star topology mode using the ApiRTC SFUs infrastructure.

Setting both meshModeEnabled and meshOnlyEnabled to true forces the conversation to remain mesh only, whatever the connection's events.

Join Conversation

Conversation.join() makes the local user enter the conversation. Note that this method returns a Promise and one must wait for it to be fulfilled before doing anything else on the conversation.

A good practice is to register all required Conversation event listeners before calling the join method:

conversation.on('streamListChanged', streamInfo => {
  // Handle the 'streamListChanged' event...
});
// and any other relevant events
// ...

conversation.join()
  .then(() => {
    // local user successfully joined the conversation.
  }).catch(error => {
    // error
  });

Leave Conversation

Conversation.leave() makes the local user leave the conversation. All ongoing streams will be automatically closed.

A good practice is to destroy the Conversation after leaving it. (Except if you want to be able to join it again afterward)

conversation.leave()
  .then(() => {
    conversation.destroy();
  });

Conversation Moderation

Moderation allows a group of users (moderators) to control the conversation's access to other participants.

Activation

In order to activate the moderation for a conversation, every party (moderator or not) must explicitly set the moderationEnabled option to true when calling getOrCreateConversation.

Additionally, the moderator participant set the moderator option to true as well when calling getOrCreateConversation.

Joining process

If the local user is moderator, then the join() will resolve immediately.

But if the local user is not moderator, then the join() will only resolve when a moderator allows it. In the meanwhile, the user will be put in a waiting room.

Waiting Room

The waiting room is a presence group associated to the conversation. It allows to identify participants who are currently waiting for a moderation answer.

Events contactJoinedWaitingRoom and contactLeftWaitingRoom will be triggered respectively upon the arrival and departure of a user to/from the waiting room:

onversation.on('contactJoinedWaitingRoom', contact => {
  // A candidate joined the waiting room.
  // Store it into a list and display it in the DOM
  // ...
});

conversation.on('contactLeftWaitingRoom', contact => {
  // A candidate left the waiting room.
  // Remove from list
  // ...
});

Then the moderator can allow or deny a contact to enter the conversation:

// Grant...
conversation.allowEntry(contact);
// ... or deny access.
conversation.denyEntry(contact);

Eject

Moderators have the ability to eject another participant from the conversation.

conversation.eject(contact, { reason: 'a reason' })
  .then(() => {
    console.log('ejected', contact);
  }).catch((error) => {
    console.error('eject error', error);
  });

To get notified of a participant ejection, listen on the participantEjected event. The event data object wears a self boolean set to true if the current local user is the ejected participant.

conversation.on('participantEjected', data => {
  console.log('on:participantEjected', data);
  if (data.self) {
    // local user was ejected,
    // unpublish streams,
    // and destroy the conversation
  }
});

Record the conversation

ApiRTC platform allows to record a composite video of a conversation. The video will be composed of all streams exchanged in the conversation and will be stored in ApiRTC's database.

To start recording a conversation:

conversation.startRecording().then(recordingInfo => {
  console.info('startRecording', recordingInfo);
}).catch((error: any) => {
  console.error('startRecording', error);
});

Refer to Conversation.startRecording(options) for details on what are the options.

Example of recordingInfo (RecordingInfo) data:

{
    "roomName": "Test",
    "callId": "COMPOSITE",
    "recordType": "composite",
    "convId": "2b0839f5-aa1e-4cb2-ba9a-46848a6b",
    "mediaId": "1261785",
    "mediaURL": "https://dashboard.apirtc.com/media/showVideo/<id>/hash/2c625610-4baa-11ec-a192-538513dee1ef",
    "recordedFileName": "vfrP9vWu-3467-composite.mp4",
    "audioOnly": false,
    "videoOnly": false,
    "mode": "complete",
    "labelEnabled": false
  }

To stop recording a conversation:

conversation.stopRecording().then(recordingInfo => {
  console.info('stopRecording', recordingInfo);
}).catch((error: any) => {
  console.error('stopRecording', error);
});

Once the recording is stopped, the ApiRTC platform will process it and make it available for download. To get notified when a record is available, listen to the recordAvailable event of the Converation's instance:

conversation.on('recordingAvailable', recordingInfo => {
  console.log("on:recordingAvailable", recordingInfo);
  ...
});

When the video is available, you can use the RecordingInfo.mediaURL to download it.

Speaker detection

To display which participant is currently talking in a Conversation, enable the feature:

userAgent.enableActiveSpeakerDetecting(true, { threshold: 50 });

Then, listen on the audioAmplitude event:

conversation.on('audioAmplitude', amplitudeInfo => {
  // handle amplitudeInfo
})

the event data object (amplitudeInfo) holds the following information:

{
  "streamId": "6725958108801516",
  "amplitude": 102.36,
  "isSpeaking": true
}

When the participant speaks and amplitude goes over the threshold configured during feature enabling, event with isSpeaking set to true is fired.

When the participant does not speak anymore, the event is fired again with initial amplitude value that triggered the event, but this time isSpeaking is false.

QoS statistics

Conversation event callStatsUpdate provides statistics information on media stream exchanges quality of service.

conversation.on('callStatsUpdate', callStats => {
  // handle callStats.stats data
});

Depending on whether the stream is sent or received, the event data object (callStats) holds the following information:

For a local stream, qos info on sent media

    "streamId": "7167592935479248",
    "stats": {
        "audioSent": {
            "bitsSentPerSecond": 22044,
            "bytesSent": 54006,
            "delay": 0,
            "kind": "audio",
            "mediaType": "audio",
            "nackCount": 0,
            "packetLossRatio": 0,
            "packetsSent": 982,
            "packetsSentPerSecond": 50,
            "remoteId": "9385e3a0",
            "samplingInterval": 10,
            "timestamp": 1633005140,
            "type": "outbound-rtp"
        },
        "videoSent": {
            "bitrateMean": 490785.10526315786,
            "bitrateStdDev": 54128.26265341604,
            "bitsSentPerSecond": 517595,
            "bytesSent": 1245545,
            "delay": 0,
            "droppedFrames": 2,
            "firCount": 0,
            "framerateMean": 30.315789473684212,
            "framerateStdDev": 0.749268649265355,
            "framesEncoded": 562,
            "framesEncodedPerSecond": 30,
            "height": 480,
            "kind": "video",
            "mediaType": "video",
            "moyDelay": null,
            "nackCount": 2,
            "packetLossRatio": 0,
            "packetsSent": 1232,
            "packetsSentPerSecond": 63,
            "pliCount": 4,
            "qpSum": 20935,
            "remoteId": "526431ec",
            "samplingInterval": 10,
            "timestamp": 1633005140,
            "type": "outbound-rtp",
            "width": 640
        },
        "quality": {
            "mosS": "NoStream",
            "mosSAV": 3.087473118525441,
            "mosSS": 4.409150284259602,
            "mosSV": 3.4956463628881274,
            "mosV": "NoStream"
        }
    }
}

For a remote stream, qos info on received media

{
    "streamId": "362307064506733",
    "stats": {
        "audioReceived": {
            "bitsReceivedPerSecond": 22044,
            "bytesReceived": 109505,
            "delay": 0,
            "jitter": 0.002,
            "kind": "audio",
            "mediaType": "audio",
            "nackCount": 0,
            "packetLossRatio": 0,
            "packetsLost": 1,
            "packetsLostPerSecond": 0,
            "packetsReceived": 1991,
            "packetsReceivedPerSecond": 50,
            "remoteId": "f14eaf8",
            "samplingInterval": 20,
            "timestamp": 1633005174,
            "type": "inbound-rtp"
        },
        "videoReceived": {
            "bitrateMean": 773107.5384615384,
            "bitrateStdDev": 176425.8098193486,
            "bitsReceivedPerSecond": 910830,
            "bytesReceived": 3874287,
            "delay": 0,
            "discardedPackets": 0,
            "firCount": 0,
            "framerateMean": 30.076923076923073,
            "framerateStdDev": 0.4220635637221745,
            "framesDecoded": 1167,
            "framesDecodedPerSecond": 30,
            "height": 480,
            "jitter": 0.009,
            "kind": "video",
            "mediaType": "video",
            "nackCount": 13,
            "packetLossRatio": 0.09086778736937756,
            "packetsLost": 3,
            "packetsLostPerSecond": 0,
            "packetsReceived": 3790,
            "packetsReceivedPerSecond": 110,
            "pliCount": 1,
            "remoteId": "8a312f14",
            "samplingInterval": 20,
            "timestamp": 1633005174,
            "type": "inbound-rtp",
            "width": 640
        },
        "quality": {
            "mosAV": 3.4344075003680574,
            "mosS": 4.409150284259602,
            "mosSS": "NoStream",
            "mosSV": "NoStream",
            "mosV": 3.860635735132783
        }
    }
}

The callStats.streamId is useful to associate data to corresponding streamreams

• See the Stream reference page for more information.

Stream

Local Streams

Camera

Acquiring camera local stream is done through the UserAgent.createStream(options) method. The browser asks the user to choose among available media devices. The Promise resolves with a Stream instance.

userAgent.createStream({
  constraints: {
    audio: true,
    video: true
  }
}).then(localStream => {
  // ...
}).catch(error => {
  // error
});

All possible options for the CreateStream method can be found in the CreateStreamOptions reference page.

constraints option is of type MediaStreamConstraints. See the Stream Constraint section for more infos.

Screen sharing

Acquiring screen sharing local stream is done through a Stream static method:

// Returns a Promise.<Stream> containing the stream
Stream.createScreensharingStream().then(localStream => {
// ...
}).catch(error => {
  // error
});

Publish/Unpublish

Publishing a local stream makes it available for remote peer participants to subscribe and view it.

The local user (UserAgent) must have joined the conversation before publishing a stream.

conversation.publish(localStream).then(publishedStream => {
  // local stream is published
}).catch(error => {
  // error
});

Conversation.publish(localStream, options) can optionally take PublishOptions second parameter object to control publication. Please check reference for details on PublishOptions.

Unpublishing a local stream makes it unavailable for remote peer participants to subscribe and stops sending media stream to peers.

conversation.unpublish(localStream);

Remote Streams

Handle remote streams availability

ApiRTC triggers an event when stream availability changes through the Conversation.streamListChanged event.

This event is triggered:

• once for each existing stream when the participant joins the Conversation

• every time a new stream is published to or unpublished from the Conversation

The data object carried by Conversation.streamListChanged event is StreamInfo: this is not a Stream yet.

conversation.on('streamListChanged', streamInfo => {
  const streamId = String(streamInfo.streamId);
  const contactId = String(streamInfo.contact.getId());
  if (streamInfo.isRemote === true) {
    if (streamInfo.listEventType === 'added') {
      // a remote stream was published
      ...
    } else if (streamInfo.listEventType === 'removed') {
      // a remote stream is not published anymore
      ...
    }
  }
});

The streamInfo.contact.getId() and streamInfo.streamId can be useful to identify which remote peer published their stream.

Subscribe to a remote stream

A remote stream is subscribed to using the Conversation.subscribeToStream(streamId) method. It takes the id of stream provided in the StreamInfo data object:

conversation.subscribeToStream(streamInfo.streamId);

Be mindful that whithout subscribing to stream's event, you will not be notified of streams updates and termination.

Conversation.subscribeToStream(streamId, options) can take optionally take SubscribeOptions as a second parameter to control subscription.

Unsubscribe from a remote stream

Unsubscribing to a remote stream is done by the Conversation.unsubscribeToStream(streamId) method.

conversation.unsubscribeToStream(streamId);

Manage media streams

Once a stream has been subscribed, ApiRTC notifies with an actual Stream instance through the streamAdded event.

This event is triggered every time the actual media stream is available to be displayed.

conversation.on('streamAdded', remoteStream => {
  // display media stream
  ...
});

Whenever a media stream becomes unavailable, ApiRTC notifies the conversation with a streamRemoved event.

conversation.on('streamRemoved', remoteStream => {
  // undisplay media stream
  ...
});

A media stream may encounter technical issues, or meet network optimization requiring to change the actual Stream instance. In this case streamRemoved event will be also fired, prior to another streamAdded event with the new instance.

Stream display

In order to display or remove media element in DOM, you can use our helpers:

• Stream.addInDiv() and Stream.removeFromDiv() to add/remove a <video> element within an existing <div>

• Stream.attachToElement(domElement) to directly attach to a <video> element.

We manage some devices specificities in our helpers that can help avoid media plays issue. (for instance with Safari iOS).

// display media stream by attaching to a media element (like <video>)
stream.attachToElement(videoDomElement)
// or insert into a container div
stream.addInDiv('container-id', 'media-element-' + stream.streamId, {}, false)

Audio/Video Mute

To control local or remote stream audio/video mute, use the following Stream methods:

Starting from apiRTC 5.0.1

Evolution has been done on apiRTC 5.0.1 version to reflect standard.

Mute state is managed by enabled/disabled attribute at application level.

// toggle audio
if (stream.isAudioEnabled()) {
  stream.disableAudio();
} else {
  stream.enableAudio();
}

// toggle video
if (stream.isVideoEnabled()) {
  stream.disableVideo();
} else {
  stream.enableVideo();
}

Older versions

// toggle audio
if (stream.isAudioMuted()) {
  stream.unmuteAudio();
} else {
  stream.muteAudio();
}

// toggle video
if (stream.isVideoMuted()) {
  stream.unmuteVideo();
} else {
  stream.muteVideo();
}

Stream constraints

Constraints are camera properties that can be set: resolution, brightness, contrast, frameRate, saturation, torch, zoom.

Capabilities are supported properties and value ranges. Settings are the current properties values.

ApiRTC allows to access constraints, capabilities, settings on both local and remote streams, using the same methods. This means you can easily control both local and remote devices.

Stream.applyConstraints(constraints) method returns a Promise resolved when all constraints are applied:

stream.applyConstraints({
  "audio": {},
  "video": {
    "frameRate": 10
  }
}).then(() => {
  ... // constraints applied
});

Note that the constraints is of type MediaStreamConstraints.

Stream.getConstraints() returns a Promise with all properties that were modified and their current values:

// get stream constraints that were applied and their values
stream.getConstraints()
  .then(constraints => {
    console.log(constraints) // constraints object
  }).catch((error) => {
    ... // error during process
  });

Constraints values depend on the device capabilities. For example, on smartphones with multiple back cameras, sometimes the torch property is only attached to one of the camera.

In addition, supported properties can be queried using Stream.getCapabilities() that returns a Promise with accepted values ranges:

// get stream capabilities values ranges
stream.getCapabilities()
  .then(capabilities => {
    console.log(capabilities) // capabilities object
  }).catch((error) => {
    ... // error during process
  });

Example of a capabilities data object:

{
  "audio": {
    "autoGainControl": [ true, false ],
    "channelCount": { "max": 1, "min": 1 },
    "deviceId": "...",
    "echoCancellation": [ true, false ],
    "groupId": "...",
    "latency": { "max": 0.085333, "min": 0.002666 },
    "noiseSuppression": [ true, false ],
    "sampleRate": { "max": 48000, "min": 48000 },
    "sampleSize": { "max": 16, "min": 16 }
  },
  "video": {
    "aspectRatio": { "max": 1920, "min": 0.001388888888888889 },
    "deviceId": "...",
    "frameRate": { "max": 30, "min": 0 },
    "groupId": "...",
    "height": { "max": 1080, "min": 1 },
    "resizeMode": ["none", "crop-and-scale"],
    "width": { "max": 1920, "min": 1 }
  }
}

In this example video.frameRate property may be set between 0 and 30.

getCapabilities() may not work with all browsers. Also, returned capabilities may differ from a device to another.

Finally, properties values can be checked with Stream.getSettings() that returns in a Promise all current settings:

// get stream actual constraints settings
stream.getSettings()
  .then(settings => {
    console.log(settings) // settings object
  }).catch((error) => {
    ... // error during process
  });

Example of a settings data object:

{
  "audio": {
    "autoGainControl": true,
    "channelCount": 1,
    "deviceId": "...",
    "echoCancellation": true,
    "groupId": "...",
    "latency": 0.01,
    "noiseSuppression": true,
    "sampleRate": 48000,
    "sampleSize": 16
  },
  "video": {
    "aspectRatio": 1.333333333333,
    "deviceId": "...",
    "frameRate": 30,
    "groupId": "...",
    "height": 480,
    "resizeMode": "none",
    "width": 640
  }
}

In this example video.frameRate is a supported property and it's actual value is 30.

video.zoom is not a supported property for this combination of device/camera/navigator as it is not present in the returned object.

Stream Transformation

Audio filters : noiseReduction - ApplyAudioProcessor()

Noise reduction feature is available on apiRTC 5.0.0

ApiRTC allows to create a stream with a noise reduction filter.

Check the noise reduction tutorial

applyAudioProcessor helper manages the different stream states for you. (ie : switch from noisedReduction to normal mode)

To start the noise reduction process on a Stream:

stream.applyAudioProcessor('noiseReduction').then((streamWithEffect) => {
...
})

This method returns a streamWithEffect Stream object; it is a encapsulated object of the base stream with an noise reduction filter applied on it.

It means that there both, the base stream and the streamWithEffect stream, are still linked :

• If the base stream audio is muted the streamWithEffect stream audio will be too,

• If the base stream is released, the streamWithEffect stream will be too.

Both streams need to be handled by the application as the noise reduction process is going on.

To stop the noise reduction process:

// stop the noise reduction from base stream
stream.applyAudioProcessor('none').then((streamWithoutEffect) => {
...
})

If an error occurs during applyAudioProcessor() process, apiRTC will reject the promise but will try to restore stream with previous effect.

Error description is available in the ApiRTC JS Library Reference

Additionally, ApiRTC gives you access to Stream.startNoiseReduction and Stream.stopNoiseReduction methods.

Background subtraction : blur, background image - applyVideoProcessor()

ApiRTC allows to create a background blurred stream or to add a background image based on an original stream.

Have you checked the blur application tutorial?

applyVideoProcessor helper manages the different stream states. (ie : switch from blur to background image ...)

To start the blur process on a stream:

stream.applyVideoProcessor('blur').then((streamWithEffect) => {
...
})

This method returns a streamWithEffect Stream object; it is a encapsulated object of the base stream with blur filter applied on it.

It means that there both, the base stream and the streamWithEffect stream, are still linked :

• If the base stream audio is muted the streamWithEffect stream audio will be too,

• If the base stream is released, the streamWithEffect stream will be too.

Both streams need to be handled by the application as the noise reduction process is going on.

Use the stream with effect as a local stream:

// display streamWithEffect media stream by attaching to a media element (like <video>)
streamWithEffect.attachToElement(videoDomElement)

// publish the streamWithEffect stream
conversation.publish(streamWithEffect).then(() => {
  ...
});

To stop the blur process:

// stop blur from original stream
stream.applyVideoProcessor('none').then((blurredStream) => {
...
})

Additionally, ApiRTC gives you access to Stream.blur(), Stream.unblur(), Stream.backgroundImage(), Stream.unBackgroundImage().

Whiteboard

The whiteboard component enables participants to interact together with:

• lines (pen)

• shapes (arrow, rectangle or ellipse)

• texts

• and also an eraser (eraser)

Lines weight and colors, and text size are configurable. Undo & redo functions are available (whiteboardClient.undo and whiteboardClient.redo). The whiteboard area can be zoomed in and out (whiteboardClient.setScale), and shifted around (whiteboardClient.setOffset). The whiteboard can be erased at once with the whiteboardClient.deleteHistory function.

Adding a whiteboard to a web page takes a few lines:

  ...
    conversation.startNewWhiteboardSession('canvas-element-id');  // instanciate a whiteboard in a canvas
    whiteboardClient = userAgent.getWhiteboardClient(); //retrieve the Whiteboardclient object
    whiteBoardClient.setFocusOnDrawing(true); //The whiteboard follows the drawings done by other users if the canvas is set on a scrollable container
    ...

See the whiteboard in action in the following github repos:

• Whiteboard example/Demo

• Whiteboard with invite example/Demo