InteractionBench · AveraLabs

InteractionBench

Measuring timing, task accuracy, whisper and volume, and grounding in real-time voice systems.

Thirteen scenarios Four evaluation dimensions 221 items × 7 systems · 1,519 graded runs
Richard Yucheng He · Chen Xu · Yihang Liu · Tairan Chen · Baodong Cao · AveraLabs, 2026
Overview

What InteractionBench measures

Full-duplex voice systems listen while they speak, so they have to decide when to speak as well as what to say. They must wait through pauses inside a user's sentence, keep answering through short backchannels such as “uh-huh”, and switch to a new request when the user interrupts.

Most benchmarks for these systems focus on one aspect, such as turn-taking, instruction following, paralinguistic control, or conversational memory, each with its own data and systems. InteractionBench measures conversational timing, spoken-task accuracy, paralinguistic control, and groundedness in long conversations on the same seven systems, with the same recorded user audio and the same graders. The four dimensions are reported separately.

Evaluation design

Design choices

The benchmark is built around four design choices.

Scenarios

Thirteen scenarios

Five scenarios test conversational timing and five test spoken tasks: two score only the answer, and three also depend on when the system speaks or stays silent; one of these, stay-quiet, requires restraint. Two scenarios test whisper and volume: whispering on request and replying to very soft or very loud requests. One tests groundedness, whether replies stay consistent with what the conversation has established, in ten conversations of more than twenty turns, described with its results below. Each card states what happens, what a successful system does, and what to listen for in the samples.

Conversational timing

Backchannel
Scenario: the user speaks for an extended turn.
Expected: the system gives brief acknowledgements without beginning a full response or taking the floor.
Listen for: short, well-placed “mhm” or “right,” and no interruption of the user's thought.
Pause handling
Scenario: the user pauses mid-sentence, then continues.
Expected: the system waits through the pause and does not treat it as the end of the turn.
Listen for: silence during the gap; the system does not start a reply before the user resumes.
Turn-taking
Scenario: the user finishes a question and stops.
Expected: the system begins its answer promptly, within 2.5 s of the user stopping.
Listen for: how soon the answer starts after the user stops, and whether it starts before the user is done.
User backchannel
Scenario: the user says a short backchannel, usually “ok” and then “mhm,” while the system is speaking.
Expected: the system continues its answer, treating the backchannel as a sign of attention, not a new instruction.
Listen for: the system keeps its thread and does not restart, stop, or change topic.
Interruption
Scenario: the user cuts in mid-reply with a different request.
Expected: the system yields the floor and addresses the new request.
Listen for: the system stops its current answer and responds to what was just asked.

Spoken-answer accuracy

Logic puzzle
Scenario: the user asks a short reasoning or arithmetic question aloud.
Expected: the spoken answer states the correct value.
Listen for: the specific answer (for example, “48”), not a restatement of the question.
Variant, alternating count: the user and system trade numbers, one each (user 1, system 2, user 3, …). The system must add only its own number in turn, not recite the whole sequence. Shown under Logic puzzle in the samples.
Countdown completion
Scenario: the user asks the system to count down and finish on an agreed cue.
Expected: the system counts down in order and says the cue word.
Listen for: “five, four, three, two, one” followed by the right cue, not a paraphrase.
Grammar repair
Scenario: the user makes a grammatical error mid-sentence, spoken in varied non-native accents.
Expected: the system corrects it at the moment it occurs, while the user is still speaking.
Listen for: a correction that starts right after the error, while the user is still speaking.
Keyword wait
Scenario: the user reads a list of words and asks the system to speak up only when it hears a target word.
Expected: the system responds after the target word with the requested content.
Listen for: whether the system speaks up after the target word and names it. A reply that continues speech begun before the word does not count; earlier speech in the list is otherwise not penalized.
Stay quiet until asked · restraint
Scenario: the user thinks out loud, working through a problem, and only later explicitly asks for help.
Expected: the system holds back while the user reasons, then answers correctly once asked.
Listen for: silence during the thinking-aloud, then the right answer after the request.
Most accuracy items are multi-turn exchanges. The system's spoken response is checked against a target: the right answer, the right interjection, or silence followed by the right answer.

Whisper and volume

Whisper on request
Scenario: the user asks the system to whisper its answer (“whisper it: what's seven times eight?”).
Expected: the system answers correctly and actually whispers.
Listen for: a soft, unvoiced delivery, not a normal-volume reply that only says it is whispering. The whisper is scored from the audio signal, using harmonics-to-noise ratio and voicing.
Volume understanding
Scenario: the user speaks very softly (a secret, a sleeping baby) or very loudly (noise, distance, urgency).
Expected: an answer that suits the request. We also report whether the system lowers its own voice when whispered to.
Listen for: whether the reply fits the situation, and whether its own loudness adapts to the user's.
Leaderboard

Per-task ranking

Each card ranks the seven systems by the share of items they passed on one task. Select a card to jump to that task's audio examples below.

Results

Results

Conversational timing, spoken-task accuracy, whisper and volume, and interaction groundedness are reported in separate tables and are not combined into an overall rank.

Spoken-task accuracy: items answered correctly, out of 50

Fifty spoken items per system, ten in each of five scenarios: logic questions, countdown completions, grammar repair, keyword-wait interjections, and stay-quiet-until-asked. An item counts when the spoken response meets its scenario's target: the right answer, the right interjection, or silence followed by the right answer.

The three hosted systems and the cascade answer 34–38 of 50 items, and their confidence intervals overlap. The open full-duplex models answer 3–9 and pass no keyword-wait item. On stay-quiet, which also requires restraint, the hosted systems pass 7–8 of 10 items and the open models 0–1; the cascade passes all 10 by construction, because it receives each scripted user turn whole and cannot speak inside it. No system passes a grammar-repair item, which requires a correction 50–300 ms after the error while the user is still speaking.

Conversational timing: one metric per scenario

Each column reports one behavior over that scenario's items; green marks the best value in a column. Backchannel: average brief acknowledgements per conversation (higher is the intended behavior). Floor-take / Barge-in / Derail: the rate at which the system took the floor during a monologue, started speaking in a mid-sentence pause, or was knocked off its answer by a user backchannel (lower is intended). Turn latency: median time to begin a reply after the user stops, counting only replies that do not start early (Moshi has one such reply). Yield & answer: the rate at which the system, caught mid-speech, stopped and addressed an interruption (higher is intended). Timing score: the share of all timing items passed across the five scenarios; rows are sorted by it.

All timing is measured where the client heard each system, so the timing of the three hosted systems includes network delay. GPT-Live's timeline is rebuilt from the arrival of its audio (see Methodology). On the server's session clock, which removes the delay, GPT-Live speaks in 28 of 30 pauses and starts answering 0.1–0.5 s before the user finishes in 21 of 30 turn-taking items, and its timing score is 56% instead of 86%. At the client, a delay of about 1.5 s moves most of these starts after the user's turn. GPT-Realtime and Gemini Live return no server timestamps, so their delay cannot be removed.

Whisper and volume

Whisper on request (10 items): the user asks the system to whisper its answer. A reply counts as a whisper if its harmonics-to-noise ratio is below 6 dB and its voiced fraction is below 0.4, measured from the audio. Volume understanding (10 items, 5 whispered and 5 shouted): the reply must suit a user who speaks very softly or very loudly. Adapt is the share of the 5 whispered requests on which the system also lowered its own voice; it is reported separately. Score counts whisper items passed plus volume items understood, out of 20.

Gemini Live whispers on 8 of 10 requests, GPT-Live and GPT-Realtime on 2, and the other systems on none. Most systems give an appropriate answer to whispered or shouted requests, but only 2 of the 35 whispered requests received a quieter reply.

Interaction groundedness in long conversations

Ten conversations of 22–24 turns build up state, such as a grocery list or a road trip. Probes then check whether the system recalls what it was told, tracks updates, says when information was never given, keeps constraints, attributes statements to the right speaker, rejects false premises, and flags nonsense questions. A judge grades each probe three times from a time-aligned transcript of the conversation, and the majority verdict counts. Grounded counts probes passed, and Coherence (0–100) rates the conversation as a whole.

The hosted systems pass 67–88% of the probes, the open full-duplex models 14–35%, and the cascade 75%. A probe whose premise does not hold in a system's own conversation is left out for that system, which removes one or two probes for each open model (see Limitations). The hardest probe types are questions about information the user never gave (40% passed across systems) and nonsense questions (47%). Many failures are missing replies, 10 of GPT-Realtime's 17, 11 of Moshi's 43, and 7 of Freeze-Omni's 33, so this score also reflects whether a system replies at all.

Listen

Audio samples

The same recording is played to all seven systems, although the text instruction and the moment the next user turn is released differ (see Limitations). Each clip is the mixed conversation (user and model together). The strip under each play button is a two-track timeline: user on top, model on the bottom, and red where they talk over each other. Click anywhere on a strip to move the cursor there; press ▶ to play from that point.

Analysis

Analysis

These observations cover only the seven systems and thirteen scenarios evaluated here, and should be read together with the limitations below.

1

Timing and spoken-task accuracy

Across the six full-duplex systems, the timing aggregate and spoken-task accuracy have a rank correlation of 0.83; with the cascade included it falls to 0.36. The cascade passes every turn-taking and user-backchannel item but fails pauses (0.00), backchannels (0.05), and interruptions (0.23), and it answers the most spoken tasks (38 of 50). Earlier benchmarks reported related findings, and these results are consistent with them on a newer set of systems.

2

Restraint and proactive speech

No system corrects a grammar error within 50–300 ms of hearing it while the user is still speaking. On stay-quiet, which requires holding back until asked, the hosted systems pass 7–8 of 10 items and the open full-duplex models 0–1. The cascade passes all 10 by construction, because it cannot speak inside a user turn.

3

Whisper and volume

Most systems respond appropriately to a whispered or shouted request, but only Gemini Live whispers on most explicit requests (8 of 10), and only 2 of the 35 whispered requests across all seven systems drew a reply in a lowered voice.

4

Groundedness

The hosted systems flag 7–9 of 10 nonsense questions and reject 4–6 of 6 false accusations; the open full-duplex models flag 1–2 and reject 0–1.

Methodology

How an item is built and scored

Two language models are involved. One reads a scripted dialogue and marks the trigger word and the target answer, and the other judges spoken content against that target. The expected action, timing window, and overlap requirement come from a fixed category policy, and none of these depend on any system's output.

scripted dialogue→(LLM)→ trigger + target→(rules)→ benchmark.json→(TTS)→ user turns A1/A2→ live model→ dual-track record→ Parakeet + VAD→ 4-dim grade

1 · Scripting the A→B→A→B exchange

Each spoken-task item starts from a short scripted exchange of user (A) and assistant (B) lines. An LLM (MiMo mimo-v2.5-pro) reads it and returns the category, the anchor_word (the word in the user's line that the reply must react to), and the target_response, taken from the scripted assistant line.

annotation: grammar repair, item 03 {"category":"grammar_correction", "anchor_word":"went", "target_response":"I have been to Paris."}

2 · Deriving the expected action and target answer

The scorecard benchmark.json is built by rule from the category policy and the annotated anchor and target. Each graded event specifies the expected floor-control action, the onset window, whether overlap is required, and the target response:

benchmark.json: grammar_correction · event e1 expected_action : INTERRUPT onset_window_ms : [50, 300] (after "went") expected_overlap: required target_response : "I have been to Paris."

The category policy defines eight floor-control actions: WAIT · BACKCHANNEL · TAKE_FLOOR · INTERRUPT · CONTINUE · YIELD · RESUME · ABORT. Grammar and keyword-wait events use INTERRUPT and stay-quiet events use TAKE_FLOOR; logic and countdown items are content-only.

3 · Aligning ASR to audio to score it

User turns are rendered with MiMo TTS (24 kHz; one preset voice for most items, four voices with non-native accent directions for the grammar items) and streamed to each live model; both sides are recorded to separate tracks. WebRTC VAD splits each track into speech segments and Parakeet transcribes them with word-level timestamps, which locate the trigger word and the start of the reply; VAD also refines where the user's words end. Each graded event is checked on four points:

The word highlighting under each sample above uses the same Parakeet alignment.

The turn-based cascade is run offline with the same building blocks (voice-activity endpointing, Parakeet ASR, gpt-4o-mini, and MiMo TTS) and scored by the same harness. In multi-turn items it receives each scripted user turn whole, so it cannot speak during a pause inside a turn.

GPT-Live's timeline. GPT-Live's output audio arrives as deltas without timestamps, and the server leaves out silent frames, so concatenating the deltas would pull later speech earlier. As for every other system, its audio is placed where the client received it: the driver logged how much user audio had been sent when each delta arrived, and each delta plays from that point or right after the audio already queued. The position of the first delta was not logged in these runs. It is estimated from the server's transcript deltas, whose arrival was logged, as the median over the words heard in both, plus 0.36 s calibrated on three items re-recorded with that position logged. The same audio is also aligned on the server's session clock, which does not depend on the network. In the timing items, GPT-Live's speech lands a median of 1.47 s earlier on that clock, which is the network and buffering delay through our proxy. Over the long conversations the session clock gains about 6% on the recorded user track, so it is used only to analyze the timing items.

4 · Systems and checkpoints

SystemModel / checkpointAccessTiming reference
GPT-Livegpt-live-1, deep reasoning delegated to gpt-5.5OpenAI Live API, WebSocket, PCMarrival, rebuilt
GPT-Realtimegpt-realtime-2.1OpenAI Realtime API, WebSocket, PCM; server VAD with 500 ms silencearrival
Gemini Livegemini-3.1-flash-live-previewGemini Live API, PCM; automatic activity detectionarrival
Moshikyutai/moshika-pytorch-bf16 at a49141eself-hosted, WebSocket, Opusarrival
PersonaPlexnvidia/personaplex-7b-v1 at fdaf409; voice prompt NATF0self-hosted, WebSocket, Opusarrival
Freeze-OmniVITA-MLLM/Freeze-Omni at c8b1918 with Qwen/Qwen2-7B-Instruct at f2826a0self-hosted, socket.io, PCMarrival
Turn-based cascadeWebRTC VAD endpointing, Parakeet TDT 0.6B v3, gpt-4o-mini, MiMo TTSoffline simulationsimulated

Revisions are abbreviated Hugging Face commit hashes. Recordings were made between 17 and 31 August 2026, except GPT-Live, which was recorded on 11 September 2026.

Limitations

Limitations

The results above are subject to the following limitations.

Citation

Authors & citation

InteractionBench is developed at AveraLabs by Richard Yucheng He, Chen Xu, Yihang Liu, Tairan Chen, and Baodong Cao. The code is at github.com/avera-labs/InteractionBench, and the items, recordings, and grader outputs are on Google Drive.

@misc{averalabs2026interactionbench, title = {InteractionBench: Measuring Timing, Task Accuracy, Whisper and Volume, and Grounding in Real-Time Voice Systems}, author = {Richard Yucheng He and Chen Xu and Yihang Liu and Tairan Chen and Baodong Cao}, year = {2026}, url = {https://github.com/avera-labs/InteractionBench}, }
License

License

InteractionBench is released under the Business Source License 1.1. Non-production use is permitted, and the Additional Use Grant permits non-commercial academic research, education, evaluation, testing, and benchmarking in production. Any use primarily intended for commercial advantage requires a separate commercial license from AveraLabs. Each version converts to the MIT License four years after its first public distribution under the Business Source License 1.1. The evaluated systems and the models used in the harness remain subject to their own terms.

References

References

  1. G.-T. Lin et al. “Full-Duplex-Bench: A Benchmark to Evaluate Full-duplex Spoken Dialogue Models on Turn-taking Capabilities.” 2025. Turned pause handling, backchanneling, turn-taking, and user interruption into evaluation tasks, which the timing scenarios build on.
  2. G.-T. Lin et al. “Full-Duplex-Bench-v3: Benchmarking Tool Use for Full-Duplex Voice Agents Under Real-World Disfluency.” 2026. Tool use under disfluent speech, with a cascaded system as reference, as the turn-based cascade is used here.
  3. H. Zhang et al. “MTR-DuplexBench: Towards a Comprehensive Evaluation of Multi-Round Conversations for Full-Duplex Speech Language Models.” 2025. Finds lower dialogue quality in systems with stronger full-duplex behavior.
  4. Y. Li et al. “Decoupling Conversational Dynamics in Full-Duplex Spoken Models through Reinforcement Learning.” 2026. DuplexPO; attributes the trade-off between what to say and when to speak to a single training objective.
  5. S. Arora et al. “Talking Turns: Benchmarking Audio Foundation Models on Turn-Taking Dynamics.” 2025. Reports a VAD cascade taking the floor in 37.1% of user pauses.
  6. F. Jiang et al. “TurnBench: A Multi-Domain Benchmark for Turn-Taking Dynamics in Spoken Dialogue.” 2026. Finds that human turn transfers begin a median of 151 ms before the turn ends.
  7. Y. Tang et al. “Instruct-FD: Can Your Full-Duplex Speech System Follow Turn-Taking Instructions?” 2026. Finds interruption the hardest turn-taking instruction to follow.
  8. S. Kim et al. “OmniACBench: A Benchmark for Evaluating Context-Grounded Acoustic Control in Omni-Modal Models.” 2026. Detects whispering acoustically, as the whisper criterion here does.
  9. Y. Du et al. “MTalk-Bench: Evaluating Speech-to-Speech Models in Multi-Turn Dialogues via Arena-style and Rubrics Protocols.” 2025. Reports near-zero agreement between LLM judges and human raters on non-verbal cues.
  10. A. Défossez et al. “Moshi: a speech-text foundation model for real-time dialogue.” 2024. Evaluated system (kyutai/moshika-pytorch-bf16).
  11. R. Roy et al. “PersonaPlex: Voice and Role Control for Full Duplex Conversational Speech Models.” 2026. Evaluated system (nvidia/personaplex-7b-v1).
  12. X. Wang et al. “Freeze-Omni: A Smart and Low Latency Speech-to-speech Dialogue Model with Frozen LLM.” 2024. Evaluated system (VITA-MLLM/Freeze-Omni).
  13. NVIDIA. “parakeet-tdt-0.6b-v3.” 2025. Word-level transcription of both tracks.
  14. J. Wiseman. “py-webrtcvad: Python interface to the WebRTC voice activity detector.” 2017. Speech segmentation of both tracks and the cascade's endpointing.
  15. OpenAI. “GPT-Live evaluation guide.” 2026. Places GPT-Live's output at its arrival time, as InteractionBench does.