@exotel-npm-dev/webrtc-client-sdk 3.0.8 → 3.0.9

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Files changed (4) hide show
  1. package/Changelog +3 -0
  2. package/dist/exotelsdk.js +3 -8
  3. package/dist/exotelsdk.js.map +1 -1
  4. package/package.json +2 -2
package/Changelog CHANGED
@@ -1,5 +1,8 @@
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  Change Log
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+ ## v3.0.9 27 March, 2026
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+ -[VST-1577] Removed the same-device-ID guard from replaceSenderTrack.
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+
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  ## v3.0.6 19 December, 2025
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  -[VST-1269] fixing noise suppression bug and added wss transport in contact params
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package/dist/exotelsdk.js CHANGED
@@ -1,6 +1,6 @@
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  /*!
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  *
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- * WebRTC CLient SIP version 3.0.8
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+ * WebRTC CLient SIP version 3.0.9
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  *
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  */
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  (function webpackUniversalModuleDefinition(root, factory) {
@@ -21438,7 +21438,7 @@ class SIPJSPhone {
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  // Extract and modify the audio m-line
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  let audioMLine = sdpLines[audioMLineIndex];
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- audioMLine = audioMLine.replace("RTP/SAVP", "RTP/SAVP");
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+ audioMLine = audioMLine.replace("RTP/SAVP", "RTP/AVP");
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  const codecs = audioMLine.split(" ");
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  const mLineStart = codecs.slice(0, 3); // "m=audio <port> <protocol>"
@@ -21462,7 +21462,7 @@ class SIPJSPhone {
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  // Extract and modify the audio m-line
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  let audioMLine = sdpLines[audioMLineIndex];
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- audioMLine = audioMLine.replace("RTP/SAVP", "RTP/SAVP");
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+ audioMLine = audioMLine.replace("RTP/SAVP", "RTP/AVP");
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  const codecs = audioMLine.split(" ");
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  const mLineStart = codecs.slice(0, 3); // "m=audio <port> <protocol>"
@@ -22416,11 +22416,6 @@ destroySocketConnection() {
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  }
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  replaceSenderTrack(stream, deviceId) {
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  try {
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-
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- if (_audioDeviceManager_js__WEBPACK_IMPORTED_MODULE_0__.audioDeviceManager.currentAudioInputDeviceId == deviceId) {
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- this.stopStreamTracks(stream);
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- return false;
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- }
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  if (this.ctxSip.callActiveID) {
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  this.ctxSip.Stream = stream;
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  const s = this.ctxSip.Sessions[this.ctxSip.callActiveID];