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IClientOptions IClientOptions

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Table of contents

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Properties

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Properties

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anonymous_login

• Optional anonymous_login: Object anonymous_login login options

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Type declaration

NameTypeDescription
target_idstringThe target ID to use for the anonymous login. this is typically the ID of the AI assistant you want to connect to.
target_params?TargetParamsOptional parameters to pass to the target. These are forwarded to voice-sdk-proxy and mapped to custom headers on the SIP INVITE. See TargetParams
target_typestringA string indicating the target type, for now only ai_assistant is supported.
target_version_id?stringThe target version ID to use for the anonymous login. This is optional and can be used to specify a particular version of the AI assistant.

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callReportInterval

• Optional callReportInterval: number Interval in milliseconds for collecting call statistics. Stats are aggregated over each interval and stored locally until call end. Default 
5000 (5 seconds)

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debug

• Optional debug: boolean Enable debug mode for this client. This will gather WebRTC debugging information.

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debugOutput

• Optional debugOutput: "file" | "socket" Debug output option

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enableCallReports

• Optional enableCallReports: boolean Enable automatic call quality reporting to voice-sdk-proxy. When enabled, WebRTC stats are collected periodically during calls and posted to the voice-sdk-proxy /call_report endpoint when the call ends. Default 
true;

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env

• Optional env: Environment Environment to use for the connection. So far this property is only for internal purposes.

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forceRelayCandidate

• Optional forceRelayCandidate: boolean Force the use of a relay ICE candidate.

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iceServers

• Optional iceServers: RTCIceServer[] ICE Servers to use for all calls within the client connection. Overrides the default ones.

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keepConnectionAliveOnSocketClose

• Optional keepConnectionAliveOnSocketClose: boolean By passing keepConnectionAliveOnSocketClose as true, the SDK will attempt to keep Peer connection alive when the WebSocket connection is closed unexpectedly (e.g. network interruption, device sleep, etc).

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login

• Optional login: string The username to authenticate with your SIP Connection. login and password will take precedence over login_token for authentication.

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login_token

• Optional login_token: string The JSON Web Token (JWT) to authenticate with your SIP Connection. This is the recommended authentication strategy. See how to create one.

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mediaPermissionsRecovery

• Optional mediaPermissionsRecovery: Object Configuration for media permissions recovery on inbound calls. When enabled and the initial getUserMedia call fails while answering, the SDK emits a recoverable telnyx.error event with resume() and reject() callbacks so the app can prompt the user to fix permissions before the call fails. Recovery is attempted only for inbound calls. If the app calls resume(), the SDK retries getUserMedia. If the app calls reject() or does not respond before timeout, recovery fails and the call is terminated with the usual media error flow. Example 
import { isMediaRecoveryErrorEvent } from '@telnyx/webrtc';

const client = new TelnyxRTC({
  login_token: '...',
  mediaPermissionsRecovery: {
    enabled: true,
    timeout: 20000,
    onSuccess: () => console.log('Media recovered'),
    onError: (err) => console.error('Recovery failed', err),
  },
});

client.on('telnyx.error', (event) => {
  if (isMediaRecoveryErrorEvent(event)) {
    showPermissionDialog({
      onContinue: () => event.resume(),
      onCancel: () => event.reject?.(),
    });
  }
});

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Type declaration

NameTypeDescription
enabledbooleanEnable the recovery flow.
onError?(error: Error) => voidCalled when retry fails, the timeout expires, or the app calls reject().
onSuccess?() => voidCalled when the retry getUserMedia succeeds after resume().
timeoutnumberMaximum time in ms to wait for the app to call resume() or reject(). Recommended max 25000.

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mutedMicOnStart

• Optional mutedMicOnStart: boolean Disabled microphone by default when the call starts or adding a new audio source.

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password

• Optional password: string The password to authenticate with your SIP Connection.

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prefetchIceCandidates

• Optional prefetchIceCandidates: boolean Enable or disable prefetching ICE candidates. Defaults to true.

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region

• Optional region: string Region to use for the connection.

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ringbackFile

• Optional ringbackFile: string A URL to a wav/mp3 ringback file that will be used when you disable “Generate Ringback Tone” in your SIP Connection.

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ringtoneFile

• Optional ringtoneFile: string A URL to a wav/mp3 ringtone file.

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rtcIp

• Optional rtcIp: string RTC connection IP address to use instead of the default one. Useful when using a custom signaling server.

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rtcPort

• Optional rtcPort: number RTC connection port to use instead of the default one. Useful when using a custom signaling server.

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trickleIce

• Optional trickleIce: boolean Enable or disable Trickle ICE.

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useCanaryRtcServer

• Optional useCanaryRtcServer: boolean Use Telnyx’s Canary RTC server