does livekit have audio to audio latency
# ask-ai
c
does livekit have audio to audio latency
t
Thanks for reaching out, Rashmi. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Hey there! So, LiveKit does have some audio-to-audio latency, but it’s built to keep that as low as possible. Thanks to its WebRTC setup, LiveKit keeps everyone in sync within about 250ms—whether you’re streaming, watching, speaking, or listening. For AI-powered voice apps, it’s pretty impressive too, offering near real-time responses with latency around ~300ms for human speech. Plus, it uses a global network of servers that’s fine-tuned to handle low-latency audio routing, even when things get busy. If you want to dive into the details for your specific situation, just give me a shout—I’m here to help!
c
I meant to ask, how to expose audio to audio latency for my python live kit agent which uses openai realtime
t
Hey there! If you want to check out the audio-to-audio latency for your Python LiveKit agent using OpenAI Realtime, here are some friendly ways to do it: 1. Using Event Listeners for Latency Tracking: You can keep an eye on latency by listening to certain events that the session sends out:
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import time
from livekit.agents import MultimodalAgent, AutoSubscribe, JobContext
from livekit.plugins.openai.realtime import RealtimeModel

class LatencyTrackingAgent:
    def __init__(self):
        self.audio_input_timestamp = None
        self.first_audio_output_timestamp = None
        self.latency_measurements = []
        
    async def setup_agent(self, ctx: JobContext):
        model = RealtimeModel()
        session = model.session()
        
        # Keep track of when audio input gets committed
        @session.on("openai_client_event_queued")
        def on_client_event(event):
            if event.get("type") == "input_audio_buffer.commit":
                self.audio_input_timestamp = time.time()
                
        # Keep track of when the first audio response comes through
        @session.on("openai_server_event_received") 
        def on_server_event(event):
            if event.get("type") == "response.audio.delta":
                if self.first_audio_output_timestamp is None:
                    self.first_audio_output_timestamp = time.time()
                    if self.audio_input_timestamp:
                        latency = self.first_audio_output_timestamp - self.audio_input_timestamp
                        self.latency_measurements.append(latency)
                        print(f"Audio-to-audio latency: {latency*1000:.2f}ms")
                        
        return session
2. Accessing Internal Timing Data: You can also peek inside the response generation objects to grab timing info and calculate how long processing takes:
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async def track_response_latency(session):
    # Check if there's a current generation object
    if hasattr(session, '_current_generation') and session._current_generation:
        generation = session._current_generation
        if generation._first_token_timestamp and generation._created_timestamp:
            processing_latency = generation._first_token_timestamp - generation._created_timestamp
            print(f"Model processing latency: {processing_latency*1000:.2f}ms")
3. End-to-End Latency Measurement: If you want the full picture from audio input all the way to output, try this:
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class ComprehensiveLatencyTracker:
    def __init__(self):
        self.input_start_time = None
        self.commit_time = None
        self.response_start_time = None
        self.first_audio_time = None
        
    def track_input_audio_buffer_events(self, session):
        @session.on("openai_client_event_queued")
        def on_client_event(event):
            event_type = event.get("type")
            if event_type == "input_audio_buffer.append" and self.input_start_time is None:
                self.input_start_time = time.time()
            elif event_type == "input_audio_buffer.commit":
                self.commit_time = time.time()
                
        @session.on("openai_server_event_received")
        def on_server_event(event):
            event_type = event.get("type")
            if event_type == "response.created":
                self.response_start_time = time.time()
            elif event_type == "response.audio.delta" and self.first_audio_time is None:
                self.first_audio_time = time.time()
                self.calculate_latencies()
                
    def calculate_latencies(self):
        if all([self.input_start_time, self.commit_time, self.response_start_time, self.first_audio_time]):
            input_to_commit = (self.commit_time - self.input_start_time) * 1000
            commit_to_response = (self.response_start_time - self.commit_time) * 1000  
            response_to_audio = (self.first_audio_time - self.response_start_time) * 1000
            total_latency = (self.first_audio_time - self.input_start_time) * 1000
            
            print(f"Input buffering: {input_to_commit:.2f}ms")
            print(f"Processing latency: {commit_to_response:.2f}ms") 
            print(f"Audio generation: {response_to_audio:.2f}ms")
            print(f"Total audio-to-audio latency: {total_latency:.2f}ms")
Hopefully, these tips help you measure and fine-tune latency throughout your audio pipeline. If you have any questions or want to dive deeper, just let me know—I'm here to help!