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-protocol/src/protocol/common.rs

@@ -1005,6 +1005,8 @@ server_notification_definitions! {
     ThreadRealtimeTranscriptUpdated => "thread/realtime/transcriptUpdated" (v2::ThreadRealtimeTranscriptUpdatedNotification),
     #[experimental("thread/realtime/outputAudio/delta")]
     ThreadRealtimeOutputAudioDelta => "thread/realtime/outputAudio/delta" (v2::ThreadRealtimeOutputAudioDeltaNotification),
+    #[experimental("thread/realtime/sdp")]
+    ThreadRealtimeSdp => "thread/realtime/sdp" (v2::ThreadRealtimeSdpNotification),
     #[experimental("thread/realtime/error")]
     ThreadRealtimeError => "thread/realtime/error" (v2::ThreadRealtimeErrorNotification),
     #[experimental("thread/realtime/closed")]
@@ -1761,6 +1763,7 @@ mod tests {
                 thread_id: "thr_123".to_string(),
                 prompt: "You are on a call".to_string(),
                 session_id: Some("sess_456".to_string()),
+                transport: None,
             },
         };
         assert_eq!(
@@ -1770,7 +1773,8 @@ mod tests {
                 "params": {
                     "threadId": "thr_123",
                     "prompt": "You are on a call",
-                    "sessionId": "sess_456"
+                    "sessionId": "sess_456",
+                    "transport": null
                 }
             }),
             serde_json::to_value(&request)?,
@@ -1850,6 +1854,7 @@ mod tests {
                 thread_id: "thr_123".to_string(),
                 prompt: "You are on a call".to_string(),
                 session_id: None,
+                transport: None,
             },
         };
         let reason = crate::experimental_api::ExperimentalApi::experimental_reason(&request);

codex-rs/app-server-protocol/src/protocol/v2.rs

@@ -3854,6 +3854,21 @@ pub struct ThreadRealtimeStartParams {
     pub prompt: String,
     #[ts(optional = nullable)]
     pub session_id: Option<String>,
+    #[ts(optional = nullable)]
+    pub transport: Option<ThreadRealtimeStartTransport>,
+}
+
+/// EXPERIMENTAL - transport used by thread realtime.
+#[derive(Serialize, Deserialize, Debug, Clone, PartialEq, JsonSchema, TS)]
+#[serde(tag = "type", rename_all = "camelCase")]
+#[ts(export_to = "v2/", tag = "type")]
+pub enum ThreadRealtimeStartTransport {
+    Websocket,
+    Webrtc {
+        /// SDP offer generated by a WebRTC RTCPeerConnection after configuring audio and the
+        /// realtime events data channel.
+        sdp: String,
+    },
 }
 
 /// EXPERIMENTAL - response for starting thread realtime.
@@ -3945,6 +3960,15 @@ pub struct ThreadRealtimeOutputAudioDeltaNotification {
     pub audio: ThreadRealtimeAudioChunk,
 }
 
+/// EXPERIMENTAL - emitted with the remote SDP for a WebRTC realtime session.
+#[derive(Serialize, Deserialize, Debug, Clone, PartialEq, JsonSchema, TS)]
+#[serde(rename_all = "camelCase")]
+#[ts(export_to = "v2/")]
+pub struct ThreadRealtimeSdpNotification {
+    pub thread_id: String,
+    pub sdp: String,
+}
+
 /// EXPERIMENTAL - emitted when thread realtime encounters an error.
 #[derive(Serialize, Deserialize, Debug, Clone, PartialEq, JsonSchema, TS)]
 #[serde(rename_all = "camelCase")]