Interface IClientOptions

anonymous_login login options

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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Enable debug mode for this client. This will gather WebRTC debugging information.

Debug output option

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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Environment to use for the connection. So far this property is only for internal purposes.

Force the use of a relay ICE candidate.

ICE Servers to use for all calls within the client connection. Overrides the default ones.

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).

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

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

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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Disabled microphone by default when the call starts or adding a new audio source.

The password to authenticate with your SIP Connection.

Enable or disable prefetching ICE candidates. Defaults to true.

Region to use for the connection.

A URL to a wav/mp3 ringback file that will be used when you disable "Generate Ringback Tone" in your SIP Connection.

A URL to a wav/mp3 ringtone file.

RTC connection IP address to use instead of the default one. Useful when using a custom signaling server.

RTC connection port to use instead of the default one. Useful when using a custom signaling server.

When reconnecting with a stored voice_sdk_id, append ?skip_last_voice_sdk_id=true to the WebSocket URL so VSP routes the connection to a different b2bua-rtc instance instead of sticky- reconnecting to the same one. Useful when retrying after errors caused by stale state on a specific b2bua-rtc node.

Default

false
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Enable or disable Trickle ICE.

Use Telnyx's Canary RTC server