We current have an eleven labs agent (In agents platform).We're wondering if we can reuse that sa | ElevenLabs | Page 1

fluid terraceBOT Oct 18, 2025, 3:47 AM

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AI-generated Answer for kishore_vb

Short answer: not directly. An Agent is an orchestration that bundles ASR, an LLM, TTS, turn‑taking, tools, and knowledge base. The standalone TTS and STT APIs don’t accept an agent_id; they take voice_id (for TTS) or a transcription model_id (for STT). (elevenlabs.io)

What you can reuse

Voice and TTS model from your Agent: fetch them via Get Agent (look at conversation_config.tts.voice_id and conversation_config.tts.model_id), then call the regular TTS endpoint POST /v1/text-to-speech/:voice_id with that model_id. (elevenlabs.io)
Speech-to-Text independently: call POST /v1/speech-to-text with a supported transcription model such as scribe_v1. This runs outside the Agents runtime. (elevenlabs.io)

Implication

Agent-only behaviors (system prompt, tools, knowledge base, turn-taking) are part of the Agents runtime and are not applied when you call the plain TTS/STT endpoints, since those endpoints don’t take an agent_id and expose only voice_id/model_id parameters. (elevenlabs.io)

If you’d like, share your agent_id and I can point to the exact voice_id/model_id to plug into TTS/STT from the Get Agent response. (elevenlabs.io)

vague dock Oct 18, 2025, 3:50 AM

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what is ASR in agents platform and how is it dfferent from the stand alone TTS and STT

fluid terraceBOT Oct 18, 2025, 3:50 AM

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AI-generated Answer for kishore_vb

Short answer

In the Agents platform, ASR means Automatic Speech Recognition: the real‑time speech‑to‑text component that transcribes the caller’s audio so the agent can reason and reply. It’s one of the built-in blocks of an agent along with turn‑taking and TTS. (elevenlabs.io)

How Agents ASR works vs standalone STT and TTS

Agents ASR (inside Agents)
- Purpose: live transcription during a conversation.
- Behavior: runs over the Agents WebSocket; you stream user audio and receive live user_transcript events and VAD scores that drive turn‑taking. Configurable via the agent’s conversation_config.asr (e.g., provider, quality, user_input_audio_format). Pricing is optimized during long silences. (elevenlabs.io)
Standalone STT (Speech to Text API)
- Purpose: batch/file transcription, not the live conversational stack.
- Capabilities: Scribe v1 model with high‑accuracy transcription, word‑level timestamps and diarization; supports multi‑channel audio and optional webhooks for async delivery. Best suited for high‑accuracy (offline) use rather than real‑time. (elevenlabs.io)
Standalone TTS (Text to Speech API)
- Purpose: synthesize the agent’s replies into audio; not recognition.
- Usage: call the Text to Speech endpoint (or its streaming variant) directly to generate speech from text. (elevenlabs.io)

In short: Agents ASR is the real‑time, streaming speech‑to‑text piece embedded in the conversational pipeline (with turn‑taking and VAD) for live calls; standalone STT is a separate API for file transcription; and TTS is for generating speech output. (elevenlabs.io)

Sources:
Introduction - Conversational voice AI agents | ElevenLabs Documentation
Agent WebSockets | ElevenLabs Documentation
Speech to Text | ElevenLabs Documentation
Create speech | ElevenLabs Documentation
Introduction - Conversational voice AI agents | ElevenLabs Documentation

vague dock Oct 18, 2025, 3:53 AM

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If we were to move out of the agents platform, we need to implement turn-taking and VAD ourselves?

fluid terraceBOT Oct 18, 2025, 3:53 AM

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AI-generated Answer for kishore_vb

Short answer: yes.

Why:

In Agents, ElevenLabs provides a built‑in turn‑taking model and handles silence/voice‑activity gating for you. The docs list “Turn taking model” as a core building block and note that after 10s of silence the system reduces inference “until voice activity is detected again,” i.e., VAD‑driven behavior. (elevenlabs.io)
ElevenLabs’ own blog describes this as a “state‑of‑the‑art turn‑taking model” that decides when to speak or wait, based on real‑time cues like hesitations. (elevenlabs.io)
Their latency guide explicitly ties turn‑taking to a Voice Activity Detector (VAD) and explains why end‑of‑turn detection isn’t just “any silence,” reinforcing that this logic is part of the conversational stack. (elevenlabs.io)

What you’d need outside Agents:

Turn‑taking and barge‑in control: detect end‑of‑user turns and interruptions yourself. The standalone TTS APIs (HTTP/streaming/WebSockets) generate/stream audio but don’t provide turn‑management. (elevenlabs.io)
VAD/silence handling: implement your own VAD to decide when the user is speaking, when to pause/resume, and when a turn ends. The Agents platform’s silence/VAD behavior would no longer apply. (elevenlabs.io)
Real‑time ASR: the public Speech‑to‑Text (Scribe v1) is positioned for high‑accuracy batch use; the docs say a low‑latency real‑time version is “coming soon.” If you leave Agents today, you’d need your own real‑time ASR layer to replicate agent behavior. (elevenlabs.io)
Optional audio processing lost from Agents/WebRTC (echo cancellation, noise removal): if you relied on the Agents/WebRTC path, you’d need to handle these client‑side or via your own media stack. (elevenlabs.io)

So, moving off Agents to raw TTS/STT means you are responsible for turn‑taking, VAD, and related duplex/interrupt logic (plus real‑time ASR and any WebRTC audio processing, if needed). (elevenlabs.io)

Sources:
Introduction - Conversational voice AI agents | ElevenLabs Documentation
ElevenLabs Conversational AI 2.0 voice agents now live | ElevenLabs
How do you optimize latency for Conversational AI? | ElevenLabs
Streaming | ElevenLabs Documentation
Introduction - Conversational voice AI agents | ElevenLabs Documentation
Speech to Text | ElevenLabs Documentation
ElevenLabs Conversational AI now supports WebRTC | ElevenLabs
Introduction - Conversational voice AI agents | ElevenLabs Documentation

vague dock Oct 18, 2025, 4:00 AM

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Is the stand alone STT of eleven labs not optimized for realtime usage?

fluid terraceBOT Oct 18, 2025, 4:00 AM

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AI-generated Answer for kishore_vb

Short answer: Correct — the standalone Speech-to-Text (Scribe v1) is tuned for high‑accuracy batch transcription, not low‑latency real‑time use. ElevenLabs’ docs explicitly recommend Scribe v1 “when high‑accuracy transcription is required rather than real‑time,” and note a low‑latency real‑time version is “coming soon.” (elevenlabs.io)

If you need real-time today, ElevenLabs directs you to the Agents Platform, which includes a fine‑tuned ASR component plus a turn‑taking model for conversational use. (elevenlabs.io)

Sources:
Speech to Text | ElevenLabs Documentation
Introduction - Conversational voice AI agents | ElevenLabs Documentation

vague dock Oct 18, 2025, 4:02 AM

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Without Agents platform, eleven labs doesn't offer a way for real time STT?

fluid terraceBOT Oct 18, 2025, 4:02 AM

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AI-generated Answer for kishore_vb

Short answer: Correct. Outside the Agents platform, ElevenLabs does not currently offer a standalone real-time/streaming STT API.

The standalone Speech to Text (Scribe v1) is positioned for high‑accuracy batch/asynchronous transcription and the docs note a low‑latency real‑time version is “coming soon.” (elevenlabs.io)
Real‑time ASR is available within the Conversational AI Agents platform, which bundles ASR, TTS, and turn‑taking for live conversations. (elevenlabs.io)
ElevenLabs does provide real‑time TTS over WebSockets (separate from STT), if that helps for your use case. (elevenlabs.io)

Sources:
Speech to Text | ElevenLabs Documentation
Introduction - Conversational voice AI agents | ElevenLabs Documentation
Generate audio in real-time | ElevenLabs Documentation

vague dock Oct 18, 2025, 4:03 AM

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But the stand alone TTS is optimized for real time use?

fluid terraceBOT Oct 18, 2025, 4:03 AM

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AI-generated Answer for kishore_vb

Yes. ElevenLabs’ standalone Text-to-Speech is built for real-time use:

Low‑latency models: Flash v2.5 is explicitly designed for real‑time applications (~75 ms model inference); Turbo v2.5 offers a quality/latency balance. (elevenlabs.io)
Real‑time transport: TTS supports streaming via Server‑Sent Events (output streaming) and full bidirectional WebSocket streaming; use SSE when your full text is ready up‑front, and WebSockets when text arrives incrementally. (elevenlabs.io)
WebSocket TTS is optimized to start generating audio from partial text to minimize latency; ElevenLabs recommends Flash v2.5 for latency‑sensitive use. (help.elevenlabs.io)
Latency guidance: With Flash models over WebSockets, typical TTFB ranges are documented by region (e.g., ~150–200 ms in the US), noting 75 ms refers to model inference only and total latency depends on network/endpoint. (elevenlabs.io)
Note: The old optimize_streaming_latency parameter is deprecated; current best practices are model choice and streaming/WebSockets. (help.elevenlabs.io)

Sources:
Models | ElevenLabs Documentation
Latency optimization | ElevenLabs Documentation
What is the Text-to-Speech WebSockets API? – ElevenLabs
Latency optimization | ElevenLabs Documentation
Can I reduce API latency? – ElevenLabs

vague dock Oct 18, 2025, 6:36 AM

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List the steps managed by agents platform ASR

#We current have an eleven labs agent (In agents platform).We're wondering if we can reuse that sa