@exotel-npm-dev/webrtc-client-sdk 3.0.8 → 3.0.10
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/Changelog +6 -0
- package/Makefile +6 -0
- package/dist/exotelsdk.js +4 -8
- package/dist/exotelsdk.js.map +1 -1
- package/package.json +2 -2
- package/src/api/callAPI/Call.js +1 -0
package/Changelog
CHANGED
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@@ -1,5 +1,11 @@
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Change Log
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## v3.0.10 01 April, 2026
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-[VST-1584]Updated webrtc-core-sdk to v3.0.10 with stack trace logging in error method
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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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## 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/Makefile
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package/dist/exotelsdk.js
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@@ -1,6 +1,6 @@
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/*!
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*
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* WebRTC CLient SIP version 3.0.
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* WebRTC CLient SIP version 3.0.10
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*
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*/
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(function webpackUniversalModuleDefinition(root, factory) {
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@@ -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/
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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>"
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@@ -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/
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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>"
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@@ -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];
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* When call is terminated
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*/
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logger.log('call ended');
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console.trace();
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webrtcSIPPhone.rejectCall();
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};
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this.MuteToggle = function () {
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