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Reconnect WebRTC Realtime sideband transports (#39257)

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## 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
This commit is contained in:
Peter Bakkum
2026-08-18 18:41:40 +00:00
committed by copyberry
parent fa9a05f2d2
commit ecb8013dfa
11 changed files with 1046 additions and 215 deletions
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codex-rs/codex-api/src/lib.rs
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@@ -59,6 +59,7 @@ pub use crate::endpoint::RealtimeEventParser;
pub use crate::endpoint::RealtimeOutputModality;
pub use crate::endpoint::RealtimeSessionConfig;
pub use crate::endpoint::RealtimeSessionMode;
pub use crate::endpoint::RealtimeTranscriptState;
pub use crate::endpoint::RealtimeWebsocketClient;
pub use crate::endpoint::RealtimeWebsocketConnection;
pub use crate::endpoint::RealtimeWebsocketEvents;