@videosdk.live/server-sdk
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    Interface CreateSipCallOptions

    Options for SipCallsResource.create.

    When the call originates from a provisioned phone number (callFrom is a provisioned number, typically together with routingRuleId), recordAudio, dtmf, hidePhoneNumber, headers, includeHeaders and mediaEncryption are taken from the matching routing rule and gateway, so the per-call values here are ignored. When the call is placed directly through an outbound gateway (gatewayId), these per-call values apply.

    interface CreateSipCallOptions {
        callFrom?: string;
        callTo?: string;
        dtmf?: boolean;
        gatewayId?: string;
        headers?: Record<string, string>;
        hidePhoneNumber?: boolean;
        includeHeaders?: SipIncludeHeaders;
        maxCallDuration?: number;
        mediaEncryption?: SipMediaEncryption;
        metadata?: Record<string, unknown>;
        participant?: Record<string, unknown>;
        participantId?: string;
        recordAudio?: boolean;
        ringingTimeout?: number;
        roomId?: string;
        routingRuleId?: string;
        waitUntilAnswered?: boolean;
    }
    Index
    callFrom?: string

    Source number. Required unless derivable from gatewayId.

    callTo?: string

    Destination number. Required.

    dtmf?: boolean
    gatewayId?: string

    Outbound gateway to place the call through.

    headers?: Record<string, string>
    hidePhoneNumber?: boolean
    includeHeaders?: SipIncludeHeaders
    maxCallDuration?: number
    mediaEncryption?: SipMediaEncryption
    metadata?: Record<string, unknown>
    participant?: Record<string, unknown>
    participantId?: string

    Id to assign the call's participant in the destination room.

    recordAudio?: boolean
    ringingTimeout?: number
    roomId?: string

    Destination room.

    routingRuleId?: string

    Outbound routing rule to place the call through.

    waitUntilAnswered?: boolean