I'm using the OpenAI Realtime model for both STT a...
# ask-ai
m
I'm using the OpenAI Realtime model for both STT and LLM capabilities, while my TTS is handled by a separate provider. For some reason, at random times, the conversation gets stuck after I say something, then I have to say "hello?" or continue talking for it to detect speech again. I want it to be as smooth as possible. I have a suspicion it might be due to the turn detection, but I could be wrong. Here's my model config:
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openai.realtime.RealtimeModel(
    model="gpt-4o-realtime-preview",
    modalities=["text"],
    temperature=0.6,
    turn_detection=TurnDetection(
        type="server_vad",
        threshold=0.6,
        prefix_padding_ms=100,
        silence_duration_ms=120,
        create_response=True,
        interrupt_response=False,
    ),
)
t
Thanks for reaching out, Anas. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
It sounds like the speech detection interruptions you're facing might be tied to how the
server_vad
turn detection is set up. No worries though—here are a few tweaks you can try to make things run more smoothly: • Lower the threshold: Try bringing down the
threshold
value (say, to 0.4) so the system picks up on quieter speech better. • Increase silence duration: Bump up the
silence_duration_ms
to something like 700 ms to avoid detecting speech stops too soon. • Adjust prefix padding: Set
prefix_padding_ms
to 300 ms for a nicer, smoother transition. Here’s a quick example of what that might look like:
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TurnDetection(
    type="server_vad",
    threshold=0.4,
    prefix_padding_ms=300,
    silence_duration_ms=700,
    create_response=True,
    interrupt_response=True,
)
If you’re still running into issues after this, it might be worth checking your network connection and audio buffering. Making sure your WebSocket connection is steady and that audio data is properly buffered before hitting the API can really help.
m
is there any better option for ultra-low latency?
t
To help you get the absolute lowest latency in your setup, here are some friendly tips you might find useful: • Use WebRTC: WebRTC is great for low-latency audio chats because it handles packet loss better and adjusts bitrate on the fly, unlike WebSocket. • Optimize Turn Detection: Tweaking settings like
threshold
,
prefix_padding_ms
, and
silence_duration_ms
in your
server_vad
config can really cut down delays. • Leverage LiveKit: LiveKit’s cloud platform and client SDK are built for real-time audio streaming and work smoothly with OpenAI’s Realtime API. • Streaming STT and TTS: Make sure your speech-to-text and text-to-speech systems support streaming so audio and responses flow in real-time. • Stable Network: A solid, reliable network connection is key to avoiding those annoying latency spikes from drops or instability. Trying out these tweaks should make your system feel much more responsive. If you want, I’m here to help walk you through any of these steps! Sources: OpenAI and LiveKit partner to turn Advanced Voice into an API | Live Conversations with AI using ChatGPT and WebRTC | livekit-plugins/livekit-plugins-openai/livekit/plugins/openai/stt.py | Adjusting Model Parameters | LiveKit Docs
m
is semantic_vad lower latency?
t
Hey there! I don’t have exact info on how the latency of
semantic_vad
stacks up against
server_vad
, but if you’re aiming to speed things up, I’d be happy to help you explore some other settings or tweaks. Just let me know what sounds good to you!
m
is it better to set the turn detection in the
AgentSession
or in the
RealtimeModel
?
t
To get the best performance, it’s a good idea to set up turn detection right in the
RealtimeModel
instead of the
AgentSession
. The
RealtimeModel
comes with built-in turn detection that’s designed to work smoothly and keep costs down. Doing it this way means you won’t need extra STT resources, and you’ll enjoy better speed and accuracy. If you’d like, I’m here to help you tweak the settings!
m
I'm hitting this error:
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2025-08-17 17:17:18 | ERROR    | livekit.plugins.openai | realtime_model.py:_recv_task:790 | failed to handle event
Traceback (most recent call last):
  File "/venv/lib/python3.13/site-packages/livekit/plugins/openai/realtime/realtime_model.py", line 770, in _recv_task
    self._handle_response_audio_delta(
    ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^
        ResponseAudioDeltaEvent.construct(**event)
        ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
    )
    ^
  File "/venv/lib/python3.13/site-packages/livekit/plugins/openai/realtime/realtime_model.py", line 1362, in _handle_response_audio_delta
    item_generation.audio_ch.send_nowait(
    ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^
        rtc.AudioFrame(
        ^^^^^^^^^^^^^^^
    ...<4 lines>...
        )
        ^
    )
    ^
  File "/venv/lib/python3.13/site-packages/livekit/agents/utils/aio/channel.py", line 95, in send_nowait
    raise ChanClosed
livekit.agents.utils.aio.channel.ChanClosed
t
Hey there! The
ChanClosed
error pops up when the audio channel (
audio_ch
) gets closed before we handle the
ResponseAudioDeltaEvent
. This usually means the session has already ended or the audio stream stopped earlier than expected. To fix this, you can tweak the
_handle_response_audio_delta
method to check if the audio channel is still open before sending any data. Here's a little example for you:
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def _handle_response_audio_delta(self, event: ResponseAudioDeltaEvent) -> None:
    assert self._current_generation is not None, "current_generation is None"
    item_generation = self._current_generation.messages[event.item_id]

    if not item_generation.modalities.done():
        item_generation.modalities.set_result(["audio", "text"])

    data = base64.b64decode(event.delta)
    if not item_generation.audio_ch.closed:
        item_generation.audio_ch.send_nowait(
            rtc.AudioFrame(
                data=data,
                sample_rate=SAMPLE_RATE,
                num_channels=NUM_CHANNELS,
                samples_per_channel=len(data) // 2,
            )
        )
This way, we double-check the channel is still good to go before sending audio data, which helps avoid that pesky error. If you want me to walk you through it or have any other questions, just let me know—I’m here to help!
m
for some reason, the LLM stops generating a response after the STT after a few interactions. I can see the transcription of what I said, but it's not triggering tts_node
t
It sounds like the LLM stopping after STT transcription might be because the turn detection isn’t quite marking the end of the user’s turn properly. No worries though—here are a few things you can try to get things running smoothly: • Make Sure Turn Detection Is Set Up Right: Try using a reliable turn detection mode like
server_vad
. Also, avoid using
turn_detection="stt"
unless you have a separate STT plugin set up. • Use an External STT Plugin: If you want real-time, incremental transcription for turn detection, you might want to connect an external STT plugin like Deepgram or Google with your
AgentSession
. • Try Manual Turn Management: Another option is to manage turns manually, where the user confirms their turn by calling something like
commit_user_turn()
. • Double-Check Modalities: Make sure your
RealtimeModel
is configured with the right modalities (like
modalities=["text"]
) and paired with a TTS plugin if you want audio output. Hopefully, these tips help your LLM respond consistently! If you want, I’m here to walk you through any of these steps or answer any questions you have. Sources: LiveKit turn detector plugin | LiveKit Docs | LiveKit turn detector plugin | LiveKit Docs | livekit-plugins/livekit-plugins-openai/livekit/plugins/openai/realtime/real... | OpenAI Realtime API integration guide | LiveKit Docs | Azure OpenAI Realtime API and LiveKit | LiveKit Docs
m
here
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self._llm = openai.realtime.RealtimeModel(
                model=self.config.llm.openai_model,
                modalities=["text"],
                temperature=self.config.llm.temperature,
                turn_detection=TurnDetection(
                    type="server_vad",
                    threshold=0.5,
                    prefix_padding_ms=200,
                    silence_duration_ms=500,
                    create_response=True,
                    interrupt_response=False,
                ),
            )
t
Hey there! Your setup with
server_vad
turn detection should help keep the LLM responses flowing smoothly after the STT transcription, as long as the parameters are dialed in just right. Here are a few things to double-check: • Silence Duration: The
silence_duration_ms=500
setting decides how long silence needs to last before we consider the speech finished. If you’re noticing responses are coming too late or too early, tweaking this might help. • Threshold: The
threshold=0.5
controls how sensitive the system is to audio. Turning this up can cut down false triggers in noisy spots but might miss softer speech. • Create Response: Make sure
create_response=True
is set so responses get generated automatically right after transcription. If things still feel off, it could be something else in the pipeline, like how events are managed after transcription. Just give me a shout if you want to dig deeper—I’m here to help! Sources: OpenAI Realtime API integration guide | LiveKit Docs | Azure OpenAI Realtime API and LiveKit | LiveKit Docs | livekit-agents/livekit/agents/voice/agent_session.py | Azure OpenAI Realtime API and LiveKit | LiveKit Docs
m
it still stops talking altogether after a few interactions
t
Hey there! If your LLM stops generating responses after a few interactions, don’t worry—here are some common reasons and easy fixes to check out: • Generation Finished Reasons: Sometimes responses stop because of things like token limits, content filters, or interruptions. Take a peek at your settings and tweak them if needed. • StopResponse Triggers: Double-check your
on_user_turn_completed
hooks to make sure they’re not accidentally causing
StopResponse
exceptions. • Session Restart Limits: Keep an eye on your logs for session restarts and tackle any connectivity or API hiccups that might be behind them. • Timeouts: Look out for timeout errors in your logs. You might want to adjust timeout settings or improve your network connection. • Text vs Audio Mode: Sometimes the API switches from audio to text mode unexpectedly—adding some recovery logic can help smooth that out. • Conversation History: If you’re loading conversation history, make sure it’s not messing with response generation. Using a separate TTS plugin can help keep audio output steady. • Connection Errors: Build in solid error handling and retry steps for things like throttling or model errors to keep things running smoothly. Turning on detailed logging can really help pinpoint what’s going on so you can fix it faster. If you want, I’m here to help you walk through any of these steps—just let me know! Sources: plugins/openai/src/realtime/realtime_model.ts | OpenAI Realtime API integration guide | LiveKit Docs | GenerationFinishedReason | LiveKit Agents | Pipeline nodes and hooks | LiveKit Docs | livekit-agents/livekit/agents/voice/agent_activity.py
m
the LLM is suddenly returning a string similar to this "{"edits": [{"path": "...", "start_line": ..., "end_line": ...}]}"
t
Hey there! When the LLM sends back a JSON-like string with edits instead of a regular response, it usually means the API switched to text mode instead of audio mode. This can happen if conversation history is loaded into the model or because of how responses are handled. Here are some steps that might help you fix this: • Use Recovery Logic: Try adding a recovery method that handles text responses and then requests a new audio response. For example:
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recoverFromTextResponse(itemId: string) {
  if (itemId) {
    this.conversation.item.delete(itemId);
  }
  this.conversation.item.create(this.#createEmptyUserAudioMessage(1));
  this.response.create();
}
Separate TTS Plugin: Consider using a separate TTS plugin (like Azure OpenAI TTS) and set the model to text-only responses to keep your audio output consistent. • Check Delta Events: Those JSON-like strings are probably delta events (like
ResponseTextDeltaEvent
). Make sure your system is handling them properly and not mistaking them for final responses. If you want, I’m here to help walk you through implementing these fixes! Sources: ResponseAudioDeltaEvent | LiveKit Agents | OpenAI Realtime API integration guide | LiveKit Docs | livekit-plugins/livekit-plugins-openai/livekit/plugins/openai/realtime/real... | plugins/openai/src/realtime/realtime_model.ts | ResponseTextDeltaEvent | LiveKit Agents
m
I'm already using a separate TTS plugin. The LLM is working fine for a few interactions, then suddenly it starts returning code
t
Hey there! That hiccup where the LLM starts giving code-like responses after a few chats usually happens because the API switches from audio responses to text-only ones. This can pop up when the conversation history loads or if there's a lot of back-and-forth. Here’s a friendly checklist to help you fix it: • Separate TTS Plugin: Since you’re already using a separate TTS plugin, double-check that the LLM is set for text-only responses (
modalities=["text"]
) so things stay consistent. • Recovery Logic: It’s a good idea to have a recovery method ready for those surprise text responses. For example:
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recoverFromTextResponse(itemId: string) {
  if (itemId) {
    this.conversation.item.delete(itemId);
  }
  this.conversation.item.create(this.#createEmptyUserAudioMessage(1));
  this.response.create();
}
Limit Conversation History: Try cutting down how much conversation history loads into the model to lower the chances of switching to text-only responses. • Monitor Logs: Keep an eye out for warnings like "Text response received from OpenAI Realtime API in audio modality"—these can be your cue to trigger recovery steps. Hopefully, these tips help smooth things out! If you hit any snags or want to chat more, just let me know—I’m here to help. Sources: plugins/openai/src/realtime/api_proto.ts | livekit-plugins/livekit-plugins-openai/livekit/plugins/openai/realtime/real... | OpenAI Realtime API integration guide | LiveKit Docs | Realtime model integrations | LiveKit Docs
m
if it suddenly switches to text mode, how can I switch it back? would manually calling generate_reply work?
t
Hey there! Manually calling
generate_reply
can get you a new response, but it doesn’t directly switch the API back to audio mode. The best way to handle this is by using the built-in recovery mechanism, which: • Deletes the text response item • Creates an empty user audio message • Requests a new response via
response.create()
This approach is made to help you bounce back from text responses and smoothly switch back to audio mode. If you keep seeing text responses even after trying this, just let me know—we can dig in together and figure out what’s going on! Sources: agents/src/multimodal/multimodal_agent.ts | RealtimeSession | LiveKit Agents | livekit-plugins/livekit-plugins-aws/livekit/plugins/aws/experimental/realti...
m
I can't find the function
_recover_from_text_response
in the latest version
t
Hey there! Just a heads-up: the function
_recover_from_text_response
isn’t available in the latest version. But don’t worry—there’s a similar method called
recoverFromTextResponse
if you’re working with JavaScript or TypeScript. This handy method: • Deletes the tricky text response item • Creates an empty user audio message • Requests a fresh response to switch back to audio mode If you’re using Python, this exact method isn’t part of the package, but usually text responses are managed within the event processing system. Just let me know whether you’re working with JavaScript/TypeScript or Python, and I’d be happy to help you figure out the best way to handle recovery or suggest some alternatives! Sources: plugins/openai/src/realtime/api_proto.ts | RealtimeSession | LiveKit Agents | livekit-plugins/livekit-plugins-openai/livekit/plugins/openai/realtime/real... | OpenAI Realtime API integration guide | LiveKit Docs
m
I'm using the python sdk
t
Hey there! Just wanted to share a quick tip about the Python SDK — it automatically handles recovery from text responses by falling back to a TTS model whenever the Realtime API sends text instead of audio. To make sure this works smoothly for you, just set up your agent with both the Realtime model and a TTS model like this:
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agent = Agent(
    instructions="Your instructions here",
    llm=openai.realtime.RealtimeModel(voice="alloy", model="gpt-4o-mini-realtime-preview"),
    tts=openai.TTS(voice="alloy"),
    chat_ctx=chat_ctx,
)
Also, if you’re loading conversation history or expecting text responses, you can configure the model for text-only mode and pair it with a TTS plugin to keep things running smoothly:
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session = AgentSession(
    llm=openai.realtime.RealtimeModel.with_azure(
        # ... endpoint and auth params ...,
        modalities=["text"]
    ),
    tts=cartesia.TTS()  # Or any other TTS plugin you prefer
)
This way, you’ll get consistent audio output without having to worry about manual recovery. If you want me to walk you through it or have any questions, just let me know — I’m here to help! Sources: Azure OpenAI Realtime API and LiveKit | LiveKit Docs | agents/src/multimodal/multimodal_agent.ts | livekit-agents/livekit/agents/tts/tts.py | OpenAI Realtime API integration guide | LiveKit Docs
m
is there a way to stop it from triggering tts_node if the LLM fails? I worry that even though it falls back to the TTS, it's still triggering tts_node, which I'm overriding
t
Hey there! To help you avoid the TTS fallback from triggering
tts_node
when the LLM hits a snag, here are a few friendly options you can try out: • Remove TTS from Configuration: Simply don’t include a TTS instance when you create your agent. This way, no fallback will happen:
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agent = Agent(
    instructions="You are a helpful assistant.",
    stt=deepgram.STT(),
    llm=openai.LLM(),
    vad=silero.VAD.load()
)
Override `tts_node`: You can create your own agent class and override the
tts_node
method to stop TTS from running:
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class NoFallbackAgent(Agent):
    async def tts_node(self, text, model_settings):
        async def empty_audio():
            return
            yield
        return empty_audio()
Dynamic Audio Control: If you want to toggle audio output on the fly, you can disable it during the session like this:
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session.output.set_audio_enabled(False)
Custom Error Handling: Another way is to catch LLM errors explicitly and stop the fallback from happening:
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@session.on("error")
def on_error(ev):
    if ev.error.recoverable:
        ev.error.recoverable = False
Feel free to let me know which one sounds like a good fit for you, or if you want a hand setting any of these up—I’m here to help! Sources: examples/voice_agents/structured_output.py | Text and transcriptions | LiveKit Docs