@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
package/lib/simple/Phone.js
CHANGED
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@@ -285,6 +285,7 @@ var updateFrom = (instance , from ) => {
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class CallSession {
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@@ -347,6 +348,8 @@ class CallSession {
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@@ -378,6 +381,7 @@ 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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@@ -406,6 +410,7 @@ class CallSession {
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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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// Useful to compare instead of instanceof with minified code
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this.type = 'CallSession';
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@@ -463,6 +468,10 @@ class 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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@@ -539,6 +548,10 @@ class CallSession {
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return this.sipStatus === sessionState.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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@@ -1670,6 +1684,7 @@ const LINE_STATE = {
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@@ -1737,6 +1753,7 @@ class Profile {
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@@ -1753,6 +1770,7 @@ 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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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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) {
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const body = {
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backend: params.backend || DEFAULT_BACKEND_USER,
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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(`${baseUrl}/token`, body, headers).then(response => Session.parse(response));
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},
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|| plain.destination_extension
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|| `meeting-${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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name: plain.source_name,
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uuid: plain.source_user_uuid,
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},
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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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if (this.host) {
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this.host.setIsConference(true);
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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$8.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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}
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if (!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$8.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$8.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
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return Promise.resolve(callSession.sipCallId);
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}
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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$8.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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}
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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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console.warn('Please note that `phone.unhold()` is being deprecated; `phone.resume()` is the preferred method');
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return this.resume(callSession,
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return this.resume(callSession, withEvent);
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}
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resume(callSession ,
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resume(callSession , withEvent = true) {
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logger$8.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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if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
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logger$8.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
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this.holdSipSession(
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this.holdSipSession(
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withEvent,
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);
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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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if (callSession) {
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this.currentCallSession = callSession;
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@@ -9495,7 +9544,6 @@ class WebRTCPhone extends Emitter {
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9495
9544
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sipSession ,
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9496
9545
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callSession ,
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9497
9546
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withEvent = true,
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9498
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-
isConference = false,
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9499
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) {
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9500
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if (!sipSession) {
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9501
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return new Promise((resolve, reject) => reject(new Error('No session to hold')));
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@@ -9516,6 +9564,8 @@ class WebRTCPhone extends Emitter {
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9516
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await this.client.downgradeToAudio(sipSession);
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9517
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}
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9518
9566
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9567
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+
const isConference = !!callSession && callSession.isConference();
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9568
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+
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9519
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const promise = this.client.hold(sipSession, isConference, hasVideo);
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9520
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if (withEvent) {
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9521
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this.eventEmitter.emit(ON_CALL_HELD, this._createCallSession(sipSession, callSession));
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@@ -9528,11 +9578,13 @@ class WebRTCPhone extends Emitter {
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9528
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sipSession ,
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9529
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callSession ,
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withEvent = true,
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9531
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-
isConference = false,
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9532
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) {
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9533
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if (!sipSession) {
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9534
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return new Promise((resolve, reject) => reject(new Error('No session to unhold')));
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9535
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}
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9585
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+
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9586
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+
const isConference = !!callSession && callSession.isConference();
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9587
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+
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9536
9588
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const sessionId = this.getSipSessionId(sipSession);
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logger$8.info('WebRTC unhold sip session', { sessionId, isConference });
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9538
9590
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@@ -9674,7 +9726,11 @@ class WebRTCPhone extends Emitter {
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9674
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await this.client.register();
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9675
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}
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9676
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if (this.currentSipSession) {
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9677
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-
this.holdSipSession(
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9729
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+
this.holdSipSession(
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9730
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+
this.currentSipSession,
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9731
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+
this.currentCallSession,
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9732
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+
true,
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9733
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+
).catch(e => {
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9678
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logger$8.warn('Unable to hold current session when making another call', e);
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9679
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});
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9680
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}
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@@ -9689,6 +9745,7 @@ class WebRTCPhone extends Emitter {
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9689
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return Promise.resolve(null);
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9690
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}
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9691
9747
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const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
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9748
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+
callSession.setIsConference(conference);
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9692
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callSession.creationTime = new Date();
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9693
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this._updateCallSession(callSession);
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9694
9751
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@@ -9996,9 +10053,17 @@ class WebRTCPhone extends Emitter {
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9996
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});
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9997
10054
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9998
10055
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this.client.on(this.client.ACCEPTED, (sipSession ) => {
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9999
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-
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10056
|
+
const sessionId = this.getSipSessionId(sipSession);
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10057
|
+
const hasSession = sessionId in this.callSessions;
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10000
10058
|
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10001
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-
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10059
|
+
logger$8.info('WebRTC call accepted', { sipId: sessionId, hasSession });
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10060
|
+
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|
10061
|
+
if (!hasSession) {
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10062
|
+
logger$8.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
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10063
|
+
return;
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10064
|
+
}
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10065
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+
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10066
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
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10002
10067
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10003
10068
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if (this.audioOutputDeviceId) {
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10004
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this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
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@@ -10134,9 +10199,14 @@ class WebRTCPhone extends Emitter {
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10134
10199
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return this.client.getSipSessionId(sipSession);
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10135
10200
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}
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10136
10201
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10137
|
-
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10202
|
+
getSipSessionIdFromCallSession(callSession ) {
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10138
10203
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const sipSession = this.findSipSession(callSession);
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10139
|
-
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|
10204
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+
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10205
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
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10206
|
+
}
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|
10207
|
+
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|
10208
|
+
_updateCallSession(callSession ) {
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|
10209
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
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10140
10210
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10141
10211
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if (sipSessionId) {
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10142
10212
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this.callSessions[sipSessionId] = callSession;
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@@ -10254,6 +10324,7 @@ class WebRTCPhone extends Emitter {
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10254
10324
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recording: fromSession ? fromSession.isRecording() : false,
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10255
10325
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recordingPaused: false, // @TODO
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10256
10326
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sipSession,
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10327
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+
conference: !!fromSession && fromSession.isConference(),
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10257
10328
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...extra,
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10258
10329
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});
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10259
10330
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@@ -13671,16 +13742,16 @@ class Phone extends Emitter {
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13671
13742
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}
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13672
13743
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}
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13673
13744
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13674
|
-
hold(callSession
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|
13675
|
-
return this.phone && this.phone.hold(callSession, true
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13745
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+
hold(callSession ) {
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13746
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+
return this.phone && this.phone.hold(callSession, true);
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13676
13747
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}
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13677
13748
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13678
|
-
unhold(callSession
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|
13679
|
-
return this.phone && this.phone.unhold(callSession, true
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13749
|
+
unhold(callSession ) {
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13750
|
+
return this.phone && this.phone.unhold(callSession, true);
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13680
13751
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}
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13681
13752
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13682
|
-
resume(callSession
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|
13683
|
-
return this.phone && this.phone.resume(callSession
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|
13753
|
+
resume(callSession ) {
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|
13754
|
+
return this.phone && this.phone.resume(callSession);
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13684
13755
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}
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13685
13756
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13686
13757
|
reject(callSession ) {
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