diff --git a/desktop/src/features/dictation/hooks/useRealtimeDictation.ts b/desktop/src/features/dictation/hooks/useRealtimeDictation.ts
index 15a7d6ed0..1b40953b1 100644
--- a/desktop/src/features/dictation/hooks/useRealtimeDictation.ts
+++ b/desktop/src/features/dictation/hooks/useRealtimeDictation.ts
@@ -11,6 +11,7 @@ import {
   BUFFER_COMMITTED_EVENT,
   TRANSCRIPT_COMPLETED_EVENT,
   TRANSCRIPT_DELTA_EVENT,
+  commitAudioBuffer,
   connectPeerConnection,
   createAudioBufferCapture,
   createPeerConnection,
@@ -18,6 +19,7 @@ import {
   flushAudioBuffer,
   getTranscriptText,
   mergeTranscriptEvent,
+  requiresManualCommit,
 } from "../lib/realtimeAudio";
 
 interface UseRealtimeDictationOptions {
@@ -58,6 +60,8 @@ export function useRealtimeDictation({
     createTranscriptSegmentState(),
   );
   const activeRunIdRef = useRef(0);
+  const manualCommitRef = useRef(false);
+  const commitIntervalRef = useRef<ReturnType<typeof setInterval> | null>(null);
   const onRecordingStartRef = useRef(onRecordingStart);
   const onTranscriptTextRef = useRef(onTranscriptText);
 
@@ -82,6 +86,45 @@ export function useRealtimeDictation({
   }, []);
 
   const cleanupResources = useCallback(() => {
+    // Clear periodic commit interval if active.
+    if (commitIntervalRef.current) {
+      clearInterval(commitIntervalRef.current);
+      commitIntervalRef.current = null;
+    }
+
+    // For manual-commit models (no server VAD), commit any buffered audio
+    // before tearing down the connection so OpenAI processes the final chunk.
+    // We keep the data channel open briefly to receive the transcript response.
+    const dc = dataChannelRef.current;
+    const needsCommit =
+      manualCommitRef.current && dc && dc.readyState === "open";
+    manualCommitRef.current = false;
+
+    if (needsCommit && dc) {
+      commitAudioBuffer(dc);
+      // Stop the mic immediately so no new audio is sent after commit.
+      for (const track of streamRef.current?.getTracks() ?? []) {
+        track.stop();
+      }
+      streamRef.current = null;
+      audioCaptureRef.current?.close();
+      audioCaptureRef.current = null;
+      // Delay full teardown to allow the final transcript to arrive.
+      const pc = peerConnectionRef.current;
+      peerConnectionRef.current = null;
+      dataChannelRef.current = null;
+      const runId = activeRunIdRef.current;
+      setTimeout(() => {
+        // Only tear down if no new run started in the meantime.
+        if (activeRunIdRef.current === runId) {
+          activeRunIdRef.current += 1;
+        }
+        dc.close();
+        pc?.close();
+      }, 3000);
+      return;
+    }
+
     activeRunIdRef.current += 1;
     closeResources({
       audioCapture: audioCaptureRef.current,
@@ -181,6 +224,7 @@ export function useRealtimeDictation({
         closeResources({ audioCapture, stream });
         return;
       }
+      manualCommitRef.current = requiresManualCommit(session.model);
 
       // 4. Set up WebRTC
       peerConnection = createPeerConnection();
@@ -203,6 +247,7 @@ export function useRealtimeDictation({
       // Flush buffered audio once data channel opens
       const channelToFlush = dataChannel;
       const captureToFlush = audioCapture;
+      const useManualCommit = manualCommitRef.current;
       dataChannel.addEventListener("open", () => {
         // If the user stopped (or restarted) recording between the SDP
         // exchange and the channel opening, drop this run's buffered audio.
@@ -211,6 +256,20 @@ export function useRealtimeDictation({
           return;
         }
         flushAudioBuffer(channelToFlush, captureToFlush.chunks);
+        // For manual-commit models, commit the initial buffered audio and
+        // start a periodic commit interval so streaming transcripts flow
+        // during recording (server VAD models commit automatically).
+        if (useManualCommit) {
+          commitAudioBuffer(channelToFlush);
+          // Commit every 2s to produce streaming transcript segments while
+          // the user is still speaking. Each commit triggers a transcription
+          // of the audio accumulated since the last commit.
+          commitIntervalRef.current = setInterval(() => {
+            if (channelToFlush.readyState === "open") {
+              commitAudioBuffer(channelToFlush);
+            }
+          }, 2000);
+        }
         captureToFlush.close();
         audioCaptureRef.current = null;
       });
diff --git a/desktop/src/features/dictation/lib/realtimeAudio.test.mjs b/desktop/src/features/dictation/lib/realtimeAudio.test.mjs
index adf0ca663..0121ef436 100644
--- a/desktop/src/features/dictation/lib/realtimeAudio.test.mjs
+++ b/desktop/src/features/dictation/lib/realtimeAudio.test.mjs
@@ -306,3 +306,29 @@ describe("mergeTranscriptEvent", () => {
     assert.equal(result, "Alpha. Bravo.");
   });
 });
+
+// Inline the logic to keep the test self-contained.
+function requiresManualCommit(model) {
+  return model.includes("realtime-whisper");
+}
+
+describe("requiresManualCommit", () => {
+  it("returns true for gpt-realtime-whisper", () => {
+    assert.equal(requiresManualCommit("gpt-realtime-whisper"), true);
+  });
+
+  it("returns true for versioned realtime-whisper model", () => {
+    assert.equal(
+      requiresManualCommit("gpt-4o-realtime-whisper-20250512"),
+      true,
+    );
+  });
+
+  it("returns false for whisper-1", () => {
+    assert.equal(requiresManualCommit("whisper-1"), false);
+  });
+
+  it("returns false for gpt-4o-transcribe", () => {
+    assert.equal(requiresManualCommit("gpt-4o-transcribe"), false);
+  });
+});
diff --git a/desktop/src/features/dictation/lib/realtimeAudio.ts b/desktop/src/features/dictation/lib/realtimeAudio.ts
index afac2ad33..4c46fec7b 100644
--- a/desktop/src/features/dictation/lib/realtimeAudio.ts
+++ b/desktop/src/features/dictation/lib/realtimeAudio.ts
@@ -235,3 +235,22 @@ export function flushAudioBuffer(
   }
   chunks.length = 0;
 }
+
+/**
+ * Send `input_audio_buffer.commit` to finalize buffered audio for transcription.
+ *
+ * Required when server VAD is disabled (e.g. `realtime-whisper` models) —
+ * without a commit, appended audio is never processed. For models using
+ * server VAD, the server commits automatically on speech boundaries.
+ */
+export function commitAudioBuffer(dataChannel: RTCDataChannel): void {
+  dataChannel.send(JSON.stringify({ type: "input_audio_buffer.commit" }));
+}
+
+/**
+ * Whether a transcription model requires manual audio commit (no server VAD).
+ * Models containing "realtime-whisper" use manual commit per OpenAI guidance.
+ */
+export function requiresManualCommit(model: string): boolean {
+  return model.includes("realtime-whisper");
+}