@exotel-npm-dev/webrtc-client-sdk 3.0.9 → 3.0.11

package/Changelog

@@ -1,5 +1,8 @@
 Change Log
 
+## v3.0.10 01 April, 2026
+-[VST-1584]Updated webrtc-core-sdk to v3.0.10 with stack trace logging in error method
+
 ## v3.0.9 27 March, 2026
 -[VST-1577] Removed the same-device-ID guard from replaceSenderTrack.
 

package/Makefile

@@ -1,3 +1,5 @@
+VERSION := $(shell npm pkg get version | tr -d '"')
+	
 build:dep
 	npm run build
 	cp -r ../webrtc-core-sdk/src/static/* dist/
@@ -16,3 +18,7 @@ dep:
 
 publish: build
 	npm publish
+
+
+tar: build
+	tar -czvf exotelsdk-$(VERSION).tar.gz dist/

package/dist/exotelsdk.js

@@ -1,6 +1,6 @@
 /*!
  * 
- * WebRTC CLient SIP version 3.0.9
+ * WebRTC CLient SIP version 3.0.11
  *
  */
 (function webpackUniversalModuleDefinition(root, factory) {
@@ -21304,8 +21304,18 @@ class SIPJSPhone {
 
 		},
 
-			// Update the sipSendDTMF method (around line 389) to use Web Audio:
+			// Local sidetone: play bundled dtmf.wav, then send SIP INFO DTMF.
 			sipSendDTMF: (digit) => {
+				logger.log("sipSendDTMF: digit", digit);
+				try {
+					if (this.ctxSip && this.ctxSip.dtmfTone) {
+						this.ctxSip.dtmfTone.currentTime = 0;
+						this.ctxSip.dtmfTone.play();
+					}
+				} catch (e) {
+					logger.log("sipSendDTMF: local DTMF tone exception:", e);
+				}
+
 				var a = this.ctxSip.callActiveID;
 				if (a) {
 					var s = this.ctxSip.Sessions[a];
@@ -21325,6 +21335,7 @@ class SIPJSPhone {
 					};
 					const requestOptions = { body };
 					return s.info({ requestOptions }).then(() => {
+						logger.log("sipSendDTMF: SIP INFO DTMF sent", dtmf);
 						return;
 					});
 
@@ -23526,6 +23537,7 @@ function Call(webrtcSIPPhone) {
      * When call is terminated
      */
     logger.log('call ended');
+    console.trace();
     webrtcSIPPhone.rejectCall();
   };
   this.MuteToggle = function () {