@exotel-npm-dev/webrtc-client-sdk 1.0.24 → 2.0.1

package/dist/exotelsdk.js

@@ -1,6 +1,6 @@
 /*!
  * 
- * WebRTC CLient SIP version 1.0.24
+ * WebRTC CLient SIP version 2.0.1
  *
  */
 (function webpackUniversalModuleDefinition(root, factory) {
@@ -20885,7 +20885,7 @@ function getLogger() {
 		uaLogger = userAgent.getLogger("sip.WebrtcLib")
 		//let loggerFactory = userAgent.getLoggerFactory()
 	} catch (e) {
-		logger.log("No userAgent.getLogger: Using console log")
+		logger.log("sipjsphone: getLogger: No userAgent.getLogger, Using console log")
 		return console;
 	}
 
@@ -20893,7 +20893,7 @@ function getLogger() {
 		return uaLogger;
 	}
 	else {
-		logger.log("No Logger: Using console log")
+		logger.log("sipjsphone: getLogger: No Logger, Using console log")
 		return logger;
 	}
 }
@@ -20928,10 +20928,10 @@ function postInit(onInitDoneCallback) {
 					ctxSip.ringtone.play()
 						.then(() => {
 							// Audio is playing.
-							logger.log("startRingTone: Audio is playing: count=" + count + " ctxSip.ringToneIntervalID=" + ctxSip.ringToneIntervalID + " ctxSip.ringtoneCount=" + ctxSip.ringtoneCount);
+							logger.log("sipjsphone: startRingTone: Audio is playing: count=" + count + " ctxSip.ringToneIntervalID=" + ctxSip.ringToneIntervalID + " ctxSip.ringtoneCount=" + ctxSip.ringtoneCount);
 						})
 						.catch(e => {
-							logger.log("startRingTone: Exception:", e);
+							logger.log("sipjsphone: startRingTone: Exception:", e);
 						});
 					count++;
 					if (count > ctxSip.ringtoneCount) {
@@ -20941,7 +20941,7 @@ function postInit(onInitDoneCallback) {
 
 
 
-			} catch (e) { logger.log("startRingTone: Exception:", e); }
+			} catch (e) { logger.log("sipjsphone: startRingTone: Exception:", e); }
 		},
 
 		stopRingTone: function () {
@@ -20951,9 +20951,9 @@ function postInit(onInitDoneCallback) {
 					ctxSip.ringtone = ringtone;
 				}
 				ctxSip.ringtone.pause();
-				logger.log("stopRingTone: intervalID:", ctxSip.ringToneIntervalID);
+				logger.log("sipjsphone: stopRingTone: intervalID:", ctxSip.ringToneIntervalID);
 				clearInterval(ctxSip.ringToneIntervalID)
-			} catch (e) { logger.log("stopRingTone: Exception:", e); }
+			} catch (e) { logger.log("sipjsphone: stopRingTone: Exception:", e); }
 		},
 
 		startRingbackTone: function () {
@@ -20963,19 +20963,19 @@ function postInit(onInitDoneCallback) {
 			try {
 				ctxSip.ringbacktone.play().then(() => {
 					// Audio is playing.
-					logger.log("startRingbackTone: Audio is playing:");
+					logger.log("sipjsphone: startRingbackTone: Audio is playing:");
 				})
 					.catch(e => {
-						logger.log("startRingbackTone: Exception:", e);
+						logger.log("sipjsphone: startRingbackTone: Exception:", e);
 					});
-			} catch (e) { logger.log("startRingbackTone: Exception:", e); }
+			} catch (e) { logger.log("sipjsphone: startRingbackTone: Exception:", e); }
 		},
 
 		stopRingbackTone: function () {
 			if (!ctxSip.ringbacktone) {
 				ctxSip.ringbacktone = ringbacktone;
 			}
-			try { ctxSip.ringbacktone.pause(); } catch (e) { logger.log("stopRingbackTone: Exception:", e); }
+			try { ctxSip.ringbacktone.pause(); } catch (e) { logger.log("sipjsphone: stopRingbackTone: Exception:", e); }
 		},
 
 		// Genereates a rendom string to ID a call
@@ -21020,7 +21020,7 @@ function postInit(onInitDoneCallback) {
 					let pc = sdh._peerConnection;
 					_webrtcSIPPhoneEventDelegate__WEBPACK_IMPORTED_MODULE_2__["default"].initGetStats(pc, callId, username);
 				} catch (e) {
-					logger.log("something went wrong while initing getstats");
+					logger.log("sipjsphone: newSession: something went wrong while initing getstats");
 					logger.log(e);
 				}
 
@@ -21125,7 +21125,7 @@ function postInit(onInitDoneCallback) {
 
 		sipSendDTMF: function (digit) {
 
-			try { ctxSip.dtmfTone.play(); } catch (e) { logger.log("sipSendDTMF: Exception:", e); }
+			try { ctxSip.dtmfTone.play(); } catch (e) { logger.log("sipjsphone: sipSendDTMF: Exception:", e); }
 
 			var a = ctxSip.callActiveID;
 			if (a) {
@@ -21159,7 +21159,7 @@ function postInit(onInitDoneCallback) {
 
 
 		phoneMuteButtonPressed: function (sessionid) {
-
+			logger.log(" sipjsphone: phoneMuteButtonPressed: bMicEnable, sessionid", bMicEnable, sessionid);
 			var s = ctxSip.Sessions[sessionid];
 
 			if (bMicEnable) {
@@ -21175,16 +21175,20 @@ function postInit(onInitDoneCallback) {
 		phoneMute: function (sessionid, bMute) {
 			if (sessionid) {
 				var s = ctxSip.Sessions[sessionid];
-				logger.log("phoneMute: bMute", bMute)
+				logger.log(" sipjsphone: phoneMute: bMute", bMute)
 				toggleMute(s, bMute);
 				bMicEnable = !bMute;
 			}
+			else{
+				logger.log(" sipjsphone: phoneMute: doing nothing as sessionid not found")
+
+			}
 		},
 
 		phoneHold: function (sessionid, bHold) {
 			if (sessionid) {
 				var s = ctxSip.Sessions[sessionid];
-				logger.log("phoneHold: bHold", bHold)
+				logger.log("sipjsphone: phoneHold: bHold", bHold)
 				toggleHold(s, bHold);
 				bHoldEnable = bHold;
 			}
@@ -21230,7 +21234,7 @@ function postInit(onInitDoneCallback) {
 		alert('Your browser don\'t support WebRTC.\naudio/video calls will be disabled.');
 	}
 	_webrtcSIPPhoneEventDelegate__WEBPACK_IMPORTED_MODULE_2__["default"].setWebRTCFSMMapper("sipjs");
-	logger.log("init: Initialization complete...")
+	logger.log("sipjsphone: init: Initialization complete...")
 	initializeComplete = true;
 	onInitDoneCallback();
 }
@@ -21353,7 +21357,7 @@ function sipRegister() {
 			},
 			stunServers: ["stun:stun.l.google.com:19302"],
 			registerOptions: {
-				expires: 60
+				expires: 300
 			}
 
 		};
@@ -21444,7 +21448,7 @@ function registerPhoneEventListeners() {
 
 	ctxSip.phone.transport.stateChange.addListener(transportStateChangeListener);
 
-	registerer = new SIP.Registerer(ctxSip.phone, { expires: 60, refreshFrequency: 80 });
+	registerer = new SIP.Registerer(ctxSip.phone, { expires: 300, refreshFrequency: 80 });
 
 
 	ctxSip.phone.delegate.onInvite = (incomingSession) => {
@@ -21495,7 +21499,7 @@ function destroySocketConnection() {
 			ctxSip.phone.transport.disconnect();
 		}
 	} catch (e) {
-		logger.log("ERROR", e);
+		logger.log("sipjsphone: destroySocketConnection: ERROR", e);
 	}
 }
 
@@ -21516,7 +21520,7 @@ function uiCallTerminated(s_description) {
 
 
 function sipCall() {
-	logger.log("testing emit accept_reject");
+	logger.log("sipjsphone: sipCall: testing emit accept_reject");
 	_webrtcSIPPhoneEventDelegate__WEBPACK_IMPORTED_MODULE_2__["default"].sendWebRTCEventsToFSM("accept_reject", "CALL");
 }
 
@@ -21531,7 +21535,7 @@ function sipPhoneLogger(level, category, label, content) {
 			} else if (content.startsWith("Received WebSocket text message")) {
 				handleWebSocketMessageContent(content, "recv");
 			}
-			logger.log(level + " sipjslog: " + category + ": " + content);
+			logger.log("sipjsphone: sipPhoneLogger:" + level + " sipjslog: " + category + ": " + content);
 		}
 	} catch (e) {
 		logger.error("sipjsphone:sipPhoneLogger ERROR", e);
@@ -21653,7 +21657,7 @@ function cleanupRegistererTimer() {
 
 
 		} catch (e) {
-			logger.log("ERROR", e);
+			logger.log("sipjsphone: cleanupRegistererTimer: ERROR", e);
 
 		}
 		registerer = null;
@@ -21809,13 +21813,13 @@ function enableReceiverTracks(s, enable) {
 			throw new Error("Peer connection closed.");
 		}
 		peerConnection.getReceivers().forEach((receiver) => {
-			logger.log("Receiver ", receiver)
+			logger.log("sipjsphone: enableReceiverTracks: Receiver ", receiver)
 			if (receiver.track) {
 				receiver.track.enabled = enable;
 			}
 		});
 	} catch (e) {
-		logger.log("enableReceiverTracks: Error in updating receiver tracks  ", e)
+		logger.log("sipjsphone: enableReceiverTracks: Error in updating receiver tracks  ", e)
 
 	}
 }
@@ -21834,7 +21838,7 @@ function enableSenderTracks(s, enable) {
 			}
 		});
 	} catch (e) {
-		logger.log("enableSenderTracks: Error in updating sender tracks  ", e)
+		logger.log("sipjsphone: enableSenderTracks: Error in updating sender tracks  ", e)
 	}
 }
 
@@ -21900,7 +21904,7 @@ function onUserSessionAcceptFailed(e) {
 		_webrtcSIPPhoneEventDelegate__WEBPACK_IMPORTED_MODULE_2__["default"].onCallStatSipJsSessionEvent('userMediaFailed');
 		_webrtcSIPPhoneEventDelegate__WEBPACK_IMPORTED_MODULE_2__["default"].onGetUserMediaErrorCallstatCallback();
 	} else {
-		logger.log("user media failed due to error ", e);
+		logger.log("sipjsphone: onUserSessionAcceptFailed: user media failed due to error ", e);
 	}
 	uiCallTerminated('Media stream permission denied');
 }
@@ -21910,13 +21914,13 @@ const SIPJSPhone = {
 	init: (onInitDoneCallback) => {
 
 		var preInit = function () {
-			logger.log("init:readyState, calling postInit")
+			logger.log("sipjsphone: init:readyState, calling postInit")
 			postInit(onInitDoneCallback);
 		}
 		var oReadyStateTimer = setInterval(function () {
 			if (document.readyState === "complete") {
 				clearInterval(oReadyStateTimer);
-				logger.log("init:readyState, calling preinit")
+				logger.log("sipjsphone: init:readyState, calling preinit")
 				preInit();
 			}
 		}, 100);
@@ -22007,11 +22011,11 @@ const SIPJSPhone = {
 	},
 
 	reRegister: () => {
-		logger.log("sipjs: registering in case of relogin");
+		logger.log("sipjsphone: reRegister: registering in case of relogin");
 		if (ctxSip.phone && registerer) {
 			registerer.register({});
 		} else {
-			logger.log("sipjs: SIP Session does not exist for re registration");
+			logger.log("sipjsphone: reRegister: SIP Session does not exist for re registration");
 		}
 
 	},
@@ -22059,7 +22063,7 @@ const SIPJSPhone = {
 
 	pickPhoneCall: () => {
 		var newSess = ctxSip.Sessions[ctxSip.callActiveID];
-		logger.log("pickphonecall ", ctxSip.callActiveID);
+		logger.log("sipjsphone: pickphonecall: ", ctxSip.callActiveID);
 		if (newSess) {
 			if (_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioInputDeviceId != "default") {
 				newSess.accept({
@@ -22092,7 +22096,7 @@ const SIPJSPhone = {
 		try {
 			ctxSip.beeptone.play();
 		} catch (e) {
-			logger.log("playBeep: Exception:", e);
+			logger.log("sipjsphone: playBeep: Exception:", e);
 		}
 	},
 
@@ -22128,22 +22132,22 @@ const SIPJSPhone = {
 	},
 	/* NL Additions - Start */
 	getSpeakerTestTone: () => {
-		logger.log("Returning speaker test tone:", ringtone);
+		logger.log("sipjsphone: getSpeakerTestTone: Returning speaker test tone:", ringtone);
 		return ringtone;
 	},
 
 
 	getWSSUrl: () => {
-		logger.log("Returning txtWebsocketURL:", txtWebsocketURL);
+		logger.log("sipjsphone: getWSSUrl: Returning txtWebsocketURL:", txtWebsocketURL);
 		return txtWebsocketURL;
 	},
 	/* NL Additions - End */
 	getTransportState: () => {
-		logger.log("Returning Transport State : ", lastTransportState);
+		logger.log("sipjsphone: getTransportState: Returning Transport State : ", lastTransportState);
 		return lastTransportState;
 	},
 	getRegistrationState: () => {
-		logger.log("Returning Registration State : ", lastRegistererState);
+		logger.log("sipjsphone: getRegistrationState: Returning Registration State : ", lastRegistererState);
 		return lastRegistererState;
 	},
 
@@ -22152,11 +22156,11 @@ const SIPJSPhone = {
 			const trackChanged = SIPJSPhone.replaceSenderTrack(stream, deviceId);
 			if (trackChanged) {
 				_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioInputDeviceId = deviceId;
-				logger.log(`SIPJSPhone:changeAudioInputDevice Input device changed to: ${deviceId}`);
+				logger.log(`sipjsphone: changeAudioInputDevice: Input device changed to: ${deviceId}`);
 
 				onSuccess();
 			} else {
-				logger.error("SIPJSPhone:changeAudioInputDevice failed");
+				logger.error("sipjsphone: changeAudioInputDevice: failed");
 				onError("replaceSenderTrack failed for webrtc");
 			}
 		}, onError);
@@ -22234,7 +22238,7 @@ const SIPJSPhone = {
 	audioOutputDeviceChangeCallback: null,
 	onDeviceChangeCallback: null,
 	registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback) {
-		logger.log(`SIPJSPhone:registerAudioDeviceChangeCallback  entry`);
+		logger.log(`sipjsphone: registerAudioDeviceChangeCallback: entry`);
 		SIPJSPhone.audioInputDeviceChangeCallback = audioInputDeviceChangeCallback;
 		SIPJSPhone.audioOutputDeviceChangeCallback = audioOutputDeviceChangeCallback;
 		SIPJSPhone.onDeviceChangeCallback = onDeviceChangeCallback;
@@ -22312,6 +22316,7 @@ let webrtcSIPEngine = null;
 const logger = _coreSDKLogger__WEBPACK_IMPORTED_MODULE_0__["default"];
 
 function sendWebRTCEventsToFSM(eventType, sipMethod) {
+	logger.log("webrtcSIPPhone: sendWebRTCEventsToFSM : ",eventType,sipMethod);
 	_webrtcSIPPhoneEventDelegate__WEBPACK_IMPORTED_MODULE_2__["default"].sendWebRTCEventsToFSM(eventType, sipMethod);
 }
 
@@ -22321,6 +22326,7 @@ const webrtcSIPPhone = {
 
 
 	isConnected: () => {
+		logger.log("webrtcSIPPhone: isConnected entry");
 		var status = phone.getStatus();
 		if (status != "offline") {
 			return true;
@@ -22330,10 +22336,12 @@ const webrtcSIPPhone = {
 	},
 
 	sendDTMFWebRTC: (dtmfValue) => {
+		logger.log("webrtcSIPPhone: sendDTMFWebRTC : ",dtmfValue);
 		phone.sipSendDTMF(dtmfValue);
 	},
 
 	registerWebRTCClient: (sipAccountInfo, handler) => {
+		logger.log("webrtcSIPPhone: registerWebRTCClient : ",sipAccountInfo,handler);
 		sipAccountInfoData = sipAccountInfo;
 		phone.init(() => {
 			phone.loadCredentials(sipAccountInfo);
@@ -22352,76 +22360,92 @@ const webrtcSIPPhone = {
 
 
 	configureWebRTCClientDevice: (handler) => {
+		logger.log("webrtcSIPPhone: configureWebRTCClientDevice : ",handler);
 		phone.registerCallBacks(handler);
 	},
 
 	setAuthenticatorServerURL(serverURL) {
+		logger.log("webrtcSIPPhone: setAuthenticatorServerURL : ",serverURL);
 		// Nothing to do here
 	},
 
 	toggleSipRegister: () => {
+		logger.log("webrtcSIPPhone: toggleSipRegister entry");
 		phone.resetRegisterAttempts();
 		phone.sipToggleRegister();
 	},
 
-	webRTCMuteUnmute: (isMuted) => {
+	webRTCMuteUnmute: () => {
+		logger.log("webrtcSIPPhone: webRTCMuteUnmute");
 		phone.sipToggleMic();
 	},
 
 	getMuteStatus: () => {
+		logger.log("webrtcSIPPhone: getMuteStatus entry");
 		return phone.getMicMuteStatus();
 	},
 
 	muteAction: (bMute) => {
+		logger.log("webrtcSIPPhone: muteAction: ",bMute);
 		phone.sipMute(bMute);
 	},
 
 	holdAction: (bHold) => {
+		logger.log("webrtcSIPPhone: holdAction: ",bHold);
 		phone.sipHold(bHold);
 	},
 
 	holdCall: () => {
+		logger.log("webrtcSIPPhone: holdCall entry");
 		phone.holdCall();
 	},
 
 	pickCall: () => {
+		logger.log("webrtcSIPPhone: pickCall entry");
 		phone.pickPhoneCall();
 
 		_webrtcSIPPhoneEventDelegate__WEBPACK_IMPORTED_MODULE_2__["default"].onPickCall();
 	},
 
 	rejectCall: () => {
+		logger.log("webrtcSIPPhone: rejectCall entry");
 		phone.sipHangUp();
 
 		_webrtcSIPPhoneEventDelegate__WEBPACK_IMPORTED_MODULE_2__["default"].onRejectCall();
 	},
 
 	reRegisterWebRTCPhone: () => {
+		logger.log("webrtcSIPPhone: reRegisterWebRTCPhone entry");
 		phone.reRegister();
 	},
 
 
 	playBeepTone: () => {
+		logger.log("webrtcSIPPhone: playBeepTone entry");
 		phone.playBeep();
 
 	},
 
 	sipUnRegisterWebRTC: () => {
+		logger.log("webrtcSIPPhone: sipUnRegisterWebRTC entry");
 		phone.sipUnRegister();
 	},
 
 	startWSNetworkTest: () => {
+		logger.log("webrtcSIPPhone: startWSNetworkTest entry");
 		undefined.testingMode = true;
 		phone.sipRegister();
 	},
 
 	stopWSNetworkTest: () => {
+		logger.log("webrtcSIPPhone stopWSNetworkTest entry");
 		phone.sipUnRegister();
 	},
 
 
 
 	registerPhone: (engine, delegate) => {
+		logger.log("webrtcSIPPhone: registerPhone : ",engine);
 		webrtcSIPEngine = engine;
 		switch (engine) {
 			case "sipjs":
@@ -22437,29 +22461,35 @@ const webrtcSIPPhone = {
 	},
 
 	getWebRTCStatus: () => {
+		logger.log("webrtcSIPPhone: getWebRTCStatus entry");
 		var status = phone.getStatus();
 		return status;
 	},
 
 	disconnect: () => {
+		logger.log("webrtcSIPPhone: disconnect entry");
 		if (phone) {
 			phone.disconnect();
 		}
 	},
 
 	connect: () => {
+		logger.log("webrtcSIPPhone: connect entry");
 		phone.connect();
 	},
 
 	getSIPAccountInfo() {
+		logger.log("webrtcSIPPhone: getSIPAccountInfo entry");
 		return sipAccountInfoData;
 	},
 	getWebRTCSIPEngine() {
+		logger.log("webrtcSIPPhone: getWebRTCSIPEngine entry");
 		return webrtcSIPEngine;
 	},
 
 	/* NL Addition - Start */
 	getSpeakerTestTone() {
+		logger.log("webrtcSIPPhone: getSpeakerTestTone entry");
 		try {
 			return _sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].getSpeakerTestTone()
 		} catch (e) {
@@ -22468,6 +22498,7 @@ const webrtcSIPPhone = {
 	},
 
 	getWSSUrl() {
+		logger.log("webrtcSIPPhone: getWSSUrl entry");
 		try {
 			return _sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].getWSSUrl()
 		} catch (e) {
@@ -22477,6 +22508,7 @@ const webrtcSIPPhone = {
 	/* NL Addition - End */
 
 	getTransportState() {
+		logger.log("webrtcSIPPhone: getTransportState entry");
 		try {
 			return _sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].getTransportState();
 		} catch (e) {
@@ -22486,6 +22518,7 @@ const webrtcSIPPhone = {
 	},
 
 	getRegistrationState() {
+		logger.log("webrtcSIPPhone: getRegistrationState entry");
 		try {
 			return _sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].getRegistrationState();
 		} catch (e) {
@@ -22495,20 +22528,20 @@ const webrtcSIPPhone = {
 	},
 
 	changeAudioInputDevice(deviceId, onSuccess, onError) {
-		logger.log(`webrtcSIPPhone:changeAudioInputDevice  entry`);
+		logger.log("webrtcSIPPhone: changeAudioInputDevice : ", deviceId, onSuccess, onError);
 		_sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].changeAudioInputDevice(deviceId, onSuccess, onError);
 	},
 
 	changeAudioOutputDevice(deviceId, onSuccess, onError) {
-		logger.log(`webrtcSIPPhone:changeAudioOutputDevice entry`);
+		logger.log("webrtcSIPPhone: changeAudioOutputDevice : ", deviceId, onSuccess, onError);
 		_sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].changeAudioOutputDevice(deviceId, onSuccess, onError);
 	},
 	setPreferredCodec(codecName) {
-		logger.log("webrtcSIPPhone:setPreferredCodec entry");
+		logger.log("webrtcSIPPhone: setPreferredCodec : ", codecName);
 		_sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].setPreferredCodec(codecName);
 	},
 	registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback) {
-		logger.log(`webrtcSIPPhone:registerAudioDeviceChangeCallback  entry`);
+		logger.log("webrtcSIPPhone: registerAudioDeviceChangeCallback entry");
 		_sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback);
 	},
 	getLogger() {
@@ -22714,6 +22747,55 @@ const webrtcSIPPhoneEventDelegate = {
 
 /***/ }),
 
+/***/ "./src/api/LogManager.js":
+/*!*******************************!*\
+  !*** ./src/api/LogManager.js ***!
+  \*******************************/
+/***/ ((__unused_webpack_module, __webpack_exports__, __webpack_require__) => {
+
+"use strict";
+__webpack_require__.r(__webpack_exports__);
+/* harmony export */ __webpack_require__.d(__webpack_exports__, {
+/* harmony export */   "default": () => (__WEBPACK_DEFAULT_EXPORT__)
+/* harmony export */ });
+const MAX_LOG_LINES = 1000;
+const LOG_STORAGE_KEY = 'webrtc_sdk_logs';
+const LogManager = {
+  onLog(level, msg, args = []) {
+    const timestamp = new Date().toISOString();
+    const line = `[${timestamp}] [${level.toUpperCase()}] ${msg} ${args.map(arg => JSON.stringify(arg)).join(" ")}`.trim();
+    let logs = JSON.parse(localStorage.getItem(LOG_STORAGE_KEY)) || [];
+    logs.push(line);
+    if (logs.length > MAX_LOG_LINES) {
+      logs = logs.slice(-MAX_LOG_LINES); // rotate
+    }
+
+    localStorage.setItem(LOG_STORAGE_KEY, JSON.stringify(logs));
+  },
+  getLogs() {
+    return JSON.parse(localStorage.getItem(LOG_STORAGE_KEY)) || [];
+  },
+  downloadLogs(filename) {
+    if (!filename) {
+      const now = new Date();
+      const formattedDate = now.toISOString().split('T')[0]; // Gets YYYY-MM-DD
+      filename = `webrtc_sdk_logs_${formattedDate}.txt`;
+    }
+    const blob = new Blob([LogManager.getLogs().join('\n')], {
+      type: 'text/plain'
+    });
+    const url = URL.createObjectURL(blob);
+    const a = document.createElement('a');
+    a.href = url;
+    a.download = filename;
+    a.click();
+    URL.revokeObjectURL(url);
+  }
+};
+/* harmony default export */ const __WEBPACK_DEFAULT_EXPORT__ = (LogManager);
+
+/***/ }),
+
 /***/ "./src/api/callAPI/Call.js":
 /*!*********************************!*\
   !*** ./src/api/callAPI/Call.js ***!
@@ -22749,25 +22831,32 @@ function Call() {
     /**
      * When agent clicks on mute
      */
-    logger.log('mute toggle clicked');
-    let dummyFlag = null;
-    _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_1__.webrtcSIPPhone.webRTCMuteUnmute(null);
+    logger.log('Call: MuteToggle');
+    _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_1__.webrtcSIPPhone.webRTCMuteUnmute();
   };
   this.Mute = function () {
     /**
      * When agent clicks on mute
      */
-    logger.log('mute clicked');
-    let dummyFlag = true;
-    _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_1__.webrtcSIPPhone.webRTCMuteUnmute(dummyFlag);
+    var isMicEnabled = _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_1__.webrtcSIPPhone.getMuteStatus();
+    logger.log('Call: Mute: isMicEnabled: ', isMicEnabled);
+    if (isMicEnabled) {
+      _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_1__.webrtcSIPPhone.muteAction(true);
+    } else {
+      logger.log('Call: Mute:  Already muted');
+    }
   };
   this.UnMute = function () {
     /**
      * When agent clicks on mute
      */
-    logger.log('unmute clicked');
-    let dummyFlag = false;
-    _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_1__.webrtcSIPPhone.webRTCMuteUnmute(dummyFlag);
+    var isMicEnabled = _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_1__.webrtcSIPPhone.getMuteStatus();
+    logger.log('Call: UnMute: isMicEnabled: ', isMicEnabled);
+    if (isMicEnabled) {
+      logger.log('Call: Unmute: Already unmuted');
+    } else {
+      _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_1__.webrtcSIPPhone.muteAction(false);
+    }
   };
   this.HoldToggle = function () {
     /**
@@ -23948,6 +24037,8 @@ __webpack_require__.r(__webpack_exports__);
 /* harmony import */ var _listeners_Callback__WEBPACK_IMPORTED_MODULE_7__ = __webpack_require__(/*! ./Callback */ "./src/listeners/Callback.js");
 /* harmony import */ var _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__ = __webpack_require__(/*! @exotel-npm-dev/webrtc-core-sdk */ "./node_modules/@exotel-npm-dev/webrtc-core-sdk/index.js");
 /* harmony import */ var _api_callAPI_CallDetails__WEBPACK_IMPORTED_MODULE_9__ = __webpack_require__(/*! ../api/callAPI/CallDetails */ "./src/api/callAPI/CallDetails.js");
+/* harmony import */ var _api_LogManager_js__WEBPACK_IMPORTED_MODULE_10__ = __webpack_require__(/*! ../api/LogManager.js */ "./src/api/LogManager.js");
+
 
 
 
@@ -24013,7 +24104,7 @@ function ExDelegationHandler(exClient_) {
     logger.log("delegationHandler: setTestingMode\n");
   };
   this.onCallStatSipJsSessionEvent = function (ev) {
-    logger.log("delegationHandler: onCallStatSipJsSessionEvent\n");
+    logger.log("delegationHandler: onCallStatSipJsSessionEvent", ev);
   };
   this.sendWebRTCEventsToFSM = function (eventType, sipMethod) {
     logger.log("delegationHandler: sendWebRTCEventsToFSM\n");
@@ -24139,6 +24230,16 @@ class ExotelWebClient {
   //this.webRTCPhones = {};
 
   sipAccountInfo = null;
+  clientSDKLoggerCallback = null;
+  constructor() {
+    /* 
+    Register the logger callback and emit the onLog event
+    */
+    logger.registerLoggerCallback(function (type, message, args) {
+      _api_LogManager_js__WEBPACK_IMPORTED_MODULE_10__["default"].onLog(type, message, args);
+      if (this.clientSDKLoggerCallback) this.clientSDKLoggerCallback("log", arg1, args);
+    });
+  }
   initWebrtc = (sipAccountInfo_, RegisterEventCallBack, CallListenerCallback, SessionCallback) => {
     if (!this.eventListener) {
       this.eventListener = new _listeners_ExotelVoiceClientListener__WEBPACK_IMPORTED_MODULE_3__.ExotelVoiceClientListener();
@@ -24152,7 +24253,7 @@ class ExotelWebClient {
     if (!this.call) {
       this.call = new _api_callAPI_Call__WEBPACK_IMPORTED_MODULE_0__.Call();
     }
-    logger.log("Exotel Client Initialised with " + JSON.stringify(sipAccountInfo_));
+    logger.log("ExWebClient: initWebrtc: Exotel Client Initialised with " + JSON.stringify(sipAccountInfo_));
     this.sipAccountInfo = sipAccountInfo_;
     if (!this.sipAccountInfo["userName"] || !this.sipAccountInfo["sipdomain"] || !this.sipAccountInfo["port"]) {
       return false;
@@ -24160,22 +24261,22 @@ class ExotelWebClient {
     this.sipAccountInfo["sipUri"] = "wss://" + this.sipAccountInfo["userName"] + "@" + this.sipAccountInfo["sipdomain"] + ":" + this.sipAccountInfo["port"];
     _listeners_Callback__WEBPACK_IMPORTED_MODULE_7__.callbacks.initializeCallback(CallListenerCallback);
     _listeners_Callback__WEBPACK_IMPORTED_MODULE_7__.registerCallback.initializeRegisterCallback(RegisterEventCallBack);
-    logger.log("Initializing session callback");
+    logger.log("ExWebClient: initWebrtc: Initializing session callback");
     _listeners_Callback__WEBPACK_IMPORTED_MODULE_7__.sessionCallback.initializeSessionCallback(SessionCallback);
     this.setEventListener(this.eventListener);
     return true;
   };
   DoRegister = () => {
-    logger.log("ExWebClient:DoRegister Entry");
+    logger.log("ExWebClient: DoRegister: Entry");
     if (!this.isReadyToRegister) {
-      logger.warn("ExWebClient:DoRegister SDK is not ready to register");
+      logger.warn("ExWebClient: DoRegister: SDK is not ready to register");
       return false;
     }
     (0,_api_registerAPI_RegisterListener__WEBPACK_IMPORTED_MODULE_1__.DoRegister)(this.sipAccountInfo, this);
     return true;
   };
   UnRegister = () => {
-    logger.log("ExWebClient:UnRegister Entry");
+    logger.log("ExWebClient: UnRegister: Entry");
     (0,_api_registerAPI_RegisterListener__WEBPACK_IMPORTED_MODULE_1__.UnRegister)(this.sipAccountInfo, this);
   };
   initDiagnostics = (saveDiagnosticsCallback, keyValueSetCallback) => {
@@ -24236,7 +24337,7 @@ class ExotelWebClient {
    */
 
   registerEventCallback = (event, phone, param) => {
-    logger.log("Dialer: registerEventCallback: Received ---> " + event + 'phone....', phone + 'param....', param);
+    logger.log("ExWebClient: registerEventCallback: Received ---> " + event + 'phone....', phone + 'param....', param);
     if (event === "connected") {
       /**
        * When registration is successful then send the phone number of the same to UI
@@ -24244,7 +24345,7 @@ class ExotelWebClient {
       this.eventListener.onInitializationSuccess(phone);
       this.registrationInProgress = false;
       if (this.unregisterInitiated) {
-        logger.log("ExWebClient:registerEventCallback unregistering due to unregisterInitiated");
+        logger.log("ExWebClient: registerEventCallback: unregistering due to unregisterInitiated");
         this.unregisterInitiated = false;
         this.unregister();
       }
@@ -24259,7 +24360,7 @@ class ExotelWebClient {
         this.isReadyToRegister = true;
       }
       if (this.shouldAutoRetry) {
-        logger.log("ExWebClient:registerEventCallback Autoretrying");
+        logger.log("ExWebClient: registerEventCallback: Autoretrying");
         (0,_api_registerAPI_RegisterListener__WEBPACK_IMPORTED_MODULE_1__.DoRegister)(this.sipAccountInfo, this, 5000);
       }
     } else if (event === "sent_request") {
@@ -24276,7 +24377,7 @@ class ExotelWebClient {
    * @param {*} param 
    */
   callEventCallback = (event, phone, param) => {
-    logger.log("Dialer: callEventCallback: Received ---> " + event + 'param sent....' + param + 'for phone....' + phone);
+    logger.log("ExWebClient: callEventCallback: Received ---> " + event + 'param sent....' + param + 'for phone....' + phone);
     if (event === "i_new_call") {
       this.callListener.onIncomingCall(param, phone);
     } else if (event === "connected") {
@@ -24301,7 +24402,7 @@ class ExotelWebClient {
    * @param {*} sipAccountInfo 
    */
   unregister = sipAccountInfo => {
-    logger.log("ExWebClient:unregister Entry");
+    logger.log("ExWebClient: unregister: Entry");
     this.shouldAutoRetry = false;
     this.unregisterInitiated = true;
     if (!this.registrationInProgress) {
@@ -24311,7 +24412,7 @@ class ExotelWebClient {
     }
   };
   webRTCStatusCallbackHandler = (msg1, arg1) => {
-    logger.log("webRTCStatusCallbackHandler: " + msg1 + " " + arg1);
+    logger.log("ExWebClient: webRTCStatusCallbackHandler: " + msg1 + " " + arg1);
   };
 
   /**
@@ -24337,7 +24438,7 @@ class ExotelWebClient {
       'port': '',
       'contactHost': ''
     };
-    logger.log('Sending register for the number..', subscriberName);
+    logger.log('ExWebClient: initialize: Sending register for the number..', subscriberName);
     fetchPublicIP(sipAccountInfo);
 
     /* Temporary till we figure out the arguments - Start */
@@ -24434,24 +24535,28 @@ class ExotelWebClient {
           }
       }
     }).catch(function (error) {
-      logger.log("something went wrong during checkClientStatus ", error);
+      logger.log("ExWebClient: checkClientStatus: something went wrong during checkClientStatus ", error);
       callback("media_permission_denied");
     });
   };
   changeAudioInputDevice(deviceId, onSuccess, onError) {
-    logger.log(`in changeAudioInputDevice() of ExWebClient.js`);
+    logger.log(`ExWebClient: changeAudioInputDevice: Entry`);
     _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.webrtcSIPPhone.changeAudioInputDevice(deviceId, onSuccess, onError);
   }
   changeAudioOutputDevice(deviceId, onSuccess, onError) {
-    logger.log(`in changeAudioOutputDevice() of ExWebClient.js`);
+    logger.log(`ExWebClient: changeAudioOutputDevice: Entry`);
     _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.webrtcSIPPhone.changeAudioOutputDevice(deviceId, onSuccess, onError);
   }
+  downloadLogs() {
+    logger.log(`ExWebClient: downloadLogs: Entry`);
+    _api_LogManager_js__WEBPACK_IMPORTED_MODULE_10__["default"].downloadLogs();
+  }
   setPreferredCodec(codecName) {
-    logger.log("ExWebClient:setPreferredCodec entry");
+    logger.log("ExWebClient: setPreferredCodec: Entry");
     _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.webrtcSIPPhone.setPreferredCodec(codecName);
   }
   registerLoggerCallback(callback) {
-    logger.registerLoggerCallback(callback);
+    this.clientSDKLoggerCallback = callback;
   }
   registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback) {
     _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.webrtcSIPPhone.registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback);