@wazo/sdk 0.43.8 → 0.43.10
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 +6 -6
- package/esm/domain/Phone/WebRTCPhone.js +32 -27
- package/esm/types/domain/Phone/WebRTCPhone.d.ts +1 -1
- package/esm/types/domain/Phone/WebRTCPhone.d.ts.map +1 -1
- package/esm/types/web-rtc-client.d.ts +1 -1
- package/esm/types/web-rtc-client.d.ts.map +1 -1
- package/esm/web-rtc-client.js +35 -28
- package/lib/domain/Phone/WebRTCPhone.js +32 -27
- package/lib/types/domain/Phone/WebRTCPhone.d.ts +1 -1
- package/lib/types/domain/Phone/WebRTCPhone.d.ts.map +1 -1
- package/lib/types/web-rtc-client.d.ts +1 -1
- package/lib/types/web-rtc-client.d.ts.map +1 -1
- package/lib/web-rtc-client.js +35 -28
- package/package.json +1 -1
- package/src/domain/Phone/WebRTCPhone.ts +34 -29
- package/src/web-rtc-client.ts +37 -30
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@@ -122,7 +122,7 @@ class WebRTCPhone extends Emitter_1.default {
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return __awaiter(this, arguments, void 0, function* (callSession, constraints = null, conference = false, audioOnly = false, iceRestart = false) {
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const sipSession = this.findSipSession(callSession);
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logger.info('WebRTC phone - send reinvite', {
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-
sessionId: sipSession
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+
sessionId: this.getSipSessionId(sipSession),
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constraints,
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audioOnly,
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conference,
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@@ -301,7 +301,9 @@ class WebRTCPhone extends Emitter_1.default {
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onCancel(message);
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const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
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const callSession = this._createCallSession(sipSession);
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-
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if (sipSessionId) {
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delete this.callSessions[sipSessionId];
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}
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this.eventEmitter.emit(exports.ON_CALL_CANCELED, callSession, elsewhere);
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};
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}
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@@ -309,14 +311,14 @@ class WebRTCPhone extends Emitter_1.default {
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switch (newState) {
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case session_state_1.SessionState.Terminating:
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logger.info('WebRTC phone - call terminating', {
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-
sipId: sipSession
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+
sipId: this.getSipSessionId(sipSession),
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});
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this.eventEmitter.emit(exports.ON_CALL_ENDING, this._createCallSession(sipSession));
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break;
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case session_state_1.SessionState.Terminated:
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{
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logger.info('WebRTC phone - call terminated', {
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-
sipId: sipSession
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+
sipId: this.getSipSessionId(sipSession),
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});
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// Should be called before `_onCallTerminated` or the callCount will not decrement...
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const callSession = this._createCallSession(sipSession);
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@@ -472,13 +474,13 @@ class WebRTCPhone extends Emitter_1.default {
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}
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_onCallAccepted(sipSession, cameraEnabled) {
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logger.info('WebRTC phone - on call accepted', {
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sipId: sipSession
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+
sipId: this.getSipSessionId(sipSession),
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cameraEnabled,
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});
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const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);
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if (this.currentSipSession && !this.isCurrentCallSipSession(callSession)) {
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logger.info('WebRTC phone - on call accepted with another ongoing call, holding it', {
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-
sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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cameraEnabled,
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});
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this.holdSipSession(this.currentSipSession, this.currentCallSession, true);
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@@ -551,7 +553,7 @@ class WebRTCPhone extends Emitter_1.default {
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_onCallTerminated(sipSession) {
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const sipSessionId = this.getSipSessionId(sipSession);
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logger.info('WebRTC phone - on call terminated', {
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-
sipId: sipSession
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+
sipId: this.getSipSessionId(sipSession),
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sipSessionId,
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ids: Object.keys(this.callSessions),
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});
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@@ -571,8 +573,8 @@ class WebRTCPhone extends Emitter_1.default {
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}
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if (sipSessionId) {
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this.removeIncomingSessions(sipSessionId);
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delete this.callSessions[sipSessionId];
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}
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-
delete this.callSessions[sipSessionId];
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if (isCurrentSession) {
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this.currentSipSession = undefined;
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this.currentCallSession = undefined;
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@@ -580,7 +582,7 @@ class WebRTCPhone extends Emitter_1.default {
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const hasIncomingCallSession = this.hasIncomingCallSession();
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// Re-trigger incoming call event for remaining incoming calls
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logger.info('WebRTC phone - check to re-trigger incoming call', {
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-
sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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shouldRetrigger,
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hasIncomingCallSession,
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});
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@@ -695,7 +697,7 @@ class WebRTCPhone extends Emitter_1.default {
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this.eventEmitter.emit(exports.ON_CALL_ANSWERED, callSession);
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const sipSession = this.client.getSipSession(callSession.getId());
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if (sipSession) {
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logger.info('accept call, session found', { sipId: sipSession
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logger.info('accept call, session found', { sipId: this.getSipSessionId(sipSession) });
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return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
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return callSession.sipCallId;
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}).catch(e => {
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@@ -731,7 +733,7 @@ class WebRTCPhone extends Emitter_1.default {
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const sipSession = this.findSipSession(callSession);
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if (sipSession) {
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logger.info('WebRTC rejecting', {
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sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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});
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this.client.hangup(sipSession);
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}
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@@ -752,7 +754,7 @@ class WebRTCPhone extends Emitter_1.default {
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const sipSession = this.findSipSession(callSession);
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if (sipSession) {
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logger.info('WebRTC atxfer', {
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sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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});
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return this.client.atxfer(sipSession);
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}
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@@ -778,12 +780,12 @@ class WebRTCPhone extends Emitter_1.default {
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return new Promise((resolve, reject) => reject(new Error('No session to resume')));
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}
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logger.info('WebRTC resuming', {
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sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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});
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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
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if (this.currentSipSession && this.getSipSessionId(this.currentSipSession) !== this.getSipSessionId(sipSession)) {
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logger.info('WebRTC hold call after resume', {
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id: this.currentSipSession
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id: this.getSipSessionId(this.currentSipSession),
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});
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this.holdSipSession(this.currentSipSession, this.currentCallSession, withEvent);
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}
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@@ -802,7 +804,7 @@ class WebRTCPhone extends Emitter_1.default {
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const sessionId = this.getSipSessionId(sipSession);
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const hasVideo = this.client.hasLocalVideo(sessionId);
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logger.info('WebRTC hold sip session', {
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sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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hasVideo,
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});
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// Stop screenshare if needed
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return;
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}
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logger.info('WebRTC muting', {
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sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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});
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this.client.mute(sipSession);
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const newCallSession = this._createCallSession(sipSession, callSession, {
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return;
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}
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logger.info('WebRTC unmuting', {
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sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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});
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this.client.unmute(sipSession);
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const newCallSession = this._createCallSession(sipSession, callSession, {
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return;
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}
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logger.info('WebRTC turn camera on', {
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sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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});
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this.client.toggleCameraOn(sipSession);
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this.eventEmitter.emit(exports.ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
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return;
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}
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logger.info('WebRTC turn camera off', {
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sipId: this.getSipSessionId(sipSession),
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});
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this.client.toggleCameraOff(sipSession);
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this.eventEmitter.emit(exports.ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
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return;
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}
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logger.info('WebRTC send key', {
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sipId: this.getSipSessionId(sipSession),
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tone,
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});
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this.client.sendDTMF(sipSession, tone);
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return;
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}
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logger.info('WebRTC transfer', {
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sipId: this.getSipSessionId(sipSession),
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target,
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});
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this.client.transfer(sipSession, target);
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const sipSession = this.client.getSipSession(source.sipCallId);
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const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
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logger.info('WebRTC indirect transfer', {
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sipId: sipSession ? sipSession
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target: sipSessionTarget ? sipSessionTarget
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sipId: sipSession ? this.getSipSessionId(sipSession) : null,
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target: sipSessionTarget ? this.getSipSessionId(sipSessionTarget) : null,
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});
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if (!sipSessionTarget) {
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return Promise.reject();
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}
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const sipSessionId = this.getSipSessionId(sipSession);
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logger.info('WebRTC hangup', {
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sipId: this.getSipSessionId(sipSession),
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sipSessionId,
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});
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yield this.client.hangup(sipSession);
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@@ -1217,7 +1219,7 @@ class WebRTCPhone extends Emitter_1.default {
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const autoAnswer = sipSession.request.getHeader('Answer-Mode') === 'Auto';
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const withVideo = this.allowVideo ? wantsToDoVideo : false;
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logger.info('WebRTC invite received', {
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sipId: sipSession
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sipId: this.getSipSessionId(sipSession),
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withVideo,
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autoAnswer,
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});
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this.eventEmitter.emit(exports.ON_CALL_ERROR, e);
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});
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this.client.on(this.client.REJECTED, session => {
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logger.info('WebRTC call rejected', session
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logger.info('WebRTC call rejected', { id: this.getSipSessionId(session) });
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const callSession = this._createCallSession(session);
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// `_createCallSession` add the CallSession in `this.callSessions`.
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// so we have to remove it or `callCount` will always return 1 even if the call was rejected
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@@ -1393,6 +1395,9 @@ class WebRTCPhone extends Emitter_1.default {
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return this.callSessions[sipSessionId];
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}
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getSipSessionId(sipSession) {
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if (!sipSession) {
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return '';
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}
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return this.client.getSipSessionId(sipSession);
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}
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getSipSessionIdFromCallSession(callSession) {
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@@ -164,7 +164,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
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bindClientEvents(): void;
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findSipSession(callSession?: CallSession): WazoSession | null | undefined;
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getCallSession(sipSessionId: string): CallSession | null | undefined;
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getSipSessionId(sipSession: WazoSession): string;
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getSipSessionId(sipSession: WazoSession | null | undefined): string;
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getSipSessionIdFromCallSession(callSession: CallSession): string | null;
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_updateCallSession(callSession: CallSession): void;
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_onMessage(message: Message & {
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@@ -1 +1 @@
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1
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-
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@@ -3,10 +3,10 @@ import type { InvitationRejectOptions } from 'sip.js/lib/api/invitation-reject-o
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|
3
3
|
import type { InviterCancelOptions } from 'sip.js/lib/api/inviter-cancel-options';
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4
4
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import type { SessionByeOptions } from 'sip.js/lib/api/session-bye-options';
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|
5
5
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import type { InvitationAcceptOptions } from 'sip.js/lib/api/invitation-accept-options';
|
|
6
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-
import type { SessionDialog } from 'sip.js/lib/core/dialogs/session-dialog';
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7
6
|
import { UserAgentState } from 'sip.js/lib/api/user-agent-state';
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8
7
|
import { URI } from 'sip.js/lib/grammar/uri';
|
|
9
8
|
import { UserAgent } from 'sip.js/lib/api/user-agent';
|
|
9
|
+
import { SessionDialog } from 'sip.js/lib/core/dialogs/session-dialog';
|
|
10
10
|
import { OutgoingByeRequest, OutgoingInviteRequest, OutgoingRequest } from 'sip.js/lib/core';
|
|
11
11
|
import { Inviter, Invitation, Registerer } from 'sip.js/lib/api';
|
|
12
12
|
import Emitter from './utils/Emitter';
|
|
@@ -1 +1 @@
|
|
|
1
|
-
{"version":3,"file":"web-rtc-client.d.ts","sourceRoot":"","sources":["../../src/web-rtc-client.ts"],"names":[],"mappings":"AAGA,OAAO,gBAAgB,CAAC;AAExB,OAAO,KAAK,EAAE,uBAAuB,EAAE,MAAM,0CAA0C,CAAC;AACxF,OAAO,KAAK,EAAE,oBAAoB,EAAE,MAAM,uCAAuC,CAAC;AAClF,OAAO,KAAK,EAAE,iBAAiB,EAAE,MAAM,oCAAoC,CAAC;AAC5E,OAAO,KAAK,EAAE,uBAAuB,EAAE,MAAM,0CAA0C,CAAC;
|
|
1
|
+
{"version":3,"file":"web-rtc-client.d.ts","sourceRoot":"","sources":["../../src/web-rtc-client.ts"],"names":[],"mappings":"AAGA,OAAO,gBAAgB,CAAC;AAExB,OAAO,KAAK,EAAE,uBAAuB,EAAE,MAAM,0CAA0C,CAAC;AACxF,OAAO,KAAK,EAAE,oBAAoB,EAAE,MAAM,uCAAuC,CAAC;AAClF,OAAO,KAAK,EAAE,iBAAiB,EAAE,MAAM,oCAAoC,CAAC;AAC5E,OAAO,KAAK,EAAE,uBAAuB,EAAE,MAAM,0CAA0C,CAAC;AAKxF,OAAO,EAAE,cAAc,EAAE,MAAM,iCAAiC,CAAC;AAGjE,OAAO,EAAE,GAAG,EAAE,MAAM,wBAAwB,CAAC;AAC7C,OAAO,EAAE,SAAS,EAAE,MAAM,2BAA2B,CAAC;AAEtD,OAAO,EAAE,aAAa,EAAE,MAAM,wCAAwC,CAAC;AASvE,OAAO,EAAE,kBAAkB,EAAE,qBAAqB,EAAE,eAAe,EAAE,MAAM,iBAAiB,CAAC;AAC7F,OAAO,EAAE,OAAO,EAAE,UAAU,EAAE,UAAU,EAA6C,MAAM,gBAAgB,CAAC;AAE5G,OAAO,OAAO,MAAM,iBAAiB,CAAC;AAGtC,OAAO,SAAS,MAAM,mBAAmB,CAAC;AAG1C,OAAO,KAAK,EAAE,WAAW,EAAE,wBAAwB,EAAE,YAAY,EAAE,gBAAgB,EAAoB,cAAc,EAAE,WAAW,EAAiB,MAAM,gBAAgB,CAAC;AAG1K,eAAO,MAAM,sBAAsB,gBAAiB,OAAO,MAAM,EAAE,GAAG,CAAC,iBAGrE,CAAC;AAyBH,eAAO,MAAM,MAAM,UAA0D,CAAC;AAC9E,eAAO,MAAM,eAAe,UAAsD,CAAC;AACnF,qBAAa,iBAAkB,SAAQ,KAAK;CAAG;AAM/C,MAAM,CAAC,OAAO,OAAO,YAAa,SAAQ,OAAO;IAC/C,QAAQ,EAAE,MAAM,CAAC;IAEjB,MAAM,EAAE,YAAY,CAAC;IAErB,iBAAiB,CAAC,EAAE,wBAAwB,CAAC;IAE7C,SAAS,EAAE,SAAS,GAAG,IAAI,CAAC;IAE5B,UAAU,EAAE,UAAU,GAAG,IAAI,CAAC;IAE9B,QAAQ,EAAE,OAAO,CAAC;IAElB,KAAK,EAAE,uBAAuB,GAAG,OAAO,GAAG,SAAS,CAAC;IAErD,aAAa,EAAE,MAAM,CAAC,MAAM,EAAE,gBAAgB,CAAC,CAAC;IAEhD,KAAK,EAAE,uBAAuB,GAAG,OAAO,GAAG,SAAS,CAAC;IAErD,YAAY,EAAE,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,CAAC;IAElC,mBAAmB,EAAE,MAAM,GAAG,IAAI,GAAG,SAAS,CAAC;IAE/C,iBAAiB,EAAE,MAAM,CAAC;IAE1B,YAAY,EAAE,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,CAAC;IAElC,iBAAiB,EAAE,OAAO,CAAC,IAAI,CAAC,GAAG,IAAI,GAAG,SAAS,CAAC;IAEpD,iBAAiB,EAAE,CAAC,GAAG,IAAI,EAAE,KAAK,CAAC,GAAG,CAAC,KAAK,GAAG,CAAC;IAEhD,SAAS,EAAE,SAAS,CAAC;IAErB,kBAAkB,EAAE,CAAC,CAAC,GAAG,IAAI,EAAE,KAAK,CAAC,GAAG,CAAC,KAAK,GAAG,CAAC,GAAG,IAAI,GAAG,SAAS,CAAC;IAEtE,WAAW,EAAE,CAAC,CAAC,GAAG,IAAI,EAAE,KAAK,CAAC,GAAG,CAAC,KAAK,GAAG,CAAC,GAAG,IAAI,GAAG,SAAS,CAAC;IAE/D,cAAc,EAAE,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,CAAC;IAEpC,WAAW,EAAE,MAAM,CAAC,MAAM,EAAE,WAAW,CAAC,CAAC;IAEzC,WAAW,EAAE,MAAM,CAAC,MAAM,EAAE,OAAO,CAAC,CAAC;IAErC,YAAY,EAAE,OAAO,CAAC;IAEtB,yBAAyB,EAAE,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,CAAC;IAE/C,mBAAmB,EAAE,MAAM,CAAC,MAAM,EAAE,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,CAAC,CAAC;IAEzD,WAAW,EAAE,OAAO,CAAC;IAGrB,aAAa,EAAE,MAAM,CAAC;IAEtB,UAAU,EAAE,MAAM,CAAC;IAEnB,YAAY,EAAE,MAAM,CAAC;IAErB,mBAAmB,EAAE,MAAM,CAAC;IAE5B,MAAM,EAAE,MAAM,CAAC;IAEf,SAAS,EAAE,MAAM,CAAC;IAElB,YAAY,EAAE,MAAM,CAAC;IAErB,eAAe,EAAE,MAAM,CAAC;IAExB,OAAO,EAAE,MAAM,CAAC;IAEhB,QAAQ,EAAE,MAAM,CAAC;IAEjB,QAAQ,EAAE,MAAM,CAAC;IAEjB,QAAQ,EAAE,MAAM,CAAC;IAEjB,WAAW,EAAE,MAAM,CAAC;IAEpB,QAAQ,EAAE,MAAM,CAAC;IAEjB,0BAA0B,EAAE,MAAM,CAAC;IAEnC,gBAAgB,EAAE,MAAM,CAAC;IAEzB,cAAc,EAAE,MAAM,CAAC;IAEvB,WAAW,EAAE,MAAM,CAAC;IAEpB,eAAe,EAAE,MAAM,CAAC;IAExB,MAAM,CAAC,YAAY,CAAC,EAAE,EAAE,MAAM,GAAG,OAAO;IAKxC,MAAM,CAAC,aAAa,CAAC,EAAE,EAAE,MAAM,GAAG,KAAK,CAAC;QACtC,IAAI,EAAE,KAAK,CAAC,MAAM,CAAC,CAAC;KACrB,CAAC;gBAUU,MAAM,EAAE,YAAY,EAAE,OAAO,CAAC,EAAE,WAAW,EAAE,iBAAiB,CAAC,EAAE,wBAAwB;IA4DrG,cAAc,CAAC,KAAK,EAAE,WAAW;IAMjC,mBAAmB,CAAC,KAAK,EAAE,sBAAsB;IAKjD,eAAe,CAAC,iBAAiB,CAAC,EAAE,wBAAwB,GAAG,SAAS;IAqDxE,WAAW,IAAI,OAAO;IAItB,YAAY,IAAI,OAAO;IAIvB,YAAY,IAAI,OAAO;IAIvB,SAAS;IAaH,YAAY,CAAC,KAAK,CAAC,EAAE,KAAK;IAkB1B,QAAQ,CAAC,KAAK,SAAI,GAAG,OAAO,CAAC,IAAI,CAAC;IA+GxC,sBAAsB,CAAC,UAAU,EAAE,UAAU,GAAG,IAAI,GAAG,SAAS,GAAG,IAAI;IA8BjE,UAAU,IAAI,OAAO,CAAC,IAAI,CAAC;IA2CjC,IAAI,IAAI,OAAO,CAAC,IAAI,CAAC;IAiBrB,IAAI,CAAC,MAAM,EAAE,MAAM,EAAE,WAAW,CAAC,EAAE,OAAO,EAAE,SAAS,UAAQ,EAAE,UAAU,UAAQ,EAAE,OAAO,KAAK,GAAG,WAAW,GAAG,IAAI;IAwFpH,MAAM,CAAC,OAAO,EAAE,UAAU,EAAE,WAAW,CAAC,EAAE,OAAO,GAAG,OAAO,CAAC,IAAI,CAAC;IAoC3D,MAAM,CAAC,OAAO,EAAE,WAAW,GAAG,OAAO,CAAC,kBAAkB,GAAG,IAAI,CAAC;IAiChE,QAAQ,CAAC,OAAO,EAAE,WAAW,GAAG,OAAO,CAAC,cAAc,GAAG,IAAI,CAAC;IAWpE,sBAAsB,CAAC,OAAO,EAAE,WAAW,EAAE,QAAQ,SAAO,GAAG,IAAI;IAUnE,qBAAqB,CAAC,OAAO,EAAE,WAAW,GAAG,IAAI;IAgB3C,MAAM,CAAC,OAAO,EAAE,WAAW,GAAG,OAAO,CAAC,IAAI,CAAC;IAmB3C,KAAK,CAAC,KAAK,UAAQ,GAAG,OAAO,CAAC,IAAI,CAAC;IA6CzC,SAAS,CAAC,OAAO,EAAE,OAAO,GAAG,MAAM,GAAG,IAAI,GAAG,SAAS;IAStD,IAAI,CAAC,OAAO,EAAE,WAAW,GAAG,IAAI;IAQhC,MAAM,CAAC,OAAO,EAAE,WAAW,GAAG,IAAI;IAQlC,YAAY,CAAC,OAAO,EAAE,WAAW,GAAG,OAAO;IAqC3C,cAAc,CAAC,OAAO,EAAE,WAAW,GAAG,IAAI;IAQ1C,eAAe,CAAC,OAAO,EAAE,WAAW,GAAG,IAAI;IAQ3C,IAAI,CAAC,OAAO,EAAE,WAAW,EAAE,YAAY,UAAQ,EAAE,QAAQ,UAAQ,GAAG,OAAO,CAAC,qBAAqB,GAAG,IAAI,CAAC;IA0DzG,MAAM,CAAC,OAAO,EAAE,WAAW,EAAE,YAAY,UAAQ,GAAG,OAAO,CAAC,qBAAqB,GAAG,IAAI,CAAC;IA4CnF,cAAc,CAAC,OAAO,EAAE,WAAW,EAAE,WAAW,EAAE,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,EAAE,YAAY,EAAE,OAAO,GAAG,OAAO,CAAC,WAAW,GAAG,SAAS,CAAC;IA+CrI,gBAAgB,CAAC,OAAO,EAAE,WAAW,GAAG,IAAI;IAqBtC,wBAAwB,CAAC,WAAW,EAAE,MAAM,CAAC,MAAM,EAAE,GAAG,CAAC,EAAE,UAAU,UAAQ,GAAG,OAAO,CAAC,WAAW,GAAG,IAAI,GAAG,SAAS,CAAC;IAoB7H,cAAc,CAAC,SAAS,EAAE,MAAM;IAIhC,UAAU,CAAC,OAAO,EAAE,WAAW,GAAG,OAAO;IAIzC,mBAAmB,CAAC,SAAS,EAAE,MAAM,GAAG,OAAO;IAY/C,QAAQ,CAAC,OAAO,EAAE,WAAW,EAAE,IAAI,EAAE,MAAM,GAAG,OAAO;IAYrD,OAAO,CAAC,WAAW,EAAE,MAAM,EAAE,OAAO,EAAE,MAAM,GAAG,IAAI;IAUnD,QAAQ,CAAC,OAAO,EAAE,WAAW,EAAE,MAAM,EAAE,MAAM,GAAG,IAAI;IAwBpD,MAAM,CAAC,O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|
package/lib/web-rtc-client.js
CHANGED
|
@@ -56,6 +56,7 @@ const constants_1 = require("sip.js/lib/core/messages/methods/constants");
|
|
|
56
56
|
const uri_1 = require("sip.js/lib/grammar/uri");
|
|
57
57
|
const user_agent_1 = require("sip.js/lib/api/user-agent");
|
|
58
58
|
const modifiers_1 = require("sip.js/lib/platform/web/modifiers/modifiers");
|
|
59
|
+
const session_dialog_1 = require("sip.js/lib/core/dialogs/session-dialog");
|
|
59
60
|
const messager_1 = require("sip.js/lib/api/messager");
|
|
60
61
|
const registerer_state_1 = require("sip.js/lib/api/registerer-state");
|
|
61
62
|
const session_state_1 = require("sip.js/lib/api/session-state");
|
|
@@ -73,7 +74,6 @@ const array_1 = require("./utils/array");
|
|
|
73
74
|
// We need to replace 0.0.0.0 to 127.0.0.1 in the sdp to avoid MOH during a createOffer.
|
|
74
75
|
const replaceLocalIpModifier = (description) => Promise.resolve(Object.assign(Object.assign({}, JSON.parse(JSON.stringify(description))), { sdp: description.sdp.replace('c=IN IP4 0.0.0.0', 'c=IN IP4 127.0.0.1') }));
|
|
75
76
|
exports.replaceLocalIpModifier = replaceLocalIpModifier;
|
|
76
|
-
const SIP_ID_LENGTH = 36;
|
|
77
77
|
const DEFAULT_ICE_TIMEOUT = 3000;
|
|
78
78
|
const SEND_STATS_DELAY = 5000;
|
|
79
79
|
const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
|
|
@@ -195,7 +195,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
195
195
|
logger.info('sdk webrtc on invite', {
|
|
196
196
|
method: 'delegate.onInvite',
|
|
197
197
|
clientId: this.clientId,
|
|
198
|
-
id: invitation
|
|
198
|
+
id: this.getSipSessionId(invitation),
|
|
199
199
|
remoteURI: invitation.remoteURI,
|
|
200
200
|
});
|
|
201
201
|
this._setupSession(invitation);
|
|
@@ -485,7 +485,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
485
485
|
},
|
|
486
486
|
onReject: (response) => {
|
|
487
487
|
logger.info('on call rejected', {
|
|
488
|
-
id: session
|
|
488
|
+
id: this.getSipSessionId(session),
|
|
489
489
|
// @ts-ignore: fromTag does not exist
|
|
490
490
|
fromTag: session === null || session === void 0 ? void 0 : session.fromTag,
|
|
491
491
|
});
|
|
@@ -515,7 +515,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
515
515
|
answer(session, enableVideo) {
|
|
516
516
|
logger.info('sdk webrtc answer call', {
|
|
517
517
|
clientId: this.clientId,
|
|
518
|
-
id: session
|
|
518
|
+
id: this.getSipSessionId(session),
|
|
519
519
|
enableVideo,
|
|
520
520
|
});
|
|
521
521
|
if (!session || !session.accept) {
|
|
@@ -531,7 +531,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
531
531
|
if (session.isCanceled) {
|
|
532
532
|
const message = 'accepted a canceled session (or was canceled during the accept phase).';
|
|
533
533
|
logger.error(message, {
|
|
534
|
-
id: session
|
|
534
|
+
id: this.getSipSessionId(session),
|
|
535
535
|
});
|
|
536
536
|
this.onCallEnded(session);
|
|
537
537
|
throw new CanceledCallError(message);
|
|
@@ -539,7 +539,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
539
539
|
logger.info('sdk webrtc answer, accepted.');
|
|
540
540
|
this._onAccepted(session);
|
|
541
541
|
}).catch(e => {
|
|
542
|
-
logger.error(`answer call error for ${session
|
|
542
|
+
logger.error(`answer call error for ${this.getSipSessionId(session)}`, e);
|
|
543
543
|
throw e;
|
|
544
544
|
});
|
|
545
545
|
}
|
|
@@ -609,7 +609,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
609
609
|
return __awaiter(this, void 0, void 0, function* () {
|
|
610
610
|
logger.info('sdk webrtc reject call', {
|
|
611
611
|
clientId: this.clientId,
|
|
612
|
-
id: session
|
|
612
|
+
id: this.getSipSessionId(session),
|
|
613
613
|
});
|
|
614
614
|
try {
|
|
615
615
|
if (session instanceof api_1.Invitation) {
|
|
@@ -673,13 +673,13 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
673
673
|
}
|
|
674
674
|
mute(session) {
|
|
675
675
|
logger.info('sdk webrtc mute', {
|
|
676
|
-
id: session
|
|
676
|
+
id: this.getSipSessionId(session),
|
|
677
677
|
});
|
|
678
678
|
this._toggleAudio(session, true);
|
|
679
679
|
}
|
|
680
680
|
unmute(session) {
|
|
681
681
|
logger.info('sdk webrtc unmute', {
|
|
682
|
-
id: session
|
|
682
|
+
id: this.getSipSessionId(session),
|
|
683
683
|
});
|
|
684
684
|
this._toggleAudio(session, false);
|
|
685
685
|
}
|
|
@@ -716,13 +716,13 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
716
716
|
}
|
|
717
717
|
toggleCameraOn(session) {
|
|
718
718
|
logger.info('sdk webrtc toggle camera on', {
|
|
719
|
-
id: session
|
|
719
|
+
id: this.getSipSessionId(session),
|
|
720
720
|
});
|
|
721
721
|
this._toggleVideo(session, false);
|
|
722
722
|
}
|
|
723
723
|
toggleCameraOff(session) {
|
|
724
724
|
logger.info('sdk webrtc toggle camera off', {
|
|
725
|
-
id: session
|
|
725
|
+
id: this.getSipSessionId(session),
|
|
726
726
|
});
|
|
727
727
|
this._toggleVideo(session, true);
|
|
728
728
|
}
|
|
@@ -979,7 +979,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
979
979
|
return;
|
|
980
980
|
}
|
|
981
981
|
logger.info('send WebRTC message', {
|
|
982
|
-
sipId: sipSession
|
|
982
|
+
sipId: this.getSipSessionId(sipSession),
|
|
983
983
|
contentType,
|
|
984
984
|
});
|
|
985
985
|
try {
|
|
@@ -1344,18 +1344,25 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
1344
1344
|
if (!sipSession) {
|
|
1345
1345
|
return '';
|
|
1346
1346
|
}
|
|
1347
|
-
const message = sipSession.message;
|
|
1348
|
-
if (message === null || message === void 0 ? void 0 : message.callId) {
|
|
1349
|
-
return message.callId.substring(0, SIP_ID_LENGTH);
|
|
1350
|
-
}
|
|
1351
1347
|
// For Inviter
|
|
1352
1348
|
// @ts-ignore: private
|
|
1353
1349
|
if (sipSession instanceof api_1.Inviter && sipSession.outgoingRequestMessage) {
|
|
1354
1350
|
// @ts-ignore: private
|
|
1355
|
-
return sipSession.outgoingRequestMessage.callId
|
|
1351
|
+
return sipSession.outgoingRequestMessage.callId;
|
|
1356
1352
|
}
|
|
1357
1353
|
// For Invitation
|
|
1358
|
-
|
|
1354
|
+
// @ts-ignore: private
|
|
1355
|
+
if (sipSession instanceof api_1.Invitation && sipSession.incomingInviteRequest.message) {
|
|
1356
|
+
// @ts-ignore: private
|
|
1357
|
+
return sipSession.incomingInviteRequest.message.callId;
|
|
1358
|
+
}
|
|
1359
|
+
// For EarlyMedia
|
|
1360
|
+
// @ts-ignore: private
|
|
1361
|
+
if (sipSession instanceof session_dialog_1.SessionDialog) {
|
|
1362
|
+
// @ts-ignore: private
|
|
1363
|
+
return sipSession.dialogState.callId;
|
|
1364
|
+
}
|
|
1365
|
+
return '';
|
|
1359
1366
|
}
|
|
1360
1367
|
waitForRegister() {
|
|
1361
1368
|
return __awaiter(this, void 0, void 0, function* () {
|
|
@@ -1415,7 +1422,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
1415
1422
|
this.sipSessions[id] = session;
|
|
1416
1423
|
}
|
|
1417
1424
|
getSipSession(id) {
|
|
1418
|
-
return id in this.sipSessions ? this.sipSessions[id
|
|
1425
|
+
return id in this.sipSessions ? this.sipSessions[id] : null;
|
|
1419
1426
|
}
|
|
1420
1427
|
getSipSessionIds() {
|
|
1421
1428
|
return Object.keys(this.sipSessions);
|
|
@@ -1791,7 +1798,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
1791
1798
|
_onProgress(session_1) {
|
|
1792
1799
|
return __awaiter(this, arguments, void 0, function* (session, early = false) {
|
|
1793
1800
|
yield this._setupMedias(session);
|
|
1794
|
-
const sessionId = this.getSipSessionId(session)
|
|
1801
|
+
const sessionId = this.getSipSessionId(session);
|
|
1795
1802
|
this.updateRemoteStream(sessionId);
|
|
1796
1803
|
logger.info('progress received', {
|
|
1797
1804
|
sessionId,
|
|
@@ -1804,7 +1811,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
1804
1811
|
return __awaiter(this, arguments, void 0, function* (session, sessionDialog, withEvent = true, initAllTracks = true) {
|
|
1805
1812
|
var _a;
|
|
1806
1813
|
logger.info('on call accepted', {
|
|
1807
|
-
id: session
|
|
1814
|
+
id: this.getSipSessionId(session),
|
|
1808
1815
|
clientId: this.clientId,
|
|
1809
1816
|
remoteTag: session.remoteTag,
|
|
1810
1817
|
});
|
|
@@ -2161,18 +2168,18 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
2161
2168
|
return __awaiter(this, arguments, void 0, function* (session, options = {}) {
|
|
2162
2169
|
if (session.state === session_state_1.SessionState.Terminated || session.state === session_state_1.SessionState.Terminating) {
|
|
2163
2170
|
const error = 'Trying to accept a terminated sipSession.';
|
|
2164
|
-
logger.warn(error, { state: session.state, sessionId: session
|
|
2171
|
+
logger.warn(error, { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2165
2172
|
throw new Error(error);
|
|
2166
2173
|
}
|
|
2167
2174
|
if (session.state !== session_state_1.SessionState.Initial) {
|
|
2168
2175
|
const error = 'Trying to accept a non Initial sipSession.';
|
|
2169
|
-
logger.warn(error, { state: session.state, sessionId: session
|
|
2176
|
+
logger.warn(error, { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2170
2177
|
throw new Error(error);
|
|
2171
2178
|
}
|
|
2172
2179
|
// @ts-ignore: private
|
|
2173
2180
|
if (!session.incomingInviteRequest.acceptable) {
|
|
2174
2181
|
const error = 'Trying to reject a non `acceptable` session';
|
|
2175
|
-
logger.warn(error, { state: session.state, sessionId: session
|
|
2182
|
+
logger.warn(error, { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2176
2183
|
throw new Error(error);
|
|
2177
2184
|
}
|
|
2178
2185
|
try {
|
|
@@ -2186,12 +2193,12 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
2186
2193
|
_reject(session_1) {
|
|
2187
2194
|
return __awaiter(this, arguments, void 0, function* (session, options = {}) {
|
|
2188
2195
|
if (session.state !== session_state_1.SessionState.Initial && session.state !== session_state_1.SessionState.Establishing) {
|
|
2189
|
-
logger.warn('Trying to reject a session in a wrong state', { state: session.state, sessionId: session
|
|
2196
|
+
logger.warn('Trying to reject a session in a wrong state', { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2190
2197
|
return;
|
|
2191
2198
|
}
|
|
2192
2199
|
// @ts-ignore: private
|
|
2193
2200
|
if (!session.incomingInviteRequest.rejectable) {
|
|
2194
|
-
logger.warn('Trying to reject a non `rejectable` session', { state: session.state, sessionId: session
|
|
2201
|
+
logger.warn('Trying to reject a non `rejectable` session', { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2195
2202
|
return;
|
|
2196
2203
|
}
|
|
2197
2204
|
try {
|
|
@@ -2205,7 +2212,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
2205
2212
|
_cancel(session_1) {
|
|
2206
2213
|
return __awaiter(this, arguments, void 0, function* (session, options = {}) {
|
|
2207
2214
|
if (session.state !== session_state_1.SessionState.Initial && session.state !== session_state_1.SessionState.Establishing) {
|
|
2208
|
-
logger.warn('Trying to cancel a session in a wrong state', { state: session.state, sessionId: session
|
|
2215
|
+
logger.warn('Trying to cancel a session in a wrong state', { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2209
2216
|
return;
|
|
2210
2217
|
}
|
|
2211
2218
|
try {
|
|
@@ -2219,7 +2226,7 @@ class WebRTCClient extends Emitter_1.default {
|
|
|
2219
2226
|
_bye(session_1) {
|
|
2220
2227
|
return __awaiter(this, arguments, void 0, function* (session, options = {}) {
|
|
2221
2228
|
if (session.state !== session_state_1.SessionState.Established) {
|
|
2222
|
-
logger.warn('Trying to end a session in a wrong state', { state: session.state, sessionId: session
|
|
2229
|
+
logger.warn('Trying to end a session in a wrong state', { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2223
2230
|
return null;
|
|
2224
2231
|
}
|
|
2225
2232
|
try {
|