Add WebRTC transport to realtime start (#16960)

Adds WebRTC startup to the experimental app-server
`thread/realtime/start` method with an optional transport enum. The
websocket path remains the default; WebRTC offers create the realtime
session through the shared start flow and emit the answer SDP via
`thread/realtime/sdp`.

---------

Co-authored-by: Codex <noreply@openai.com>

codex-rs/app-server/README.md

@@ -152,7 +152,7 @@ Example with notification opt-out:
 - `turn/start` — add user input to a thread and begin Codex generation; responds with the initial `turn` object and streams `turn/started`, `item/*`, and `turn/completed` notifications. For `collaborationMode`, `settings.developer_instructions: null` means "use built-in instructions for the selected mode".
 - `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. Review and manual compaction turns reject `turn/steer`.
 - `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"`.
-- `thread/realtime/start` — start a thread-scoped realtime session (experimental); returns `{}` and streams `thread/realtime/*` notifications.
+- `thread/realtime/start` — start a thread-scoped realtime session (experimental); 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`.
 - `thread/realtime/appendAudio` — append an input audio chunk to the active realtime session (experimental); returns `{}`.
 - `thread/realtime/appendText` — append text input to the active realtime session (experimental); returns `{}`.
 - `thread/realtime/stop` — stop the active realtime session for the thread (experimental); returns `{}`.
@@ -563,6 +563,51 @@ Invoke a plugin by including a UI mention token such as `@sample` in the text in
 } } }
 ```
 
+### Example: Start realtime with WebRTC
+
+Use `thread/realtime/start` with `transport.type: "webrtc"` when a browser or webview owns the `RTCPeerConnection` and app-server should create the server-side realtime session. The transport `sdp` must be the offer SDP produced by `RTCPeerConnection.createOffer()`, not a hand-written or minimal SDP string.
+
+The offer should include the media sections the client wants to negotiate. For the standard realtime UI flow, create the audio track/transceiver and the `oai-events` data channel before calling `createOffer()`:
+
+```javascript
+const pc = new RTCPeerConnection();
+
+audioElement.autoplay = true;
+pc.ontrack = (event) => {
+  audioElement.srcObject = event.streams[0];
+};
+
+const mediaStream = await navigator.mediaDevices.getUserMedia({ audio: true });
+pc.addTrack(mediaStream.getAudioTracks()[0], mediaStream);
+pc.createDataChannel("oai-events");
+
+const offer = await pc.createOffer();
+await pc.setLocalDescription(offer);
+```
+
+Then send `offer.sdp` to app-server. Core uses `experimental_realtime_ws_backend_prompt` for the backend instructions and the thread conversation id for the realtime session id. The start response is `{}`; the remote answer SDP arrives later as `thread/realtime/sdp` and should be passed to `setRemoteDescription()`:
+
+```json
+{ "method": "thread/realtime/start", "id": 40, "params": {
+    "threadId": "thr_123",
+    "prompt": "You are on a call.",
+    "sessionId": null,
+    "transport": { "type": "webrtc", "sdp": "v=0\r\no=..." }
+} }
+{ "id": 40, "result": {} }
+{ "method": "thread/realtime/sdp", "params": {
+    "threadId": "thr_123",
+    "sdp": "v=0\r\no=..."
+} }
+```
+
+```javascript
+await pc.setRemoteDescription({
+  type: "answer",
+  sdp: notification.params.sdp,
+});
+```
+
 ### Example: Interrupt an active turn
 
 You can cancel a running Turn with `turn/interrupt`.