JavaScript
Unleash real-time audio/video, live streaming, cloud recording, transcriptions, language translation, and more in your web and mobile apps with the versatile Sariska Media JavaScript APIs.
Key Features:
Seamlessly integrate with various JavaScript frameworks (Vanilla, React, Angular, Vue, Electron, NW, React Native, and more).
Access a rich set of features for audio/video conferencing, live streaming, cloud recording, transcriptions, language translation, virtual backgrounds, and more.
Maintain persistent, low-latency connections for real-time data exchange.
Installation
Step 1 : Install Sariska Media Transport Library
With NPM
npm i sariska-media-transport
With CDN
Add a script tag to your HTML head
<script src="https://sdk.sariska.io/umd/sariska-media-transport.min.js"></script>
Compatibility for React Native: Polyfill Required
At the very beginning of your index.js
file, insert the following import statement:
import 'sariska-media-transport/dist/esm/modules/mobile/polyfills';
Step 2 : Initialize the SDK
Kickstart the SDK with this simple command:
import SariskaMediaTransport from "sariska-media-transport";
SariskaMediaTransport.initialize();
Step 3 : Establish a WebSocket Connection
Create a persistent connection for real-time communication.
const token = {your-token};
const connection = new SariskaMediaTransport.JitsiConnection(token, "roomName", isNightly);
connection.addEventListener(SariskaMediaTransport.events.connection.CONNECTION_ESTABLISHED, () => {
console.log('connection successful!!!');
});
// Handle connection events
connection.addEventListener(SariskaMediaTransport.events.connection.CONNECTION_FAILED, (error) => {
// Token expired, set again
if (error === SariskaMediaTransport.events.connection.PASSWORD_REQUIRED) {
// Set a new token
connection.setToken(token)
console.log('connection disconnect!!!', error);
}
});
connection.addEventListener(SariskaMediaTransport.events.connection.CONNECTION_DISCONNECTED, (error) => {
console.log('connection disconnect!!!', error);
});
connection.connect();
Step 4 : Initiate a Conference
Create a Jitsi-powered conference for real-time audio and video
const conference = connection.initJitsiConference(options);
conference.join();
Step 5 : Capture Local Stream
Media Stream
A MediaStream is a collection of audio or video tracks, represented by MediaStreamTrack objects.
Each MediaStreamTrack can have multiple channels (e.g., left and right channels in a stereo audio track).
Capture Local Tracks
Define options:
Specify desired devices ("audio", "video", or "desktop").
Set preferred video resolution.
Optionally configure specific devices, frame rates, screen sharing options, and facing mode.
const options = {
devices: ["audio", "video"],
resolution: 240,
}
Create Local Tracks
const localTracks = await SariskaMediaTransport.createLocalTracks(options);
Step 6 : Play Local Stream
// Access local media tracks
const audioTrack = localTracks.find(track=>track.getType()==="audio");
const videoTrack = localTracks.find(track=>track.getType()==="video");
// Play video
videoTrack.attach(document.getElementById("videoElement"))
// Play audio
audioTrack.attach(document.getElementById("audioElement"))
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.
conference.addEventListener(SariskaMediaTransport.events.conference.USER_JOINED, function(id, participant) {
console.log("user joined!!!", id, participant);
});
Step 8 : Publish Streams
Make audio and video streams visible to others in the conference by publishing them using the following code:
localTracks.forEach(track => conference.addTrack(track));
Step 9 : Play Remote Peers Streams
const remoteTracks = [];
conference.addEventListener(SariskaMediaTransport.events.conference.TRACK_ADDED, function(track) {
remoteTracks.push(track);
});
remoteTracks.forEach(track => {
if (track.getType() === "audio") {
RemoteContainer.append(track.attach(document.getElementById("remoteAudioElemId")));
}
if (track.getType() === "video") {
RemoteContainer.append(track.attach(document.getElementById("remoteVideoElemId")));
}
});
Additional methods for remote tracks:
getType():
Returns the track type (audio, video, or desktop)stream.toURL():
Returns the stream URLgetId():
Returns the track IDisMuted():
Checks if the track is mutedgetParticipantId():
Returns the participant ID associated with the trackisLocal():
Checks if the track is localattach():
Attaches the track to an audio or video elementdetach():
Detaches the track from an audio or video element
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
Browser performance stats
Add Event Listener to Track Events
conference.addEventListener(SariskaMediaTransport.events.conference.ANALYTICS_EVENT_RECEIVED, (payload)=> {
// Construct the payload
const { name, action, actionSubject, source, attributes } = payload;
})
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.
// Listen for changes in the dominant speaker
conference.addEventListener(SariskaMediaTransport.events.conference.DOMINANT_SPEAKER_CHANGED, id=> {
console.log(id) // Dominant speaker ID
});
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.
// Track changes in the "last N" speakers
conference.addEventListener(SariskaMediaTransport.events.conference.LAST_N_ENDPOINTS_CHANGED, (leavingEndpointIds, enteringEndpointIds)=> {
console.log(leavingEndpointIds) //Array of ID's of users leaving lastN
console.log(enteringEndpointIds) //Array of ID's of users entering lastN
});
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 a 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(SariskaMediaTransport.events.conference.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
Participant Lists
Access all participants: Obtain a list of all participants, including their IDs and detailed information.
// Get all participants
conference.getParticipants(); // Array of {participantId: details} objects
// Get participants without hidden users
conference.getParticipantsWithoutHidden(); // Array of {participantId: details} objects
All Participants as an Object
Advanced manipulation: You can directly access the conference
object for more granular control over conference behavior.
conference.participants; // {participantId: details}
Pinning Participants
Pin a single participant: Pin a specific participant to always receive their video, even when "last n" is enabled.
conference.pinParticipant(participantId)
Pin multiple participants: Pin an array of participants to always receive their videos.
conference.pinParticipant(participantIds)
Access Local User Details
Retrieve the local user's ID: Get the ID of the local user.
conference.myUserId();
Retrieve the local user's information: Get comprehensive details about the local user, including name, email, ID, and avatar.
conference.getLocalUser();
Retrieve the local user's role: Get the role of the local user (For example, participant, moderator, owner).
conference.getRole();
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();
Breakout Rooms
Split your conference meeting into smaller, focused groups with unique audio and video. Moderators can create rooms, assign participants, and reunite everyone seamlessly.
Access breakout rooms: Get an instance to manage breakout rooms.
const breakoutRooms = conference.getBreakoutRooms();
Create a breakout room: Create a new breakout room with the specified subject.
breakoutRooms.createBreakoutRoom("room subject");
Remove a breakout room: Remove the current breakout room (if applicable).
breakoutRooms.removeBreakoutRoom();
Check for breakout room status: Determine if the current room is a breakout room.
breakoutRooms.isBreakoutRoom();
Send a participant to a breakout room: Move a participant to a specific breakout room.
breakoutRooms.sendParticipantToRoom(participantJid, roomJid)
Kick Participants
Listen for participant kick events
conference.addEventListener(SariskaMediaTransport.events.conference.KICKED, (id)=> { // Handle a participant being kicked by a moderator
// The kicked participant's ID is available in the `id` variable
});
conference.addEventListener(SariskaMediaTransport.events.conference.PARTICIPANT_KICKED, (actorParticipant, kickedParticipant, reason) => { // Handle a moderator being kicked by another moderator
// Information about the actor, kicked participant, and reason is available in the event arguments
});
Kick a participant
confernece.kickParticipant(id)
Kick a moderator
confernece.kickParticipant(id, reason) // Kick a moderator, providing a reason for the kick
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(SariskaMediaTransport.events.conference.USER_ROLE_CHANGED, (id, role) => {
if (cofenerece.user.id === id ) {
console.log(`My role changed, new role: ${role}`);
} else {
console.log(`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(SariskaMediaTransport.events.conference.DISPLAY_NAME_CHANGED, (id, displayName)=> { // Handle display name changes for other participants
// 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
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(SariskaMediaTransport.events.conference.SUBTITLES_RECEIVED, (id, name, text)=> {
// Handle received subtitle data (id, speaker name, text)
});
Stop subtitles: Disable subtitles.
confernence.setLocalParticipantProperty("requestingTranscription", false);
Screen Share
Start screen share: Share your screen with other participants.
const desktopTrack = await SariskaMediaTransport.createLocalTracks({devices: ["desktop"]});
conference.addtrack(desktopTrack[0]);
Stop screen share: Stop sharing your screen.
await conference.removeTrack(desktopTrack);
Messages
Sariska offers robust messaging capabilities for both private and group communication scenarios.
Send and Receive Private Text Messages
// Send a private text message to a specific participant
conference.sendMessage("message", participantId);
// Listen for incoming private text messages
conference.addEventListener(SariskaMediaTransport.events.conference.PRIVATE_MESSAGE_RECEIVED, (participantId, message)=>{
});
Send and Receive Private Payload
// Send a private payload to a specific participant
conference.sendEndpointMessage(to, payload);
// Listen for incoming private payloads
conference.addEventListener(SariskaMediaTransport.events.conference.ENDPOINT_MESSAGE_RECEIVED, (participant, payload)=>{
});
Send and Receive Group Text Messages
// Send a group text message to all participants
conference.sendMessage("message", participantId);
// Listen for incoming group text messages
conference.addEventListener(SariskaMediaTransport.events.conference.MESSAGE_RECEIVED, (participantId, message)=>{
});
Send and Receive Group Payload
// Send a group payload to all participants
conference.sendEndpointMessage('', payload);
// Listen for incoming group payloads
conference.addEventListener(SariskaMediaTransport.events.conference.ENDPOINT_MESSAGE_RECEIVED, (participant, payload)=>{
});
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')
Mute/Unmute Local Participant
// Mute a local track (audio or video)
track.mute()
// Unmute a previously muted local track
track.unmute()
// Check if a local track is currently muted
track.isMuted()
Connection Quality
Local Connection Statistics Received
conference.addEventListener(SariskaMediaTransport.events.conference.LOCAL_STATS_UPDATED, (stats)=>{
// Handle local connection statistics
});
Remote Connection Statistics Received
conference.addEventListener(SariskaMediaTransport.events.conference.REMOTE_STATS_UPDATED, (id, stats)=>{
// Handle remote connection statistics
});
Audio
No Audio Signal
// Triggered when the conference audio input loses signal
conference.addEventListener(SariskaMediaTransport.events.conference.NO_AUDIO_INPUT, () => {
// Handle the absence of audio input
});
Audio Input State Changed
// Triggered when the audio input state switches between having or not having audio input
conference.addEventListener(SariskaMediaTransport.events.conference.AUDIO_INPUT_STATE_CHANGE, hasAudioInput => {
// Handle changes in audio input state
});
Audio Level Indicator
conference.addEventListener(SariskaMediaTransport.events.conference.TRACK_AUDIO_LEVEL_CHANGED, function() {
// Handle audio level change events
});
Noise Detection
Detect excessive noise from the microphone used in the conference.
conference.addEventListener(SariskaMediaTransport.events.conference.NOISY_MIC, function () {
// Handle noisy mic events, such as notifying the user or adjusting settings
});
Talk While Muted Detection
conference.addEventListener(SariskaMediaTransport.events.conference.TALK_WHILE_MUTED, function () {
// Handle talk while muted events, such as providing a visual indicator
});
Noise Suppression/Cancellation
Reduces background noise from audio signals using a recurrent neural network (RNN).
await SariskaMediaTransport.effects.createRnnoiseProcessor();
Virtual Background
Change the background behind you in video calls with various options:
Image: Define a static image as the background
const options = {
// Enable virtual background
backgroundEffectEnabled: true,
// Choose image background
backgroundType: "image",
// URL of the image
virtualSource: "https://image.shutterstock.com/z/stock-photo-jasna-lake-with-beautiful-reflections-of-the-mountains-triglav-national-park-slovenia-1707906793.jpg"
};
const effect = await SariskaMediaTransport.effects.createVirtualBackgroundEffect(options);
Blur: Blur the background for a subtle effect
const options = {
// Enable virtual background
backgroundEffectEnabled: true,
// Choose blur background
backgroundType: "blur",
// Adjust blur intensity (0-100)
blurValue: 25
}
const effect = await SariskaMediaTransport.effects.createVirtualBackgroundEffect(options);
Screen Share: Show your computer screen as the background
const [ desktopTrack ] = await SariskaMediaTransport.createLocalTracks({devices: ["desktop"]});
const options = {
// Enable virtual background
backgroundEffectEnabled: true,
// Choose screen share background
backgroundType: "desktop-share",
virtualSource: desktopTrack
}
const effect = await SariskaMediaTransport.effects.createVirtualBackgroundEffect(options);
Start and Stop Virtual Background Effect
Start: Apply the effect to your local video track
const videoTrack = localTracks.find(track=>track.getType()==="video"); // Get your video track
await videoTrack.setEffect(effect);
Stop: Remove the effect from your video track
await videoTrack.setEffect(undefined);
Capture Screenshots of Shared Screen
Periodically capture screenshots of your screen share (e.g., every 30 seconds) and upload them to your server for analysis.
Start capturing
const [ desktopTrack ] = await SariskaMediaTransport.createLocalTracks({devices: ["desktop"]});
const effect = await SariskaMediaTransport.effects.createScreenshotCaptureEffect(processScreenShot);
await effect.startEffect(
desktopTrack.getOriginalStream(),
desktopTrack.videoType
);
// Process the captured screenshot
const processScreenShot = (canvas) => {
var dataURL = canvas.toDataURL();
console.log("data", dataURL);
// Upload dataURL to your server
}
Stop capturing
effect.stopEffect()
CallStats Integration
Monitor your WebRTC application performance using CallStats (or build your own). See the "RTC Stats" section for details.
const options = {callStatsID: 'callstats-id', callStatsSecret: 'callstats-secret'}
const conference = connection.initJitsiConference(options);
Internet Connectivity Status
This ensures seamless connectivity even in the face of fluctuating internet connections. It automatically manages connections and disconnections as needed.
// Update network status and notify Sariska Media Transport
function updateNetwork() {
// Communicate the current online status to SariskaMediaTransport
SariskaMediaTransport.setNetworkInfo({isOnline: window.navigator.onLine});
}
// When the browser goes offline, updateNetwork() is called to inform SariskaMediaTransport
window.addEventListener("offline", updateNetwork);
// When the browser comes back online, updateNetwork() is called again to update the status
window.addEventListener("online", updateNetwork);
Speaker Stats
Get valuable insights into the current conversation, specifically focusing on speaker dominance. It analyzes the interaction and estimates how long each participant has held the dominant speaker role.
conference.getSpeakerStats();
Connection Stats
Gain insights into the connection quality of conference participants.
const connectionStats = conference.getConnectionState();
audioSsrc
The audio SSRC (Synchronization Source identifier)
bandwidth
The estimated available bandwidth in bits per second
bitrate
The current media bitrate in bits per second
bridgeCount
The number of bridges in use
codec
The codec being used for media transmission
connectionSummary
A brief summary of the connection quality (e.g., "Good", "Fair", "Poor")
e2eRtt
The estimated end-to-end round-trip time in milliseconds
participantId
The ID of the participant
framerate
The current media framerate in frames per second
isLocalVideo
Indicates whether the stats are for the local participant's video
maxEnabledResolution
The maximum resolution enabled for the participant
packetLoss
The percentage of packet loss
region
The region where the participant is located
resolution
The current resolution of the media stream
serverRegion
The region of the server handling the connection
shouldShowMore
Indicates whether more detailed connection stats are available
videoSsrc
The video SSRC
transport
The transport protocol being used (e.g., "UDP", "TCP")
Web Audio Mixer
Combine multiple audio tracks into a single, unified audio stream.
// Obtain the audio tracks to be mixed
const audioTrack1 = getTracks()[0];
const audioTrack2 = getTracks()[1];
// Create an audio mixer instance
const audioMixer = SariskaMediaTransport.createAudioMixer();
// Add individual audio streams to the mixer
audioMixer.addMediaStream(audioTrack1.getOriginalStream());
audioMixer.addMediaStream(audioTrack2.getOriginalStream());
// Initiate the mixing process and retrieve the resulting mixed stream
const mixedMediaStream = audioMixer.start();
// Extract the mixed audio track from the mixed stream
const mixedMediaTrack = mixedMediaStream.getTracks()[0];
// Maintain synchronization between the mixed track's enabled state and the track using the effect
End-to-End Encryption
Empower your application with robust end-to-end encryption, ensuring secure communication for your users.
Enable End-to-End Encryption
// Initialize Olm early for E2EE readiness
window.Olm.init().catch(e => {
console.error('Failed to initialize Olm, E2EE will be disabled', e);
delete window.Olm; // Remove Olm if initialization fails
});
// Activate E2EE:
conference.toggleE2EE(true); // Enable end-to-end encryption
// Verify E2EE support
conference.isE2EESupported() // Check if E2EE is available
Disable End-to-End Encryption
conference.toggleE2EE(false); // Disable end-to-end encryption
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
// Join the conference with both audio and video muted initially
const conference = connection.initJitsiConference({
startAudioMuted: true, // Mute audio upon joining
startVideoMuted: true // Mute video upon joining
});
Join in Silent Mode
// Join the conference in silent mode, disabling both audio input and output
const conference = connection.initJitsiConference({
startSilent: true // Enter the conference in silent mode
});
Live Stream
Broadcast your conference to multiple platforms simultaneously. Embed live streams directly into your app or website using various formats.
// Define streaming destinations and settings
const appData = {
// Keys for platforms to stream
streamKeys: [
{streamKey: "youtube", streamValue: "youtube-stream-key"},
{streamKey: "facebook", streamValue: "facebook-stream-key"},
{streamKey: "twitch", streamValue: "twitch-stream-key"},
{streamKey: "vimeo", streamValue: "vimeo-stream-key"},
{streamKey: "periscope", streamValue: "periscope-stream-key"},
{streamKey: "instagram",streamValue: "instagram-stream-key"}
// Add keys for other platforms as needed
],
// Optional list of additional RTMP URLs for streaming
streamUrls: ["rtmp://test-rtmp-url-1", "rtmp://test-rtmp-url-2", "rtmp://test-rtmp-url-n"],
isRecording: false,
// Specify "live" for embedding the stream
app: "live",
stream: "livestream"
}
// app and stream: If you want to embed live streaming to your app or website just pass app and stream then you can embed and play live streaming in your platform for HTTP-FLV, HLS, DASH and HDS, mp3, aac
* Play HTTP-FLV: https://edge.sariska.io/http-flv/live/livestream.flv
* Play HLS: https://edge.sariska.io/hls/live/livestream.m3u8
* Play DASH: https://edge.sariska.io/dash/live/livestream.mpd
* Play MP3: https://edge.sariska.io/mp3/live/livestream.mp3
* Play AAC: https://edge.sariska.io/acc/live/livestream.aac
// Start live recording with configuration data
conference.startRecording({
mode: SariskaMediaTransport.constants.recording.mode.STREAM, // Set mode to "stream"
appData: JSON.stringify(appData) // Pass app data as JSON string
});
// Listen for RECORDER_STATE_CHANGED event to track streaming status
conference.addEventListener("RECORDER_STATE_CHANGED", (payload)=>{
// Verify mode is "stream" again
const mode = payload._mode;
// Get the live streaming session ID
const sessionId = payload._sessionID;
// Check the streaming status: on, off, or pending
const status = payload._status;
});
// Stop live streaming using the session ID
conference.stopRecording(sessionId);
Cloud Recording
Store your recordings and transcriptions in various cloud storage services.
// Configure for Object-based storage
const appData = {
file_recording_metadata : {
'share': true // Enable sharing
}
}
// Configure for Dropbox
const appData = {
file_recording_metadata: {
upload_credentials: {
service_name: "dropbox",
token: "your_dropbox_oauth_token"
}
}
}
// Start recording
conference.startRecording({
mode: SariskaMediaTransport.constants.recording.mode.FILE,
appData: JSON.stringify(appData)
});
// Monitor recording state
conference.addEventListener("RECORDER_STATE_CHANGED", (payload)=>{
const mode = payload._mode; // Recording mode (e.g., "file")
const sessionId = payload._sessionID; // Unique identifier for the cloud recording session
const status = payload._status; // Current recording status ("on", "off", or "pending")
// Handle recording state changes based on mode, sessionId, and status
});
// Stop recording
conference.stopRecording(sessionId); // Provide the session ID
PSTN
Dial-in(PSTN)
// Retrieve the phone pin and number for users to join via PSTN:
const phonePin = conference.getPhonePin(); // Get the phone pin for PSTN access
const phoneNumber = conference.getPhoneNumber() // Get the phone number for PSTN access
Dial-out(PSTN)
// Dial a phone number to invite a participant to the conference
conference.dial(phoneNumber)
Lobby/Waiting Room
// Join the lobby
conference.joinLobby(displayName, email); // Request to join the conference lobby
// Event listeners for lobby-related actions:
conference.addEventListener(SariskaMediaTransport.events.conference.LOBBY_USER_JOINED, (id, name) => {
// Handle events when a user joins the lobby
})
conference.addEventListener(SariskaMediaTransport.events.conference.LOBBY_USER_UPDATED, (id, participant)=> {
// Handle events when a user's information in the lobby is updated
})
// Additional event listeners for lobby actions:
conference.addEventListener(SariskaMediaTransport.events.conference.LOBBY_USER_LEFT, id=> {
})
conference.addEventListener(SariskaMediaTransport.events.conference.MEMBERS_ONLY_CHANGED, enabled=> {
})
// 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("[email protected]", "display name"); // Start a SIP video call with the specified address and display name
// Terminate a SIP video call
conference.stopSIPVideoCall('[email protected]'); // End the SIP video call with the specified address
// 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)
console.log("state", state);
})
// 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)
console.log("status", status);
})
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();
conference.stopP2PSession();
One-to-One Calling (React Native Only)
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
HTTP Call to Sariska Server
{API Method}
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
Reads Push Notification (using react-native-callkeep)
Processes the notification even if the app is closed or in the background.
HTTP Call to Update Status
{API Method}
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(SariskaMediaTransport.events.conference.USER_STATUS_CHANGED, (id, status) => {
// - id: receiver's user id
// - status: "ringing", "busy", "rejected", "connected", "expired"
});
Step 4 : After Connection Established
The call proceeds like a normal conference call.
React Native Libraries:
react-native-callkeep: Handles notifications and call events even when the app is closed or in the background.
react-native-incall-manager: Manages device events like headset state, proximity sensors, and audio routing.
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