Adding real-time streaming transcription to an async STT pipeline: a build guide
Blog post from Gladia
The guide describes how to add real-time speech-to-text to an existing asynchronous transcription system through a hybrid architecture that streams audio to Gladia’s Solaria-1 model for low-latency partial and final transcripts while simultaneously buffering raw audio for Solaria-3 post-call processing with diarization, entity extraction, and analytics. It emphasizes WebSocket connection management, correct audio configuration, secure token-based authentication, retry and fallback behavior, packet resequencing, duplicate-event prevention, and proper end-of-stream handling to avoid data loss. Partial transcripts should support live interfaces, while only committed final segments should trigger downstream LLM, CRM, or analytics workflows; VAD endpointing settings must be tuned to balance fast turn-taking against premature commits. The article recommends monitoring latency, connection failures, queue depth, and end-to-end processing performance, rolling out streaming through session-level feature flags, and retaining the asynchronous path as a resilient source of complete records. It positions streaming for applications such as live agent assistance, compliance alerts, voice agents, and captioning, while asynchronous transcription remains better suited to post-call summaries, quality assurance, and speaker attribution.
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