diff --git a/shared/js/ConnectionHandler.ts b/shared/js/ConnectionHandler.ts
index b8882c0e..e5fbad2f 100644
--- a/shared/js/ConnectionHandler.ts
+++ b/shared/js/ConnectionHandler.ts
@@ -38,6 +38,7 @@ import {ChannelTree} from "./tree/ChannelTree";
 import {LocalClientEntry} from "./tree/Client";
 import {ServerAddress} from "./tree/Server";
 import {ChannelVideoFrame} from "tc-shared/ui/frames/video/Controller";
+import {global_client_actions} from "tc-shared/events/GlobalEvents";
 
 export enum InputHardwareState {
     MISSING,
@@ -392,18 +393,13 @@ export class ConnectionHandler {
                 control_bar.apply_server_voice_state();
             */
 
-            /*
-            this.serverConnection.getVoiceConnection().startWhisper({ target: "echo" }).catch(error => {
-                logError(LogCategory.CLIENT, tr("Failed to start local echo: %o"), error);
-            });
-             */
-            if(__build.target === "web") {
+            if(__build.target === "web" && settings.static_global(Settings.KEY_VOICE_ECHO_TEST_ENABLED)) {
                 this.serverFeatures.awaitFeatures().then(result => {
                     if(!result) {
                         return;
                     }
                     if(this.serverFeatures.supportsFeature(ServerFeature.WHISPER_ECHO)) {
-                        spawnEchoTestModal(this);
+                        global_client_actions.fire("action_open_window", { window: "server-echo-test", connection: this });
                     }
                 });
             }
diff --git a/shared/js/events/ClientGlobalControlHandler.ts b/shared/js/events/ClientGlobalControlHandler.ts
index fca41d03..b64fdd16 100644
--- a/shared/js/events/ClientGlobalControlHandler.ts
+++ b/shared/js/events/ClientGlobalControlHandler.ts
@@ -19,6 +19,7 @@ import {spawnAbout} from "tc-shared/ui/modal/ModalAbout";
 import {spawnVideoSourceSelectModal} from "tc-shared/ui/modal/video-source/Controller";
 import {LogCategory, logError} from "tc-shared/log";
 import {getVideoDriver} from "tc-shared/video/VideoSource";
+import {spawnEchoTestModal} from "tc-shared/ui/modal/echo-test/Controller";
 
 /*
 function initialize_sounds(event_registry: Registry<ClientGlobalControlEvents>) {
@@ -158,6 +159,13 @@ export function initialize(event_registry: Registry<ClientGlobalControlEvents>)
                 spawnAbout();
                 break;
 
+            case "server-echo-test":
+                const connection = event.connection || server_connections.active_connection();
+                if(connection) {
+                    spawnEchoTestModal(connection);
+                }
+                break;
+
             default:
                 console.warn(tr("Received open window event for an unknown window: %s"), event.window);
         }
diff --git a/shared/js/events/GlobalEvents.ts b/shared/js/events/GlobalEvents.ts
index 403595eb..6bd9a675 100644
--- a/shared/js/events/GlobalEvents.ts
+++ b/shared/js/events/GlobalEvents.ts
@@ -17,7 +17,8 @@ export interface ClientGlobalControlEvents {
             "ban-list" |
             "permissions" |
             "token-list" |
-            "token-use",
+            "token-use" |
+            "server-echo-test",
         connection?: ConnectionHandler
     },
 
diff --git a/web/app/rtc/Connection.ts b/web/app/rtc/Connection.ts
index 52677341..6e64d94f 100644
--- a/web/app/rtc/Connection.ts
+++ b/web/app/rtc/Connection.ts
@@ -217,7 +217,7 @@ class CommandHandler extends AbstractCommandHandler {
                 }).then(() => this.handle["peer"].createAnswer())
                 .then(async answer => {
                     if(RTCConnection.kEnableSdpTrace) {
-                        logTrace(LogCategory.WEBRTC, tr("Applying local description from remote %s:\n%s"), data.mode, answer.sdp);
+                        logTrace(LogCategory.WEBRTC, tr("Generated local answer due to remote %s:\n%s"), data.mode, answer.sdp);
                     }
                     answer.sdp = this.sdpProcessor.processOutgoingSdp(answer.sdp, "answer");
 
@@ -782,6 +782,10 @@ export class RTCConnection {
 
     private async updateTracks() {
         for(const type of kRtcSourceTrackTypes) {
+            if(!this.currentTransceiver[type]?.sender) {
+                continue;
+            }
+
             let fallback;
             switch (type) {
                 case "audio":
@@ -794,7 +798,15 @@ export class RTCConnection {
                     fallback = getIdleTrack("video");
                     break;
             }
-            await this.currentTransceiver[type]?.sender.replaceTrack(this.currentTracks[type] || fallback);
+
+            let target = this.currentTracks[type] || fallback;
+            if(this.currentTransceiver[type].sender.track === target) {
+                continue;
+            }
+
+            await this.currentTransceiver[type].sender.replaceTrack(target);
+            target.enabled = true;
+            console.error("Replaced track for %o (Fallback: %o)", type, target === fallback);
         }
     }
 
diff --git a/web/app/rtc/SdpUtils.ts b/web/app/rtc/SdpUtils.ts
index e0a24198..8e23e325 100644
--- a/web/app/rtc/SdpUtils.ts
+++ b/web/app/rtc/SdpUtils.ts
@@ -11,7 +11,7 @@ interface SdpCodec {
 }
 
 /* For optimal performance these payload formats should match the servers one */
-const OPUS_VOICE_PAYLOAD_TYPE = 111
+const OPUS_VOICE_PAYLOAD_TYPE = 111;
 const OPUS_MUSIC_PAYLOAD_TYPE = 112;
 const H264_PAYLOAD_TYPE = 126;
 
@@ -24,25 +24,27 @@ type SdpMedia = {
 
 export class SdpProcessor {
     private static readonly kAudioCodecs: SdpCodec[] = [
-        /* Primary audio format */
+        // Primary audio format
+        // Attention, the payload id should be the same as the server once.
+        // If not, Firefox start to make trouble and isn't replaying the sound...
         {
-            /* Opus Mono/Opus Voice */
+            // Opus Mono/Opus Voice
             payload: OPUS_VOICE_PAYLOAD_TYPE,
             codec: "opus",
             rate: 48000,
             encoding: 2,
-            fmtp: { minptime: 1, maxptime: 20, useinbandfec: 1, stereo: 0 },
+            fmtp: { minptime: 1, maxptime: 20, useinbandfec: 1, usedtx: 1, stereo: 0, "sprop-stereo": 0 },
             rtcpFb: [ "transport-cc" ]
         },
         {
-            /* Opus Stereo/Opus Music */
+            // Opus Stereo/Opus Music
             payload: OPUS_MUSIC_PAYLOAD_TYPE,
             codec: "opus",
             rate: 48000,
             encoding: 2,
-            fmtp: { minptime: 1, maxptime: 20, useinbandfec: 1, stereo: 1 },
+            fmtp: { minptime: 1, maxptime: 20, useinbandfec: 1, usedtx: 1, stereo: 1, "sprop-stereo": 1 },
             rtcpFb: [ "transport-cc" ]
-        }
+        },
     ];
 
     private static readonly kVideoCodecs: SdpCodec[] = [
@@ -193,6 +195,7 @@ export class SdpProcessor {
                      * This causes us only to add one, the primary, codec and hope for the best
                      * (Opus Stereo and Mono mixing seem to work right now 11.2020)
                      */
+                    /* TODO: Test this again since we're not sending a end signal via RTCP. This might change the behaviour? */
                     break;
                 }
             }
diff --git a/web/app/rtc/voice/Connection.ts b/web/app/rtc/voice/Connection.ts
index 9646a1b2..4d154e4e 100644
--- a/web/app/rtc/voice/Connection.ts
+++ b/web/app/rtc/voice/Connection.ts
@@ -365,6 +365,7 @@ export class RtpVoiceConnection extends AbstractVoiceConnection {
     stopWhisper() {
         if(this.whisperTarget) {
             this.whisperTarget.canceled = true;
+            this.whisperTarget = undefined;
             this.whisperTargetInitialize = undefined;
             this.connection.send_command("whispersessionreset").catch(error => {
                 logWarn(LogCategory.CLIENT, tr("Failed to clear the whisper target: %o"), error);