@exotel-npm-dev/webrtc-client-sdk 3.0.2 → 3.0.4

package/Changelog

@@ -1,5 +1,11 @@
 Change Log
 
+## v3.0.4 16 September, 2025
+-[VST-1093] Enabled support for noise suppression
+
+## v3.0.3 15 September, 2025
+-[VST-1063] granular control over different audio streams, added static registerLogger method for logger callback registration
+
 ## v3.0.2 27 August, 2025
 -[VST-1063] prevent exotel / sipjs from throwing logs in the console based on a flag, added missing iceconnectionstate in session event callback, make auto audio device change handling configurable and fixing Missing sent_request event in registerCallback
 

package/dist/exotelsdk.js

@@ -1,6 +1,6 @@
 /*!
  * 
- * WebRTC CLient SIP version 3.0.2
+ * WebRTC CLient SIP version 3.0.4
  *
  */
 (function webpackUniversalModuleDefinition(root, factory) {
@@ -50,6 +50,7 @@ __webpack_require__.r(__webpack_exports__);
 
 
 const logger = _coreSDKLogger__WEBPACK_IMPORTED_MODULE_0__["default"];
+const AudioManagerCtx = window.AudioContext || window.webkitAudioContext;
 const audioDeviceManager = {
     resetInputDevice: false,
     resetOutputDevice: false,
@@ -57,6 +58,7 @@ const audioDeviceManager = {
     currentAudioOutputDeviceId: "default",
     mediaDevices: [],
     enableAutoAudioDeviceChangeHandling: false,
+    webAudioCtx : new AudioManagerCtx(),
     // Method to set the resetInputDevice flag
     setResetInputDeviceFlag(value) {
         this.resetInputDevice = value;
@@ -186,6 +188,33 @@ const audioDeviceManager = {
         if (callback) callback();
     },
 
+    configureAudioGainNode(sourceNode) {
+        logger.log("audioDeviceManager:configureAudioGainNode entry");
+        let gainNode = this.webAudioCtx.createGain();
+        
+        sourceNode.connect(gainNode).connect(this.webAudioCtx.destination);
+        return gainNode;
+    },
+    
+    createAndConfigureAudioGainNode(audioElement) {
+    
+        logger.log("audioDeviceManager:createAndConfigureAudioGainNode entry for audioElement", audioElement);
+        // get audio track from audio element
+        let sourceNode = this.webAudioCtx.createMediaElementSource(audioElement);
+        // Create a GainNode
+        let gainNode = this.configureAudioGainNode(sourceNode);
+
+         // resume audio context when audio element is played
+         audioElement.addEventListener("play", () => {
+            if (this.webAudioCtx.state === "suspended") {
+                this.webAudioCtx.resume();
+            }
+        });
+        return gainNode;
+        
+        
+    }
+
 };
 
 /* harmony default export */ const __WEBPACK_DEFAULT_EXPORT__ = (audioDeviceManager);
@@ -20850,8 +20879,7 @@ const name = "sip.js";
 "use strict";
 __webpack_require__.r(__webpack_exports__);
 /* harmony export */ __webpack_require__.d(__webpack_exports__, {
-/* harmony export */   "default": () => (__WEBPACK_DEFAULT_EXPORT__),
-/* harmony export */   getLogger: () => (/* binding */ getLogger)
+/* harmony export */   "default": () => (__WEBPACK_DEFAULT_EXPORT__)
 /* harmony export */ });
 /* harmony import */ var _audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__ = __webpack_require__(/*! ./audioDeviceManager.js */ "./node_modules/@exotel-npm-dev/webrtc-core-sdk/src/audioDeviceManager.js");
 /* harmony import */ var _coreSDKLogger_js__WEBPACK_IMPORTED_MODULE_1__ = __webpack_require__(/*! ./coreSDKLogger.js */ "./node_modules/@exotel-npm-dev/webrtc-core-sdk/src/coreSDKLogger.js");
@@ -20860,7 +20888,6 @@ var SIP = __webpack_require__(/*! ./sip-0.20.0.js */ "./node_modules/@exotel-npm
 ;
 
 
-
 let logger = _coreSDKLogger_js__WEBPACK_IMPORTED_MODULE_1__["default"];
 
 var beeptone = document.createElement("audio");
@@ -20872,31 +20899,25 @@ ringbacktone.src = __webpack_require__(/*! ./static/ringbacktone.wav */ "./node_
 var dtmftone = document.createElement("audio");
 dtmftone.src = __webpack_require__(/*! ./static/dtmf.wav */ "./node_modules/@exotel-npm-dev/webrtc-core-sdk/src/static/dtmf.wav");
 
-var audioRemote = document.createElement("audio");
+class SIPJSPhone {
 
-function getLogger() {
-	let uaLogger;
+	static toBeConfigure = true;
+	static audioElementNameVsAudioGainNodeMap = {};
 
-	try {
-		let userAgent = SIP.UserAgent
-		uaLogger = userAgent.getLogger("sip.WebrtcLib")
-	} catch (e) {
-		logger.log("sipjsphone: getLogger: No userAgent.getLogger, Using console log")
-		return console;
-	}
-
-	if (uaLogger) {
-		return uaLogger;
+	static configure() {
+		logger.log("SIPJSPhone: configure: entry");
+		SIPJSPhone.audioElementNameVsAudioGainNodeMap["ringtone"] = _audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.createAndConfigureAudioGainNode(ringtone);
+		SIPJSPhone.audioElementNameVsAudioGainNodeMap["ringbacktone"] = _audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.createAndConfigureAudioGainNode(ringbacktone);
+		SIPJSPhone.audioElementNameVsAudioGainNodeMap["dtmftone"] = _audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.createAndConfigureAudioGainNode(dtmftone);
+		SIPJSPhone.audioElementNameVsAudioGainNodeMap["beeptone"] = _audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.createAndConfigureAudioGainNode(beeptone);
 	}
-	else {
-		logger.log("sipjsphone: getLogger: No Logger, Using console log")
-		return logger;
-	}
-}
-
-class SIPJSPhone {
+	
 
 	constructor(delegate, username) {
+		if(SIPJSPhone.toBeConfigure) {
+			SIPJSPhone.toBeConfigure = false;
+			SIPJSPhone.configure();
+		}
 		this.webrtcSIPPhoneEventDelegate = delegate;
 		this.username = username;
 		this.ctxSip = {};
@@ -20978,6 +20999,7 @@ class SIPJSPhone {
 		this.bHoldEnable = false;
 		this.register_flag = false;
 		this.enableAutoAudioDeviceChangeHandling = false;
+		this.addPreferredCodec = this.addPreferredCodec.bind(this);
 
 		this.ringtone = ringtone;
 		this.beeptone = beeptone;
@@ -20985,15 +21007,48 @@ class SIPJSPhone {
 		this.dtmftone = dtmftone;
 		this.audioRemote = document.createElement("audio");
 		this.audioRemote.style.display = 'none';
-		document.body.appendChild(this.audioRemote);
+		document.body.appendChild(this.audioRemote);		
+		this.callAudioOutputVolume = 1;
+		
+	}
 
-		this.addPreferredCodec = this.addPreferredCodec.bind(this);
 
-		// In the constructor, after initializing audio elements:
-		[this.ringtone, this.beeptone, this.ringbacktone, this.dtmftone].forEach(audio => {
-			audio.muted = false;
-			audio.volume = 1.0;
-		});
+	setCallAudioOutputVolume(value) {
+		logger.log(`sipjsphone: setCallAudioOutputVolume: ${value}`);
+		this.callAudioOutputVolume = Math.max(0, Math.min(1, value));
+		this.audioRemote.volume = this.callAudioOutputVolume;
+		return true;
+	}
+
+	getCallAudioOutputVolume() {
+		logger.log(`sipjsphone: getCallAudioOutputVolume`);
+		return this.callAudioOutputVolume;
+	}
+
+	// Volume control methods
+	static setAudioOutputVolume(audioElementName, value) {
+		
+		logger.log(`SIPJSPhone: setAudioOutputVolume: ${audioElementName} volume set to ${value}`);
+		if(!SIPJSPhone.audioElementNameVsAudioGainNodeMap.hasOwnProperty(audioElementName)) {
+			logger.error(`SIPJSPhone: setAudioOutputVolume: Invalid audio element name: ${audioElementName}`);
+			throw new Error(`Invalid audio element name: ${audioElementName}`);
+		}
+
+		let gainNode = SIPJSPhone.audioElementNameVsAudioGainNodeMap[audioElementName];
+		gainNode.gain.value = Math.max(0, Math.min(1, value));
+		logger.log(`SIPJSPhone: setAudioOutputVolume: ${audioElementName} volume set to ${value}`);
+		return true;
+	
+	}
+
+	static getAudioOutputVolume(audioElementName) {
+		logger.log(`SIPJSPhone: getAudioOutputVolume: ${audioElementName}`);
+		if(!SIPJSPhone.audioElementNameVsAudioGainNodeMap.hasOwnProperty(audioElementName)) {
+			logger.error(`SIPJSPhone: getAudioOutputVolume: Invalid audio element name: ${audioElementName}`);
+			throw new Error(`Invalid audio element name: ${audioElementName}`);
+		}
+		let gainNode = SIPJSPhone.audioElementNameVsAudioGainNodeMap[audioElementName];
+		return gainNode.gain.value;	
 	}
 
 	attachGlobalDeviceChangeListener() {
@@ -21060,11 +21115,11 @@ class SIPJSPhone {
 					if (count > this.ctxSip.ringtoneCount) {
 						clearInterval(this.ctxSip.ringToneIntervalID);
 					}
-				}, 500);
-			} catch (e) {
-				logger.log("DEBUG: startRingTone: Exception:", e);
-			}
-		},
+					}, 500);
+				} catch (e) {
+					logger.log("DEBUG: startRingTone: Exception:", e);
+				}
+			},
 
 			stopRingTone: () => {
 				try {
@@ -21078,11 +21133,12 @@ class SIPJSPhone {
 			} catch (e) { logger.log("sipjsphone: stopRingTone: Exception:", e); }
 		},
 
+			// Update the startRingbackTone method (around line 223) to use Web Audio:
 			startRingbackTone: () => {
 				if (!this.ctxSip.ringbacktone) {
 					this.ctxSip.ringbacktone = this.ringbacktone;
-			}
-			try {
+				}
+				try {
 					this.ctxSip.ringbacktone.play()
 						.then(() => {
 							logger.log("sipjsphone: startRingbackTone: Audio is playing...");
@@ -21091,8 +21147,8 @@ class SIPJSPhone {
 							logger.log("sipjsphone: startRingbackTone: Exception:", e);
 							// Optionally, prompt user to interact with the page to enable audio
 						});
-			} catch (e) { logger.log("sipjsphone: startRingbackTone: Exception:", e); }
-		},
+				} catch (e) { logger.log("sipjsphone: startRingbackTone: Exception:", e); }
+			},
 
 			stopRingbackTone: () => {
 				if (!this.ctxSip.ringbacktone) {
@@ -21244,35 +21300,32 @@ class SIPJSPhone {
 
 		},
 
+			// Update the sipSendDTMF method (around line 389) to use Web Audio:
 			sipSendDTMF: (digit) => {
-
-				try { this.ctxSip.dtmfTone.play(); } catch (e) { logger.log("sipjsphone: sipSendDTMF: Exception:", e); }
-
 				var a = this.ctxSip.callActiveID;
-			if (a) {
+				if (a) {
 					var s = this.ctxSip.Sessions[a];
 
-				if (!/^[0-9A-D#*,]$/.exec(digit)) {
-					return Promise.reject(new Error("Invalid DTMF tone."));
-				}
-				if (!s) {
-					return Promise.reject(new Error("Session does not exist."));
-				}
-
-				const dtmf = digit;
-				const duration = 240;
-				const body = {
-					contentDisposition: "render",
-					contentType: "application/dtmf-relay",
-					content: "Signal=" + dtmf + "\r\nDuration=" + duration
-				};
-				const requestOptions = { body };
-				return s.info({ requestOptions }).then(() => {
-					return;
-				});
+					if (!/^[0-9A-D#*,]$/.exec(digit)) {
+						return Promise.reject(new Error("Invalid DTMF tone."));
+					}
+					if (!s) {
+						return Promise.reject(new Error("Session does not exist."));
+					}	
+					const dtmf = digit;
+					const duration = 240;
+					const body = {
+						contentDisposition: "render",
+						contentType: "application/dtmf-relay",
+						content: "Signal=" + dtmf + "\r\nDuration=" + duration
+					};
+					const requestOptions = { body };
+					return s.info({ requestOptions }).then(() => {
+						return;
+					});
 
-			}
-		},
+				}
+			},
 
 			setError: (err, title, msg, closable) => { },
 
@@ -21927,37 +21980,41 @@ destroySocketConnection() {
 	});
 }
 
-	 assignStream(stream, element) {
-	if (_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioOutputDeviceId != "default")
-		element.setSinkId(_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioOutputDeviceId);
-	// Set element source.
-	element.autoplay = true; // Safari does not allow calling .play() from a
-	// non user action
-	element.srcObject = stream;
-
-	// Load and start playback of media.
-	element.play().catch((error) => {
-		logger.error("Failed to play media");
-		logger.error(error);
-	});
 
-	// If a track is added, load and restart playback of media.
-	stream.onaddtrack = () => {
-		element.load(); // Safari does not work otheriwse
-		element.play().catch((error) => {
-			logger.error("Failed to play remote media on add track");
-			logger.error(error);
-		});
-	};
+	assignStream(stream, element) {
+	logger.log("sipjsphone: assignStream: entry");
+		
+    if (_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioOutputDeviceId != "default")
+        element.setSinkId(_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioOutputDeviceId);
+        
+    // Set element source.
+    element.autoplay = true;
+    element.srcObject = stream;
+    
+
+    // Set HTML audio element volume to 0 to prevent direct audio output
+    element.volume = this.callAudioOutputVolume;
+    
+    // Load and start playback of media.
+    element.play().catch((error) => {
+        logger.error("sipjsphone: assignStream: Failed to play media", error);
+    });
 
-	// If a track is removed, load and restart playback of media.
-	stream.onremovetrack = () => {
-		element.load(); // Safari does not work otheriwse
-		element.play().catch((error) => {
-			logger.error("Failed to play remote media on remove track");
-			logger.error(error);
-		});
-	};
+    // If a track is added, load and restart playback of media.
+    stream.onaddtrack = () => {
+        element.load();
+        element.play().catch((error) => {
+            logger.error("sipjsphone: assignStream: Failed to play remote media on add track", error);
+        });
+    };
+    
+    // If a track is removed, load and restart playback of media.
+    stream.onremovetrack = () => {
+        element.load();
+        element.play().catch((error) => {
+            logger.error("sipjsphone: assignStream: Failed to play remote media on remove track", error);
+        });
+    };
 }
 
 	 onUserSessionAcceptFailed(e) {
@@ -22242,7 +22299,7 @@ destroySocketConnection() {
 		logger.log("sipjsphone: changeAudioOutputDevice : ", deviceId, onSuccess, onError, "forceDeviceChange = ", forceDeviceChange, "enableAutoAudioDeviceChangeHandling = ", this.enableAutoAudioDeviceChangeHandling);
 		try {
 			// Ensure device list is up-to-date
-			await _audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.enumerateDevices();
+			await _audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.enumerateDevices(() => {});
 			if (!this.audioRemote) {
 				const errorMsg = 'SIPJSPhone:changeAudioOutputDevice audioRemote element is not set.';
 				logger.error(errorMsg);
@@ -22841,6 +22898,26 @@ class WebrtcSIPPhone {
 		logger.log("webrtcSIPPhone: registerAudioDeviceChangeCallback entry");
 		this.phone.registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback);
 	}
+
+	static setAudioOutputVolume(audioElementName, value){
+		logger.log("WebrtcSIPPhone: setAudioOutputVolume: ", audioElementName, value);
+		return _sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].setAudioOutputVolume(audioElementName, value);
+	}
+
+	static getAudioOutputVolume(audioElementName) {
+		logger.log("webrtcSIPPhone: getAudioOutputVolume: ", audioElementName);
+		return _sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].getAudioOutputVolume(audioElementName);
+	}
+
+	setCallAudioOutputVolume(value) {
+		logger.log("webrtcSIPPhone: setCallAudioOutputVolume: ", value);
+		return this.phone.setCallAudioOutputVolume(value);
+	}
+
+	getCallAudioOutputVolume() {
+		logger.log("webrtcSIPPhone: getCallAudioOutputVolume");
+		return this.phone.getCallAudioOutputVolume();
+	}
 }
 
 /* harmony default export */ const __WEBPACK_DEFAULT_EXPORT__ = (WebrtcSIPPhone);
@@ -24782,7 +24859,6 @@ class ExSynchronousHandler {
     logger.log("synchronousHandler: onResponse, phone is connected.\n");
   }
 }
-
 class ExotelWebClient {
   /**
    * @param {Object} sipAccntInfo 
@@ -24803,6 +24879,7 @@ class ExotelWebClient {
   registerCallback = null;
   sessionCallback = null;
   logger = (0,_exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.getLogger)();
+  static clientSDKLoggerCallback = null;
   constructor() {
     // Initialize properties
     this.ctrlr = null;
@@ -24816,20 +24893,9 @@ class ExotelWebClient {
     this.currentSIPUserName = "";
     this.isReadyToRegister = true;
     this.sipAccountInfo = null;
-    this.clientSDKLoggerCallback = null;
     this.callbacks = new _listeners_Callback__WEBPACK_IMPORTED_MODULE_7__.Callback();
     this.registerCallback = new _listeners_Callback__WEBPACK_IMPORTED_MODULE_7__.RegisterCallback();
     this.sessionCallback = new _listeners_Callback__WEBPACK_IMPORTED_MODULE_7__.SessionCallback();
-    this.logger = (0,_exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.getLogger)();
-
-    // Register logger callback
-    let exwebClientOb = this;
-    this.logger.registerLoggerCallback((type, message, args) => {
-      _api_LogManager_js__WEBPACK_IMPORTED_MODULE_10__["default"].onLog(type, message, args);
-      if (exwebClientOb.clientSDKLoggerCallback) {
-        exwebClientOb.clientSDKLoggerCallback("log", message, args);
-      }
-    });
   }
   initWebrtc = async (sipAccountInfo_, RegisterEventCallBack, CallListenerCallback, SessionCallback, enableAutoAudioDeviceChangeHandling = false) => {
     const userName = sipAccountInfo_?.userName;
@@ -25142,8 +25208,9 @@ class ExotelWebClient {
     }
     this.webrtcSIPPhone.setPreferredCodec(codecName);
   }
-  registerLoggerCallback(callback) {
-    this.clientSDKLoggerCallback = callback;
+  static registerLoggerCallback(callback) {
+    logger.log("ExWebClient: registerLoggerCallback: Entry");
+    ExotelWebClient.clientSDKLoggerCallback = callback;
   }
   registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback) {
     logger.log("ExWebClient: registerAudioDeviceChangeCallback: Entry");
@@ -25153,13 +25220,40 @@ class ExotelWebClient {
     }
     this.webrtcSIPPhone.registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback);
   }
-  setEnableConsoleLogging(enable) {
+  static setEnableConsoleLogging(enable) {
     if (enable) {
-      logger.log(`ExWebClient: setEnableConsoleLogging: ${enable}`);
+      logger.log("ExWebClient: setEnableConsoleLogging: Entry, enable: " + enable);
     }
     logger.setEnableConsoleLogging(enable);
   }
+  static setAudioOutputVolume(audioElementName, value) {
+    logger.log(`ExWebClient: setAudioOutputVolume: Entry, audioElementName: ${audioElementName}, value: ${value}`);
+    _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.WebrtcSIPPhone.setAudioOutputVolume(audioElementName, value);
+  }
+  static getAudioOutputVolume(audioElementName) {
+    logger.log(`ExWebClient: getAudioOutputVolume: Entry, audioElementName: ${audioElementName}`);
+    return _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.WebrtcSIPPhone.getAudioOutputVolume(audioElementName);
+  }
+  setCallAudioOutputVolume(value) {
+    logger.log(`ExWebClient: setCallAudioOutputVolume: Entry, value: ${value}`);
+    this.webrtcSIPPhone.setCallAudioOutputVolume(value);
+  }
+  getCallAudioOutputVolume() {
+    logger.log(`ExWebClient: getCallAudioOutputVolume: Entry`);
+    return this.webrtcSIPPhone.getCallAudioOutputVolume();
+  }
+  setNoiseSuppression(enabled = false) {
+    logger.log(`ExWebClient: setNoiseSuppression: ${enabled}`);
+    this.webrtcSIPPhone.setNoiseSuppression(enabled);
+  }
 }
+logger.registerLoggerCallback((type, message, args) => {
+  _api_LogManager_js__WEBPACK_IMPORTED_MODULE_10__["default"].onLog(type, message, args);
+  if (ExotelWebClient.clientSDKLoggerCallback) {
+    ExotelWebClient.clientSDKLoggerCallback("log", message, args);
+  }
+});
+
 /* harmony default export */ const __WEBPACK_DEFAULT_EXPORT__ = (ExotelWebClient);
 
 /***/ }),