@signalwire/js 3.29.2 → 4.0.0-beta.1

package/dist/webrtc/src/utils/getDisplayMedia.d.ts

@@ -1,17 +0,0 @@
-/**
- * Prompts the user to share the screen and asynchronously returns a
- * [MediaStream](https://developer.mozilla.org/en-US/docs/Web/API/MediaStream)
- * object associated with a display or part of it.
- *
- * @param constraints an optional
- * [MediaStreamConstraints](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamConstraints)
- * object specifying requirements for the returned [MediaStream](https://developer.mozilla.org/en-US/docs/Web/API/MediaStream).
- *
- * @example
- *  ```typescript
- * await SignalWire.WebRTC.getDisplayMedia()
- * // MediaStream {id: "HCXy...", active: true, ...}
- * ```
- */
-export declare const getDisplayMedia: (constraints?: MediaStreamConstraints) => Promise<MediaStream>;
-//# sourceMappingURL=getDisplayMedia.d.ts.map

package/dist/webrtc/src/utils/getDisplayMedia.d.ts.map

@@ -1 +0,0 @@
-{"version":3,"file":"getDisplayMedia.d.ts","sourceRoot":"","sources":["../../../../../webrtc/src/utils/getDisplayMedia.ts"],"names":[],"mappings":"AAEA;;;;;;;;;;;;;;GAcG;AACH,eAAO,MAAM,eAAe,GAAI,cAAc,sBAAsB,yBAEnE,CAAA"}

package/dist/webrtc/src/utils/getUserMedia.d.ts

@@ -1,35 +0,0 @@
-/**
- * Prompts the user to share one or more media devices and asynchronously
- * returns an associated [MediaStream](https://developer.mozilla.org/en-US/docs/Web/API/MediaStream)
- * object.
- *
- * For more information, see [`MediaDevices.getUserMedia()`](https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia).
- *
- * @param constraints an optional [MediaStreamConstraints](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamConstraints)
- *                    object specifying requirements for the returned [MediaStream](https://developer.mozilla.org/en-US/docs/Web/API/MediaStream).
- *
- * @example
- * To only request audio media:
- *
- * ```typescript
- * await SignalWire.WebRTC.getUserMedia({audio: true, video: false})
- * // MediaStream {id: "HCXy...", active: true, ...}
- * ```
- *
- * @example
- * To request both audio and video, specifying constraints for the video:
- *
- * ```typescript
- * const constraints = {
- *   audio: true,
- *   video: {
- *     width: { min: 1024, ideal: 1280, max: 1920 },
- *     height: { min: 576, ideal: 720, max: 1080 }
- *   }
- * }
- * await SignalWire.WebRTC.getUserMedia(constraints)
- * // MediaStream {id: "EDVk...", active: true, ...}
- * ```
- */
-export declare const getUserMedia: (constraints?: MediaStreamConstraints) => Promise<MediaStream>;
-//# sourceMappingURL=getUserMedia.d.ts.map

package/dist/webrtc/src/utils/getUserMedia.d.ts.map

@@ -1 +0,0 @@
-{"version":3,"file":"getUserMedia.d.ts","sourceRoot":"","sources":["../../../../../webrtc/src/utils/getUserMedia.ts"],"names":[],"mappings":"AAoCA;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;GAgCG;AACH,eAAO,MAAM,YAAY,GACvB,cAAa,sBAAqD,yBAuEnE,CAAA"}

package/dist/webrtc/src/utils/helpers.d.ts

@@ -1,18 +0,0 @@
-import { ConnectionOptions } from './interfaces';
-export declare const getUserMedia: (constraints: MediaStreamConstraints) => Promise<MediaStream> | undefined;
-export declare const getMediaConstraints: (options: ConnectionOptions) => Promise<MediaStreamConstraints>;
-interface FilterIceServersOptions {
-    disableUdpIceServers?: boolean;
-}
-export declare const filterIceServers: (servers: RTCIceServer[], options: FilterIceServersOptions) => {
-    urls: string | string[];
-    credential?: string;
-    username?: string;
-}[];
-/**
- * Check if the negotiation is single media (audio or video only).
- * This is used to determine if we should handle SDP negotiation differently.
- */
-export declare const isSingleMediaNegotiation: (options: ConnectionOptions) => boolean;
-export {};
-//# sourceMappingURL=helpers.d.ts.map

package/dist/webrtc/src/utils/helpers.d.ts.map

@@ -1 +0,0 @@
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package/dist/webrtc/src/utils/helpers.test.d.ts

@@ -1,2 +0,0 @@
-export {};
-//# sourceMappingURL=helpers.test.d.ts.map

package/dist/webrtc/src/utils/helpers.test.d.ts.map

@@ -1 +0,0 @@
-{"version":3,"file":"helpers.test.d.ts","sourceRoot":"","sources":["../../../../../webrtc/src/utils/helpers.test.ts"],"names":[],"mappings":""}

package/dist/webrtc/src/utils/index.d.ts

@@ -1,8 +0,0 @@
-export * from './primitives';
-export * from './enumerateDevices';
-export * from './getUserMedia';
-export * from './getDisplayMedia';
-export * from './permissions';
-export * from './requestPermissions';
-export * from './sdpHelpers';
-//# sourceMappingURL=index.d.ts.map

package/dist/webrtc/src/utils/index.d.ts.map

@@ -1 +0,0 @@
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package/dist/webrtc/src/utils/interfaces.d.ts

@@ -1,139 +0,0 @@
-import type { VideoPositions } from '@signalwire/core';
-import { BaseConnectionState, VideoRoomDeviceEventParams, VideoRoomDeviceEventNames } from '@signalwire/core';
-export interface ConnectionOptions {
-    /** @internal */
-    destinationNumber?: string;
-    /** @internal */
-    remoteCallerName?: string;
-    /** @internal */
-    remoteCallerNumber?: string;
-    /** @internal */
-    callerName?: string;
-    /** @internal */
-    callerNumber?: string;
-    /** @internal */
-    remoteSdp?: string;
-    /** @internal */
-    localStream?: MediaStream;
-    /** @internal */
-    remoteStream?: MediaStream;
-    /** List of ICE servers. */
-    iceServers?: RTCIceServer[];
-    /** Disable ICE UDP transport policy */
-    disableUdpIceServers?: boolean;
-    /** Audio constraints to use when joining the room. Default: `true`. */
-    audio?: MediaStreamConstraints['audio'];
-    /** Video constraints to use when joining the room. Default: `true`. */
-    video?: MediaStreamConstraints['video'];
-    /** @internal */
-    attach?: boolean;
-    /** @internal */
-    useStereo?: boolean;
-    /** @internal */
-    micId?: string;
-    /** @internal */
-    micLabel?: string;
-    /** @internal */
-    camId?: string;
-    /** @internal */
-    camLabel?: string;
-    /** Id of the speaker device to use for audio output. If undefined, picks a default speaker. */
-    speakerId?: string;
-    /** @internal */
-    userVariables?: {
-        [key: string]: any;
-    };
-    /** @internal */
-    screenShare?: boolean;
-    /** @internal */
-    additionalDevice?: boolean;
-    /** @internal */
-    recoverCall?: boolean;
-    /** @internal */
-    googleMaxBitrate?: number;
-    /** @internal */
-    googleMinBitrate?: number;
-    /** @internal */
-    googleStartBitrate?: number;
-    /** @internal */
-    negotiateAudio?: boolean;
-    /** @internal */
-    negotiateVideo?: boolean;
-    /** @internal */
-    sfu?: boolean;
-    /** @internal */
-    simulcast?: boolean;
-    /** @internal */
-    msStreamsNumber?: number;
-    /** @internal */
-    requestTimeout?: number;
-    /** @internal */
-    shakenCheck?: string;
-    /** @internal */
-    shakenResult?: string;
-    /** @internal */
-    autoApplyMediaParams?: boolean;
-    /** @internal */
-    rtcPeerConfig?: {
-        [key: string]: any;
-    };
-    /** @internal */
-    iceGatheringTimeout?: number;
-    /** @internal */
-    maxIceGatheringTimeout?: number;
-    /** @internal */
-    maxConnectionStateTimeout?: number;
-    /** @internal */
-    watchMediaPackets?: boolean;
-    /** @internal */
-    watchMediaPacketsTimeout?: number;
-    /** @internal */
-    pingSupported?: boolean;
-    /** @internal */
-    prevCallId?: string;
-    /** @internal */
-    nodeId?: string;
-    /** @internal */
-    fromFabricAddressId?: string;
-    layout?: string;
-    positions?: VideoPositions;
-    /**
-     * Number of connections to maintain in the connection pool.
-     * Set to 0 to disable connection pooling. Default: 3
-     * @internal
-     */
-    connectionPoolSize?: number;
-    /**
-     * Whether to enable connection pooling for faster call setup.
-     * When enabled, maintains pre-warmed RTCPeerConnections with gathered ICE candidates.
-     * Default: true
-     * @internal
-     */
-    enableConnectionPool?: boolean;
-    /**
-     * Size of the ICE candidate pool for reuse.
-     * Allows ICE candidates to be reused when tracks are replaced.
-     * Default: 10
-     * @internal
-     */
-    iceCandidatePoolSize?: number;
-}
-export interface EmitDeviceUpdatedEventsParams {
-    newTrack: MediaStreamTrack;
-    prevAudioTrack?: MediaStreamTrack | null;
-    prevVideoTrack?: MediaStreamTrack | null;
-}
-export type UpdateMediaOptionsParams = Pick<ConnectionOptions, 'video' | 'audio' | 'negotiateVideo' | 'negotiateAudio'>;
-export interface OnVertoByeParams {
-    byeCause: string;
-    byeCauseCode: string;
-    rtcPeerId: string;
-    redirectDestination?: string;
-}
-export type MediaEventNames = 'media.connected' | 'media.reconnecting' | 'media.disconnected';
-type BaseConnectionEventsHandlerMap = Record<BaseConnectionState, (params: any) => void> & Record<MediaEventNames, () => void> & Record<VideoRoomDeviceEventNames, (params: VideoRoomDeviceEventParams) => void>;
-export type BaseConnectionEvents = {
-    [k in keyof BaseConnectionEventsHandlerMap]: BaseConnectionEventsHandlerMap[k];
-};
-export {};
-//# sourceMappingURL=interfaces.d.ts.map

package/dist/webrtc/src/utils/interfaces.d.ts.map

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package/dist/webrtc/src/utils/mockTracks.d.ts

@@ -1,26 +0,0 @@
-/**
- * Mock track utilities for pre-warming RTCPeerConnections
- * These tracks are used to trigger ICE gathering without real media
- */
-/**
- * Creates a mock audio track using Web Audio API
- * @returns A MediaStreamTrack that produces silence
- */
-export declare function createMockAudioTrack(): MediaStreamTrack;
-/**
- * Creates a mock video track using canvas
- * Falls back to null on Safari which doesn't support captureStream
- * @returns A MediaStreamTrack or null if not supported
- */
-export declare function createMockVideoTrack(): MediaStreamTrack | null;
-/**
- * Properly stops and cleans up a mock audio track
- * @param track The mock audio track to clean up
- */
-export declare function cleanupMockAudioTrack(track: MediaStreamTrack): void;
-/**
- * Properly stops and cleans up a mock video track
- * @param track The mock video track to clean up
- */
-export declare function cleanupMockVideoTrack(track: MediaStreamTrack | null): void;
-//# sourceMappingURL=mockTracks.d.ts.map

package/dist/webrtc/src/utils/mockTracks.d.ts.map

@@ -1 +0,0 @@
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package/dist/webrtc/src/utils/permissions.d.ts

@@ -1,48 +0,0 @@
-type DevicePermissionName = 'camera' | 'microphone' | 'speaker';
-/**
- * Asynchronously returns whether we have permissions to access the specified
- * resource. Some common parameter values for `name` are `"camera"`,
- * `"microphone"`, and `"speaker"`. In those cases, prefer the dedicated methods
- * {@link checkCameraPermissions}, {@link checkMicrophonePermissions}, and
- * {@link checkSpeakerPermissions}.
- * @param name name of the resource
- *
- * @example
- * ```typescript
- * await SignalWire.WebRTC.checkPermissions("camera")
- * // true: we have permission for using the camera
- * ```
- */
-export declare const checkPermissions: (name?: DevicePermissionName) => Promise<boolean | null>;
-/**
- * Asynchronously returns whether we have permissions to access the camera.
- *
- * @example
- * ```typescript
- * await SignalWire.WebRTC.checkCameraPermissions()
- * // true
- * ```
- */
-export declare const checkCameraPermissions: () => Promise<boolean | null>;
-/**
- * Asynchronously returns whether we have permissions to access the microphone.
- *
- * @example
- * ```typescript
- * await SignalWire.WebRTC.checkMicrophonePermissions()
- * // true
- * ```
- */
-export declare const checkMicrophonePermissions: () => Promise<boolean | null>;
-/**
- * Asynchronously returns whether we have permissions to access the speakers.
- *
- * @example
- * ```typescript
- * await SignalWire.WebRTC.checkSpeakerPermissions()
- * // true
- * ```
- */
-export declare const checkSpeakerPermissions: () => Promise<boolean | null>;
-export {};
-//# sourceMappingURL=permissions.d.ts.map

package/dist/webrtc/src/utils/permissions.d.ts.map

@@ -1 +0,0 @@
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package/dist/webrtc/src/utils/primitives.d.ts

@@ -1,54 +0,0 @@
-export declare const RTCPeerConnection: (config: RTCConfiguration) => RTCPeerConnection;
-/**
- * Returns whether the current environment supports the media devices API.
- */
-export declare const supportsMediaDevices: () => boolean;
-/**
- * Returns the mediaDevices object if supported, otherwise throws an error.
- */
-export declare const getMediaDevicesApi: () => MediaDevices;
-/**
- * Returns whether the current environment supports `getUserMedia`.
- */
-export declare const supportsGetUserMedia: () => boolean;
-/**
- * Returns whether the current environment supports `getDisplayMedia`.
- */
-export declare const supportsGetDisplayMedia: () => boolean;
-/**
- * Returns a dictionary whose fields specify the constrainable properties the user agent understands.
- */
-export declare const getSupportedConstraints: () => MediaTrackSupportedConstraints;
-export declare const streamIsValid: (stream?: MediaStream) => boolean | undefined;
-/**
- * Returns whether the current environment supports the selection of a media output device.
- */
-export declare const supportsMediaOutput: () => boolean;
-/**
- * Assigns the specified audio output device to the specified HTMLMediaElement.
- * The device with id `deviceId` must be an audio output device. Asynchronously
- * returns whether the operation had success.
- *
- * > 📘
- * > Some browsers do not support output device selection. You can check by
- * > calling [`supportsMediaOutput`](supportsMediaOutput).
- *
- * @param el target element
- * @param deviceId id of the audio output device
- * @returns a promise of whether the operation had success
- *
- * @example
- * ```typescript
- * const el = document.querySelector('video')
- * const outDevices = await SignalWire.WebRTC.getSpeakerDevicesWithPermissions()
- * await SignalWire.WebRTC.setMediaElementSinkId(el, outDevices[0].deviceId)
- * // true
-```
- */
-export declare const setMediaElementSinkId: (el: HTMLMediaElement | null, deviceId: string) => Promise<undefined>;
-export declare const sdpToJsonHack: (sdp: any) => any;
-export declare const stopStream: (stream?: MediaStream) => void;
-export declare const stopTrack: (track: MediaStreamTrack) => void;
-export type DevicePermissionName = 'camera' | 'microphone' | 'speaker';
-export declare const _getMediaDeviceKindByName: (name?: DevicePermissionName) => MediaDeviceKind | undefined;
-//# sourceMappingURL=primitives.d.ts.map

package/dist/webrtc/src/utils/primitives.d.ts.map

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package/dist/webrtc/src/utils/primitives.native.d.ts

@@ -1,36 +0,0 @@
-import { MediaStream as RNMediaStream } from 'react-native-webrtc';
-export declare const RTCPeerConnection: (config: RTCConfiguration) => any;
-/**
- * Returns whether the current environment supports the media devices API.
- */
-export declare const supportsMediaDevices: () => boolean;
-/**
- * Returns the mediaDevices object if supported, otherwise throws an error.
- */
-export declare const getMediaDevicesApi: () => any;
-/**
- * Returns whether the current environment supports `getUserMedia`.
- */
-export declare const supportsGetUserMedia: () => boolean;
-/**
- * Returns whether the current environment supports `getDisplayMedia`.
- */
-export declare const supportsGetDisplayMedia: () => boolean;
-/**
- * Returns a dictionary whose fields specify the constrainable properties the user agent understands.
- * Not supported on React Native
- */
-export declare const getSupportedConstraints: () => {};
-export declare const streamIsValid: (stream: RNMediaStream) => any;
-/**
- * Returns whether the current environment supports the selection of a media output device.
- * Not supported in React Native.
- */
-export declare const supportsMediaOutput: () => boolean;
-export declare const setMediaElementSinkId: (_htmlElementId: string, _deviceId: string) => Promise<boolean>;
-export declare const sdpToJsonHack: (sdp: any) => any;
-export declare const stopStream: (stream: RNMediaStream) => void;
-export declare const stopTrack: (track: MediaStreamTrack) => void;
-export type DevicePermissionName = 'camera' | 'microphone' | 'speaker';
-export declare const _getMediaDeviceKindByName: (name?: DevicePermissionName) => MediaDeviceKind | undefined;
-//# sourceMappingURL=primitives.native.d.ts.map

package/dist/webrtc/src/utils/primitives.native.d.ts.map

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package/dist/webrtc/src/utils/requestPermissions.d.ts

@@ -1,27 +0,0 @@
-/**
- * Prompts the user to grant permissions for the devices matching the specified set of constraints.
- * @param constraints an optional [MediaStreamConstraints](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamConstraints)
- *                    object specifying requirements for the returned [MediaStream](https://developer.mozilla.org/en-US/docs/Web/API/MediaStream).
- *
- * @example
- * To only request audio permissions:
- *
- * ```typescript
- * await SignalWire.WebRTC.requestPermissions({audio: true, video: false})
- * ```
- *
- * @example
- * To request permissions for both audio and video, specifying constraints for the video:
- * ```typescript
- * const constraints = {
- *   audio: true,
- *   video: {
- *     width: { min: 1024, ideal: 1280, max: 1920 },
- *     height: { min: 576, ideal: 720, max: 1080 }
- *   }
- * }
- * await SignalWire.WebRTC.requestPermissions(constraints)
- * ```
- */
-export declare const requestPermissions: (constraints: MediaStreamConstraints) => Promise<void>;
-//# sourceMappingURL=requestPermissions.d.ts.map

package/dist/webrtc/src/utils/requestPermissions.d.ts.map

@@ -1 +0,0 @@
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package/dist/webrtc/src/utils/sdpHelpers.d.ts

@@ -1,61 +0,0 @@
-/**
- * Check if SDP has a media section (audio or video)
- */
-export declare const sdpHasMediaSection: (sdp: string, media: "audio" | "video") => boolean;
-/**
- * Check if SDP includes video
- */
-export declare const sdpHasVideo: (sdp: string) => boolean;
-/**
- * Check if SDP includes audio
- */
-export declare const sdpHasAudio: (sdp: string) => boolean;
-/**
- * Add stereo support hacking the SDP
- * @return the SDP modified
- */
-export declare const sdpStereoHack: (sdp: string) => string;
-export declare const sdpMediaOrderHack: (answer: string, localOffer: string) => string;
-/**
- * Modify the SDP to increase video bitrate
- * @return the SDP modified
- */
-export declare const sdpBitrateHack: (sdp: string, max: number, min: number, start: number) => string;
-/**
- * Check for srflx, prflx or relay candidates
- * TODO: improve the logic check private/public IP for typ host
- *
- * @param sdp string
- * @returns boolean
- */
-export declare const sdpHasValidCandidates: (sdp: string) => boolean;
-/**
- * Check for ice candidates
- *
- * @param sdp string
- * @returns boolean
- */
-export declare const sdpHasCandidatesForEachMedia: (sdp: string) => boolean;
-/**
- * Remove "a=candidate" lines with local candidates
- * https://bloggeek.me/psa-mdns-and-local-ice-candidates-are-coming/
- */
-export declare const sdpRemoveLocalCandidates: (sdp: string) => string;
-/**
- * Get the SDP direction for the specified media type.
- * Returns 'sendrecv' if no direction attribute is found, as per SDP standards.
- */
-export declare const getSdpDirection: (sdp: string, media: "audio" | "video") => RTCRtpTransceiverDirection;
-/**
- * Returns the opposite SDP direction based on the provided direction.
- */
-export declare const getOppositeSdpDirection: (direction: RTCRtpTransceiverDirection) => RTCRtpTransceiverDirection;
-/**
- * Returns boolean indicating remote and local SDPs has the expected opposite direction
- */
-export declare const hasMatchingSdpDirection: ({ localSdp, remoteSdp, media, }: {
-    localSdp: string;
-    remoteSdp: string;
-    media: "audio" | "video";
-}) => boolean;
-//# sourceMappingURL=sdpHelpers.d.ts.map

package/dist/webrtc/src/utils/sdpHelpers.d.ts.map

@@ -1 +0,0 @@
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package/dist/webrtc/src/utils/sdpHelpers.test.d.ts

@@ -1,2 +0,0 @@
-export {};
-//# sourceMappingURL=sdpHelpers.test.d.ts.map

package/dist/webrtc/src/utils/sdpHelpers.test.d.ts.map

@@ -1 +0,0 @@
-{"version":3,"file":"sdpHelpers.test.d.ts","sourceRoot":"","sources":["../../../../../webrtc/src/utils/sdpHelpers.test.ts"],"names":[],"mappings":""}

package/dist/webrtc/src/utils/watchRTCPeerMediaPackets.d.ts

@@ -1,7 +0,0 @@
-import { EventEmitter } from '@signalwire/core';
-import type RTCPeer from '../RTCPeer';
-export declare const watchRTCPeerMediaPackets: <T extends EventEmitter.ValidEventTypes>(rtcPeer: RTCPeer<T>) => {
-    start: () => void;
-    stop: () => void;
-} | undefined;
-//# sourceMappingURL=watchRTCPeerMediaPackets.d.ts.map

package/dist/webrtc/src/utils/watchRTCPeerMediaPackets.d.ts.map

@@ -1 +0,0 @@
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package/dist/webrtc/src/utils/webrtcHelpers.test.d.ts

@@ -1,2 +0,0 @@
-export {};
-//# sourceMappingURL=webrtcHelpers.test.d.ts.map

package/dist/webrtc/src/utils/webrtcHelpers.test.d.ts.map

@@ -1 +0,0 @@
-{"version":3,"file":"webrtcHelpers.test.d.ts","sourceRoot":"","sources":["../../../../../webrtc/src/utils/webrtcHelpers.test.ts"],"names":[],"mappings":""}