Interface IClientOptions

IClientOptions IClientOptions

Hierarchy

  • IClientOptions

Properties

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

Properties

Optional anonymous_login

anonymous_login?: {
    target_id: string;
    target_type: string;
    target_version_id?: string;
}

anonymous_login login options

Type declaration

  • 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_type: string

    A string indicating the target type, for now only ai_assistant is supported.

  • Optional 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.

Optional callReportInterval

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)

Optional debug

debug?: boolean

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

Optional debugOutput

debugOutput?: "file" | "socket"

Debug output option

Optional enableCallReports

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

Optional env

env?: Environment

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

Optional forceRelayCandidate

forceRelayCandidate?: boolean

Force the use of a relay ICE candidate.

Optional iceServers

iceServers?: RTCIceServer[]

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

Optional keepConnectionAliveOnSocketClose

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

login?: string

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

Optional login_token

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.

Optional mutedMicOnStart

mutedMicOnStart?: boolean

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

Optional password

password?: string

The password to authenticate with your SIP Connection.

Optional prefetchIceCandidates

prefetchIceCandidates?: boolean

Enable or disable prefetching ICE candidates. Defaults to true.

Optional region

region?: string

Region to use for the connection.

Optional ringbackFile

ringbackFile?: string

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

Optional ringtoneFile

ringtoneFile?: string

A URL to a wav/mp3 ringtone file.

Optional rtcIp

rtcIp?: string

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

Optional rtcPort

rtcPort?: number

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

Optional trickleIce

trickleIce?: boolean

Enable or disable Trickle ICE.

Optional useCanaryRtcServer

useCanaryRtcServer?: boolean

Use Telnyx's Canary RTC server

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