import type { MediaTrackConstraints } from 'react-native-webrtc/lib/typescript/Constraints';
import { MediaStream } from 'react-native-webrtc';
export interface CallOptions {
    /**
     * The destination for the call.
     * This can be a phone number, SIP URI, or any other valid Telnyx destination.
     */
    destinationNumber?: string;
    /**
     * WebRTC audio constraints for the call.
     * This can be a boolean to enable/disable audio or an object with specific constraints.
     * @default true
     * @see https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints
     * @example
     * ```typescript
     * {
     *   audio: {
     *     echoCancellation: true,
     *     noiseSuppression: true,
     *     autoGainControl: true,
     *   },
     * }
     * ```
     * @example
     * ```typescript
     * {
     *   audio: true, // Enable audio with default constraints
     * }
     * ```
     */
    audio?: boolean | MediaTrackConstraints;
    /**
     * The caller ID name to display for the call.
     */
    callerIdName?: string;
    /**
     * The caller ID number to display for the call.
     */
    callerIdNumber?: string;
    /**
     * Custom headers to include in the call request.
     * This can be used to pass additional information or metadata with the call.
     * Each header should be an object with `name` and `value` properties.
     * @example
     * ```typescript
     * [
     *   { name: 'X-Custom-Header', value: 'CustomValue' },
     *   { name: 'X-Another-Header', value: 'AnotherValue' },
     * ]
     * ```
     */
    customHeaders?: {
        name: string;
        value: string;
    }[];
    /**
     * Client state to persist across calls.
     * This can be used to store any custom state information that should be maintained.
     * This is typically specified as base64 encoded string.
     * @example
     * ```typescript
     * btoa(JSON.stringify({ key: 'value' }))
     * ```
     * @see https://developer.mozilla.org/en-US/docs/Web/API/WindowOrWorkerGlobalScope/btoa
     * @see https://developer.mozilla.org/en-US/docs/Web/API/WindowOrWorkerGlobalScope/atob
     *
     */
    clientState?: string;
    /**
     * optional MediaStream for the local audio  tracks.
     */
    localStream?: MediaStream;
    /**
     * optional MediaStream for the remote audio tracks.
     * This is typically used to play the remote audio stream in the application.
     */
    remoteStream?: MediaStream;
    /**
     * Peer connection options for the WebRTC call.
     * These options can be used to customize the behavior of the WebRTC peer connection.
     * For example, you can specify ICE servers, transport policies, and more.
     */
    peerConnectionOptions?: {
        prefetchIceCandidates?: boolean;
        /**
         * The ICE servers to use for the WebRTC connection.
         * This can include STUN and TURN servers.
         * @example
         * ```typescript
         * [
         *   { urls: 'stun:stun.l.google.com:19302' },
         *   { urls: 'turn:turn.example.com', username: 'user', credential: 'password' },
         * ]
         * ```
         */
        iceServers?: RTCIceServer[];
        /**
         * The ICE transport policy to use for the WebRTC connection.
         * This can be 'all', 'relay', 'nohost', or 'none'.
         * @default 'all'
         */
        iceTransportPolicy?: RTCIceTransportPolicy;
        /**
         * The bundle policy to use for the WebRTC connection.
         * This can be 'balanced', 'max-compat', or 'max-bundle'.
         * @default 'balanced'
         */
        bundlePolicy?: RTCBundlePolicy;
        /**
         * The RTCP mux policy to use for the WebRTC connection.
         * This can be 'require' or 'negotiate'.
         * @default 'require'
         */
        rtcpMuxPolicy?: RTCRtcpMuxPolicy;
    };
    remoteCallerIdName?: string;
    remoteCallerIdNumber?: string;
}