what are the best practices for low latency voice ...
# ask-ai
r
what are the best practices for low latency voice agent.
t
Thanks for reaching out, rahul. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
To help you achieve low latency in your voice agents, here are some friendly best practices to consider: • Network Proximity: It’s really helpful to deploy your agents close to Speech-to-Text (STT), Text-to-Speech (TTS), and Large Language Model (LLM) services. This can significantly reduce network latency and make interactions smoother. • Provider Selection: Choosing AI services that are optimized for low latency, like Groq for STT and Cartesia for TTS, can make a big difference in performance. • Latency Metrics Monitoring: Keep an eye on metrics such as Time To First Token (TTFT) and Time To First Byte (TTFB). Aiming for around 300ms latency helps create natural, seamless conversations. LiveKit’s Agents Metrics API is a great tool to help you monitor these. • Optimized Processing: Techniques like chunked processing (for example, 50ms audio chunks), running STT, LLM, and TTS concurrently, and preemptive generation to overlap inference with audio input can really boost responsiveness. • Real-Time Protocols: Using WebRTC for client-server audio transmission and WebSockets for server-to-server communication with real-time LLMs ensures your data flows efficiently. • Agent Design: LiveKit’s
AgentSession
is fantastic for turn detection, and adjusting endpointing delays can help you find the perfect balance between responsiveness and accuracy. If you’d like, I can also share practical examples like Groq’s voice assistant and how OpenAI’s Realtime API can be integrated for effective low-latency setups. Just let me know if you want to dive deeper into any of these points—I’m here to help! Sources: README.md | avatars/hedra/README.md | How can I reduce latency in voice agents using STT, TTS and LLM? | OpenAI and LiveKit partner to turn Advanced Voice into an API | examples/voice_agents/basic_agent.py
r
my agent takes nearly 5-8seconds to respond. While my TTFT is 500ms and TTFB is 800ms sometimes. What could be the reasons for this high latency.
t
I understand how frustrating high latency in voice agents can be, especially when your TTFT (500ms) and TTFB (800ms) seem low. There are a few common reasons this might happen: • End-of-Utterance (EOU) Detection Delay: Sometimes, voice activity detection (VAD) adds a delay (like 500ms or more) to make sure it doesn’t mistake pauses in speech for endpoints. This helps with accuracy but can add to latency. • Network Proximity: Even if TTFT and TTFB look good, the distance or routing between your agent and AI services (like STT, LLM, TTS) can still add up and cause delays. • Sequential Processing: If your STT, LLM, and TTS steps happen one after another instead of at the same time, it can really slow down the response. • Chunking Configuration: Larger audio chunks (more than 50ms) can also slow the processing pipeline. • System-Level Bottlenecks: Other elements, such as video rendering in multimodal setups, might be adding to the perceived latency. To get to the bottom of this, I recommend monitoring your end-to-end latency with LiveKit’s Agents Metrics API. Checking metrics like
end_of_utterance_delay
and
transcription_delay
can help pinpoint where things might be slowing down. If you want, I’d be happy to guide you through optimizing any specific areas—just let me know! Sources: Speech & Audio Processing | livekit/livekit_composite | DeepWiki | Agent Testing & Evaluation | livekit/livekit_composite | DeepWiki | Improving voice AI's turn detection with transformers | How can I reduce latency in voice agents using STT, TTS and LLM? | tests/test_agent_session.py