diff --git a/tests/test_audio.py b/tests/test_audio.py
new file mode 100644
index 00000000..d2756a59
--- /dev/null
+++ b/tests/test_audio.py
@@ -0,0 +1,231 @@
+"""Unit tests for the STT/TTS audio helper (model-role resolution + backends).
+
+``transcribe`` / ``synthesize`` are exercised through the registry boundary
+with a mocked OpenAI-SDK client (mocking ``client.audio.*``), so the real
+helper code runs end-to-end without a network call.
+"""
+
+from __future__ import annotations
+
+from unittest.mock import MagicMock
+
+import pytest
+
+from turnstone.core import audio
+
+
+class _Cfg:
+    """Stand-in for ModelConfig — only the fields audio.py reads."""
+
+    def __init__(self, model: str, capabilities: dict | None = None) -> None:
+        self.model = model
+        self.capabilities = capabilities or {}
+
+
+class _FakeConfigStore:
+    def __init__(self, **values: str) -> None:
+        self._values = values
+
+    def get(self, key: str, default: str = "") -> str:
+        return self._values.get(key, default)
+
+
+class _FakeRegistry:
+    """Minimal registry exposing the surface audio.py uses."""
+
+    def __init__(self, alias: str, cfg: _Cfg, client: object) -> None:
+        self._alias = alias
+        self._cfg = cfg
+        self._client = client
+
+    def has_alias(self, alias: str) -> bool:
+        return alias == self._alias
+
+    def get_config(self, alias: str) -> _Cfg:
+        if alias != self._alias:
+            raise ValueError(alias)
+        return self._cfg
+
+    def resolve(self, alias: str | None = None):
+        if alias not in (None, self._alias):
+            raise ValueError(alias)
+        return self._client, self._cfg.model, self._cfg
+
+
+# ---------------------------------------------------------------------------
+# Capability gating
+# ---------------------------------------------------------------------------
+
+
+class TestModelSupportsRole:
+    def test_explicit_flag_wins(self):
+        assert audio.model_supports_role(_Cfg("anything", {"supports_transcription": True}), "stt")
+        # Explicit False overrides the would-be inference from the model name.
+        assert not audio.model_supports_role(
+            _Cfg("gpt-4o-mini-tts", {"supports_speech_synthesis": False}), "tts"
+        )
+
+    def test_infers_known_openai_audio_models(self):
+        assert audio.model_supports_role(_Cfg("gpt-4o-mini-transcribe"), "stt")
+        assert audio.model_supports_role(_Cfg("whisper-1"), "stt")
+        assert audio.model_supports_role(_Cfg("gpt-4o-mini-tts"), "tts")
+        assert audio.model_supports_role(_Cfg("tts-1"), "tts")
+
+    def test_chat_model_not_eligible(self):
+        assert not audio.model_supports_role(_Cfg("gpt-5"), "stt")
+        # Anthropic has no audio API — gated out of every audio role.
+        assert not audio.model_supports_role(_Cfg("claude-opus-4-8"), "tts")
+        assert not audio.model_supports_role(_Cfg("claude-opus-4-8"), "stt")
+
+    def test_unknown_role(self):
+        assert not audio.model_supports_role(_Cfg("whisper-1"), "vision_eval")
+
+    def test_hint_seed_lists_are_pinned(self):
+        # Mirrored verbatim in admin.js AUDIO_MODEL_HINTS — if these change,
+        # update the JS dropdown gate too (this pin makes the change deliberate).
+        assert audio._AUDIO_MODEL_HINTS == {
+            "stt": ("transcribe", "whisper", "-asr"),
+            "tts": ("tts-", "-tts"),
+        }
+
+
+# ---------------------------------------------------------------------------
+# Role resolution
+# ---------------------------------------------------------------------------
+
+
+class TestResolveRoleAlias:
+    def test_resolves_configured_capable_alias(self):
+        reg = _FakeRegistry("voice", _Cfg("gpt-4o-mini-transcribe"), MagicMock())
+        cs = _FakeConfigStore(**{"audio.stt_model_alias": "voice"})
+        assert audio.resolve_role_alias(config_store=cs, registry=reg, role="stt") == "voice"
+
+    def test_none_when_unset(self):
+        reg = _FakeRegistry("voice", _Cfg("gpt-4o-mini-transcribe"), MagicMock())
+        assert (
+            audio.resolve_role_alias(config_store=_FakeConfigStore(), registry=reg, role="stt")
+            is None
+        )
+
+    def test_none_when_alias_missing_from_registry(self):
+        reg = _FakeRegistry("voice", _Cfg("gpt-4o-mini-transcribe"), MagicMock())
+        cs = _FakeConfigStore(**{"audio.stt_model_alias": "ghost"})
+        assert audio.resolve_role_alias(config_store=cs, registry=reg, role="stt") is None
+
+    def test_none_when_alias_not_capability_eligible(self):
+        # Alias exists but its model can't do TTS -> gated out (Anthropic case).
+        reg = _FakeRegistry("brain", _Cfg("claude-opus-4-8"), MagicMock())
+        cs = _FakeConfigStore(**{"audio.tts_model_alias": "brain"})
+        assert audio.resolve_role_alias(config_store=cs, registry=reg, role="tts") is None
+
+    def test_none_when_no_registry_or_store(self):
+        assert audio.resolve_role_alias(config_store=None, registry=None, role="stt") is None
+
+
+# ---------------------------------------------------------------------------
+# transcribe / synthesize — boundary: mocked OpenAI-SDK client
+# ---------------------------------------------------------------------------
+
+
+class TestTranscribe:
+    def test_calls_audio_transcriptions_and_returns_text(self):
+        client = MagicMock()
+        client.audio.transcriptions.create.return_value = MagicMock(text="  hello world  ")
+        reg = _FakeRegistry("voice", _Cfg("gpt-4o-mini-transcribe"), client)
+        res = audio.transcribe(
+            registry=reg, alias="voice", data=b"RIFFfake", filename="speech.webm"
+        )
+        assert res.transcript == "hello world"
+        assert res.model_alias == "voice"
+        assert res.model == "gpt-4o-mini-transcribe"
+        kwargs = client.audio.transcriptions.create.call_args.kwargs
+        assert kwargs["model"] == "gpt-4o-mini-transcribe"
+        assert kwargs["file"] == ("speech.webm", b"RIFFfake")
+
+    def test_prompt_forwarded_when_set(self):
+        client = MagicMock()
+        client.audio.transcriptions.create.return_value = MagicMock(text="ok")
+        reg = _FakeRegistry("voice", _Cfg("whisper-1"), client)
+        audio.transcribe(
+            registry=reg, alias="voice", data=b"x", filename="a.wav", prompt="ACME jargon"
+        )
+        assert client.audio.transcriptions.create.call_args.kwargs["prompt"] == "ACME jargon"
+
+    def test_prompt_omitted_when_blank(self):
+        client = MagicMock()
+        client.audio.transcriptions.create.return_value = MagicMock(text="ok")
+        reg = _FakeRegistry("voice", _Cfg("whisper-1"), client)
+        audio.transcribe(registry=reg, alias="voice", data=b"x", filename="a.wav")
+        assert "prompt" not in client.audio.transcriptions.create.call_args.kwargs
+
+    def test_backend_failure_raises_backend_error(self):
+        client = MagicMock()
+        client.audio.transcriptions.create.side_effect = RuntimeError("boom")
+        reg = _FakeRegistry("voice", _Cfg("whisper-1"), client)
+        with pytest.raises(audio.AudioBackendError):
+            audio.transcribe(registry=reg, alias="voice", data=b"x", filename="a.wav")
+
+
+class TestSynthesize:
+    def test_calls_audio_speech_and_returns_bytes(self):
+        client = MagicMock()
+        speech = MagicMock()
+        speech.read.return_value = b"RIFF...wavbytes"
+        client.audio.speech.create.return_value = speech
+        reg = _FakeRegistry("voice", _Cfg("gpt-4o-mini-tts"), client)
+        res = audio.synthesize(registry=reg, alias="voice", text="hi", voice="nova")
+        assert res.audio_bytes == b"RIFF...wavbytes"
+        assert res.media_type == "audio/mpeg"
+        assert res.model_alias == "voice"
+        kwargs = client.audio.speech.create.call_args.kwargs
+        assert kwargs["voice"] == "nova"
+        assert kwargs["input"] == "hi"
+
+    def test_default_voice_when_empty(self):
+        client = MagicMock()
+        client.audio.speech.create.return_value = MagicMock(read=lambda: b"a")
+        reg = _FakeRegistry("voice", _Cfg("gpt-4o-mini-tts"), client)
+        audio.synthesize(registry=reg, alias="voice", text="hi", voice="")
+        assert client.audio.speech.create.call_args.kwargs["voice"] == "alloy"
+
+    def test_backend_failure_raises_backend_error(self):
+        client = MagicMock()
+        client.audio.speech.create.side_effect = RuntimeError("down")
+        reg = _FakeRegistry("voice", _Cfg("gpt-4o-mini-tts"), client)
+        with pytest.raises(audio.AudioBackendError):
+            audio.synthesize(registry=reg, alias="voice", text="hi", voice="nova")
+
+
+class TestOpenAIAudioModelsKnown:
+    """The current OpenAI STT/TTS lineup is registered in the static capability
+    table, so the admin 'suggested capabilities' recognizes them and they show
+    in the known-models list. (Role gating also works via name inference for
+    openai-compatible/local backends that aren't in the static table.)"""
+
+    def test_stt_models_flagged(self):
+        from turnstone.core.providers import lookup_model_capabilities
+
+        for m in (
+            "whisper-1",
+            "gpt-4o-transcribe",
+            "gpt-4o-mini-transcribe",
+            "gpt-4o-transcribe-diarize",  # prefix variant
+        ):
+            caps = lookup_model_capabilities("openai", m) or {}
+            assert caps.get("supports_transcription") is True, m
+            assert caps.get("supports_speech_synthesis") is False, m
+
+    def test_tts_models_flagged(self):
+        from turnstone.core.providers import lookup_model_capabilities
+
+        for m in ("tts-1", "tts-1-hd", "gpt-4o-mini-tts"):  # tts-1-hd is a prefix variant
+            caps = lookup_model_capabilities("openai", m) or {}
+            assert caps.get("supports_speech_synthesis") is True, m
+            assert caps.get("supports_transcription") is False, m
+
+    def test_chat_model_has_no_audio_flags(self):
+        from turnstone.core.providers import lookup_model_capabilities
+
+        caps = lookup_model_capabilities("openai", "gpt-5") or {}
+        assert not caps.get("supports_transcription")
+        assert not caps.get("supports_speech_synthesis")
diff --git a/tests/test_openapi.py b/tests/test_openapi.py
index 03ff500e..3004a660 100644
--- a/tests/test_openapi.py
+++ b/tests/test_openapi.py
@@ -41,12 +41,29 @@ class TestServerSpec:
             "/v1/api/command",
             "/v1/api/events/global",
             "/v1/api/workstreams/new",
+            "/v1/api/workstreams/{ws_id}/speech-to-text",
+            "/v1/api/tts",
             "/v1/api/auth/login",
             "/v1/api/auth/logout",
             "/health",
         }
         assert expected.issubset(paths), f"Missing: {expected - paths}"
 
+    def test_voice_endpoints_documented(self):
+        from turnstone.api.server_spec import build_server_spec
+
+        spec = build_server_spec()
+        stt = spec["paths"]["/v1/api/workstreams/{ws_id}/speech-to-text"]["post"]
+        tts = spec["paths"]["/v1/api/tts"]["post"]
+        assert "responses" in stt
+        assert "requestBody" in tts
+        assert "application/json" in tts["requestBody"]["content"]
+        schemas = spec["components"]["schemas"]
+        assert "capabilities" in schemas["AvailableModelInfo"]["properties"]
+        models_props = schemas["ListAvailableModelsResponse"]["properties"]
+        assert "stt_default_alias" in models_props
+        assert "tts_default_alias" in models_props
+
     def test_workstream_history_has_limit_query_param(self):
         """Mirror of the coord-side history limit param test — server now
         exposes the same endpoint via the lifted factory."""
diff --git a/tests/test_server_attachments_endpoints.py b/tests/test_server_attachments_endpoints.py
index ab587f2b..34aacf73 100644
--- a/tests/test_server_attachments_endpoints.py
+++ b/tests/test_server_attachments_endpoints.py
@@ -1082,3 +1082,210 @@ class TestServiceScopedActorFlow:
         atts = captured["attachments"]
         assert atts is not None and len(atts) == 1
         assert atts[0].attachment_id == aid
+
+
+# ---------------------------------------------------------------------------
+# Voice I/O (STT / TTS) endpoints
+# ---------------------------------------------------------------------------
+
+
+class _VoiceConfigStore:
+    def __init__(self, **values: str) -> None:
+        self._values = dict(values)
+
+    def get(self, key: str, default: str = "") -> str:
+        return self._values.get(key, default)
+
+
+@pytest.fixture
+def voice_app_client(tmp_path):
+    """App wired with an audio-capable registry alias + a mocked OpenAI client.
+
+    The mock is injected into ``registry._clients`` so the real endpoint →
+    resolve_role_alias → transcribe/synthesize path runs end-to-end with only
+    the SDK network call stubbed.
+    """
+    import sqlalchemy as sa
+
+    import turnstone.server as srv_mod
+    from turnstone.core.memory import register_workstream
+    from turnstone.core.metrics import MetricsCollector
+    from turnstone.core.model_registry import ModelConfig, ModelRegistry
+    from turnstone.core.storage import init_storage, reset_storage
+    from turnstone.core.storage._registry import get_storage
+    from turnstone.core.storage._schema import workstreams as ws_tbl
+
+    db_path = tmp_path / "voice.db"
+    reset_storage()
+    init_storage("sqlite", path=str(db_path), run_migrations=False)
+
+    srv_mod._metrics = MetricsCollector()
+    srv_mod._metrics.model = "test-model"
+
+    register_workstream("ws-A", name="A")
+    with get_storage()._conn() as conn:
+        conn.execute(sa.update(ws_tbl).where(ws_tbl.c.ws_id == "ws-A").values(user_id="userA"))
+        conn.commit()
+
+    registry = ModelRegistry(
+        models={
+            "voice": ModelConfig(
+                "voice",
+                "http://localhost:9/v1",
+                "none",
+                "gpt-4o-mini-tts",
+                capabilities={
+                    "supports_transcription": True,
+                    "supports_speech_synthesis": True,
+                },
+            ),
+        },
+        default="voice",
+    )
+    mock_client = MagicMock()
+    mock_client.audio.transcriptions.create.return_value = MagicMock(text="hello from speech")
+    speech = MagicMock()
+    speech.read.return_value = b"RIFF\x00\x00fakeaudio"
+    mock_client.audio.speech.create.return_value = speech
+    registry._clients["voice"] = mock_client  # bypass real SDK client construction
+
+    config_store = _VoiceConfigStore(
+        **{
+            "audio.stt_model_alias": "voice",
+            "audio.tts_model_alias": "voice",
+            "audio.tts_voice": "alloy",
+        }
+    )
+
+    mock_mgr = MagicMock()
+    mock_mgr.get.return_value = None
+    mock_mgr.list_all.return_value = []
+    mock_mgr.max_active = 10
+
+    app = srv_mod.create_app(
+        workstreams=mock_mgr,
+        global_queue=queue.Queue(),
+        global_listeners=[],
+        global_listeners_lock=threading.Lock(),
+        skip_permissions=False,
+        jwt_secret=_TEST_JWT_SECRET,
+        registry=registry,
+        config_store=config_store,
+    )
+    client = TestClient(app, raise_server_exceptions=False)
+    try:
+        yield client, mock_client
+    finally:
+        client.close()
+        reset_storage()
+
+
+class TestSpeechToText:
+    def test_unconfigured_returns_503(self, app_client):
+        client, _ = app_client
+        resp = client.post(
+            "/v1/api/workstreams/ws-A/speech-to-text",
+            files={"audio": ("speech.webm", b"RIFFfake", "audio/webm")},
+            headers=_auth("userA"),
+        )
+        assert resp.status_code == 503
+        assert "not configured" in resp.json()["error"]
+
+    def test_happy_path_returns_transcript(self, voice_app_client):
+        client, mock_client = voice_app_client
+        resp = client.post(
+            "/v1/api/workstreams/ws-A/speech-to-text",
+            files={"audio": ("speech.webm", b"RIFFfake", "audio/webm")},
+            headers=_auth("userA"),
+        )
+        assert resp.status_code == 200, resp.text
+        body = resp.json()
+        assert body["transcript"] == "hello from speech"
+        assert body["model_alias"] == "voice"
+        assert mock_client.audio.transcriptions.create.called
+
+    def test_empty_upload_returns_400(self, voice_app_client):
+        client, _ = voice_app_client
+        resp = client.post(
+            "/v1/api/workstreams/ws-A/speech-to-text",
+            files={"audio": ("speech.webm", b"", "audio/webm")},
+            headers=_auth("userA"),
+        )
+        assert resp.status_code == 400
+
+    def test_silence_returns_422(self, voice_app_client):
+        # A successful transcription with no speech is not a backend failure.
+        client, mock_client = voice_app_client
+        mock_client.audio.transcriptions.create.return_value = MagicMock(text="   ")
+        resp = client.post(
+            "/v1/api/workstreams/ws-A/speech-to-text",
+            files={"audio": ("speech.webm", b"RIFFfake", "audio/webm")},
+            headers=_auth("userA"),
+        )
+        assert resp.status_code == 422
+        assert "No speech detected" in resp.json()["error"]
+
+    def test_backend_failure_returns_masked_502(self, voice_app_client):
+        # Backend SDK error detail must not leak into the client-facing body.
+        client, mock_client = voice_app_client
+        mock_client.audio.transcriptions.create.side_effect = RuntimeError(
+            "Error code: 401 - internal-host:9 invalid_api_key"
+        )
+        resp = client.post(
+            "/v1/api/workstreams/ws-A/speech-to-text",
+            files={"audio": ("speech.webm", b"RIFFfake", "audio/webm")},
+            headers=_auth("userA"),
+        )
+        assert resp.status_code == 502
+        body = resp.json()
+        assert body["error"] == "Speech transcription backend failed"
+        assert "internal-host" not in body["error"]
+
+    def test_unknown_workstream_404(self, voice_app_client):
+        # Trusted-team semantics: ownership isn't row-enforced, but a
+        # nonexistent workstream is masked as 404 (no enumeration).
+        client, _ = voice_app_client
+        resp = client.post(
+            "/v1/api/workstreams/ws-DOES-NOT-EXIST/speech-to-text",
+            files={"audio": ("speech.webm", b"RIFFfake", "audio/webm")},
+            headers=_auth("userA"),
+        )
+        assert resp.status_code == 404
+
+
+class TestTextToSpeech:
+    def test_unconfigured_returns_503(self, app_client):
+        client, _ = app_client
+        resp = client.post("/v1/api/tts", json={"text": "hello"}, headers=_auth("userA"))
+        assert resp.status_code == 503
+
+    def test_happy_path_returns_audio(self, voice_app_client):
+        client, mock_client = voice_app_client
+        resp = client.post("/v1/api/tts", json={"text": "hello"}, headers=_auth("userA"))
+        assert resp.status_code == 200, resp.text
+        assert resp.headers["content-type"].startswith("audio/")
+        assert resp.content == b"RIFF\x00\x00fakeaudio"
+        assert resp.headers.get("x-model-alias") == "voice"
+        # audio.tts_voice setting supplies the voice when the body omits one.
+        assert mock_client.audio.speech.create.call_args.kwargs["voice"] == "alloy"
+
+    def test_empty_text_returns_400(self, voice_app_client):
+        client, _ = voice_app_client
+        resp = client.post("/v1/api/tts", json={"text": "   "}, headers=_auth("userA"))
+        assert resp.status_code == 400
+
+    def test_too_long_text_returns_400(self, voice_app_client):
+        client, _ = voice_app_client
+        resp = client.post("/v1/api/tts", json={"text": "x" * 9000}, headers=_auth("userA"))
+        assert resp.status_code == 400
+
+    def test_backend_failure_returns_masked_502(self, voice_app_client):
+        client, mock_client = voice_app_client
+        mock_client.audio.speech.create.side_effect = RuntimeError(
+            "Error code: 500 - internal-host:9 boom"
+        )
+        resp = client.post("/v1/api/tts", json={"text": "hello"}, headers=_auth("userA"))
+        assert resp.status_code == 502
+        body = resp.json()
+        assert body["error"] == "Speech synthesis backend failed"
+        assert "internal-host" not in body["error"]
diff --git a/tests/test_server_available_models.py b/tests/test_server_available_models.py
index 958adca3..81368521 100644
--- a/tests/test_server_available_models.py
+++ b/tests/test_server_available_models.py
@@ -52,6 +52,7 @@ class _StubRegistry:
             alias=alias,
             model=self._aliases[alias],
             provider="openai-compatible",
+            capabilities={},
         )
 
 
diff --git a/tests/test_settings_registry.py b/tests/test_settings_registry.py
index db581cfe..db44149e 100644
--- a/tests/test_settings_registry.py
+++ b/tests/test_settings_registry.py
@@ -28,6 +28,24 @@ class TestValidateKey:
         with pytest.raises(ValueError, match="Unknown setting"):
             validate_key("nonexistent.key")
 
+    def test_audio_role_settings_registered(self):
+        for key in (
+            "audio.stt_model_alias",
+            "audio.stt_prompt",
+            "audio.tts_model_alias",
+            "audio.tts_voice",
+        ):
+            defn = validate_key(key)
+            assert defn.key == key
+            assert defn.type == "str"
+            assert defn.section == "audio"
+            assert key in SETTINGS
+        # Voice has a concrete default; the role aliases + prompt default to empty.
+        assert validate_key("audio.stt_model_alias").default == ""
+        assert validate_key("audio.tts_model_alias").default == ""
+        assert validate_key("audio.stt_prompt").default == ""
+        assert validate_key("audio.tts_voice").default == "alloy"
+
 
 # ---------------------------------------------------------------------------
 # validate_value — type coercion
diff --git a/turnstone/api/server_schemas.py b/turnstone/api/server_schemas.py
index 52c9a6d3..018e6cff 100644
--- a/turnstone/api/server_schemas.py
+++ b/turnstone/api/server_schemas.py
@@ -87,6 +87,21 @@ class ListAttachmentsResponse(BaseModel):
     )
 
 
+class SpeechToTextResponse(BaseModel):
+    """Transcript returned for the browser to place into the composer."""
+
+    status: str = Field(default="ok", description="Request outcome")
+    transcript: str = Field(description="Transcribed text")
+    model_alias: str = Field(default="", description="STT role alias used")
+
+
+class TextToSpeechRequest(BaseModel):
+    text: str = Field(description="Text to synthesize")
+    voice: str = Field(
+        default="", description="Optional voice override (else audio.tts_voice setting)"
+    )
+
+
 class ApproveRequest(BaseModel):
     approved: bool = Field(description="True to approve, false to deny")
     feedback: str | None = Field(default=None, description="Optional denial reason")
@@ -628,6 +643,10 @@ class AvailableModelInfo(BaseModel):
     alias: str
     model: str
     provider: str
+    capabilities: dict[str, Any] = Field(
+        default_factory=dict,
+        description="Operator-set capability flags for this alias (e.g. supports_transcription)",
+    )
 
 
 class ListAvailableModelsResponse(BaseModel):
@@ -635,3 +654,11 @@ class ListAvailableModelsResponse(BaseModel):
     default_alias: str = ""
     channel_default_alias: str = ""
     judge_default_alias: str = ""
+    stt_default_alias: str = Field(
+        default="",
+        description="Effective speech-to-text role alias (blank = voice input disabled)",
+    )
+    tts_default_alias: str = Field(
+        default="",
+        description="Effective text-to-speech role alias (blank = voice output disabled)",
+    )
diff --git a/turnstone/api/server_spec.py b/turnstone/api/server_spec.py
index d7107c94..d3265ce2 100644
--- a/turnstone/api/server_spec.py
+++ b/turnstone/api/server_spec.py
@@ -43,6 +43,8 @@ from turnstone.api.server_schemas import (
     SendRequest,
     SendResponse,
     SkillSummary,
+    SpeechToTextResponse,
+    TextToSpeechRequest,
     UploadAttachmentResponse,
     WorkstreamDetailResponse,
     WorkstreamHistoryResponse,
@@ -313,6 +315,28 @@ SERVER_ENDPOINTS: list[EndpointSpec] = [
         error_codes=[403, 404],
         tags=["Attachments"],
     ),
+    # --- Voice I/O ---
+    EndpointSpec(
+        "/v1/api/workstreams/{ws_id}/speech-to-text",
+        "POST",
+        "Transcribe a short audio clip (multipart/form-data, field 'audio') "
+        "using the configured STT model role.  Returns the transcript for the "
+        "client to place into the composer; this endpoint never sends on the "
+        "user's behalf.  Returns 503 when no STT role is configured.",
+        response_model=SpeechToTextResponse,
+        error_codes=[400, 403, 404, 413, 502, 503],
+        tags=["Attachments"],
+    ),
+    EndpointSpec(
+        "/v1/api/tts",
+        "POST",
+        "Synthesize text to speech audio for browser playback using the "
+        "configured TTS model role.  Returns audio bytes; 503 when no TTS "
+        "role is configured.",
+        request_model=TextToSpeechRequest,
+        error_codes=[400, 502, 503],
+        tags=["Chat"],
+    ),
     # --- Saved workstreams ---
     EndpointSpec(
         "/v1/api/workstreams/saved",
@@ -495,6 +519,8 @@ _ALL_MODELS: list[type[BaseModel]] = [
     ListSavedWorkstreamsResponse,
     UploadAttachmentResponse,
     ListAttachmentsResponse,
+    SpeechToTextResponse,
+    TextToSpeechRequest,
     HealthResponse,
     SaveMemoryRequest,
     MemoryInfo,
diff --git a/turnstone/console/static/admin.js b/turnstone/console/static/admin.js
index 66e28d31..b22a27e0 100644
--- a/turnstone/console/static/admin.js
+++ b/turnstone/console/static/admin.js
@@ -22,6 +22,8 @@ const ALIAS_SETTING_KEYS = [
   "model.plan_alias",
   "model.task_alias",
   "channels.default_model_alias",
+  "audio.stt_model_alias",
+  "audio.tts_model_alias",
 ];
 
 // Settings whose empty option means "inherit from a fallback chain", as
@@ -2747,6 +2749,7 @@ function _settingsSectionLabel(section) {
     server: "Server",
     cluster: "Cluster",
     channels: "Channels",
+    audio: "Voice",
     mcp: "MCP",
     ratelimit: "Rate Limiting",
     health: "Health",
@@ -5192,8 +5195,58 @@ const MODEL_ROLES = [
     aliasKey: "channels.default_model_alias",
     fallbackKind: "default",
   },
+  {
+    label: "Speech-to-text",
+    description:
+      "Transcribes microphone audio in the workstream composer (voice input). Empty disables the mic affordance — there is no audio-capable session fallback.",
+    aliasKey: "audio.stt_model_alias",
+    fallbackKind: "disabled",
+    mediaCapability: "supports_transcription",
+    mediaRole: "stt",
+  },
+  {
+    label: "Text-to-speech",
+    description:
+      "Synthesizes assistant replies for playback (voice output). Empty disables the play affordance.",
+    aliasKey: "audio.tts_model_alias",
+    fallbackKind: "disabled",
+    mediaCapability: "supports_speech_synthesis",
+    mediaRole: "tts",
+  },
 ];
 
+// Whether a model definition is eligible for an audio role.  Mirrors
+// turnstone/core/audio.py model_supports_role: an explicit capability flag
+// wins; otherwise infer from well-known OpenAI audio model names so a stock
+// gpt-4o-mini-transcribe / -tts / whisper alias shows up without hand-ticking.
+// Known-model-name hints, mirrored VERBATIM from _AUDIO_MODEL_HINTS in
+// turnstone/core/audio.py (the canonical source). A substring match marks
+// eligibility. Keep these two lists in sync — the Python side is pinned by
+// tests/test_audio.py so any change there is deliberate.
+const AUDIO_MODEL_HINTS = {
+  stt: ["transcribe", "whisper", "-asr"],
+  tts: ["tts-", "-tts"],
+};
+
+function _audioModelEligible(md, capFlag, mediaRole) {
+  let caps = md && md.capabilities;
+  if (typeof caps === "string") {
+    try {
+      caps = JSON.parse(caps || "{}");
+    } catch (e) {
+      caps = {};
+    }
+  }
+  if (!caps || typeof caps !== "object") caps = {};
+  if (Object.prototype.hasOwnProperty.call(caps, capFlag))
+    return !!caps[capFlag];
+  const name = ((md && md.model) || "").toLowerCase();
+  const hints = AUDIO_MODEL_HINTS[mediaRole] || [];
+  return hints.some(function (h) {
+    return name.indexOf(h) !== -1;
+  });
+}
+
 // Roles sub-tab reads/writes via ``/v1/api/admin/settings`` which
 // requires ``admin.settings`` — different from the ``admin.models``
 // permission gating the Models tab itself.  When the user has Models
@@ -5387,7 +5440,9 @@ function _renderModelRoles(container, values, schema) {
     // "(default — <coordinator-alias>)".
     const blank = document.createElement("option");
     blank.value = "";
-    if (role.fallbackKind === "inherit") {
+    if (role.fallbackKind === "disabled") {
+      blank.textContent = "(disabled — voice off)";
+    } else if (role.fallbackKind === "inherit") {
       blank.textContent = "(inherit)";
     } else {
       let defaultDef = null;
@@ -5412,10 +5467,18 @@ function _renderModelRoles(container, values, schema) {
       }
     }
     aliasSel.appendChild(blank);
+    // Audio roles only list aliases capable of the role (capability flag or
+    // known-model inference); other roles list every enabled alias.
+    let roleAliases = enabledAliases;
+    if (role.mediaCapability) {
+      roleAliases = enabledAliases.filter(function (md) {
+        return _audioModelEligible(md, role.mediaCapability, role.mediaRole);
+      });
+    }
     const currentAlias = aliasInfo.value || "";
     let matched = false;
-    for (let m = 0; m < enabledAliases.length; m++) {
-      const md = enabledAliases[m];
+    for (let m = 0; m < roleAliases.length; m++) {
+      const md = roleAliases[m];
       const opt = document.createElement("option");
       opt.value = md.alias;
       opt.textContent =
diff --git a/turnstone/core/audio.py b/turnstone/core/audio.py
new file mode 100644
index 00000000..a4436846
--- /dev/null
+++ b/turnstone/core/audio.py
@@ -0,0 +1,186 @@
+"""Speech-to-text and text-to-speech over the OpenAI audio wire protocol.
+
+STT/TTS are model *roles* (settings ``stt.model_alias`` / ``tts.model_alias``),
+resolved the same way as ``judge.model``.  A role resolves to a registry alias
+whose client is an OpenAI-SDK-compatible client (provider ``openai`` /
+``openai-compatible`` / ``google`` / ``xai``); the same ``client.audio.*`` calls
+then work against OpenAI, a local vLLM / vLLM-Omni server, or any compatible
+backend — selected purely by the alias's ``base_url``.  Anthropic models have no
+audio API, so they are capability-gated out of these roles (Claude stays valid
+as the agent model).
+
+No local in-process models and no silent fallbacks: an unconfigured or failing
+backend is surfaced as a typed error the endpoint maps to 503 / 502.
+"""
+
+from __future__ import annotations
+
+from dataclasses import dataclass
+from typing import Any
+
+# Setting key + capability flag per media role.  Kept deliberately small; the
+# perception/eval roles (vision_eval/av_eval/intent_eval) are a later slice.
+_ROLE_SETTING: dict[str, str] = {
+    "stt": "audio.stt_model_alias",
+    "tts": "audio.tts_model_alias",
+}
+_ROLE_CAPABILITY: dict[str, str] = {
+    "stt": "supports_transcription",
+    "tts": "supports_speech_synthesis",
+}
+
+# Known-model-name hints for capability inference. The explicit
+# ``capabilities`` flag is ALWAYS canonical (see ``model_supports_role``); these
+# only fill the gap so a stock OpenAI audio model alias works without an
+# operator hand-ticking a box. A substring match against the lowercased model
+# name marks eligibility.
+#
+# IMPORTANT: these lists are mirrored verbatim in the admin UI
+# (``_audioModelEligible`` / ``AUDIO_*_MODEL_HINTS`` in
+# ``turnstone/console/static/admin.js``) so the Models -> Roles dropdown offers
+# exactly the aliases the endpoints will accept. Keep the two in sync;
+# ``tests/test_audio.py`` pins these values so a change here is deliberate.
+_AUDIO_MODEL_HINTS: dict[str, tuple[str, ...]] = {
+    "stt": ("transcribe", "whisper", "-asr"),
+    "tts": ("tts-", "-tts"),
+}
+
+# response_format -> Content-Type for synthesized audio.
+_MEDIA_TYPES: dict[str, str] = {
+    "mp3": "audio/mpeg",
+    "opus": "audio/ogg",
+    "aac": "audio/aac",
+    "flac": "audio/flac",
+    "wav": "audio/wav",
+    "pcm": "audio/pcm",
+}
+
+_DEFAULT_VOICE = "alloy"
+
+
+@dataclass(frozen=True)
+class TranscriptionResult:
+    transcript: str
+    model_alias: str
+    model: str
+
+
+@dataclass(frozen=True)
+class SpeechResult:
+    audio_bytes: bytes
+    media_type: str
+    model_alias: str
+    model: str
+
+
+class AudioUnavailableError(RuntimeError):
+    """No capable backend is configured for the requested role (maps to 503)."""
+
+
+class AudioBackendError(RuntimeError):
+    """A configured audio backend failed during execution (maps to 502)."""
+
+
+def _infer_audio_capability(model: str, role: str) -> bool:
+    """Best-effort capability default for well-known audio model names.
+
+    The explicit ``capabilities`` flag always wins (see ``model_supports_role``);
+    this only fills the gap so a stock ``gpt-4o-mini-transcribe`` /
+    ``gpt-4o-mini-tts`` / ``whisper-1`` alias works without an operator
+    hand-ticking a capability box. Rules live in :data:`_AUDIO_MODEL_HINTS`
+    (mirrored in admin.js).
+    """
+    name = (model or "").strip().lower()
+    if not name:
+        return False
+    return any(hint in name for hint in _AUDIO_MODEL_HINTS.get(role, ()))
+
+
+def model_supports_role(cfg: Any, role: str) -> bool:
+    """Whether the alias's model is eligible for a media *role*.
+
+    Explicit ``capabilities[<flag>]`` wins; otherwise fall back to a
+    known-model-name inference for OpenAI audio models.
+    """
+    flag = _ROLE_CAPABILITY.get(role)
+    if not flag:
+        return False
+    caps = getattr(cfg, "capabilities", None) or {}
+    if flag in caps:
+        return bool(caps.get(flag))
+    return _infer_audio_capability(getattr(cfg, "model", ""), role)
+
+
+def resolve_role_alias(*, config_store: Any | None, registry: Any | None, role: str) -> str | None:
+    """Return the configured, capability-eligible alias for *role*, or ``None``.
+
+    Resolution: ``<role>.model_alias`` setting → must exist in the registry →
+    must be capability-eligible for the role.  Any miss returns ``None`` so the
+    caller surfaces a 503 (or hides the affordance) rather than calling a
+    backend that can't serve audio.
+    """
+    if registry is None or config_store is None:
+        return None
+    key = _ROLE_SETTING.get(role)
+    if not key:
+        return None
+    alias = (config_store.get(key) or "").strip()
+    if not alias or not registry.has_alias(alias):
+        return None
+    if not model_supports_role(registry.get_config(alias), role):
+        return None
+    return alias
+
+
+def transcribe(
+    *, registry: Any, alias: str, data: bytes, filename: str, prompt: str = ""
+) -> TranscriptionResult:
+    """Transcribe ``data`` using the STT role alias's audio backend.
+
+    ``prompt`` (when non-empty) is forwarded as the transcription ``prompt``
+    parameter to bias the model toward domain vocabulary / instructions; it is
+    omitted entirely when blank so backends that don't accept it aren't sent it.
+    """
+    try:
+        client, model, _cfg = registry.resolve(alias)
+    except Exception as exc:  # unknown/removed alias
+        raise AudioUnavailableError(f"STT model alias {alias!r} is not available") from exc
+    kwargs: dict[str, Any] = {
+        "model": model,
+        "file": (filename or "speech.webm", data),
+        "response_format": "json",
+    }
+    if prompt:
+        kwargs["prompt"] = prompt
+    try:
+        resp = client.audio.transcriptions.create(**kwargs)
+        transcript = (getattr(resp, "text", "") or "").strip()
+    except Exception as exc:
+        raise AudioBackendError(f"Transcription backend failed: {exc}") from exc
+    return TranscriptionResult(transcript=transcript, model_alias=alias, model=model)
+
+
+def synthesize(
+    *, registry: Any, alias: str, text: str, voice: str, response_format: str = "mp3"
+) -> SpeechResult:
+    """Synthesize ``text`` to speech using the TTS role alias's audio backend."""
+    try:
+        client, model, _cfg = registry.resolve(alias)
+    except Exception as exc:
+        raise AudioUnavailableError(f"TTS model alias {alias!r} is not available") from exc
+    try:
+        resp = client.audio.speech.create(
+            model=model,
+            voice=voice or _DEFAULT_VOICE,
+            input=text,
+            response_format=response_format,
+        )
+        audio_bytes = resp.read() if hasattr(resp, "read") else bytes(getattr(resp, "content", b""))
+    except Exception as exc:
+        raise AudioBackendError(f"TTS backend failed: {exc}") from exc
+    return SpeechResult(
+        audio_bytes=audio_bytes,
+        media_type=_MEDIA_TYPES.get(response_format, "audio/mpeg"),
+        model_alias=alias,
+        model=model,
+    )
diff --git a/turnstone/core/auth.py b/turnstone/core/auth.py
index ec19d6e1..b97a0f57 100644
--- a/turnstone/core/auth.py
+++ b/turnstone/core/auth.py
@@ -309,6 +309,7 @@ WRITE_PATHS: frozenset[str] = frozenset(
         "/api/workstreams/new",
         "/api/cluster/workstreams/new",
         "/api/memories",
+        "/api/tts",
     }
 )
 
@@ -694,6 +695,7 @@ def required_scope(method: str, path: str) -> str:
             "refresh-title",
             "title",
             "attachments",
+            "speech-to-text",
             "send",
             "cancel",
             "rewind",
@@ -744,6 +746,7 @@ def required_scope(method: str, path: str) -> str:
                 "refresh-title",
                 "title",
                 "attachments",
+                "speech-to-text",
                 "send",
                 "cancel",
                 "rewind",
diff --git a/turnstone/core/providers/_openai_common.py b/turnstone/core/providers/_openai_common.py
index 86ad0ba9..44c6d600 100644
--- a/turnstone/core/providers/_openai_common.py
+++ b/turnstone/core/providers/_openai_common.py
@@ -197,6 +197,39 @@ OPENAI_CAPABILITIES: dict[str, ModelCapabilities] = {
         reasoning_effort_values=(),
         supports_vision=True,
     ),
+    # Audio models — not chat/session models; used only as STT/TTS roles via
+    # the /v1/audio/transcriptions and /v1/audio/speech endpoints. Prefixes
+    # cover variants: "gpt-4o-transcribe" → "-diarize", "tts-1" → "tts-1-hd".
+    "whisper-1": ModelCapabilities(
+        supports_temperature=False,
+        supports_streaming=False,
+        supports_tools=False,
+        supports_transcription=True,
+    ),
+    "gpt-4o-transcribe": ModelCapabilities(
+        supports_temperature=False,
+        supports_streaming=False,
+        supports_tools=False,
+        supports_transcription=True,
+    ),
+    "gpt-4o-mini-transcribe": ModelCapabilities(
+        supports_temperature=False,
+        supports_streaming=False,
+        supports_tools=False,
+        supports_transcription=True,
+    ),
+    "tts-1": ModelCapabilities(
+        supports_temperature=False,
+        supports_streaming=False,
+        supports_tools=False,
+        supports_speech_synthesis=True,
+    ),
+    "gpt-4o-mini-tts": ModelCapabilities(
+        supports_temperature=False,
+        supports_streaming=False,
+        supports_tools=False,
+        supports_speech_synthesis=True,
+    ),
 }
 
 # Default for unknown models (local servers: vLLM, llama.cpp, etc.)
diff --git a/turnstone/core/providers/_protocol.py b/turnstone/core/providers/_protocol.py
index ccd86d7d..58ea5094 100644
--- a/turnstone/core/providers/_protocol.py
+++ b/turnstone/core/providers/_protocol.py
@@ -86,6 +86,10 @@ class ModelCapabilities:
     supports_web_search: bool = False
     supports_tool_search: bool = False
     supports_vision: bool = False
+    # Audio I/O roles (STT / TTS) — not chat behavior; consumed by the audio
+    # endpoints and the Models -> Roles capability gate (turnstone/core/audio.py).
+    supports_transcription: bool = False
+    supports_speech_synthesis: bool = False
     # Server-side tool types to auto-inject into Responses-API ``tools[]``
     # for this model (e.g. ``("web_search",)`` for OpenAI search models,
     # ``("web_search", "x_search")`` for Grok variants).  The
diff --git a/turnstone/core/settings_registry.py b/turnstone/core/settings_registry.py
index fa8117e0..92c85683 100644
--- a/turnstone/core/settings_registry.py
+++ b/turnstone/core/settings_registry.py
@@ -404,6 +404,54 @@ def _build_registry() -> dict[str, SettingDef]:
             "Degraded backends are deprioritised in the fallback chain but requests are never "
             "blocked. The backend recovers automatically when a request succeeds.",
         ),
+        # -- audio / voice roles -------------------------------------------
+        # Keys are section-prefixed (audio.*) with distinct leaves so the
+        # Settings tab (which labels by the key's last segment) doesn't render
+        # two identical "model_alias" rows.
+        SettingDef(
+            "audio.stt_model_alias",
+            "str",
+            "",
+            "Model alias for speech-to-text (empty = voice input disabled)",
+            "audio",
+            help="Which registered model alias transcribes microphone audio. Point it at an "
+            "audio-capable backend (e.g. an OpenAI gpt-4o-transcribe alias, or a local vLLM "
+            "whisper endpoint). Empty disables the microphone affordance — there is no "
+            "audio-capable session fallback, so this must name a transcription model. The "
+            "curated, capability-gated picker lives in Models -> Roles.",
+        ),
+        SettingDef(
+            "audio.stt_prompt",
+            "str",
+            "",
+            "Optional prompt sent with each transcription request",
+            "audio",
+            help="Optional text passed to the speech-to-text backend to bias the transcription "
+            "— useful for domain vocabulary, names, or acronyms, and required by some models "
+            "(e.g. Gemma-style ASR) that take an instruction prompt. Sent as the OpenAI "
+            "transcription `prompt` parameter; leave empty to omit it.",
+        ),
+        SettingDef(
+            "audio.tts_model_alias",
+            "str",
+            "",
+            "Model alias for text-to-speech (empty = voice output disabled)",
+            "audio",
+            help="Which registered model alias synthesizes assistant speech. Point it at an "
+            "audio-capable backend (e.g. an OpenAI gpt-4o-mini-tts alias, or a local "
+            "vLLM-Omni speech endpoint). Empty disables the playback affordance. The curated, "
+            "capability-gated picker lives in Models -> Roles.",
+        ),
+        SettingDef(
+            "audio.tts_voice",
+            "str",
+            "alloy",
+            "Voice identifier for text-to-speech",
+            "audio",
+            help="Voice passed to the TTS backend. OpenAI voices include alloy, echo, fable, "
+            "onyx, nova, shimmer; local backends (vLLM-Omni, Kokoro) define their own — set "
+            "this to a voice your configured TTS model backend supports.",
+        ),
         # -- judge ----------------------------------------------------------
         SettingDef(
             "judge.enabled",
diff --git a/turnstone/server.py b/turnstone/server.py
index bd9b7adb..1937a91a 100644
--- a/turnstone/server.py
+++ b/turnstone/server.py
@@ -1176,6 +1176,7 @@ async def list_available_models(request: Request) -> JSONResponse:
                 "alias": cfg.alias,
                 "model": cfg.model,
                 "provider": cfg.provider,
+                "capabilities": cfg.capabilities,
             }
         )
     # Include effective defaults for clients (web UI, channel gateway).
@@ -1210,16 +1211,165 @@ async def list_available_models(request: Request) -> JSONResponse:
     # an explicitly-configured, enabled alias surfaces a concrete default.
     if judge_default_alias and judge_default_alias not in enabled_aliases:
         judge_default_alias = ""
+    # STT/TTS are media roles: report a default only when the configured alias
+    # exists AND is capability-eligible (the same gate the endpoints apply), so
+    # the UI shows the mic / playback affordances only when they actually work.
+    from turnstone.core.audio import resolve_role_alias
+
+    stt_default_alias = resolve_role_alias(config_store=cs, registry=registry, role="stt") or ""
+    tts_default_alias = resolve_role_alias(config_store=cs, registry=registry, role="tts") or ""
     return JSONResponse(
         {
             "models": models,
             "default_alias": default_alias,
             "channel_default_alias": channel_default_alias,
             "judge_default_alias": judge_default_alias,
+            "stt_default_alias": stt_default_alias,
+            "tts_default_alias": tts_default_alias,
         }
     )
 
 
+_STT_UPLOAD_CAP = 25 * 1024 * 1024  # 25 MiB — generous for short dictation clips
+_TTS_TEXT_CAP = 8000  # characters per synthesis request
+
+
+async def speech_to_text(request: Request) -> JSONResponse:
+    """POST /v1/api/workstreams/{ws_id}/speech-to-text — transcribe one audio clip.
+
+    Multipart body with a single ``audio`` field.  Returns the transcript for
+    the browser to place into the composer; this endpoint never sends on the
+    user's behalf (no auto-send, no request rewriting).
+    """
+    from turnstone.core.audio import (
+        AudioBackendError,
+        AudioUnavailableError,
+        resolve_role_alias,
+        transcribe,
+    )
+    from turnstone.core.web_helpers import read_multipart_file_or_400
+
+    ws_id = request.path_params.get("ws_id", "")
+    if not ws_id:
+        return JSONResponse({"error": "ws_id is required"}, status_code=400)
+    _user_id, err = _require_ws_access(request, ws_id)
+    if err:
+        return err
+
+    registry = getattr(request.app.state, "registry", None)
+    config_store = getattr(request.app.state, "config_store", None)
+    alias = resolve_role_alias(config_store=config_store, registry=registry, role="stt")
+    if not alias:
+        return JSONResponse(
+            {
+                "error": (
+                    "Speech-to-text is not configured. Assign an STT model role in Models → Roles."
+                )
+            },
+            status_code=503,
+        )
+
+    got = await read_multipart_file_or_400(request, field="audio", max_bytes=_STT_UPLOAD_CAP)
+    if isinstance(got, JSONResponse):
+        return got
+    filename, _claimed_mime, data = got
+    if not data:
+        return JSONResponse({"error": "Empty audio upload"}, status_code=400)
+
+    stt_prompt = ""
+    if config_store is not None:
+        stt_prompt = (config_store.get("audio.stt_prompt") or "").strip()
+    try:
+        # Blocking SDK round-trip — offload so the shared event loop (and SSE
+        # streaming) stays responsive.
+        result = await asyncio.to_thread(
+            transcribe,
+            registry=registry,
+            alias=alias,
+            data=data,
+            filename=filename or "speech.webm",
+            prompt=stt_prompt,
+        )
+    except AudioUnavailableError as exc:
+        return JSONResponse({"error": str(exc)}, status_code=503)
+    except AudioBackendError as exc:
+        # str(exc) wraps the backend SDK error, which can carry upstream
+        # response detail — log it, but return a static body to the caller.
+        log.warning("speech_to_text.backend_failed", error=str(exc), exc_info=True)
+        return JSONResponse({"error": "Speech transcription backend failed"}, status_code=502)
+
+    if not result.transcript:
+        # Successful call that detected no speech (silence / non-speech audio)
+        # is not a backend failure — surface it as 422 so the UI can say so
+        # rather than treating a healthy backend as a bad gateway.
+        return JSONResponse({"error": "No speech detected"}, status_code=422)
+
+    return JSONResponse(
+        {
+            "status": "ok",
+            "transcript": result.transcript,
+            "model_alias": result.model_alias,
+        }
+    )
+
+
+async def text_to_speech(request: Request) -> Response:
+    """POST /v1/api/tts — synthesize assistant text into playable audio."""
+    from turnstone.core.audio import (
+        AudioBackendError,
+        AudioUnavailableError,
+        resolve_role_alias,
+        synthesize,
+    )
+    from turnstone.core.web_helpers import read_json_or_400
+
+    body = await read_json_or_400(request)
+    if isinstance(body, JSONResponse):
+        return body
+    text = str(body.get("text") or "").strip()
+    if not text:
+        return JSONResponse({"error": "text is required"}, status_code=400)
+    if len(text) > _TTS_TEXT_CAP:
+        return JSONResponse(
+            {"error": f"text too long (cap {_TTS_TEXT_CAP} chars)"}, status_code=400
+        )
+
+    registry = getattr(request.app.state, "registry", None)
+    config_store = getattr(request.app.state, "config_store", None)
+    alias = resolve_role_alias(config_store=config_store, registry=registry, role="tts")
+    if not alias:
+        return JSONResponse(
+            {
+                "error": (
+                    "Text-to-speech is not configured. Assign a TTS model role in Models → Roles."
+                )
+            },
+            status_code=503,
+        )
+
+    voice = str(body.get("voice") or "").strip()
+    if not voice and config_store is not None:
+        voice = (config_store.get("audio.tts_voice") or "").strip()
+
+    try:
+        # Blocking SDK round-trip — offload off the event loop.
+        speech = await asyncio.to_thread(
+            synthesize, registry=registry, alias=alias, text=text, voice=voice
+        )
+    except AudioUnavailableError as exc:
+        return JSONResponse({"error": str(exc)}, status_code=503)
+    except AudioBackendError as exc:
+        # Static body to the caller; backend SDK detail stays in the log.
+        log.warning("text_to_speech.backend_failed", error=str(exc), exc_info=True)
+        return JSONResponse({"error": "Speech synthesis backend failed"}, status_code=502)
+
+    return Response(
+        speech.audio_bytes,
+        media_type=speech.media_type,
+        headers={"X-Model-Alias": speech.model_alias},
+    )
+
+
 def _count_ws_states(wss: list[Workstream]) -> dict[str, int]:
     """Count workstream states for health/metrics endpoints."""
     counts = dict.fromkeys(("idle", "thinking", "running", "attention", "error"), 0)
@@ -3830,6 +3980,14 @@ def create_app(
         ),
     )
     v1_routes.append(Route("/api/dashboard", dashboard))
+    v1_routes.append(
+        Route(
+            "/api/workstreams/{ws_id}/speech-to-text",
+            speech_to_text,
+            methods=["POST"],
+        )
+    )
+    v1_routes.append(Route("/api/tts", text_to_speech, methods=["POST"]))
 
     app = Starlette(
         routes=[
diff --git a/turnstone/shared_static/chat.css b/turnstone/shared_static/chat.css
index 6dbec6bf..3d819fb4 100644
--- a/turnstone/shared_static/chat.css
+++ b/turnstone/shared_static/chat.css
@@ -364,8 +364,115 @@
   border-color: var(--border-strong);
 }
 
+/* Voice I/O — mic dictation button (composer row) mirrors .composer-attach,
+   plus a recording-pulse state and the per-message TTS playback states. */
+.composer-mic-btn {
+  display: inline-flex;
+  align-items: center;
+  justify-content: center;
+  background: transparent;
+  border: 1px solid var(--border-strong);
+  border-radius: var(--radius-sm);
+  color: var(--fg-dim);
+  cursor: pointer;
+  padding: 4px 8px;
+  line-height: 1;
+  height: 32px;
+  transition:
+    background 0.12s,
+    color 0.12s,
+    border-color 0.12s;
+}
+.composer-mic-btn:hover {
+  background: var(--bg-highlight);
+  color: var(--accent);
+  border-color: var(--accent-dim);
+}
+.composer-mic-btn:focus-visible {
+  outline: 2px solid var(--accent);
+  outline-offset: 1px;
+}
+.composer-mic-btn:disabled {
+  opacity: 0.5;
+  cursor: not-allowed;
+}
+.composer-mic-btn.is-recording {
+  color: var(--red, #f87171);
+  border-color: var(--red, #f87171);
+  animation: ts-mic-pulse 1.2s ease-in-out infinite;
+}
+@keyframes ts-mic-pulse {
+  0%,
+  100% {
+    opacity: 1;
+  }
+  50% {
+    opacity: 0.55;
+  }
+}
+.msg-tts-btn.is-playing {
+  color: var(--accent);
+}
+.msg-tts-btn.is-busy {
+  opacity: 0.6;
+  cursor: progress;
+}
+@media (prefers-reduced-motion: reduce) {
+  .composer-mic-btn.is-recording {
+    /* No pulse — keep an unmistakable static cue instead of the only signal
+       being a subtle icon swap. */
+    animation: none;
+    background: var(--red-glow);
+  }
+}
+/* Coarse-pointer touch target (WCAG 2.5.5) — matches the .msg-action-btn rule. */
+@media (hover: none) and (pointer: coarse) {
+  .composer-mic-btn {
+    min-width: 36px;
+    min-height: 36px;
+  }
+}
+
+/* Monochrome currentColor glyphs for the voice controls (CSS-mask technique),
+   so they theme and sit beside the line-art .icon-retry set rather than reading
+   as full-color, OS-dependent emoji. */
+.composer-mic-icon,
+.icon-mic,
+.icon-speaker {
+  display: inline-block;
+  width: 14px;
+  height: 14px;
+  background-color: currentColor;
+  -webkit-mask-repeat: no-repeat;
+  mask-repeat: no-repeat;
+  -webkit-mask-position: center;
+  mask-position: center;
+  -webkit-mask-size: contain;
+  mask-size: contain;
+}
+.icon-mic {
+  -webkit-mask-image: url("data:image/svg+xml,%3Csvg xmlns='http://www.w3.org/2000/svg' width='24' height='24' viewBox='0 0 24 24' fill='none' stroke='%23000' stroke-width='2' stroke-linecap='round' stroke-linejoin='round'%3E%3Cpath d='M12 2a3 3 0 0 0-3 3v6a3 3 0 0 0 6 0V5a3 3 0 0 0-3-3z'/%3E%3Cpath d='M19 10v1a7 7 0 0 1-14 0v-1'/%3E%3Cline x1='12' y1='19' x2='12' y2='22'/%3E%3C/svg%3E");
+  mask-image: url("data:image/svg+xml,%3Csvg xmlns='http://www.w3.org/2000/svg' width='24' height='24' viewBox='0 0 24 24' fill='none' stroke='%23000' stroke-width='2' stroke-linecap='round' stroke-linejoin='round'%3E%3Cpath d='M12 2a3 3 0 0 0-3 3v6a3 3 0 0 0 6 0V5a3 3 0 0 0-3-3z'/%3E%3Cpath d='M19 10v1a7 7 0 0 1-14 0v-1'/%3E%3Cline x1='12' y1='19' x2='12' y2='22'/%3E%3C/svg%3E");
+}
+.icon-speaker {
+  -webkit-mask-image: url("data:image/svg+xml,%3Csvg xmlns='http://www.w3.org/2000/svg' width='24' height='24' viewBox='0 0 24 24' fill='none' stroke='%23000' stroke-width='2' stroke-linecap='round' stroke-linejoin='round'%3E%3Cpolygon points='11 5 6 9 2 9 2 15 6 15 11 19 11 5'/%3E%3Cpath d='M15.54 8.46a5 5 0 0 1 0 7.07'/%3E%3Cpath d='M19.07 4.93a10 10 0 0 1 0 14.14'/%3E%3C/svg%3E");
+  mask-image: url("data:image/svg+xml,%3Csvg xmlns='http://www.w3.org/2000/svg' width='24' height='24' viewBox='0 0 24 24' fill='none' stroke='%23000' stroke-width='2' stroke-linecap='round' stroke-linejoin='round'%3E%3Cpolygon points='11 5 6 9 2 9 2 15 6 15 11 19 11 5'/%3E%3Cpath d='M15.54 8.46a5 5 0 0 1 0 7.07'/%3E%3Cpath d='M19.07 4.93a10 10 0 0 1 0 14.14'/%3E%3C/svg%3E");
+}
+/* Recording: a solid stop square (no mask) in place of the mic glyph. */
+.composer-mic-icon.icon-stop {
+  width: 11px;
+  height: 11px;
+  border-radius: 2px;
+  -webkit-mask-image: none;
+  mask-image: none;
+}
+
 .composer-input {
   flex: 1;
+  /* Allow the textarea to shrink below its content width so a long unbroken
+     token can't push the fixed-width row buttons (attach/mic/send) past the
+     edge of a narrow split pane. */
+  min-width: 0;
   min-height: 32px;
   max-height: 160px;
   font-family: var(--font-mono);
diff --git a/turnstone/shared_static/composer.js b/turnstone/shared_static/composer.js
index 66658146..c4138d52 100644
--- a/turnstone/shared_static/composer.js
+++ b/turnstone/shared_static/composer.js
@@ -177,6 +177,10 @@
       textRow = actionRow;
     }
 
+    // Supported surface for callers that need to add page-specific controls
+    // (e.g. a mic button) beside attach/send/stop.
+    this.actionsRowEl = actionRow;
+
     this._buildAttachButton(actionRow, opts);
     this._buildInput(textRow, opts);
     this._buildOptionsToggle(actionRow, opts);
diff --git a/turnstone/ui/static/app.js b/turnstone/ui/static/app.js
index b076211c..b1d639a2 100644
--- a/turnstone/ui/static/app.js
+++ b/turnstone/ui/static/app.js
@@ -9,6 +9,43 @@
 
 let _paneCounter = 0;
 
+// Voice-role availability comes from /v1/api/models (stt_default_alias /
+// tts_default_alias — present only when an audio-capable model role is
+// configured).  Memoized so all panes share a single fetch; affordances stay
+// hidden until it resolves.
+let _voiceRolesPromise = null;
+function getVoiceRoles() {
+  if (!_voiceRolesPromise) {
+    _voiceRolesPromise = authFetch("/v1/api/models")
+      .then((r) => (r.ok ? r.json() : {}))
+      .then((d) => ({
+        stt: !!(d && d.stt_default_alias),
+        tts: !!(d && d.tts_default_alias),
+      }))
+      .catch(() => ({ stt: false, tts: false }));
+  }
+  return _voiceRolesPromise;
+}
+
+// Visually-hidden polite live region for voice status (recording / playback)
+// so screen-reader users perceive state changes otherwise conveyed only by
+// color/icon. Errors go through showToast (already a live region). Single
+// shared node; clear-then-set so repeated identical messages re-announce.
+let _voiceStatusEl = null;
+function voiceAnnounce(msg) {
+  if (!_voiceStatusEl) {
+    _voiceStatusEl = document.createElement("div");
+    _voiceStatusEl.className = "sr-only";
+    _voiceStatusEl.setAttribute("role", "status");
+    _voiceStatusEl.setAttribute("aria-live", "polite");
+    document.body.appendChild(_voiceStatusEl);
+  }
+  _voiceStatusEl.textContent = "";
+  window.setTimeout(() => {
+    if (_voiceStatusEl) _voiceStatusEl.textContent = msg;
+  }, 30);
+}
+
 class Pane {
   constructor(wsId) {
     this.id = "p" + ++_paneCounter;
@@ -35,6 +72,21 @@ class Pane {
     this._cancelTimeout = null;
     this._forceTimeout = null;
     this._pendingEditSend = null;
+    // Voice I/O (mic STT + per-message TTS playback)
+    this._voiceRoles = { stt: false, tts: false };
+    this._micBtn = null;
+    this._micIcon = null;
+    this._micDenied = false;
+    this._recorder = null;
+    this._recordingStream = null;
+    this._isRecording = false;
+    this._discardRecording = false;
+    this._recordAborted = false;
+    this._recordTimer = null;
+    this._recordStartMs = 0;
+    this._ttsAudio = null;
+    this._ttsBtnActive = null;
+    this._ttsSeq = 0;
     this._createDOM();
   }
 
@@ -48,6 +100,8 @@ class Pane {
     this._pendingEditSend = null;
     this.inputEl.disabled = false;
     this.attachments.clearChips();
+    this._stopRecording(true);
+    this._stopTTS();
   }
 
   updateWsName() {
@@ -73,6 +127,8 @@ class Pane {
       this.evtSource.close();
       this.evtSource = null;
     }
+    this._stopRecording(true);
+    this._stopTTS();
   }
 
   // composer.setBusy runs unconditionally so the Stop button label /
@@ -88,6 +144,16 @@ class Pane {
     const edge = next !== this.busy;
     this.busy = next;
     if (edge && !next) this.queue.onIdleEdge();
+    // The mic produces composer input the user can only send when idle — keep
+    // it in lockstep with the send button (which composer.setBusy gates).
+    if (this._micBtn && !this._micDenied) {
+      if (next) {
+        this._stopRecording(true); // abandon any in-flight recording
+        this._micBtn.disabled = true;
+      } else if (!this._micBtn.classList.contains("is-busy")) {
+        this._micBtn.disabled = false;
+      }
+    }
   }
 
   showEmptyState() {
@@ -672,6 +738,14 @@ class Pane {
         }
       },
     });
+
+    // Voice input: mic button, hidden until the STT role is confirmed
+    // available (so it never appears when voice isn't configured).
+    this._buildMicButton();
+    getVoiceRoles().then((roles) => {
+      this._voiceRoles = roles;
+      if (this._micBtn) this._micBtn.style.display = roles.stt ? "" : "none";
+    });
   }
 
   connectSSE(wsId) {
@@ -1350,6 +1424,364 @@ class Pane {
     bar.insertBefore(btn, bar.firstChild);
   }
 
+  // -------------------------------------------------------------------------
+  // Voice I/O: microphone dictation (STT) + per-message playback (TTS)
+  // -------------------------------------------------------------------------
+
+  _buildMicButton() {
+    if (!this.composer || !this.composer.actionsRowEl) return;
+    const btn = document.createElement("button");
+    btn.type = "button";
+    btn.className = "composer-mic-btn";
+    btn.style.display = "none"; // revealed once the STT role is confirmed
+    btn.title = "Record speech to text";
+    btn.setAttribute("aria-label", "Record speech to text");
+    btn.setAttribute("aria-pressed", "false");
+    const icon = document.createElement("span");
+    icon.className = "composer-mic-icon icon-mic";
+    icon.setAttribute("aria-hidden", "true");
+    btn.appendChild(icon);
+    this._micIcon = icon;
+    btn.addEventListener("click", () => this._toggleRecording());
+    this.composer.actionsRowEl.insertBefore(btn, this.sendBtn || null);
+    this._micBtn = btn;
+  }
+
+  _syncMicButton() {
+    if (!this._micBtn) return;
+    const rec = !!this._isRecording;
+    this._micBtn.classList.toggle("is-recording", rec);
+    this._micBtn.setAttribute("aria-pressed", rec ? "true" : "false");
+    if (this._micIcon) {
+      this._micIcon.classList.toggle("icon-stop", rec);
+      this._micIcon.classList.toggle("icon-mic", !rec);
+    }
+    const label = rec
+      ? "Stop recording and transcribe"
+      : "Record speech to text";
+    this._micBtn.title = label;
+    this._micBtn.setAttribute("aria-label", label);
+  }
+
+  _toggleRecording() {
+    if (this._isRecording) {
+      this._stopRecording(false);
+    } else {
+      this._startRecording();
+    }
+  }
+
+  _startRecording() {
+    if (this._isRecording || this.busy) return;
+    if (
+      !navigator.mediaDevices ||
+      !navigator.mediaDevices.getUserMedia ||
+      typeof MediaRecorder === "undefined"
+    ) {
+      showToast("Microphone capture is not supported in this browser", "error");
+      return;
+    }
+    // Synchronous abort latch: if teardown (reset / pane switch) runs while the
+    // permission prompt is open, the stream the promise later hands us must be
+    // stopped instead of going hot after teardown.
+    this._recordAborted = false;
+    navigator.mediaDevices
+      .getUserMedia({ audio: true })
+      .then((stream) => {
+        if (this._recordAborted) {
+          stream.getTracks().forEach((t) => t.stop());
+          return;
+        }
+        this._recordingStream = stream;
+        let mimeType = "";
+        const candidates = [
+          "audio/webm;codecs=opus",
+          "audio/webm",
+          "audio/ogg;codecs=opus",
+          "audio/mp4",
+        ];
+        for (let i = 0; i < candidates.length; i++) {
+          if (
+            MediaRecorder.isTypeSupported &&
+            MediaRecorder.isTypeSupported(candidates[i])
+          ) {
+            mimeType = candidates[i];
+            break;
+          }
+        }
+        const rec = mimeType
+          ? new MediaRecorder(stream, { mimeType })
+          : new MediaRecorder(stream);
+        const chunks = [];
+        rec.addEventListener("dataavailable", (e) => {
+          if (e.data && e.data.size) chunks.push(e.data);
+        });
+        rec.addEventListener("stop", () => {
+          this._teardownRecordingStream();
+          this._isRecording = false;
+          this._stopRecordTimer();
+          this._syncMicButton();
+          const discard = this._discardRecording;
+          this._discardRecording = false;
+          if (discard) return;
+          const blob = new Blob(chunks, {
+            type: rec.mimeType || mimeType || "audio/webm",
+          });
+          if (blob.size) this._uploadForTranscription(blob);
+        });
+        this._recorder = rec;
+        this._isRecording = true;
+        this._discardRecording = false;
+        this._startRecordTimer();
+        this._syncMicButton();
+        voiceAnnounce(
+          "Recording. Activate the microphone button again to stop.",
+        );
+        rec.start();
+      })
+      .catch(() => {
+        // Denied / hardware unavailable: leave a persistent disabled state with
+        // guidance — a hot button reads as dead once the browser blocks re-prompts.
+        this._teardownRecordingStream();
+        this._isRecording = false;
+        this._stopRecordTimer();
+        this._setMicDenied();
+      });
+  }
+
+  _setMicDenied() {
+    this._micDenied = true;
+    showToast("Microphone access was denied", "error");
+    this._syncMicButton();
+    if (this._micBtn) {
+      this._micBtn.disabled = true;
+      const msg =
+        "Microphone blocked — enable it in your browser's site settings";
+      this._micBtn.title = msg;
+      this._micBtn.setAttribute("aria-label", msg);
+    }
+  }
+
+  _startRecordTimer() {
+    this._recordStartMs = Date.now();
+    this._stopRecordTimer();
+    this._recordTimer = window.setInterval(() => this._tickRecordTimer(), 500);
+  }
+
+  _stopRecordTimer() {
+    if (this._recordTimer) {
+      window.clearInterval(this._recordTimer);
+      this._recordTimer = null;
+    }
+  }
+
+  _tickRecordTimer() {
+    if (!this._isRecording || !this._micBtn) return;
+    const secs = Math.max(
+      0,
+      Math.floor((Date.now() - this._recordStartMs) / 1000),
+    );
+    const mmss =
+      Math.floor(secs / 60) + ":" + String(secs % 60).padStart(2, "0");
+    this._micBtn.title = "Recording " + mmss + " — activate to stop";
+  }
+
+  _stopRecording(discard) {
+    this._discardRecording = !!discard;
+    this._recordAborted = true; // abort a getUserMedia still in flight
+    if (this._recorder && this._recorder.state !== "inactive") {
+      try {
+        this._recorder.stop();
+      } catch (e) {
+        /* already stopped */
+      }
+      return;
+    }
+    this._teardownRecordingStream();
+    this._stopRecordTimer();
+    if (this._isRecording) {
+      this._isRecording = false;
+      this._syncMicButton();
+    }
+  }
+
+  _teardownRecordingStream() {
+    if (this._recordingStream) {
+      this._recordingStream.getTracks().forEach((t) => t.stop());
+      this._recordingStream = null;
+    }
+    this._recorder = null;
+  }
+
+  _uploadForTranscription(blob) {
+    if (!this.wsId) return;
+    const ext =
+      blob.type.indexOf("ogg") !== -1
+        ? "ogg"
+        : blob.type.indexOf("mp4") !== -1
+          ? "mp4"
+          : "webm";
+    const fd = new FormData();
+    fd.append("audio", blob, "speech." + ext);
+    if (this._micBtn) {
+      this._micBtn.disabled = true;
+      this._micBtn.classList.add("is-busy");
+    }
+    voiceAnnounce("Transcribing…");
+    authFetch(
+      "/v1/api/workstreams/" +
+        encodeURIComponent(this.wsId) +
+        "/speech-to-text",
+      { method: "POST", body: fd },
+    )
+      .then((r) => r.json().then((body) => ({ ok: r.ok, body })))
+      .then((res) => {
+        if (!res.ok) {
+          showToast(
+            (res.body && res.body.error) || "Transcription failed",
+            "error",
+          );
+          return;
+        }
+        const text = (res.body && res.body.transcript) || "";
+        if (text && this.inputEl) {
+          const cur = this.inputEl.value || "";
+          this.inputEl.value = cur
+            ? cur.replace(/\s*$/, "") + " " + text
+            : text;
+          // Drive the composer's auto-resize + send-enable listeners.
+          this.inputEl.dispatchEvent(new Event("input", { bubbles: true }));
+          this.inputEl.focus();
+          voiceAnnounce("Transcript added to message.");
+        }
+      })
+      .catch((err) => {
+        showToast(
+          "Transcription failed: " + (err && err.message ? err.message : err),
+          "error",
+        );
+      })
+      .finally(() => {
+        if (this._micBtn && !this._micDenied) {
+          this._micBtn.disabled = !!this.busy;
+          this._micBtn.classList.remove("is-busy");
+        }
+      });
+  }
+
+  _addTtsAction(el) {
+    let bar = el.querySelector(".msg-actions");
+    if (!bar) {
+      bar = document.createElement("div");
+      bar.className = "msg-actions";
+      bar.setAttribute("role", "toolbar");
+      bar.setAttribute("aria-label", "Message actions");
+      el.appendChild(bar);
+    }
+    if (bar.querySelector(".msg-tts-btn")) return; // already added
+    const btn = document.createElement("button");
+    btn.className = "msg-action-btn msg-tts-btn";
+    btn.title = "Play response aloud";
+    btn.setAttribute("aria-label", "Play response aloud");
+    btn.setAttribute("aria-pressed", "false");
+    const icon = document.createElement("span");
+    icon.className = "icon-speaker";
+    icon.setAttribute("aria-hidden", "true");
+    btn.appendChild(icon);
+    btn.addEventListener("click", (e) => {
+      e.stopPropagation();
+      this._playMessageTTS(el, btn);
+    });
+    bar.appendChild(btn);
+  }
+
+  // Strip code blocks / inline code / rendered math so TTS doesn't read source
+  // or KaTeX accessibility text out character-by-character.
+  _extractSpeakableText(bodyEl) {
+    const clone = bodyEl.cloneNode(true);
+    clone
+      .querySelectorAll("pre, code, .katex, .katex-display")
+      .forEach((n) =>
+        n.replaceWith(document.createTextNode(" (code omitted) ")),
+      );
+    return (clone.textContent || "").replace(/\s+/g, " ").trim();
+  }
+
+  _playMessageTTS(el, btn) {
+    // Toggle: clicking the active button (or any while playing) stops first.
+    if (this._ttsAudio) {
+      const wasThis = this._ttsBtnActive === btn;
+      this._stopTTS();
+      if (wasThis) return;
+    }
+    const bodyEl = el.querySelector(".msg-body") || el;
+    const text = this._extractSpeakableText(bodyEl);
+    if (!text) return;
+    // Serialize: a monotonic token guards against an earlier (slower) request
+    // resolving after a newer one — which would double-play and leak the blob.
+    const token = ++this._ttsSeq;
+    btn.classList.add("is-busy");
+    btn.disabled = true;
+    authFetch("/v1/api/tts", {
+      method: "POST",
+      headers: { "Content-Type": "application/json" },
+      body: JSON.stringify({ text }),
+    })
+      .then((r) => {
+        if (!r.ok) {
+          return r.json().then((b) => {
+            throw new Error((b && b.error) || "Speech synthesis failed");
+          });
+        }
+        return r.blob();
+      })
+      .then((audioBlob) => {
+        const url = URL.createObjectURL(audioBlob);
+        if (token !== this._ttsSeq) {
+          URL.revokeObjectURL(url); // superseded — don't play or leak
+          return;
+        }
+        const audio = new Audio(url);
+        this._ttsAudio = audio;
+        this._ttsBtnActive = btn;
+        btn.classList.add("is-playing");
+        btn.setAttribute("aria-pressed", "true");
+        voiceAnnounce("Playing response.");
+        audio.addEventListener("ended", () => this._stopTTS());
+        audio.addEventListener("error", () => this._stopTTS());
+        audio.play().catch(() => this._stopTTS());
+      })
+      .catch((err) => {
+        showToast(
+          err && err.message ? err.message : "Speech synthesis failed",
+          "error",
+        );
+      })
+      .finally(() => {
+        btn.classList.remove("is-busy");
+        btn.disabled = false;
+      });
+  }
+
+  _stopTTS() {
+    this._ttsSeq++; // invalidate any in-flight request
+    if (this._ttsAudio) {
+      try {
+        this._ttsAudio.pause();
+      } catch (e) {
+        /* ignore */
+      }
+      const src = this._ttsAudio.src || "";
+      if (src.indexOf("blob:") === 0) URL.revokeObjectURL(src);
+      this._ttsAudio = null;
+    }
+    if (this._ttsBtnActive) {
+      this._ttsBtnActive.classList.remove("is-playing");
+      this._ttsBtnActive.setAttribute("aria-pressed", "false");
+      this._ttsBtnActive = null;
+    }
+  }
+
   _retryLast() {
     if (this.busy) return;
     // Path-keyed retry (#549). Truncation + re-dispatch happen
@@ -1836,7 +2268,11 @@ class Pane {
     }
     const assistants = this.messagesEl.querySelectorAll(".msg.assistant");
     if (assistants.length) {
-      this._addRetryAction(assistants[assistants.length - 1]);
+      const lastAssistant = assistants[assistants.length - 1];
+      this._addRetryAction(lastAssistant);
+      if (this._voiceRoles && this._voiceRoles.tts) {
+        this._addTtsAction(lastAssistant);
+      }
     }
   }