Documentation Index
Fetch the complete documentation index at: https://docs.cognigy.com/llms.txt
Use this file to discover all available pages before exploring further.
This page documents all TypeScript types and interfaces exported by the Cognigy Click To Call SDK.
WebRTCClientConfig
The configuration object passed to createWebRTCClient().
interface WebRTCClientConfig {
endpointUrl: string;
userId?: string;
captureAudio?: boolean;
pcConfig?: RTCConfiguration;
}
| Property | Type | Description | Required |
|---|
endpointUrl | string | URL to fetch SIP configuration from the Cognigy Voice Gateway Endpoint. | Yes |
userId | string | Optional user identifier for the session. | No |
captureAudio | boolean | When set to true, the SDK emits the captureAudio event with the remote MediaStream during a call. Defaults to false. | No |
pcConfig | RTCConfiguration | Custom WebRTC peer connection configuration. Use this to specify STUN/TURN servers or other RTCPeerConnection settings. | No |
WebRTCClient
The main client interface returned by createWebRTCClient(). Provides all call control, state query, and event handling methods.
| Method | Reference |
|---|
connect() | connect |
disconnect() | disconnect |
connectAndCall() | connectAndCall |
startCall() | startCall |
endCall() | endCall |
mute() | mute |
unmute() | unmute |
sendInfo() | sendInfo |
isConnected() | isConnected |
getCurrentSession() | getCurrentSession |
on() | on |
off() | off |
destroy() | destroy |
CallSession
Represents an active call session.
interface CallSession {
id: string;
status: SessionStatus;
direction: 'incoming' | 'outgoing';
startTime: Date;
answerTime?: Date;
duration: number;
muted: boolean;
localHold: boolean;
remoteHold: boolean;
}
| Property | Type | Description |
|---|
id | string | Unique identifier for the call session. |
status | SessionStatus | Current session status: 'init', 'ringing', 'answered', 'failed', or 'ended'. |
direction | 'incoming' | 'outgoing' | Whether the call was initiated locally or received from a remote party. |
startTime | Date | Timestamp when the session was created. |
answerTime | Date | undefined | Timestamp when the call was answered, or undefined if not yet answered. |
duration | number | Call duration in seconds since the call was answered. |
muted | boolean | Whether the local microphone is muted. |
localHold | boolean | Whether the call is on hold locally. |
remoteHold | boolean | Whether the call is on hold by the remote party. |
SessionStatus
type SessionStatus = 'init' | 'ringing' | 'answered' | 'failed' | 'ended';
CallEndInfo
Contains information about why a call ended.
interface CallEndInfo {
originator: 'local' | 'remote' | null;
cause: string | null;
description?: string | null;
}
| Property | Type | Description |
|---|
originator | 'local' | 'remote' | null | Who initiated the call end — the local user, the remote party, or null if unknown. |
cause | string | null | The reason the call ended, for example, Terminated, Canceled, Rejected. |
description | string | null | undefined | Additional description, such as a SIP status code or SIP Reason header value. |
Transcription Data
The transcription event emits the value of the _transcription property from incoming SIP INFO messages. The structure depends on your Cognigy Voice Gateway configuration, but typically has the following shape:
interface TranscriptionData {
messages: { text: string }[];
originator: 'user' | 'bot';
}
| Property | Type | Description |
|---|
originator | string | Indicates the source of the transcription, for example, 'user' or 'bot'. |
messages | { text: string }[] | Array of message objects, each containing a text field. |
This is not a formally defined SDK interface. The transcription data is passed through as-is from the SIP INFO message body. The actual structure depends on your Voice Gateway configuration.
WebRTCClientEvents
Type map of all event names to their callback signatures. Use this for type-safe event handling.
The following events are emitted by the client:
| Event | Callback Signature |
|---|
connecting | () => void |
connected | () => void |
disconnected | () => void |
registered | () => void |
unregistered | () => void |
ringing | (session: CallSession) => void |
answered | (session: CallSession) => void |
ended | (session: CallSession, endInfo: CallEndInfo) => void |
failed | (session: CallSession, endInfo: CallEndInfo) => void |
muted | (session: CallSession) => void |
unmuted | (session: CallSession) => void |
audioEnded | () => void |
infoSent | (text: string, data: Record<string, any>) => void |
infoReceived | (data: { originator: string, info: { body: string } }) => void |
transcription | (transcription: { originator: string; messages: { text: string }[] }) => void |
error | (error: Error) => void |
captureAudio | (stream: MediaStream) => void |
import type { WebRTCClientEvents } from '@cognigy/click-to-call-sdk';
const handleAnswered: WebRTCClientEvents['answered'] = (session) => {
console.log('Call answered:', session.id);
};
client.on('answered', handleAnswered);
TypeScript Usage
The SDK is written in TypeScript and ships comprehensive type definitions. All interfaces and types can be imported directly:
import type {
WebRTCClient,
CallSession,
CallEndInfo,
WebRTCClientEvents
} from '@cognigy/click-to-call-sdk';
Type-Safe Event Handling
Use the WebRTCClientEvents type map for fully typed event callbacks:
const handleAnswered: WebRTCClientEvents['answered'] = (session) => {
console.log('Call answered:', session.id);
};
client.on('answered', handleAnswered);
Type-Safe Info Messages
When sending structured data via sendInfo(), you can define your data shape:
interface MyInfoData {
action: string;
payload: { key: string; value: number };
}
const data: MyInfoData = {
action: 'update',
payload: { key: 'score', value: 42 }
};
await client.sendInfo('custom-action', data);
