Nari Labs is opening up a streaming Speech-to-Text endpoint aimed at developers building real-time voice applications like agents, meeting tools, and live captioning. The service runs Qwen3-ASR 1.7B, an open-source ASR model developed by the Qwen team at Alibaba Cloud that supports multilingual speech, music, and song recognition along with language detection and timestamp prediction , served on the company's own inference engine. The pitch on this launch is a 40 ms time-to-final-segment and pricing of $0.06 per hour of audio, positioned as the fastest and cheapest streaming transcription endpoint available, with a Top 3 placement on the Coval accuracy benchmarks.
The launch matters because streaming ASR latency directly caps how natural a voice agent can feel, and current hosted options for Qwen3-ASR sit much higher on the latency curve. For reference, Baseten's own Qwen3-ASR-1.7B streaming deployment reports P95 finalization latency under 0.50 seconds across up to 64 concurrent streams on RTX 6000 and H100 GPUs , so a 40 ms TTFS claim on the same underlying model is a meaningful step for teams stitching together listen-think-speak loops.
Nari Labs is a two-person startup that introduced Dia, a 1.6 billion parameter text-to-speech model designed to produce naturalistic dialogue directly from text prompts , and has since been working down the latency stack across the voice pipeline. The team, led by Toby Kim, previously shipped Nari Qwen3-TTS, a high-performance single-H100 serving implementation of Qwen3-TTS 1.7B CustomVoice that exposes streaming and non-streaming speech generation over HTTP plus WebSocket-based input streaming, targeting sub-50 ms time-to-first-audio at 10 requests per second . This ASR endpoint is the listening half of that same effort, giving builders a matched-latency pair for full-duplex voice systems.
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Priya VaidyaSep 12, 2026
40ms TTFS is the kind of number that either changes voice agents forever or gets an asterisk buried in the docs. Which one is it?
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Tomasz KowalskiSep 12, 2026
$0.06/hr streaming ASR makes me want to see the gross margin math on a napkin. Are you eating cost to grab share or is the inference engine actually that lean?
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kenjiSep 12, 2026
The tweet buried the lede by putting pricing second. Lead with the 3x claim next time and watch the quote tweets do the work.
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Adaobi N.Sep 12, 2026
Qwen3-ASR under the hood is a spicy choice given where the weights come from. Hoping the data residency page is more than a footer link.
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Milo ForsbergSep 12, 2026
One of my portfolio voice agents was begging for sub-100ms last month. Sliding into DMs, this is exactly the primitive they needed.
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Rhea R.Sep 12, 2026
Nice, now do diarization, code-switching, and a webhook for silence detection. I promise I'll stop after that (I won't).
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kwabenaSep 12, 2026
Any ASR eng candidates watching this drop and updating their resumes? I have a role that would love a Coval top-3 name on the team.
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Nari Labs is opening up a streaming Speech-to-Text endpoint aimed at developers building real-time voice applications like agents, meeting tools, and live captioning. The service runs Qwen3-ASR 1.7B, an open-source ASR model developed by the Qwen team at Alibaba Cloud that supports multilingual speech, music, and song recognition along with language detection and timestamp prediction , served on the company's own inference engine. The pitch on this launch is a 40 ms time-to-final-segment and pricing of $0.06 per hour of audio, positioned as the fastest and cheapest streaming transcription endpoint available, with a Top 3 placement on the Coval accuracy benchmarks. The launch matters because streaming ASR latency directly caps how natural a voice agent can feel, and current hosted options for Qwen3-ASR sit much higher on the latency curve. For reference, Baseten's own Qwen3-ASR-1.7B streaming deployment reports P95 finalization latency under 0.50 seconds across up to 64 concurrent streams on RTX 6000 and H100 GPUs , so a 40 ms TTFS claim on the same underlying model is a meaningful step for teams stitching together listen-think-speak loops. Nari Labs is a two-person startup that introduced Dia, a 1.6 billion parameter text-to-speech model designed to produce naturalistic dialogue directly from text prompts , and has since been working down the latency stack across the voice pipeline. The team, led by Toby Kim, previously shipped Nari Qwen3-TTS, a high-performance single-H100 serving implementation of Qwen3-TTS 1.7B CustomVoice that exposes streaming and non-streaming speech generation over HTTP plus WebSocket-based input streaming, targeting sub-50 ms time-to-first-audio at 10 requests per second . This ASR endpoint is the listening half of that same effort, giving builders a matched-latency pair for full-duplex voice systems.
Comments (7)
40ms TTFS is the kind of number that either changes voice agents forever or gets an asterisk buried in the docs. Which one is it?
$0.06/hr streaming ASR makes me want to see the gross margin math on a napkin. Are you eating cost to grab share or is the inference engine actually that lean?
The tweet buried the lede by putting pricing second. Lead with the 3x claim next time and watch the quote tweets do the work.
Qwen3-ASR under the hood is a spicy choice given where the weights come from. Hoping the data residency page is more than a footer link.
One of my portfolio voice agents was begging for sub-100ms last month. Sliding into DMs, this is exactly the primitive they needed.
Nice, now do diarization, code-switching, and a webhook for silence detection. I promise I'll stop after that (I won't).
Any ASR eng candidates watching this drop and updating their resumes? I have a role that would love a Coval top-3 name on the team.