@wazo/sdk 0.43.6 → 1.0.0-alpha.1

package/lib/types/utils/sdp.d.ts.map

@@ -1 +1 @@
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package/lib/types/utils/sip-logger.d.ts

@@ -0,0 +1,4 @@
+import { WebRtcConfig } from '../domain/types';
+declare const configureLogger: (rawOptions?: Partial<WebRtcConfig>) => Partial<WebRtcConfig>;
+export default configureLogger;
+//# sourceMappingURL=sip-logger.d.ts.map

package/lib/types/utils/sip-logger.d.ts.map

@@ -0,0 +1 @@
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package/lib/types/utils/webrtc.d.ts

@@ -1,3 +1,4 @@
+import { SipCall, UserAgentOptions, WebRtcConfig, UserAgentConfigOverrides } from '../domain/types';
 type Candidate = {
     foundation: string;
     component: string;
@@ -10,6 +11,45 @@ type Candidate = {
     relatedPort?: number;
     tcpType?: string;
 };
+type Constraint = MediaTrackConstraintSet | boolean | undefined;
+export declare const isWeb: () => boolean;
+export declare const isFirefox: () => boolean;
+export declare const isAPrivateIp: (ip: string) => boolean;
+export declare const getIceServers: (ip: string) => Array<{
+    urls: Array<string>;
+}>;
 export declare const parseCandidate: (line: string) => Candidate;
+export declare const getPeerConnection: (sipCall: SipCall) => RTCPeerConnection | null | undefined;
+export declare const getLocalStream: (sipCall: SipCall) => MediaStream | null;
+export declare const getLocalTracks: (sipCall: SipCall) => MediaStreamTrack[];
+export declare const getRemoteStream: (sipCall: SipCall) => MediaStream | null;
+export declare const isVideoTrack: (track: MediaStreamTrack) => boolean;
+export declare const hasLocalVideo: (sipCall: SipCall) => boolean;
+export declare const hasALocalVideoTrack: (sipCall: SipCall) => boolean;
+export declare const getLocalVideoStream: (sipCall: SipCall) => MediaStream | null | undefined;
+export declare const getRemoteTracks: (sipCall: SipCall) => MediaStreamTrack[];
+export declare const isVideoRemotelyHeld: (sipCall: SipCall) => boolean;
+export declare const hasRemoteVideo: (sipCall: SipCall) => boolean;
+export declare const hasARemoteVideoTrack: (sipCall: SipCall) => boolean;
+export declare const getRemoteVideoStream: (sipCall: SipCall) => MediaStream | null | undefined;
+export declare const getRemoteVideoStreamFromPc: (sipCall: SipCall) => MediaStream | null | undefined;
+export declare const hasVideoTrack: (sipCall: SipCall) => boolean;
+export declare const hasAVideoTrack: (sipCall: SipCall) => boolean;
+export declare const sessionWantsToDoVideo: (sipCall: SipCall) => boolean;
+export declare const getWebRtcStats: (sipCall: SipCall) => Promise<RTCStatsReport | null>;
+export declare const toggleAudio: (sipCall: SipCall, muteAudio: boolean) => void;
+export declare const toggleVideo: (sipCall: SipCall, muteCamera: boolean) => void;
+export declare const cleanupStream: (stream: MediaStream) => void;
+export declare const setLocalMediaStream: (sipCall: SipCall, newStream: MediaStream) => void;
+export declare const updateRemoteStream: (sipCall: SipCall, allTracks?: boolean) => void;
+export declare const isAudioOnly: (sipCall: SipCall) => boolean;
+export declare const getAudioConstraints: (audio: Constraint) => MediaTrackConstraints | boolean;
+export declare const getVideoConstraints: (video?: Constraint) => MediaTrackConstraints | boolean;
+export declare const getMediaConfiguration: (enableVideo: boolean, conference?: boolean, constraints?: Record<string, any> | null | undefined) => Record<string, any>;
+export declare const getStreamFromConstraints: (constraints: Record<string, any>, conference?: boolean) => Promise<MediaStream | null | undefined>;
+export declare const createUaOptions: (config: WebRtcConfig, uaOptionsOverrides?: UserAgentConfigOverrides, audio?: Constraint, video?: Constraint) => UserAgentOptions;
+export declare const downgradeToAudio: (sipCall: SipCall) => void;
+export declare const upgradeToVideo: (sipCall: SipCall, constraints: Record<string, any>, isConference: boolean) => Promise<MediaStream | undefined>;
+export declare const fetchNetworkStats: (sipCall: SipCall, lastNetworkStats: Record<string, any>) => Promise<Record<string, any> | null | undefined>;
 export {};
 //# sourceMappingURL=webrtc.d.ts.map

package/lib/types/utils/webrtc.d.ts.map

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package/lib/types/voice/call.d.ts

@@ -0,0 +1,110 @@
+/// <reference types="node" />
+import { EventEmitter } from 'events';
+import { OutgoingInviteRequest } from 'sip.js/lib/core';
+import type { IncomingRequestMessage } from 'sip.js/lib/core/messages/incoming-request-message';
+import { type ActionTypes, type EstablishedActionTypes, type CallActorRef } from '../state-machine/call-state-machine';
+import { SipCall } from '../domain/types';
+import { Softphone } from './softphone';
+export type AcceptOptions = {
+    withCamera?: boolean;
+};
+export type ReinviteOptions = {
+    constraints?: Record<string, any>;
+    conference?: boolean;
+    audioOnly?: boolean;
+    iceRestart?: boolean;
+};
+export declare const EVENT_CALL_ACCEPTED = "accepted";
+export declare const EVENT_CALL_CANCELED = "canceled";
+export declare const EVENT_PROGRESS = "progress";
+export declare const EVENT_EARLY_MEDIA = "earlyMedia";
+export declare const EVENT_CALL_ERROR = "error";
+export declare const EVENT_CALL_REJECTED = "rejected";
+export declare const EVENT_CALL_TERMINATING = "terminating";
+export declare const EVENT_CALL_ENDED = "ended";
+export declare const EVENT_CALL_MUTED = "muted";
+export declare const EVENT_CALL_UN_MUTED = "unMuted";
+export declare const EVENT_CALL_HELD = "held";
+export declare const EVENT_CALL_RESUMED = "resumed";
+export declare const EVENT_REINVITE = "reinvite";
+export declare const EVENT_TRACK = "track";
+export declare const EVENT_AUDIO_STREAM = "onAudioStream";
+export declare const EVENT_VIDEO_STREAM = "onVideoStream";
+export declare const EVENT_REMOVE_STREAM = "onRemoveStream";
+export declare const EVENT_SHARE_SCREEN_ENDING = "onScreenShareEnding";
+export declare const EVENT_SHARE_SCREEN_STARTED = "onScreenShareStarted";
+export declare const EVENT_SHARE_SCREEN_ENDED = "onScreenShareEnded";
+export declare const EVENT_WEB_RTC_STATS = "webRtcStats";
+export declare const EVENT_CAMERA_RESUMED = "onCameraResumed";
+export declare const EVENT_CAMERA_DISABLED = "onCameraDisabled";
+export declare const EVENT_TERMINATE_SOUND = "terminateSound";
+export declare const MESSAGE_TYPE_CHAT = "message/TYPE_CHAT";
+export declare const MESSAGE_TYPE_SIGNAL = "message/TYPE_SIGNAL";
+export declare const MESSAGE_TRACK_UPDATED = "onTrackUpdated";
+declare class Call extends EventEmitter {
+    eventEmitter: EventEmitter;
+    callActor: CallActorRef;
+    apiId?: string;
+    sipCall: SipCall;
+    talkingToIds: string[];
+    creationTime: Date;
+    answerTime: Date;
+    endTime: Date;
+    phone: Softphone;
+    shouldSendReinvite: boolean;
+    currentScreenShare: Record<string, any> | null;
+    constructor(sipCall: SipCall, phone: Softphone);
+    accept(options?: AcceptOptions): Promise<string | null>;
+    reject(): Promise<boolean>;
+    hangup(): Promise<boolean>;
+    forceCancel(): void;
+    startConference(withCalls: Call[]): void;
+    mute(): void;
+    unMute(): void;
+    hold(): Promise<void | OutgoingInviteRequest>;
+    resume(): Promise<MediaStream | null | undefined>;
+    turnCameraOn(): void;
+    turnCameraOff(): void;
+    isHeld(): boolean;
+    isEstablished(): boolean;
+    isConference(): boolean;
+    transfer(target: string): void;
+    atxfer(): Record<string, any> | null | undefined;
+    sendDigits(digits: string): void;
+    startNetworkMonitoring(interval?: number): void;
+    stopNetworkMonitoring(): void;
+    getStats(): Promise<RTCStatsReport | null | undefined>;
+    sendReinvite(options?: ReinviteOptions): Promise<OutgoingInviteRequest | void>;
+    sendMessage(body: string, contentType?: string): void;
+    sendChat(content: string): void;
+    sendSignal(content: any): void;
+    startScreenSharing(constraints: Record<string, any>): Promise<MediaStream>;
+    stopScreenSharing(restoreLocalStream?: boolean): Promise<OutgoingInviteRequest | void | null>;
+    onScreenShareReinvite(response: any, desktop: boolean): void;
+    onAccepted(): void;
+    onProgress(): void;
+    onEarlyMedia(): void;
+    onTrack(event: RTCTrackEvent | MediaStreamTrackEvent): void;
+    onError(error: Error): void;
+    onRejected(): void;
+    onNetworkStats(stats: Record<string, any>, previousStats: Record<string, any>): void;
+    getRemoteStream(): MediaStream | null;
+    getRemoteVideoStream(): MediaStream | null | undefined;
+    hasRemoteVideo(): boolean;
+    hasLocalVideo(): boolean;
+    getLocalStream(): MediaStream | null;
+    getLocalVideoStream(): MediaStream | null | undefined;
+    hasALocalVideoTrack(): boolean;
+    isVideoRemotelyHeld(): boolean;
+    onReinvite(request: IncomingRequestMessage, updatedCalleeName: string, updatedNumber: string, hadRemoteVideo: boolean): void;
+    get id(): string;
+    get number(): any;
+    _bindEvents(): void;
+    _downgradeToAudio(withMessage?: boolean): boolean;
+    _upgradeToVideo(options?: ReinviteOptions): Promise<boolean>;
+    _sendReinviteMessage(isUpgrade: boolean): void;
+    _onScreenSharing(screenStream: MediaStream, hadVideo: boolean, desktop?: boolean | null | undefined): void;
+    _sendAction(action: ActionTypes | EstablishedActionTypes): void;
+}
+export default Call;
+//# sourceMappingURL=call.d.ts.map

package/lib/types/voice/call.d.ts.map

@@ -0,0 +1 @@
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package/lib/types/voice/softphone.d.ts

@@ -0,0 +1,77 @@
+/// <reference types="node" />
+import { EventEmitter } from 'events';
+import type { Message } from 'sip.js/lib/api/message';
+import type Session from '../domain/Session';
+import type SipLine from '../domain/SipLine';
+import { type SoftphoneActorRef, type StateTypes, ActionTypes } from '../state-machine/softphone-state-machine';
+import { WebRtcConfig, SipCall } from '../domain/types';
+import WazoWebRTCClient from '../web-rtc-client';
+import Call from './call';
+export declare const EVENT_INCOMING = "incoming";
+export declare const EVENT_HEARTBEAT = "heartbeat";
+export declare const EVENT_HEARTBEAT_TIMEOUT = "heartbeatTimeout";
+export declare const EVENT_REGISTERED = "registered";
+export declare const EVENT_UNREGISTERED = "unregistered";
+export declare const EVENT_DISCONNECTED = "disconnected";
+export declare const EVENT_VIDEO_INPUT_CHANGE = "videoInputChange";
+export declare const EVENT_ON_MESSAGE = "message";
+export declare const EVENT_ON_CHAT = "phone/ON_CHAT";
+export declare const EVENT_ON_SIGNAL = "phone/ON_SIGNAL";
+export declare const MESSAGE_TYPE_CHAT = "message/TYPE_CHAT";
+export declare const MESSAGE_TYPE_SIGNAL = "message/TYPE_SIGNAL";
+export type CallOptions = {
+    params: {
+        To: string;
+    };
+    withCamera?: boolean;
+    audioOnly?: boolean;
+    conference?: boolean;
+};
+export declare class Softphone extends EventEmitter {
+    audioOutputDeviceId?: string;
+    audioRingDeviceId?: string;
+    audioOutputVolume: number;
+    audioRingVolume: number;
+    eventEmitter: EventEmitter;
+    calls: Call[];
+    client: WazoWebRTCClient;
+    session: Session;
+    sipLine: SipLine | null;
+    softphoneActor: SoftphoneActorRef;
+    options: Partial<WebRtcConfig>;
+    shouldSendReinvite: boolean;
+    constructor();
+    connect(options?: Partial<WebRtcConfig>): Promise<void>;
+    disconnect(): Promise<void | null>;
+    getPrimaryWebRtcLine(): SipLine | null;
+    connectWithCredentials(server: string, sipLine: SipLine, displayName: string, rawOptions?: Partial<WebRtcConfig>): Promise<void>;
+    reconnect(): void;
+    call(options: CallOptions): Promise<Call | null>;
+    removeCall(callToRemove: Call): void;
+    holdAllCalls(exceptCall?: Call | null): void;
+    startHeartbeat(): void;
+    stopHeartbeat(): void;
+    setMediaConstraints(media: MediaStreamConstraints): void;
+    changeAudioOutputDevice(id: string): Promise<void>;
+    changeAudioInputDevice(id: string, force?: boolean): Promise<MediaStream | null>;
+    changeVideoInputDevice(id: string): Promise<void | MediaStream | null | undefined>;
+    changeRingDevice(id: string): void;
+    hasSfu(): boolean;
+    changeAudioVolume(volume: number): void;
+    changeRingVolume(volume: number): void;
+    getCalls(): Call[];
+    getCurrentCall(): Call | undefined;
+    getState(): StateTypes;
+    getUserAgent(): string;
+    _bindEvents(): void;
+    _sendAction(action: ActionTypes): void;
+    _getCall(sipCall: SipCall): Call | undefined;
+    _onCallEvent(sipCall: SipCall, eventName: string, ...args: any[]): void;
+    _onMessage(message: Message & {
+        method?: string;
+        body?: string;
+    }): void;
+}
+declare const _default: Softphone;
+export default _default;
+//# sourceMappingURL=softphone.d.ts.map

package/lib/types/voice/softphone.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"softphone.d.ts","sourceRoot":"","sources":["../../../src/voice/softphone.ts"],"names":[],"mappings":";AACA,OAAO,EAAE,YAAY,EAAE,MAAM,QAAQ,CAAC;AAEtC,OAAO,KAAK,EAAE,OAAO,EAAE,MAAM,wBAAwB,CAAC;AAGtD,OAAO,KAAK,OAAO,MAAM,mBAAmB,CAAC;AAC7C,OAAO,KAAK,OAAO,MAAM,mBAAmB,CAAC;AAI7C,OAA8B,EAAE,KAAK,iBAAiB,EAAW,KAAK,UAAU,EAAU,WAAW,EAAE,MAAM,0CAA0C,CAAC;AAExJ,OAAO,EAAE,YAAY,EAAE,OAAO,EAAE,MAAM,iBAAiB,CAAC;AACxD,OAAO,gBAAgB,MAAM,mBAAmB,CAAC;AAGjD,OAAO,IAAI,MAAM,QAAQ,CAAC;AAG1B,eAAO,MAAM,cAAc,aAAa,CAAC;AACzC,eAAO,MAAM,eAAe,cAAc,CAAC;AAC3C,eAAO,MAAM,uBAAuB,qBAAqB,CAAC;AAC1D,eAAO,MAAM,gBAAgB,eAAe,CAAC;AAC7C,eAAO,MAAM,kBAAkB,iBAAiB,CAAC;AACjD,eAAO,MAAM,kBAAkB,iBAAiB,CAAC;AACjD,eAAO,MAAM,wBAAwB,qBAAqB,CAAC;AAC3D,eAAO,MAAM,gBAAgB,YAAY,CAAC;AAC1C,eAAO,MAAM,aAAa,kBAAkB,CAAC;AAC7C,eAAO,MAAM,eAAe,oBAAoB,CAAC;AAEjD,eAAO,MAAM,iBAAiB,sBAAsB,CAAC;AACrD,eAAO,MAAM,mBAAmB,wBAAwB,CAAC;AAEzD,MAAM,MAAM,WAAW,GAAG;IACxB,MAAM,EAAE;QACN,EAAE,EAAE,MAAM,CAAC;KACZ,CAAC;IACF,UAAU,CAAC,EAAE,OAAO,CAAC;IAErB,SAAS,CAAC,EAAE,OAAO,CAAC;IACpB,UAAU,CAAC,EAAE,OAAO,CAAC;CACtB,CAAC;AAIF,qBAAa,SAAU,SAAQ,YAAY;IACzC,mBAAmB,CAAC,EAAE,MAAM,CAAC;IAE7B,iBAAiB,CAAC,EAAE,MAAM,CAAC;IAE3B,iBAAiB,EAAE,MAAM,CAAC;IAE1B,eAAe,EAAE,MAAM,CAAC;IAExB,YAAY,EAAE,YAAY,CAAC;IAE3B,KAAK,EAAE,IAAI,EAAE,CAAC;IAEd,MAAM,EAAE,gBAAgB,CAAC;IAEzB,OAAO,EAAE,OAAO,CAAC;IAEjB,OAAO,EAAE,OAAO,GAAG,IAAI,CAAC;IAExB,cAAc,EAAE,iBAAiB,CAAC;IAElC,OAAO,EAAE,OAAO,CAAC,YAAY,CAAC,CAAC;IAE/B,kBAAkB,EAAE,OAAO,CAAC;;IAW5B,OAAO,CAAC,OAAO,GAAE,OAAO,CAAC,YAAY,CAAM;IAyBrC,UAAU;IAYhB,oBAAoB,IAAI,OAAO,GAAG,IAAI;IAIhC,sBAAsB,CAC1B,MAAM,EAAE,MAAM,EACd,OAAO,EAAE,OAAO,EAChB,WAAW,EAAE,MAAM,EACnB,UAAU,GAAE,OAAO,CAAC,YAAY,CAAM,GACrC,OAAO,CAAC,IAAI,CAAC;IAqChB,SAAS;IASH,IAAI,CAAC,OAAO,EAAE,WAAW,GAAG,OAAO,CAAC,IAAI,GAAG,IAAI,CAAC;IAmCtD,UAAU,CAAC,YAAY,EAAE,IAAI;IAI7B,YAAY,CAAC,UAAU,GAAE,IAAI,GAAG,IAAW;IAI3C,cAAc;IASd,aAAa;IASb,mBAAmB,CAAC,KAAK,EAAE,sBAAsB,GAAG,IAAI;IAIxD,uBAAuB,CAAC,EAAE,EAAE,MAAM;IAOlC,sBAAsB,CAAC,EAAE,EAAE,MAAM,EAAE,KAAK,CAAC,EAAE,OAAO;IAOlD,sBAAsB,CAAC,EAAE,EAAE,MAAM,GAAG,OAAO,CAAC,IAAI,GAAG,WAAW,GAAG,IAAI,GAAG,SAAS,CAAC;IAOlF,gBAAgB,CAAC,EAAE,EAAE,MAAM;IAK3B,MAAM,IAAI,OAAO;IAKjB,iBAAiB,CAAC,MAAM,EAAE,MAAM;IAOhC,gBAAgB,CAAC,MAAM,EAAE,MAAM;IAK/B,QAAQ,IAAI,IAAI,EAAE;IAIlB,cAAc,IAAI,IAAI,GAAG,SAAS;IAIlC,QAAQ,IAAI,UAAU;IAItB,YAAY;IAIZ,WAAW,IAAI,IAAI;IAoInB,WAAW,CAAC,MAAM,EAAE,WAAW;IAI/B,QAAQ,CAAC,OAAO,EAAE,OAAO,GAAG,IAAI,GAAG,SAAS;IAI5C,YAAY,CAAC,OAAO,EAAE,OAAO,EAAE,SAAS,EAAE,MAAM,EAAE,GAAG,IAAI,EAAE,GAAG,EAAE;IAahE,UAAU,CAAC,OAAO,EAAE,OAAO,GAAG;QAAE,MAAM,CAAC,EAAE,MAAM,CAAC;QAAC,IAAI,CAAC,EAAE,MAAM,CAAA;KAAE,GAAG,IAAI;CAiCxE;;AAED,wBAA+B"}

package/lib/types/web-rtc-client.d.ts

@@ -5,14 +5,12 @@ import type { SessionByeOptions } from 'sip.js/lib/api/session-bye-options';
 import type { InvitationAcceptOptions } from 'sip.js/lib/api/invitation-accept-options';
 import type { SessionDialog } from 'sip.js/lib/core/dialogs/session-dialog';
 import { UserAgentState } from 'sip.js/lib/api/user-agent-state';
-import { URI } from 'sip.js/lib/grammar/uri';
 import { UserAgent } from 'sip.js/lib/api/user-agent';
 import { OutgoingByeRequest, OutgoingInviteRequest, OutgoingRequest } from 'sip.js/lib/core';
 import { Inviter, Invitation, Registerer } from 'sip.js/lib/api';
 import Emitter from './utils/Emitter';
 import Heartbeat from './utils/Heartbeat';
-import type { MediaConfig, UserAgentConfigOverrides, WebRtcConfig, UserAgentOptions, PeerConnection, WazoSession } from './domain/types';
-export declare const replaceLocalIpModifier: (description: Record<string, any>) => Promise<any>;
+import type { MediaConfig, UserAgentConfigOverrides, WebRtcConfig, WazoSession } from './domain/types';
 export declare const events: string[];
 export declare const transportEvents: string[];
 export declare class CanceledCallError extends Error {
@@ -62,10 +60,6 @@ export default class WebRTCClient extends Emitter {
     ON_EARLY_MEDIA: string;
     ON_PROGRESS: string;
     ON_DISCONNECTED: string;
-    static isAPrivateIp(ip: string): boolean;
-    static getIceServers(ip: string): Array<{
-        urls: Array<string>;
-    }>;
     constructor(config: WebRtcConfig, session?: WazoSession, uaConfigOverrides?: UserAgentConfigOverrides);
     configureMedia(media: MediaConfig): void;
     setMediaConstraints(media: MediaStreamConstraints): void;
@@ -82,12 +76,10 @@ export default class WebRTCClient extends Emitter {
     call(number: string, enableVideo?: boolean, audioOnly?: boolean, conference?: boolean, options?: {}): WazoSession | null;
     answer(session: Invitation, enableVideo?: boolean): Promise<void>;
     hangup(session: WazoSession): Promise<OutgoingByeRequest | null>;
-    getStats(session: WazoSession): Promise<RTCStatsReport | null>;
     startNetworkMonitoring(session: WazoSession, interval?: number): void;
     stopNetworkMonitoring(session: WazoSession): void;
     reject(session: WazoSession): Promise<void>;
     close(force?: boolean): Promise<void>;
-    getNumber(session: Inviter): string | null | undefined;
     mute(session: WazoSession): void;
     unmute(session: WazoSession): void;
     isAudioMuted(session: WazoSession): boolean;
@@ -95,9 +87,6 @@ export default class WebRTCClient extends Emitter {
     toggleCameraOff(session: WazoSession): void;
     hold(session: WazoSession, isConference?: boolean, hadVideo?: boolean): Promise<OutgoingInviteRequest | void>;
     unhold(session: WazoSession, isConference?: boolean): Promise<OutgoingInviteRequest | void>;
-    upgradeToVideo(session: WazoSession, constraints: Record<string, any>, isConference: boolean): Promise<MediaStream | undefined>;
-    downgradeToAudio(session: WazoSession): void;
-    getStreamFromConstraints(constraints: Record<string, any>, conference?: boolean): Promise<MediaStream | null | undefined>;
     getHeldSession(sessionId: string): any;
     isCallHeld(session: WazoSession): boolean;
     isVideoRemotelyHeld(sessionId: string): boolean;
@@ -109,7 +98,6 @@ export default class WebRTCClient extends Emitter {
     pingServer(): OutgoingRequest | void;
     getState(): UserAgentState;
     getContactIdentifier(): string | null;
-    isFirefox(): boolean;
     changeAudioOutputVolume(volume: number): void;
     changeAudioOutputDevice(id: string): Promise<void>;
     changeAudioInputDevice(id: string, session: WazoSession | null | undefined, force?: boolean): Promise<MediaStream | null>;
@@ -134,9 +122,7 @@ export default class WebRTCClient extends Emitter {
     getRemoteVideoStreamFromPc(sessionId: string): MediaStream | null | undefined;
     hasVideo(sessionId: string): boolean;
     hasAVideoTrack(sessionId: string): boolean;
-    getSipSessionId(sipSession: WazoSession | null | undefined): string;
     waitForRegister(): Promise<void>;
-    sessionWantsToDoVideo(session: WazoSession): boolean;
     hasHeartbeat(): boolean;
     startHeartbeat(): void;
     stopHeartbeat(): void;
@@ -147,47 +133,26 @@ export default class WebRTCClient extends Emitter {
     storeSipSession(session: WazoSession): void;
     getSipSession(id: string): WazoSession | null | undefined;
     getSipSessionIds(): string[];
-    setLocalMediaStream(sipSessionId: string, newStream: MediaStream): void;
     updateLocalStream(sipSessionId: string, newStream: MediaStream): Promise<void>;
-    updateRemoteStream(sessionId: string, allTracks?: boolean): void;
-    getMediaConfiguration(enableVideo: boolean, conference?: boolean, constraints?: Record<string, any> | null | undefined): Record<string, any>;
     isConference(sessionId: string): boolean;
     createAudioElementFor(sessionId: string): Promise<HTMLAudioElement>;
     _onTransportError(): void;
     enableLogger(logConnector: (level: any, className: string, label: any, content: string) => void): void;
     _onHeartbeat(message: string | Record<string, any>): void;
     _onHeartbeatTimeout(): Promise<void>;
-    _isWeb(): boolean;
-    _isVideoTrack(track: MediaStreamTrack): boolean;
     _hasAudio(): boolean;
     _getAudioConstraints(): MediaTrackConstraints | boolean;
     _getVideoConstraints(video?: boolean): MediaTrackConstraints | boolean;
     _connectIfNeeded(): Promise<any>;
     _buildConfig(config: WebRtcConfig, session: WazoSession | null | undefined): Promise<WebRtcConfig>;
-    _createUaOptions(uaOptionsOverrides?: UserAgentConfigOverrides): UserAgentOptions;
-    _getRtcOptions(): {
-        mandatory: {
-            OfferToReceiveAudio: boolean;
-            OfferToReceiveVideo: boolean;
-        };
-    };
     _setupSession(session: WazoSession): void;
     _onProgress(session: WazoSession, early?: boolean): Promise<void>;
     _onAccepted(session: WazoSession, sessionDialog?: SessionDialog, withEvent?: boolean, initAllTracks?: boolean): Promise<void>;
-    _isAudioOnly(session: WazoSession): boolean;
     _setupMedias(session: WazoSession, newStream?: MediaStream | null | undefined): Promise<void>;
     _cleanupMedia(session?: WazoSession): void;
-    _cleanupStream(stream: MediaStream): void;
-    _toggleAudio(session: WazoSession, muteAudio: boolean): void;
-    _toggleVideo(session: WazoSession, muteCamera: boolean): void;
-    /**
-     * @param pc RTCPeerConnection
-     */
-    _getRemoteStream(pc: PeerConnection): MediaStream | null;
     _cleanupRegister(): void;
     _startSendingStats(session: WazoSession): void;
     _stopSendingStats(session: WazoSession): void;
-    _makeURI(target: string): URI | undefined;
     _disconnectTransport(force?: boolean): Promise<void>;
     _fetchNetworkStats(sessionId: string): Promise<Record<string, any> | null | undefined>;
     _accept(session: Invitation, options?: InvitationAcceptOptions): Promise<void>;

package/lib/types/web-rtc-client.d.ts.map

@@ -1 +1 @@
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package/lib/utils/sdp.js

@@ -3,8 +3,11 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
     return (mod && mod.__esModule) ? mod : { "default": mod };
 };
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.addIcesInAllBundles = exports.fixSdp = exports.hasAnActiveVideo = exports.activateVideoModifier = exports.deactivateVideoModifier = exports.getVideoDirection = exports.toggleVideoDirection = exports.fixBundle = exports.isSdpValid = exports.areCandidateValid = exports.parseCandidate = exports.getCandidates = void 0;
+exports.makeURI = exports.replaceLocalIpModifier = exports.getSipSessionId = exports.addIcesInAllBundles = exports.fixSdp = exports.hasAnActiveVideo = exports.activateVideoModifier = exports.deactivateVideoModifier = exports.getVideoDirection = exports.toggleVideoDirection = exports.fixBundle = exports.isSdpValid = exports.areCandidateValid = exports.parseCandidate = exports.getCandidates = exports.SIP_ID_LENGTH = void 0;
 const sdp_transform_1 = __importDefault(require("sdp-transform"));
+const user_agent_1 = require("sip.js/lib/api/user-agent");
+const api_1 = require("sip.js/lib/api");
+exports.SIP_ID_LENGTH = 36;
 const getCandidates = (rawSdp) => {
     if (!rawSdp) {
         return [];
@@ -113,3 +116,26 @@ const addIcesInAllBundles = (sdp) => {
     return sdp_transform_1.default.write(parsedSdp);
 };
 exports.addIcesInAllBundles = addIcesInAllBundles;
+const getSipSessionId = (sipSession) => {
+    if (!sipSession) {
+        return '';
+    }
+    const message = sipSession.message;
+    if (message === null || message === void 0 ? void 0 : message.callId) {
+        return message.callId.substring(0, exports.SIP_ID_LENGTH);
+    }
+    // For Inviter
+    // @ts-ignore: private
+    if (sipSession instanceof api_1.Inviter && sipSession.outgoingRequestMessage) {
+        // @ts-ignore: private
+        return sipSession.outgoingRequestMessage.callId.substring(0, exports.SIP_ID_LENGTH);
+    }
+    // For Invitation
+    return (sipSession.id || '').substring(0, exports.SIP_ID_LENGTH);
+};
+exports.getSipSessionId = getSipSessionId;
+// We need to replace 0.0.0.0 to 127.0.0.1 in the sdp to avoid MOH during a createOffer.
+const replaceLocalIpModifier = (description) => Promise.resolve(Object.assign(Object.assign({}, JSON.parse(JSON.stringify(description))), { sdp: description.sdp.replace('c=IN IP4 0.0.0.0', 'c=IN IP4 127.0.0.1') }));
+exports.replaceLocalIpModifier = replaceLocalIpModifier;
+const makeURI = (target, host) => user_agent_1.UserAgent.makeURI(`sip:${target}@${host}`);
+exports.makeURI = makeURI;

package/lib/utils/sip-logger.js

@@ -0,0 +1,37 @@
+"use strict";
+var __importDefault = (this && this.__importDefault) || function (mod) {
+    return (mod && mod.__esModule) ? mod : { "default": mod };
+};
+Object.defineProperty(exports, "__esModule", { value: true });
+const IssueReporter_1 = __importDefault(require("../service/IssueReporter"));
+const sipLogger = IssueReporter_1.default.loggerFor('sip.js');
+const protocolLogger = IssueReporter_1.default.loggerFor('sip');
+const protocolDebugMessages = ['Received WebSocket text message:', 'Sending WebSocket message:'];
+const logConnector = (level, className, label, content) => {
+    const protocolIndex = content && content.indexOf ? protocolDebugMessages.findIndex(prefix => content.indexOf(prefix) !== -1) : -1;
+    if (className === 'sip.Transport' && protocolIndex !== -1) {
+        const direction = protocolIndex === 0 ? 'receiving' : 'sending';
+        const message = content.replace(`${protocolDebugMessages[protocolIndex]}\n\n`, '').replace('\r\n', '\n');
+        protocolLogger.trace(message, {
+            className,
+            direction,
+        });
+    }
+    else {
+        sipLogger.trace(content, {
+            className,
+        });
+    }
+};
+const configureLogger = (rawOptions = {}) => {
+    const options = rawOptions;
+    if (options.uaConfigOverrides) {
+        options.uaConfigOverrides.traceSip = true;
+    }
+    options.log = options.log || {};
+    options.log.builtinEnabled = false;
+    options.log.logLevel = 'debug';
+    options.log.connector = logConnector;
+    return options;
+};
+exports.default = configureLogger;