WebRTC JS Client Options

IClientOptions IClientOptions

Properties

anonymous_login
callReportFlushInterval
callReportInterval
debug
debugOutput
enableCallReports
env
forceRelayCandidate
hangupOnBeforeUnload
iceServers
keepConnectionAliveOnSocketClose
login
login_token
maxReconnectAttempts
mediaPermissionsRecovery
mutedMicOnStart
password
prefetchIceCandidates
region
ringbackFile
ringtoneFile
rtcIp
rtcPort
skipLastVoiceSdkId
skipTrailing
trickleIce
useCanaryRtcServer

Properties

anonymous_login

• Optional anonymous_login: Object anonymous_login login options

Type declaration

Name	Type	Description
`target_id`	`string`	The target ID to use for the anonymous login. this is typically the ID of the AI assistant you want to connect to.
`target_params?`	`TargetParams`	Optional parameters to pass to the target. These are forwarded to voice-sdk-proxy and mapped to custom headers on the SIP INVITE. Use `target_params.conversation_id` only to join an existing Telnyx AI conversation; omit it to start a new conversation. `See` TargetParams
`target_type`	`string`	A string indicating the target type, for now only `ai_assistant` is supported.
`target_version_id?`	`string`	The 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.

callReportFlushInterval

• Optional callReportFlushInterval: number Interval in milliseconds for submitting intermediate call reports while a call is active. Set to 0 to disable time-based intermediate reports. Default

180000 (3 minutes)

callReportInterval

• Optional callReportInterval: number Interval in milliseconds for collecting call statistics after the initial high-resolution startup window. Stats are aggregated over each interval and submitted as intermediate reports while the call is active. Default

5000 (5 seconds)

debug

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

debugOutput

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

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;

env

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

forceRelayCandidate

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

hangupOnBeforeUnload

• Optional hangupOnBeforeUnload: boolean Controls whether the SDK attempts to send BYE for active calls during browser page unload. Enabled by default to preserve graceful call cleanup, but applications that handle page lifecycle themselves can disable it to avoid best-effort unload BYE races. Default

true;

iceServers

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

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).

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

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.

maxReconnectAttempts

• Optional maxReconnectAttempts: number Maximum number of automatic socket reconnection attempts after an unexpected disconnect. When the limit is reached, no further automatic reconnects are scheduled and a telnyx.error event with code RECONNECTION_EXHAUSTED (45003) is emitted. A manual connect() call resets the counter and starts a fresh retry sequence. Set to 0 to allow unlimited automatic reconnect attempts. When omitted, defaults to 10. Default

10;

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. In the recovery flow the emitted error carries event.recoverable === true (top-level) plus the resume/reject/retryDeadline helpers, and event.error.fatal === false (the SDK is actively handling recovery via the app’s cooperation). Outside the recovery flow, media failures are terminal (event.error.fatal === true, no top-level recoverable). 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)) {
    // event.recoverable === true, event.error.fatal === false
    showPermissionDialog({
      onContinue: () => event.resume(),
      onCancel: () => event.reject?.(),
    });
  } else if (event.error.fatal) {
    // Terminal error — give up on this call/session
  }
});

Type declaration

Name	Type	Description
`enabled`	`boolean`	Enable the recovery flow.
`onError?`	(`error`: `Error`) => `void`	Called when retry fails, the timeout expires, or the app calls `reject()`.
`onSuccess?`	() => `void`	Called when the retry `getUserMedia` succeeds after `resume()`.
`timeout`	`number`	Maximum time in ms to wait for the app to call `resume()` or `reject()`. Recommended max 25000.

mutedMicOnStart

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

password

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

prefetchIceCandidates

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

region

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

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.

ringtoneFile

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

rtcIp

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

rtcPort

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

skipLastVoiceSdkId

• Optional skipLastVoiceSdkId: boolean 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;

skipTrailing

• Optional skipTrailing: boolean When set to true, appends skip_trailing=true to the VSP WebSocket URL so VSP skips pre-routing identity resolution (telephony-tokens validation and UsersClass trailing checks) for this connection. This is intended for internal/test-infra usage (e.g. BBT-generated credentials) where the connection should not participate in trailing release routing. The actual login still goes to the upstream RTC service for normal authentication — this only skips VSP’s pre-routing lookup used for trailing target selection. Default

false;

trickleIce

• Optional trickleIce: boolean Enable or disable Trickle ICE.

useCanaryRtcServer

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

​Table of contents

​Properties

​Properties

​anonymous_login

​Type declaration

​callReportFlushInterval

​callReportInterval

​debug

​debugOutput

​enableCallReports

​env

​forceRelayCandidate

​hangupOnBeforeUnload

​iceServers

​keepConnectionAliveOnSocketClose

​login

​login_token

​maxReconnectAttempts

​mediaPermissionsRecovery

​Type declaration

​mutedMicOnStart

​password

​prefetchIceCandidates

​region

​ringbackFile

​ringtoneFile

​rtcIp

​rtcPort

​skipLastVoiceSdkId

​skipTrailing

​trickleIce

​useCanaryRtcServer

Table of contents

Properties

Properties

anonymous_login

Type declaration

callReportFlushInterval

callReportInterval

debug

debugOutput

enableCallReports

env

forceRelayCandidate

hangupOnBeforeUnload

iceServers

keepConnectionAliveOnSocketClose

login

login_token

maxReconnectAttempts

mediaPermissionsRecovery

Type declaration

mutedMicOnStart

password

prefetchIceCandidates

region

ringbackFile

ringtoneFile

rtcIp

rtcPort

skipLastVoiceSdkId

skipTrailing

trickleIce

useCanaryRtcServer