@azure/communication-call-automation/dist/react-native/models/models.d.ts

Azure client library for Azure Communication Call Automation services

import type { CommunicationIdentifier, CommunicationUserIdentifier, MicrosoftTeamsUserIdentifier, MicrosoftTeamsAppIdentifier, PhoneNumberIdentifier } from "@azure/communication-common";
import type { CallConnectionStateModel, MediaStreamingSubscription, TranscriptionSubscription, MediaStreamingAudioChannelType, MediaStreamingContentType, AudioFormat } from "../generated/src/index.js";
export { CallConnectionStateModel, CallRejectReason, KnownCallRejectReason, KnownMediaStreamingAudioChannelType, KnownMediaStreamingContentType, KnownStreamingTransportType, RecognitionType, ChoiceResult, DtmfResult, SpeechResult, RecordingState, Tone, MediaStreamingAudioChannelType, MediaStreamingContentType, AudioFormat, TranscriptionUpdate, MediaStreamingUpdate, } from "../generated/src/models/index.js";
/** Properties of a call connection */
export interface CallConnectionProperties {
    /** The call connection id. */
    callConnectionId?: string;
    /** The server call id. */
    serverCallId?: string;
    /**
     * The source caller Id, a phone number, that's shown to the PSTN participant being invited.
     * Required only when calling a PSTN callee.
     */
    sourceCallerIdNumber?: PhoneNumberIdentifier;
    /** Display name of the call if dialing out to a pstn number. */
    sourceDisplayName?: string;
    /** Source identity. */
    source?: CommunicationIdentifier;
    /** The targets of the call. */
    targetParticipants?: CommunicationIdentifier[];
    /** The state of the call connection. */
    callConnectionState?: CallConnectionStateModel;
    /** The callback URL. */
    callbackUrl?: string;
    /** The correlation ID. */
    correlationId?: string;
    /** Identity of the answering entity. Only populated when identity is provided in the request. */
    answeredby?: CommunicationUserIdentifier;
    /** Identity of the original Pstn target of an incoming Call. Only populated when the original target is a Pstn number. */
    answeredFor?: PhoneNumberIdentifier;
    /** Media streaming subscription */
    mediaStreamingSubscription?: MediaStreamingSubscription;
    /** Transcription Subscription. */
    transcriptionSubscription?: TranscriptionSubscription;
}
/** Contract model of an ACS call participant */
export interface CallParticipant {
    /** Communication identifier of the participant */
    identifier?: CommunicationIdentifier;
    /** Is participant muted */
    isMuted?: boolean;
    /** Is participant on hold. */
    isOnHold?: boolean;
}
/** The locator used for joining or taking action on a call. */
export interface CallLocator {
    id: string;
    kind: CallLocatorType;
}
/** Defines values for VoiceKind that the service accepts. */
export declare enum VoiceKind {
    /** Male */
    Male = "male",
    /** Female */
    Female = "female"
}
/** The PlaySource model. */
export interface PlaySource {
    /** @deprecated Not in use, instead use playsourcecacheid for similar functionality*/
    playsourcacheid?: string;
    /** Sets the play source cache id.*/
    playSourceCacheId?: string;
}
/** The FileSource model. */
export interface FileSource extends PlaySource {
    url: string;
    readonly kind: "fileSource";
}
/** The TextSource model. */
export interface TextSource extends PlaySource {
    text: string;
    sourceLocale?: string;
    voiceKind?: VoiceKind;
    voiceName?: string;
    customVoiceEndpointId?: string;
    readonly kind: "textSource";
}
/** The SsmlSource model. */
export interface SsmlSource extends PlaySource {
    ssmlText: string;
    customVoiceEndpointId?: string;
    readonly kind: "ssmlSource";
}
/** A Dtmf Tone. */
export declare enum DtmfTone {
    /** Zero */
    Zero = "zero",
    /** One */
    One = "one",
    /** Two */
    Two = "two",
    /** Three */
    Three = "three",
    /** Four */
    Four = "four",
    /** Five */
    Five = "five",
    /** Six */
    Six = "six",
    /** Seven */
    Seven = "seven",
    /** Eight */
    Eight = "eight",
    /** Nine */
    Nine = "nine",
    /** A */
    A = "a",
    /** B */
    B = "b",
    /** C */
    C = "c",
    /** D */
    D = "d",
    /** Pound */
    Pound = "pound",
    /** Asterisk */
    Asterisk = "asterisk"
}
/** A Recognition Choice */
export interface RecognitionChoice {
    /** Identifier for a given choice */
    label: string;
    /** List of phrases to recognize */
    phrases: string[];
    tone?: DtmfTone;
}
/** The type of the recognition that the service accepts. */
export declare enum RecognizeInputType {
    /** Dtmf */
    Dtmf = "dtmf"
}
/** Call invitee details. */
export interface CallInvite {
    /** The Target's PhoneNumberIdentifier, CommunicationUserIdentifier, MicrosoftTeamsUserIdentifier or MicrosoftTeamsAppIdentifier. */
    readonly targetParticipant: PhoneNumberIdentifier | CommunicationUserIdentifier | MicrosoftTeamsUserIdentifier | MicrosoftTeamsAppIdentifier;
    /** Caller's phone number identifier. */
    readonly sourceCallIdNumber?: PhoneNumberIdentifier;
    sourceDisplayName?: string;
    /** Used by customer to send custom context to targets. */
    customCallingContext?: CustomCallingContext;
}
/** The locator type of a call. */
export type CallLocatorType = "serverCallLocator" | "groupCallLocator" | "roomCallLocator";
/** The content type of a call recording. */
export type RecordingContent = "audio" | "audioVideo";
/** The channel type of a call recording. */
export type RecordingChannel = "mixed" | "unmixed";
/** The format type of a call recording. */
export type RecordingFormat = "mp3" | "mp4" | "wav";
/** The format type of a call recording. */
export type RecordingKind = "azureCommunicationServices" | "teams" | "teamsCompliance";
/** The storage type of a call recording. */
export type RecordingStorageKind = "azureCommunicationServices" | "azureBlobStorage";
/** Channel affinity for a participant */
export interface ChannelAffinity {
    /** Channel number to which bitstream from a particular participant will be written. */
    channel?: number;
    /**
     * The identifier for the participant whose bitstream will be written to the channel
     * represented by the channel number.
     */
    targetParticipant: CommunicationIdentifier;
}
/** The recording storage */
export interface RecordingStorage {
    /** Defines the kind of recording storage */
    recordingStorageKind: RecordingStorageKind;
    /** Uri of a container or a location within a container */
    recordingDestinationContainerUrl?: string;
}
interface CustomCallingContextHeader {
    key: string;
    value: string;
}
/** VOIP header. */
export interface VoipHeader extends CustomCallingContextHeader {
    kind: "voip";
}
/** SIP User To User header. */
export interface SipUserToUserHeader extends CustomCallingContextHeader {
    kind: "sipuui";
}
/** SIP Custom header. */
export interface SipCustomHeader extends CustomCallingContextHeader {
    kind: "sipx";
}
/** Custom Calling Context */
export type CustomCallingContext = (VoipHeader | SipUserToUserHeader | SipCustomHeader)[];
/** AI options for the call. */
export interface CallIntelligenceOptions {
    /** The identifier of the Cognitive Service resource assigned to this call. */
    cognitiveServicesEndpoint?: string;
}
/** Options for media streaming. */
export interface MediaStreamingOptions {
    /** Polymorphic discriminator, which specifies the different types this object can be */
    transportType: "websocket";
    /** The audio channel type to stream, e.g., unmixed audio, mixed audio. */
    audioChannelType: MediaStreamingAudioChannelType;
    /** The transport URL for media streaming. */
    transportUrl?: string;
    contentType?: MediaStreamingContentType;
    /** A value indicating whether the media streaming should start immediately after the call is answered. */
    startMediaStreaming?: boolean;
    /** A value indicating whether bidirectional streaming is enabled. */
    enableBidirectional?: boolean;
    /** The audio format used for encoding, including sample rate and channel type. The default is Pcm16KMono. */
    audioFormat?: AudioFormat;
    /** A value that indicates whether to stream the DTMF tones. */
    enableDtmfTones?: boolean;
}
/** Options for media streaming. */
export interface TranscriptionOptions {
    /** Specifies the Locale used for transcription, e.g., en-CA or en-AU. */
    locale: string;
    /** Polymorphic discriminator, which specifies the different types this object can be */
    transportType: "websocket";
    /** The URL used for live transcription transport. */
    transportUrl?: string;
    /** The ID of the deployed custom model in GUID format. The GUID is generated by Azure Speech Studio, e.g., a259c255-1cdw-4ed7-a693-dd58563b6f6a. */
    speechModelEndpointId?: string;
    /** Indicates whether the transcription should start immediately after the call is answered. */
    startTranscription?: boolean;
    /** Enables intermediate results for the transcribed speech. */
    enableIntermediateResults?: boolean;
}
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