@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
package/esm/checker/Checker.js
CHANGED
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@@ -1722,6 +1722,12 @@ 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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+
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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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@@ -2346,9 +2352,17 @@ class WebRTCPhone extends Emitter {
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});
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this.client.on(this.client.ACCEPTED, (sipSession ) => {
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-
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const sessionId = this.getSipSessionId(sipSession);
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const hasSession = sessionId in this.callSessions;
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logger.info('WebRTC call accepted', { sipId: sessionId, hasSession });
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if (!hasSession) {
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logger.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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@@ -2484,9 +2498,14 @@ 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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package/esm/index.js
CHANGED
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@@ -6225,7 +6225,7 @@ 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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@@ -10089,6 +10089,12 @@ 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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@@ -10713,9 +10719,17 @@ 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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@@ -10851,9 +10865,14 @@ 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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package/esm/simple/Auth.js
CHANGED
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@@ -6399,7 +6399,7 @@ 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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@@ -9362,6 +9362,12 @@ 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$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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@@ -9986,9 +9992,17 @@ 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$8.info('WebRTC call accepted', { sipId: sessionId, hasSession });
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if (!hasSession) {
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logger$8.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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@@ -10124,9 +10138,14 @@ 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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@@ -6399,7 +6399,7 @@ 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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@@ -9362,6 +9362,12 @@ 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$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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@@ -9986,9 +9992,17 @@ 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$8.info('WebRTC call accepted', { sipId: sessionId, hasSession });
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+
if (!hasSession) {
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logger$8.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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@@ -10124,9 +10138,14 @@ 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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package/esm/simple/Phone.js
CHANGED
|
@@ -6399,7 +6399,7 @@ 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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6403
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6405
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@@ -9362,6 +9362,12 @@ 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$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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9369
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+
}
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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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|
@@ -9986,9 +9992,17 @@ class WebRTCPhone extends Emitter {
|
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});
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this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
9989
|
-
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9995
|
+
const sessionId = this.getSipSessionId(sipSession);
|
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9996
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+
const hasSession = sessionId in this.callSessions;
|
|
9997
|
+
|
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9998
|
+
logger$8.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
9999
|
+
|
|
10000
|
+
if (!hasSession) {
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10001
|
+
logger$8.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
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|
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+
return;
|
|
10003
|
+
}
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|
9990
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|
|
|
9991
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10005
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
9992
10006
|
|
|
9993
10007
|
if (this.audioOutputDeviceId) {
|
|
9994
10008
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10124,9 +10138,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10124
10138
|
return this.client.getSipSessionId(sipSession);
|
|
10125
10139
|
}
|
|
10126
10140
|
|
|
10127
|
-
|
|
10141
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10128
10142
|
const sipSession = this.findSipSession(callSession);
|
|
10129
|
-
|
|
10143
|
+
|
|
10144
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10145
|
+
}
|
|
10146
|
+
|
|
10147
|
+
_updateCallSession(callSession ) {
|
|
10148
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10130
10149
|
|
|
10131
10150
|
if (sipSessionId) {
|
|
10132
10151
|
this.callSessions[sipSessionId] = callSession;
|
package/esm/simple/index.js
CHANGED
|
@@ -6663,7 +6663,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6663
6663
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6664
6664
|
class CanceledCallError extends Error {}
|
|
6665
6665
|
|
|
6666
|
-
const MAX_REGISTER_TRIES =
|
|
6666
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6667
6667
|
|
|
6668
6668
|
|
|
6669
6669
|
|
|
@@ -9626,6 +9626,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9626
9626
|
return Promise.resolve(null);
|
|
9627
9627
|
}
|
|
9628
9628
|
|
|
9629
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
9630
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
9631
|
+
logger$7.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
9632
|
+
return Promise.resolve(null);
|
|
9633
|
+
}
|
|
9634
|
+
|
|
9629
9635
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9630
9636
|
logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9631
9637
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -10250,9 +10256,17 @@ class WebRTCPhone extends Emitter {
|
|
|
10250
10256
|
});
|
|
10251
10257
|
|
|
10252
10258
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
10253
|
-
|
|
10259
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10260
|
+
const hasSession = sessionId in this.callSessions;
|
|
10261
|
+
|
|
10262
|
+
logger$7.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10263
|
+
|
|
10264
|
+
if (!hasSession) {
|
|
10265
|
+
logger$7.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10266
|
+
return;
|
|
10267
|
+
}
|
|
10254
10268
|
|
|
10255
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10269
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10256
10270
|
|
|
10257
10271
|
if (this.audioOutputDeviceId) {
|
|
10258
10272
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10388,9 +10402,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10388
10402
|
return this.client.getSipSessionId(sipSession);
|
|
10389
10403
|
}
|
|
10390
10404
|
|
|
10391
|
-
|
|
10405
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10392
10406
|
const sipSession = this.findSipSession(callSession);
|
|
10393
|
-
|
|
10407
|
+
|
|
10408
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10409
|
+
}
|
|
10410
|
+
|
|
10411
|
+
_updateCallSession(callSession ) {
|
|
10412
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10394
10413
|
|
|
10395
10414
|
if (sipSessionId) {
|
|
10396
10415
|
this.callSessions[sipSessionId] = callSession;
|
|
@@ -6663,7 +6663,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6663
6663
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6664
6664
|
class CanceledCallError extends Error {}
|
|
6665
6665
|
|
|
6666
|
-
const MAX_REGISTER_TRIES =
|
|
6666
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6667
6667
|
|
|
6668
6668
|
|
|
6669
6669
|
|
|
@@ -9626,6 +9626,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9626
9626
|
return Promise.resolve(null);
|
|
9627
9627
|
}
|
|
9628
9628
|
|
|
9629
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
9630
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
9631
|
+
logger$7.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
9632
|
+
return Promise.resolve(null);
|
|
9633
|
+
}
|
|
9634
|
+
|
|
9629
9635
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9630
9636
|
logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9631
9637
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -10250,9 +10256,17 @@ class WebRTCPhone extends Emitter {
|
|
|
10250
10256
|
});
|
|
10251
10257
|
|
|
10252
10258
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
10253
|
-
|
|
10259
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10260
|
+
const hasSession = sessionId in this.callSessions;
|
|
10261
|
+
|
|
10262
|
+
logger$7.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10263
|
+
|
|
10264
|
+
if (!hasSession) {
|
|
10265
|
+
logger$7.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10266
|
+
return;
|
|
10267
|
+
}
|
|
10254
10268
|
|
|
10255
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10269
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10256
10270
|
|
|
10257
10271
|
if (this.audioOutputDeviceId) {
|
|
10258
10272
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10388,9 +10402,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10388
10402
|
return this.client.getSipSessionId(sipSession);
|
|
10389
10403
|
}
|
|
10390
10404
|
|
|
10391
|
-
|
|
10405
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10392
10406
|
const sipSession = this.findSipSession(callSession);
|
|
10393
|
-
|
|
10407
|
+
|
|
10408
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10409
|
+
}
|
|
10410
|
+
|
|
10411
|
+
_updateCallSession(callSession ) {
|
|
10412
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10394
10413
|
|
|
10395
10414
|
if (sipSessionId) {
|
|
10396
10415
|
this.callSessions[sipSessionId] = callSession;
|
|
@@ -6663,7 +6663,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6663
6663
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6664
6664
|
class CanceledCallError extends Error {}
|
|
6665
6665
|
|
|
6666
|
-
const MAX_REGISTER_TRIES =
|
|
6666
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6667
6667
|
|
|
6668
6668
|
|
|
6669
6669
|
|
|
@@ -9626,6 +9626,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9626
9626
|
return Promise.resolve(null);
|
|
9627
9627
|
}
|
|
9628
9628
|
|
|
9629
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
9630
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
9631
|
+
logger$7.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
9632
|
+
return Promise.resolve(null);
|
|
9633
|
+
}
|
|
9634
|
+
|
|
9629
9635
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9630
9636
|
logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9631
9637
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -10250,9 +10256,17 @@ class WebRTCPhone extends Emitter {
|
|
|
10250
10256
|
});
|
|
10251
10257
|
|
|
10252
10258
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
10253
|
-
|
|
10259
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10260
|
+
const hasSession = sessionId in this.callSessions;
|
|
10261
|
+
|
|
10262
|
+
logger$7.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10263
|
+
|
|
10264
|
+
if (!hasSession) {
|
|
10265
|
+
logger$7.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10266
|
+
return;
|
|
10267
|
+
}
|
|
10254
10268
|
|
|
10255
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10269
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10256
10270
|
|
|
10257
10271
|
if (this.audioOutputDeviceId) {
|
|
10258
10272
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10388,9 +10402,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10388
10402
|
return this.client.getSipSessionId(sipSession);
|
|
10389
10403
|
}
|
|
10390
10404
|
|
|
10391
|
-
|
|
10405
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10392
10406
|
const sipSession = this.findSipSession(callSession);
|
|
10393
|
-
|
|
10407
|
+
|
|
10408
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10409
|
+
}
|
|
10410
|
+
|
|
10411
|
+
_updateCallSession(callSession ) {
|
|
10412
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10394
10413
|
|
|
10395
10414
|
if (sipSessionId) {
|
|
10396
10415
|
this.callSessions[sipSessionId] = callSession;
|
|
@@ -6663,7 +6663,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6663
6663
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6664
6664
|
class CanceledCallError extends Error {}
|
|
6665
6665
|
|
|
6666
|
-
const MAX_REGISTER_TRIES =
|
|
6666
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6667
6667
|
|
|
6668
6668
|
|
|
6669
6669
|
|
|
@@ -9626,6 +9626,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9626
9626
|
return Promise.resolve(null);
|
|
9627
9627
|
}
|
|
9628
9628
|
|
|
9629
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
9630
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
9631
|
+
logger$7.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
9632
|
+
return Promise.resolve(null);
|
|
9633
|
+
}
|
|
9634
|
+
|
|
9629
9635
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9630
9636
|
logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9631
9637
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -10250,9 +10256,17 @@ class WebRTCPhone extends Emitter {
|
|
|
10250
10256
|
});
|
|
10251
10257
|
|
|
10252
10258
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
10253
|
-
|
|
10259
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10260
|
+
const hasSession = sessionId in this.callSessions;
|
|
10261
|
+
|
|
10262
|
+
logger$7.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10263
|
+
|
|
10264
|
+
if (!hasSession) {
|
|
10265
|
+
logger$7.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10266
|
+
return;
|
|
10267
|
+
}
|
|
10254
10268
|
|
|
10255
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10269
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10256
10270
|
|
|
10257
10271
|
if (this.audioOutputDeviceId) {
|
|
10258
10272
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
|
|
@@ -10388,9 +10402,14 @@ class WebRTCPhone extends Emitter {
|
|
|
10388
10402
|
return this.client.getSipSessionId(sipSession);
|
|
10389
10403
|
}
|
|
10390
10404
|
|
|
10391
|
-
|
|
10405
|
+
getSipSessionIdFromCallSession(callSession ) {
|
|
10392
10406
|
const sipSession = this.findSipSession(callSession);
|
|
10393
|
-
|
|
10407
|
+
|
|
10408
|
+
return sipSession ? this.getSipSessionId(sipSession) : null;
|
|
10409
|
+
}
|
|
10410
|
+
|
|
10411
|
+
_updateCallSession(callSession ) {
|
|
10412
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
10394
10413
|
|
|
10395
10414
|
if (sipSessionId) {
|
|
10396
10415
|
this.callSessions[sipSessionId] = callSession;
|
package/esm/simple/room/Room.js
CHANGED
|
@@ -6663,7 +6663,7 @@ const events$1 = [REGISTERED, UNREGISTERED, REGISTRATION_FAILED, INVITE];
|
|
|
6663
6663
|
const transportEvents = [CONNECTED, DISCONNECTED, TRANSPORT_ERROR, MESSAGE];
|
|
6664
6664
|
class CanceledCallError extends Error {}
|
|
6665
6665
|
|
|
6666
|
-
const MAX_REGISTER_TRIES =
|
|
6666
|
+
const MAX_REGISTER_TRIES = 5;
|
|
6667
6667
|
|
|
6668
6668
|
|
|
6669
6669
|
|
|
@@ -9626,6 +9626,12 @@ class WebRTCPhone extends Emitter {
|
|
|
9626
9626
|
return Promise.resolve(null);
|
|
9627
9627
|
}
|
|
9628
9628
|
|
|
9629
|
+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
|
|
9630
|
+
if (sipSessionId && !(sipSessionId in this.callSessions)) {
|
|
9631
|
+
logger$7.warn('Call session already ended or not found, can\'t accept it.', { sipSessionId });
|
|
9632
|
+
return Promise.resolve(null);
|
|
9633
|
+
}
|
|
9634
|
+
|
|
9629
9635
|
if (callSession.getId() in this.acceptedSessions) {
|
|
9630
9636
|
logger$7.warn('CallSession already accepted.', { id: callSession ? callSession.getId() : 'n/a' });
|
|
9631
9637
|
return Promise.resolve(callSession.sipCallId);
|
|
@@ -10250,9 +10256,17 @@ class WebRTCPhone extends Emitter {
|
|
|
10250
10256
|
});
|
|
10251
10257
|
|
|
10252
10258
|
this.client.on(this.client.ACCEPTED, (sipSession ) => {
|
|
10253
|
-
|
|
10259
|
+
const sessionId = this.getSipSessionId(sipSession);
|
|
10260
|
+
const hasSession = sessionId in this.callSessions;
|
|
10261
|
+
|
|
10262
|
+
logger$7.info('WebRTC call accepted', { sipId: sessionId, hasSession });
|
|
10263
|
+
|
|
10264
|
+
if (!hasSession) {
|
|
10265
|
+
logger$7.warn('Call accepted ignored, session is no longer present in the WebRtcPhone', { sessionId });
|
|
10266
|
+
return;
|
|
10267
|
+
}
|
|
10254
10268
|
|
|
10255
|
-
this._onCallAccepted(sipSession, this.client.hasVideo(
|
|
10269
|
+
this._onCallAccepted(sipSession, this.client.hasVideo(sessionId));
|
|
10256
10270
|
|
|
10257
10271
|
if (this.audioOutputDeviceId) {
|
|
10258
10272
|
this.client.changeAudioOutputDevice(this.audioOutputDeviceId);
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@@ -10388,9 +10402,14 @@ class WebRTCPhone extends Emitter {
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10388
10402
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return this.client.getSipSessionId(sipSession);
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10389
10403
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}
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10390
10404
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10391
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-
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10405
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+
getSipSessionIdFromCallSession(callSession ) {
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10392
10406
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const sipSession = this.findSipSession(callSession);
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10393
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-
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10407
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+
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10408
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+
return sipSession ? this.getSipSessionId(sipSession) : null;
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10409
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+
}
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10410
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+
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10411
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+
_updateCallSession(callSession ) {
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10412
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+
const sipSessionId = this.getSipSessionIdFromCallSession(callSession);
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10394
10413
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10395
10414
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if (sipSessionId) {
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10396
10415
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this.callSessions[sipSessionId] = callSession;
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