diff --git a/ui/src/pages/chat/realtime-talk-audio.test.ts b/ui/src/pages/chat/realtime-talk-audio.test.ts
index b0db4684e9b6..bb771a9e6263 100644
--- a/ui/src/pages/chat/realtime-talk-audio.test.ts
+++ b/ui/src/pages/chat/realtime-talk-audio.test.ts
@@ -95,6 +95,44 @@ describe("RealtimeTalkMediaStreamMeter", () => {
     expect(close).toHaveBeenCalledOnce();
   });
 
+  it("reclaims its interval when the initial level callback stops it", () => {
+    vi.useFakeTimers();
+    const close = vi.fn(async () => undefined);
+    const disconnectSource = vi.fn();
+    const disconnectAnalyser = vi.fn();
+    class MockAudioContext {
+      readonly close = close;
+      createMediaStreamSource() {
+        return { connect: vi.fn(), disconnect: disconnectSource };
+      }
+      createAnalyser() {
+        return {
+          fftSize: 0,
+          smoothingTimeConstant: 0,
+          disconnect: disconnectAnalyser,
+          getFloatTimeDomainData: (samples: Float32Array) => samples.fill(0.25),
+        };
+      }
+    }
+    vi.stubGlobal("AudioContext", MockAudioContext);
+    const onLevel = vi.fn((level: number) => {
+      if (level > 0) {
+        meter.stop();
+      }
+    });
+    const meter = new RealtimeTalkMediaStreamMeter(onLevel);
+
+    meter.start({} as MediaStream);
+    meter.stop();
+    meter.stop();
+    vi.advanceTimersByTime(1_000);
+
+    expect(vi.getTimerCount()).toBe(0);
+    expect(disconnectSource).toHaveBeenCalledOnce();
+    expect(disconnectAnalyser).toHaveBeenCalledOnce();
+    expect(close).toHaveBeenCalledOnce();
+  });
+
   it("closes an owned AudioContext when analyser setup fails", () => {
     const close = vi.fn(async () => undefined);
     class MockAudioContext {
diff --git a/ui/src/pages/chat/realtime-talk-gateway-relay.test.ts b/ui/src/pages/chat/realtime-talk-gateway-relay.test.ts
index aac2ba302117..a729faa6a7e8 100644
--- a/ui/src/pages/chat/realtime-talk-gateway-relay.test.ts
+++ b/ui/src/pages/chat/realtime-talk-gateway-relay.test.ts
@@ -614,6 +614,25 @@ describe("GatewayRelayRealtimeTalkTransport", () => {
     expect(onInputLevel).toHaveBeenLastCalledWith(0);
   });
 
+  it("reclaims the input meter when its first level update stops the transport", async () => {
+    vi.useFakeTimers();
+    const client = createClient();
+    const onInputLevel = vi.fn((level: number) => {
+      if (level > 0) {
+        transport.stop();
+      }
+    });
+    const transport = createTransport({ client, callbacks: { onInputLevel } });
+
+    await expect(transport.start()).resolves.toBe("ready");
+    transport.stop();
+    transport.stop();
+    vi.advanceTimersByTime(1_000);
+
+    expect(vi.getTimerCount()).toBe(0);
+    expect(requestCallsFor(client, "talk.session.close")).toHaveLength(1);
+  });
+
   it("bounds stalled microphone appends and aborts every owner on stop", async () => {
     const onStatus = vi.fn();
     const client = createClient();
diff --git a/ui/src/pages/chat/realtime-talk-webrtc.test.ts b/ui/src/pages/chat/realtime-talk-webrtc.test.ts
index 6bef1ac0e370..8a662e211f2d 100644
--- a/ui/src/pages/chat/realtime-talk-webrtc.test.ts
+++ b/ui/src/pages/chat/realtime-talk-webrtc.test.ts
@@ -223,6 +223,42 @@ describe("WebRtcSdpRealtimeTalkTransport", () => {
     expect(close).toHaveBeenCalledOnce();
   });
 
+  it("reclaims the input meter when its first level update stops the transport", async () => {
+    vi.useFakeTimers();
+    stubAnswerSdpFetch();
+    const close = vi.fn(async () => undefined);
+    class MockAudioContext {
+      readonly close = close;
+      createMediaStreamSource() {
+        return { connect: vi.fn(), disconnect: vi.fn() };
+      }
+      createAnalyser() {
+        return {
+          fftSize: 0,
+          smoothingTimeConstant: 0,
+          disconnect: vi.fn(),
+          getFloatTimeDomainData: (samples: Float32Array) => samples.fill(0.25),
+        };
+      }
+    }
+    vi.stubGlobal("AudioContext", MockAudioContext);
+    const onInputLevel = vi.fn((level: number) => {
+      if (level > 0) {
+        transport.stop();
+      }
+    });
+    const transport = createOpenAiTransport({}, { onInputLevel });
+
+    await expect(transport.start()).resolves.toBe("cancelled");
+    transport.stop();
+    transport.stop();
+    vi.advanceTimersByTime(1_000);
+
+    expect(vi.getTimerCount()).toBe(0);
+    expect(stopInputTrack).toHaveBeenCalledOnce();
+    expect(close).toHaveBeenCalledOnce();
+  });
+
   it("does not continue WebRTC setup when stopped while microphone access is pending", async () => {
     const fetchMock = vi.fn(async () => new Response("answer-sdp"));
     vi.stubGlobal("fetch", fetchMock as unknown as typeof fetch);