@exotel-npm-dev/webrtc-client-sdk 1.0.17 → 1.0.18

package/Changelog

@@ -1,11 +1,15 @@
 Change Log
 
-## v1.0.17 08 January, 2025
+## v1.0.18 08 January, 2025
 -[VST-865] Added option in websdk to select the codec preference
 
 ## v1.0.16 21 November, 2024
 -[VST-885] Retry Support for SDK Websocket Connection
 
+## v1.0.15 25 October, 2024
+-[VST-863] Added option provide ondevice change callback
+
+
 ## v1.0.14 12 September, 2024
 -[VST-807] Added call details with callsid and sip headers
 

package/dist/exotelsdk.js

@@ -1,6 +1,6 @@
 /*!
  * 
- * WebRTC CLient SIP version 1.0.17
+ * WebRTC CLient SIP version 1.0.18
  *
  */
 (function webpackUniversalModuleDefinition(root, factory) {
@@ -53,7 +53,7 @@ const audioDeviceManager = {
     resetOutputDevice: false,
     currentAudioInputDeviceId: "default",
     currentAudioOutputDeviceId: "default",
-
+    mediaDevices: [],
 
     // Method to set the resetInputDevice flag
     setResetInputDeviceFlag(value) {
@@ -69,14 +69,14 @@ const audioDeviceManager = {
         logger.log(`SIPJSPhone:changeAudioInputDevice entry`);
         try {
             if (deviceId == audioDeviceManager.currentAudioInputDeviceId) {
-                logger.log(`SIPJSPhone:changeAudioInputDevice current input device is same as ${deviceId}`);
+                logger.log(`SIPJSPhone:changeAudioInputDevice current input device is same as ${deviceId} hence not changing`);
+                if (onError) onError("current input device is same as " + deviceId + " hence not changing");
                 return;
             }
-            const devices = await navigator.mediaDevices.enumerateDevices();
-            const inputDevice = devices.find(device => device.deviceId === deviceId && device.kind === 'audioinput');
+            const inputDevice = audioDeviceManager.mediaDevices.find(device => device.deviceId === deviceId && device.kind === 'audioinput');
             if (!inputDevice) {
                 logger.error("input device id " + deviceId + "not found");
-                onError("deviceIdNotFound");
+                if (onError) onError("deviceIdNotFound");
                 return;
             }
             logger.log(`SIPJSPhone:changeAudioInputDevice acquiring input device ${deviceId} : ${inputDevice.label}`);
@@ -86,7 +86,7 @@ const audioDeviceManager = {
             onSuccess(stream);
         } catch (error) {
             logger.error('SIPJSPhone:changeAudioInputDevice Error changing input device:', error);
-            onError(error);
+            if (onError) onError(error);
         }
     },
 
@@ -94,19 +94,26 @@ const audioDeviceManager = {
         logger.log(`audioDeviceManager:changeAudioOutputDevice : entry`);
         if (deviceId == audioDeviceManager.currentAudioOutputDeviceId) {
             logger.log(`SIPJSPhone:changeAudioOutputDevice current output device is same as ${deviceId}`);
+            if (onError) onError("current output device is same as " + deviceId);
             return;
         }
         const audioElement = audioRemote;
         if (typeof audioElement.sinkId !== 'undefined') {
             try {
-                const devices = await navigator.mediaDevices.enumerateDevices();
-                const outputDevice = devices.find(device => device.deviceId === deviceId && device.kind === 'audiooutput');
+
+                if (!audioDeviceManager.mediaDevices || audioDeviceManager.mediaDevices.length == 0) {
+                    logger.error("audioDeviceManager:changeAudioOutputDevice mediaDeviceList is empty ");
+                    if (onError) logger.error(deviceId + "not found in mediaDeviceList in audioManager");
+                    return;
+                }
+                const outputDevice = audioDeviceManager.mediaDevices.find(device => device.deviceId === deviceId && device.kind === 'audiooutput');
                 if (!outputDevice) {
                     logger.error("audioDeviceManager:changeAudioOutputDevice output device id " + deviceId + "not found");
-                    onError("deviceIdNotFound");
+                    if (onError) onError("deviceIdNotFound");
                     return;
                 }
-                logger.log(`SIPJSPhone:changeAudioOutputDevice acquiring output device ${deviceId} : ${outputDevice.label}`);
+                logger.log(`audioDeviceManager:changeAudioOutputDevice acquiring output device ${deviceId} : ${outputDevice.label}`);
+                // audioElement.load();
                 await audioElement.setSinkId(deviceId);
                 audioDeviceManager.currentAudioOutputDeviceId = deviceId;
                 logger.log(`audioDeviceManager:changeAudioOutputDevice Output device changed to: ${deviceId}`);
@@ -136,21 +143,20 @@ const audioDeviceManager = {
     async _resetAudioDevice(audioRemote, onInputDeviceChangeCallback, onOutputDeviceChangecallback, resetOutputDevice, resetInputDevice) {
         logger.log("audioDeviceManager:_resetAudioDevice entry");
         try {
-            const devices = await navigator.mediaDevices.enumerateDevices();
 
             if (resetOutputDevice) {
-                const defaultOutputDevice = devices.find(device => device.deviceId === "default" && device.kind === 'audiooutput');
-                const outputDevice = devices.find(device => device.groupId == defaultOutputDevice.groupId && device.kind === 'audiooutput' && device.deviceId != 'default');
+                const defaultOutputDevice = audioDeviceManager.mediaDevices.find(device => device.deviceId === "default" && device.kind === 'audiooutput');
+                const outputDevice = audioDeviceManager.mediaDevices.find(device => device.groupId == defaultOutputDevice.groupId && device.kind === 'audiooutput' && device.deviceId != 'default');
 
                 audioDeviceManager.changeAudioOutputDevice(audioRemote,
                     outputDevice.deviceId,
                     () => onOutputDeviceChangecallback(outputDevice.deviceId),
-                    (error) => logger.log(`audioDeviceManager:_resetAudioDevice Failed to change output device: ${error}`)
+                    (error) => logger.error(`audioDeviceManager:_resetAudioDevice Failed to change output device: ${error}`)
                 );
             }
             if (resetInputDevice) {
-                const defaultInputDevice = devices.find(device => device.deviceId === "default" && device.kind === 'audioinput');
-                const inputDevice = devices.find(device => device.groupId == defaultInputDevice.groupId && device.kind === 'audioinput' && device.deviceId != 'default');
+                const defaultInputDevice = audioDeviceManager.mediaDevices.find(device => device.deviceId === "default" && device.kind === 'audioinput');
+                const inputDevice = audioDeviceManager.mediaDevices.find(device => device.groupId == defaultInputDevice.groupId && device.kind === 'audioinput' && device.deviceId != 'default');
                 audioDeviceManager.changeAudioInputDevice(
                     inputDevice.deviceId,
                     (stream) => onInputDeviceChangeCallback(stream, inputDevice.deviceId),
@@ -158,13 +164,23 @@ const audioDeviceManager = {
                 );
             }
         } catch (error) {
-            logger.log("audioDeviceManager:_resetAudioDevice something went wrong", error);
+            logger.error("audioDeviceManager:_resetAudioDevice reset audio device failed", error);
         }
     },
 
-};
+    async enumerateDevices(callback) {
+        logger.log("audioDeviceManager:enumerateDevices entry")
+        try {
+            audioDeviceManager.mediaDevices = await navigator.mediaDevices.enumerateDevices();
+        } catch (e) {
+            logger.log("audioDeviceManager:enumerateDevices device enumeration failed", e);
+        }
+        if (callback) callback();
+    },
 
+};
 
+audioDeviceManager.enumerateDevices();
 /* harmony default export */ const __WEBPACK_DEFAULT_EXPORT__ = (audioDeviceManager);
 
 /***/ }),
@@ -21219,86 +21235,86 @@ function postInit(onInitDoneCallback) {
 
 const addPreferredCodec = (description) => {
 	logger.log("sipjsphone:addPreferredCodec entry");
-    // Ensure a preferred codec is set
-    if (!SIPJSPhone.preferredCodec) {
-        logger.info("sipjsphone:addPreferredCodec: No preferred codec set. Using default.");
-        return Promise.resolve(description);
-    }
+	// Ensure a preferred codec is set
+	if (!SIPJSPhone.preferredCodec) {
+		logger.info("sipjsphone:addPreferredCodec: No preferred codec set. Using default.");
+		return Promise.resolve(description);
+	}
 
-    const { payloadType, rtpMap, fmtp } = SIPJSPhone.preferredCodec;
-    const codecRtpMap = `a=rtpmap:${payloadType} ${rtpMap}`;
-    const codecFmtp = fmtp ? `a=fmtp:${payloadType} ${fmtp}` : "";
+	const { payloadType, rtpMap, fmtp } = SIPJSPhone.preferredCodec;
+	const codecRtpMap = `a=rtpmap:${payloadType} ${rtpMap}`;
+	const codecFmtp = fmtp ? `a=fmtp:${payloadType} ${fmtp}` : "";
 
-    logger.log("sipjsphone:addPreferredCodec: Original SDP:", description.sdp);
+	logger.log("sipjsphone:addPreferredCodec: Original SDP:", description.sdp);
 
-    // Parse SDP into lines
-    let sdpLines = description.sdp.split("\r\n");
+	// Parse SDP into lines
+	let sdpLines = description.sdp.split("\r\n");
 
-    // Check if Opus is already in the SDP
-    const existingOpusIndex = sdpLines.findIndex((line) => line.includes(`a=rtpmap`) && line.includes("opus/48000/2"));
-    const audioMLineIndex = sdpLines.findIndex((line) => line.startsWith("m=audio"));
+	// Check if Opus is already in the SDP
+	const existingOpusIndex = sdpLines.findIndex((line) => line.includes(`a=rtpmap`) && line.includes("opus/48000/2"));
+	const audioMLineIndex = sdpLines.findIndex((line) => line.startsWith("m=audio"));
 
-    if (existingOpusIndex !== -1 && audioMLineIndex !== -1) {
-        logger.log("sipjsphone:addPreferredCodec: Opus codec already exists. Prioritizing it.");
+	if (existingOpusIndex !== -1 && audioMLineIndex !== -1) {
+		logger.log("sipjsphone:addPreferredCodec: Opus codec already exists. Prioritizing it.");
 
-        // Extract and modify the audio m-line
-        let audioMLine = sdpLines[audioMLineIndex];
-        audioMLine = audioMLine.replace("RTP/SAVP", "RTP/AVP");
+		// Extract and modify the audio m-line
+		let audioMLine = sdpLines[audioMLineIndex];
+		audioMLine = audioMLine.replace("RTP/SAVP", "RTP/AVP");
 
-        const codecs = audioMLine.split(" ");
-        const mLineStart = codecs.slice(0, 3); // "m=audio <port> <protocol>"
-        const mLineCodecs = codecs.slice(3);
+		const codecs = audioMLine.split(" ");
+		const mLineStart = codecs.slice(0, 3); // "m=audio <port> <protocol>"
+		const mLineCodecs = codecs.slice(3);
 
-        // Move existing Opus payload type to the top
-        const opusPayloadType = sdpLines[existingOpusIndex].match(/a=rtpmap:(\d+)/)[1];
-        const opusIndex = mLineCodecs.indexOf(opusPayloadType);
+		// Move existing Opus payload type to the top
+		const opusPayloadType = sdpLines[existingOpusIndex].match(/a=rtpmap:(\d+)/)[1];
+		const opusIndex = mLineCodecs.indexOf(opusPayloadType);
 
-        if (opusIndex !== -1) {
-            // Remove Opus from its current position
-            mLineCodecs.splice(opusIndex, 1);
-        }
-        // Add Opus to the beginning of the codec list
-        mLineCodecs.unshift(opusPayloadType);
+		if (opusIndex !== -1) {
+			// Remove Opus from its current position
+			mLineCodecs.splice(opusIndex, 1);
+		}
+		// Add Opus to the beginning of the codec list
+		mLineCodecs.unshift(opusPayloadType);
 
-        // Update the audio m-line
-        sdpLines[audioMLineIndex] = `${mLineStart.join(" ")} ${mLineCodecs.join(" ")}`;
-    } else if (audioMLineIndex !== -1) {
-        logger.log("sipjsphone:addPreferredCodec: Opus codec not found. Adding it to SDP.");
+		// Update the audio m-line
+		sdpLines[audioMLineIndex] = `${mLineStart.join(" ")} ${mLineCodecs.join(" ")}`;
+	} else if (audioMLineIndex !== -1) {
+		logger.log("sipjsphone:addPreferredCodec: Opus codec not found. Adding it to SDP.");
 
-        // Extract and modify the audio m-line
-        let audioMLine = sdpLines[audioMLineIndex];
-        audioMLine = audioMLine.replace("RTP/SAVP", "RTP/AVP");
+		// Extract and modify the audio m-line
+		let audioMLine = sdpLines[audioMLineIndex];
+		audioMLine = audioMLine.replace("RTP/SAVP", "RTP/AVP");
 
-        const codecs = audioMLine.split(" ");
-        const mLineStart = codecs.slice(0, 3); // "m=audio <port> <protocol>"
-        const mLineCodecs = codecs.slice(3);
+		const codecs = audioMLine.split(" ");
+		const mLineStart = codecs.slice(0, 3); // "m=audio <port> <protocol>"
+		const mLineCodecs = codecs.slice(3);
 
-        // Add Opus payload type to the top
-        mLineCodecs.unshift(payloadType.toString());
+		// Add Opus payload type to the top
+		mLineCodecs.unshift(payloadType.toString());
 
-        // Update the audio m-line
-        sdpLines[audioMLineIndex] = `${mLineStart.join(" ")} ${mLineCodecs.join(" ")}`;
+		// Update the audio m-line
+		sdpLines[audioMLineIndex] = `${mLineStart.join(" ")} ${mLineCodecs.join(" ")}`;
 
-        // Add Opus-specific attributes to the SDP
-        if (!sdpLines.includes(codecRtpMap)) {
-            sdpLines.splice(audioMLineIndex + 1, 0, codecRtpMap); // Add rtpmap after m=audio
-        }
-        if (fmtp && !sdpLines.includes(codecFmtp)) {
-            sdpLines.splice(audioMLineIndex + 2, 0, codecFmtp); // Add fmtp after rtpmap
-        }
-    } else {
-        logger.error("sipjsphone:addPreferredCodec: No audio m-line found in SDP. Cannot modify.");
-        return Promise.resolve(description);
-    }
+		// Add Opus-specific attributes to the SDP
+		if (!sdpLines.includes(codecRtpMap)) {
+			sdpLines.splice(audioMLineIndex + 1, 0, codecRtpMap); // Add rtpmap after m=audio
+		}
+		if (fmtp && !sdpLines.includes(codecFmtp)) {
+			sdpLines.splice(audioMLineIndex + 2, 0, codecFmtp); // Add fmtp after rtpmap
+		}
+	} else {
+		logger.error("sipjsphone:addPreferredCodec: No audio m-line found in SDP. Cannot modify.");
+		return Promise.resolve(description);
+	}
 
-    // Remove any duplicate lines
-    sdpLines = [...new Set(sdpLines)];
+	// Remove any duplicate lines
+	sdpLines = [...new Set(sdpLines)];
 
-    // Combine back into SDP
-    description.sdp = sdpLines.join("\r\n");
-    logger.log("sipjsphone:addPreferredCodec: Modified SDP:", description.sdp);
+	// Combine back into SDP
+	description.sdp = sdpLines.join("\r\n");
+	logger.log("sipjsphone:addPreferredCodec: Modified SDP:", description.sdp);
 
-    return Promise.resolve(description);
+	return Promise.resolve(description);
 };
 
 function sipRegister() {
@@ -22024,14 +22040,14 @@ const SIPJSPhone = {
 		const codecPayloadTypes = {
 			opus: { payloadType: 111, rtpMap: "opus/48000/2", fmtp: "minptime=10;useinbandfec=1" },
 		};
-	
+
 		const preferredCodec = codecPayloadTypes[codecName.toLowerCase()];
 		if (!preferredCodec) {
 			logger.error("sipjsphone:setPreferredCodec: Unsupported code" + codecName + "specified.");
 			SIPJSPhone.preferredCodec = null; // Clear codec details if unsupported
 			return;
 		}
-	
+
 		SIPJSPhone.preferredCodec = preferredCodec;
 		logger.log("sipjsphone:setPreferredCodec: Preferred codec set to " + codecName);
 	},
@@ -22043,13 +22059,14 @@ const SIPJSPhone = {
 			if (_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioInputDeviceId != "default") {
 				newSess.accept({
 					sessionDescriptionHandlerOptions: {
-						constraints: { audio: { deviceId: _audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioInputDeviceId }, video: false },
+						constraints: { audio: { deviceId: _audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioInputDeviceId }, video: false }
 					},
-					sessionDescriptionHandlerModifiers: [addPreferredCodec],
+					sessionDescriptionHandlerModifiers: [addPreferredCodec]
 				}).catch((e) => {
 					onUserSessionAcceptFailed(e);
 				});
 			} else {
+
 				newSess.accept({
 					sessionDescriptionHandlerModifiers: [addPreferredCodec]
 				}).catch((e) => {
@@ -22135,21 +22152,30 @@ const SIPJSPhone = {
 				onSuccess();
 			} else {
 				logger.error("SIPJSPhone:changeAudioInputDevice failed");
-				onError("something went wrong , try again");
+				onError("replaceSenderTrack failed for webrtc");
 			}
 		}, onError);
 	},
-	onRemoteAudioOutputDeviceChanged(deviceId) {
-		ringtone.setSinkId(deviceId).catch((e) => {
-			logger.error("sipjsphone:onRemoteAudioOutputDeviceChanged ringtone changedevice failure ", e);
-		});
+	changeAudioOutputDeviceForAdditionalAudioElement(deviceId) {
+		const additionalAudioElements = [ringtone, beeptone, ringbacktone, dtmftone];
+		let i = 0;
+		let elem;
+		try {
+			for (i = 0; i < additionalAudioElements.length; i++) {
+				elem = additionalAudioElements[i];
+				elem.load();
+				elem.setSinkId(deviceId);
+			}
+		} catch (e) {
+			logger.error("sipjsphone:changeAudioOutputDeviceForAdditionalAudioElement failed to setSink for additonal AudioElements", e);
+		}
 	},
 	changeAudioOutputDevice(deviceId, onSuccess, onError) {
 		if (!ctxSip.callActiveID) {
 			audioRemote = document.createElement("audio");
 		}
 		_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.changeAudioOutputDevice(audioRemote, deviceId, function () {
-			SIPJSPhone.onRemoteAudioOutputDeviceChanged(deviceId);
+			SIPJSPhone.changeAudioOutputDeviceForAdditionalAudioElement(deviceId);
 			onSuccess();
 		}, onError);
 	},
@@ -22201,11 +22227,12 @@ const SIPJSPhone = {
 	},
 	audioInputDeviceChangeCallback: null,
 	audioOutputDeviceChangeCallback: null,
-	registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback) {
+	onDeviceChangeCallback: null,
+	registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback) {
 		logger.log(`SIPJSPhone:registerAudioDeviceChangeCallback  entry`);
 		SIPJSPhone.audioInputDeviceChangeCallback = audioInputDeviceChangeCallback;
 		SIPJSPhone.audioOutputDeviceChangeCallback = audioOutputDeviceChangeCallback;
-
+		SIPJSPhone.onDeviceChangeCallback = onDeviceChangeCallback;
 	}
 
 };
@@ -22216,22 +22243,31 @@ navigator.mediaDevices.addEventListener('devicechange', function (event) {
 		if (!ctxSip.callActiveID) {
 			audioRemote = document.createElement("audio");
 		}
-		_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.onAudioDeviceChange(audioRemote,
-			function (stream, deviceId) {
-				const trackChanged = SIPJSPhone.replaceSenderTrack(stream, deviceId);
-				if (trackChanged) {
-					_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioInputDeviceId = deviceId;
-					if (SIPJSPhone.audioInputDeviceChangeCallback) {
-						SIPJSPhone.audioInputDeviceChangeCallback(deviceId);
+		_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.enumerateDevices(function () {
+
+			if (SIPJSPhone.onDeviceChangeCallback) {
+				logger.info("SIPJSPhone:ondevicechange relaying event to callback");
+				SIPJSPhone.onDeviceChangeCallback(event);
+				return;
+			}
+			_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.onAudioDeviceChange(audioRemote,
+				function (stream, deviceId) {
+					const trackChanged = SIPJSPhone.replaceSenderTrack(stream, deviceId);
+					if (trackChanged) {
+						_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioInputDeviceId = deviceId;
+						if (SIPJSPhone.audioInputDeviceChangeCallback) {
+							SIPJSPhone.audioInputDeviceChangeCallback(deviceId);
+						}
 					}
-				}
-			}, function (deviceId) {
-				SIPJSPhone.onRemoteAudioOutputDeviceChanged(deviceId);
-				_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioOutputDeviceId = deviceId;
-				if (SIPJSPhone.audioOutputDeviceChangeCallback) {
-					SIPJSPhone.audioOutputDeviceChangeCallback(deviceId);
-				}
-			});
+				}, function (deviceId) {
+					SIPJSPhone.changeAudioOutputDeviceForAdditionalAudioElement(deviceId);
+					_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioOutputDeviceId = deviceId;
+					if (SIPJSPhone.audioOutputDeviceChangeCallback) {
+						SIPJSPhone.audioOutputDeviceChangeCallback(deviceId);
+					}
+				});
+		});
+
 	} catch (e) {
 		logger.error("SIPJSPhone:ondevicechange something went wrong during device change", e);
 	}
@@ -22462,15 +22498,13 @@ const webrtcSIPPhone = {
 		logger.log(`webrtcSIPPhone:changeAudioOutputDevice entry`);
 		_sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].changeAudioOutputDevice(deviceId, onSuccess, onError);
 	},
-
 	setPreferredCodec(codecName) {
 		logger.log("webrtcSIPPhone:setPreferredCodec entry");
 		_sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].setPreferredCodec(codecName);
 	},
-
-	registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback) {
+	registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback) {
 		logger.log(`webrtcSIPPhone:registerAudioDeviceChangeCallback  entry`);
-		_sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback);
+		_sipjsphone__WEBPACK_IMPORTED_MODULE_1__["default"].registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback);
 	},
 	getLogger() {
 		return _coreSDKLogger__WEBPACK_IMPORTED_MODULE_0__["default"];
@@ -24407,8 +24441,8 @@ class ExotelWebClient {
   registerLoggerCallback(callback) {
     logger.registerLoggerCallback(callback);
   }
-  registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback) {
-    _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.webrtcSIPPhone.registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback);
+  registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback) {
+    _exotel_npm_dev_webrtc_core_sdk__WEBPACK_IMPORTED_MODULE_8__.webrtcSIPPhone.registerAudioDeviceChangeCallback(audioInputDeviceChangeCallback, audioOutputDeviceChangeCallback, onDeviceChangeCallback);
   }
 }
 /* harmony default export */ const __WEBPACK_DEFAULT_EXPORT__ = (ExotelWebClient);