move mediaServer.start() into try catch

2 files changed, 2 insertions, 2 deletions

diff --git a/src/webrtc/Server.ts b/src/webrtc/Server.ts

index 312a97c9..bc26f3d1 100644
--- a/src/webrtc/Server.ts
+++ b/src/webrtc/Server.ts
@@ -66,12 +66,12 @@ export class Server {
         // try to load webrtc library, if failed just don't start webrtc endpoint
         try {
             await loadWebRtcLibrary();
+            await mediaServer.start(WRTC_PUBLIC_IP, WRTC_PORT_MIN, WRTC_PORT_MAX);
         } catch (e) {
             console.log(`[WebRTC] ${yellow("WEBRTC disabled")}`);
             return;
         }
 
-        await mediaServer.start(WRTC_PUBLIC_IP, WRTC_PORT_MIN, WRTC_PORT_MAX);
         if (!this.server.listening) {
             this.server.listen(this.port);
             console.log(`[WebRTC] ${green(`online on 0.0.0.0:${this.port}`)}`);
diff --git a/src/webrtc/opcodes/Identify.ts b/src/webrtc/opcodes/Identify.ts

index 50976794..d61d4bbb 100644
--- a/src/webrtc/opcodes/Identify.ts
+++ b/src/webrtc/opcodes/Identify.ts
@@ -91,7 +91,7 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
     });
 
     // the server generates a unique ssrc for the audio and video stream. Must be unique among users connected to same server
-    // UDP clients will respect this ssrc, but websocket clients will generate and replace it with their own
+    // UDP clients will respect this ssrc, but webrtc clients will generate and replace it with their own
     const generatedSsrc: SSRCs = {
         audio_ssrc: generateSsrc(),
         video_ssrc: generateSsrc(),