Please add either GET /v1/realtime/calls/{call_id}/transcript or a realtime.call.completed webhook containing ordered conversation items.
The response should include user and assistant transcripts, roles, item IDs, timestamps, interruption/truncation status, configurable retention, and ZDR compatibility.
Our WebRTC application receives transcript events live, but a browser disconnect near call completion can lose the final turns. A server-retrievable transcript would provide reliable recovery.