IClientOptions IClientOptions

• anonymous_login
• callReportInterval
• debug
• debugOutput
• enableCallReports
• env
• forceRelayCandidate
• iceServers
• keepConnectionAliveOnSocketClose
• login
• login_token
• mutedMicOnStart
• password
• prefetchIceCandidates
• region
• ringbackFile
• ringtoneFile
• rtcIp
• rtcPort
• trickleIce
• useCanaryRtcServer

• Optional anonymous_login: Object

anonymous_login login options

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. 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.

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• 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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• Optional debug: boolean

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

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• Optional debugOutput: "file" | "socket"

Debug output option

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• 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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• Optional env: Environment

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

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• Optional forceRelayCandidate: boolean

Force the use of a relay ICE candidate.

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• Optional iceServers: RTCIceServer[]

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

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• 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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• 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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• 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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• Optional mutedMicOnStart: boolean

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

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• Optional password: string

The password to authenticate with your SIP Connection.

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• Optional prefetchIceCandidates: boolean

Enable or disable prefetching ICE candidates. Defaults to true.

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• Optional region: string

Region to use for the connection.

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• 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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• Optional ringtoneFile: string

A URL to a wav/mp3 ringtone file.

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• 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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• Optional rtcPort: number

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

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• Optional trickleIce: boolean

Enable or disable Trickle ICE.

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• Optional useCanaryRtcServer: boolean

Use Telnyx's Canary RTC server