From f923faa0b88ce5bdd0072b8cf61da1830cbb9b13 Mon Sep 17 00:00:00 2001
From: Teknium <127238744+teknium1@users.noreply.github.com>
Date: Fri, 11 Sep 2026 03:36:27 -0700
Subject: [PATCH] =?UTF-8?q?feat(voice):=20GPT-Live=20voice=20chat=20mode?=
 =?UTF-8?q?=20=E2=80=94=20a=20full-duplex=20voice=20frontend=20that=20dele?=
 =?UTF-8?q?gates=20to=20Hermes=20(Desktop)?=
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Content-Transfer-Encoding: 8bit

`voice.voice_chat_mode: gpt-live` swaps the desktop's chained STT → turn → TTS
loop for OpenAI's gpt-live-1: one voice model that listens while it speaks and
has no tools of its own. Every real request it hears becomes a normal Hermes
turn on the open chat — any model/provider the session selected, full toolset,
memory, approvals — and the voice paraphrases the reply aloud.

Backend
- tools/voice_live.py: mode/credential/persona resolution and the one server-side
  step the API needs — POST /v1/live/sessions exchanging the renderer's SDP offer,
  pinned to client delegation; the OpenAI key never reaches the renderer. Voice
  persona follows the vendor prompting guide (role, style, labelled delegation
  policy describing Hermes as the backend). VOICE_LIVE_TURN_NOTE is the per-turn
  model-input note (transcript in, speakable prose out).
- REST: GET /api/audio/voice-live/status (mode + readiness, non-secret),
  POST /api/audio/voice-live/session (SDP exchange). The offer is passed
  byte-exact: a stripped trailing CRLF is a vendor 400 "unmarshal SDP: EOF".
- prompt.submit accepts surface=voice-live (+ voice_context) beside hud; the
  note rides the model input via the existing _prepend_note seam, the persisted
  user row stays the user's words, the system prompt stays byte-stable.
- config_defaults: voice.voice_chat_mode (chained|gpt-live), voice.gpt_live.*.

Desktop
- lib/voice-live.ts: RTCPeerConnection + oai-events data channel owner, transcript
  accumulation, session.commentary/thinking/instructions appends (500-token
  chunking), mute, graceful close waiting for session.closed.
- hooks/use-voice-live-conversation.ts: same public shape as useVoiceConversation;
  delegation → prompt.submit(surface=voice-live); tool activity → quiet thinking
  appends; reply streamed back per sentence; spoken stop phrase ends the chat;
  a newer delegation interrupts an in-flight turn.
- use-composer-voice mounts both engines and latches one at conversation start
  from the backend-resolved status; gpt-live without a key falls back to chained
  with a notice. Settings → Voice gets the mode dropdown, voice picker, persona.

Live-verified on the worktree desktop build (headless Electron, CDP, synthetic
mic): "what is 17 times 23 and which model are you on" → delegation → Hermes
(Claude Sonnet 4.5 via OpenRouter) → spoken "391 … Claude Sonnet 4.5 through
OpenRouter"; follow-up "double that" resolved from the spoken context → 782;
"run uname -r" ran the terminal tool with "Hermes is working: terminal" fed as
quiet context → spoken kernel version; "stop" closed the session
(reason=close_requested). Chained mode creates no RTCPeerConnection.
---
 .../chat/composer/hooks/use-composer-voice.ts |  97 +++-
 .../hooks/use-voice-live-conversation.test.ts |  47 ++
 .../hooks/use-voice-live-conversation.ts      | 435 ++++++++++++++
 .../app/session/hooks/use-hermes-config.ts    |   3 +
 .../hooks/use-prompt-actions/submit.ts        |   4 +
 .../session/hooks/use-prompt-actions/utils.ts |   7 +
 apps/desktop/src/app/settings/constants.ts    |  26 +-
 apps/desktop/src/i18n/en.ts                   |   6 +-
 apps/desktop/src/i18n/types.ts                |   4 +
 apps/desktop/src/i18n/zh.ts                   |  21 +-
 apps/desktop/src/lib/voice-live.ts            | 532 ++++++++++++++++++
 apps/desktop/src/store/voice-live.ts          |  36 ++
 hermes_cli/config_defaults.py                 |  13 +
 hermes_cli/web_models.py                      |   6 +
 hermes_cli/web_routers/audio.py               |  35 +-
 .../tui_gateway/test_voice_live_delegation.py | 123 ++++
 tools/voice_live.py                           | 186 ++++++
 tui_gateway/methods_prompt.py                 |  13 +-
 tui_gateway/session_notifications.py          |  15 +-
 .../docs/user-guide/features/voice-mode.md    |  18 +
 20 files changed, 1604 insertions(+), 23 deletions(-)
 create mode 100644 apps/desktop/src/app/chat/composer/hooks/use-voice-live-conversation.test.ts
 create mode 100644 apps/desktop/src/app/chat/composer/hooks/use-voice-live-conversation.ts
 create mode 100644 apps/desktop/src/lib/voice-live.ts
 create mode 100644 apps/desktop/src/store/voice-live.ts
 create mode 100644 tests/tui_gateway/test_voice_live_delegation.py
 create mode 100644 tools/voice_live.py

diff --git a/apps/desktop/src/app/chat/composer/hooks/use-composer-voice.ts b/apps/desktop/src/app/chat/composer/hooks/use-composer-voice.ts
index eb8307b234..f000c7b9ab 100644
--- a/apps/desktop/src/app/chat/composer/hooks/use-composer-voice.ts
+++ b/apps/desktop/src/app/chat/composer/hooks/use-composer-voice.ts
@@ -6,11 +6,13 @@ import { chatMessageText, collectUnspokenTurnSpeech } from '@/lib/chat-messages'
 import { triggerHaptic } from '@/lib/haptics'
 import { adoptSpokenReplySession, markAssistantIdSpoken, resolveSpokenReply } from '@/lib/spoken-reply'
 import { CONVERSATION_LEASE, READ_ALOUD_LEASE, syncTtsLease } from '@/lib/tts-lease'
+import { toLiveHistory } from '@/lib/voice-live'
 import { clearWakeIndicator, syncWakeIndicatorWithVoice } from '@/lib/wake-indicator'
 import { $voiceConversationStartRequest, takeVoiceConversationStart } from '@/store/composer'
 import { resetBrowseState } from '@/store/composer-input-history'
 import { $gateway } from '@/store/gateway'
 import { notify, notifyError } from '@/store/notifications'
+import { $voiceLiveStatus, refreshVoiceLiveStatus, selectedVoiceChatMode } from '@/store/voice-live'
 import { $autoSpeakReplies, $voiceStopPhrase, setAutoSpeakReplies } from '@/store/voice-prefs'
 import { resumeWakeAfterVoice } from '@/store/wake-word'
 
@@ -21,6 +23,7 @@ import type { ChatBarProps } from '../types'
 
 import { useAutoSpeakReplies } from './use-auto-speak-replies'
 import { useVoiceConversation } from './use-voice-conversation'
+import { useVoiceLiveConversation } from './use-voice-live-conversation'
 import { useVoiceRecorder } from './use-voice-recorder'
 
 interface UseComposerVoiceArgs {
@@ -64,6 +67,9 @@ export function useComposerVoice({
   // A tile's composer speaks ITS transcript, not the primary chat's.
   const { $messages } = useComposerScope()
   const [voiceConversationActive, setVoiceConversationActive] = useState(false)
+  // Engine selection is latched at conversation START (a Settings change
+  // applies to the next conversation, never mid-call).
+  const [liveEngineActive, setLiveEngineActive] = useState(false)
   const ownsWakeIndicatorRef = useRef(false)
   const previousSessionIdRef = useRef(sessionId)
   const voiceStartRequest = useStore($voiceConversationStartRequest)
@@ -135,6 +141,32 @@ export function useComposerVoice({
     await onSubmit(text)
   }
 
+  /** A GPT-Live delegation → Hermes turn. The bubble and the persisted row are
+   *  what the user said; the transcript window rides the model input only. */
+  const submitLiveDelegation = async (text: string, voiceContext: string) => {
+    triggerHaptic('submit')
+    resetBrowseState(sessionId)
+    clearDraft()
+    await onSubmit(text, { surface: 'voice-live', voiceContext })
+  }
+
+  /** Recent text turns of this chat, as GPT-Live startup history. */
+  const seedLiveHistory = () =>
+    toLiveHistory(
+      $messages
+        .get()
+        .filter(m => !m.hidden && (m.role === 'user' || m.role === 'assistant'))
+        .map(m => ({ role: m.role as 'assistant' | 'user', text: chatMessageText(m) }))
+    )
+
+  /** The tool Hermes is running right now, for quiet progress in the voice. */
+  const activeToolLabel = () => {
+    const last = $messages.get().findLast(m => m.role === 'assistant' && !m.hidden)
+    const running = last?.parts.findLast(part => part.type === 'tool-call' && part.result === undefined)
+
+    return running && running.type === 'tool-call' ? running.toolName : null
+  }
+
   const wakePausedRef = useRef(false)
   // Resolves once the in-flight wake.pause round-trip completes (mic released by
   // the wake listener). The conversation awaits this before opening its own mic
@@ -143,10 +175,10 @@ export function useComposerVoice({
   // fail and the conversation never starts listening.
   const wakePauseBarrierRef = useRef<Promise<void> | null>(null)
 
-  const conversation = useVoiceConversation({
+  const chainedConversation = useVoiceConversation({
     busy,
     consumePendingResponse,
-    enabled: voiceConversationActive,
+    enabled: voiceConversationActive && !liveEngineActive,
     onFatalError: () => setVoiceConversationActive(false),
     // Speaking over the model mid-generation interrupts the in-flight turn —
     // the same seam as the Stop button — so the interjection becomes the next
@@ -165,6 +197,53 @@ export function useComposerVoice({
     beforeMicOpen: () => wakePauseBarrierRef.current ?? undefined
   })
 
+  const liveConversation = useVoiceLiveConversation({
+    activeToolLabel,
+    beforeMicOpen: () => wakePauseBarrierRef.current ?? undefined,
+    busy,
+    consumePendingResponse,
+    enabled: voiceConversationActive && liveEngineActive,
+    onFatalError: () => setVoiceConversationActive(false),
+    onInterrupt,
+    onStopWord: () => setVoiceConversationActive(false),
+    onSubmit: submitLiveDelegation,
+    pendingResponse: pendingTurnResponse,
+    seedHistory: seedLiveHistory
+  })
+
+  const conversation = liveEngineActive ? liveConversation : chainedConversation
+
+  /** Turn the conversation on with the engine `voice.voice_chat_mode` selects,
+   *  decided in the same state batch so the other engine never sees a frame of
+   *  `enabled`. gpt-live selected but not startable (no OpenAI key on the
+   *  gateway) falls back to chained with a notice rather than a dead button. */
+  const activateConversation = useCallback(() => {
+    const status = $voiceLiveStatus.get()
+    let live = false
+
+    if (selectedVoiceChatMode(status) === 'gpt-live') {
+      if (status?.available) {
+        live = true
+      } else {
+        notify({
+          id: 'voice-live-unavailable',
+          kind: 'warning',
+          message: t.notifications.voice.liveUnavailable(status?.reason ?? 'not configured')
+        })
+      }
+    }
+
+    setLiveEngineActive(live)
+    setVoiceConversationActive(true)
+  }, [t])
+
+  useEffect(() => {
+    if (!voiceConversationActive) {
+      // Prefetch so the first press picks the right engine without a round trip.
+      void refreshVoiceLiveStatus().catch(() => undefined)
+    }
+  }, [voiceConversationActive])
+
   // eslint-disable-next-line no-restricted-syntax -- ownership token used only by unmount cleanup
   useEffect(() => {
     if (target !== 'main') {
@@ -197,9 +276,9 @@ export function useComposerVoice({
       setVoiceConversationActive(false)
       void conversation.end()
     } else {
-      setVoiceConversationActive(true)
+      activateConversation()
     }
-  }, [conversation, disabled, voiceConversationActive])
+  }, [activateConversation, conversation, disabled, voiceConversationActive])
 
   useEffect(
     () => onComposerVoiceToggleRequest(toggled => toggled === target && toggleVoiceConversation()),
@@ -208,9 +287,9 @@ export function useComposerVoice({
 
   useEffect(() => {
     if (target === 'main' && !disabled && takeVoiceConversationStart(voiceStartRequest) && !voiceConversationActive) {
-      setVoiceConversationActive(true)
+      activateConversation()
     }
-  }, [disabled, target, voiceConversationActive, voiceStartRequest])
+  }, [activateConversation, disabled, target, voiceConversationActive, voiceStartRequest])
 
   const resumeWakeIfPaused = useCallback(() => {
     if (!wakePausedRef.current) {
@@ -279,8 +358,8 @@ export function useComposerVoice({
   // lease, and the backend unloads resident local models once no surface holds
   // one. Fire-and-forget — the toggle never waits on or fails from this.
   useEffect(() => {
-    void syncTtsLease(CONVERSATION_LEASE, voiceConversationActive)
-  }, [voiceConversationActive])
+    void syncTtsLease(CONVERSATION_LEASE, voiceConversationActive && !liveEngineActive)
+  }, [liveEngineActive, voiceConversationActive])
 
   useEffect(() => () => void syncTtsLease(CONVERSATION_LEASE, false), [])
 
@@ -295,7 +374,7 @@ export function useComposerVoice({
 
   // Explicit start/end for the on-screen conversation controls (the hotkey uses
   // the gated toggle above).
-  const startConversation = useCallback(() => setVoiceConversationActive(true), [])
+  const startConversation = activateConversation
 
   const endConversation = useCallback(() => {
     setVoiceConversationActive(false)
diff --git a/apps/desktop/src/app/chat/composer/hooks/use-voice-live-conversation.test.ts b/apps/desktop/src/app/chat/composer/hooks/use-voice-live-conversation.test.ts
new file mode 100644
index 0000000000..5046a8f59e
--- /dev/null
+++ b/apps/desktop/src/app/chat/composer/hooks/use-voice-live-conversation.test.ts
@@ -0,0 +1,47 @@
+// @vitest-environment jsdom
+import { describe, expect, it } from 'vitest'
+
+import { chunkForCommentary, toLiveHistory } from '@/lib/voice-live'
+
+import { delegationPrompt } from './use-voice-live-conversation'
+
+describe('GPT-Live delegation → Hermes turn', () => {
+  it('sends the latest user words as the turn and the exchange as model-only context', () => {
+    // The delegation event carries no text: both are reconstructed from
+    // transcript deltas, fragments of one speaker concatenated as received.
+    const { context, prompt } = delegationPrompt([
+      { endMs: 1000, speaker: 'assistant', startMs: 0, text: 'Hi, how ' },
+      { endMs: 1500, speaker: 'assistant', startMs: 1000, text: 'can I help?' },
+      { endMs: 2500, speaker: 'user', startMs: 1500, text: 'What is ' },
+      { endMs: 3200, speaker: 'user', startMs: 2500, text: 'the weather in Paris?' }
+    ])
+
+    expect(prompt).toBe('What is the weather in Paris?')
+    expect(context).toContain('Voice assistant: Hi, how can I help?')
+    expect(context).toContain('User: What is the weather in Paris?')
+  })
+
+  it('splits a long reply into vendor-sized commentary appends on sentence boundaries', () => {
+    const sentence = 'This is a sentence about the result. '
+    const chunks = chunkForCommentary(sentence.repeat(80), 400)
+
+    expect(chunks.length).toBeGreaterThan(1)
+    expect(chunks.every(chunk => chunk.length <= 400)).toBe(true)
+    expect(chunks.every(chunk => chunk.endsWith('.'))).toBe(true)
+    expect(chunks.join(' ')).toBe(sentence.repeat(80).trim())
+  })
+
+  it('seeds the live session with the most recent text turns within budget', () => {
+    const turns = Array.from({ length: 40 }, (_, index) => ({
+      role: (index % 2 === 0 ? 'user' : 'assistant') as 'assistant' | 'user',
+      text: `turn ${index}`
+    }))
+
+    const history = toLiveHistory(turns, 6)
+
+    expect(history).toHaveLength(6)
+    expect(history.at(-1)?.content[0]?.text).toBe('turn 39')
+    expect(history[0]?.role).toBe('user')
+    expect(history.find(m => m.role === 'assistant')?.content[0]?.type).toBe('output_text')
+  })
+})
diff --git a/apps/desktop/src/app/chat/composer/hooks/use-voice-live-conversation.ts b/apps/desktop/src/app/chat/composer/hooks/use-voice-live-conversation.ts
new file mode 100644
index 0000000000..d3d4a02ad4
--- /dev/null
+++ b/apps/desktop/src/app/chat/composer/hooks/use-voice-live-conversation.ts
@@ -0,0 +1,435 @@
+import { useCallback, useEffect, useRef, useState } from 'react'
+
+import { useI18n } from '@/i18n'
+import { sanitizeTextForSpeech } from '@/lib/speech-text'
+import {
+  type LiveHistoryMessage,
+  type LiveTranscriptFragment,
+  VoiceLiveSession
+} from '@/lib/voice-live'
+import { isVoiceStopCommand } from '@/lib/voice-stop-word'
+import { notify, notifyError } from '@/store/notifications'
+
+import type { ConversationStatus } from './use-voice-conversation'
+
+/** How long an accepted delegation may sit before the gateway shows the turn running. */
+const SUBMIT_SETTLE_GRACE_MS = 15_000
+/** Quiet after the last user transcript fragment before the utterance is judged
+ *  as a whole ("stop" ends the chat; "stop the container" is a request). */
+const UTTERANCE_SETTLE_MS = 1_500
+
+interface PendingVoiceResponse {
+  id: string
+  pending: boolean
+  text: string
+}
+
+interface VoiceLiveConversationOptions {
+  busy: boolean
+  enabled: boolean
+  onFatalError?: () => void
+  /** Interrupt the in-flight Hermes turn (Stop-button seam). Fired when a new
+   *  delegation supersedes one still running. */
+  onInterrupt?: () => Promise<void> | void
+  onStopWord?: () => void
+  /** Submit a Hermes turn: `text` is the user's last words (the bubble and the
+   *  persisted row), `voiceContext` the recent spoken exchange for the model. */
+  onSubmit: (text: string, voiceContext: string) => Promise<void> | void
+  pendingResponse: () => PendingVoiceResponse | null
+  consumePendingResponse: () => void
+  /** Text turns to seed the live model with when the session opens. */
+  seedHistory: () => LiveHistoryMessage[]
+  /** Names of tools currently running in the turn (quiet progress for the voice). */
+  activeToolLabel?: () => null | string
+  beforeMicOpen?: () => Promise<void> | void
+}
+
+/** Turn transcript fragments into the Hermes turn: `prompt` is what the user
+ *  last said (the persisted user row), `context` the recent spoken exchange
+ *  that rides the model input only (see tools/voice_live.py). */
+export function delegationPrompt(context: LiveTranscriptFragment[]): { context: string; prompt: string } {
+  const turns: Array<{ speaker: 'assistant' | 'user'; text: string }> = []
+
+  for (const fragment of context) {
+    const last = turns.at(-1)
+
+    if (last && last.speaker === fragment.speaker) {
+      last.text += fragment.text
+    } else {
+      turns.push({ speaker: fragment.speaker, text: fragment.text })
+    }
+  }
+
+  const lastUser = [...turns].reverse().find(turn => turn.speaker === 'user')
+  const prompt = (lastUser?.text ?? '').replace(/\s+/g, ' ').trim()
+
+  const transcript = turns
+    .map(turn => `${turn.speaker === 'user' ? 'User' : 'Voice assistant'}: ${turn.text.replace(/\s+/g, ' ').trim()}`)
+    .filter(line => !line.endsWith(': '))
+    .join('\n')
+
+  return { context: transcript, prompt: prompt || transcript.slice(-400) }
+}
+
+/**
+ * GPT-Live conversation engine — same public shape as `useVoiceConversation`
+ * so the composer can mount either from `voice.voice_chat_mode`.
+ *
+ * Status mapping: `listening` = session up, voice idle; `speaking` = the
+ * remote track is producing audio; `thinking` = a delegation is in flight in
+ * Hermes. There is no `transcribing` phase: the voice model owns speech.
+ */
+export function useVoiceLiveConversation({
+  busy,
+  enabled,
+  onFatalError,
+  onInterrupt,
+  onStopWord,
+  onSubmit,
+  pendingResponse,
+  consumePendingResponse,
+  seedHistory,
+  activeToolLabel,
+  beforeMicOpen
+}: VoiceLiveConversationOptions) {
+  const { t } = useI18n()
+  const voiceCopy = t.notifications.voice
+  const [status, setStatus] = useState<ConversationStatus>('idle')
+  const [muted, setMuted] = useState(false)
+  const [level, setLevel] = useState(0)
+  // Mirrors delegationRef for the reply-drive effect: a new delegation must
+  // restart the feed loop, and a ref write alone does not re-render.
+  const [activeDelegation, setActiveDelegation] = useState<null | string>(null)
+  const sessionRef = useRef<null | VoiceLiveSession>(null)
+  // Bumped by every start/end so an in-flight start() that lost the race
+  // (StrictMode double-effect, quick toggle) closes its session instead of
+  // leaving a second billed one running.
+  const startEpochRef = useRef(0)
+  const startingRef = useRef(false)
+  // Set at delegation submit; a turn is only "settled" once it has been seen
+  // running (busy) or produced a reply — the gateway ack lags the submit.
+  const turnObservedRef = useRef(false)
+  const submittedAtRef = useRef(0)
+  const enabledRef = useRef(enabled)
+  const busyRef = useRef(busy)
+  const speakingRef = useRef(false)
+  const userUtteranceRef = useRef('')
+  const utteranceTimerRef = useRef<null | number>(null)
+  const delegationRef = useRef<null | string>(null)
+  const spokenLengthRef = useRef(0)
+  const spokenResponseIdRef = useRef<null | string>(null)
+  const lastToolLabelRef = useRef<null | string>(null)
+  const wasEnabledRef = useRef(enabled)
+  const latest = useRef({ activeToolLabel, beforeMicOpen, onFatalError, onInterrupt, onStopWord, onSubmit, pendingResponse, consumePendingResponse, seedHistory })
+  latest.current = { activeToolLabel, beforeMicOpen, onFatalError, onInterrupt, onStopWord, onSubmit, pendingResponse, consumePendingResponse, seedHistory }
+
+  // eslint-disable-next-line no-restricted-syntax -- legitimate non-atom ref write (see eslint rule comment)
+  useEffect(() => {
+    enabledRef.current = enabled
+  }, [enabled])
+
+  // eslint-disable-next-line no-restricted-syntax -- legitimate non-atom ref write (see eslint rule comment)
+  useEffect(() => {
+    busyRef.current = busy
+  }, [busy])
+
+  const setDelegation = useCallback((id: null | string) => {
+    delegationRef.current = id
+    setActiveDelegation(id)
+  }, [])
+
+  const refreshStatus = useCallback(() => {
+    if (!sessionRef.current) {
+      setStatus('idle')
+
+      return
+    }
+
+    if (speakingRef.current) {
+      setStatus('speaking')
+    } else if (delegationRef.current) {
+      setStatus('thinking')
+    } else {
+      setStatus('listening')
+    }
+  }, [])
+
+  const end = useCallback(async () => {
+    startEpochRef.current += 1
+    startingRef.current = false
+
+    if (utteranceTimerRef.current) {
+      window.clearTimeout(utteranceTimerRef.current)
+      utteranceTimerRef.current = null
+    }
+
+    userUtteranceRef.current = ''
+    const session = sessionRef.current
+    sessionRef.current = null
+    setDelegation(null)
+    spokenResponseIdRef.current = null
+    spokenLengthRef.current = 0
+    speakingRef.current = false
+    session?.close()
+    setMuted(false)
+    setLevel(0)
+    setStatus('idle')
+  }, [setDelegation])
+
+  const start = useCallback(async () => {
+    if (sessionRef.current || startingRef.current) {
+      return
+    }
+
+    startingRef.current = true
+    const epoch = ++startEpochRef.current
+
+    try {
+      await latest.current.beforeMicOpen?.()
+    } catch {
+      // A wake-pause failure must not block an explicit start.
+    }
+
+    if (!enabledRef.current || startEpochRef.current !== epoch) {
+      startingRef.current = false
+
+      return
+    }
+
+    const session = new VoiceLiveSession({
+      // The voice model answers a bare "stop" itself (it just goes quiet) and
+      // never delegates it, so the spoken stop phrase is judged on the user
+      // transcript once the utterance settles.
+      onTranscript: fragment => {
+        if (fragment.speaker !== 'user') {
+          return
+        }
+
+        userUtteranceRef.current += fragment.text
+
+        if (utteranceTimerRef.current) {
+          window.clearTimeout(utteranceTimerRef.current)
+        }
+
+        utteranceTimerRef.current = window.setTimeout(() => {
+          utteranceTimerRef.current = null
+          const utterance = userUtteranceRef.current
+          userUtteranceRef.current = ''
+
+          if (sessionRef.current === session && isVoiceStopCommand(utterance)) {
+            void end()
+            latest.current.onStopWord?.()
+          }
+        }, UTTERANCE_SETTLE_MS)
+      },
+      onClosed: (reason, usageSeconds) => {
+        if (sessionRef.current !== session) {
+          return
+        }
+
+        sessionRef.current = null
+        setDelegation(null)
+        setStatus('idle')
+
+        if (reason !== 'close_requested') {
+          notify({
+            kind: 'warning',
+            message: usageSeconds != null ? `${reason} (${Math.round(usageSeconds)}s)` : reason,
+            title: voiceCopy.liveEnded
+          })
+          latest.current.onFatalError?.()
+        }
+      },
+      onDelegation: (delegationId, context) => {
+        if (sessionRef.current !== session) {
+          return
+        }
+
+        const { context: voiceContext, prompt } = delegationPrompt(context)
+
+        // A spoken stop command ends the conversation instead of becoming a turn.
+        if (prompt && isVoiceStopCommand(prompt)) {
+          void end()
+          latest.current.onStopWord?.()
+
+          return
+        }
+
+        // A newer request supersedes an in-flight turn: stop it so the answer
+        // the voice speaks is for what the user asked last.
+        if (busyRef.current) {
+          void latest.current.onInterrupt?.()
+        }
+
+        setDelegation(delegationId)
+        spokenResponseIdRef.current = null
+        spokenLengthRef.current = 0
+        lastToolLabelRef.current = null
+        turnObservedRef.current = false
+        submittedAtRef.current = Date.now()
+        latest.current.consumePendingResponse()
+        refreshStatus()
+        void Promise.resolve(latest.current.onSubmit(prompt, voiceContext)).catch(error => {
+          notifyError(error, voiceCopy.liveDelegationFailed)
+          session.speak(delegationId, 'Sorry, I could not reach Hermes for that request.')
+          setDelegation(null)
+          refreshStatus()
+        })
+      },
+      onError: (message, fatal) => {
+        notify({ kind: fatal ? 'error' : 'warning', message, title: voiceCopy.liveError })
+      },
+      onSpeakingChange: speaking => {
+        speakingRef.current = speaking
+        setLevel(speaking ? 0.6 : 0)
+        refreshStatus()
+      }
+    })
+
+    sessionRef.current = session
+    startingRef.current = false
+    setMuted(false)
+    setStatus('thinking')
+
+    try {
+      await session.start(latest.current.seedHistory())
+
+      if (sessionRef.current !== session || startEpochRef.current !== epoch) {
+        session.close()
+
+        return
+      }
+
+      refreshStatus()
+    } catch (error) {
+      if (sessionRef.current === session) {
+        sessionRef.current = null
+      }
+
+      session.close()
+
+      if (startEpochRef.current !== epoch) {
+        return
+      }
+
+      notifyError(error, voiceCopy.couldNotStartSession)
+      setStatus('idle')
+      latest.current.onFatalError?.()
+    }
+  }, [end, refreshStatus, setDelegation, voiceCopy.couldNotStartSession, voiceCopy.liveDelegationFailed, voiceCopy.liveEnded, voiceCopy.liveError])
+
+  // Drive the reply back into the voice: stream commentary as Hermes writes
+  // it (sentence-chunked), quiet tool progress as thinking appends, and clear
+  // the delegation when the turn settles.
+  // eslint-disable-next-line no-restricted-syntax -- turn-coordination refs (delegation id / spoken cursor), not atom mirrors
+  useEffect(() => {
+    const session = sessionRef.current
+    const delegationId = delegationRef.current
+
+    if (!session || !delegationId) {
+      return undefined
+    }
+
+    const tick = () => {
+      if (sessionRef.current !== session || delegationRef.current !== delegationId) {
+        return
+      }
+
+      if (busyRef.current) {
+        turnObservedRef.current = true
+      }
+
+      const tool = latest.current.activeToolLabel?.() ?? null
+
+      if (tool && tool !== lastToolLabelRef.current) {
+        lastToolLabelRef.current = tool
+        session.think(delegationId, `Hermes is working: ${tool}. Not done yet.`)
+      }
+
+      const response = latest.current.pendingResponse()
+
+      if (response) {
+        turnObservedRef.current = true
+
+        if (spokenResponseIdRef.current !== response.id) {
+          spokenResponseIdRef.current = response.id
+          spokenLengthRef.current = 0
+        }
+
+        const spoken = sanitizeTextForSpeech(response.text)
+
+        // Append only completed sentences while streaming; the tail lands on settle.
+        if (response.pending || busyRef.current) {
+          const boundary = spoken.lastIndexOf('. ', spoken.length - 2)
+          const cut = boundary > spokenLengthRef.current ? boundary + 1 : spokenLengthRef.current
+
+          if (cut > spokenLengthRef.current) {
+            session.speak(delegationId, spoken.slice(spokenLengthRef.current, cut))
+            spokenLengthRef.current = cut
+          }
+
+          return
+        }
+
+        if (spoken.length > spokenLengthRef.current) {
+          session.speak(delegationId, spoken.slice(spokenLengthRef.current))
+          spokenLengthRef.current = spoken.length
+        }
+
+        latest.current.consumePendingResponse()
+        setDelegation(null)
+        refreshStatus()
+
+        return
+      }
+
+      // The submit ack lags: give the turn time to be seen running before
+      // reading "idle and no reply" as a finished turn.
+      if (!busyRef.current && (turnObservedRef.current || Date.now() - submittedAtRef.current > SUBMIT_SETTLE_GRACE_MS)) {
+        // Turn settled without a speakable reply (tool-only, error, interrupted).
+        if (spokenLengthRef.current === 0) {
+          session.think(delegationId, 'Hermes finished that request without a spoken result.')
+        }
+
+        setDelegation(null)
+        refreshStatus()
+      }
+    }
+
+    const timer = window.setInterval(tick, 200)
+    tick()
+
+    return () => window.clearInterval(timer)
+  }, [activeDelegation, busy, refreshStatus, setDelegation, status])
+
+  const toggleMute = useCallback(() => {
+    setMuted(value => {
+      const next = !value
+      sessionRef.current?.setMuted(next)
+
+      return next
+    })
+  }, [])
+
+  /** No explicit turn boundary in full duplex; a nudge tells the voice to answer now. */
+  const stopTurn = useCallback(() => {
+    sessionRef.current?.instruct('The user has finished speaking. Respond now to what they said.')
+  }, [])
+
+  // eslint-disable-next-line no-restricted-syntax -- legitimate non-atom ref write (see eslint rule comment)
+  useEffect(() => {
+    if (enabled && !wasEnabledRef.current) {
+      void start()
+    }
+
+    if (!enabled && wasEnabledRef.current) {
+      void end()
+    }
+
+    wasEnabledRef.current = enabled
+  }, [enabled, end, start])
+
+  useEffect(() => () => void end(), [end])
+
+  return { end, level, muted, start, status, stopTurn, toggleMute }
+}
diff --git a/apps/desktop/src/app/session/hooks/use-hermes-config.ts b/apps/desktop/src/app/session/hooks/use-hermes-config.ts
index c560ee0d5b..ebf6a85dd9 100644
--- a/apps/desktop/src/app/session/hooks/use-hermes-config.ts
+++ b/apps/desktop/src/app/session/hooks/use-hermes-config.ts
@@ -16,6 +16,7 @@ import {
   setDefaultReasoningEffort,
   setIntroPersonality
 } from '@/store/session'
+import { refreshVoiceLiveStatus } from '@/store/voice-live'
 import {
   applyAutoSpeakFromConfig,
   applyThinkingSoundFromConfig,
@@ -147,6 +148,8 @@ export function useHermesConfig({ activeSessionIdRef }: HermesConfigOptions) {
         applyAutoSpeakFromConfig(config)
         applyVoiceStopPhraseFromConfig(config)
         applyThinkingSoundFromConfig(config)
+        // Resolved server-side (mode + whether a key resolves); non-critical.
+        void refreshVoiceLiveStatus().catch(() => undefined)
       } catch {
         // Config is nice-to-have; chat still works without it.
       }
diff --git a/apps/desktop/src/app/session/hooks/use-prompt-actions/submit.ts b/apps/desktop/src/app/session/hooks/use-prompt-actions/submit.ts
index bea099da65..a47ab2e25b 100644
--- a/apps/desktop/src/app/session/hooks/use-prompt-actions/submit.ts
+++ b/apps/desktop/src/app/session/hooks/use-prompt-actions/submit.ts
@@ -765,6 +765,10 @@ export function useSubmitPrompt(deps: SubmitPromptDeps) {
           // rather than at Hermes. The gateway turns this into a per-turn hint
           // to read the window underneath and work in it.
           ...($hudMode.get() && { surface: 'hud' }),
+          // A GPT-Live delegation: the text is a voice transcript and the reply
+          // will be spoken by the voice model. Wins over HUD for this turn.
+          ...(options?.surface && { surface: options.surface }),
+          ...(options?.surface && options.voiceContext && { voice_context: options.voiceContext }),
           // A queue drain is a "run after" message, never a live-turn
           // correction. The flag tells the gateway's busy path to hold it for
           // the next turn untouched — without it, losing the settle race
diff --git a/apps/desktop/src/app/session/hooks/use-prompt-actions/utils.ts b/apps/desktop/src/app/session/hooks/use-prompt-actions/utils.ts
index ef315bbcc7..57ca8e4293 100644
--- a/apps/desktop/src/app/session/hooks/use-prompt-actions/utils.ts
+++ b/apps/desktop/src/app/session/hooks/use-prompt-actions/utils.ts
@@ -719,6 +719,13 @@ export interface SubmitTextOptions {
    *  renders anywhere — the off-screen path for widget intents. The agent
    *  still receives the text as a normal user turn. */
   displayKind?: 'hidden'
+  /** Per-turn client surface the gateway turns into a model-bound note. The
+   *  HUD sets `hud` from its own store; a GPT-Live delegation passes
+   *  `voice-live` (spoken transcript in, speakable prose out). */
+  surface?: 'voice-live'
+  /** With `surface: 'voice-live'`: the recent spoken exchange, appended to the
+   *  model-bound note by the gateway (never persisted, never rendered). */
+  voiceContext?: string
   fromQueue?: boolean
   /** Runtime session id to submit into. Queue drains pass this so a
    *  backgrounded/source session cannot be replaced by the current foreground
diff --git a/apps/desktop/src/app/settings/constants.ts b/apps/desktop/src/app/settings/constants.ts
index daa1d05cc3..3a344aca97 100644
--- a/apps/desktop/src/app/settings/constants.ts
+++ b/apps/desktop/src/app/settings/constants.ts
@@ -251,6 +251,13 @@ export const ENUM_OPTIONS: Record<string, string[]> = {
   // Speech-to-text backends — kept in sync with the stt block in
   // hermes_cli/config.py (local/groq/openai/mistral/elevenlabs).
   'stt.provider': ['local', 'groq', 'openai', 'mistral', 'xai', 'elevenlabs'],
+  // How the desktop voice conversation is wired — tools/voice_live.py owns the
+  // gpt-live branch (one full-duplex voice model delegating to Hermes).
+  'voice.voice_chat_mode': ['chained', 'gpt-live'],
+  'voice.gpt_live.voice': [
+    'marin', 'cedar', 'quartz', 'ripple', 'vesper', 'willow', 'stone', 'gleam', 'meridian',
+    'bossa', 'tempo', 'beacon', 'delta', 'cinder'
+  ],
   // OpenAI TTS voices — the union across models (per the OpenAI TTS API
   // docs). Model-specific narrowing happens in enumOptionsFor():
   // tts-1 / tts-1-hd support 9 voices; gpt-4o-mini-tts supports all 13.
@@ -355,6 +362,7 @@ export const ENUM_OPTIONS: Record<string, string[]> = {
 // suggestions rather than a gate for these keys.
 export const FREE_INPUT_KEYS = new Set([
   'tts.edge.voice',
+  'voice.gpt_live.voice',
   'tts.openai.model',
   'tts.openai.voice',
   'tts.elevenlabs.voice_id',
@@ -437,7 +445,12 @@ export const FIELD_LABELS: Record<string, string> = defineFieldCopy({
   voice: {
     recordKey: 'Voice Shortcut',
     maxRecordingSeconds: 'Max Recording Length',
-    autoTts: 'Read Responses Aloud'
+    autoTts: 'Read Responses Aloud',
+    voiceChatMode: 'Voice Chat Mode',
+    gptLive: {
+      voice: 'GPT-Live Voice',
+      instructions: 'GPT-Live Persona'
+    }
   },
   stt: {
     enabled: 'Speech To Text',
@@ -598,7 +611,13 @@ export const FIELD_DESCRIPTIONS: Record<string, string> = defineFieldCopy({
     enabled: 'Summarize older context when conversations get large.'
   },
   voice: {
-    autoTts: 'Automatically speak assistant responses.'
+    autoTts: 'Automatically speak assistant responses.',
+    voiceChatMode:
+      'chained: speech-to-text → Hermes → text-to-speech with the providers below. gpt-live: one full-duplex OpenAI voice model (gpt-live-1) listens and talks, and hands every real request to Hermes — any model you have selected answers with the full toolset. Needs an OpenAI API key; the voice layer bills $0.05 per minute.',
+    gptLive: {
+      voice: 'Voice for GPT-Live mode. Custom voice IDs are accepted.',
+      instructions: 'Extra sentences for the live voice persona (tone, pace, language). Hermes keeps its own system prompt.'
+    }
   },
   tts: {
     xai: {
@@ -704,6 +723,9 @@ export const SECTIONS: DesktopConfigSection[] = [
     label: 'Voice',
     icon: Mic,
     keys: [
+      'voice.voice_chat_mode',
+      'voice.gpt_live.voice',
+      'voice.gpt_live.instructions',
       'tts.provider',
       'stt.enabled',
       'stt.echo_transcripts',
diff --git a/apps/desktop/src/i18n/en.ts b/apps/desktop/src/i18n/en.ts
index 304627c190..824430cf30 100644
--- a/apps/desktop/src/i18n/en.ts
+++ b/apps/desktop/src/i18n/en.ts
@@ -246,7 +246,11 @@ export const en: Translations = {
       transcriptionFailed: 'Voice transcription failed',
       transcriptionUnavailable: 'Voice transcription is not available yet.',
       tryRecordingAgain: 'Try recording again.',
-      unavailable: 'Voice unavailable'
+      unavailable: 'Voice unavailable',
+      liveEnded: 'Live voice session ended',
+      liveError: 'Live voice',
+      liveDelegationFailed: 'Could not hand the request to Hermes',
+      liveUnavailable: reason => `GPT-Live voice chat is not available: ${reason}. Using speech-to-text instead.`
     },
     native: {
       approvalTitle: 'Approval needed',
diff --git a/apps/desktop/src/i18n/types.ts b/apps/desktop/src/i18n/types.ts
index 861a429854..f010715dfd 100644
--- a/apps/desktop/src/i18n/types.ts
+++ b/apps/desktop/src/i18n/types.ts
@@ -285,6 +285,10 @@ export interface Translations {
       transcriptionUnavailable: string
       tryRecordingAgain: string
       unavailable: string
+      liveEnded: string
+      liveError: string
+      liveDelegationFailed: string
+      liveUnavailable: (reason: string) => string
     }
     // Native OS notification copy (titles + generic fallback bodies). Dynamic
     // bodies (the agent's reply, a command, an error) are passed through raw.
diff --git a/apps/desktop/src/i18n/zh.ts b/apps/desktop/src/i18n/zh.ts
index 895cb83c3b..4b85a69467 100644
--- a/apps/desktop/src/i18n/zh.ts
+++ b/apps/desktop/src/i18n/zh.ts
@@ -240,7 +240,11 @@ export const zh: Translations = {
       transcriptionFailed: '语音转写失败',
       transcriptionUnavailable: '语音转写暂不可用。',
       tryRecordingAgain: '请再录一次。',
-      unavailable: '语音不可用'
+      unavailable: '语音不可用',
+      liveEnded: '实时语音会话已结束',
+      liveError: '实时语音',
+      liveDelegationFailed: '无法将请求交给 Hermes',
+      liveUnavailable: reason => `GPT-Live 语音聊天不可用：${reason}。已改用语音转文字。`
     },
     native: {
       approvalTitle: '需要批准',
@@ -859,7 +863,12 @@ export const zh: Translations = {
       voice: {
         recordKey: '语音快捷键',
         maxRecordingSeconds: '最长录音时长',
-        autoTts: '朗读回复'
+        autoTts: '朗读回复',
+        voiceChatMode: '语音聊天模式',
+        gptLive: {
+          voice: 'GPT-Live 音色',
+          instructions: 'GPT-Live 人设'
+        }
       },
       stt: {
         enabled: '语音转文字',
@@ -1006,7 +1015,13 @@ export const zh: Translations = {
         enabled: '当对话变大时对较早的上下文进行摘要。'
       },
       voice: {
-        autoTts: '自动朗读助手回复。'
+        autoTts: '自动朗读助手回复。',
+        voiceChatMode:
+          'chained：语音转文字 → Hermes → 文字转语音，使用下方的提供商。gpt-live：一个全双工的 OpenAI 语音模型（gpt-live-1）负责听和说，并把每个实际请求交给 Hermes——由你选择的任意模型带着完整工具集作答。需要 OpenAI API 密钥；语音层按每分钟 $0.05 计费。',
+        gptLive: {
+          voice: 'GPT-Live 模式使用的音色，可填写自定义音色 ID。',
+          instructions: '附加到实时语音人设的句子（语气、语速、语言）。Hermes 保留自己的系统提示词。'
+        }
       },
       stt: {
         enabled: '启用本地或提供方支持的语音转写。',
diff --git a/apps/desktop/src/lib/voice-live.ts b/apps/desktop/src/lib/voice-live.ts
new file mode 100644
index 0000000000..aca8e4aae3
--- /dev/null
+++ b/apps/desktop/src/lib/voice-live.ts
@@ -0,0 +1,532 @@
+import { profileScoped } from '@/api/client'
+import { hermesApi } from '@/hermes'
+
+/**
+ * GPT-Live voice chat: the full-duplex voice frontend that DELEGATES to Hermes.
+ *
+ * `voice.voice_chat_mode: gpt-live` swaps the chained mic → STT → turn → TTS
+ * loop for one OpenAI voice model (`gpt-live-1`) that listens and speaks at
+ * the same time over WebRTC and has no tools of its own. Whenever the user
+ * asks for real work it emits `session.delegation.created`; the desktop turns
+ * that into an ordinary Hermes turn on the open session and streams the reply
+ * back with `session.commentary.append`, which the voice paraphrases aloud.
+ * Hermes keeps every capability — model choice, tools, memory, approvals.
+ *
+ * This module owns the transport only: session creation via the gateway
+ * (the OpenAI key never reaches the renderer), the RTCPeerConnection, the
+ * `oai-events` data channel, transcript accumulation and the command
+ * surface the conversation hook drives. Vendor contract:
+ * https://developers.openai.com/api/docs/guides/live-delegation
+ */
+
+export type VoiceChatMode = 'chained' | 'gpt-live'
+
+export interface VoiceLiveStatus {
+  mode: VoiceChatMode
+  available: boolean
+  reason: null | string
+  model: string
+  voice: string
+}
+
+export interface LiveHistoryMessage {
+  type: 'message'
+  role: 'assistant' | 'developer' | 'user'
+  content: Array<{ type: 'input_text' | 'output_text'; text: string }>
+}
+
+interface LiveServerEvent {
+  type: string
+  event_id?: string
+  client_event_id?: string
+  delta?: string
+  start_ms?: number
+  end_ms?: number
+  delegation?: { id: string; type: string; target: string }
+  error?: { type?: string; code?: null | string; message?: string; client_event_id?: string }
+  usage?: { seconds?: number }
+  reason?: string
+  session?: { id: string }
+}
+
+export interface LiveTranscriptFragment {
+  speaker: 'assistant' | 'user'
+  text: string
+  startMs: number
+  endMs: number
+}
+
+export interface VoiceLiveHandlers {
+  /** GPT-Live asked the backend (Hermes) for help. `context` is the recent
+   *  transcript window, newest last — the delegation itself carries no text. */
+  onDelegation: (delegationId: string, context: LiveTranscriptFragment[]) => void
+  /** Vendor-side error. `fatal` when the session is gone. */
+  onError: (message: string, fatal: boolean) => void
+  /** `session.closed` arrived (or the transport dropped without it). */
+  onClosed: (reason: string, usageSeconds: null | number) => void
+  /** Transcript deltas, for captions / live UI. */
+  onTranscript?: (fragment: LiveTranscriptFragment) => void
+  /** Assistant audio output level hint: the remote track is speaking. */
+  onSpeakingChange?: (speaking: boolean) => void
+}
+
+const CLOSE_TIMEOUT_MS = 15_000
+const ICE_GATHER_TIMEOUT_MS = 10_000
+// Vendor cap: 500 tokens per append. ~4 chars/token, keep headroom.
+const APPEND_CHAR_LIMIT = 1_400
+// How much conversation the backend receives per delegation.
+const CONTEXT_WINDOW_MS = 5 * 60_000
+const CONTEXT_MAX_FRAGMENTS = 80
+
+export async function fetchVoiceLiveStatus(): Promise<null | VoiceLiveStatus> {
+  try {
+    const response = await hermesApi<{ ok: boolean } & VoiceLiveStatus>({
+      ...profileScoped(),
+      path: '/api/audio/voice-live/status'
+    })
+
+    if (!response?.ok) {
+      return null
+    }
+
+    return {
+      available: Boolean(response.available),
+      mode: response.mode === 'gpt-live' ? 'gpt-live' : 'chained',
+      model: response.model,
+      reason: response.reason ?? null,
+      voice: response.voice
+    }
+  } catch {
+    // Older backend without the endpoint → chained.
+    return null
+  }
+}
+
+/** Split a reply into append-sized chunks on sentence boundaries. */
+export function chunkForCommentary(text: string, limit = APPEND_CHAR_LIMIT): string[] {
+  const clean = text.replace(/\s+/g, ' ').trim()
+
+  if (!clean) {
+    return []
+  }
+
+  if (clean.length <= limit) {
+    return [clean]
+  }
+
+  const chunks: string[] = []
+  let current = ''
+
+  for (const sentence of clean.split(/(?<=[.!?])\s+/)) {
+    if (sentence.length > limit) {
+      if (current) {
+        chunks.push(current)
+        current = ''
+      }
+
+      for (let index = 0; index < sentence.length; index += limit) {
+        chunks.push(sentence.slice(index, index + limit))
+      }
+
+      continue
+    }
+
+    const candidate = current ? `${current} ${sentence}` : sentence
+
+    if (candidate.length > limit) {
+      chunks.push(current)
+      current = sentence
+    } else {
+      current = candidate
+    }
+  }
+
+  if (current) {
+    chunks.push(current)
+  }
+
+  return chunks
+}
+
+/** Seed history for a new Live session from the chat transcript (text turns only). */
+export function toLiveHistory(
+  turns: Array<{ role: 'assistant' | 'user'; text: string }>,
+  maxMessages = 24,
+  maxChars = 6_000
+): LiveHistoryMessage[] {
+  const out: LiveHistoryMessage[] = []
+  let budget = maxChars
+
+  for (const turn of [...turns].reverse()) {
+    const text = turn.text.replace(/\s+/g, ' ').trim().slice(0, 1_200)
+
+    if (!text) {
+      continue
+    }
+
+    if (out.length >= maxMessages || budget - text.length < 0) {
+      break
+    }
+
+    budget -= text.length
+    out.unshift({
+      content: [{ text, type: turn.role === 'assistant' ? 'output_text' : 'input_text' }],
+      role: turn.role,
+      type: 'message'
+    })
+  }
+
+  return out
+}
+
+async function waitForIceGathering(connection: RTCPeerConnection): Promise<void> {
+  if (connection.iceGatheringState === 'complete') {
+    return
+  }
+
+  await new Promise<void>((resolve, reject) => {
+    const timeout = window.setTimeout(() => {
+      connection.removeEventListener('icegatheringstatechange', onState)
+      // Trickle is fine: the vendor answers with the candidates it has.
+      resolve()
+    }, ICE_GATHER_TIMEOUT_MS)
+
+    function onState() {
+      if (connection.iceGatheringState !== 'complete') {
+        return
+      }
+
+      window.clearTimeout(timeout)
+      connection.removeEventListener('icegatheringstatechange', onState)
+      resolve()
+    }
+
+    connection.addEventListener('icegatheringstatechange', onState)
+    connection.addEventListener('connectionstatechange', () => {
+      if (connection.connectionState === 'failed') {
+        window.clearTimeout(timeout)
+        reject(new Error('WebRTC connection failed'))
+      }
+    })
+  })
+}
+
+export class VoiceLiveSession {
+  readonly audio: HTMLAudioElement
+  private peer: null | RTCPeerConnection = null
+  private events: null | RTCDataChannel = null
+  private microphone: null | MediaStream = null
+  private closeTimer: null | number = null
+  private finalized = false
+  private started = false
+  private eventCounter = 0
+  private transcript: LiveTranscriptFragment[] = []
+  private speakingProbe: null | number = null
+  private analyser: null | AnalyserNode = null
+  private audioContext: null | AudioContext = null
+  private lastSpeaking = false
+  sessionId: null | string = null
+  /** The delegation currently being answered by Hermes; late results for an
+   *  older id are dropped by the conversation hook. */
+  activeDelegationId: null | string = null
+
+  constructor(private readonly handlers: VoiceLiveHandlers) {
+    this.audio = new Audio()
+    this.audio.autoplay = true
+  }
+
+  get connected(): boolean {
+    return this.started && this.events?.readyState === 'open'
+  }
+
+  private nextEventId(prefix: string): string {
+    this.eventCounter += 1
+
+    return `${prefix}_${this.eventCounter}`
+  }
+
+  private send(event: Record<string, unknown>): boolean {
+    if (!this.events || this.events.readyState !== 'open') {
+      return false
+    }
+
+    this.events.send(JSON.stringify(event))
+
+    return true
+  }
+
+  /** Recent conversation, oldest first, bounded by time and count. */
+  contextWindow(): LiveTranscriptFragment[] {
+    const last = this.transcript.at(-1)
+
+    if (!last) {
+      return []
+    }
+
+    const floor = last.endMs - CONTEXT_WINDOW_MS
+
+    return this.transcript.filter(fragment => fragment.endMs >= floor).slice(-CONTEXT_MAX_FRAGMENTS)
+  }
+
+  async start(history: LiveHistoryMessage[]): Promise<void> {
+    if (this.peer) {
+      throw new Error('GPT-Live session already started')
+    }
+
+    const connection = new RTCPeerConnection()
+    this.peer = connection
+
+    connection.addEventListener('track', event => {
+      const stream = new MediaStream([event.track])
+      this.audio.srcObject = stream
+      void this.audio.play().catch(() => undefined)
+      this.armSpeakingProbe(stream)
+    })
+    connection.addEventListener('connectionstatechange', () => {
+      if (connection.connectionState === 'failed' || connection.connectionState === 'disconnected') {
+        this.finish('connection_lost', null)
+      }
+    })
+
+    this.microphone = await navigator.mediaDevices.getUserMedia({
+      audio: { autoGainControl: true, echoCancellation: true, noiseSuppression: true }
+    })
+
+    for (const track of this.microphone.getAudioTracks()) {
+      connection.addTrack(track, this.microphone)
+    }
+
+    // Register the data channel before the offer so its m-line is negotiated.
+    const events = connection.createDataChannel('oai-events')
+    this.events = events
+    events.addEventListener('message', ({ data }) => this.handleEvent(String(data)))
+    events.addEventListener('close', () => {
+      if (!this.finalized) {
+        this.finish('connection_lost', null)
+      }
+    })
+
+    const offer = await connection.createOffer()
+    await connection.setLocalDescription(offer)
+    await waitForIceGathering(connection)
+
+    const sdp = connection.localDescription?.sdp
+
+    if (!sdp) {
+      throw new Error('Missing local SDP offer')
+    }
+
+    const response = await hermesApi<{
+      ok: boolean
+      session?: { id: string }
+      transport?: { sdp: string; type: string }
+    }>({
+      ...profileScoped(),
+      body: { history, sdp },
+      method: 'POST',
+      path: '/api/audio/voice-live/session',
+      timeoutMs: 45_000
+    })
+
+    if (!response?.ok || !response.transport?.sdp) {
+      throw new Error('GPT-Live session creation failed')
+    }
+
+    this.sessionId = response.session?.id ?? null
+    await connection.setRemoteDescription({ sdp: response.transport.sdp, type: 'answer' })
+  }
+
+  private armSpeakingProbe(stream: MediaStream): void {
+    try {
+      const context = new AudioContext()
+      const source = context.createMediaStreamSource(stream)
+      const analyser = context.createAnalyser()
+      analyser.fftSize = 512
+      source.connect(analyser)
+      this.audioContext = context
+      this.analyser = analyser
+      const buffer = new Uint8Array(analyser.frequencyBinCount)
+      let quietFrames = 0
+
+      this.speakingProbe = window.setInterval(() => {
+        analyser.getByteTimeDomainData(buffer)
+        let peak = 0
+
+        for (const sample of buffer) {
+          peak = Math.max(peak, Math.abs(sample - 128))
+        }
+
+        const loud = peak > 6
+        quietFrames = loud ? 0 : quietFrames + 1
+        const speaking = loud || quietFrames < 4
+
+        if (speaking !== this.lastSpeaking) {
+          this.lastSpeaking = speaking
+          this.handlers.onSpeakingChange?.(speaking)
+        }
+      }, 100)
+    } catch {
+      // No analyser → no speaking indicator; the conversation still works.
+    }
+  }
+
+  private handleEvent(raw: string): void {
+    let event: LiveServerEvent
+
+    try {
+      event = JSON.parse(raw) as LiveServerEvent
+    } catch {
+      return
+    }
+
+    switch (event.type) {
+      case 'session.started':
+        this.started = true
+        this.sessionId = event.session?.id ?? this.sessionId
+
+        return
+
+      case 'session.input_transcript.delta':
+      case 'session.output_transcript.delta': {
+        const fragment: LiveTranscriptFragment = {
+          endMs: event.end_ms ?? 0,
+          speaker: event.type === 'session.input_transcript.delta' ? 'user' : 'assistant',
+          startMs: event.start_ms ?? 0,
+          text: event.delta ?? ''
+        }
+
+        this.transcript.push(fragment)
+
+        if (this.transcript.length > 2_000) {
+          this.transcript.splice(0, this.transcript.length - 1_500)
+        }
+
+        this.handlers.onTranscript?.(fragment)
+
+        return
+      }
+
+      case 'session.delegation.created': {
+        const id = event.delegation?.id
+
+        if (id) {
+          this.activeDelegationId = id
+          this.handlers.onDelegation(id, this.contextWindow())
+        }
+
+        return
+      }
+
+      case 'error': {
+        const code = event.error?.code ?? ''
+
+        // Late appends after our own close are expected noise.
+        if (code === 'context_injection_incomplete') {
+          return
+        }
+
+        this.handlers.onError(event.error?.message ?? 'GPT-Live error', false)
+
+        return
+      }
+
+      case 'session.closed':
+        this.finish(event.reason ?? 'closed', event.usage?.seconds ?? null)
+
+        return
+
+      default:
+        return
+    }
+  }
+
+  /** Quiet progress for the live model ("Hermes is running the tests…"). */
+  think(delegationId: null | string, content: string): void {
+    const text = content.replace(/\s+/g, ' ').trim().slice(0, APPEND_CHAR_LIMIT)
+
+    if (text) {
+      this.send({
+        content: text,
+        delegation_id: delegationId,
+        event_id: this.nextEventId('think'),
+        type: 'session.thinking.append'
+      })
+    }
+  }
+
+  /** A result the voice should say aloud (paraphrased). */
+  speak(delegationId: null | string, content: string): void {
+    for (const chunk of chunkForCommentary(content)) {
+      this.send({
+        content: chunk,
+        delegation_id: delegationId,
+        event_id: this.nextEventId('say'),
+        type: 'session.commentary.append'
+      })
+    }
+  }
+
+  /** Steer the live persona mid-conversation (session-wide). */
+  instruct(content: string): void {
+    const text = content.trim().slice(0, APPEND_CHAR_LIMIT)
+
+    if (text) {
+      this.send({
+        content: text,
+        delegation_id: null,
+        event_id: this.nextEventId('instr'),
+        type: 'session.instructions.append'
+      })
+    }
+  }
+
+  setMuted(muted: boolean): void {
+    for (const track of this.microphone?.getAudioTracks() ?? []) {
+      track.enabled = !muted
+    }
+
+    this.send({ event_id: this.nextEventId(muted ? 'mute' : 'unmute'), type: muted ? 'session.input_audio.mute' : 'session.input_audio.unmute' })
+  }
+
+  /** Graceful close: ask for `session.closed`, tear down after it (or a timeout). */
+  close(): void {
+    if (this.finalized) {
+      return
+    }
+
+    if (!this.send({ type: 'session.close' })) {
+      this.finish('close_requested', null)
+
+      return
+    }
+
+    this.closeTimer = window.setTimeout(() => this.finish('close_requested', null), CLOSE_TIMEOUT_MS)
+  }
+
+  private finish(reason: string, usageSeconds: null | number): void {
+    if (this.finalized) {
+      return
+    }
+
+    this.finalized = true
+
+    if (this.closeTimer) {
+      window.clearTimeout(this.closeTimer)
+      this.closeTimer = null
+    }
+
+    if (this.speakingProbe) {
+      window.clearInterval(this.speakingProbe)
+      this.speakingProbe = null
+    }
+
+    this.analyser?.disconnect()
+    void this.audioContext?.close().catch(() => undefined)
+    this.microphone?.getTracks().forEach(track => track.stop())
+    this.events?.close()
+    this.peer?.close()
+    this.audio.srcObject = null
+    this.audio.pause()
+    this.handlers.onClosed(reason, usageSeconds)
+  }
+}
diff --git a/apps/desktop/src/store/voice-live.ts b/apps/desktop/src/store/voice-live.ts
new file mode 100644
index 0000000000..fd6bbedf92
--- /dev/null
+++ b/apps/desktop/src/store/voice-live.ts
@@ -0,0 +1,36 @@
+import { atom } from 'nanostores'
+
+import { fetchVoiceLiveStatus, type VoiceLiveStatus } from '@/lib/voice-live'
+
+/**
+ * `voice.voice_chat_mode` as the backend resolves it, plus whether GPT-Live can
+ * actually start (an OpenAI key resolves on the gateway host). The composer
+ * mounts the chained or the live conversation engine from this; refreshed with
+ * the config snapshot so a Settings change applies to the next conversation.
+ */
+export const $voiceLiveStatus = atom<null | VoiceLiveStatus>(null)
+
+let inflight: null | Promise<null | VoiceLiveStatus> = null
+
+export async function refreshVoiceLiveStatus(): Promise<null | VoiceLiveStatus> {
+  if (inflight) {
+    return inflight
+  }
+
+  inflight = fetchVoiceLiveStatus()
+    .then(status => {
+      $voiceLiveStatus.set(status)
+
+      return status
+    })
+    .finally(() => {
+      inflight = null
+    })
+
+  return inflight
+}
+
+/** Selected mode. `chained` until the backend answers, or when the backend predates the mode. */
+export function selectedVoiceChatMode(status: null | VoiceLiveStatus = $voiceLiveStatus.get()): 'chained' | 'gpt-live' {
+  return status?.mode === 'gpt-live' ? 'gpt-live' : 'chained'
+}
diff --git a/hermes_cli/config_defaults.py b/hermes_cli/config_defaults.py
index 6111e98eb2..4a6fbb0fd4 100644
--- a/hermes_cli/config_defaults.py
+++ b/hermes_cli/config_defaults.py
@@ -1116,6 +1116,19 @@ DEFAULT_CONFIG = {
     },
 
     "voice": {
+        # How the Desktop voice conversation is wired:
+        #   chained  — STT → Hermes turn → TTS (the stt.* / tts.* providers below)
+        #   gpt-live — one full-duplex voice model (OpenAI GPT-Live) owns the mic and speaker and
+        #              DELEGATES every real request to Hermes (any model / provider you have
+        #              selected); needs an OpenAI API key. $0.05/min voice layer billing.
+        "voice_chat_mode": "chained",
+        "gpt_live": {
+            "model": "gpt-live-1",
+            "voice": "marin",  # marin | quartz | ripple | vesper | willow | stone | gleam | meridian | ...
+            # Extra sentences appended to the live model's conversation persona (tone, pacing, language).
+            "instructions": "",
+            # optional "api_key" / "base_url" keys override the OpenAI audio credentials for this mode only
+        },
         "record_key": "ctrl+b",
         "submit_mode": "direct",  # TUI: direct submits immediately; draft = editable transcript
         "max_recording_seconds": 120,
diff --git a/hermes_cli/web_models.py b/hermes_cli/web_models.py
index 3f4faf5ebd..8aeacb378b 100644
--- a/hermes_cli/web_models.py
+++ b/hermes_cli/web_models.py
@@ -217,6 +217,12 @@ class DebugShareRequest(BaseModel):
 class TTSSpeakRequest(BaseModel):
     text: str
 
+class VoiceLiveSessionRequest(BaseModel):
+    """POST /api/audio/voice-live/session: the renderer's WebRTC SDP offer plus optional prior
+    text turns (``{"type":"message","role":..,"content":[..]}``) to seed the live voice model."""
+    sdp: str
+    history: Optional[List[Dict[str, Any]]] = None
+
 class TTSLeaseRequest(BaseModel):
     """POST /api/audio/tts-lease: ``lease`` names the toggle/surface holding the lease
     (``desktop:read-aloud``, ``desktop:conversation``); ``active`` True acquires + warms, False releases."""
diff --git a/hermes_cli/web_routers/audio.py b/hermes_cli/web_routers/audio.py
index ec49c90467..d70b194bbc 100644
--- a/hermes_cli/web_routers/audio.py
+++ b/hermes_cli/web_routers/audio.py
@@ -22,7 +22,7 @@ from hermes_cli.web_deps import late
 from hermes_cli.web_server_chat import _ws_auth_ok, _ws_request_is_allowed
 from hermes_cli.web_server_gateway import _split_text_for_speak_stream
 from fastapi import HTTPException, WebSocket, WebSocketDisconnect
-from hermes_cli.web_models import AudioTranscriptionRequest, TTSSpeakRequest, TTSLeaseRequest
+from hermes_cli.web_models import AudioTranscriptionRequest, TTSSpeakRequest, TTSLeaseRequest, VoiceLiveSessionRequest
 from typing import Any, Dict, Optional
 
 _log = logging.getLogger("hermes_cli.web_server")
@@ -164,6 +164,39 @@ async def get_client_voice_config(profile: Optional[str] = None):
     return {"ok": True, **result}
 
 
+@router.get("/api/audio/voice-live/status")
+async def get_voice_live_status(profile: Optional[str] = None):
+    """Which voice chat mode the profile selected (``chained`` | ``gpt-live``) and whether GPT-Live
+    can start. Non-secret: the desktop decides which conversation engine to mount from this."""
+    from tools.voice_live import resolve_gpt_live_status
+    with http_failure("GPT-Live status resolution failed", 500, "GPT-Live status failed"):
+        result = await _run_config_scoped(profile, resolve_gpt_live_status)
+    return {"ok": True, **result}
+
+
+@router.post("/api/audio/voice-live/session")
+async def create_voice_live_session(payload: VoiceLiveSessionRequest, profile: Optional[str] = None):
+    """Exchange the renderer's WebRTC SDP offer for a GPT-Live session answer.
+
+    The project API key stays on this host; the renderer only receives the session id and the
+    SDP answer. Client delegation is fixed at creation: every ``session.delegation.created`` the
+    renderer receives becomes a Hermes turn on the session it belongs to.
+    """
+    from tools.voice_live import create_webrtc_session
+    # Validate emptiness only: the vendor's SDP parser needs the offer byte-exact, including the
+    # trailing CRLF (a stripped offer answers 400 "failed to unmarshal SDP: EOF").
+    sdp = payload.sdp or ""
+    if not sdp.strip():
+        raise HTTPException(status_code=400, detail="An SDP offer is required")
+    try:
+        result = await _run_config_scoped(profile, lambda: create_webrtc_session(sdp, payload.history))
+    except ValueError as exc:
+        raise HTTPException(status_code=503, detail=str(exc))
+    except RuntimeError as exc:
+        raise HTTPException(status_code=502, detail=str(exc))
+    return {"ok": True, **result}
+
+
 def _elevenlabs_voice_label(voice: Dict[str, Any]) -> str:
     name = str(voice.get("name") or voice.get("voice_id") or "Voice").strip()
     category = str(voice.get("category") or "").strip()
diff --git a/tests/tui_gateway/test_voice_live_delegation.py b/tests/tui_gateway/test_voice_live_delegation.py
new file mode 100644
index 0000000000..34ba98f14b
--- /dev/null
+++ b/tests/tui_gateway/test_voice_live_delegation.py
@@ -0,0 +1,123 @@
+"""GPT-Live voice chat mode: the full-duplex voice frontend that delegates to Hermes.
+
+The live voice model owns the microphone and speaker and has no tools; every real
+request is delegated to Hermes as a normal turn on the open session. Two contracts
+matter and are pinned here:
+
+* the gateway never hands the OpenAI key to the renderer — ``POST /v1/live/sessions``
+  is performed server-side from the renderer's SDP offer, with the session pinned to
+  client delegation so Hermes (any model) is the backend;
+* a turn submitted from the live voice surface carries the spoken-delegation note on
+  the MODEL INPUT only (the byte-stable system prompt is untouched), exactly like the
+  HUD note it sits beside.
+"""
+
+import json
+import threading
+import types
+
+import pytest
+
+from tools import voice_live
+from tui_gateway import server
+
+
+def _session(**extra):
+    return {
+        "agent": types.SimpleNamespace(valid_tool_names=set()),
+        "session_key": "session-key",
+        "history": [],
+        "history_lock": threading.Lock(),
+        "history_version": 0,
+        "running": True,
+        "transport": None,
+        "attached_images": [],
+        **extra,
+    }
+
+
+class TestSessionCreation:
+    def test_client_delegation_and_key_stay_server_side(self, monkeypatch):
+        """Whatever the renderer sends, the vendor request pins ``delegation.type == client``
+        (Hermes is the backend) and authenticates with the resolved key; the client only ever
+        sees the vendor answer."""
+        captured = {}
+
+        class _Resp:
+            def __init__(self, body):
+                self._body = body
+
+            def read(self):
+                return self._body
+
+            def __enter__(self):
+                return self
+
+            def __exit__(self, *exc):
+                return False
+
+        def fake_urlopen(req, timeout=0):
+            captured["url"] = req.full_url
+            captured["auth"] = req.get_header("Authorization")
+            captured["body"] = json.loads(req.data)
+            return _Resp(json.dumps({"session": {"id": "live_x"}, "transport": {"type": "webrtc", "sdp": "answer"}}).encode())
+
+        monkeypatch.setattr(voice_live.urllib.request, "urlopen", fake_urlopen)
+        monkeypatch.setattr(voice_live, "_live_section", lambda voice=None: {"voice": "willow", "instructions": "Speak Spanish."})
+        monkeypatch.setattr(voice_live, "_resolve_credentials", lambda live: ("sk-test", "https://api.example/v1"))
+
+        result = voice_live.create_webrtc_session("v=0 offer", history=[{"type": "message", "role": "user", "content": []}])
+
+        assert result["transport"]["sdp"] == "answer"
+        assert captured["url"] == "https://api.example/v1/live/sessions"
+        assert captured["auth"] == "Bearer sk-test"
+        session = captured["body"]["session"]
+        assert session["delegation"] == {"type": "client"}
+        assert session["audio"]["output"]["voice"] == "willow"
+        assert session["instructions"].endswith("Speak Spanish.")
+        assert session["input"][0]["role"] == "user"
+        assert captured["body"]["transport"] == {"type": "webrtc", "sdp": "v=0 offer"}
+        assert "sk-test" not in json.dumps(result)
+
+    def test_missing_key_refuses_before_any_network(self, monkeypatch):
+        monkeypatch.setattr(voice_live, "_resolve_credentials", lambda live: ("", voice_live.DEFAULT_LIVE_BASE_URL))
+        monkeypatch.setattr(voice_live.urllib.request, "urlopen", lambda *a, **k: pytest.fail("must not call the vendor"))
+
+        with pytest.raises(ValueError):
+            voice_live.create_webrtc_session("v=0 offer")
+        assert voice_live.resolve_gpt_live_status()["available"] is False
+
+
+class TestVoiceLiveTurnNote:
+    @pytest.fixture
+    def busy_session(self):
+        session = _session()
+        server._sessions["sid"] = session
+        yield session
+        server._sessions.pop("sid", None)
+
+    def test_live_surface_recorded_and_noted_with_spoken_context(self, busy_session):
+        """The persisted row is the user's words; the transcript window reaches the model only."""
+        server._methods["prompt.submit"](
+            "r1", {"session_id": "sid", "text": "what's the weather", "queued": True, "surface": "voice-live",
+                   "voice_context": "Voice assistant: Hi\nUser: what's the weather"})
+
+        assert busy_session["client_surface"] == "voice-live"
+        note = server._hud_surface_note(busy_session)
+        assert note.startswith(voice_live.VOICE_LIVE_TURN_NOTE)
+        assert "spoken" in note and "no markdown" in note
+        assert "User: what's the weather" in note
+
+    def test_voice_context_ignored_off_the_live_surface(self, busy_session):
+        server._methods["prompt.submit"](
+            "r1", {"session_id": "sid", "text": "x", "queued": True, "voice_context": "User: smuggled"})
+
+        assert busy_session["voice_live_context"] == ""
+        assert server._hud_surface_note(busy_session) == ""
+
+    def test_plain_window_submit_clears_the_live_surface(self, busy_session):
+        server._methods["prompt.submit"]("r1", {"session_id": "sid", "text": "x", "queued": True, "surface": "voice-live"})
+        server._methods["prompt.submit"]("r2", {"session_id": "sid", "text": "y", "queued": True})
+
+        assert busy_session["client_surface"] == ""
+        assert server._hud_surface_note(busy_session) == ""
diff --git a/tools/voice_live.py b/tools/voice_live.py
new file mode 100644
index 0000000000..b9ab6f7863
--- /dev/null
+++ b/tools/voice_live.py
@@ -0,0 +1,186 @@
+"""GPT-Live voice chat mode: the full-duplex voice frontend that delegates to Hermes.
+
+``voice.voice_chat_mode: gpt-live`` replaces the chained STT → turn → TTS loop with ONE
+full-duplex voice model (OpenAI ``gpt-live-1``) that owns the microphone and the speaker and
+delegates every real request to Hermes as its *client-delegation* backend. Hermes stays the
+agent: whatever model/provider the session has selected answers, with the full toolset.
+
+Division of labour (the Live API has no tools of its own in client mode):
+
+* the desktop renderer holds the WebRTC media session (mic in, speech out) and the data channel;
+* this module resolves WHICH credentials/voice/persona to use and performs the one server-side
+  step the API requires — exchanging the browser's SDP offer for an answer with the project key
+  (``POST /v1/live/sessions``), so the key never reaches the client;
+* the renderer turns each ``session.delegation.created`` into a normal ``prompt.submit`` on the
+  active session (surface ``voice-live``) and streams the reply back as
+  ``session.commentary.append`` — Hermes' answer is what the voice speaks.
+
+Vendor contract: https://developers.openai.com/api/docs/guides/live (+ live-delegation,
+voice-webrtc). Billing is $0.05/min of session time on the OpenAI key, separate from the
+Hermes turn.
+"""
+
+from __future__ import annotations
+
+import json
+import logging
+import urllib.error
+import urllib.request
+from typing import Any, Dict, Optional
+
+logger = logging.getLogger(__name__)
+
+GPT_LIVE_MODE = "gpt-live"
+CHAINED_MODE = "chained"
+DEFAULT_LIVE_MODEL = "gpt-live-1"
+DEFAULT_LIVE_VOICE = "marin"
+DEFAULT_LIVE_BASE_URL = "https://api.openai.com/v1"
+# Voices the vendor lists for gpt-live-1 (live-conversations guide) plus the realtime defaults it
+# accepts; free text stays allowed for custom voices.
+GPT_LIVE_VOICES = (
+    "marin", "cedar", "quartz", "ripple", "vesper", "willow", "stone", "gleam", "meridian",
+    "bossa", "tempo", "beacon", "delta", "cinder",
+)
+
+# Persona for the voice layer. Short on purpose: the live model has a small context window and
+# the vendor guide asks for role + style + a labelled delegation policy, nothing more. The
+# backend (Hermes) carries the real instructions, tools and memory.
+LIVE_PERSONA = (
+    "You are Hermes, a calm and friendly voice assistant. Speak naturally at an unhurried pace. "
+    "Be clear and direct, not overly cheerful. If the user is frustrated, acknowledge it briefly "
+    "and focus on the next helpful step.\n\n"
+    "Backchannel policy: Use moderate backchannels. Acknowledge naturally without competing with "
+    "the main response.\n\n"
+    "Interruption policy: Stop speaking when the user interrupts. Listen to what they say.\n\n"
+    "Delegation policy:\n"
+    "Backend tools:\n"
+    "- Hermes agent: a full AI agent with tools — it can run commands, read and edit files, "
+    "browse the web, search, remember things across sessions, schedule tasks, and reason "
+    "carefully about anything. It is the one who actually does work and knows facts.\n\n"
+    "Delegate to the backend when:\n"
+    "- The user asks a question that needs facts, current information, or careful reasoning.\n"
+    "- The user asks you to do, check, find, make, fix, run or remember anything.\n"
+    "- A correction changes work already requested.\n\n"
+    "Do not delegate to the backend when:\n"
+    "- The user greets you, makes small talk, or asks you to repeat a result already provided.\n"
+    "- You need a brief clarification to understand the request.\n\n"
+    "Delegate before giving an answer that depends on backend work. Do not guess the result "
+    "while waiting; say briefly that you are checking, then wait for the result."
+)
+
+# Per-turn note prepended to the MODEL INPUT (never the byte-stable system prompt) when a turn
+# arrives from the live voice layer. Same seam as the HUD note.
+VOICE_LIVE_TURN_NOTE = (
+    "[Note: this message is a delegation from a live spoken conversation. The text is a voice "
+    "transcript (it may contain mis-hearings, hesitations and later corrections; use the latest "
+    "intent). Your reply will be spoken aloud by a voice model that paraphrases it: answer in plain "
+    "conversational sentences, keep it short (a few sentences unless the user asked for detail), no "
+    "markdown, no lists, no code blocks, no URLs read out character by character. Do the work with "
+    "your tools as usual; only the final facts need to be spoken. Do not claim an action succeeded "
+    "before it actually did.]"
+)
+
+
+def voice_live_turn_note(context: str = "") -> str:
+    """The per-turn note plus, when the client sent one, the recent spoken exchange the delegation
+    refers to (the user's last words alone are often "yes" or "Thursday, not Friday")."""
+    context = context.strip()
+    if not context:
+        return VOICE_LIVE_TURN_NOTE
+    return f"{VOICE_LIVE_TURN_NOTE}\n[Recent spoken conversation, newest last:\n{context}]"
+
+
+def _voice_section() -> Dict[str, Any]:
+    try:
+        from hermes_cli.config import load_config
+        voice = load_config().get("voice")
+    except Exception:
+        return {}
+    return voice if isinstance(voice, dict) else {}
+
+
+def _live_section(voice: Optional[Dict[str, Any]] = None) -> Dict[str, Any]:
+    section = (voice if voice is not None else _voice_section()).get("gpt_live")
+    return section if isinstance(section, dict) else {}
+
+
+def voice_chat_mode(voice: Optional[Dict[str, Any]] = None) -> str:
+    """``chained`` (default) or ``gpt-live``. Accepts the underscore spelling too."""
+    raw = (voice if voice is not None else _voice_section()).get("voice_chat_mode")
+    mode = str(raw or CHAINED_MODE).strip().lower().replace("_", "-")
+    return GPT_LIVE_MODE if mode in {GPT_LIVE_MODE, "gptlive", "live"} else CHAINED_MODE
+
+
+def _resolve_credentials(live: Dict[str, Any]) -> tuple[str, str]:
+    """``(api_key, base_url)`` — ``voice.gpt_live.api_key`` first, else the same OpenAI audio
+    chain the STT/TTS providers use (``VOICE_TOOLS_OPENAI_KEY`` → ``OPENAI_API_KEY`` → pool).
+
+    The Nous-managed audio proxy does not carry ``/live/sessions``; this mode is direct-key only.
+    """
+    from tools.tool_backend_helpers import resolve_openai_audio_api_key
+    api_key = str(live.get("api_key") or "").strip() or resolve_openai_audio_api_key()
+    base_url = str(live.get("base_url") or DEFAULT_LIVE_BASE_URL).strip().rstrip("/")
+    return api_key, base_url
+
+
+def live_instructions(live: Optional[Dict[str, Any]] = None) -> str:
+    extra = str((live if live is not None else _live_section()).get("instructions") or "").strip()
+    return f"{LIVE_PERSONA}\n\n{extra}" if extra else LIVE_PERSONA
+
+
+def resolve_gpt_live_status() -> Dict[str, Any]:
+    """Non-secret readiness verdict for the client: which mode is selected and whether GPT-Live
+    can start (a key resolves). Never returns the key."""
+    voice = _voice_section()
+    mode = voice_chat_mode(voice)
+    live = _live_section(voice)
+    api_key, _base = _resolve_credentials(live)
+    return {
+        "mode": mode,
+        "available": bool(api_key),
+        "reason": None if api_key else "no OpenAI API key (set OPENAI_API_KEY or voice.gpt_live.api_key)",
+        "model": str(live.get("model") or DEFAULT_LIVE_MODEL),
+        "voice": str(live.get("voice") or DEFAULT_LIVE_VOICE),
+    }
+
+
+def build_session_config(history: Optional[list] = None) -> Dict[str, Any]:
+    """The ``session`` object for ``POST /v1/live/sessions`` (client delegation, WebRTC — the
+    transport negotiates the audio format, so none is set)."""
+    live = _live_section()
+    config: Dict[str, Any] = {
+        "model": str(live.get("model") or DEFAULT_LIVE_MODEL),
+        "instructions": live_instructions(live),
+        "audio": {"output": {"voice": str(live.get("voice") or DEFAULT_LIVE_VOICE)}},
+        "delegation": {"type": "client"},
+    }
+    if history:
+        config["input"] = history
+    return config
+
+
+def create_webrtc_session(sdp_offer: str, history: Optional[list] = None) -> Dict[str, Any]:
+    """Exchange the renderer's SDP offer for the Live session answer.
+
+    Returns the vendor response ``{"session": {"id": ...}, "transport": {"type": "webrtc",
+    "sdp": ...}}``. Raises ``ValueError`` for a missing key and ``RuntimeError`` (with the vendor
+    status/detail) for a rejected request.
+    """
+    live = _live_section()
+    api_key, base_url = _resolve_credentials(live)
+    if not api_key:
+        raise ValueError("GPT-Live needs an OpenAI API key (OPENAI_API_KEY or voice.gpt_live.api_key)")
+    body = json.dumps({
+        "session": build_session_config(history),
+        "transport": {"type": "webrtc", "sdp": sdp_offer},
+    }).encode("utf-8")
+    req = urllib.request.Request(
+        f"{base_url}/live/sessions", data=body, method="POST",
+        headers={"Authorization": f"Bearer {api_key}", "Content-Type": "application/json"})
+    try:
+        with urllib.request.urlopen(req, timeout=30) as resp:
+            return json.loads(resp.read().decode("utf-8"))
+    except urllib.error.HTTPError as exc:
+        detail = exc.read().decode("utf-8", "replace")[:600]
+        logger.warning("GPT-Live session creation failed: %s %s", exc.code, detail)
+        raise RuntimeError(f"GPT-Live session creation failed ({exc.code}): {detail}") from exc
diff --git a/tui_gateway/methods_prompt.py b/tui_gateway/methods_prompt.py
index 0ff67007c2..bdb0a40b92 100644
--- a/tui_gateway/methods_prompt.py
+++ b/tui_gateway/methods_prompt.py
@@ -537,6 +537,10 @@ def _lock_in_submit_turn(
     return None, fields
 
 
+# Per-turn client surfaces that carry a model-bound note (session_notifications._surface_note).
+_CLIENT_SURFACES = frozenset({"hud", "voice-live"})
+
+
 @method("prompt.submit")
 def _(rid, params: dict) -> dict:
     from hermes_cli.input_sanitize import sanitize_user_prompt_text
@@ -575,8 +579,13 @@ def _(rid, params: dict) -> dict:
         # leaves the session untouched.  The reason travels as machine-readable data.
         reason = getattr(limit_message, "reason", None)
         return _err(rid, 4090, str(limit_message), {"reason": reason} if reason else None)
-    # Rewritten every submit: a session alternates app window / HUD; stale "hud" misinforms.
-    session["client_surface"] = "hud" if params.get("surface") == "hud" else ""
+    # Rewritten every submit: a session alternates app window / HUD / live voice; a stale value misinforms.
+    session["client_surface"] = params.get("surface") if params.get("surface") in _CLIENT_SURFACES else ""
+    # Live-voice delegations carry the recent spoken transcript for the MODEL INPUT only (the persisted
+    # user row stays the words the user said); anything else clears it.
+    voice_context = params.get("voice_context")
+    session["voice_live_context"] = (
+        voice_context[:6000] if session["client_surface"] == "voice-live" and isinstance(voice_context, str) else "")
     has_truncation = any(params.get(k) is not None for k in _TRUNCATION_PARAMS)
     if has_truncation and isinstance(text, str):
         # A rewind replays what the transcript shows: re-expand a skill invocation or
diff --git a/tui_gateway/session_notifications.py b/tui_gateway/session_notifications.py
index c6d6368f53..5d16abf471 100644
--- a/tui_gateway/session_notifications.py
+++ b/tui_gateway/session_notifications.py
@@ -662,11 +662,16 @@ def _start_notification_poller(sid: str, session: dict) -> threading.Event:
 
 
 def _hud_surface_note(session: dict) -> str:
-    """The HUD-mode note for this turn, or "" when it was not typed there."""
-    if session.get("client_surface") != "hud":
-        return ""
-    from agent.prompt_builder import hud_surface_note
-    return hud_surface_note(getattr(session.get("agent"), "valid_tool_names", None))
+    """The per-surface note for this turn ("" for the plain app window): HUD → the read-the-window-below
+    prior; voice-live → the spoken-delegation contract (transcript in, speakable prose out)."""
+    surface = session.get("client_surface")
+    if surface == "hud":
+        from agent.prompt_builder import hud_surface_note
+        return hud_surface_note(getattr(session.get("agent"), "valid_tool_names", None))
+    if surface == "voice-live":
+        from tools.voice_live import voice_live_turn_note
+        return voice_live_turn_note(session.get("voice_live_context") or "")
+    return ""
 
 
 def _prepend_note(run_message: Any, note: str) -> Any:
diff --git a/website/docs/user-guide/features/voice-mode.md b/website/docs/user-guide/features/voice-mode.md
index 349b8936d2..8750029e2b 100644
--- a/website/docs/user-guide/features/voice-mode.md
+++ b/website/docs/user-guide/features/voice-mode.md
@@ -193,6 +193,24 @@ voice:
 
 Client-direct wire support: OpenAI (incl. Nous-managed audio), Groq, Mistral, and DeepInfra via the OpenAI-compatible shapes, xAI Grok STT, and ElevenLabs STT + TTS. xAI configured through OAuth stays on the relay (the OAuth bearer refreshes server-side).
 
+### Desktop: GPT-Live voice chat mode (full duplex, delegates to Hermes)
+
+The chained loop above is one of two voice chat modes in the desktop app. The other replaces the whole STT → turn → TTS chain with **one full-duplex voice model**, OpenAI's `gpt-live-1`: it listens while it speaks, handles interruptions, backchannels and background noise itself, and has **no tools of its own**. Whenever you ask for real work it *delegates* to Hermes, which answers as usual — with whatever model and provider the session has selected, the full toolset, memory and approvals — and the voice paraphrases the answer aloud.
+
+```yaml
+voice:
+  voice_chat_mode: gpt-live     # chained (default) | gpt-live
+  gpt_live:
+    voice: marin                # marin, cedar, quartz, ripple, vesper, willow, stone, gleam, meridian, …
+    instructions: ""            # optional extra persona sentences (tone, pace, language)
+```
+
+Requirements: an OpenAI API key (`OPENAI_API_KEY`, `VOICE_TOOLS_OPENAI_KEY`, or `voice.gpt_live.api_key`). The voice layer is billed by OpenAI at **$0.05 per minute of session time** (idle time counts); the Hermes turn is billed on its own provider as always. The mode is also in Settings → Voice → *Voice Chat Mode*.
+
+How it works: pressing the voice button opens a WebRTC session from the desktop to GPT-Live; the desktop only ever receives a session id and an SDP answer — the key stays on the gateway host, which performs the session creation (`POST /api/audio/voice-live/session`). Each `session.delegation.created` becomes a normal turn on the open chat (the bubble shows what you said; the recent spoken exchange rides the model input as a per-turn note, never the system prompt, so the reply is speakable prose). Tool activity is fed to the voice as quiet context ("Hermes is working: terminal") so it can tell you what is happening if you ask; the final answer is streamed back sentence by sentence. Saying the stop phrase ends the conversation. If `gpt-live` is selected but no key resolves, the button falls back to the chained mode with a notice.
+
+Not supported in this mode: the Nous-managed audio proxy (direct key only), the CLI/TUI (`/voice` keeps the chained loop), and the `tts` tool (it keeps using `tts.provider`).
+
 ### Barge-in
 
 You can interrupt the agent at ANY point in its turn — the microphone stays live from the moment you finish speaking until the reply has fully played (full duplex):