im using the gemini realtime model and trying to create a push to talk interface. however, i dont seem to be getting audio responses from the model, despite it transcribing my audio correcrtly (suggesting it is receiving the audio). what is the suggested convo framework for gemini realtime model

Thanks for reaching out, adit-kulkarni. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.

is push to talk supported for gemini realtime?

Yes — push‑to‑talk is supported with the Gemini realtime model. Key points to implement it: • Use

turn_detection="manual"

in your

AgentSession

so you control when the agent processes user input. • Enable audio capture while the user holds the button and disable it when they release: ◦ Start with

session.input.set_audio_enabled(False)

. ◦ On press:

session.interrupt()

,

session.clear_user_turn()

,

session.input.set_audio_enabled(True)

. ◦ On release:

session.input.set_audio_enabled(False)

, then

session.commit_user_turn()

to trigger the model response (which will include audio if audio modalities are enabled). • Configure the session LLM as

google.realtime.RealtimeModel(model="gemini-...")

and do not disable audio modalities so the model can produce built‑in TTS audio. If you’d like, share the model name you’re using and any logs around

commit_user_turn()

and assistant output — I can help debug further.

do these apply to the google realtime plugin or are they only applicable for stt/tts pipleins

Good question — the push‑to‑talk framework I described applies to the Google Realtime plugin (the realtime model) itself, not only to separate STT/TTS pipelines. • The realtime plugin handles both STT and built‑in TTS, so using

turn_detection="manual"

, toggling

session.input.set_audio_enabled(...)

, and calling

session.commit_user_turn()

will work with the Google Realtime model and cause it to produce audio responses when enabled. • The same pattern (manual turn control, enable/disable capture, then commit) also works conceptually if you build your own separate STT/TTS pipeline, but you don’t need a separate TTS if you use the realtime model’s built‑in audio output. If you want, tell me which integration you’re using (the realtime plugin name or a custom STT/TTS setup) and I’ll give exact code snippets for that environment.

and is it possible to append audio to a buffer whilst the user has the button pushed, then when the user releases the button, the audio gets committed?

Yes — you can buffer audio while the button is held and commit it on release. How to do it (concepts + quick pattern) • Enable audio capture when the user presses the button and keep capturing into the session's input buffer while held. • On release, disable audio capture and call

session.commit_user_turn()

to signal end of input and trigger the model response (including built-in TTS). • If you need explicit control over buffering (e.g., pre-buffer from mic, stitch multiple chunks, or replay/trim before commit), you can append audio to the session input manually and then commit. Use the session input APIs (start/stop audio capture or

append_audio

if available) to accumulate audio while pressed, then

session.commit_user_turn()

on release. Best practices • Call

session.interrupt()

before enabling capture on press to stop any in-progress assistant audio. • Call

session.clear_user_turn()

on press if you want to discard partial user audio state before starting a new capture. • Keep

turn_detection="manual"

so the session waits for your explicit

commit_user_turn()

. If you want, tell me: • Which client/plugin you’re using (Google Realtime plugin name or a custom pipeline) and whether you need a code example for automatic mic capture vs. manual

append_audio

. I’ll provide the exact snippet.

how do i do "keep capturing into the session's input buffer" - is this supported directly or do i need to manage that myself?

Good question — both options are supported. Short answer • The Google Realtime plugin will automatically accumulate audio into the session input while you have capture enabled via

session.input.set_audio_enabled(True)

. • If you need finer control (pre‑buffering, trimming, stitching, or replaying chunks) you can also append audio manually with

session.input.append_audio()

and then call

session.commit_user_turn()

. Quick pattern (manual turn detection + buffering) 1. Create session with

turn_detection="manual"

. 2. Start with audio disabled: ◦

session.input.set_audio_enabled(False)

1. On button press: ◦

session.interrupt()

to stop assistant audio ◦

session.clear_user_turn()

if you want to discard prior partial input ◦

session.input.set_audio_enabled(True)

— realtime plugin will start buffering mic audio into the session input ◦ (Optional) instead of enabling mic capture, capture audio yourself and call

session.input.append_audio(chunk)

to add chunks to the buffer 1. On button release: ◦

session.input.set_audio_enabled(False)

(if using mic capture) ◦

session.commit_user_turn()

to signal end of turn and trigger the model response When to use manual append vs automatic capture • Use automatic capture (

set_audio_enabled(True)

) for straightforward push‑to‑talk — the plugin handles mic streaming and buffering. • Use

append_audio()

when you need to manipulate audio (trim, combine multiple sources, or implement custom capture logic) before committing. If you tell me which client/plugin or environment you’re using (browser WebRTC, native SDK, or a custom pipeline), I can provide a concise code snippet for that setup.

so just to be crystal clear, will the audio be sent to gemini (from my swift frontend) only on commit_user_turn and importantly NOT before that?

Thanks — you're asking whether audio from your Swift frontend is sent to Gemini only when you call

commit_user_turn()

or earlier. Here's the clear answer. Short answer • Audio can be streamed to Gemini for realtime transcription while the button is held (when you enable mic capture). The model will not begin generating an assistant response until you call

commit_user_turn()

. • If you use

session.input.set_audio_enabled(True)

the realtime plugin will send audio as it's captured for STT. If you instead capture audio yourself and use

session.input.append_audio(...)

, those appended chunks are sent when you append them (and still require

commit_user_turn()

to trigger a response). • In both cases,

commit_user_turn()

is what signals the end of the user turn and causes the model to generate (and stream) the assistant response and its audio output. If you want a Swift-specific snippet showing mic capture vs manual append, tell me which SDK/client you're using (browser WebRTC, iOS realtime SDK, or a custom websocket client) and I’ll provide the concise example.

im using "gemini-2.5-flash-native-audio-preview-09-2025" and ive been doing your suggested pattern but i seem to be getting responses from gemini before ive released the mic/cimmied the user turn. could you put together an example agent pythin code and swift frontend code for managing this push to talk pipeline

Thanks — I’ll show a minimal Python agent example plus a Swift frontend pattern that enforce manual turns so Gemini only generates responses after you call

commit_user_turn()

. I’ll also list the common reasons you might still see early responses and how to avoid them. Acknowledgement • This responds to your request for example Python agent code and Swift frontend code for model "gemini-2.5-flash-native-audio-preview-09-2025" using push‑to‑talk with manual turn detection. Python agent (minimal outline) • Create AgentSession with manual turn detection and RealtimeModel LLM. • Expose endpoints or a websocket so the Swift client can start/stop capture and send appended audio if you choose manual append.

# Python (server / agent) - conceptual example
from some_agent_sdk import AgentSession, google

# Create session with manual turn detection and realtime LLM
session = AgentSession(
    turn_detection="manual",
    llm=google.realtime.RealtimeModel(model="gemini-2.5-flash-native-audio-preview-09-2025"),
    # other components as needed
)

# Start session (framework-specific)
session.start()

# OPTIONAL: If you accept appended audio from client:
def receive_audio_chunk(chunk_bytes):
    # append chunk to session buffer
    session.input.append_audio(chunk_bytes)

# If using mic capture on the client (preferred):
# Provide endpoint or signaling so client calls server only to control session:
def on_client_press():
    session.interrupt()
    session.clear_user_turn()
    # tell client to start capturing locally (or server can instruct plugin)
    # If server-side capture is used, session.input.set_audio_enabled(True)

def on_client_release():
    # If server-side capture was enabled: session.input.set_audio_enabled(False)
    # Commit the user turn to trigger generation (and audio response)
    session.commit_user_turn()

# Hook session events to stream assistant audio back to client as produced.
def on_assistant_audio_chunk(chunk):
    # forward to client for playback
    pass

Swift frontend (conceptual pattern) • Two options: let the realtime plugin handle mic capture (preferred) or capture mic yourself and send chunks to the server to append. Both require calling commit only on release. Option A — Let realtime plugin handle capture (client controls session via RPC/signaling) 1. On press: ◦ Send "press" signal to server. ◦ Start local mic capture and direct audio into the session (or instruct server/plugin to enable capture). 1. While held: ◦ Continue capturing; audio accumulates in session input. 1. On release: ◦ Stop local capture (or instruct server/plugin to disable capture). ◦ Send "release" signal to server, which calls

commit_user_turn()

. 1. Play assistant audio as server streams it back. Pseudo Swift (high-level)

// UI handlers
func pushPressed() {
    sendSignalToServer("press") // server will interrupt + clear_user_turn
    startMicCapture()          // or tell server/plugin to enable capture
}

func pushReleased() {
    stopMicCapture()           // or tell server/plugin to disable capture
    sendSignalToServer("release") // server will call commit_user_turn()
}

// playback: receive assistant audio chunks from server and play
func onAssistantAudioChunk(_ data: Data) {
    audioPlayer.append(data)
}

Option B — Client captures and appends audio manually • Capture microphone into PCM chunks locally. • On press: notify server to prepare, clear previous user turn. • While held: send each chunk to server endpoint that calls

session.input.append_audio(chunk)

. • On release: send release signal; server calls

commit_user_turn()

. Why you might see responses before release (and how to avoid them) • turn_detection not set to "manual": ensure

turn_detection="manual"

. • An earlier

commit_user_turn()

call is triggered by code (search for accidental calls). • Using automatic capture + an automatic turn detector elsewhere: confirm no other component triggers end‑of‑turn. • If you append audio chunks to the session as they arrive, they are sent for STT immediately — that’s fine; the model still waits for

commit_user_turn()

to generate the assistant response. If you see generation start early, something is signaling commit. • Concurrent interruptions:

session.interrupt()

may cause state changes; call it only to stop assistant audio before starting capture. • Multi-client/session confusion: ensure the session you append audio to is the same session you call commit on. Next steps I can provide • A concrete Swift snippet tailored to your SDK (iOS realtime SDK vs custom WebRTC/datachannel) if you tell me which one you’re using. • Help inspect logs around any

commit_user_turn()

calls or session events if you paste them. Would you like a full runnable Swift + Python example for the iOS realtime SDK, or are you using a custom websocket/WebRTC client?

do we need to do anything with automatic_activity_detection.disabled=true?

Short answer: yes — set automatic_activity_detection.disabled=true (or otherwise disable automatic activity/turn detection) when using manual push‑to‑talk. That ensures the model won’t auto‑end the user turn and start generating before you call

commit_user_turn()

. What to do • Use

turn_detection="manual"

on the AgentSession and disable automatic activity detection:

automatic_activity_detection.disabled=true

(or the equivalent flag in your SDK). • Keep audio capture enabled only while the button is pressed (

session.input.set_audio_enabled(True)

), then disable it and call

session.commit_user_turn()

on release. Why this matters • If automatic activity detection is enabled the system can detect end‑of‑speech and commit a turn automatically, causing early assistant responses. Disabling it hands full control to your push‑to‑talk UI so responses only start after your explicit commit. Quick checklist • Set

turn_detection="manual"

. • Disable automatic activity detection (

automatic_activity_detection.disabled=true

). • Call

session.interrupt()

+

session.clear_user_turn()

on press, enable capture, then disable capture +

session.commit_user_turn()

on release. • Confirm there are no accidental

commit_user_turn()

calls or other turn detectors running. If you want, tell me which SDK/version you’re using (iOS realtime SDK or a custom client) and I’ll give the exact flag placement and a tiny code snippet.

should i use generate_reply or commit_user_turn for gemini realtime

Use commit_user_turn for Gemini Realtime in a push‑to‑talk flow. Why •

commit_user_turn()

signals the end of the user’s turn in a manual turn‑detection session and triggers the agent/model to generate a reply (including realtime TTS). That aligns with push‑to‑talk semantics: audio is captured/ buffered while the button is held, and the reply is produced only after you explicitly commit. •

generate_reply()

is for manually asking the model to produce a response outside the AgentSession turn lifecycle. It can work, but it’s not the recommended pattern for the realtime AgentSession push‑to‑talk flow and can conflict with session state, turn tracking, and the session’s built‑in audio streaming behavior. Recommended setup •

turn_detection="manual"

•

automatic_activity_detection.disabled=true

• Capture audio while pressed (

session.input.set_audio_enabled(True)

or

session.input.append_audio(...)

) • On release: disable capture and call

session.commit_user_turn()

If you want, I can provide a short code snippet showing

commit_user_turn()

usage in your Python agent + Swift frontend for

gemini-2.5-flash-native-audio-preview-09-2025

. Which client SDK are you using on iOS (iOS realtime SDK or custom WebRTC/datachannel)?

Goal: Implement push-to-talk with Gemini Realtime model (gemini-2.5-flash-native-audio-preview-09-2025) Current approach: - AgentSession(turn_detection="manual") with google.realtime.RealtimeModel - Disabled Gemini's built-in VAD via automatic_activity_detection.disabled=True - RPC handlers: start_turn enables audio, end_turn disables audio and calls commit_user_turn() - Audio is disabled at session start Problem: 1. Without automatic_activity_detection.disabled=True: Gemini responds automatically when it detects silence (before end_turn RPC is called) - not true push-to-talk 2. With automatic_activity_detection.disabled=True: Gemini never responds after commit_user_turn() is called Question: How do we properly trigger a Gemini Realtime response in push-to-talk mode after audio is buffered and the user releases the button? Should we use commit_user_turn(), generate_reply(), or something else?

Thanks — this is a clear summary. Short answer: use commit_user_turn(), but ensure automatic activity detection is truly disabled and your session is receiving the buffered audio before committing. If commit_user_turn() still does nothing, the usual causes and fixes are below. What to do (concise) • Use:

session.commit_user_turn()

to trigger the Gemini realtime response. • Must set: ◦

turn_detection="manual"

◦

realtime_input_config.automatic_activity_detection.disabled = True

• Buffer audio while button held: ◦ Either enable plugin mic capture:

session.input.set_audio_enabled(True)

(plugin buffers and streams STT) ◦ Or capture locally and call

session.input.append_audio(chunk)

to buffer manually • On release: ◦ Stop capture:

session.input.set_audio_enabled(False)

(or stop sending chunks) ◦ Call:

session.commit_user_turn()

If commit_user_turn() produces no response — checklist to verify • Confirm the

realtime_input_config

flag is actually applied to the active RealtimeModel instance (no config mismatch). • Confirm audio was streamed/appended to the same session prior to commit (check session input/transcription logs). • Confirm no other turn detector or accidental `commit_user_turn()`/`generate_reply()` calls are running. • Ensure you’re not using

generate_reply()

in place of commit;

generate_reply()

is for agent-initiated replies and can conflict with session state. • Check server/client logs for events: audio chunks received, transcription results, and the commit_user_turn call — there should be transcription/segments before the commit. If you want, I can: • Review a small excerpt of your session initialization and the RPC handlers (start_turn / end_turn) to spot config or ordering issues. • Provide a minimal runnable Python + Swift snippet targeted to your iOS client type. Which iOS client are you using (iOS realtime SDK or custom WebRTC/datachannel)? Sources: Gemini Live API plugin | LiveKit docs | Turns overview | LiveKit docs

Goal: Implement push-to-talk with Gemini Realtime model (gemini-2.5-flash-native-audio-preview-09-2025) in a LiveKit agent. Current setup: - AgentSession(turn_detection="manual") with google.realtime.RealtimeModel - automatic_activity_detection.disabled=True in realtime_input_config - RPC handlers: start_turn calls session.interrupt(), session.clear_user_turn(), session.input.set_audio_enabled(True) - end_turn calls session.input.set_audio_enabled(False) then either commit_user_turn() or generate_reply() What we've tried: 1. commit_user_turn() → No response from Gemini 2. generate_reply() → No response from Gemini 3. Various orderings of enable/disable audio with these calls Observations: - Audio IS reaching Gemini (we can see transcription in some modes) - With automatic_activity_detection.disabled=True, Gemini never responds - Without it disabled, Gemini auto-responds when it detects silence (before end_turn RPC) Key insight from non-LiveKit implementation: Our working push-to-talk uses OpenAI Realtime API which has an explicit server-side audio buffer with events: - input_audio_buffer.clear - Clear buffer on button press - input_audio_buffer.commit - Commit buffer on release (triggers response) This gives explicit control over when audio is "submitted" for processing. Question: With Gemini Realtime + automatic_activity_detection.disabled=True: 1. How do we properly signal activity_start and activity_end to Gemini through the LiveKit agents framework? 2. Is there an equivalent to OpenAI's buffer commit pattern? 3. Or is push-to-talk with Gemini Realtime simply not supported in the same way, and we should use STT-LLM-TTS pipeline instead?

Thanks — I see you're blocked: with automatic activity detection disabled Gemini never responds after you call commit_user_turn(), but with it enabled Gemini auto-responds on silence. I'll answer your three questions directly and give the minimal, actionable path forward. Summary answers 1) How to signal activity_start / activity_end in LiveKit AgentSession • You don’t send explicit activity_start/activity_end events. Instead: ◦ Enable audio capture for the session while the user holds the button (

session.input.set_audio_enabled(True)

or

session.input.append_audio(...)

). ◦ Disable audio capture on release (

session.input.set_audio_enabled(False)

). ◦ Call

session.commit_user_turn()

to mark end of the user turn and trigger the model’s response. • Also set

turn_detection="manual"

on the AgentSession. 2) Is there an equivalent to OpenAI’s buffer commit? • Yes:

session.commit_user_turn()

is the LiveKit/AgentSession equivalent of a buffer commit — it signals the session that the user’s turn is finished and requests generation. • Important:

commit_user_turn()

will only trigger a response if the realtime model’s automatic activity detection is actually disabled at the model layer (see next point). 3) If commit_user_turn() doesn't trigger a response, why — and should you use a STT→LLM→TTS pipeline instead? • Cause: If

automatic_activity_detection.disabled=True

is not actually applied to the active RealtimeModel instance, the model may remain waiting for internal activity signals or be in a state where it won’t react to commit. In practice this is the most common cause when commit does nothing. • Confirm these three things: a. Your RealtimeModel was created with

realtime_input_config.automatic_activity_detection.disabled = True

(the flag must be set on the model you attach to the session). b.

turn_detection="manual"

on the AgentSession. c. Audio was indeed streamed or appended to the same session before calling

commit_user_turn()

(check transcription or audio-received events). • If all three are correct and commit still does nothing, two pragmatic options: ◦ Fix configuration/order so

automatic_activity_detection.disabled=True

is applied before starting audio capture. This is the preferred path — it allows native Gemini realtime TTS and simpler flow. ◦ If you cannot get that to work reliably, use an explicit STT → call model via generate_reply()/generate API → TTS pipeline. That gives you an explicit client/server buffer commit you control, but requires managing STT/TTS yourself and loses builtin native audio behavior. Practical checklist to fix your setup (in this order) 1. When creating the RealtimeModel ensure realtime_input_config gets passed correctly:

# Python (model init)
realtime_model = google.realtime.RealtimeModel(
    model="gemini-2.5-flash-native-audio-preview-09-2025",
    realtime_input_config=RealtimeInputConfig(
        automatic_activity_detection=AutomaticActivityDetection(disabled=True)
    ),
    voice="Puck",
)
session = AgentSession(llm=realtime_model, turn_detection="manual")
session.input.set_audio_enabled(False)

1. RPC handlers • start_turn: ◦ session.interrupt() ◦ session.clear_user_turn() ◦ session.input.set_audio_enabled(True) • end_turn: ◦ session.input.set_audio_enabled(False) ◦ session.commit_user_turn() 1. Verify events/logs • Confirm you see audio chunks or interim transcriptions in the session before commit. • Confirm commit_user_turn() call appears in logs and that session transitions to generation. 1. If you still see no response: • Check for a config mismatch (a different model instance or session created without the disabled flag). • Ensure no other automatic turn detector (client or server) is active. • Share minimal logs showing: audio-received events, transcription, commit_user_turn call, and model response (or lack thereof). If you want, I can: • Review a short copy of the session creation and your start_turn / end_turn handlers to spot misordering or config issues. • Provide a tiny runnable example (Python LiveKit agent + iOS snippet) showing the exact sequence. Which would you prefer?

please can you provide a tuny runnable example showing the exact sequence as im unable to get it to work reliably

Below is a minimal, focused runnable example showing the exact sequence (Python LiveKit agent + Swift iOS frontend pattern). It enforces manual turn control, disables automatic activity detection on the RealtimeModel, buffers audio while the button is held, and calls

commit_user_turn()

on release. Python agent (minimal, asyncio) • Requirements: livekit agent SDK that provides AgentSession and google.realtime.RealtimeModel (adjust imports to match your installed package). • Purpose: host an AgentSession, expose simple RPCs "start_turn" and "end_turn", and forward assistant audio back to client.

# server.py
import asyncio
import logging

# Replace with actual SDK imports for your LiveKit Agent environment
from livekit_agent import AgentSession, google  # adjust to real package names
from google.genai.types import RealtimeInputConfig, AutomaticActivityDetection

logging.basicConfig(level=logging.DEBUG)

async def main():
    # Create realtime model with automatic activity detection disabled
    realtime_model = google.realtime.RealtimeModel(
        model="gemini-2.5-flash-native-audio-preview-09-2025",
        realtime_input_config=RealtimeInputConfig(
            automatic_activity_detection=AutomaticActivityDetection(disabled=True)
        ),
        voice="Puck",
    )

    session = AgentSession(
        llm=realtime_model,
        turn_detection="manual",
    )

    # Ensure audio is disabled at start
    session.input.set_audio_enabled(False)

    # Hook up RPC handlers (pseudo-code; wire to your actual signaling/room)
    async def on_start_turn(_):
        # stop assistant audio, clear previous user turn, start capture
        session.interrupt()
        session.clear_user_turn()
        session.input.set_audio_enabled(True)
        <http://logging.info|logging.info>("start_turn: audio enabled")

    async def on_end_turn(_):
        # stop capture, commit the buffered user turn to trigger response
        session.input.set_audio_enabled(False)
        <http://logging.info|logging.info>("end_turn: audio disabled, committing user turn")
        session.commit_user_turn()

    # Register handlers with your room/participant RPC mechanism
    # e.g., room.register_rpc("start_turn", on_start_turn), etc.
    # For demo, simulate a run loop
    print("Agent running. Wire RPCs to on_start_turn/on_end_turn in your app.")
    while True:
        await asyncio.sleep(1)

if __name__ == "__main__":
    asyncio.run(main())

Notes for server: • Confirm the RealtimeModel constructor actually applies

realtime_input_config

. If your SDK differs, ensure the

automatic_activity_detection.disabled=True

flag is set on the active model instance. • Ensure the same session instance receives the audio chunks (if you append manually) or has plugin mic capture enabled. Swift frontend (conceptual, iOS) • Purpose: control push-to-talk UI, capture microphone into PCM, and either let plugin stream audio to the session (preferred) or upload PCM to server for append_audio. • This example shows local mic capture + RPC signaling to server. Use AVAudioEngine for capture and send chunks to server endpoint that either calls

session.input.append_audio()

or the plugin handles the streaming.

// PushToTalkController.swift (conceptual)
import Foundation
import AVFoundation

class PushToTalkController: NSObject {
    let engine = AVAudioEngine()
    var inputNode: AVAudioInputNode!
    var isCapturing = false

    // Replace with your signaling client to call RPCs on server
    var signaling: SignalingClient!

    override init() {
        super.init()
        inputNode = engine.inputNode
    }

    func startCapture() throws {
        guard !isCapturing else { return }
        let format = inputNode.outputFormat(forBus: 0)

        inputNode.installTap(onBus: 0, bufferSize: 1024, format: format) { buffer, when in
            // Convert buffer to PCM data
            let data = self.pcmData(from: buffer)
            // Send chunk to server (append_audio) or let plugin stream if using SDK
            self.signaling.sendAudioChunk(data)
        }

        try engine.start()
        isCapturing = true
    }

    func stopCapture() {
        guard isCapturing else { return }
        inputNode.removeTap(onBus: 0)
        engine.stop()
        isCapturing = false
    }

    func pcmData(from buffer: AVAudioPCMBuffer) -> Data {
        let audioBuffer = buffer.audioBufferList.pointee.mBuffers
        let data = Data(bytes: audioBuffer.mData!, count: Int(audioBuffer.mDataByteSize))
        return data
    }

    // UI handlers
    func onPushPressed() {
        // Tell server to prepare: interrupt + clear_user_turn
        signaling.callRPC("start_turn", params: nil)
        // Start capturing locally and streaming chunks
        try? startCapture()
    }

    func onPushReleased() {
        // Stop capturing
        stopCapture()
        // Tell server we finished: server should call commit_user_turn()
        signaling.callRPC("end_turn", params: nil)
    }
}

Signaling expectations •

signaling.sendAudioChunk(data)

should POST or stream to your server endpoint that calls

session.input.append_audio(data)

(if you’re appending on server). Or, if your client uses a LiveKit realtime SDK that directly streams mic into the session, you can skip manual chunk uploads and let the plugin stream (then server-side only needs start/end RPCs). Common pitfalls and how to verify • The realtime model flag must be set on the model instance attached to the session. If you create the session first and later attach a model or recreate it, the flag may be lost. • Verify logs: ◦ Audio-chunk received or interim transcription before commit. ◦

commit_user_turn()

call logged. ◦ Model transitions to “generating” / assistant audio events after commit. • If you see transcription but no generation after commit, it means commit is not triggering generation — usually a sign the model instance still has automatic detection or the flag wasn’t applied. If this still fails for you • Paste the exact session creation snippet and start_turn/end_turn handlers you run (remove secrets). I’ll scan for ordering/config issues. • Tell me whether you stream audio via the client SDK (plugin handles mic) or upload chunks to server and call

session.input.append_audio

— that determines small but important differences. Want a fully runnable sample? • I can produce a complete zip with: ◦ Minimal asyncio server that exposes HTTP endpoints

/start_turn

,

/end_turn

,

/append_audio

◦ Swift sample that posts PCM chunks and calls

/start_turn

and

/end_turn

If so, confirm you want the full runnable repo and whether you prefer plugin mic streaming or client→server append_audio flow.

We're using LiveKit's iOS SDK (LiveKitComponents) which streams audio directly via RoomIO. We're NOT manually calling session.input.append_audio(). With automatic_activity_detection.disabled=True and turn_detection="manual": - Audio streams via RoomIO when session.input.set_audio_enabled(True) - We call session.commit_user_turn() on button release - But Gemini never responds (no transcription events, no generation) Does commit_user_turn() work with RoomIO-streamed audio, or do we need to use append_audio() manually? Is there something special about how RoomIO feeds audio to the Gemini realtime session?

Thanks — I see you're streaming mic audio via LiveKit RoomIO (LiveKit iOS SDK) and after disabling automatic activity detection + using manual turns, commit_user_turn() does not trigger Gemini to transcribe or generate. I'll be direct: commit_user_turn() does work with RoomIO-streamed audio — you do NOT have to switch to manual

append_audio()

— but only if a few conditions are met exactly. Below are the precise checks, a minimal sequence to enforce, and the most common pitfalls to inspect. What must be true • The RealtimeModel attached to the AgentSession has automatic activity detection disabled on the active model instance: ◦

realtime_input_config.automatic_activity_detection.disabled = True

• The AgentSession is created with

turn_detection="manual"

. • RoomIO is actually streaming into that same AgentSession when you call

session.input.set_audio_enabled(True)

. The session must be the one tied to the RealtimeModel instance above. • You call

session.commit_user_turn()

only after you stop RoomIO capture (i.e., after

session.input.set_audio_enabled(False)

). Minimal sequence (server + iOS RoomIO) 1. Session init (server) ◦ Create RealtimeModel with realtime_input_config disabling automatic activity detection. ◦ Create AgentSession with that model and

turn_detection="manual"

. ◦ Ensure

session.input.set_audio_enabled(False)

at start. 1. On button press (client -> server RPC) ◦ server:

session.interrupt()

◦ server:

session.clear_user_turn()

◦ server:

session.input.set_audio_enabled(True)

// RoomIO begins streaming into session ◦ client: start local UI capture (RoomIO already wired to the LiveKit SDK will send audio) 1. While held ◦ audio should stream from iOS RoomIO to the session (you should see audio-received events or interim STT if enabled) 1. On release ◦ server:

session.input.set_audio_enabled(False)

// stop RoomIO->session streaming ◦ server:

session.commit_user_turn()

// signal end-of-turn -> triggers transcribe+generate Key verification steps (check logs/events) • Confirm model config: log the active RealtimeModel instance and show

automatic_activity_detection.disabled=True

. • Confirm audio-in: server logs should show RoomIO audio frames received or STT interim segments while capture is enabled. • Confirm commit call: server logs should show the

commit_user_turn()

invocation immediately after stopping audio. • Confirm session state change: after commit, you should see session transition to "generating" and assistant audio/transcript events. Most common reasons commit_user_turn() appears to do nothing • Config not applied: the

realtime_input_config

flag was not actually attached to the model instance used by the session (e.g., session created before the model config or model re-created later). • Wrong session: RoomIO is streaming to a different Room/Session than the one you call commit on (multi-session mixups). • Timing/order: calling

commit_user_turn()

while RoomIO still streaming (don’t commit until after

set_audio_enabled(False)

). • Race with client-side VAD or other detectors: ensure no other automatic turn detector on the client or server is interfering. • Missing permissions or audio routing issues: RoomIO may start but not actually deliver frames to the session (verify audio-frame counts). If you still see no transcription/generation after commit • Share these three minimal logs (redact secrets): a. The code snippet that constructs the RealtimeModel + AgentSession. b. Server logs showing: start_turn RPC, audio-received events (or lack), the set_audio_enabled False call, and the commit_user_turn() call. c. A short iOS log showing RoomIO started/stopped capture around the press/release. Workaround if you need an immediate alternative • As a fallback, capture on iOS and POST PCM chunks to your server and call

session.input.append_audio()

while held, then

commit_user_turn()

on release. That always gives explicit buffer control, but isn’t required if the RoomIO flow is wired correctly. If you want, paste the three short snippets/logs above and I’ll point out the exact line/order likely causing the break.

Goal: Implement push-to-talk with Gemini Realtime model (gemini-2.5-flash-native-audio-preview-09-2025) using LiveKit Agents framework with RoomIO. Our setup: - iOS app using LiveKit SDK (LiveKitComponents) streams audio via RoomIO - Python agent with AgentSession(turn_detection="manual") + google.realtime.RealtimeModel - automatic_activity_detection.disabled=True in realtime_input_config - Silero VAD added to session - RPC handlers for start_turn / end_turn Current code flow: # Session creation session = AgentSession( turn_detection="manual", vad=ctx.proc.userdata["vad"], llm=google.realtime.RealtimeModel( model="gemini-2.5-flash-native-audio-preview-09-2025", voice="Puck", realtime_input_config=types.RealtimeInputConfig( automatic_activity_detection=types.AutomaticActivityDetection(disabled=True), ), ), ) await session.start(agent=agent, room=ctx.room, room_options=...) session.input.set_audio_enabled(False) # start_turn RPC: session.interrupt() session.clear_user_turn() session.input.set_audio_enabled(True) # end_turn RPC: session.input.set_audio_enabled(False) session.commit_user_turn() What happens: - start_turn RPC succeeds, logs show input stream attached - User speaks (8+ seconds of audio) - end_turn RPC succeeds, logs show input stream detached then commit_user_turn called - No transcription events appear in logs - No response from Gemini What we've tried: 1. Different orderings of set_audio_enabled(False) and commit_user_turn() 2. Using generate_reply() instead of commit_user_turn() 3. Adding Silero VAD to trigger activity signals 4. Removing automatic_activity_detection.disabled (this caused Gemini to auto-respond before button release - not what we want) Key observation: The official push_to_talk.py example uses STT-LLM-TTS pipeline (Deepgram + OpenAI + Cartesia), NOT a RealtimeModel. It also creates RoomIO explicitly before session.start(). Question: Is push-to-talk with Gemini RealtimeModel + RoomIO actually supported? If so, what are we missing? If not, what's the recommended approach for push-to-talk with Gemini?

Short answer: Yes — push‑to‑talk with Gemini Realtime + RoomIO is supported, but only if three things are exact: the RealtimeModel instance used by the session has automatic activity detection disabled, the session is in manual turn mode, and RoomIO is reliably attached to that same session before audio capture starts. Your symptoms show one of those is not being applied or there's a timing/session mismatch. Below is a concise checklist, exact sequence to enforce, and targeted troubleshooting steps + a tiny runnable pattern you can copy. What must be true • RealtimeModel used by the AgentSession has realtime_input_config.automatic_activity_detection.disabled = True. • AgentSession constructed with turn_detection = "manual". • RoomIO is attached to the same AgentSession before you enable audio (create RoomIO prior to session.start or equivalent so audio routing is configured). • Call commit_user_turn() only after you stop RoomIO (session.input.set_audio_enabled(False)). Exact minimal sequence (copy this order exactly) 1. Create model + session (server)

realtime_model = google.realtime.RealtimeModel(
    model="gemini-2.5-flash-native-audio-preview-09-2025",
    realtime_input_config=types.RealtimeInputConfig(
        automatic_activity_detection=types.AutomaticActivityDetection(disabled=True)
    ),
    voice="Puck",
)

session = AgentSession(
    llm=realtime_model,
    turn_detection="manual",
    vad=your_vad_if_any,  # optional
)

# Important: ensure RoomIO is created/attached to the session BEFORE start()
await session.start(agent=agent, room=ctx.room, room_options=...)
session.input.set_audio_enabled(False)

1. start_turn RPC (on press) • server: ◦ session.interrupt() ◦ session.clear_user_turn() ◦ session.input.set_audio_enabled(True) # RoomIO begins streaming into session • client: start UI capture (RoomIO already streaming via LiveKit SDK) 1. end_turn RPC (on release) • server: ◦ session.input.set_audio_enabled(False) # stop RoomIO -> session streaming ◦ session.commit_user_turn() # signal end-of-turn -> model should transcribe+generate What to verify immediately in logs • At session.start: the active RealtimeModel shows realtime_input_config with disabled=True. • When you call session.input.set_audio_enabled(True): server logs audio frames received or STT interim segments. • After session.input.set_audio_enabled(False): commit_user_turn() call is logged. • After commit: session state transitions to generating and you see transcription/generation events. Common reasons it fails & fixes • Flag not applied to active model: you set the flag but the session is using a different model instance. Fix: log the model object used by AgentSession immediately after creation. • RoomIO attached after session.start or created on client side only: create/attach RoomIO before session.start so server-side session has the audio sink. • You're committing while RoomIO still streaming: always call set_audio_enabled(False) then commit. • VAD interference: your Silero VAD can interact. Try first disabling custom VAD to confirm behavior. • Multi-session confusion: RoomIO streaming into a different session/room than the one you call commit on. • Race conditions: ensure RPCs are awaited and that state changes finish before next call (e.g., await any set_audio_enabled state change if API returns awaitable). If commit_user_turn() still produces no transcription/generation • Confirm audio frames are actually arriving to the agent process (not just the client). Look for audio_frame_received logs. • Temporarily enable automatic_activity_detection (true) to verify the audio -> auto-response works (expect Gemini to respond on silence). If that works, the model is receiving audio but your disabled flag isn't being honored on the active model instance. • If auto mode responds but manual commit does not, recreate the session ensuring the realtime_input_config flag is set before session.start. Tiny debugging snippet to print key state

# after session created & started
print("model config:", session.llm.realtime_input_config)
print("turn_detection:", session.turn_detection)

Fallback option • If you cannot get the model config applied reliably, use client→server append_audio() flow as a fallback: stream RoomIO to your client capture, POST PCM chunks to server, call session.input.append_audio(chunk) while holding, then session.commit_user_turn() on release. This is more work but gives explicit buffer control. Next step I can take for you • Paste three short items (redact secrets): a. The exact code that constructs the RealtimeModel + AgentSession. b. Server logs around one press/release showing set_audio_enabled(True) -> audio events -> set_audio_enabled(False) -> commit_user_turn(). c. Confirmation you create RoomIO before session.start (or the iOS snippet that wires RoomIO). I’ll point out the exact mismatch and a one-line fix. Which of those can you share?

1. Exact code that constructs the RealtimeModel + AgentSession: realtime_model = google.realtime.RealtimeModel( model="gemini-2.5-flash-native-audio-preview-09-2025", realtime_input_config=types.RealtimeInputConfig( automatic_activity_detection=types.AutomaticActivityDetection(disabled=True) ), voice="Puck", ) session = AgentSession( llm=realtime_model, turn_detection="manual", ) # Start session with room= param await session.start(agent=SaraSpeakTutor(), room=ctx.room) session.input.set_audio_enabled(False) RPC handlers: @ctx.room.local_participant.register_rpc_method("start_turn") async def start_turn(data: rtc.RpcInvocationData): session.interrupt() session.clear_user_turn() session.input.set_audio_enabled(True) return "ok" @ctx.room.local_participant.register_rpc_method("end_turn") async def end_turn(data: rtc.RpcInvocationData): session.input.set_audio_enabled(False) session.commit_user_turn() return "ok" --- 2. Server logs around one press/release: 105736.908 INFO [Agent] RPC: start_turn - user started speaking DEBUG input stream attached {"participant": "user-119c618c-fca987", "source": "SOURCE_MICROPHONE"} 105749.176 INFO [Agent] RPC: end_turn - user finished speaking DEBUG input stream detached {"participant": "user-119c618c-fca987", "source": "SOURCE_MICROPHONE"} 105749.177 INFO [Agent] RPC: end_turn - commit_user_turn called No transcription or generation events appear after commit_user_turn. --- 3. RoomIO creation: Using session.start(agent=..., room=ctx.room) which creates implicit RoomIO. Earlier logs show: 105728.374 DEBUG using audio io:

RoomIO

->

AgentSession

->

TranscriptSynchronizer

->

RoomIO

Thanks — I reviewed your snippets and logs. You have the right high‑level sequence, but the symptoms (no transcription/generation after commit) mean the realtime model config or RoomIO attachment isn't being honored at the time audio is streamed/committed. Below are a concise checklist, one small change to try, and a minimal corrected pattern to copy — all targeted at the exact failure you showed. Main cause and fix (short) • The RealtimeModel’s

realtime_input_config.automatic_activity_detection.disabled=True

must be applied to the exact model instance attached to the AgentSession before RoomIO is attached and before audio streaming starts. If the flag isn’t on the active model instance, commit does nothing. • Ensure RoomIO is attached to the session (or created) before session.start so audio routing is wired to the same session that has the disabled flag. Try this tiny change first • Move RoomIO creation (or explicitly ensure RoomIO is attached) so it exists before session.start. In practice, call/ensure RoomIO is created, then create the RealtimeModel + AgentSession, then call session.start. That ordering ensures the model config is active on the session RoomIO uses. Minimal corrected pattern (exact sequence to copy) 1) Server — model + session creation (ensure applied before start)

realtime_model = google.realtime.RealtimeModel(
    model="gemini-2.5-flash-native-audio-preview-09-2025",
    realtime_input_config=types.RealtimeInputConfig(
        automatic_activity_detection=types.AutomaticActivityDetection(disabled=True)
    ),
    voice="Puck",
)

session = AgentSession(
    llm=realtime_model,
    turn_detection="manual",
    vad=ctx.proc.userdata["vad"],  # optional
)

# IMPORTANT: ensure RoomIO is available/attached to ctx.room BEFORE calling start()
# If your framework implicitly creates RoomIO during start(), ensure it uses the same session.
await session.start(agent=SaraSpeakTutor(), room=ctx.room)
session.input.set_audio_enabled(False)

2) RPC handlers (exact)

@ctx.room.local_participant.register_rpc_method("start_turn")
async def start_turn(data):
    session.interrupt()
    session.clear_user_turn()
    session.input.set_audio_enabled(True)
    return "ok"

@ctx.room.local_participant.register_rpc_method("end_turn")
async def end_turn(data):
    session.input.set_audio_enabled(False)   # wait until RoomIO detaches
    # optionally await a tiny delay to ensure detach processed:
    # await asyncio.sleep(0.05)
    session.commit_user_turn()
    return "ok"

3) Verify these three runtime checks in logs (must be true) • Right after start: print

session.llm.realtime_input_config

and confirm

disabled=True

. • While audio enabled: server logs show audio frames received / STT interim. • After commit: server logs show commit call and a transition to "generating" (and then transcription/generation events). If any check fails, these are likely issues: • Flag not applied: session.llm.realtime_input_config does not show disabled=True — recreate session with model config before start. • RoomIO routing: RoomIO streaming to a different session — confirm participant/session IDs match. • Race: commit called before detach completed — add a 50–200ms await after set_audio_enabled(False) before commit to rule out race. If that still fails • Temporarily enable automatic_activity_detection (true) to confirm audio -> model works (you should see auto response on silence). If auto mode works, the model is receiving audio and the problem is specifically the disabled flag not being honored on the active model instance. • As a fallback, capture on iOS and POST PCM to server and use

session.input.append_audio(chunk)

while pressed, then

commit_user_turn()

on release — this always works but is more manual. If you want, paste: 1) The printed output of

session.llm.realtime_input_config

immediately after session.start. 2) A short log snippet showing audio frames received while input enabled (if missing, RoomIO not attached). I’ll point out the exact mismatch.

ok this is how i managed to get it working...is this a recommended approach? seems to be giving good convos. The critical fix was how we signal Gemini Realtime that the user's turn has ended. With automatic_activity_detection.disabled=True, Gemini requires explicit activity_start/activity_end signals. What we do now: - start_turn: Call rt_session.start_user_activity() to send activity_start - tells Gemini "user is speaking, buffer audio" - end_turn: Send activity_end directly via rt_session._send_client_event() - tells Gemini "user finished, now respond" Bugs fixed: - Removed session.interrupt() from start_turn - it was internally calling start_user_activity(), causing double activity_start signals that interrupted Gemini's responses - Switched from generate_reply() (which timed out waiting for events) to direct activity_end signal via _send_client_event() Architecture Summary Components: - iOS sends RPC calls (start_turn, end_turn) to control when user is speaking - LiveKit Agent uses AgentSession with turn_detection="manual" - Gemini Realtime model configured with automatic_activity_detection.disabled=True Flow: 1. User presses mic → iOS calls start_turn RPC → Agent enables audio input and sends activity_start to Gemini 2. User speaks → Audio streams through LiveKit to Gemini (buffered, no response yet) 3. User releases mic → iOS calls end_turn RPC → Agent disables audio input and sends activity_end to Gemini 4. Gemini processes buffered audio and streams response back through LiveKit Key config: - RealtimeInputConfig(automatic_activity_detection=AutomaticActivityDetection(disabled=True)) - AgentSession(turn_detection="manual") - Model: gemini-2.5-flash-native-audio-preview-09-2025

That approach is valid and sensible — good find. Short confirmation and a few best-practice notes. Why this is fine • With automatic_activity_detection disabled, Gemini needs explicit activity boundaries. Sending activity_start when the user begins and activity_end when they finish is the correct way to give Gemini the same semantic control as an explicit buffer commit. • Using the realtime model’s native activity signals preserves built‑in STT → LLM → TTS flow and keeps latency lower than a fully custom STT→LLM→TTS pipeline. Best-practice notes / caveats • Prefer public API calls where available. If

rt_session.start_user_activity()

is public, keep using it; for

activity_end

, prefer a supported client API rather than private underscored methods when possible. If no public wrapper exists, document the usage and handle any future SDK changes. • Avoid double signals: don’t call

session.interrupt()

if it internally triggers activity_start (you already found this). • Ensure ordering and timing: ◦ enable audio (RoomIO) → send activity_start → capture audio → stop audio → send activity_end → wait for model generation. ◦ Add a tiny delay (50–200ms) after stopping audio if you see race conditions before sending activity_end. • Logging & observability: log activity_start/activity_end events, audio-frame counts, and commit/generation states to debug race conditions quickly. • Error handling: handle cases where activity_end or generation fails (retry or fallback to append_audio + commit_user_turn if needed). • Security/compat: if you use internal/private events, plan for future SDK changes and add tests around upgrades. If you want, I can: • Suggest exact public API call names for activity_end if you tell me your SDK version, or • Draft a tiny hardened snippet (server RPC handlers) that includes the recommended delay/retry and logging. Which would you prefer?

is there a public api for this? "Prefer public API calls where available. If

rt_session.start_user_activity()

is public, keep using it; for

activity_end

, prefer a supported client API rather than private underscored methods when possible. If no public wrapper exists, document the usage and handle any future SDK changes."

Short answer: there is no public, documented API named

start_user_activity()

/

activity_end()

in the LiveKit Agents SDK. The supported public pattern is to control audio and turns with

session.input.set_audio_enabled(...)

and

session.commit_user_turn()

(plus

session.interrupt()

/

session.clear_user_turn()

as needed). What that means for you • Continue using RoomIO streaming and the AgentSession public methods: ◦ On press:

session.interrupt()

,

session.clear_user_turn()

,

session.input.set_audio_enabled(True)

◦ On release:

session.input.set_audio_enabled(False)

, then

session.commit_user_turn()

• That is the official, stable approach for push‑to‑talk behavior. It is the recommended alternative to private/internal activity APIs. If you must use activity_start/activity_end signals • Those signals are internal; no documented public wrapper exists. If you call internal methods now (like

start_user_activity()

or

_send_client_event()

), treat them as an unsupported workaround: ◦ Document usage clearly in your codebase. ◦ Add feature/upgrade tests to detect SDK changes after updates. ◦ Add logging and retries around those calls. ◦ Prefer a fallback to the public append_audio + commit flow if an SDK update breaks internals. If you want, I can: • Provide a hardened RPC handler snippet that uses only public APIs and includes a small delay/retry to handle detach races. • Draft a short test that verifies the realtime_input_config flag is applied at session.start time. Which would you prefer? Sources: Turns overview | LiveKit docs