@wazo/sdk 0.37.33 → 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 +2 -2
- package/esm/__tests__/web-rtc-client.test.js +1 -1
- package/esm/checker/Checker.js +1 -1
- package/esm/checker/checks/index.js +1 -1
- package/esm/checker/checks/webrtc-transport.js +1 -1
- package/esm/checker/checks/webrtc.js +1 -1
- package/esm/domain/Phone/WebRTCPhone.js +23 -4
- package/esm/index.js +24 -5
- package/esm/simple/Auth.js +24 -5
- package/esm/simple/Configuration.js +24 -5
- package/esm/simple/Phone.js +24 -5
- package/esm/simple/index.js +24 -5
- package/esm/simple/room/LocalParticipant.js +24 -5
- package/esm/simple/room/Participant.js +24 -5
- package/esm/simple/room/RemoteParticipant.js +24 -5
- package/esm/simple/room/Room.js +24 -5
- package/esm/simple/room/SipRoom.js +24 -5
- package/esm/simple/room/Stream.js +24 -5
- package/esm/simple/room/__tests__/RemoteParticipant.test.js +24 -5
- package/esm/simple/utils.js +24 -5
- package/esm/web-rtc-client.js +1 -1
- package/lib/__tests__/web-rtc-client.test.js +1 -1
- package/lib/checker/Checker.js +1 -1
- package/lib/checker/checks/index.js +1 -1
- package/lib/checker/checks/webrtc-transport.js +1 -1
- package/lib/checker/checks/webrtc.js +1 -1
- package/lib/domain/Phone/WebRTCPhone.js +23 -4
- package/lib/index.js +24 -5
- package/lib/simple/Auth.js +24 -5
- package/lib/simple/Configuration.js +24 -5
- package/lib/simple/Phone.js +24 -5
- package/lib/simple/index.js +24 -5
- package/lib/simple/room/LocalParticipant.js +24 -5
- package/lib/simple/room/Participant.js +24 -5
- package/lib/simple/room/RemoteParticipant.js +24 -5
- package/lib/simple/room/Room.js +24 -5
- package/lib/simple/room/SipRoom.js +24 -5
- package/lib/simple/room/Stream.js +24 -5
- package/lib/simple/room/__tests__/RemoteParticipant.test.js +24 -5
- package/lib/simple/utils.js +24 -5
- package/lib/web-rtc-client.js +1 -1
- package/package.json +1 -1
- package/src/domain/Phone/WebRTCPhone.js +23 -4
- package/src/web-rtc-client.js +1 -1
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@@ -6663,7 +6663,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
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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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@@ -9626,6 +9626,12 @@ class WebRTCPhone extends Emitter {
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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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@@ -10250,9 +10256,17 @@ class WebRTCPhone extends Emitter {
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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(
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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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@@ -10388,9 +10402,14 @@ class WebRTCPhone extends Emitter {
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return this.client.getSipSessionId(sipSession);
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}
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getSipSessionIdFromCallSession(callSession ) {
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const sipSession = this.findSipSession(callSession);
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return sipSession ? this.getSipSessionId(sipSession) : null;
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}
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_updateCallSession(callSession ) {
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const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
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if (sipSessionId) {
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this.callSessions[sipSessionId] = callSession;
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@@ -6663,7 +6663,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
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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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@@ -9626,6 +9626,12 @@ class WebRTCPhone extends Emitter {
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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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@@ -10250,9 +10256,17 @@ class WebRTCPhone extends Emitter {
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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(
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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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@@ -10388,9 +10402,14 @@ class WebRTCPhone extends Emitter {
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return this.client.getSipSessionId(sipSession);
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}
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-
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getSipSessionIdFromCallSession(callSession ) {
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const sipSession = this.findSipSession(callSession);
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return sipSession ? this.getSipSessionId(sipSession) : null;
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}
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_updateCallSession(callSession ) {
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const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
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if (sipSessionId) {
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this.callSessions[sipSessionId] = callSession;
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@@ -6663,7 +6663,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
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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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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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@@ -10250,9 +10256,17 @@ class WebRTCPhone extends Emitter {
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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(
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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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return this.client.getSipSessionId(sipSession);
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}
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getSipSessionIdFromCallSession(callSession ) {
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const sipSession = this.findSipSession(callSession);
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return sipSession ? this.getSipSessionId(sipSession) : null;
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}
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_updateCallSession(callSession ) {
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const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
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if (sipSessionId) {
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this.callSessions[sipSessionId] = callSession;
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package/esm/simple/utils.js
CHANGED
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@@ -6663,7 +6663,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
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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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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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@@ -10250,9 +10256,17 @@ class WebRTCPhone extends Emitter {
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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(
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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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@@ -10388,9 +10402,14 @@ class WebRTCPhone extends Emitter {
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return this.client.getSipSessionId(sipSession);
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}
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-
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getSipSessionIdFromCallSession(callSession ) {
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const sipSession = this.findSipSession(callSession);
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-
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return sipSession ? this.getSipSessionId(sipSession) : null;
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}
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_updateCallSession(callSession ) {
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const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
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if (sipSessionId) {
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this.callSessions[sipSessionId] = callSession;
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package/esm/web-rtc-client.js
CHANGED
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@@ -5365,7 +5365,7 @@ const events = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
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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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package/lib/checker/Checker.js
CHANGED
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@@ -1731,6 +1731,12 @@ class WebRTCPhone extends Emitter {
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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.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.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
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return Promise.resolve(callSession.sipCallId);
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@@ -2355,9 +2361,17 @@ class WebRTCPhone extends Emitter {
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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;
|
|
2366
|
+
|
|
2367
|
+
logger.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
2368
|
+
|
|
2369
|
+
if (!hasSession) {
|
|
2370
|
+
logger.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
2371
|
+
return;
|
|
2372
|
+
}
|
|
2359
2373
|
|
|
2360
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
2374
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
2361
2375
|
|
|
2362
2376
|
if (this.audioOutputDeviceId) {
|
|
2363
2377
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -2493,9 +2507,14 @@ class WebRTCPhone extends Emitter {
|
|
|
2493
2507
|
return this.client.getSipSessionId(sipSession);
|
|
2494
2508
|
}
|
|
2495
2509
|
|
|
2496
|
-
|
|
2510
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
2497
2511
|
const sipSession = this.findSipSession(callSession);
|
|
2498
|
-
|
|
2512
|
+
|
|
2513
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
2514
|
+
}
|
|
2515
|
+
|
|
2516
|
+
_updateCallSession(callSession ) {
|
|
2517
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
2499
2518
|
|
|
2500
2519
|
if (sipSessionId) {
|
|
2501
2520
|
this.callSessions[sipSessionId] = callSession;
|
package/lib/index.js
CHANGED
|
@@ -6236,7 +6236,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6236
6236
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6237
6237
|
class CanceledCallError extends Error {}
|
|
6238
6238
|
|
|
6239
|
-
const MAX_REGISTER_TRIES =
|
|
6239
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6240
6240
|
|
|
6241
6241
|
|
|
6242
6242
|
|
|
@@ -10100,6 +10100,12 @@ class WebRTCPhone extends Emitter {
|
|
|
10100
10100
|
return Promise.resolve(null);
|
|
10101
10101
|
}
|
|
10102
10102
|
|
|
10103
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10104
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
10105
|
+
logger$7.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
10106
|
+
return Promise.resolve(null);
|
|
10107
|
+
}
|
|
10108
|
+
|
|
10103
10109
|
if (callSession.getId() in this.acceptedSessions) {
|
|
10104
10110
|
logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
10105
10111
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -10724,9 +10730,17 @@ class WebRTCPhone extends Emitter {
|
|
|
10724
10730
|
});
|
|
10725
10731
|
|
|
10726
10732
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
10727
|
-
|
|
10733
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10734
|
+
const hasSession = sessionId in this.callSessions;
|
|
10735
|
+
|
|
10736
|
+
logger$7.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10737
|
+
|
|
10738
|
+
if (!hasSession) {
|
|
10739
|
+
logger$7.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10740
|
+
return;
|
|
10741
|
+
}
|
|
10728
10742
|
|
|
10729
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10743
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10730
10744
|
|
|
10731
10745
|
if (this.audioOutputDeviceId) {
|
|
10732
10746
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10862,9 +10876,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10862
10876
|
return this.client.getSipSessionId(sipSession);
|
|
10863
10877
|
}
|
|
10864
10878
|
|
|
10865
|
-
|
|
10879
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10866
10880
|
const sipSession = this.findSipSession(callSession);
|
|
10867
|
-
|
|
10881
|
+
|
|
10882
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10883
|
+
}
|
|
10884
|
+
|
|
10885
|
+
_updateCallSession(callSession ) {
|
|
10886
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10868
10887
|
|
|
10869
10888
|
if (sipSessionId) {
|
|
10870
10889
|
this.callSessions[sipSessionId] = callSession;
|
package/lib/simple/Auth.js
CHANGED
|
@@ -6411,7 +6411,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6411
6411
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6412
6412
|
class CanceledCallError extends Error {}
|
|
6413
6413
|
|
|
6414
|
-
const MAX_REGISTER_TRIES =
|
|
6414
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6415
6415
|
|
|
6416
6416
|
|
|
6417
6417
|
|
|
@@ -9374,6 +9374,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9374
9374
|
return Promise.resolve(null);
|
|
9375
9375
|
}
|
|
9376
9376
|
|
|
9377
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
9378
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
9379
|
+
logger$8.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
9380
|
+
return Promise.resolve(null);
|
|
9381
|
+
}
|
|
9382
|
+
|
|
9377
9383
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9378
9384
|
logger$8.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9379
9385
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -9998,9 +10004,17 @@ class WebRTCPhone extends Emitter {
|
|
|
9998
10004
|
});
|
|
9999
10005
|
|
|
10000
10006
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
10001
|
-
|
|
10007
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10008
|
+
const hasSession = sessionId in this.callSessions;
|
|
10009
|
+
|
|
10010
|
+
logger$8.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10011
|
+
|
|
10012
|
+
if (!hasSession) {
|
|
10013
|
+
logger$8.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10014
|
+
return;
|
|
10015
|
+
}
|
|
10002
10016
|
|
|
10003
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10017
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10004
10018
|
|
|
10005
10019
|
if (this.audioOutputDeviceId) {
|
|
10006
10020
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10136,9 +10150,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10136
10150
|
return this.client.getSipSessionId(sipSession);
|
|
10137
10151
|
}
|
|
10138
10152
|
|
|
10139
|
-
|
|
10153
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10140
10154
|
const sipSession = this.findSipSession(callSession);
|
|
10141
|
-
|
|
10155
|
+
|
|
10156
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10157
|
+
}
|
|
10158
|
+
|
|
10159
|
+
_updateCallSession(callSession ) {
|
|
10160
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10142
10161
|
|
|
10143
10162
|
if (sipSessionId) {
|
|
10144
10163
|
this.callSessions[sipSessionId] = callSession;
|
|
@@ -6409,7 +6409,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6409
6409
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6410
6410
|
class CanceledCallError extends Error {}
|
|
6411
6411
|
|
|
6412
|
-
const MAX_REGISTER_TRIES =
|
|
6412
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6413
6413
|
|
|
6414
6414
|
|
|
6415
6415
|
|
|
@@ -9372,6 +9372,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9372
9372
|
return Promise.resolve(null);
|
|
9373
9373
|
}
|
|
9374
9374
|
|
|
9375
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
9376
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
9377
|
+
logger$8.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
9378
|
+
return Promise.resolve(null);
|
|
9379
|
+
}
|
|
9380
|
+
|
|
9375
9381
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9376
9382
|
logger$8.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9377
9383
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -9996,9 +10002,17 @@ class WebRTCPhone extends Emitter {
|
|
|
9996
10002
|
});
|
|
9997
10003
|
|
|
9998
10004
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
9999
|
-
|
|
10005
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10006
|
+
const hasSession = sessionId in this.callSessions;
|
|
10007
|
+
|
|
10008
|
+
logger$8.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10009
|
+
|
|
10010
|
+
if (!hasSession) {
|
|
10011
|
+
logger$8.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10012
|
+
return;
|
|
10013
|
+
}
|
|
10000
10014
|
|
|
10001
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10015
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10002
10016
|
|
|
10003
10017
|
if (this.audioOutputDeviceId) {
|
|
10004
10018
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10134,9 +10148,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10134
10148
|
return this.client.getSipSessionId(sipSession);
|
|
10135
10149
|
}
|
|
10136
10150
|
|
|
10137
|
-
|
|
10151
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10138
10152
|
const sipSession = this.findSipSession(callSession);
|
|
10139
|
-
|
|
10153
|
+
|
|
10154
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10155
|
+
}
|
|
10156
|
+
|
|
10157
|
+
_updateCallSession(callSession ) {
|
|
10158
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10140
10159
|
|
|
10141
10160
|
if (sipSessionId) {
|
|
10142
10161
|
this.callSessions[sipSessionId] = callSession;
|
package/lib/simple/Phone.js
CHANGED
|
@@ -6409,7 +6409,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6409
6409
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6410
6410
|
class CanceledCallError extends Error {}
|
|
6411
6411
|
|
|
6412
|
-
const MAX_REGISTER_TRIES =
|
|
6412
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6413
6413
|
|
|
6414
6414
|
|
|
6415
6415
|
|
|
@@ -9372,6 +9372,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9372
9372
|
return Promise.resolve(null);
|
|
9373
9373
|
}
|
|
9374
9374
|
|
|
9375
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
9376
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
9377
|
+
logger$8.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
9378
|
+
return Promise.resolve(null);
|
|
9379
|
+
}
|
|
9380
|
+
|
|
9375
9381
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9376
9382
|
logger$8.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9377
9383
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -9996,9 +10002,17 @@ class WebRTCPhone extends Emitter {
|
|
|
9996
10002
|
});
|
|
9997
10003
|
|
|
9998
10004
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
9999
|
-
|
|
10005
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10006
|
+
const hasSession = sessionId in this.callSessions;
|
|
10007
|
+
|
|
10008
|
+
logger$8.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10009
|
+
|
|
10010
|
+
if (!hasSession) {
|
|
10011
|
+
logger$8.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10012
|
+
return;
|
|
10013
|
+
}
|
|
10000
10014
|
|
|
10001
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10015
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10002
10016
|
|
|
10003
10017
|
if (this.audioOutputDeviceId) {
|
|
10004
10018
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10134,9 +10148,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10134
10148
|
return this.client.getSipSessionId(sipSession);
|
|
10135
10149
|
}
|
|
10136
10150
|
|
|
10137
|
-
|
|
10151
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10138
10152
|
const sipSession = this.findSipSession(callSession);
|
|
10139
|
-
|
|
10153
|
+
|
|
10154
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10155
|
+
}
|
|
10156
|
+
|
|
10157
|
+
_updateCallSession(callSession ) {
|
|
10158
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10140
10159
|
|
|
10141
10160
|
if (sipSessionId) {
|
|
10142
10161
|
this.callSessions[sipSessionId] = callSession;
|
package/lib/simple/index.js
CHANGED
|
@@ -6673,7 +6673,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6673
6673
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6674
6674
|
class CanceledCallError extends Error {}
|
|
6675
6675
|
|
|
6676
|
-
const MAX_REGISTER_TRIES =
|
|
6676
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6677
6677
|
|
|
6678
6678
|
|
|
6679
6679
|
|
|
@@ -9636,6 +9636,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9636
9636
|
return Promise.resolve(null);
|
|
9637
9637
|
}
|
|
9638
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
|
+
|
|
9639
9645
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9640
9646
|
logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9641
9647
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -10260,9 +10266,17 @@ class WebRTCPhone extends Emitter {
|
|
|
10260
10266
|
});
|
|
10261
10267
|
|
|
10262
10268
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
10263
|
-
|
|
10269
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10270
|
+
const hasSession = sessionId in this.callSessions;
|
|
10271
|
+
|
|
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
|
+
}
|
|
10264
10278
|
|
|
10265
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10279
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10266
10280
|
|
|
10267
10281
|
if (this.audioOutputDeviceId) {
|
|
10268
10282
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10398,9 +10412,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10398
10412
|
return this.client.getSipSessionId(sipSession);
|
|
10399
10413
|
}
|
|
10400
10414
|
|
|
10401
|
-
|
|
10415
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10402
10416
|
const sipSession = this.findSipSession(callSession);
|
|
10403
|
-
|
|
10417
|
+
|
|
10418
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10419
|
+
}
|
|
10420
|
+
|
|
10421
|
+
_updateCallSession(callSession ) {
|
|
10422
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10404
10423
|
|
|
10405
10424
|
if (sipSessionId) {
|
|
10406
10425
|
this.callSessions[sipSessionId] = callSession;
|