From 4550c88fdc20b87adc9acc6755955cc05b03243c Mon Sep 17 00:00:00 2001
From: Seungmin Kim <8457324+ehfd@users.noreply.github.com>
Date: Sat, 21 Sep 2024 13:47:56 +0900
Subject: [PATCH] Fix issues with relative path

---
 addons/gst-web/src/app.js | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

diff --git a/addons/gst-web/src/app.js b/addons/gst-web/src/app.js
index cfdfdbc9..e98b29a3 100644
--- a/addons/gst-web/src/app.js
+++ b/addons/gst-web/src/app.js
@@ -369,10 +369,12 @@ if (audioElement === null) {
 
 // WebRTC entrypoint, connect to the signalling server
 /*global WebRTCDemoSignalling, WebRTCDemo*/
+var pathname = window.location.pathname;
+pathname = pathname.slice(0, pathname.lastIndexOf("/") + 1);
 var protocol = (location.protocol == "http:" ? "ws://" : "wss://");
-var signalling = new WebRTCDemoSignalling(new URL(protocol + window.location.host + "/" + app.appName + "/signalling/"));
+var signalling = new WebRTCDemoSignalling(new URL(protocol + window.location.host + pathname + app.appName + "/signalling/"));
 var webrtc = new WebRTCDemo(signalling, videoElement, 1);
-var audio_signalling = new WebRTCDemoSignalling(new URL(protocol + window.location.host + "/" + app.appName + "/signalling/"));
+var audio_signalling = new WebRTCDemoSignalling(new URL(protocol + window.location.host + pathname + app.appName + "/signalling/"));
 var audio_webrtc = new WebRTCDemo(audio_signalling, audioElement, 3);
 
 // Function to add timestamp to logs.