@wazo/sdk 0.37.33 → 0.37.36
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/README.md +1 -0
- package/dist/wazo-sdk.js +2 -2
- package/esm/__tests__/api-client.test.js +17 -0
- package/esm/__tests__/web-rtc-client.test.js +18 -1
- package/esm/api/auth.js +5 -0
- package/esm/api/call-logd.js +6 -0
- package/esm/api/chatd.js +6 -0
- package/esm/api/confd.js +6 -0
- package/esm/api-client.js +17 -0
- package/esm/checker/Checker.js +18 -1
- package/esm/checker/checks/aor.js +17 -0
- package/esm/checker/checks/api.js +17 -0
- package/esm/checker/checks/index.js +18 -1
- package/esm/checker/checks/webrtc-transport.js +18 -1
- package/esm/checker/checks/webrtc.js +18 -1
- package/esm/domain/AdHocAPIConference.js +21 -0
- package/esm/domain/CallLog.js +6 -0
- package/esm/domain/CallSession.js +14 -0
- package/esm/domain/Features.js +17 -0
- package/esm/domain/Phone/CTIPhone.js +31 -0
- package/esm/domain/Phone/WebRTCPhone.js +65 -15
- package/esm/domain/Profile.js +6 -0
- package/esm/domain/__tests__/CallLog.test.js +10 -0
- package/esm/domain/__tests__/CallSession.test.js +14 -0
- package/esm/domain/__tests__/IndirectTransfer.test.js +14 -0
- package/esm/domain/__tests__/Profile.test.js +6 -0
- package/esm/domain/__tests__/Room.test.js +14 -0
- package/esm/domain/__tests__/Session.test.js +6 -0
- package/esm/index.js +93 -22
- package/esm/service/CallApi.js +17 -0
- package/esm/service/getApiClient.js +17 -0
- package/esm/simple/Auth.js +93 -22
- package/esm/simple/Configuration.js +93 -22
- package/esm/simple/Directory.js +17 -0
- package/esm/simple/Phone.js +93 -22
- package/esm/simple/index.js +93 -22
- package/esm/simple/room/LocalParticipant.js +93 -22
- package/esm/simple/room/Participant.js +93 -22
- package/esm/simple/room/RemoteParticipant.js +93 -22
- package/esm/simple/room/Room.js +93 -22
- package/esm/simple/room/SipRoom.js +93 -22
- package/esm/simple/room/Stream.js +93 -22
- package/esm/simple/room/__tests__/RemoteParticipant.test.js +93 -22
- package/esm/simple/utils.js +93 -22
- package/esm/web-rtc-client.js +18 -1
- package/lib/__tests__/api-client.test.js +17 -0
- package/lib/__tests__/web-rtc-client.test.js +18 -1
- package/lib/api/auth.js +5 -0
- package/lib/api/call-logd.js +6 -0
- package/lib/api/chatd.js +6 -0
- package/lib/api/confd.js +6 -0
- package/lib/api-client.js +17 -0
- package/lib/checker/Checker.js +18 -1
- package/lib/checker/checks/aor.js +17 -0
- package/lib/checker/checks/api.js +17 -0
- package/lib/checker/checks/index.js +18 -1
- package/lib/checker/checks/webrtc-transport.js +18 -1
- package/lib/checker/checks/webrtc.js +18 -1
- package/lib/domain/AdHocAPIConference.js +21 -0
- package/lib/domain/CallLog.js +6 -0
- package/lib/domain/CallSession.js +14 -0
- package/lib/domain/Features.js +17 -0
- package/lib/domain/Phone/CTIPhone.js +31 -0
- package/lib/domain/Phone/WebRTCPhone.js +65 -15
- package/lib/domain/Profile.js +6 -0
- package/lib/domain/__tests__/CallLog.test.js +10 -0
- package/lib/domain/__tests__/CallSession.test.js +14 -0
- package/lib/domain/__tests__/IndirectTransfer.test.js +14 -0
- package/lib/domain/__tests__/Profile.test.js +6 -0
- package/lib/domain/__tests__/Room.test.js +14 -0
- package/lib/domain/__tests__/Session.test.js +6 -0
- package/lib/index.js +93 -22
- package/lib/service/CallApi.js +17 -0
- package/lib/service/getApiClient.js +17 -0
- package/lib/simple/Auth.js +93 -22
- package/lib/simple/Configuration.js +93 -22
- package/lib/simple/Directory.js +17 -0
- package/lib/simple/Phone.js +93 -22
- package/lib/simple/index.js +93 -22
- package/lib/simple/room/LocalParticipant.js +93 -22
- package/lib/simple/room/Participant.js +93 -22
- package/lib/simple/room/RemoteParticipant.js +93 -22
- package/lib/simple/room/Room.js +93 -22
- package/lib/simple/room/SipRoom.js +93 -22
- package/lib/simple/room/Stream.js +93 -22
- package/lib/simple/room/__tests__/RemoteParticipant.test.js +93 -22
- package/lib/simple/utils.js +93 -22
- package/lib/web-rtc-client.js +18 -1
- package/package.json +1 -1
- package/src/api/auth.js +5 -0
- package/src/domain/AdHocAPIConference.js +4 -0
- package/src/domain/CallLog.js +6 -0
- package/src/domain/CallSession.js +14 -0
- package/src/domain/Phone/WebRTCPhone.js +51 -15
- package/src/domain/Profile.js +6 -0
- package/src/domain/__tests__/CallLog.test.js +4 -0
- package/src/simple/Phone.js +6 -6
- package/src/web-rtc-client.js +1 -1
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@@ -384,6 +384,7 @@ const LINE_STATE = {
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@@ -420,6 +421,7 @@ const LINE_STATE = {
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@@ -451,6 +453,7 @@ class Profile {
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@@ -467,6 +470,7 @@ class Profile {
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lines: plain.lines.map(line => Line.parse(line)),
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username: plain.username,
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mobileNumber: plain.mobile_phone_number || '',
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ringSeconds: plain.ring_seconds,
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forwards: [
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ForwardOption.parse(plain.forwards.unconditional, FORWARD_KEYS.UNCONDITIONAL),
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ForwardOption.parse(plain.forwards.noanswer, FORWARD_KEYS.NO_ANSWER),
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@@ -503,6 +507,7 @@ class Profile {
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switchboards,
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agent,
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status,
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ringSeconds,
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sipLines,
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callPickupTargetUsers,
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onlineCallRecordEnabled,
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this.status = status;
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this.callPickupTargetUsers = callPickupTargetUsers;
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this.onlineCallRecordEnabled = onlineCallRecordEnabled;
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this.ringSeconds = ringSeconds;
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this.sipLines = sipLines || [];
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}
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@@ -2216,6 +2222,7 @@ var authMethods = (client , baseUrl ) => ({
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) {
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const body = {
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backend: params.backend || DEFAULT_BACKEND_USER,
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@@ -2235,6 +2242,10 @@ var authMethods = (client , baseUrl ) => ({
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headers['Wazo-Session-Type'] = 'mobile';
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}
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if (params.tenantId) {
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body.tenant_id = params.tenantId;
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}
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return client.post(`${baseUrl}/token`, body, headers).then(response => Session.parse(response));
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},
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class CallSession {
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videoRemotelyDowngraded,
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sipSession,
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answerTime,
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conference,
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} ) {
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this.callId = callId;
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this.sipCallId = sipCallId;
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this.videoRemotelyDowngraded = videoRemotelyDowngraded;
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this.sipSession = sipSession;
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this.answerTime = answerTime || this.answerTime;
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this.conference = conference;
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this.type = 'CallSession';
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this.screensharing = false;
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}
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setIsConference(conference ) {
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this.conference = conference;
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}
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isIncoming() {
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return !this.isCaller && !this.answered;
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}
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return this.sipStatus === SessionState.Terminated;
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}
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isConference() {
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return this.conference;
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}
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getElapsedTimeInSeconds() {
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if (!this.startTime) {
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return 0;
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cameraEnabled: false,
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dialedExtension: call.dialedExtension,
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call,
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conference: false, // @FIXME?
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});
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}
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}
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|| plain.destination_extension
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|| `meeting-${plain.requested_name || plain.destination_name || ''}`,
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name: plain.requested_name || plain.destination_name || '',
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uuid: plain.destination_user_uuid,
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},
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source: {
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extension: plain.source_extension,
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id: plain.id,
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duration: (plain.duration || 0) * 1000, // duration is in seconds
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this.conferenceId = conferenceId || '';
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this.muted = muted || false;
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this.paused = paused || false;
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}
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}
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async start() {
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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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accept(callSession , cameraEnabled ) {
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logger$7.info('WebRTC phone - accept call', { id: callSession ? callSession.getId() : 'n/a', cameraEnabled });
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if (this.currentSipSession) {
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this.holdSipSession(
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this.holdSipSession(
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this.currentSipSession,
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this.currentCallSession,
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true,
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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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hold(callSession , withEvent = true
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hold(callSession , withEvent = true) {
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logger$7.info('WebRTC hold', { id: callSession.getId() });
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const sipSession = this.findSipSession(callSession);
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return sipSession ? this.holdSipSession(sipSession, callSession, withEvent
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return sipSession ? this.holdSipSession(sipSession, callSession, withEvent) : null;
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// @Deprecated
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unhold(callSession , withEvent = true
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unhold(callSession , withEvent = true) {
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resume(callSession , withEvent = true) {
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logger$7.info('WebRTC resume called', { id: callSession ? callSession.getId() : null });
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const sipSession = this.findSipSession(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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this.holdSipSession(
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this.holdSipSession(
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}
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const promise = this.unholdSipSession(sipSession, callSession, withEvent
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const promise = this.unholdSipSession(sipSession, callSession, withEvent);
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this.currentSipSession = sipSession;
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@@ -9749,7 +9798,6 @@ class WebRTCPhone extends Emitter {
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9749
9798
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sipSession ,
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9750
9799
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callSession ,
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9751
9800
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withEvent = true,
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9752
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-
isConference = false,
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9753
9801
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) {
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9754
9802
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if (!sipSession) {
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9755
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return new Promise((resolve, reject) => reject(new Error('No session to hold')));
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@@ -9770,6 +9818,8 @@ class WebRTCPhone extends Emitter {
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9770
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await this.client.downgradeToAudio(sipSession);
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9771
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}
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9772
9820
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9821
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+
const isConference = !!callSession && callSession.isConference();
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9822
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+
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9773
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const promise = this.client.hold(sipSession, isConference, hasVideo);
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9774
9824
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if (withEvent) {
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9775
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this.eventEmitter.emit(ON_CALL_HELD, this._createCallSession(sipSession, callSession));
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@@ -9782,11 +9832,13 @@ class WebRTCPhone extends Emitter {
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9782
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sipSession ,
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9783
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callSession ,
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9784
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withEvent = true,
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9785
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-
isConference = false,
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9786
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) {
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9787
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if (!sipSession) {
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9788
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return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
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9789
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}
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9839
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+
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9840
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+
const isConference = !!callSession && callSession.isConference();
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9841
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+
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9790
9842
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const sessionId = this.getSipSessionId(sipSession);
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9791
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logger$7.info('WebRTC unhold sip session', { sessionId, isConference });
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9792
9844
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@@ -9928,7 +9980,11 @@ class WebRTCPhone extends Emitter {
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9928
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await this.client.register();
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9929
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}
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9930
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if (this.currentSipSession) {
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9931
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-
this.holdSipSession(
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9983
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+
this.holdSipSession(
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9984
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+
this.currentSipSession,
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9985
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+
this.currentCallSession,
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9986
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+
true,
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9987
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+
).catch(e => {
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9932
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logger$7.warn('Unable to hold current session when making another call', e);
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9933
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});
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9934
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}
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@@ -9943,6 +9999,7 @@ class WebRTCPhone extends Emitter {
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9943
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return Promise.resolve(null);
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9944
10000
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}
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9945
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const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
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10002
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+
callSession.setIsConference(conference);
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9946
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callSession.creationTime = new Date();
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9947
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this._updateCallSession(callSession);
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9948
10005
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@@ -10250,9 +10307,17 @@ class WebRTCPhone extends Emitter {
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10250
10307
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});
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10251
10308
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10252
10309
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this.client.on(this.client.ACCEPTED, (sipSession ) => {
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10253
|
-
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|
10310
|
+
const sessionId = this.getSipSessionId(sipSession);
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10311
|
+
const hasSession = sessionId in this.callSessions;
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10254
10312
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|
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10255
|
-
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10313
|
+
logger$7.info('WebRTC call accepted', { sipId: sessionId, hasSession });
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10314
|
+
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|
10315
|
+
if (!hasSession) {
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10316
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+
logger$7.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
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10317
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+
return;
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10318
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+
}
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10319
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+
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10320
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
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10256
10321
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10257
10322
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if (this.audioOutputDeviceId) {
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10258
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this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
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@@ -10388,9 +10453,14 @@ class WebRTCPhone extends Emitter {
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10388
10453
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return this.client.getSipSessionId(sipSession);
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10389
10454
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}
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10390
10455
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10391
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-
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10456
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+
getSipSessionIdFromCallSession(callSession ) {
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10392
10457
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const sipSession = this.findSipSession(callSession);
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10393
|
-
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10458
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+
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10459
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+
return sipSession ? this.getSipSessionId(sipSession) : null;
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10460
|
+
}
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10461
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+
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10462
|
+
_updateCallSession(callSession ) {
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10463
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
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10394
10464
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10395
10465
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if (sipSessionId) {
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10396
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this.callSessions[sipSessionId] = callSession;
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@@ -10508,6 +10578,7 @@ class WebRTCPhone extends Emitter {
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10508
10578
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recording: fromSession ? fromSession.isRecording() : false,
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10509
10579
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recordingPaused: false, // @TODO
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10510
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sipSession,
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10581
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+
conference: !!fromSession && fromSession.isConference(),
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10511
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...extra,
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10512
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});
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10513
10584
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@@ -10764,16 +10835,16 @@ class Phone extends Emitter {
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10764
10835
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}
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10765
10836
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}
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10766
10837
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10767
|
-
hold(callSession
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10768
|
-
return this.phone && this.phone.hold(callSession, true
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10838
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+
hold(callSession ) {
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10839
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+
return this.phone && this.phone.hold(callSession, true);
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10769
10840
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}
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10770
10841
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10771
|
-
unhold(callSession
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10772
|
-
return this.phone && this.phone.unhold(callSession, true
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10842
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+
unhold(callSession ) {
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10843
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+
return this.phone && this.phone.unhold(callSession, true);
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10773
10844
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}
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10774
10845
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10775
|
-
resume(callSession
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|
10776
|
-
return this.phone && this.phone.resume(callSession
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10846
|
+
resume(callSession ) {
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10847
|
+
return this.phone && this.phone.resume(callSession);
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10777
10848
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}
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10778
10849
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10779
10850
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reject(callSession ) {
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