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https://github.com/pchuan98/codex.git
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Add per-session realtime model and version overrides (#24999)
## Why Clients need to select a realtime session configuration for an individual start without rewriting persisted configuration or restarting the app-server process. ## What Changed - Add optional `model` and `version` fields to `thread/realtime/start` - Forward those optional values through the realtime start operation and apply them only for that session - Preserve existing configured/default behavior when the new fields are omitted - Update generated protocol schema and app-server documentation ## Validation - Added/updated protocol serialization coverage for the new optional request fields - Added focused core coverage for a session override taking precedence over configured realtime selection - Added focused app-server coverage that a request override reaches the realtime WebSocket handshake
This commit is contained in:
guinness-oai
@@ -162,7 +162,7 @@ Example with notification opt-out:
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- `thread/inject_items` — append raw Responses API items to a loaded thread’s model-visible history without starting a user turn; returns `{}` on success.
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- `turn/steer` — add user input to an already in-flight regular turn without starting a new turn; returns the active `turnId` that accepted the input. `clientUserMessageId` is optional; when supplied, the corresponding `userMessage` item echoes it as `clientId`. Review and manual compaction turns reject `turn/steer`.
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- `turn/interrupt` — request cancellation of an in-flight turn by `(thread_id, turn_id)`; success is an empty `{}` response and the turn finishes with `status: "interrupted"`.
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- `thread/realtime/start` — start a thread-scoped realtime session (experimental); pass `outputModality: "text"` or `outputModality: "audio"` to choose model output, returns `{}` and streams `thread/realtime/*` notifications. Omit `transport` for the websocket transport, or pass `{ "type": "webrtc", "sdp": "..." }` to create a WebRTC session from a browser-generated SDP offer; the remote answer SDP is emitted as `thread/realtime/sdp`.
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- `thread/realtime/start` — start a thread-scoped realtime session (experimental); pass `outputModality: "text"` or `outputModality: "audio"` to choose model output, and optionally pass `model` and `version` to override configured realtime selection for this session only. Returns `{}` and streams `thread/realtime/*` notifications. Omit `transport` for the websocket transport, or pass `{ "type": "webrtc", "sdp": "..." }` to create a WebRTC session from a browser-generated SDP offer; the remote answer SDP is emitted as `thread/realtime/sdp`.
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- `thread/realtime/appendAudio` — append an input audio chunk to the active realtime session (experimental); returns `{}`.
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- `thread/realtime/appendText` — append text input to the active realtime session (experimental); returns `{}`.
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- `thread/realtime/stop` — stop the active realtime session for the thread (experimental); returns `{}`.
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@@ -837,6 +837,8 @@ Then send `offer.sdp` to app-server. Core uses `experimental_realtime_ws_backend
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Omit `prompt` to use Codex's default realtime backend prompt. Send `prompt: null` or
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`prompt: ""` when the session should start without that default backend prompt.
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Clients may also pass `model` and `version` on `thread/realtime/start` to select a
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different realtime session configuration without changing thread or user config.
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```javascript
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await pc.setRemoteDescription({
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@@ -934,6 +934,7 @@ impl TurnRequestProcessor {
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request_id,
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thread.as_ref(),
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Op::RealtimeConversationStart(ConversationStartParams {
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model: params.model,
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output_modality: params.output_modality,
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prompt: params.prompt,
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realtime_session_id: params.realtime_session_id,
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@@ -945,6 +946,7 @@ impl TurnRequestProcessor {
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ConversationStartTransport::Webrtc { sdp }
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}
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}),
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version: params.version,
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voice: params.voice,
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}),
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)
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@@ -80,10 +80,12 @@ async fn realtime_conversation_start_requires_experimental_api_capability() -> R
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let request_id = mcp
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.send_thread_realtime_start_request(ThreadRealtimeStartParams {
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thread_id: "thr_123".to_string(),
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model: None,
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output_modality: RealtimeOutputModality::Audio,
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prompt: Some(Some("hello".to_string())),
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realtime_session_id: None,
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transport: None,
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version: None,
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voice: None,
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})
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.await?;
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@@ -186,12 +188,14 @@ async fn realtime_webrtc_start_requires_experimental_api_capability() -> Result<
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let request_id = mcp
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.send_thread_realtime_start_request(ThreadRealtimeStartParams {
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thread_id: "thr_123".to_string(),
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model: None,
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output_modality: RealtimeOutputModality::Audio,
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prompt: Some(Some("hello".to_string())),
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realtime_session_id: None,
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transport: Some(ThreadRealtimeStartTransport::Webrtc {
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sdp: "v=offer\r\n".to_string(),
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}),
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version: None,
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voice: None,
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})
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.await?;
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@@ -311,12 +311,14 @@ impl RealtimeE2eHarness {
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.mcp
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.send_thread_realtime_start_request(ThreadRealtimeStartParams {
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thread_id: self.thread_id.clone(),
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model: None,
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output_modality: RealtimeOutputModality::Audio,
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prompt: Some(Some("backend prompt".to_string())),
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realtime_session_id: None,
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transport: Some(ThreadRealtimeStartTransport::Webrtc {
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sdp: offer_sdp.to_string(),
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}),
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version: None,
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voice: None,
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})
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.await?;
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@@ -556,10 +558,12 @@ async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
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let start_request_id = mcp
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.send_thread_realtime_start_request(ThreadRealtimeStartParams {
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thread_id: thread_start.thread.id.clone(),
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model: Some("realtime-treatment-model".to_string()),
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output_modality: RealtimeOutputModality::Audio,
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prompt: None,
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realtime_session_id: None,
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transport: None,
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version: None,
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voice: Some(RealtimeVoice::Cedar),
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})
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.await?;
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@@ -588,6 +592,10 @@ async fn realtime_conversation_streams_v2_notifications() -> Result<()> {
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startup_context_request.body_json()["session"]["audio"]["output"]["voice"],
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"cedar"
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);
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assert_eq!(
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realtime_server.single_handshake().uri(),
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"/v1/realtime?model=realtime-treatment-model"
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);
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assert_eq!(
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startup_context_request.body_json()["session"]["output_modalities"],
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json!(["audio"])
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@@ -805,10 +813,12 @@ async fn realtime_text_output_modality_requests_text_output_and_final_transcript
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let start_request_id = mcp
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.send_thread_realtime_start_request(ThreadRealtimeStartParams {
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thread_id: thread_start.thread.id.clone(),
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model: None,
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output_modality: RealtimeOutputModality::Text,
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prompt: None,
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realtime_session_id: None,
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transport: None,
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version: None,
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voice: None,
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})
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.await?;
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@@ -979,10 +989,12 @@ async fn realtime_conversation_stop_emits_closed_notification() -> Result<()> {
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let start_request_id = mcp
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.send_thread_realtime_start_request(ThreadRealtimeStartParams {
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thread_id: thread_start.thread.id.clone(),
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model: None,
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output_modality: RealtimeOutputModality::Audio,
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prompt: Some(Some("backend prompt".to_string())),
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realtime_session_id: None,
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transport: None,
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version: None,
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voice: None,
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})
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.await?;
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@@ -1076,12 +1088,14 @@ async fn realtime_webrtc_start_emits_sdp_notification() -> Result<()> {
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let start_request_id = mcp
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.send_thread_realtime_start_request(ThreadRealtimeStartParams {
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thread_id: thread_id.clone(),
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model: None,
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output_modality: RealtimeOutputModality::Audio,
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prompt: Some(Some("backend prompt".to_string())),
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realtime_session_id: None,
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transport: Some(ThreadRealtimeStartTransport::Webrtc {
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sdp: "v=offer\r\n".to_string(),
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}),
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version: None,
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voice: None,
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})
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.await?;
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@@ -1991,12 +2005,14 @@ async fn realtime_webrtc_start_surfaces_backend_error() -> Result<()> {
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let start_request_id = mcp
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.send_thread_realtime_start_request(ThreadRealtimeStartParams {
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thread_id: thread_start.thread.id,
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model: None,
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output_modality: RealtimeOutputModality::Audio,
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prompt: Some(Some("backend prompt".to_string())),
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realtime_session_id: None,
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transport: Some(ThreadRealtimeStartTransport::Webrtc {
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sdp: "v=offer\r\n".to_string(),
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}),
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version: None,
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voice: None,
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})
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.await?;
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@@ -2050,10 +2066,12 @@ async fn realtime_conversation_requires_feature_flag() -> Result<()> {
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let start_request_id = mcp
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.send_thread_realtime_start_request(ThreadRealtimeStartParams {
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thread_id: thread_start.thread.id.clone(),
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model: None,
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output_modality: RealtimeOutputModality::Audio,
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prompt: Some(Some("backend prompt".to_string())),
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realtime_session_id: None,
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transport: None,
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version: None,
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voice: None,
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})
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.await?;
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