Kotlin

Sariska Media provides a robust suite of Kotlin APIs designed to streamline the development of real-time android applications. With Sariska, you can seamlessly integrate a variety of features.

Installation

Step 1 : Install Sariska Media Transport Client

With Pre-built Artifacts

  • For older Android Studio versions:

  1. In your project's root directory, locate the build.gradle file.

  2. Add the following code block within the repositories section:

allprojects {
    // Add the Sariska repository
    repositories {
        maven {
            url "https://github.com/SariskaIO/sariska-maven-repository/raw/master/releases"
        }
        
        // Add other repositories
        google()
        mavenCentral()
        maven { url 'https://www.jitpack.io' }
    }
}

In recent versions of Android Studio, the allprojects section might be absent in build.gradle

In that case, the repository goes into the settings.gradle file in the root of your project.

  • For newer Android Studio versions:

  1. In your project's root directory, locate the settings.gradle file.

  2. Add the following code block within the repositories section:

dependencyResolutionManagement {
    repositoriesMode.set(RepositoriesMode.FAIL_ON_PROJECT_REPOS)
    repositories {
        google()
        mavenCentral()
        maven {
            url "https://github.com/SariskaIO/sariska-maven-repository/raw/master/releases"
        }
        maven {
            url "https://maven.google.com"
        }
    }
}

With Maven

Add the dependency:

dependencies {
    implementation 'io.sariska:sariska-media-transport:5.4.9'
}

Step 2 : Initialize SDK

After installing the Sariska Media Transport SDK, begin by initializing it.

import io.sariska.sdk.SariskaMediaTransport;
import io.sariska.sdk.Connection;
import io.sariska.sdk.Conference;
import io.sariska.sdk.JitsiRemoteTrack;
import io.sariska.sdk.JitsiLocalTrack;
import io.sariska.sdk.Participant;

SariskaMediaTransport.initializeSdk();

Step 3 : Establish a WebSocket Connection

Media Services utilizes WebSockets to establish a continuous, two-way communication channel between your application and the server. This persistent connection offers significant advantages over traditional HTTP requests, particularly in terms of responsiveness.

// Replace {your-token} with your actual Sariska token
val token = {your-token}
  
// Create a connection object
val connection = SariskaMediaTransport.JitsiConnection(token, "roomName", isNightly)
  
  
// Add event listeners for connection status
  
connection.addEventListener("CONNECTION_ESTABLISHED"), { 
// Handle successful connection establishment
}

connection.addEventListener("CONNECTION_FAILED"), { error->
  if (error === PASSWORD_REQUIRED) {
    // Token expired, update the connection with a new token
    connection.setToken(token)
  }
}

connection.addEventListener("CONNECTION_DISCONNECTED"), {
// Handle connection disconnection  
}
  

// Connect to the Sariska server
connection.connect()

Step 4 : Initiate a Conference

Create a Jitsi-powered conference for real-time audio and video.

// Create a conference object using the connection and optional configuration
val conference = connection.initJitsiConference(options);

// Join the conference
conference.join();
Customize conference

To join with audio muted

startAudioMuted: true

To join with video muted

startVideoMuted: true

To join with startSilent no audio receive/send

startSilent: true

To enable rtcstats tracking analytics

rtcstatsServer: ""

CallStats ID

callStatsID: ""

CallStats Secret

callStatsSecret: ""

Last n speakers

channelLastN: 10

Step 5 : Capture Local Stream

A MediaStream consists of various audio or video tracks represented by MediaStreamTrack objects. Each track can have multiple channels, which are the smallest units of the stream (e.g., left and right audio channels in stereo).

// Define options for local stream capture
val options = Bundle()

// Enable audio and video tracks in the stream
options.putBoolean("audio", true)
options.putBoolean("video", true)

// Set desired video resolution
options.putInt("resolution", 240) // Specify desired resolution
// ... (additional options for desktop sharing, facing mode, devices, etc.)

// Create local audio and video tracks based on the options
SariskaMediaTransport.createLocalTracks(options) { tracks ->
localTracks = tracks;
}
Customize tracks

devices:

  • Type: Array of strings

  • Values: "desktop", "video", "audio"

  • Purpose: Specifies which devices to request from the browser's GetUserMedia (GUM) API.

  • Default: If this property is not set, GUM will attempt to access all available devices.

resolution:

  • Type: String

  • Values: 180, 240, 360, vga, 480, qhd, 540, hd, 720, fullhd, 1080, 4k, 2160

  • Purpose: Sets the preferred resolution for the local video stream.

cameraDeviceId

  • Type: String

  • Purpose: Specifies the device ID of the camera to use.

micDeviceId

  • Type: String

  • Purpose: Specifies the device ID of the microphone to use.

minFps

  • Type: Integer

  • Purpose: Sets the minimum frame rate for the video stream.

maxFps

  • Type: Integer

  • Purpose: Sets the maximum frame rate for the video stream.

desktopSharingFrameRate

  • Type: Object

  • Properties:

    • min: Minimum frame rate for desktop sharing

    • max: Maximum frame rate for desktop sharing

desktopSharingSourceDevice

  • Type: String

  • Purpose: Specifies the device ID or label of the video input source to use for screen sharing.

facingMode

  • Type: String

  • Values: "user", "environment"

  • Purpose: Sets the camera's facing mode (front-facing or back-facing).

Step 6 : Play Local Stream

runOnUiThread(() -> {
    // Loop through each local track
    for (JitsiLocalTrack track : localTracks) {
        if (track.getType().equals("video")) {
            // Get the view for rendering the video track
            val view = track.render();
            
            // Set the view to fill the container
            view.setObjectFit("cover");
            
            // Add the video view to the container layout
            mLocalContainer.addView(view);
        }
    }
})
Additional view properties

view.setMirror(mirror): Set whether to mirror your video (true/false).

view.setObjectFit("cover"): Set how the video fills the view ("cover" or "contain").

view.setZOrderMediaOverlay(0): Set the layering order (0 or 1).

view.setZOrderOnTop(0): Set the layering order (0, 1, or 2).

Audio playback is handled automatically with the video stream.

Step 7 : Handle User Joined Event

This event is triggered when a new user joins the conference. Moderators have exclusive control over meetings and can manage participants. To assign a moderator, set the moderator value to true when generating the token.

Moderator permissions
  • Password Protection:

Ability to add a password to the meeting room, restricting access.

  • Role Assignment:

Ability to grant moderator privileges to other participants.

  • Participant Removal:

Ability to kick non-moderators or even other moderators from the meeting.

  • Audio Control:

Ability to mute individual participants or all participants at once.

  • Video Focus:

Ability to make everyone's video view follow the moderator's video.

  • Joining Settings: Ability to:

    • Set participants to join with audio muted by default.

    • Set participants to join with video disabled by default.

  • Lobby Management:

Ability to enable or disable the lobby room, requiring approval for participants to join.

  • Join Approval:

Ability to approve or deny join requests when the lobby is enabled.

  • Moderator Transfer:

If the current moderator leaves the meeting, a new moderator is automatically selected.

conference.addEventListener("USER_JOINED", (id,  participant)=>{
});
Joined participant properties

avatar:

  • Type: String

  • Purpose: Used to display their profile picture in the UI.

email:

  • Type: String

  • Purpose: May be used for identification or communication purposes.

moderator:

  • Type: Boolean

  • Purpose: Used to control moderation-related features in the UI.

audioMuted:

  • Type: Boolean

  • Purpose: Used to display the audio muted state in the UI.

videoMuted:

  • Type: Boolean

  • Purpose: Used to display the video muted state in the UI.

displayName:

  • Type: String

  • Purpose: Used to identify them in the UI.

role:

  • Type: String

  • Purpose: Used to determine their permissions and UI features.

status:

  • Type: String

  • Purpose: Used to display ("online", "offline", "away") their availability in the UI.

hidden:

  • Type: Boolean

  • Purpose: Typically used for bots like transcribers or recorders.

botType:

  • Type: String

  • Purpose: Used to identify the bot's purpose and capabilities.

Step 8 : Publish Streams

Make audio and video streams visible to others in the conference by publishing them using the following code:

// Loop through all local tracks
for (JitsiLocalTrack track : localTracks) {
    // Add the track to the conference, allowing others to see/hear you
    conference.addTrack(track);
}

Step 9 : Play Remote Peers Streams

// Add an event listener to the conference for "TRACK_ADDED" events
conference.addEventListener("TRACK_ADDED") { p ->
    // Cast the event object to a JitsiRemoteTrack instance
    val track: JitsiRemoteTrack = p as JitsiRemoteTrack
    
    // Run code on the UI thread
    runOnUiThread {
        // Check if the track type is video
        if (track.getType().equals("video")) {
            // Render the video track using Jitsi's WebRTCView component
            val view: WebRTCView = track.render()
            
            // Set the view to cover the entire container
            view.setObjectFit("cover")
            
            // Store the view for potential later reference
            remoteView = view
            
            // Add the view to the container layout, displaying the remote video
            mRemoteContainer!!.addView(view)
        }
    }
}

Analytics

Sariska-media-transport offers pre-configured events to help you track and analyze user interactions, media usage, and overall performance. This data can be used to enhance your product, improve user experience, and make informed decisions.

Available Events

Here are some of the key events you can track:

  • User Actions:

    • User joined

    • User left

  • Media Usage:

    • Conference duration

    • Camera duration

    • Audio track duration

    • Video track duration

  • Recording:

    • Recording started

    • Recording stopped

    • Local recording started

    • Local recording stopped

  • Transcription:

    • Transcription started

    • Transcription stopped

  • Performance:

    • Speaker stats

    • Connection stats

    • Performance stats

Add Event Listener to Track Events

conference.startTracking();

Features

Sariska offers powerful features to enhance your application's capabilities. Find your desired feature using the search bar or explore below!

Active/Dominant Speaker

Identify the main speaker: Easily detect the active or dominant speaker in a conference. Choose to stream only their video for improved resolution and reduced bandwidth usage. Ideal for one-way streaming scenarios like virtual concerts.

conference.addEventListener("DOMINANT_SPEAKER_CHANGED") { p ->
    // participantId is a string containing the ID of the dominant speaker
    val participantId = p as String 
}

Last N Speakers

Dynamically showcase recent speakers: Focus on the active conversation by displaying video only for the last N participants who spoke. This automatically adjusts based on speech activity, offering a more efficient and relevant view.

// Listen for last N speakers changed event
conference.addEventListener("LAST_N_ENDPOINTS_CHANGED") { leavingEndpointIds,  enteringEndpointIds -> 
    // leavingEndpointIds: Array of IDs of users leaving lastN
    // enteringEndpointIds: Array of IDs of users entering lastN
};

// Set the number of last speakers to show
conference.setLastN(10)

Participant Information

Set local participant properties: Define additional information about participants beyond the default settings. This can include screen-sharing status, custom roles, or any other relevant attributes.

// Set local participant property 
conference.setLocalParticipantProperty(key, value);

// Remove a local participant property
conference.rempveLocalParticipantProperty(key)

// Get the value of a local participant property 
conference.getLocalParticipantProperty(key)

// Listen for changes in participant properties
conference.addEventListener("PARTICIPANT_PROPERTY_CHANGED"){ participant, key, oldValue, newValue ->
}

Participant Count

Get the total number of participants: Retrieve the complete participant count, including both visible and hidden members.

conference.getParticipantCount(); 
// Pass true to include hidden participants

Some conferences may include hidden participants besides attendees, such as bots assisting with recording, transcription, or pricing.

Participant Lists

Access all participants: Obtain a list of all participants, including their IDs and detailed information.

// Get all participants
conference.getParticipants(); // List of all participants

// Get participants excluding hidden users
conference.getParticipantsWithoutHidden(); // List of all participants

Pinning Participants

  • Pin a single participant: Give a participant higher video quality (if simulcast is enabled).

conference.selectParticipant(participantId) // Select participant with ID

  • Pin multiple participants: Give multiple participants higher video quality.

conference.selectParticipants(participantIds) // Select participant with IDs

Access Local User Details

Retrieve information about the local user directly from the conference object.

// Check if the local user is hidden
conference.isHidden()

// Get local user details
confernece.getUserId()
conference.getUserRole()
conference.getUserEmail()
conference.getUserAvatar()
conference.getUserName()

Set Meeting Subject

Set the subject of the meeting.

conference.setSubject(subject)

Manage Tracks

  • Get all remote tracks: Retrieve a list of all remote tracks (audio and video) in the conference.

conference.getRemoteTracks();

  • Get all local tracks: Retrieve a list of all local tracks (audio and video)

conference.getLocalTracks()

Kick Participants

  • Listen for participant kick events

conference.addEventListener("KICKED"){ id ->
   // Handle participant kicked
};

conference.addEventListener("PARTICIPANT_KICKED") { actorParticipant, kickedParticipant, reason ->
}

  • Kick a participant

confernece.kickParticipant(id)

Manage Owner Roles

The room creator has a moderator role, while other users have a participatory role.

  • Grant owner rights

conference.grantOwner(id) // Grant owner rights to a participant

  • Listen for role changes

conference.addEventListener("USER_ROLE_CHANGED"){ id, role ->
    
    if (confernece.getUserId() === id ) {
        // My role changed, new role: role;
    } else {
       // Participant role changed: role;
    }
};

  • Revoke owner rights

conference.revokeOwner(id) // Revoke owner rights from a participant

Change Display Names

  • Setting a new display name

conference.setDisplayName(name); // Change the local user's display name

  • Listen for display name changes

conference.addEventListener("DISPLAY_NAME_CHANGED"){ id, displayName->
    // Access the participant ID
};

Lock/Unlock Room

  • Lock room: Moderators can restrict access to the room with a password.

conference.lock(password); // Lock the room with the specified password

  • Unlock room: Removes any existing password restriction.

conference.unlock();

Subtitles

  • Request subtitles: Enable subtitles for spoken content.

conference.setLocalParticipantProperty("requestingTranscription", true);

  • Request language translation: Translate subtitles into a specific language.

conference.setLocalParticipantProperty("translation_language", 'hi'); // Example for Hindi

Language Code
Language Name

af

Afrikaans

ar

Arabic

bg

Bulgarian

ca

Catalan

cs

Czech

da

Danish

de

German

el

Greek

en

English

enGB

English (United Kingdom)

eo

Esperanto

es

Spanish

esUS

Spanish (Latin America)

et

Estonian

eu

Basque

fi

Finnish

fr

French

frCA

French (Canadian)

he

Hebrew

hi

Hindi

hr

Croatian

hu

Hungarian

hy

Armenian

id

Indonesian

it

Italian

ja

Japanese

kab

Kabyle

ko

Korean

lt

Lithuanian

ml

Malayalam

lv

Latvian

nl

Dutch

oc

Occitan

fa

Persian

pl

Polish

pt

Portuguese

ptBR

Portuguese (Brazil)

ru

Russian

ro

Romanian

sc

Sardinian

sk

Slovak

sl

Slovenian

sr

Serbian

sq

Albanian

sv

Swedish

te

Telugu

th

Thai

tr

Turkish

uk

Ukrainian

vi

Vietnamese

zhCN

Chinese (China)

zhTW

Chinese (Taiwan)

mr

Marathi

  • Receive subtitles: Listen for incoming subtitles.

conference.addEventListener("SUBTITLES_RECEIVED"){ id, name, text->
// Handle received subtitle data (id, speaker name, text)
};

  • Stop subtitles: Disable subtitles.

conference.setLocalParticipantProperty("requestingTranscription", false);

Screen Share

Each participant can contribute two types of data to a meeting: audio and video. Screen sharing counts as a video track. If you want to share your screen while also showing your own video (like when presenting), you need to enable "Presenter mode". This mode hides the gallery of other participants' videos and gives you more control over the meeting layout.

  • Start screen share: Share your screen with other participants.

// Create a desktop track
val options = new Bundle();  
options.putBoolean("desktop", true);

val videoTrack = localTracks[1]; // Participant's local video track

SariskaMediaTransport.createLocalTracks(options, tracks -> {
    conference.replaceTrack(videoTrack, tracks[0]);
});

Messages

Sariska offers robust messaging capabilities for both private and group communication scenarios.

  • Send a group message to all participants

conference.sendMessage("message");

  • Send a private message to a specific participant

conference.sendMessage("message", participantId);

  • Listen for incoming messages

// Add an event listener to handle incoming messages
conference.addEventListener("MESSAGE_RECEIVED" ){ message, senderId->
// Process the received message
});

Transcription

  • Start Transcription: Initiate transcription for the ongoing conference.

conference.startTranscriber();

  • Stop Transcription: Stop transcription and get a download link for the transcript.

conference.stopTranscriber();

Mute/Unmute Participants

  • Mute Remote Participant

conference.muteParticipant(participantId, mediaType)
// participantId: ID of the participant to be muted
// mediaType: Type of media to mute ('audio' or 'video')

The moderator has the ability to temporarily silence the microphone of any participant who is not physically present in the meeting room.

  • Mute/Unmute Local Participant

// Mute a local track (audio or video)
track.mute()

// Unmute a previously muted local track
track.unmute()

Connection Quality

  • Local Connection Statistics Received

conference.addEventListener(event: "LOCAL_STATS_UPDATED"){ stats in
// Handle local connection statistics
}

  • Remote Connection Statistics Received

conference.addEventListener(event: "REMOTE_STATS_UPDATED") { id, stats-> 
// Handle remote connection statistics
}

Internet Connectivity Status

The SDK features intelligent auto-join/leave functionality based on internet connectivity status. This helps optimize network resources and improve user experience.

Peer-to-Peer Mode

Designed for efficient communication between two participants.

  • Start Peer-to-Peer Mode

Sariska automatically activates Peer-to-Peer mode when your conference has exactly two participants. This mode bypasses the central server and directly connects participants, maximizing bandwidth efficiency and reducing latency. However, even with more than two participants, you can forcefully start Peer-to-Peer mode.

conference.startP2PSession();

Starting Peer-to-Peer mode eliminates server charges as long as the central turn server remains unused.

  • Stop Peer-to-Peer Mode

If you need to revert to server-mediated communication, you can easily stop Peer-to-Peer mode.

conference.stopP2PSession();

CallStats Integration

Monitor your WebRTC application performance using CallStats (or build your own). See the "RTC Stats" section for details.

val options = new Bundle();
options.putString("callStatsID", 'callstats-id');
options.putString("callStatsSecret", 'callstats-secret');

val conference = connection.initJitsiConference(options);

Join Muted/Silent

Join conferences with audio and video already muted, or in a silent mode where no audio is transmitted or received. This ensures a seamless experience and respects participant preferences.

  • Join with Muted Audio and Video

val options = new Bundle();
options.putBoolean("startAudioMuted", true);
options.putBoolean("starVideoMuted", true);

val conference = connection.initJitsiConference(options);

  • Join in Silent Mode

val options = new Bundle();
options.putBoolean("startSilent", true);

val confernce = connection.initJitsiConference(options);

Live Stream

Broadcast your conference to multiple platforms simultaneously. Embed live streams directly into your app or website using various formats.

  • Stream to YouTube

val options = new Bundle();
options.putString("broadcastId", "youtubeBroadcastID"); // Put any string this will become part of your publish URL
options.putString("mode", "stream");
options.putString("streamId", "youtubeStreamKey"); 

// Start live stream
conference.startRecording(options);

  • Stream to Facebook

val options = new Bundle();
options.putString("mode", "stream");
options.putString("streamId", "rtmps://live-api-s.facebook.com:443/rtmp/FB-4742724459088842-0-AbwKHwKiTd9lFMPy");

// Start live stream
conference.startRecording(options);

  • Stream to Twitch

val options = new Bundle();     
options.putString("mode", "stream");        
options.putString("streamId", "rtmp://live.twitch.tv/app/STREAM_KEY");      

// Start live stream
conference.startRecording(options);     

  • Stream to any RTMP Server

val options = new Bundle();
options.putString("mode", "stream");
options.putString("streamId", "rtmps://rtmp-server/rtmp"); // RTMP server URL

// Start live stream
conference.startRecording(options);

Listen for RECORDER_STATE_CHANGED event to track streaming status

conference.addEventListener("RECORDER_STATE_CHANGED"){ sessionId, mode, status in
// Verify mode is "stream"
// Get the live streaming session ID
// Check the streaming status: on, off, or pending
};

Stop Live Stream

conference.stopRecording(sessionId);

Cloud Recording

Store your recordings and transcriptions in various cloud storage services.

Supported storage providers
  1. Object-based storage: Amazon S3, Google Cloud Storage, Azure Blob Storage, other S3-compatible cloud providers.

  2. Dropbox

Set credentials
  1. Object-based storage

    • Access your Sariska dashboard

    • Locate the storage credentials section

    • Enter the required credentials for your chosen provider

  2. Dropbox

    • Obtain a Dropbox OAuth token

// Configure for Object-based storage
val options = new Bundle();
options.putString("mode", "file");
options.putString("serviceName", "s3");


// Configure for Dropbox
val options = new Bundle();
options.putString("mode", "file");
options.putString("serviceName", "dropbox");
options.putString("token", "dropbox_oauth_token");


// Start recording
conference.startRecording(options);


// Monitor recording state
conference.addEventListener("RECORDER_STATE_CHANGED"){ sessionId, mode, status ->
   val sessionId = sessionId as String; // Unique identifier for the cloud recording session
   val mode = mode as String; // Recording mode (e.g., "file")
   val status = status as String; // Current recording status ("on", "off", or "pending")
// Handle recording state changes based on mode, sessionId, and status
});

// Stop recording
conference.stopRecording(sessionId)

PSTN

  • Dial-in(PSTN)

// Retrieve the phone pin and number for users to join via PSTN:

val phonePin = conference.getPhonePin() // Get the phone pin for PSTN access
val phoneNumber = conference.getPhoneNumber() // Get the phone number for PSTN access 

Share this phone number and pin with users to enable PSTN conference call participation.

  • Dial-out(PSTN)

// Dial a phone number to invite a participant to the conference
conference.dial(phoneNumber)

This initiates a call to the provided phone number, inviting them to join the conference.

Lobby/Waiting Room

// Join the lobby
conference.joinLobby(displayName, email); // Request to join the conference lobby

// Event listeners for lobby-related actions:
conference.addEventListener("LOBBY_USER_JOINED"){ id, name ->
    // Handle events when a user joins the lobby
    val id = id as String;
    val name = name as String
})

conference.addEventListener(event: "LOBBY_USER_UPDATED"), { id, participant ->
    // Handle events when a user's information in the lobby is updated
    val id = id as String;
    val name = participant as Participant
}


// Additional event listeners for lobby actions:
conference.addEventListener(event: "LOBBY_USER_LEFT") { id ->
    val id = id as String;
}

conference.addEventListener(event: "MEMBERS_ONLY_CHANGED") { enabled ->
    val enabled = enabled as Boolean;
}


// Moderator actions for lobby access:
conference.lobbyDenyAccess(participantId); // Deny access to a participant in the lobby
conference.lobbyApproveAccess(participantId); // Approve access to a participant in the lobby


// Lobby management methods:
conference.enableLobby() // Enable lobby mode for the conference (moderator only)
conference.disableLobby(); // Disable lobby mode for the conference (moderator only)
conference.isMembersOnly(); // Check if the conference is in members-only mode (lobby disabled)

SIP Video Gateway

// Initiate a SIP video call
conference.startSIPVideoCall("address@sip.domain.com", "display name"); // Start a SIP video call with the specified address and display name

// Terminate a SIP video call
conference.stopSIPVideoCall('address@sip.domain.com');

// Event listeners for SIP gateway state changes
conference.addEventListener("VIDEO_SIP_GW_SESSION_STATE_CHANGED", (state)=>{
    // Handle events when the SIP gateway session state changes (on, off, pending, retrying, failed)
}

// Event listener for SIP gateway availability changes
conference.addEventListener("VIDEO_SIP_GW_AVAILABILITY_CHANGED"){ status ->
    // Handle events when the SIP gateway availability changes (busy or available)
}

One-to-One Calling

This allows synchronous phone calls, similar to WhatsApp, even if the receiver's app is closed or in the background.

Initiating Calls:

  • Make a call even if the callee's app is closed or in the background.

  • Play a busy tone if the callee is on another call or disconnects your call.

  • Play ringtone/ringback/DTMF tones.

Step 1 : Caller Initiates Call

  1. HTTP Call to Sariska Server

{API Method}

  1. Push Notification to callee using Firebase or APNS

This notifies the receiver even if their app is closed or in the background.

Step 2 : Callee Responds to Call

  1. Reads Push Notification

Processes the notification even if the app is closed or in the background.

  1. HTTP Call to Update Status

{API Method}

  1. No need to join conference via SDK

Status update through the HTTP call suffices.

Step 3 : Caller Receives Response

Listens for USER_STATUS_CHANGED event

conference.addEventListener(event: "USER_STATUS_CHANGED"){ id, status ->
    val id = id as! String;
    val status = status as!
// - id: receiver's user id
// - status: "ringing", "busy", "rejected", "connected", "expired"
};
Event status details
  • ringing: receiver's phone is ringing

  • busy: receiver is on another call

  • rejected: receiver declined your call

  • connected: receiver accepted your call

  • expired: receiver didn't answer within 40 seconds

Step 4 : After Connection Established

The call proceeds like a normal conference call.

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