Streaming Audio in Real Time

This section talks about streaming the audio in real-time using the Javascript SDK. We can use this API to pass in audio via a single stream and it also supports sending multiple isolated streams of audio, each of which can contain one or more speaker's audio data.


If you plan to use the multiple audio streams we recommend using single streams for each speaker involved to get the most accurate of transcription and speaker separation.

You can also consume the processed results in real-time, which include:

  1. Real Time Transcription
  2. Real Time Insights (Action Items and Questions)
  3. When using multiple audio streams (Each stream for 1 speaker) you also get access to speaker-separated data (including transcription and messages)

Example with Single Stream#

The example below utilises mic package to stream audio in real-time. This will be a single stream of audio obtained through mic which may have one or more than one speaker's audio. The link to the complete example below can be found here

Import required packages#

const {sdk} = require('symbl-node');
const uuid = require('uuid').v4;
// For demo purposes, we're using mic to simply get audio from microphone and pass it on to websocket connection
const mic = require('mic');

In the above snippet we import the sdk, uuid and mic npm packages. The uuid package is used for generating a unique ID to represent this stream and it's strongly recommended to use it. The mic package is used to obtain the audio stream in real-time to pass to the SDK.

Initialise an instance of mic#

const sampleRateHertz = 16000;
const micInstance = mic({
rate: sampleRateHertz,
channels: '1',
debug: false,
exitOnSilence: 6

We now declare the sampleRateHertz variable to specify the Sample Rate of the audio obtained from the mic. It is imperative to use the same Sample Rate used for initialising the mic package and for passing in to the startRealtimeRequest of Javascript SDK as we will see below. Otherwise the transcription will be completely in-accurate.

We also initialise mic with channels: '1' (mono channel) audio as currently only mono channel audio data is supported.

Initialise the Javascript SDK#

// Initialize the SDK
await sdk.init({
appId: '__appId__',
appSecret: '__appSecret__',
basePath: ''
// Need unique Id
const id = uuid();

Next we initialise a helper function to execute our code in the async/await style. The following code snippets (including the one just above) will be a part of the same function. We now initialise the Javascript SDK with the init call, passing in appId and appSecret which you can be obtain by signing up on Symbl Developer Platform We also initialise variable id with uuid function for the unique ID required for this stream as was also mentioned above in the import section snippet.

Call the startRealtimeRequest#

// Start Real-time Request (Uses Realtime WebSocket API behind the scenes)
const connection = await sdk.startRealtimeRequest({
insightTypes: ["action_item", "question"],
config: {
meetingTitle: 'My Test Meeting',
confidenceThreshold: 0.7,
timezoneOffset: 480, // Offset in minutes from UTC
languageCode: "en-US",
speaker: { // Optional, if not specified, will simply not send an email in the end.
userId: '', // Update with valid email
name: 'John'
handlers: {
'onSpeechDetected': (data) => {
// For live transcription
if (data) {
const {punctuated} = data;
console.log('Live: ', punctuated && punctuated.transcript);
'onMessageResponse': (data) => {
// When a processed message is available
console.log('onMessageResponse', JSON.stringify(data));
'onInsightResponse': (data) => {
// When an insight is detected
console.log('onInsightResponse', JSON.stringify(data));

The next call is made to startRealtimeRequest of the Javascript SDK and includes various parameters passed in. Lets breakdown the configuration and take a look at them one by one.

  1. id: The unique ID that represents this stream. (This needs to be unique, which is why we are using uuid)

  2. insightTypes: This array represents the type of insights that are to be detected. Today the supported ones are action_item and question.

  3. config: This configuration object encapsulates the properties which directly relate to the conversation generated by the audio being passed.

    a. meetingTitle : This optional parameter specifies the name of the conversation generated. You can get more info on conversations here

    b. confidenceThreshold : This optional parameter specifies the confidence threshold for detecting the insights. Only the insights that have confidenceScore more than this value will be returned.

    c. timezoneOffset : This specifies the actual timezoneOffset used for detecting the time/date related entities.

    d. languageCode : It specifies the language to be used for transcribing the audio in BCP-47 format. (Needs to be same as the language in which audio is spoken)

    e. sampleRateHertz : It specifies the sampleRate for this audio stream.

  4. speaker: Optionally specify the details of the speaker whose data is being passed in the stream. This enables an e-mail with the Summary UI URL to be sent after the end of the stream.

  5. handlers: This object has the callback functions for different events a. onSpeechDetected: To retrieve the real-time transcription results as soon as they are detected. We can use this callback to render live transcription which is specific to the speaker of this audio stream.

    b. onMessageResponse: This callback function contains the "finalized" transcription data for this speaker and if used with multiple streams with other speakers this callback would also provide their messages. The "finalized" messages mean that the ASR has finalised the state of this part of transcription and has declared it "final".

    c. onInsightResponse: This callback would provide with any of the detected insights in real-time as they are detected. As with the onMessageCallback above this would also return every speaker's insights in case of multiple streams.

Retrieve audio data from mic#

console.log('Successfully connected.');
const micInputStream = micInstance.getAudioStream();
micInputStream.on('data', (data) => {
// Push audio from Microphone to websocket connection
console.log('Started listening to Microphone.');

After the startRealtimeRequest returns successfully, it signifies that the connection has been established successfully with the passed configuration. In the above snippet we now obtain the audio data from the micInputStream and as it's received we relay it to the active connection instance we now have with Javascript SDK.

Stop the stream#

setTimeout(async () => {
// Stop listening to microphone
console.log('Stopped listening to Microphone.');
try {
// Stop connection
await conversationData = connection.stop();
console.log('Conversation ID: ' + conversationData.conversationId);
console.log('Connection Stopped.');
} catch (e) {
console.error('Error while stopping the connection.', e);
}, 60 * 1000); // Stop connection after 1 minute i.e. 60 secs

For the purpose of demoing a continuous audio stream we now simulate a stop on the above stream after 60 seconds. The connection.stop() would close the active connection and will trigger the optional email if the speaker config is included. Here the conversationData variable includes the conversationId you can use with the Conversation API to retrieve this conversation's data.

And that's it! This marks the completion of streaming audio in real-time (Single Audio Stream) with Javascript SDK. The complete code for the example explained above can be found here

With Multiple Streams#

The same example explained above can be deployed on multiple machines, each with one speaker to simulate the multiple streams use-case. The only thing common needs to be the unique ID created in the above example which is used to initialize startRealtimeRequest request.

Having this unique ID in common across all different ensures that the audio streams of all the speakers are bound the context of a single conversation. This conversation can be retrieved by the conversationId via the Conversation API which will include the data of all the speakers connecting using the same common ID.