Reconnect WebRTC Realtime sideband transports (#39257)

## What changed

- Reconnect frameless bidirectional WebRTC sideband sockets after unexpected transport loss with capped exponential backoff.
- Preserve pending text or handoff output and bounded transcript state across reconnects.
- Treat `404 Not Found` and `410 Gone` handshake responses as terminal session completion, and preserve HTTP status information from failed WebSocket handshakes.
- Bound Realtime delegation input and transcript fields while retaining the most relevant portion of each.

## Testing

- Add unit coverage for reconnect backoff, terminal status handling, transport error classification, transcript continuity and truncation, and pending outbound messages.
- Add an end-to-end WebRTC conversation test covering an unclean disconnect, reconnect, transcript reconciliation, outbound progress during sustained inbound traffic, and terminal session completion.

GitOrigin-RevId: 544fd7e542c3434d0bb42cfbcc7143604419a037

codex-rs/codex-api/tests/realtime_websocket_e2e.rs

@@ -10,6 +10,7 @@ use codex_api::RealtimeEventParser;
 use codex_api::RealtimeOutputModality;
 use codex_api::RealtimeSessionConfig;
 use codex_api::RealtimeSessionMode;
+use codex_api::RealtimeTranscriptState;
 use codex_api::RealtimeWebsocketClient;
 use codex_api::RetryConfig;
 use codex_protocol::protocol::RealtimeHandoffRequested;
@@ -263,6 +264,7 @@ async fn realtime_ws_connect_webrtc_sideband_retries_join_until_server_is_availa
             "rtc_test",
             HeaderMap::new(),
             HeaderMap::new(),
+            RealtimeTranscriptState::default(),
         )
         .await
         .expect("connect on retry");