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从零构建企业级实时语音代理:技术教程

Building Enterprise Realtime Voice Agents from Scratch: A Technical Tutorial

Salesforce AI Research(Salesforce AI研究院)
Jielin Qiu, Zixiang Chen, Liangwei Yang, Ming Zhu, Zhiwei Liu, Juntao Tan, Wenting Zhao, Rithesh Murthy, Roshan Ram, Akshara Prabhakar, Shelby Heinecke, Caiming… 展开作者

Jielin Qiu, Zixiang Chen, Liangwei Yang, Ming Zhu, Zhiwei Liu, Juntao Tan, Wenting Zhao, Rithesh Murthy, Roshan Ram, Akshara Prabhakar, Shelby Heinecke, Caiming Xiong, Silvio Savarese, Huan Wang

AI总结:

本文介绍如何从零构建企业级实时语音代理,通过Deepgram、vLLM和ElevenLabs实现端到端流程,展示最佳实践与代码实现。

摘要:

我们提供了一项技术教程,介绍如何从基础原理构建企业级实时语音代理。尽管端到端的语音到语音模型可能最终提供最佳延迟,但完全自托管的端到端解决方案尚未可用。我们评估了最接近的候选者Qwen3-Omni,其云服务DashScope实时API实现约702ms的音频到音频延迟,但无法自托管;其本地vLLM部署仅支持Thinker(从音频生成文本,516ms),不支持Talker(音频合成);其本地Transformers部署运行完整流程但延迟约146秒,远不足以实现实时。因此,STT→LLM→TTS的级联流式管道仍然是自托管实时语音代理的实用架构。本文通过Deepgram(流式STT)、vLLM服务的LLMs(流式文本生成)和ElevenLabs(流式TTS)构建完整语音代理,实现了755ms(最佳情况729ms)的首次音频时间,支持完整功能调用。本文发布完整的代码库作为9章渐进教程,为每个组件提供经过测试的代码。

英文摘要:

We present a technical tutorial for building enterprise-grade realtime voice agents from first principles. While end-to-end speech-to-speech models may ultimately provide the best latency for voice agents, fully self-hosted end-to-end solutions are not yet available. We evaluate the closest candidate, Qwen3-Omni, across three configurations: its cloud-only DashScope Realtime API achieves $\sim$702ms audio-to-audio latency with streaming, but is not self-hostable; its local vLLM deployment supports only the Thinker (text generation from audio, 516ms), not the Talker (audio synthesis); and its local Transformers deployment runs the full pipeline but at $\sim$146s -- far too slow for realtime. The cascaded streaming pipeline (STT $\rightarrow$ LLM $\rightarrow$ TTS) therefore remains the practical architecture for self-hosted realtime voice agents, and the focus of this tutorial. We build a complete voice agent using Deepgram (streaming STT), vLLM-served LLMs with function calling (streaming text generation), and ElevenLabs (streaming TTS), achieving a measured time-to-first-audio of 755ms (best case 729ms) with full function calling support. We release the full codebase as a 9-chapter progressive tutorial with working, tested code for every component.

机构、总结、中文摘要均由AI分析而来,可能存在错误,请以论文正文为准。

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