what can i do to redue latency here in between the...
# ask-ai
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what can i do to redue latency here in between the models? what works and what does livekit offer for reducing latency? model = openai.realtime.RealtimeModel( voice="shimmer", temperature=0.8, #model="gpt-4o-realtime-preview-2025-06-03", # Tell Whisper to expect Turkish (“tr”) audio _input_audio_transcription_=None, _turn_detection_=None, modalities=["text"] ) # Create AgentSession with the multimodal model session = AgentSession( llm=model, tts=elevenlabs.TTS( language=LANG_MODE, _voice_id_="cgSgspJ2msm6clMCkdW9", _streaming_latency_=3, ), vad=silero.VAD.load(), stt = deepgram.STT( model="nova-2-general", language=LANG_MODE, _sample_rate_=8000, # critical ), _#turn_detection=MultilingualModel(),_ )
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Thanks for reaching out, Rhythm. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
To reduce latency in your multimodal pipeline, consider the following strategies and features supported by LiveKit: • Network Proximity: Deploy your agent and all external services (LLM, STT, TTS) in the same region or as close as possible to minimize network round-trip time. • Stream All Components: Ensure every stage in your pipeline (audio capture, STT, LLM, TTS) streams data as soon as it is available, rather than waiting for full completion at each step. This is critical for minimizing end-to-end latency. • Direct Publishing: For video or avatar generation, have the generation server join the LiveKit room directly and publish media streams, rather than routing through intermediate servers. This avoids redundant encoding and transmission delays. • Monitor Latency: Use LiveKit’s Agents metrics API to track key metrics like Time To First Token (TTFT) and Time To First Byte (TTFB) to identify bottlenecks. • Network Congestion Handling: Enable automatic video pausing for congested subscribers and use UI feedback to inform users of network issues. • Transport Optimization: Leverage LiveKit’s WebRTC-based architecture for efficient, low-latency media transport between clients and servers. Applying these principles—especially streaming at every stage and optimizing network placement—will have the greatest impact on reducing latency in your real-time multimodal applications. Let me know if you want details on implementing any of these strategies in your specific setup. Sources: Live Conversations with AI using ChatGPT and WebRTC | Live Conversations with AI using ChatGPT and WebRTC | An open source stack for real-time multimodal AI | Voice Assistants | livekit/livekit_composite | DeepWiki | Bringing AI avatars to voice agents
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• Stream All Components: Ensure every stage in your pipeline (audio capture, STT, LLM, TTS) streams data as soon as it is available, rather than waiting for full completion at each step. This is critical for minimizing end-to-end latency.
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Hey there! Want to make your audio streaming super smooth and lightning-fast? I've got some friendly tips to help you minimize those pesky delays and keep your pipeline running like a well-oiled machine. • Audio Capture: Let's grab that audio in real-time! LiveKit's WebRTC is your new best friend - it'll help you stream audio directly from user devices. Pro tip: subscribe to those audio tracks ASAP to keep the sound flowing continuously. • STT (Speech-to-Text): Look for an STT provider that loves streaming as much as you do. If your current tool isn't streaming-friendly, no worries! A handy streaming adapter like
STTStreamAdapter
with VAD can help you process those audio frames on the fly. • LLM (Language Model): Speed is the name of the game! Choose models and APIs that can stream tokens. Instead of waiting around, start processing and forwarding tokens the moment they're generated. • TTS (Text-to-Speech): Smooth audio is happy audio! Seek out TTS engines that stream like pros. If yours needs a little help, a
TTSStreamAdapter
with sentence-based tokenization can generate and play audio segments piece by piece. Quick Pro Tips: • Go for models that are all about speed and streaming. • Break up your TTS output by sentence to keep things sounding natural. • No native streaming support? LiveKit's VoicePipelineAgent has got your back with built-in streaming adapters. • Use LiveKit's WebRTC transport for media that moves at lightning speed. By processing and forwarding data the moment it's available, you'll create a super-smooth, low-latency pipeline that goes from user input to AI response in the blink of an eye. You've got this! Sources: Live Conversations with AI using ChatGPT and WebRTC | agents/src/pipeline/pipeline_agent.ts | Pipeline nodes and hooks | LiveKit Docs | Live Conversations with AI using ChatGPT and WebRTC | LiveKit Docs
Hey there! I’m not quite sure what you mean by 'abhi123.' Could you please share a bit more info or clarify what you need help with? Are you asking about a specific part of the latency reduction process or something else? I’m here to help!