@wazo/sdk 0.37.30 → 0.37.34
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/dist/wazo-sdk.js +4 -4
- package/esm/__tests__/api-client.test.js +30 -0
- package/esm/__tests__/web-rtc-client.test.js +57 -6
- package/esm/api/calld.js +30 -0
- package/esm/api-client.js +30 -0
- package/esm/checker/Checker.js +57 -6
- package/esm/checker/checks/aor.js +30 -0
- package/esm/checker/checks/api.js +30 -0
- package/esm/checker/checks/index.js +57 -6
- package/esm/checker/checks/webrtc-transport.js +57 -6
- package/esm/checker/checks/webrtc.js +57 -6
- package/esm/domain/AdHocAPIConference.js +30 -0
- package/esm/domain/CallSession.js +9 -1
- package/esm/domain/Features.js +30 -0
- package/esm/domain/MeetingStatus.js +29 -0
- package/esm/domain/Phone/CTIPhone.js +39 -1
- package/esm/domain/Phone/WebRTCPhone.js +43 -8
- package/esm/domain/__tests__/CallSession.test.js +41 -5
- package/esm/domain/__tests__/IndirectTransfer.test.js +9 -1
- package/esm/domain/__tests__/Room.test.js +9 -1
- package/esm/index.js +100 -14
- package/esm/service/CallApi.js +30 -0
- package/esm/service/getApiClient.js +30 -0
- package/esm/simple/Auth.js +100 -14
- package/esm/simple/Configuration.js +100 -14
- package/esm/simple/Directory.js +30 -0
- package/esm/simple/Phone.js +100 -14
- package/esm/simple/index.js +100 -14
- package/esm/simple/room/LocalParticipant.js +100 -14
- package/esm/simple/room/Participant.js +100 -14
- package/esm/simple/room/RemoteParticipant.js +100 -14
- package/esm/simple/room/Room.js +100 -14
- package/esm/simple/room/SipRoom.js +100 -14
- package/esm/simple/room/Stream.js +100 -14
- package/esm/simple/room/__tests__/RemoteParticipant.test.js +100 -14
- package/esm/simple/utils.js +100 -14
- package/esm/web-rtc-client.js +57 -6
- package/lib/__tests__/api-client.test.js +30 -0
- package/lib/__tests__/web-rtc-client.test.js +57 -6
- package/lib/api/calld.js +30 -0
- package/lib/api-client.js +30 -0
- package/lib/checker/Checker.js +57 -6
- package/lib/checker/checks/aor.js +30 -0
- package/lib/checker/checks/api.js +30 -0
- package/lib/checker/checks/index.js +57 -6
- package/lib/checker/checks/webrtc-transport.js +57 -6
- package/lib/checker/checks/webrtc.js +57 -6
- package/lib/domain/AdHocAPIConference.js +30 -0
- package/lib/domain/CallSession.js +9 -1
- package/lib/domain/Features.js +30 -0
- package/lib/domain/MeetingStatus.js +31 -0
- package/lib/domain/Phone/CTIPhone.js +39 -1
- package/lib/domain/Phone/WebRTCPhone.js +43 -8
- package/lib/domain/__tests__/CallSession.test.js +41 -5
- package/lib/domain/__tests__/IndirectTransfer.test.js +9 -1
- package/lib/domain/__tests__/Room.test.js +9 -1
- package/lib/index.js +100 -14
- package/lib/service/CallApi.js +30 -0
- package/lib/service/getApiClient.js +30 -0
- package/lib/simple/Auth.js +100 -14
- package/lib/simple/Configuration.js +100 -14
- package/lib/simple/Directory.js +30 -0
- package/lib/simple/Phone.js +100 -14
- package/lib/simple/index.js +100 -14
- package/lib/simple/room/LocalParticipant.js +100 -14
- package/lib/simple/room/Participant.js +100 -14
- package/lib/simple/room/RemoteParticipant.js +100 -14
- package/lib/simple/room/Room.js +100 -14
- package/lib/simple/room/SipRoom.js +100 -14
- package/lib/simple/room/Stream.js +100 -14
- package/lib/simple/room/__tests__/RemoteParticipant.test.js +100 -14
- package/lib/simple/utils.js +100 -14
- package/lib/web-rtc-client.js +57 -6
- package/package.json +1 -1
- package/src/api/calld.js +3 -0
- package/src/domain/CallSession.js +9 -1
- package/src/domain/MeetingStatus.js +27 -0
- package/src/domain/Phone/WebRTCPhone.js +34 -7
- package/src/domain/__tests__/CallSession.test.js +32 -4
- package/src/web-rtc-client.js +27 -6
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@@ -3620,13 +3620,21 @@ class CallSession {
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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this.answerTime = value ? new Date() : null;
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this.answerTime = value ? this.answerTime || new Date() : null;
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}
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get answered() {
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return !!this.answerTime;
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}
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toJSON() {
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const jsonObj = { ...this };
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// $FlowFixMe
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jsonObj.answered = this.answered;
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return jsonObj;
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}
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static parseCall(call ) {
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return new CallSession({
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callId: call.id,
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@@ -4852,6 +4860,34 @@ var chatdMethods = (client , baseUrl ) => ({
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client.get(`${baseUrl}/users/me/rooms/messages`, options),
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});
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//
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class MeetingStatus {
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constructor({
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full,
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} = {}) {
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this.full = full;
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// Useful to compare instead of instanceof with minified code
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this.type = 'MeetingStatus';
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}
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static parse(plain ) {
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return new MeetingStatus({
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full: plain.full,
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});
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}
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}
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/* */
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client.put(`${baseUrl}/users/me/calls/${callId}/record/resume`, null, null,
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ApiRequester.successResponseParser),
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guestGetMeetingStatus: (meetingUuid ) =>
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client.get(`${baseUrl}/guests/me/meetings/${meetingUuid}/status`).then(MeetingStatus.parse),
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});
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//
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const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
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class CanceledCallError extends Error {}
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const MAX_REGISTER_TRIES =
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const MAX_REGISTER_TRIES = 5;
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// sugar
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this.conferences = {};
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this.networkMonitoringInterval = {};
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this.sessionNetworkStats = {};
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this.forceClosed = false;
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this._boundOnHeartbeat = this._onHeartbeat.bind(this);
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this.heartbeat = new Heartbeat(config.heartbeatDelay, config.heartbeatTimeout, config.maxHeartbeats);
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skipRegister: this.skipRegister,
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};
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this.forceClosed = false;
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if (this.skipRegister) {
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logger$8.info('sdk webrtc skip register...', logInfo);
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return Promise.resolve();
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const registerOptions = this._isWeb() ? {} : { extraContactHeaderParams: ['mobility=mobile'] };
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const onRegisterFailed = () => {
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logger$8.info('sdk webrtc registering failed', {
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logger$8.info('sdk webrtc registering failed', {
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tries,
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registerer: !!this.registerer,
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forceClosed: this.forceClosed,
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});
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if (this.forceClosed) {
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return;
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}
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this.connectionPromise = null;
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if (this.registerer && this.registerer.waiting) {
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this.registerer.waitingToggle(false);
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}
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}
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async close() {
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logger$8.info('sdk webrtc closing client', { userAgent: !!this.userAgent });
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async close(force = false) {
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logger$8.info('sdk webrtc closing client', { userAgent: !!this.userAgent, force });
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this.forceClosed = force;
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this._cleanupMedia();
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this.connectionPromise = null;
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this.userAgent.delegate = null;
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this.userAgent.stateChange.removeAllListeners();
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await this._disconnectTransport();
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await this._disconnectTransport(force);
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this._cleanupRegister();
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return userAgent.UserAgent.makeURI(`sip:${target}@${this.config.host}`);
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}
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async _disconnectTransport() {
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async _disconnectTransport(force = false) {
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if (force) {
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// Bypass sip.js state machine that prevent to close WS with the state `Connecting`
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this.userAgent.transport.disconnectResolve = () => {};
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// eslint-disable-next-line
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this.userAgent.transport._ws.close(1000);
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return;
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}
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// Check if `disconnectPromise` is not present to avoid `Disconnect promise must not be defined` errors.
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if (this.userAgent && this.userAgent.transport && !this.userAgent.transport.disconnectPromise) {
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try {
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}
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callSession.answerTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
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return callSession;
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return Promise.resolve(null);
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}
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const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
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if (sipSessionId && !(sipSessionId in this.callSessions)) {
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logger$7.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
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return Promise.resolve(null);
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}
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if (callSession.getId() in this.acceptedSessions) {
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logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
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return Promise.resolve(callSession.sipCallId);
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this.shouldSendReinvite = false;
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callSession.answerTime = new Date();
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this._updateCallSession(callSession);
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this.acceptedSessions[callSession.getId()] = true;
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this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
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// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
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if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
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logger$7.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
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this.holdSipSession(this.currentSipSession,
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this.holdSipSession(this.currentSipSession, this.currentCallSession);
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}
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const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
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}
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const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
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callSession.creationTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
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return this.client && this.client.isConference(callSession.sipCallId);
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}
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async close() {
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async close(force = false) {
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logger$7.info('WebRTC close');
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try {
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logger$7.error('WebRTC close, unregister error', e);
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}
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this.client.close();
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this.client.close(force);
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this.unbind();
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this.incomingSessions = [];
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}
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callSession.creationTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
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});
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});
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this.client.on(this.client.ACCEPTED, (sipSession ) => {
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const sessionId = this.getSipSessionId(sipSession);
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const hasSession = sessionId in this.callSessions;
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logger$7.info('WebRTC call accepted', { sipId: sessionId, hasSession });
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if (!hasSession) {
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logger$7.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
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return;
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}
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this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
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if (this.audioOutputDeviceId) {
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this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
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logger$7.info('WebRTC call rejected', session.id);
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const callSession = this._createCallSession(session);
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callSession.endTime = new Date();
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this._updateCallSession(callSession);
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10298
|
this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
|
|
10219
10299
|
});
|
|
10220
10300
|
|
|
@@ -10332,9 +10412,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10332
10412
|
return this.client.getSipSessionId(sipSession);
|
|
10333
10413
|
}
|
|
10334
10414
|
|
|
10335
|
-
|
|
10415
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10336
10416
|
const sipSession = this.findSipSession(callSession);
|
|
10337
|
-
|
|
10417
|
+
|
|
10418
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10419
|
+
}
|
|
10420
|
+
|
|
10421
|
+
_updateCallSession(callSession ) {
|
|
10422
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10338
10423
|
|
|
10339
10424
|
if (sipSessionId) {
|
|
10340
10425
|
this.callSessions[sipSessionId] = callSession;
|
|
@@ -10439,6 +10524,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10439
10524
|
displayName: identity ? identity.displayName || number : number,
|
|
10440
10525
|
startTime: fromSession ? fromSession.startTime : new Date(),
|
|
10441
10526
|
creationTime: fromSession ? fromSession.creationTime : null,
|
|
10527
|
+
answerTime: fromSession ? fromSession.answerTime : null,
|
|
10442
10528
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10443
10529
|
answered: state === sessionState.SessionState.Established,
|
|
10444
10530
|
paused: this.client.isCallHeld(sipSession),
|
|
@@ -3620,13 +3620,21 @@ class CallSession {
|
|
|
3620
3620
|
|
|
3621
3621
|
// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
|
|
3622
3622
|
set answered(value ) {
|
|
3623
|
-
this.answerTime = value ? new Date() : null;
|
|
3623
|
+
this.answerTime = value ? this.answerTime || new Date() : null;
|
|
3624
3624
|
}
|
|
3625
3625
|
|
|
3626
3626
|
get answered() {
|
|
3627
3627
|
return !!this.answerTime;
|
|
3628
3628
|
}
|
|
3629
3629
|
|
|
3630
|
+
toJSON() {
|
|
3631
|
+
const jsonObj = { ...this };
|
|
3632
|
+
// $FlowFixMe
|
|
3633
|
+
jsonObj.answered = this.answered;
|
|
3634
|
+
|
|
3635
|
+
return jsonObj;
|
|
3636
|
+
}
|
|
3637
|
+
|
|
3630
3638
|
static parseCall(call ) {
|
|
3631
3639
|
return new CallSession({
|
|
3632
3640
|
callId: call.id,
|
|
@@ -4852,6 +4860,34 @@ var chatdMethods = (client , baseUrl ) => ({
|
|
|
4852
4860
|
client.get(`${baseUrl}/users/me/rooms/messages`, options),
|
|
4853
4861
|
});
|
|
4854
4862
|
|
|
4863
|
+
//
|
|
4864
|
+
|
|
4865
|
+
|
|
4866
|
+
|
|
4867
|
+
|
|
4868
|
+
|
|
4869
|
+
class MeetingStatus {
|
|
4870
|
+
|
|
4871
|
+
|
|
4872
|
+
|
|
4873
|
+
|
|
4874
|
+
constructor({
|
|
4875
|
+
full,
|
|
4876
|
+
} = {}) {
|
|
4877
|
+
this.full = full;
|
|
4878
|
+
|
|
4879
|
+
// Useful to compare instead of instanceof with minified code
|
|
4880
|
+
this.type = 'MeetingStatus';
|
|
4881
|
+
}
|
|
4882
|
+
|
|
4883
|
+
static parse(plain ) {
|
|
4884
|
+
return new MeetingStatus({
|
|
4885
|
+
full: plain.full,
|
|
4886
|
+
});
|
|
4887
|
+
}
|
|
4888
|
+
|
|
4889
|
+
}
|
|
4890
|
+
|
|
4855
4891
|
/* */
|
|
4856
4892
|
|
|
4857
4893
|
|
|
@@ -5046,6 +5082,8 @@ var calldMethods = (client , baseUrl ) => ({
|
|
|
5046
5082
|
client.put(`${baseUrl}/users/me/calls/${callId}/record/resume`, null, null,
|
|
5047
5083
|
ApiRequester.successResponseParser),
|
|
5048
5084
|
|
|
5085
|
+
guestGetMeetingStatus: (meetingUuid ) =>
|
|
5086
|
+
client.get(`${baseUrl}/guests/me/meetings/${meetingUuid}/status`).then(MeetingStatus.parse),
|
|
5049
5087
|
});
|
|
5050
5088
|
|
|
5051
5089
|
//
|
|
@@ -6635,7 +6673,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6635
6673
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6636
6674
|
class CanceledCallError extends Error {}
|
|
6637
6675
|
|
|
6638
|
-
const MAX_REGISTER_TRIES =
|
|
6676
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6639
6677
|
|
|
6640
6678
|
|
|
6641
6679
|
|
|
@@ -6689,6 +6727,7 @@ class WebRTCClient extends Emitter {
|
|
|
6689
6727
|
|
|
6690
6728
|
|
|
6691
6729
|
|
|
6730
|
+
|
|
6692
6731
|
|
|
6693
6732
|
// sugar
|
|
6694
6733
|
|
|
@@ -6751,6 +6790,7 @@ class WebRTCClient extends Emitter {
|
|
|
6751
6790
|
this.conferences = {};
|
|
6752
6791
|
this.networkMonitoringInterval = {};
|
|
6753
6792
|
this.sessionNetworkStats = {};
|
|
6793
|
+
this.forceClosed = false;
|
|
6754
6794
|
|
|
6755
6795
|
this._boundOnHeartbeat = this._onHeartbeat.bind(this);
|
|
6756
6796
|
this.heartbeat = new Heartbeat(config.heartbeatDelay, config.heartbeatTimeout, config.maxHeartbeats);
|
|
@@ -6872,6 +6912,8 @@ class WebRTCClient extends Emitter {
|
|
|
6872
6912
|
skipRegister: this.skipRegister,
|
|
6873
6913
|
};
|
|
6874
6914
|
|
|
6915
|
+
this.forceClosed = false;
|
|
6916
|
+
|
|
6875
6917
|
if (this.skipRegister) {
|
|
6876
6918
|
logger$8.info('sdk webrtc skip register...', logInfo);
|
|
6877
6919
|
return Promise.resolve();
|
|
@@ -6896,7 +6938,15 @@ class WebRTCClient extends Emitter {
|
|
|
6896
6938
|
const registerOptions = this._isWeb() ? {} : { extraContactHeaderParams: ['mobility=mobile'] };
|
|
6897
6939
|
|
|
6898
6940
|
const onRegisterFailed = () => {
|
|
6899
|
-
logger$8.info('sdk webrtc registering failed', {
|
|
6941
|
+
logger$8.info('sdk webrtc registering failed', {
|
|
6942
|
+
tries,
|
|
6943
|
+
registerer: !!this.registerer,
|
|
6944
|
+
forceClosed: this.forceClosed,
|
|
6945
|
+
});
|
|
6946
|
+
if (this.forceClosed) {
|
|
6947
|
+
return;
|
|
6948
|
+
}
|
|
6949
|
+
|
|
6900
6950
|
this.connectionPromise = null;
|
|
6901
6951
|
if (this.registerer && this.registerer.waiting) {
|
|
6902
6952
|
this.registerer.waitingToggle(false);
|
|
@@ -7174,8 +7224,9 @@ class WebRTCClient extends Emitter {
|
|
|
7174
7224
|
}
|
|
7175
7225
|
}
|
|
7176
7226
|
|
|
7177
|
-
async close() {
|
|
7178
|
-
logger$8.info('sdk webrtc closing client', { userAgent: !!this.userAgent });
|
|
7227
|
+
async close(force = false) {
|
|
7228
|
+
logger$8.info('sdk webrtc closing client', { userAgent: !!this.userAgent, force });
|
|
7229
|
+
this.forceClosed = force;
|
|
7179
7230
|
this._cleanupMedia();
|
|
7180
7231
|
this.connectionPromise = null;
|
|
7181
7232
|
|
|
@@ -7194,7 +7245,7 @@ class WebRTCClient extends Emitter {
|
|
|
7194
7245
|
this.userAgent.delegate = null;
|
|
7195
7246
|
this.userAgent.stateChange.removeAllListeners();
|
|
7196
7247
|
|
|
7197
|
-
await this._disconnectTransport();
|
|
7248
|
+
await this._disconnectTransport(force);
|
|
7198
7249
|
|
|
7199
7250
|
this._cleanupRegister();
|
|
7200
7251
|
|
|
@@ -8630,7 +8681,15 @@ class WebRTCClient extends Emitter {
|
|
|
8630
8681
|
return userAgent.UserAgent.makeURI(`sip:${target}@${this.config.host}`);
|
|
8631
8682
|
}
|
|
8632
8683
|
|
|
8633
|
-
async _disconnectTransport() {
|
|
8684
|
+
async _disconnectTransport(force = false) {
|
|
8685
|
+
if (force) {
|
|
8686
|
+
// Bypass sip.js state machine that prevent to close WS with the state `Connecting`
|
|
8687
|
+
this.userAgent.transport.disconnectResolve = () => {};
|
|
8688
|
+
// eslint-disable-next-line
|
|
8689
|
+
this.userAgent.transport._ws.close(1000);
|
|
8690
|
+
return;
|
|
8691
|
+
}
|
|
8692
|
+
|
|
8634
8693
|
// Check if `disconnectPromise` is not present to avoid `Disconnect promise must not be defined` errors.
|
|
8635
8694
|
if (this.userAgent && this.userAgent.transport && !this.userAgent.transport.disconnectPromise) {
|
|
8636
8695
|
try {
|
|
@@ -9367,6 +9426,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9367
9426
|
}
|
|
9368
9427
|
|
|
9369
9428
|
callSession.answerTime = new Date();
|
|
9429
|
+
this._updateCallSession(callSession);
|
|
9370
9430
|
this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
|
|
9371
9431
|
|
|
9372
9432
|
return callSession;
|
|
@@ -9576,6 +9636,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9576
9636
|
return Promise.resolve(null);
|
|
9577
9637
|
}
|
|
9578
9638
|
|
|
9639
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
9640
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
9641
|
+
logger$7.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
9642
|
+
return Promise.resolve(null);
|
|
9643
|
+
}
|
|
9644
|
+
|
|
9579
9645
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9580
9646
|
logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9581
9647
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -9583,6 +9649,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9583
9649
|
|
|
9584
9650
|
this.shouldSendReinvite = false;
|
|
9585
9651
|
callSession.answerTime = new Date();
|
|
9652
|
+
this._updateCallSession(callSession);
|
|
9586
9653
|
this.acceptedSessions[callSession.getId()] = true;
|
|
9587
9654
|
|
|
9588
9655
|
this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
|
|
@@ -9682,7 +9749,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9682
9749
|
// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
|
|
9683
9750
|
if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
|
|
9684
9751
|
logger$7.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
|
|
9685
|
-
this.holdSipSession(this.currentSipSession,
|
|
9752
|
+
this.holdSipSession(this.currentSipSession, this.currentCallSession);
|
|
9686
9753
|
}
|
|
9687
9754
|
|
|
9688
9755
|
const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
|
|
@@ -9893,6 +9960,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9893
9960
|
}
|
|
9894
9961
|
const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
|
|
9895
9962
|
callSession.creationTime = new Date();
|
|
9963
|
+
this._updateCallSession(callSession);
|
|
9896
9964
|
|
|
9897
9965
|
this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
|
|
9898
9966
|
|
|
@@ -10034,7 +10102,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10034
10102
|
return this.client && this.client.isConference(callSession.sipCallId);
|
|
10035
10103
|
}
|
|
10036
10104
|
|
|
10037
|
-
async close() {
|
|
10105
|
+
async close(force = false) {
|
|
10038
10106
|
logger$7.info('WebRTC close');
|
|
10039
10107
|
|
|
10040
10108
|
try {
|
|
@@ -10046,7 +10114,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10046
10114
|
logger$7.error('WebRTC close, unregister error', e);
|
|
10047
10115
|
}
|
|
10048
10116
|
|
|
10049
|
-
this.client.close();
|
|
10117
|
+
this.client.close(force);
|
|
10050
10118
|
this.unbind();
|
|
10051
10119
|
|
|
10052
10120
|
this.incomingSessions = [];
|
|
@@ -10172,6 +10240,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10172
10240
|
}
|
|
10173
10241
|
|
|
10174
10242
|
callSession.creationTime = new Date();
|
|
10243
|
+
this._updateCallSession(callSession);
|
|
10244
|
+
|
|
10175
10245
|
this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
|
|
10176
10246
|
});
|
|
10177
10247
|
|
|
@@ -10196,9 +10266,17 @@ class WebRTCPhone extends Emitter {
|
|
|
10196
10266
|
});
|
|
10197
10267
|
|
|
10198
10268
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
10199
|
-
|
|
10269
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10270
|
+
const hasSession = sessionId in this.callSessions;
|
|
10200
10271
|
|
|
10201
|
-
|
|
10272
|
+
logger$7.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10273
|
+
|
|
10274
|
+
if (!hasSession) {
|
|
10275
|
+
logger$7.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10276
|
+
return;
|
|
10277
|
+
}
|
|
10278
|
+
|
|
10279
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10202
10280
|
|
|
10203
10281
|
if (this.audioOutputDeviceId) {
|
|
10204
10282
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10215,6 +10293,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10215
10293
|
logger$7.info('WebRTC call rejected', session.id);
|
|
10216
10294
|
const callSession = this._createCallSession(session);
|
|
10217
10295
|
callSession.endTime = new Date();
|
|
10296
|
+
this._updateCallSession(callSession);
|
|
10297
|
+
|
|
10218
10298
|
this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
|
|
10219
10299
|
});
|
|
10220
10300
|
|
|
@@ -10332,9 +10412,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10332
10412
|
return this.client.getSipSessionId(sipSession);
|
|
10333
10413
|
}
|
|
10334
10414
|
|
|
10335
|
-
|
|
10415
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10336
10416
|
const sipSession = this.findSipSession(callSession);
|
|
10337
|
-
|
|
10417
|
+
|
|
10418
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10419
|
+
}
|
|
10420
|
+
|
|
10421
|
+
_updateCallSession(callSession ) {
|
|
10422
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10338
10423
|
|
|
10339
10424
|
if (sipSessionId) {
|
|
10340
10425
|
this.callSessions[sipSessionId] = callSession;
|
|
@@ -10439,6 +10524,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10439
10524
|
displayName: identity ? identity.displayName || number : number,
|
|
10440
10525
|
startTime: fromSession ? fromSession.startTime : new Date(),
|
|
10441
10526
|
creationTime: fromSession ? fromSession.creationTime : null,
|
|
10527
|
+
answerTime: fromSession ? fromSession.answerTime : null,
|
|
10442
10528
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10443
10529
|
answered: state === sessionState.SessionState.Established,
|
|
10444
10530
|
paused: this.client.isCallHeld(sipSession),
|