@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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@@ -123,7 +123,7 @@ export default class WebRTCPhone extends Emitter {
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async sendReinvite(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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@@ -298,7 +298,9 @@ export default class WebRTCPhone extends Emitter {
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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(ON_CALL_CANCELED, callSession, elsewhere);
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};
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}
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@@ -306,14 +308,14 @@ export default class WebRTCPhone extends Emitter {
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switch (newState) {
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case 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(ON_CALL_ENDING, this._createCallSession(sipSession));
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break;
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case 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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@@ -465,13 +467,13 @@ export default class WebRTCPhone extends Emitter {
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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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@@ -540,7 +542,7 @@ export default class WebRTCPhone extends Emitter {
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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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@@ -560,8 +562,8 @@ export default class WebRTCPhone extends Emitter {
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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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@@ -569,7 +571,7 @@ export default class WebRTCPhone extends Emitter {
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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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@@ -684,7 +686,7 @@ export default class WebRTCPhone extends Emitter {
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this.eventEmitter.emit(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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@@ -719,7 +721,7 @@ export default class WebRTCPhone extends Emitter {
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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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@@ -737,7 +739,7 @@ export default class WebRTCPhone extends Emitter {
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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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@@ -763,12 +765,12 @@ export default class WebRTCPhone extends Emitter {
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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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@@ -786,7 +788,7 @@ export default class WebRTCPhone extends Emitter {
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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(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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this.client.toggleCameraOff(sipSession);
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this.eventEmitter.emit(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: sipSession
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sipId: this.getSipSessionId(sipSession),
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});
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await this.client.hangup(sipSession);
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@@ -1188,7 +1190,7 @@ export default class WebRTCPhone extends Emitter {
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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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@@ -1248,7 +1250,7 @@ export default class WebRTCPhone extends Emitter {
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this.eventEmitter.emit(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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@@ -1363,6 +1365,9 @@ export default class WebRTCPhone extends Emitter {
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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/esm/web-rtc-client.js
CHANGED
|
@@ -7,6 +7,7 @@ import { C } from 'sip.js/lib/core/messages/methods/constants';
|
|
|
7
7
|
import { URI } from 'sip.js/lib/grammar/uri';
|
|
8
8
|
import { UserAgent } from 'sip.js/lib/api/user-agent';
|
|
9
9
|
import { holdModifier, stripVideo } from 'sip.js/lib/platform/web/modifiers/modifiers';
|
|
10
|
+
import { SessionDialog } from 'sip.js/lib/core/dialogs/session-dialog';
|
|
10
11
|
import { Messager } from 'sip.js/lib/api/messager';
|
|
11
12
|
import { RegistererState } from 'sip.js/lib/api/registerer-state';
|
|
12
13
|
import { SessionState } from 'sip.js/lib/api/session-state';
|
|
@@ -26,7 +27,6 @@ export const replaceLocalIpModifier = (description) => Promise.resolve({
|
|
|
26
27
|
...JSON.parse(JSON.stringify(description)),
|
|
27
28
|
sdp: description.sdp.replace('c=IN IP4 0.0.0.0', 'c=IN IP4 127.0.0.1'),
|
|
28
29
|
});
|
|
29
|
-
const SIP_ID_LENGTH = 36;
|
|
30
30
|
const DEFAULT_ICE_TIMEOUT = 3000;
|
|
31
31
|
const SEND_STATS_DELAY = 5000;
|
|
32
32
|
const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
|
|
@@ -191,7 +191,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
191
191
|
logger.info('sdk webrtc on invite', {
|
|
192
192
|
method: 'delegate.onInvite',
|
|
193
193
|
clientId: this.clientId,
|
|
194
|
-
id: invitation
|
|
194
|
+
id: this.getSipSessionId(invitation),
|
|
195
195
|
remoteURI: invitation.remoteURI,
|
|
196
196
|
});
|
|
197
197
|
this._setupSession(invitation);
|
|
@@ -473,7 +473,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
473
473
|
},
|
|
474
474
|
onReject: (response) => {
|
|
475
475
|
logger.info('on call rejected', {
|
|
476
|
-
id: session
|
|
476
|
+
id: this.getSipSessionId(session),
|
|
477
477
|
// @ts-ignore: fromTag does not exist
|
|
478
478
|
fromTag: session?.fromTag,
|
|
479
479
|
});
|
|
@@ -503,7 +503,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
503
503
|
answer(session, enableVideo) {
|
|
504
504
|
logger.info('sdk webrtc answer call', {
|
|
505
505
|
clientId: this.clientId,
|
|
506
|
-
id: session
|
|
506
|
+
id: this.getSipSessionId(session),
|
|
507
507
|
enableVideo,
|
|
508
508
|
});
|
|
509
509
|
if (!session || !session.accept) {
|
|
@@ -519,7 +519,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
519
519
|
if (session.isCanceled) {
|
|
520
520
|
const message = 'accepted a canceled session (or was canceled during the accept phase).';
|
|
521
521
|
logger.error(message, {
|
|
522
|
-
id: session
|
|
522
|
+
id: this.getSipSessionId(session),
|
|
523
523
|
});
|
|
524
524
|
this.onCallEnded(session);
|
|
525
525
|
throw new CanceledCallError(message);
|
|
@@ -527,7 +527,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
527
527
|
logger.info('sdk webrtc answer, accepted.');
|
|
528
528
|
this._onAccepted(session);
|
|
529
529
|
}).catch(e => {
|
|
530
|
-
logger.error(`answer call error for ${session
|
|
530
|
+
logger.error(`answer call error for ${this.getSipSessionId(session)}`, e);
|
|
531
531
|
throw e;
|
|
532
532
|
});
|
|
533
533
|
}
|
|
@@ -591,7 +591,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
591
591
|
async reject(session) {
|
|
592
592
|
logger.info('sdk webrtc reject call', {
|
|
593
593
|
clientId: this.clientId,
|
|
594
|
-
id: session
|
|
594
|
+
id: this.getSipSessionId(session),
|
|
595
595
|
});
|
|
596
596
|
try {
|
|
597
597
|
if (session instanceof Invitation) {
|
|
@@ -652,13 +652,13 @@ export default class WebRTCClient extends Emitter {
|
|
|
652
652
|
}
|
|
653
653
|
mute(session) {
|
|
654
654
|
logger.info('sdk webrtc mute', {
|
|
655
|
-
id: session
|
|
655
|
+
id: this.getSipSessionId(session),
|
|
656
656
|
});
|
|
657
657
|
this._toggleAudio(session, true);
|
|
658
658
|
}
|
|
659
659
|
unmute(session) {
|
|
660
660
|
logger.info('sdk webrtc unmute', {
|
|
661
|
-
id: session
|
|
661
|
+
id: this.getSipSessionId(session),
|
|
662
662
|
});
|
|
663
663
|
this._toggleAudio(session, false);
|
|
664
664
|
}
|
|
@@ -695,13 +695,13 @@ export default class WebRTCClient extends Emitter {
|
|
|
695
695
|
}
|
|
696
696
|
toggleCameraOn(session) {
|
|
697
697
|
logger.info('sdk webrtc toggle camera on', {
|
|
698
|
-
id: session
|
|
698
|
+
id: this.getSipSessionId(session),
|
|
699
699
|
});
|
|
700
700
|
this._toggleVideo(session, false);
|
|
701
701
|
}
|
|
702
702
|
toggleCameraOff(session) {
|
|
703
703
|
logger.info('sdk webrtc toggle camera off', {
|
|
704
|
-
id: session
|
|
704
|
+
id: this.getSipSessionId(session),
|
|
705
705
|
});
|
|
706
706
|
this._toggleVideo(session, true);
|
|
707
707
|
}
|
|
@@ -953,7 +953,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
953
953
|
return;
|
|
954
954
|
}
|
|
955
955
|
logger.info('send WebRTC message', {
|
|
956
|
-
sipId: sipSession
|
|
956
|
+
sipId: this.getSipSessionId(sipSession),
|
|
957
957
|
contentType,
|
|
958
958
|
});
|
|
959
959
|
try {
|
|
@@ -1309,18 +1309,25 @@ export default class WebRTCClient extends Emitter {
|
|
|
1309
1309
|
if (!sipSession) {
|
|
1310
1310
|
return '';
|
|
1311
1311
|
}
|
|
1312
|
-
const message = sipSession.message;
|
|
1313
|
-
if (message?.callId) {
|
|
1314
|
-
return message.callId.substring(0, SIP_ID_LENGTH);
|
|
1315
|
-
}
|
|
1316
1312
|
// For Inviter
|
|
1317
1313
|
// @ts-ignore: private
|
|
1318
1314
|
if (sipSession instanceof Inviter && sipSession.outgoingRequestMessage) {
|
|
1319
1315
|
// @ts-ignore: private
|
|
1320
|
-
return sipSession.outgoingRequestMessage.callId
|
|
1316
|
+
return sipSession.outgoingRequestMessage.callId;
|
|
1321
1317
|
}
|
|
1322
1318
|
// For Invitation
|
|
1323
|
-
|
|
1319
|
+
// @ts-ignore: private
|
|
1320
|
+
if (sipSession instanceof Invitation && sipSession.incomingInviteRequest.message) {
|
|
1321
|
+
// @ts-ignore: private
|
|
1322
|
+
return sipSession.incomingInviteRequest.message.callId;
|
|
1323
|
+
}
|
|
1324
|
+
// For EarlyMedia
|
|
1325
|
+
// @ts-ignore: private
|
|
1326
|
+
if (sipSession instanceof SessionDialog) {
|
|
1327
|
+
// @ts-ignore: private
|
|
1328
|
+
return sipSession.dialogState.callId;
|
|
1329
|
+
}
|
|
1330
|
+
return '';
|
|
1324
1331
|
}
|
|
1325
1332
|
async waitForRegister() {
|
|
1326
1333
|
return new Promise(resolve => this.on(REGISTERED, resolve));
|
|
@@ -1378,7 +1385,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
1378
1385
|
this.sipSessions[id] = session;
|
|
1379
1386
|
}
|
|
1380
1387
|
getSipSession(id) {
|
|
1381
|
-
return id in this.sipSessions ? this.sipSessions[id
|
|
1388
|
+
return id in this.sipSessions ? this.sipSessions[id] : null;
|
|
1382
1389
|
}
|
|
1383
1390
|
getSipSessionIds() {
|
|
1384
1391
|
return Object.keys(this.sipSessions);
|
|
@@ -1769,7 +1776,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
1769
1776
|
}
|
|
1770
1777
|
async _onProgress(session, early = false) {
|
|
1771
1778
|
await this._setupMedias(session);
|
|
1772
|
-
const sessionId = this.getSipSessionId(session)
|
|
1779
|
+
const sessionId = this.getSipSessionId(session);
|
|
1773
1780
|
this.updateRemoteStream(sessionId);
|
|
1774
1781
|
logger.info('progress received', {
|
|
1775
1782
|
sessionId,
|
|
@@ -1779,7 +1786,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
1779
1786
|
}
|
|
1780
1787
|
async _onAccepted(session, sessionDialog, withEvent = true, initAllTracks = true) {
|
|
1781
1788
|
logger.info('on call accepted', {
|
|
1782
|
-
id: session
|
|
1789
|
+
id: this.getSipSessionId(session),
|
|
1783
1790
|
clientId: this.clientId,
|
|
1784
1791
|
remoteTag: session.remoteTag,
|
|
1785
1792
|
});
|
|
@@ -2130,18 +2137,18 @@ export default class WebRTCClient extends Emitter {
|
|
|
2130
2137
|
async _accept(session, options = {}) {
|
|
2131
2138
|
if (session.state === SessionState.Terminated || session.state === SessionState.Terminating) {
|
|
2132
2139
|
const error = 'Trying to accept a terminated sipSession.';
|
|
2133
|
-
logger.warn(error, { state: session.state, sessionId: session
|
|
2140
|
+
logger.warn(error, { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2134
2141
|
throw new Error(error);
|
|
2135
2142
|
}
|
|
2136
2143
|
if (session.state !== SessionState.Initial) {
|
|
2137
2144
|
const error = 'Trying to accept a non Initial sipSession.';
|
|
2138
|
-
logger.warn(error, { state: session.state, sessionId: session
|
|
2145
|
+
logger.warn(error, { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2139
2146
|
throw new Error(error);
|
|
2140
2147
|
}
|
|
2141
2148
|
// @ts-ignore: private
|
|
2142
2149
|
if (!session.incomingInviteRequest.acceptable) {
|
|
2143
2150
|
const error = 'Trying to reject a non `acceptable` session';
|
|
2144
|
-
logger.warn(error, { state: session.state, sessionId: session
|
|
2151
|
+
logger.warn(error, { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2145
2152
|
throw new Error(error);
|
|
2146
2153
|
}
|
|
2147
2154
|
try {
|
|
@@ -2153,12 +2160,12 @@ export default class WebRTCClient extends Emitter {
|
|
|
2153
2160
|
}
|
|
2154
2161
|
async _reject(session, options = {}) {
|
|
2155
2162
|
if (session.state !== SessionState.Initial && session.state !== SessionState.Establishing) {
|
|
2156
|
-
logger.warn('Trying to reject a session in a wrong state', { state: session.state, sessionId: session
|
|
2163
|
+
logger.warn('Trying to reject a session in a wrong state', { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2157
2164
|
return;
|
|
2158
2165
|
}
|
|
2159
2166
|
// @ts-ignore: private
|
|
2160
2167
|
if (!session.incomingInviteRequest.rejectable) {
|
|
2161
|
-
logger.warn('Trying to reject a non `rejectable` session', { state: session.state, sessionId: session
|
|
2168
|
+
logger.warn('Trying to reject a non `rejectable` session', { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2162
2169
|
return;
|
|
2163
2170
|
}
|
|
2164
2171
|
try {
|
|
@@ -2170,7 +2177,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
2170
2177
|
}
|
|
2171
2178
|
async _cancel(session, options = {}) {
|
|
2172
2179
|
if (session.state !== SessionState.Initial && session.state !== SessionState.Establishing) {
|
|
2173
|
-
logger.warn('Trying to cancel a session in a wrong state', { state: session.state, sessionId: session
|
|
2180
|
+
logger.warn('Trying to cancel a session in a wrong state', { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2174
2181
|
return;
|
|
2175
2182
|
}
|
|
2176
2183
|
try {
|
|
@@ -2182,7 +2189,7 @@ export default class WebRTCClient extends Emitter {
|
|
|
2182
2189
|
}
|
|
2183
2190
|
async _bye(session, options = {}) {
|
|
2184
2191
|
if (session.state !== SessionState.Established) {
|
|
2185
|
-
logger.warn('Trying to end a session in a wrong state', { state: session.state, sessionId: session
|
|
2192
|
+
logger.warn('Trying to end a session in a wrong state', { state: session.state, sessionId: this.getSipSessionId(session) });
|
|
2186
2193
|
return null;
|
|
2187
2194
|
}
|
|
2188
2195
|
try {
|