56 KiB
Realtime Duplex API
vLLM-Omni serves full-duplex speech models over a WebSocket endpoint,
/v1/realtime, that speaks the OpenAI Realtime vocabulary plus a
set of duplex extensions: the model decides when to listen and when to speak,
the user can talk over the assistant, playback progress is reported back so
history stays honest, and a dropped connection can resume the same session.
This page covers how to run a duplex deployment, how to drive it from Python
with vllm_omni.clients.duplex.DuplexClient, and the complete wire contract.
The handler is enabled by either a model-native session_mode: duplex
deployment or an explicit duplex_session configuration for turn-based
Server VAD. The runtime architecture is described in
Full-Duplex Runtime (MiniCPM-o 4.5).
Quick Start
Start the Server
vllm-omni serve openbmb/MiniCPM-o-4_5 \
--omni \
--deploy-config vllm_omni/deploy/minicpmo_4_5.yaml \
--trust-remote-code \
--host 0.0.0.0 --port 8099
vllm_omni/deploy/minicpmo_4_5.yaml declares session_mode: duplex and
duplex_session.max_sessions: 4, which mounts the WebSocket routes
ws://<host>:8099/v1/realtime (this page) and ws://<host>:8099/v1/duplex
(the native dialect for raw-protocol clients) next to the usual HTTP API.
PersonaPlex and Nemotron VoiceChat use vllm_omni/deploy/personaplex.yaml and
vllm_omni/deploy/nemotron_labs_voicechat_duplex.yaml in the same way.
Run the Example Client
examples/online_serving/barge_in_client.py asks a question, interrupts the
answer mid-stream, and writes one WAV per response:
python examples/online_serving/barge_in_client.py \
--url ws://127.0.0.1:8099/v1/realtime \
--model openbmb/MiniCPM-o-4_5 \
--ref-audio /path/to/reference_voice.wav \
--question-wav question_16k.wav \
--interrupt-wav follow_up_16k.wav \
--output-dir ./duplex_out
Input WAVs must be mono 16 kHz PCM16. --ref-audio is required by the
MiniCPM-o preset (the session is otherwise rejected with
ref_audio_required). --preset personaplex switches the session shape to
the PersonaPlex preset. The flow is diagrammed in
examples/online_serving/barge_in_client_flow.md.
Using DuplexClient
vllm_omni.clients.duplex is a client-side package: it depends only on the
standard library, pybase64, and the websockets package, and it is never
imported by server code. Full signatures are on the
API reference page.
Open a session
DuplexClient is an async context manager. Entering it connects, sends
session.update, and waits for session.created; exiting sends
session.close and waits for session.closed.
from vllm_omni.clients.duplex import DuplexClient, audio_data_url
from vllm_omni.clients.minicpmo_4_5 import create_duplex_session_config
config = create_duplex_session_config(ref_audio=audio_data_url(Path("reference_voice.wav")))
async with DuplexClient("ws://127.0.0.1:8099/v1/realtime", model="openbmb/MiniCPM-o-4_5", config=config) as client:
print(client.session_id, client.session_info["capabilities"])
The session shape comes from SessionConfig; model-specific knobs ride in
its extra_body, so the client itself stays model-neutral and each model
ships a preset:
| Preset | Input / output audio | What it sets |
|---|---|---|
vllm_omni.clients.minicpmo_4_5.create_duplex_session_config(ref_audio=...) |
16 kHz pcm16 / 24 kHz pcm16 |
extra_body.native_duplex=True (opt in to the model-native lane), force_listen_count=0, overlap_policy="listen_only", playback_commit_policy="ack_only"; ref_audio is the assistant voice clip |
vllm_omni.clients.personaplex.create_duplex_session_config(voice="NATF2.pt", persona="") |
24 kHz pcm_f32le / 24 kHz pcm16 |
bundled .pt voice prompt and the persona as instructions |
SessionConfig(...) |
16 kHz pcm16 / 24 kHz pcm16 |
model-neutral defaults; pass extra_body, turn_detection, overlap_policy, playback_commit_policy, instructions, voice, temperature yourself |
Keyword overrides on a preset replace the corresponding SessionConfig
field, e.g. create_duplex_session_config(ref_audio=..., temperature=0.6).
Constructor options: session_id (names the session the client creates; it
does not resume or take over an existing session — entering the client always
performs the session.update handshake), reconnect (a ReconnectPolicy,
default five attempts with 0.25–4 s jittered backoff; None disables
auto-resume), heartbeat_interval_s (default 30 s; None disables the lease
heartbeat), handshake_timeout_s, and connect (a custom transport factory,
used by tests and benchmarks). Resume is supported only as automatic reconnect
within the same DuplexClient instance (see
Reconnect and resume); re-attaching from a new client
or process requires the wire-level session.resume handshake
(resume_token, incarnation, last_received_server_event_seq), which this
client does not expose.
Stream audio and commit a turn
from vllm_omni.clients.duplex import read_pcm16_wav
question = read_pcm16_wav(Path("question_16k.wav"))
await client.stream_pcm(question, chunk_ms=100) # paced 100 ms appends
await client.commit(create_response=False) # the model decides listen/speak
stream_pcm slices PCM into chunk_ms appends and paces them in real time
(realtime=False sends as fast as possible). append_audio sends one chunk
and accepts is_speech and video_frames (base64 JPEG/PNG strings riding
the append that closes a one-second model unit — data URLs are accepted and
their prefix stripped; stream_pcm(video_frames=..., stacked_video_frames=...) interleaves a frame track automatically and
returns the number of frames sent). A turn is ended with commit, never by
a flag on the append.
commit seals the buffered utterance into a user item. With the default
auto_response=True a commit lets the model respond; pass
create_response=False to leave the listen/speak decision entirely to a
model-native session, or set auto_response=False in the config and call
client.send({"type": "response.create"}) when you want an answer. In the
model-native lane the audio is already streaming into the model before the
commit, so the model may start answering — or emit a listen decision —
without any commit at all.
Consume responses
responses() demultiplexes the event stream into one ResponseHandle per
assistant response. A model listen decision surfaces as an already-finished
handle with decision == "listen".
async for response in client.responses():
if response.decision == "listen":
continue # the model kept listening
async for chunk in response.audio(): # decoded PCM16 at 24 kHz
play(chunk)
await client.ack_playback(response.played_ms, response_id=response.response_id)
await response.wait()
print(response.transcript, response.done_event.raw["response"]["status"])
break
ResponseHandle exposes response_id, decision ("speak" / "listen"),
the accumulated transcript and text, played_ms, created_event,
done_event, finished, audio() and wait().
ack_playback(played_ms, response_id=..., committed_ms=...) reports the
cumulative playback position. Call it while audio plays: with the MiniCPM-o
preset's playback_commit_policy="ack_only" the assistant turn enters the
conversation history only as far as you acknowledge it, which is what keeps
history truthful after an interruption. acknowledge_collected_playback(client, collector) acknowledges everything an EventCollector has received, which is
what the example and the benchmarks do at the end of a turn.
Interrupt
await client.cancel_response() # cancel the active response (epoch advances)
await client.clear_input() # drop un-committed input audio
In the model-native lane you usually do not need either: keep streaming
microphone audio while the assistant speaks and the server's overlap policy
decides — a short remark is acknowledged, meaningful overlap is deferred to
the turn end, and on models that advertise supports_barge_in (MiniCPM-o
4.5) the in-flight response is cancelled. Each decision is reported as an
overlap.decision event. cancel_response and clear_input compose a
client-forced barge-in for clients that run their own VAD.
Work with raw events
events() (or async for event in client) yields every server event from
now on as a typed DuplexEvent (SessionCreated, ResponseCreated,
AudioDelta, TranscriptDelta, TextDelta, ListenDecision,
SpeakDecision, ResponseDone, SessionClosed, SessionExpired,
ErrorEvent, and ConnectionResumed after an automatic resume; unknown
types arrive as plain DuplexEvent). Common accessors are type,
event_id, session_id, response_id, item_id, server_event_seq,
audio (decoded bytes), text, and raw (the wire dict).
wait_for("response.done", timeout_s=30) blocks for the next event of the
given type(s), and send({...}) is the escape hatch for any client event
the wrapper does not have a method for (response.create,
conversation.item.create, conversation.item.truncate, ...); it returns
the stamped event_id.
Reconnect and resume
On a transport drop the client reconnects with the ReconnectPolicy backoff
and sends session.resume from the last acknowledged server_event_seq.
Replayed events are deduplicated, consumers see one ConnectionResumed
event, and client.incarnation / client.resume_token are refreshed. The
server keeps a detached session alive for disconnect_grace_s (30 s by
default); after that a resume fails with session_resume_expired. Resume is
gated by capabilities.supports_session_resume (MiniCPM-o 4.5 and Nemotron
VoiceChat: yes; PersonaPlex: no). With reconnect=None the same drop surfaces
as DuplexSessionClosedError.
Errors
Server error events are delivered as ErrorEvent (with code, message,
related_event_id) and raised as DuplexProtocolError from the awaiting
call; transport failures raise DuplexConnectionError; a session that ends
while you are awaiting it raises DuplexSessionClosedError. All three derive
from DuplexClientError.
Measure latency
EventCollector accumulates events for assertions and metrics:
from vllm_omni.clients.duplex import EventCollector
collector = EventCollector()
consume_task = asyncio.create_task(collector.consume(client))
commit_sent_at = time.monotonic()
await client.commit(create_response=False)
...
summary = collector.timing_summary(after_s=commit_sent_at, input_committed_at_s=commit_sent_at)
timing_summary reports client-observed first-text / first-audio latency,
audio cadence, and, when the deployment attaches engine stage metrics to the
response events (the benchmark requests them with
extra_body.return_stage_metrics=True), the Stage 0 token metrics
(ttft_ms, tpot_ms, itls_ms). It reports raw measurements only —
derived metrics such as the audio RTF are computed by the caller from
audio_generation_ms / audio_duration_ms (the benchmarks use
vllm_omni.metrics.definitions.compute_audio_rtf), keeping metric
definitions out of the dependency-free client. audio_bytes(response_id), count(type),
errors(), and first_received_at(type) cover the common assertions; the
openai-realtime-duplex benchmark backend (vllm-omni bench serve) is built
on the same helpers.
API Reference
This section is the complete wire contract of /v1/realtime: how
the dialect relates to the OpenAI Realtime protocol, which surfaces each
model's capability flags gate, a catalogue of every client-to-server and
server-to-client event, and a JSON example for each. DuplexClient speaks
exactly this contract; raw WebSocket clients (any OpenAI Realtime client, or
your own) can use it directly. The ordering and cardinality rules behind it
are specified in the normative contract section of
Full-Duplex Runtime (MiniCPM-o 4.5); a
compatibility change must update this section together with that table and
its protocol tests.
Endpoint and transport
The endpoint is ws(s)://<host>/v1/realtime. Bare connections select the
session handler when the deployment declares either session_mode: duplex
or duplex_session. ?duplex=1 remains a compatibility alias, while
?duplex=0 selects the legacy Realtime handler. Optional query parameters
are model, session_id,
autostart (0 means resume-only), resume, and native_duplex
(minicpmo45_native_duplex is accepted as a deprecated alias and folded
into the canonical name). Every message is one JSON object per WebSocket
text frame, discriminated by type. Inbound events are applied in arrival
order through one per-session mailbox; outbound events preserve that order
and carry a monotonically increasing server_event_seq, which is the replay
cursor for session.resume. Client events may carry an event_id, echoed in
error.error.event_id when the event is rejected.
Session identity is (session_id, incarnation); attachment_generation
names the current socket; epoch is the interruption fence that advances on
every cancel, clear, or barge-in; turn_id counts committed model turns;
response_id (item id item_<response_id>) names one assistant utterance.
Tier in the tables below: 1 means identical to OpenAI Realtime in name and semantics; 2 means an OpenAI event name carrying vLLM-Omni extensions that a stock OpenAI client ignores; 3 means vLLM-Omni only, with no OpenAI counterpart.
Capability negotiation by model
The event vocabulary is uniform, but several surfaces are gated by the
capabilities object the server returns in session.created; a client must
branch on those flags rather than on the model name. The current model
plugins advertise:
| Capability | MiniCPM-o 4.5 | PersonaPlex | Nemotron VoiceChat | Gated surface |
|---|---|---|---|---|
implementation_level |
model_native_duplex (when extra_body.native_duplex=true) |
model_native_duplex |
model_native_duplex |
model-owned response.listen / response.speak; the chat-fallback lane otherwise |
chunk_period_ms |
1000 | 80 | 80 | the model unit that response.listen decisions and camera frames align to |
supports_session_resume |
yes | no | yes | session.resume, session.resumed, session.replaced, replay after a transport drop |
supports_barge_in |
yes | no | no | barge_in, turn.signal{event:"barge_in"}, overlap_policy=barge_in_on_speech, turn_detection.server_vad |
supports_audio_truncate |
yes | no | no | conversation.item.truncate and truncating playback.ack adjusting the stored assistant item |
video input (video_frames on append) |
consumed by Stage 0 | ignored | ignored | omni camera track |
| tool calls | no | no | yes | response.function_call_arguments.*, function_call items |
Everything else in the catalogue — session lifecycle, heartbeat and event acknowledgement, append/commit/clear, the response envelope, playback acknowledgement, and the error envelope — behaves identically for every model.
Compatibility with the OpenAI Realtime protocol
The Realtime dialect is, by design, an OpenAI-Realtime-compatible surface:
RealtimeInputTranslator / RealtimeOutputProjector exist to translate
the native duplex vocabulary onto OpenAI's schema so stock Realtime clients
work unmodified. The overlap falls into the three tiers used by the event catalogue below.
Tier 1: identical to OpenAI Realtime
| Direction | Messages |
|---|---|
| Client → server | session.update, input_audio_buffer.append / .commit / .clear, conversation.item.create / .retrieve / .truncate / .delete, response.create, response.cancel, output_audio_buffer.clear (OpenAI: WebRTC-only; here also a WebSocket event) |
| Server → client | error (same {type, code, message, event_id, param} envelope and the same classes invalid_request_error / server_error / rate_limit_error), session.created, session.updated, input_audio_buffer.committed / .cleared / .speech_started / .speech_stopped, conversation.item.created / .added / .done / .deleted / .retrieved / .truncated, conversation.item.input_audio_transcription.completed, response.created, response.done, response.output_item.added / .done, response.content_part.added / .done, response.output_audio.delta / .done, response.output_audio_transcript.delta / .done, response.output_text.delta / .done, response.function_call_arguments.delta / .done, rate_limits.updated, output_audio_buffer.cleared |
| Session fields | model, modalities / output_modalities, instructions, voice, input_audio_format / output_audio_format (beta spelling) and audio.input.format / audio.output.format objects (GA spelling), turn_detection.server_vad {threshold, prefix_padding_ms, silence_duration_ms}, input_audio_transcription, input_audio_noise_reduction, tools, tool_choice, temperature, max_response_output_tokens, speed, tracing |
| Object shapes | realtime.session, realtime.item, realtime.response; item content parts input_text / input_audio / output_text / output_audio; function_call and function_call_output items; previous_item_id chaining; output_index / content_index addressing |
| Sequencing | response.created → output_item.added → content_part.added → deltas → *.done → content_part.done → output_item.done → response.done → rate_limits.updated; speech_started → speech_stopped → committed |
Audio/transcript output uses only the current (non-beta) OpenAI event names
(response.output_audio.delta, not the deprecated response.audio.delta),
matching openai.types.realtime in the official openai Python SDK and what
clients built against it (e.g. livekit-plugins-openai's non-Azure code path)
expect; a client still on the old beta SDK types would not recognize these
events. Session fields and conversation.item.* events, in contrast, are
deliberately dual: the server emits and accepts both the beta and the GA
spellings at once (output_modalities alongside modalities,
conversation.item.added alongside conversation.item.created) so either
generation of OpenAI client parses the stream.
Tier 2: OpenAI names carrying vLLM-Omni extensions
A stock client ignores the extra keys; the extensions are additive.
| Message | Extension |
|---|---|
session.created / session.updated |
top-level incarnation, attachment_generation, resume_token; inside session: state, turn_state, epoch, turn_id, active_request_id, active_response_id, active_response_turn_id, overlap_policy, overlap_*_ms/rms, playback_commit_policy, playback, capabilities, ref_audio, extra_body, idle_timeout_s, response_format |
response.created / response.done / response.listen |
response_id at top level; the raw duplex event under response.metadata (duplex_event); status_details.reason uses vLLM reasons (barge_in, client_cancelled, new_response, …) |
response.output_audio.delta |
format, sample_rate_hz, metadata{session_id, epoch, model_speak, end_of_turn, audio_duration_ms, audio_text_marks, playback} |
response.speak (inserted before the first delta) |
not an OpenAI event, but rides the OpenAI response envelope (response_id, item_id, output_index, content_index) |
input_audio_buffer.append |
is_speech, video_frames (+ max_slice_nums), duration_ms, audio_end_ms, per-event format / sample_rate_hz |
input_audio_buffer.commit |
final, response_create, is_speech:false (silence declaration) |
input_audio_buffer.committed |
the native input.committed event wrapped under event (turn_id, epoch, history_len, message, response_create_deferred) |
conversation.item.* server events |
echo of the originating event under event |
error |
the error codes are vLLM-Omni's (OpenAI standardises only the class); handshake-stage errors use the flat native shape {type:"error", error:"…", code} |
Semantic divergences hidden behind shared names:
- Commit ≠ response. In the native lane audio streams into the model
before any commit; a commit may end in
response.listenand no response, and the model may open a response with no commit at all. OpenAI: commit ⇒ item,response.create⇒ exactly one response. - With
turn_detection={"type":"server_vad"}, omittedinterrupt_responsedefaults tofalseon turn-based models such as Qwen3-Omni, enabling endpointing and automatic responses without interruption. Explicittrueis rejected there. Model-native duplex keeps thetruedefault and rejectsfalse; its VAD impliesoverlap_policy=barge_in_on_speech. This capability-dependent default differs from OpenAI'struedefault; the effective configuration is returned insession.created/session.updated.semantic_vadis unsupported. rate_limits.updatedis always an empty list (compatibility only).- Cancellation never reuses a
response_id; truncation is driven byplayback.ackas well asconversation.item.truncate.
Tier 3: vLLM-Omni only
| Area | Messages / fields |
|---|---|
| Session lifetime | session.heartbeat / session.heartbeat_ack, session.event_ack, session.close / session.closed, session.resume / session.resumed / session.replaced / session.expired / session.resync_required; server_event_seq, resume_token, incarnation, attachment_generation; query params ?duplex=1, autostart, resume, native_duplex |
| Turn-taking | response.speak, response.listen, overlap.decision, overlap_policy, barge_in, turn.signal, input.cancel, input.text.append, epoch / turn_id fencing, force_listen |
| Playback truth | playback.ack / playback.acknowledged, playback_commit_policy (ack_only | commit_all_on_done), playback{generated_ms, sent_ms, played_ms, committed_ms}, history_committed, audio_text_marks |
| Model negotiation | capabilities{implementation_level, supports_input_append, supports_barge_in, supports_session_resume, chunk_period_ms, input_modes, …}, ref_audio, extra_body (auto_response, native_duplex — deprecated alias minicpmo45_native_duplex is folded into it, force_listen_count, duplex_initial_user_text) |
| Diagnostics | runtime.control, duplex.function_call.done, the full error-code vocabulary (REALTIME_ERROR_TYPES_BY_CODE in vllm_omni/entrypoints/duplex/realtime_state.py) |
Consequences for clients
- Event names are roughly 70 % OpenAI's; payloads are OpenAI's plus additive extensions; everything that makes the endpoint full-duplex — model-owned turn-taking, playback truth, interruption fencing, resumable sessions — lives in Tier 3.
- An unmodified OpenAI Realtime client can hold a conversation
(commit-driven or
turn_detection: server_vad), but it will never seeresponse.listen/response.speak, cannot resume after a transport drop, will not receive replay after reconnect, and its server-side history is only as honest ascommit_all_on_doneallows because it never sendsplayback.ack. - A duplex-aware client (
vllm_omni.clients.duplex.DuplexClient) uses the Tier 3 surface on top of the same Tier 1 events, so both kinds of client can share one deployment and one event log.
Turn-based Chat audio metadata
For turn-based models served through Chat fallback, non-empty audio choices
produced by the Omni Chat audio encoder include audio_metadata, in both
streaming and complete Chat responses. Text-only choices do not require it.
| Field | Meaning |
|---|---|
format |
Encoding of the returned audio, such as pcm or wav; pcm16 requests normalize to pcm. |
sample_rate_hz |
Sample rate used by the encoder, in samples per second. |
frame_count |
Samples per channel in this chunk, or in the complete waveform for a non-streaming choice. Not a byte count. |
channels |
Number of audio channels. |
These values describe the waveform after audio transformations. Compressed
formats may add codec padding that is not included in frame_count.
The encoder uses a model-reported sample rate when available; Qwen3-Omni
currently omits it and uses the existing 24 kHz default. This metadata does
not independently determine a model's sample rate.
Chat fallback requests the session's output format explicitly. Its internal
response.output_audio.delta carries format, sample_rate_hz, channels,
and audio_duration_ms. The Realtime endpoint exposes output format and
sample_rate_hz on response.audio.delta, with duration in
metadata.audio_duration_ms; it does not currently forward channels.
The duration is cumulative for the response:
integer frames are summed before converting to milliseconds and rounding down.
Non-empty audio without valid metadata, or a sample-rate change within one
response, fails that fallback response with response_error.
In this Chat fallback path, response.done.response.metadata.playback reports:
generated_ms: cumulative source-waveform duration.sent_ms: cumulative audio accepted into the server output path, including queued or journaled audio. A journaled event remains accepted if the subsequent send fails; this is not proof of socket delivery.played_ms: cumulative playback reported by the client.committed_ms: playback position used for history reconciliation.
The native model path still records audio before sending; it does not yet share Chat fallback's output-acceptance timing. Only client playback reports establish what was played. The existing duration-proportional text truncation remains an approximation, not word-level audio alignment.
Event catalogue
Every event type the Realtime route accepts or emits, with a worked example in the next section: 21
client→server and 42 server→client. Aliases (push_chunk,
input.audio.append, input_text.append, push_text, signal_turn,
audio.playback_ack, close_session, close, session_close) share the
payload of their canonical event and are not listed separately; the alias
table lives in the normative contract section of Full-Duplex Runtime (MiniCPM-o 4.5).
Client to server
| Event type | Tier | Description |
|---|---|---|
session.update |
2 | Open handshake on first send (creates the session); later sends reconfigure it. OpenAI name; session object carries vLLM-Omni keys (ref_audio, overlap_policy, playback_commit_policy, extra_body, ...). |
session.heartbeat |
3 | Keepalive; refreshes the engine session lease. |
session.event_ack |
3 | Acknowledges received server events by server_event_seq; trims the resume replay journal. |
session.close |
3 | Graceful close; answered with session.closed. |
session.resume |
3 | Re-attach a live session on a new socket (resume_token, incarnation, last_received_server_event_seq). |
input_audio_buffer.append |
2 | Append one audio chunk. OpenAI name; extended with is_speech, video_frames, duration_ms, audio_end_ms, per-event format/sample_rate_hz. |
input_audio_buffer.commit |
2 | Seal the buffered utterance into a user item. OpenAI name; extended with final, response_create, is_speech:false. |
input.commit |
3 | Alias of input_audio_buffer.commit accepted by the runner. |
input_audio_buffer.clear |
1 | Drop un-committed input audio. |
input.cancel |
3 | Cancel pending input; advances epoch. |
input.text.append |
3 | Append text to the open input item (chat-fallback lane only). |
conversation.item.create |
1 | Add a completed user item (text, audio, or function_call_output) to history. |
conversation.item.retrieve |
1 | Fetch a stored item by id. |
conversation.item.delete |
1 | Delete a stored item by id. |
conversation.item.truncate |
1 | Truncate an assistant item's audio/transcript at audio_end_ms. |
response.create |
1 | Explicitly request a response with per-response overrides. |
response.cancel |
1 | Cancel the active (or named) response; advances epoch. |
output_audio_buffer.clear |
1 | Discard queued output audio; advances epoch (OpenAI: WebRTC-only, here also WebSocket). |
barge_in |
3 | Explicit hard interrupt; requires capabilities.supports_barge_in. |
turn.signal |
3 | External turn-taking signal (e.g. event:"barge_in"); same capability gate. |
playback.ack |
3 | Report cumulative playback progress (played_ms, committed_ms, optional truncate). |
Server to client
| Event type | Tier | Description |
|---|---|---|
session.created |
2 | Session opened. OpenAI name; adds incarnation, attachment_generation, resume_token and vLLM-Omni keys inside session (epoch, turn_id, playback, capabilities, ...). |
session.updated |
2 | Echo of the effective session config after every session.update (same extended session object). |
session.heartbeat_ack |
3 | Reply to session.heartbeat. |
session.closed |
3 | Last event on the socket; reason ∈ session_close, timeout, disconnect, ... |
session.resumed |
3 | Resume accepted; carries the new attachment_generation and rotated resume_token; journaled events are replayed after it. |
session.replaced |
3 | Sent to the superseded socket when another socket resumes the session. |
session.expired |
3 | Engine lease reaped (disconnect_grace_expired, idle_ttl_expired); socket closes after it. |
session.resync_required |
3 | Replay impossible (journal_gap / journal_overflow); client must start a new session. |
input_audio_buffer.speech_started |
1 | Speech onset detected (server VAD, client hint, or RMS heuristic). |
input_audio_buffer.speech_stopped |
1 | Speech end detected. |
input_audio_buffer.committed |
2 | Commit accepted. OpenAI name; wraps the native input.committed payload under event (turn_id, epoch, response_create_deferred, ...). |
input_audio_buffer.cleared |
1 | Input buffer cleared (after input_audio_buffer.clear or input.cancel). |
conversation.item.added |
2 | Item entered history (GA spelling). OpenAI name; carries previous_item_id and echoes the originating event. |
conversation.item.created |
2 | Same as conversation.item.added in the beta spelling; both are emitted. |
conversation.item.done |
2 | Item finalized (GA spelling). |
conversation.item.input_audio_transcription.completed |
1 | Transcript of a committed user audio item (model-produced). |
conversation.item.deleted |
2 | Reply to conversation.item.delete (echoes the request under event). |
conversation.item.retrieved |
1 | Reply to conversation.item.retrieve. |
conversation.item.truncated |
2 | Reply to conversation.item.truncate (echoes the request under event). |
response.created |
2 | A visible assistant response starts. OpenAI name; top-level response_id, raw duplex event under response.metadata. |
response.output_item.added |
1 | Assistant item (or function_call item) attached to the response. |
response.content_part.added |
1 | Audio/text content part opened on the assistant item. |
response.speak |
3 | Model chose to speak (native lane); at most once per response, before the first audio delta. |
response.output_audio.delta |
2 | One ordered audio chunk. OpenAI name; adds format, sample_rate_hz, metadata{epoch, model_speak, end_of_turn, audio_text_marks, playback}. |
response.output_audio_transcript.delta |
1 | Transcript text paired one-to-one with each audio delta. |
response.output_text.delta |
1 | Text delta for text-modality responses. |
response.output_text.done |
1 | Final text of a text-modality response. |
response.output_audio.done |
1 | Audio stream closed for the response. |
response.output_audio_transcript.done |
1 | Full transcript (concatenation of the deltas). |
response.content_part.done |
1 | Content part finalized with its transcript/text. |
response.output_item.done |
1 | Assistant item finalized. |
response.done |
2 | Terminal event; status ∈ completed | cancelled | failed. OpenAI name; vLLM-Omni status_details.reason values and raw duplex event under metadata. |
rate_limits.updated |
1 | Compatibility event after every response.done; always an empty list. |
response.listen |
3 | Model chose to keep listening at a unit boundary or after a silence commit. Carries response_id when the decision terminates a precreated response (a response.done with the same id follows); otherwise no response.created precedes it. |
response.function_call_arguments.delta |
1 | Tool-call arguments delta (Nemotron VoiceChat). |
response.function_call_arguments.done |
1 | Tool-call arguments complete. |
duplex.function_call.done |
3 | Raw fallback when a model tool call is malformed. |
output_audio_buffer.cleared |
1 | Reply to output_audio_buffer.clear. |
playback.acknowledged |
3 | Reply to playback.ack; returns the playback ledger and history_committed. |
overlap.decision |
3 | How an append that overlapped an active response was handled (drop | listen | barge_in). |
runtime.control |
3 | Diagnostic: an engine control signal was partially unsupported. |
error |
2 | Error envelope. OpenAI shape and classes; vLLM-Omni error codes. |
Message examples
Field values are illustrative; base64 payloads are shortened to "…".
C→S = client sends, S→C = server sends. Every S→C event on the wire
additionally carries "server_event_seq": <int> (omitted below for brevity).
Session lifecycle
C→S session.update (first one = open handshake)
{
"type": "session.update",
"event_id": "evt_0001",
"session": {
"model": "openbmb/MiniCPM-o-4_5",
"session_id": "sess_demo_01",
"modalities": ["audio", "text"],
"instructions": "You are a concise voice assistant.",
"input_audio_format": "pcm16",
"output_audio_format": "pcm16",
"ref_audio": "data:audio/wav;base64,…",
"turn_detection": null,
"overlap_policy": "listen_only",
"playback_commit_policy": "ack_only",
"temperature": 0.7,
"idle_timeout_s": 300,
"extra_body": {
"auto_response": true,
"native_duplex": true,
"force_listen_count": 0
}
}
}
S→C session.created
{
"type": "session.created",
"incarnation": 0,
"attachment_generation": 1,
"resume_token": "rt_3f9c…",
"session": {
"object": "realtime.session",
"type": "realtime",
"id": "sess_demo_01",
"model": "openbmb/MiniCPM-o-4_5",
"state": "open",
"turn_state": "idle",
"epoch": 0,
"turn_id": 0,
"active_request_id": null,
"active_response_id": null,
"active_response_turn_id": null,
"modalities": ["audio", "text"],
"output_modalities": ["audio", "text"],
"instructions": "You are a concise voice assistant.",
"voice": null,
"response_format": "pcm16",
"input_audio_format": "pcm16",
"output_audio_format": "pcm16",
"audio": {
"input": {"format": {"type": "audio/pcm", "rate": 16000}, "sample_rate_hz": 16000},
"output": {"format": {"type": "audio/pcm", "rate": 24000}}
},
"temperature": 0.7,
"max_tokens": null,
"speed": null,
"idle_timeout_s": 300.0,
"overlap_policy": "listen_only",
"overlap_short_ack_ms": 700,
"overlap_barge_in_ms": 1200,
"overlap_silence_rms": 0.003,
"playback_commit_policy": "ack_only",
"turn_detection": null,
"input_audio_transcription": null,
"tracing": null,
"playback": {"generated_ms": 0, "sent_ms": 0, "played_ms": 0, "committed_ms": 0},
"capabilities": {
"implementation_level": "model_native_duplex",
"supports_input_append": true,
"supports_barge_in": true,
"supports_session_resume": true,
"supports_session_lease": true,
"supports_multi_session": true,
"chunk_period_ms": 1000,
"input_modes": ["append_audio_chunk"],
"signal_sources": ["model_native", "client_event", "server_policy"],
"target_barge_in_latency_ms": null
}
}
}
S→C session.updated (same session object; follows the first
session.created and every later session.update)
{"type": "session.updated", "session": {"object": "realtime.session", "id": "sess_demo_01", "…": "…"}}
C→S session.heartbeat / S→C session.heartbeat_ack
{"type": "session.heartbeat"}
{"type": "session.heartbeat_ack", "session_id": "sess_demo_01"}
C→S session.event_ack
{"type": "session.event_ack", "server_event_seq": 48}
C→S session.close / S→C session.closed
{"type": "session.close"}
{"type": "session.closed", "session_id": "sess_demo_01", "reason": "session_close"}
C→S session.resume (new socket, ?duplex=1&autostart=0)
{
"type": "session.resume",
"session_id": "sess_demo_01",
"incarnation": 0,
"resume_token": "rt_3f9c…",
"last_received_server_event_seq": 40
}
S→C session.resumed (activation; the token is rotated, then journaled
events with server_event_seq > 40 are replayed in order)
{
"type": "session.resumed",
"session_id": "sess_demo_01",
"incarnation": 0,
"attachment_generation": 2,
"resume_token": "rt_a71e…"
}
S→C session.replaced (sent to the superseded socket, which is then closed)
{"type": "session.replaced", "session_id": "sess_demo_01", "attachment_generation": 1}
S→C session.expired
{"type": "session.expired", "session_id": "sess_demo_01", "incarnation": 0, "reason": "disconnect_grace_expired"}
S→C session.resync_required
{"type": "session.resync_required", "session_id": "sess_demo_01", "reason": "journal_gap"}
Input
C→S input_audio_buffer.append (200 ms of 16 kHz PCM16 = 6400 bytes)
{
"type": "input_audio_buffer.append",
"event_id": "evt_0042",
"audio": "…",
"input_audio_format": "pcm16",
"sample_rate_hz": 16000,
"duration_ms": 200,
"audio_end_ms": 1200,
"is_speech": true
}
with camera frames (omni video; rides the append that closes a 1 s model
unit; each entry is a bare base64 JPEG/PNG — no data: URL prefix)
{
"type": "input_audio_buffer.append",
"audio": "…",
"input_audio_format": "pcm16",
"sample_rate_hz": 16000,
"duration_ms": 200,
"audio_end_ms": 1200,
"video_frames": ["/9j/4AAQSkZJRg…", "/9j/4AAQSkZJRg…"]
}
S→C input_audio_buffer.speech_started / speech_stopped
{"type": "input_audio_buffer.speech_started", "audio_start_ms": 1040, "item_id": "item_9b2d…"}
{"type": "input_audio_buffer.speech_stopped", "audio_end_ms": 2860, "item_id": "item_9b2d…"}
C→S input_audio_buffer.commit
{"type": "input_audio_buffer.commit", "event_id": "evt_0060", "final": true}
alias form accepted by the runner (input.commit)
{"type": "input.commit", "final": true}
silence/noise declaration (no runtime append; answered with response.listen)
{"type": "input_audio_buffer.commit", "final": true, "is_speech": false}
S→C commit fan-out (in this order)
{
"type": "conversation.item.added",
"previous_item_id": null,
"item": {
"id": "item_9b2d…",
"object": "realtime.item",
"type": "message",
"role": "user",
"status": "completed",
"content": [{"type": "input_audio", "transcript": "what is the weather today"}]
}
}
{"type": "conversation.item.created", "previous_item_id": null, "item": {"id": "item_9b2d…", "…": "…"}}
{
"type": "input_audio_buffer.committed",
"previous_item_id": null,
"item_id": "item_9b2d…",
"event": {
"type": "input.committed",
"session_id": "sess_demo_01",
"turn_id": 1,
"epoch": 0,
"history_len": 1,
"message": {"role": "user", "content": [{"type": "audio_url", "audio_url": {"url": "native-duplex:input-audio"}}]},
"realtime_item_id": "item_9b2d…"
}
}
{
"type": "conversation.item.input_audio_transcription.completed",
"item_id": "item_9b2d…",
"content_index": 0,
"transcript": "what is the weather today"
}
{"type": "conversation.item.done", "previous_item_id": null, "item": {"id": "item_9b2d…", "status": "completed", "…": "…"}}
deferred commit during an active response (event.response_create_deferred)
{
"type": "input_audio_buffer.committed",
"previous_item_id": "item_9b2d…",
"item_id": "item_c044…",
"event": {"type": "input.committed", "session_id": "sess_demo_01", "turn_id": 2, "epoch": 0, "history_len": 3, "response_create_deferred": true, "message": {"…": "…"}}
}
C→S input_audio_buffer.clear / S→C input_audio_buffer.cleared
{"type": "input_audio_buffer.clear"}
{"type": "input_audio_buffer.cleared"}
C→S input.cancel (advances epoch; also answered with input_audio_buffer.cleared)
{"type": "input.cancel"}
C→S input.text.append
{"type": "input.text.append", "text": "and tomorrow?"}
C→S conversation.item.create (text turn, or a tool result)
{
"type": "conversation.item.create",
"previous_item_id": null,
"item": {
"type": "message",
"role": "user",
"content": [{"type": "input_text", "text": "Summarise what I said."}]
}
}
{
"type": "conversation.item.create",
"item": {"type": "function_call_output", "call_id": "call_7d1f", "output": "{\"temperature_c\": 24}"}
}
C→S conversation.item.delete / S→C conversation.item.deleted
{"type": "conversation.item.delete", "item_id": "item_9b2d…"}
{"type": "conversation.item.deleted", "item_id": "item_9b2d…", "event": {"type": "conversation.item.deleted", "item_id": "item_9b2d…"}}
C→S conversation.item.retrieve / S→C conversation.item.retrieved
{"type": "conversation.item.retrieve", "item_id": "item_resp_01"}
{"type": "conversation.item.retrieved", "item": {"id": "item_resp_01", "object": "realtime.item", "type": "message", "role": "assistant", "status": "completed", "content": [{"type": "output_audio", "transcript": "It is sunny, 24 degrees."}]}}
C→S conversation.item.truncate / S→C conversation.item.truncated
{"type": "conversation.item.truncate", "item_id": "item_resp_01", "content_index": 0, "audio_end_ms": 1850}
{"type": "conversation.item.truncated", "item_id": "item_resp_01", "content_index": 0, "audio_end_ms": 1850, "event": {"…": "…"}}
Response lifecycle
C→S response.create
{
"type": "response.create",
"event_id": "evt_0070",
"response": {
"modalities": ["audio", "text"],
"instructions": "Answer in one sentence.",
"output_audio_format": "pcm16",
"temperature": 0.6,
"max_output_tokens": 256,
"metadata": {"trace": "turn-3"}
}
}
S→C response.created
{
"type": "response.created",
"response_id": "resp_01",
"response": {
"id": "resp_01",
"object": "realtime.response",
"status": "in_progress",
"status_details": null,
"output": [],
"modalities": ["audio", "text"],
"metadata": {"duplex_event": {"type": "response.created", "session_id": "sess_demo_01", "response_id": "resp_01", "epoch": 0, "turn_id": 1}}
}
}
S→C conversation.item.added / conversation.item.created (assistant item, status: in_progress)
{
"type": "conversation.item.added",
"previous_item_id": "item_9b2d…",
"item": {"id": "item_resp_01", "object": "realtime.item", "type": "message", "role": "assistant", "status": "in_progress", "content": []}
}
S→C response.output_item.added
{"type": "response.output_item.added", "response_id": "resp_01", "output_index": 0, "item": {"id": "item_resp_01", "object": "realtime.item", "type": "message", "role": "assistant", "status": "in_progress", "content": []}}
S→C response.content_part.added
{"type": "response.content_part.added", "response_id": "resp_01", "item_id": "item_resp_01", "output_index": 0, "content_index": 0, "part": {"type": "audio", "transcript": ""}}
S→C response.speak
{
"type": "response.speak",
"response_id": "resp_01",
"item_id": "item_resp_01",
"output_index": 0,
"content_index": 0,
"metadata": {"session_id": "sess_demo_01", "epoch": 0, "model_speak": true}
}
S→C response.output_audio.delta (one ~1 s unit, 24 kHz PCM16)
{
"type": "response.output_audio.delta",
"response_id": "resp_01",
"item_id": "item_resp_01",
"output_index": 0,
"content_index": 0,
"delta": "…",
"format": "pcm16",
"sample_rate_hz": 24000,
"metadata": {
"session_id": "sess_demo_01",
"epoch": 0,
"model_speak": true,
"end_of_turn": false,
"audio_duration_ms": 1000,
"audio_text_marks": [{"text_chars": 11, "audio_end_ms": 1000}],
"playback": {"generated_ms": 1000, "sent_ms": 1000, "played_ms": 0, "committed_ms": 0}
}
}
S→C response.output_audio_transcript.delta (exactly one per audio delta)
{"type": "response.output_audio_transcript.delta", "response_id": "resp_01", "item_id": "item_resp_01", "output_index": 0, "content_index": 0, "delta": "It is sunny"}
S→C response.output_text.delta / response.output_text.done (text modality)
{"type": "response.output_text.delta", "response_id": "resp_02", "item_id": "item_resp_02", "output_index": 0, "content_index": 0, "delta": "Tomorrow looks"}
{"type": "response.output_text.done", "response_id": "resp_02", "item_id": "item_resp_02", "output_index": 0, "content_index": 0, "text": "Tomorrow looks cloudy."}
S→C response.output_audio.done / response.output_audio_transcript.done
{"type": "response.output_audio.done", "response_id": "resp_01", "item_id": "item_resp_01", "output_index": 0, "content_index": 0}
{"type": "response.output_audio_transcript.done", "response_id": "resp_01", "item_id": "item_resp_01", "output_index": 0, "content_index": 0, "transcript": "It is sunny, 24 degrees."}
S→C response.content_part.done
{"type": "response.content_part.done", "response_id": "resp_01", "item_id": "item_resp_01", "output_index": 0, "content_index": 0, "part": {"type": "audio", "transcript": "It is sunny, 24 degrees."}}
S→C response.output_item.done / conversation.item.done
{"type": "response.output_item.done", "response_id": "resp_01", "output_index": 0, "item": {"id": "item_resp_01", "object": "realtime.item", "type": "message", "role": "assistant", "status": "completed", "content": [{"type": "output_audio", "transcript": "It is sunny, 24 degrees."}]}}
{"type": "conversation.item.done", "previous_item_id": "item_9b2d…", "item": {"id": "item_resp_01", "status": "completed", "…": "…"}}
S→C response.done (completed)
{
"type": "response.done",
"response_id": "resp_01",
"response": {
"id": "resp_01",
"object": "realtime.response",
"status": "completed",
"status_details": null,
"output": [{"id": "item_resp_01", "object": "realtime.item", "type": "message", "role": "assistant", "status": "completed", "content": [{"type": "output_audio", "transcript": "It is sunny, 24 degrees."}]}],
"metadata": {"type": "response.done", "session_id": "sess_demo_01", "response_id": "resp_01", "epoch": 0, "committed": true, "playback": {"generated_ms": 2400, "sent_ms": 2400, "played_ms": 2400, "committed_ms": 2400}}
}
}
S→C response.done (cancelled by barge-in)
{
"type": "response.done",
"response_id": "resp_01",
"response": {"id": "resp_01", "object": "realtime.response", "status": "cancelled", "status_details": {"type": "cancelled", "reason": "barge_in"}, "output": [{"id": "item_resp_01", "status": "cancelled", "…": "…"}], "metadata": {"…": "…"}}
}
S→C rate_limits.updated
{"type": "rate_limits.updated", "rate_limits": []}
S→C response.listen
{
"type": "response.listen",
"session_id": "sess_demo_01",
"epoch": 0,
"response": {"object": "realtime.response", "status": "listening", "metadata": {"type": "response.listen", "session_id": "sess_demo_01", "epoch": 0, "reason": "model_listen"}}
}
C→S response.cancel
{"type": "response.cancel", "response_id": "resp_01"}
C→S output_audio_buffer.clear / S→C output_audio_buffer.cleared
{"type": "output_audio_buffer.clear"}
{"type": "output_audio_buffer.cleared", "response_id": "resp_01"}
C→S barge_in / turn.signal
{"type": "barge_in"}
{"type": "turn.signal", "event": "barge_in", "payload": {"source": "client_vad"}}
function-call response (Nemotron VoiceChat tools)
{"type": "response.output_item.added", "response_id": "resp_fc_01", "output_index": 0, "item": {"id": "item_fc_01", "object": "realtime.item", "type": "function_call", "status": "completed", "name": "get_weather", "call_id": "call_7d1f", "arguments": "{\"city\":\"Hong Kong\"}"}}
{"type": "response.function_call_arguments.delta", "response_id": "resp_fc_01", "item_id": "item_fc_01", "output_index": 0, "call_id": "call_7d1f", "delta": "{\"city\":\"Hong Kong\"}"}
{"type": "response.function_call_arguments.done", "response_id": "resp_fc_01", "item_id": "item_fc_01", "output_index": 0, "call_id": "call_7d1f", "arguments": "{\"city\":\"Hong Kong\"}"}
raw fallback when a model tool call is malformed (call_id/name missing)
{"type": "duplex.function_call.done", "event": {"type": "function_call.done", "session_id": "sess_demo_01", "epoch": 0, "call_id": null, "name": null, "arguments": "{"}}
Playback
C→S playback.ack / S→C playback.acknowledged
{"type": "playback.ack", "response_id": "resp_01", "item_id": "item_resp_01", "played_ms": 1000, "committed_ms": 1000}
{
"type": "playback.acknowledged",
"event": {
"type": "playback.acknowledged",
"session_id": "sess_demo_01",
"epoch": 0,
"item_id": "item_resp_01",
"played_ms": 1000,
"committed_ms": 1000,
"truncate": false,
"playback": {"generated_ms": 2400, "sent_ms": 2400, "played_ms": 1000, "committed_ms": 1000},
"history_committed": false
}
}
truncating ack after an interruption (what was actually heard)
{"type": "playback.ack", "response_id": "resp_01", "played_ms": 1850, "committed_ms": 1850, "truncate": true}
Overlap and barge-in
S→C overlap.decision
{
"type": "overlap.decision",
"session_id": "sess_demo_01",
"epoch": 0,
"policy": "listen_only",
"action": "listen",
"reason": "long_overlap_speech",
"overlap_ms": 1400,
"buffer_audio": true,
"defer_runtime_append": true,
"force_listen": true
}
{"type": "overlap.decision", "session_id": "sess_demo_01", "epoch": 0, "policy": "barge_in_on_speech", "action": "barge_in", "reason": "server_vad_utterance_active"}
server-VAD session (session.update fragment) and the annotated append the
translator forwards internally (shown for reference; clients never see it)
{"type": "session.update", "session": {"turn_detection": {"type": "server_vad", "threshold": 0.5, "prefix_padding_ms": 300, "silence_duration_ms": 500}, "overlap_policy": "barge_in_on_speech"}}
{"type": "input_audio_buffer.append", "audio": "…", "format": "pcm16", "sample_rate_hz": 16000, "is_speech": true, "force_listen": true, "vad": {"backend": "silero", "is_speech": true, "speech_active": true, "speech_started": true, "speech_stopped": false, "speech_probability": 0.93}}
S→C runtime.control (only when an engine signal was partially unsupported)
{"type": "runtime.control", "session_id": "sess_demo_01", "epoch": 1, "result": {"…": "redacted engine control result"}}
Errors
S→C error — Realtime shape
{
"type": "error",
"error": {
"type": "invalid_request_error",
"code": "input_audio_buffer_empty",
"message": "input_audio_buffer.commit requires a non-empty input audio buffer.",
"event_id": "evt_0060"
}
}
{"type": "error", "error": {"type": "invalid_request_error", "code": "unsupported_turn_detection", "message": "turn_detection.interrupt_response=false is unsupported; use turn_detection=null for model-owned listen/speak", "event_id": "evt_0001", "param": "turn_detection"}}
{"type": "error", "error": {"type": "server_error", "code": "runtime_append_failed", "message": "stale epoch: expected 1, got 0"}}
{"type": "error", "error": {"type": "rate_limit_error", "code": "duplex_session_busy", "message": "duplex session capacity reached (max_sessions=4)"}}
Handshake-stage errors (before the Realtime projector is bound) use the flat native shape:
{"type": "error", "error": "Unknown or expired duplex session: sess_demo_01", "code": "session_resume_expired"}
Coverage
The examples above cover every event type the Realtime route emits or
accepts (verified by diffing the type literals in
vllm_omni/entrypoints/duplex/ against this document). Not shown, because
they never reach a /v1/realtime?duplex=1 client unprojected: the native
/v1/duplex dialect (session.create / open_session / session.config,
input.committed, input.cancelled, audio.cancelled,
response.output_audio.delta, response.text.delta, response.message,
function_call.done) — their Realtime projections are the examples above
(see the name map at the end of this section) — and the runner-internal markers __timeout__,
__disconnect__, __replaced_attachment__. Input aliases (push_chunk,
input.audio.append, input_text.append, push_text, signal_turn,
audio.playback_ack, close_session, close, session_close) share the
payload of their canonical event (see the alias table in Full-Duplex Runtime (MiniCPM-o 4.5)).
Native to Realtime name map
The native /v1/duplex dialect is what the session runner produces; the
Realtime projector renames and fans out these events before they reach a
/v1/realtime?duplex=1 client.
| Native / internal event | Realtime projection |
|---|---|
session.create (open_session, session.config) |
first session.update |
input.committed |
conversation.item.added / conversation.item.created, input_audio_buffer.committed, conversation.item.input_audio_transcription.completed, conversation.item.done |
input.cancelled |
input_audio_buffer.cleared |
response.output_audio.delta (with audio_transcript) |
optional response.speak, then response.output_audio.delta and response.output_audio_transcript.delta |
response.output_audio.done / response.output_text.done |
response.output_audio.done / response.output_text.done |
response.text.delta |
response.output_text.delta |
response.message (chat-fallback raw chunk) |
passed through |
audio.cancelled |
optional output_audio_buffer.cleared, then the cancelled terminal events and response.done with status: "cancelled" |
function_call.done |
response.created, response.output_item.added (function_call item), response.function_call_arguments.delta / .done, response.output_item.done, response.done |
playback.acknowledged |
playback.acknowledged (wrapped verbatim under event) |
runtime.control, session.resumed, session.replaced, session.expired, session.resync_required, session.heartbeat_ack |
passed through unchanged |
Known Limitations
- Several surfaces are capability-gated per model (see Capability negotiation by model above): PersonaPlex does not support session resume, barge-in, or audio truncation; Nemotron VoiceChat does not support barge-in or audio truncation; camera frames are consumed only by MiniCPM-o 4.5; tool calls are produced only by Nemotron VoiceChat.
turn_detectionsupportsserver_vadornull, notsemantic_vad. Turn-based models support endpointing without interruption, includingcreate_response=falsefor explicit response creation. Model-native duplex requiresinterrupt_response=trueandcreate_response=true.input_audio_transcriptionandinput_audio_noise_reductionare accepted and echoed but no separate transcription or noise-reduction stage runs; transcripts come from the model.rate_limits.updatedis emitted for compatibility only and always carries an empty list.- Session capacity is bounded by
duplex_session.max_sessionsin the deploy configuration; admission beyond it fails withduplex_session_busyorresource_exhausted. - The native
/v1/duplexroute speaks the internal dialect and does not supportsession.resume; use/v1/realtime?duplex=1for the contract on this page.