Real-time latency for meeting transcription: latency budgets and live note-taking requirements

Real-time latency in meeting transcription is crucial for delivering responsive live note-taking experiences, requiring careful management of end-to-end delays from audio capture to display rendering. While asynchronous transcription provides higher accuracy and lower costs for post-meeting notes, real-time transcription must keep latency under 500ms to maintain user engagement during calls. This involves managing five key components: client audio chunking, network routing, STT model inference, post-processing, and client rendering. Many teams mistakenly focus solely on STT inference speed, overlooking the cumulative delays contributed by other stages. Effective latency management requires a comprehensive understanding of the pipeline, allowing teams to make informed trade-offs between real-time user experience and asynchronous accuracy. Furthermore, the choice of transcription workflow—real-time for live UX or asynchronous for final notes—depends on the specific use case, with each offering distinct advantages and costs. The challenge lies in balancing the need for immediate interim results during live interactions with the accuracy provided by batch processing for final transcripts, particularly in multilingual and complex audio environments.

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