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Simulating thousands of conversations to test a voice agent requires moving away from manual QA and adopting an automated, agent-to-agent simulation pipeline . Borrowing a page from autonomous vehicle testing, the modern approach uses a "Simulated User" powered by an LLM to talk to your "Agent Under Test" at scale, supervised by an automated "Judge".
The standard architecture and step-by-step framework to execute this effectively involve several key components:
Step-by-Step Implementation Strategy
Happy Path: Direct, cooperative user completing a transaction.
Adversarial/Edge Case: User changes their mind mid-sentence, speaks in fragments, or goes off-topic.
Acoustic Variance: Inject background noise profiles or different accents into the simulation configuration.
Text-level Simulation (Fast & Cheap): Strip the STT (Speech-to-Text) and TTS (Text-to-Speech) layers and test the core conversational logic via text APIs. This lets you run 10,000 simulations in minutes for pennies to catch logical loops, prompt injections, or goal failures.
End-to-End Audio Simulation (Realistic & Costly): Route synthetic audio through text-to-speech generators for the user simulator, hitting your actual telephony endpoints. Use platforms designed for this scale.
Explore dedicated platforms like Cekura AI or Roark which specialize in running thousands of persona-driven voice simulations, handling telephony, and tracking aggregate success metrics.
Utilize evaluation and scenario orchestration libraries like the LangWatch Scenario framework to code custom multi-turn audio-to-audio agent tests.
Whenever a live user call fails in production, convert that exact conversation path into a programmatic test case.
Add it to your nightly batch simulation suite so that bug is permanently locked out from regressing.
To help narrow this down, are you looking to test pure conversational logic/prompts (text-based) at high volume, or do you need full audio-to-audio pipeline testing including latency and telephony integration?
The best approach is LLM-driven user simulation + real audio + deterministic/LLM evaluation, rather than simply generating thousands of text transcripts.
A good architecture looks like this:
Scenario generator
↓
Synthetic caller personas
↓
TTS audio
↓
┌─────────────────┐
│ Voice Agent │
│ STT → LLM → TTS │
└─────────────────┘
↓
Transcript + audio + tool calls + timings
↓
Graders
↓
Pass/fail + metrics + failure clusters
Create perhaps 100–500 scenario templates, each specifying:
For example:
Goal: Reschedule a doctor's appointment Persona: Busy, impatient caller Facts: Appointment is Tuesday at 3 PM Behavior: Interrupts twice, initially gives the wrong date, asks an unrelated question Success: Appointment moved to an available slot and confirmation provided Then generate 10–100 variations of each scenario. Conversation simulation is increasingly being used specifically this way: scenario descriptions become simulated multi-turn conversations, allowing large-scale and repeatable testing.
This is probably the most important part.
Don't tell the simulator:
"Have a conversation that successfully completes the task." Instead give it an objective and behavioral characteristics:
Research on synthetic users suggests that naive simulators can be substantially more cooperative and predictable than real users, producing overly optimistic evaluations.
Don't run everything as text if the product is a voice agent.
For a representative subset—or ideally the entire suite—run:
simulated user → TTS → phone/WebRTC → agent → audio → simulated user
Vary:
This catches failures that a transcript-only simulator completely misses. Recent voice-agent evaluation work specifically emphasizes bot-to-bot audio conversations and validation of the simulated user's behavior.
Don't run one conversation per scenario.
If you have:
that's 30,000 conversations.
The repetitions matter because the agent itself is stochastic. You want:
"This scenario succeeds 97% of the time" rather than:
"This scenario worked."
I'd use several independent graders.
Hard/deterministic checks
LLM graders
Voice metrics
Then produce something like:
30,000 conversations
Task success 96.8%
Correct escalation 98.1%
Policy compliance 99.4%
Tool correctness 97.9%
Median latency 720 ms
P95 latency 1.8 s
Failures:
412 missed user intent
287 interruption failures
193 incorrect tool arguments
106 context loss
74 inappropriate escalation
This is where the system becomes genuinely powerful.
Production call
↓
Failure detected
↓
Anonymize + classify
↓
Convert into scenario
↓
Generate 20–100 variants
↓
Run against new agent
↓
Compare versions
That gives you a continuously growing test suite instead of repeatedly inventing hypothetical tests. OpenAI describes a related deployment-simulation approach in which realistic historical conversations are replayed against candidate systems to make pre-deployment evaluation more representative of actual traffic.
I'd structure a 30,000-test run approximately like:
| Layer | Example | Volume |
|---|---|---|
| Unit/tool tests | Correct API arguments | 10,000+ |
| Text simulations | Conversation logic | 10,000 |
| Audio simulations | STT/TTS/interruptions | 5,000 |
| Full real-call simulations | End-to-end | 5,000 |
That gives you scale without making every iteration expensive.
If you're building this yourself, I'd make the core test case a structured object:
{
"scenario": "reschedule_appointment",
"goal": "Move appointment to another available time",
"persona": {
"patience": 0.2,
"verbosity": 0.7,
"interruptions": 0.4,
"confusion": 0.3
},
"facts": {
"current_time": "Tuesday 3 PM"
},
"constraints": [
"Do not cancel without confirmation",
"Must verify identity"
],
"success_criteria": [
"identity_verified",
"new_slot_booked",
"confirmation_given"
]
}
Then your simulator can generate thousands of distinct callers from the same underlying scenario, while your evaluator checks the same objective ground truth.
If I were starting today, I'd prioritize this order:
The key insight is that "10,000 synthetic conversations" isn't inherently useful. Ten thousand nearly identical, cooperative LLM callers can give you a beautifully misleading 99.8% success rate. The value comes from distributional coverage, behavioral realism, audio realism, and reliable grading. Recent research specifically finds a gap between synthetic and real-user behavior, so I'd validate your simulator against a sample of real conversations before trusting its scores.
If you tell me what your stack is (e.g. Vapi, Retell, LiveKit, Twilio, ElevenLabs, custom WebRTC), I can sketch the exact architecture and a 10k-conversation test harness.
The best approach is not to simply generate thousands of random transcripts. For a voice agent, I’d build a simulation + evaluation harness that exercises the actual audio path and treats each call as a test case.
Scenario-driven simulation is now a common pattern for agent evaluation because it produces diverse multi-turn conversations while keeping the intended goal controlled. MLflow AI Platform Microsoft Learn 2. Use an LLM as the simulated caller
Give the caller a goal and persona, then let it decide what to say next based on what your agent actually says. Don't pre-script every turn. This exposes failures such as the agent misunderstanding a clarification or getting stuck in a loop.
For example:
Goal: Cancel an existing reservation.
Persona:
- impatient
- frequently interrupts
- initially gives the wrong reservation date
- becomes cooperative if the agent explains why information is needed
Success:
- reservation is cancelled
- agent confirms the cancellation
- no unauthorized cancellation occurs
For voice specifically, vary:
This matters because a text-only simulator won't catch problems in STT, turn detection, latency, TTS, or interruption handling. Recent voice-agent evaluation work similarly emphasizes bot-to-bot audio conversations rather than transcript-only simulation. arXiv 4. Run several layers of testing
I would use roughly this hierarchy:
| Test | Purpose |
|---|---|
| 100–500 scripted calls | Regression / deterministic behavior |
| 1,000–10,000 simulated calls | Scenario and edge-case coverage |
| Load test with concurrent calls | Capacity, latency, rate limits |
| Replay production conversations | Real-world regression |
| Human review of failures | Validate the evaluator |
Multi-turn load testing is particularly important: a backend call may only happen after several conversational turns, so single-message load tests can miss major bottlenecks. Microsoft Learn 5. Score every conversation automatically
Don't use one vague “good/bad” score. Capture things like:
Then retain the full audio + transcript + tool trace for failures. 6. Make failures become new tests
This is probably the most valuable part of the system:
production call → failure → scenario extracted → simulator → regression test
That gives you a continually growing test suite rather than regenerating thousands of unrelated conversations every time. Platforms such as MLflow explicitly support generating scenarios from existing conversations for this reason.
I'd actually build two separate systems.
Simulation testing asks:
“Can my agent handle 10,000 different kinds of callers?” Load testing asks:
“Can my infrastructure handle 2,000 simultaneous calls without latency exploding?” Don't conflate them. You can have excellent conversational coverage while still discovering that your telephony provider, STT service, tool backend, or model API falls over at concurrency.
A solid stack looks like:
Scenario generator
↓
Persona / user simulator
↓
Voice input generator
↓
┌───────────────┐
│ Actual agent │
│ STT → LLM → │
│ tools → TTS │
└───────────────┘
↓
Conversation recorder
↓
┌─────────────────────────┐
│ Deterministic checks │
│ LLM judge │
│ Latency measurements │
│ Audio/voice metrics │
└─────────────────────────┘
↓
Results DB
↓
Failures → new regression scenarios
For deterministic pieces of the pipeline, you don't necessarily need to invoke real models. For example, the OpenAI Agents SDK provides scripted STT/TTS/model components specifically for testing voice pipelines without making provider requests.
Don't trust an LLM judge blindly. Have a small, manually reviewed benchmark—say 100–300 conversations—with humans determining the correct outcome. Then measure how well your automated evaluator agrees with those humans. Research on voice-agent testing has found that both simulation quality and evaluation quality can vary substantially between testing approaches.
If you tell me what voice stack you're using (e.g. Twilio + OpenAI, Retell, Vapi, LiveKit, ElevenLabs, custom WebRTC), I can lay out a concrete architecture for running 10,000 simulated calls, including concurrency, audio generation, scoring, and cost control.
Simulating thousands of conversations to test a voice agent requires a structured pipeline that couples synthetic user personas with an automated evaluation (LLM-as-a-judge) framework . Running this at scale bypasses the limitations of manual QA and lets you stress-test edge cases, latency, interruptions, and prompt regressions.
The industry-standard architecture to achieve this involves a 4-step execution model:
Building a concurrency engine from scratch is resource-heavy. Instead, use specialized GenAI voice evaluation platforms or open-source testing wrappers:
Don't just test happy paths. Feed your agent’s system prompt, knowledge base, and tool definitions into a generator model to automatically output varied test scenarios. Configure your synthetic callers with different behavioral dimensions:
Simulating thousands of real-time audio streams via telephony or WebSockets simultaneously is expensive and slow. The most efficient approach at scale is a hybrid evaluation pipeline:
Every simulated conversation must be programmatically graded against strict, atomic assertions (e.g., "Did the agent trigger the refund tool?", "Did it avoid revealing restricted info?").
If you'd like to dive deeper, let me know:
I can help map out a specific testing script or architecture for your setup.
Simulating thousands of conversations to test a voice agent requires moving away from manual testing and implementing a bot-to-bot evaluation pipeline.
The industry-standard approach leverages a synthetic user simulator (an LLM with a voice/text persona) to talk to your voice agent under test , followed by an LLM-as-a-judge framework to score the output.
Step 1: Choose Your Architecture (Text-Level vs. Real Audio)
Before spinning up thousands of runs, decide how deep you need to test:
Step 2: Generate Diverse Personas and Scenarios
Don’t just write "happy path" tests. Real users are messy, impatient, and uncooperative. Use an LLM to auto-generate a synthetic dataset of thousands of variations based on your agent's system prompt and tool definitions:
Step 3: Select an Orchestration & Evaluation Tool
Building this infrastructure from scratch is a massive engineering undertaking. Most teams plug into specialized developer platforms that support massive parallel simulation runs and native integrations with orchestration layers like Vapi, Retell AI, LiveKit, or Pipecat:
Step 4: Define Automated Scorers (LLM-as-a-Judge)
Reviewing thousands of audio files manually is impossible. Automate 80% of evaluation using specific criteria:
If you'd like to dive deeper, let me know:
Yes. The best approach is not to make thousands of real phone calls. Use a layered simulation/evaluation system:
Create a library of scenarios such as:
Then have an LLM act as the simulated caller, with a goal like “Book an appointment, but refuse to give your address unless asked twice.”
This is dramatically cheaper and faster than testing through audio. Both Vapi's chat testing docs and LiveKit's testing docs recommend text-based simulations for high-volume behavioral testing.
This is important.
A weak test is:
User: "I'd like to cancel my appointment."
User: "Yes."
User: "Thanks."
A better simulator gets an objective:
Goal: Cancel an appointment scheduled for Tuesday.
Personality: Distracted, mildly impatient.
Hidden information: The appointment is under a different phone number.
Behavior: If the agent asks an unnecessary question, complain. If the agent successfully cancels it, end the call.
The simulator then improvises its turns. That's much closer to real conversations. Vapi's newer Simulations model and LiveKit's agent simulations both use this general approach.
For each conversation, collect structured metrics such as:
| Metric | Example |
|---|---|
| Task success | Appointment actually booked |
| Tool correctness | Correct API + arguments |
| Policy compliance | Didn't disclose restricted info |
| Factuality | Didn't invent availability |
| Conversation quality | Clear, coherent, relevant |
| Efficiency | Number of unnecessary turns |
| Handoff | Correctly transferred when required |
| Recovery | Recovered from misunderstanding |
| Audio quality | STT/TTS/interruptions worked |
Use deterministic assertions wherever possible—e.g. “book_appointment was called with date=X”—and an LLM judge for subjective criteria such as tone or conversational quality. Vapi's eval framework supports exact match, regex, tool-call validation, and AI judges; LiveKit similarly provides conversation-level judges.
A practical architecture looks like:
Scenario generator
│
10k user scenarios
│
┌─────────┴─────────┐
▼ ▼
Text simulation Text simulation
│ │
└─────────┬─────────┘
▼
Voice agent
│
transcript/events
│
┌──────────┴──────────┐
▼ ▼
deterministic LLM judges
checks │
└──────────┬──────────┘
▼
Eval database
│
failures / regressions
Run the cheap text tests continuously—e.g. every prompt/model/tool change—and reserve actual voice calls for a much smaller representative sample.
Text simulation won't catch:
So I'd use something like 90–99% simulated/text tests + 1–10% full voice tests, depending on how critical the audio behavior is.
Vapi's voice testing actually connects two AI agents over a real phone call and records/transcribes the interaction, while LiveKit explicitly distinguishes its inexpensive text simulations from full audio-pipeline testing.
If you have 100 core scenarios, generate 50–100 variations of each:
Scenario: customer wants to reschedule
Variations can alter:
That gets you to 5,000–10,000 conversations without having to manually author 10,000 scripts.
If you're building this yourself, I'd start with a scenario generator → goal-driven simulated user → your agent → structured event/transcript collector → deterministic checks + LLM judge → regression dashboard. If you're already using Vapi or LiveKit, their current simulation/eval tooling can eliminate a lot of that infrastructure.
If you tell me what voice stack you're using (Vapi, Retell, LiveKit, ElevenLabs, Twilio, custom, etc.), I can give you a concrete architecture for running 10,000 conversations in parallel, including how I'd generate the users and score the results.
The best approach is synthetic caller simulation + automated evaluation, with a smaller amount of human testing to validate that your simulator itself is realistic.
A good architecture looks like this:
Scenario generator → synthetic caller → real voice agent → audio/transcript/tool logs → evaluator → failure database
Start with your important intents and combine them with caller behaviors:
This matters because simulated conversations can otherwise be unrealistically cooperative; recent research finds that naive synthetic users often miss the friction present in real conversations.
Give the simulator:
Goal:
Reschedule an appointment from Tuesday to Thursday.
Persona:
Impatient customer. Gives short answers. Interrupts occasionally.
Hidden facts:
- Original appointment is at 2 PM Tuesday.
- Customer actually wants Thursday after 4 PM.
- They will become frustrated if asked for information they've
already provided.
Success:
Appointment is changed to an available Thursday slot.
Failure:
Wrong appointment changed, unnecessary escalation, or conversation
ends without completing the task.
Don't tell it exactly what sentences to say. Let it decide what to say based on the agent's responses. That's what gives you meaningful multi-turn testing rather than thousands of slightly randomized scripts. Microsoft's current conversation-simulation approach similarly uses scenario descriptions to generate simulated conversations and evaluate them.
For a voice agent, don't stop at text-to-text simulation.
Ideally:
Synthetic caller text → TTS → telephony/WebRTC → your agent → STT → LLM → TTS → synthetic caller
That exposes problems that transcript-only tests miss:
End-to-end bot-to-bot audio testing is also the approach used in recent voice-agent evaluation work such as EVA-Bench.
Don't ask simply, "Was this conversation good?"
Create explicit evaluators:
| Metric | Example |
|---|---|
| Task success | Was the appointment actually changed? |
| Tool correctness | Was the correct API/action invoked? |
| Factuality | Did the agent give correct information? |
| Policy/compliance | Were required disclosures made? |
| Conversation state | Did it remember information 5 turns later? |
| Turn-taking | Did it interrupt or respond too slowly? |
| Latency | Time to first response / total latency |
| Escalation | Did it transfer appropriately? |
| User experience | Did the caller have to repeat themselves? |
Then produce something like:
10,000 calls
────────────
Task success: 94.2%
Tool correctness: 97.8%
Required disclosure: 99.6%
Unnecessary transfer: 2.1%
Median latency: 820 ms
Top failures:
1. 37% — loses appointment date after user changes it
2. 24% — misunderstands numbers spoken quickly
3. 18% — fails after API timeout
4. 11% — talks over caller
5. 10% — other
The important part is that every failure becomes a reusable regression test.
I'd structure the workload roughly like:
That quickly gets you into the thousands.
Repeated runs are particularly important because the agent is stochastic: one successful run doesn't establish that a scenario reliably works.
A. Deterministic tests
Specific inputs → expected tool calls/state transitions.
B. Synthetic conversational tests
Thousands of realistic, variable multi-turn calls.
C. Human tests
A much smaller set where people deliberately try to break the agent and judge things automated evaluators can't reliably capture.
The synthetic layer gives you scale; humans give you reality checks. Current voice-testing guidance and research both point toward this combination rather than relying exclusively on synthetic callers.
If you're building this yourself, I'd use:
┌─────────────────┐
│ Scenario Store │
└────────┬────────┘
↓
┌─────────────────┐
│ Caller Simulator│
│ persona + goal │
└────────┬────────┘
↓
TTS / audio
↓
┌─────────────────┐
│ Voice Agent │
│ (real system) │
└────────┬────────┘
↓
transcript + audio
tool calls + timing
↓
┌─────────────────┐
│ Evaluators │
└────────┬────────┘
↓
┌─────────────────┐
│ Failure / Eval │
│ Database │
└─────────────────┘
↓
regression set
For a quick implementation, there are now platforms specifically built around this pattern, including Microsoft's conversation simulation tooling and commercial voice-testing systems such as Maxim and Future AGI.
The biggest mistake I'd avoid: generating 10,000 polite callers saying essentially the same thing. You'll get an impressive-looking pass rate that tells you very little. Make the simulator behaviorally difficult, ground its personas in real production conversations, and use real audio whenever voice-specific behavior matters.
If you tell me what your stack is (e.g. Vapi, Retell, LiveKit, Twilio, custom WebRTC, etc.), I can lay out a concrete architecture for running 10k+ calls in parallel, including the simulator, queueing, scoring, and regression pipeline.
Simulating thousands of conversations to test a voice agent requires moving away from manual testing or simple text-only scripts . Because voice involves real-time audio dynamics (latency, interruptions, accent variations, and background noise), testing at scale requires an end-to-end, bot-to-bot audio simulation architecture.
The industry-standard approach to executing this effectively involves a structured blueprint:
If you'd like to narrow this down, let me know:
The best approach is bot-to-bot, end-to-end audio simulation: have a simulated caller pursue a goal against your actual voice agent, over the same audio/telephony path real users use, then automatically score the resulting conversation.
That’s better than generating thousands of text transcripts because voice-specific failures—interruptions, latency, barge-in, STT mistakes, awkward turn-taking, silence, background noise—only appear when you exercise the audio loop. Recent voice-agent evaluation work is moving toward exactly this architecture.
┌──────────────────┐
│ Scenario Generator│
│ goals + personas │
└────────┬─────────┘
│
▼
┌──────────────────┐
│ Caller Simulator │
│ LLM + TTS │
└────────┬─────────┘
│ audio
▼
┌──────────────────┐
│ Your Voice Agent │
│ STT/LLM/tools/TTS│
└────────┬─────────┘
│
▼
┌──────────────────┐
│ Transcript/audio │
│ tool-call traces │
└────────┬─────────┘
▼
┌──────────────────┐
│ Evaluator │
│ rules + LLM judge│
└──────────────────┘
Create a few hundred scenario templates, each with:
Then generate variations from each scenario.
This gives you reproducibility: the same scenarios can be run against agent version A and B. Conversation-simulation frameworks such as MLflow and Microsoft Foundry use essentially this goal/persona/scenario model.
Don't use an LLM that simply generates the next utterance from the transcript.
Give the caller an explicit state machine:
goal = "change flight"
current_state = "has not provided booking number"
patience = 0.35
speaking_style = "brief"
will_cooperate = 0.8
rules:
- don't volunteer booking number unless asked
- if agent asks twice, become annoyed
- if offered an unacceptable flight, reject it
- after successful change, confirm and end call
The LLM decides how to express the next action; the scenario definition decides what the caller is trying to accomplish.
This is important because synthetic users can otherwise become unrealistically cooperative. Research on simulated users has found that they can miss the communication friction present in real conversations, producing overly optimistic evaluations.
For a voice agent, I'd strongly prefer:
LLM → TTS → your actual voice interface → STT/agent → TTS → simulator
rather than:
LLM → text → agent → text
You want to measure:
The open-source EVA framework is a useful reference architecture: its simulated caller and voice agent communicate through live audio, and it evaluates both task accuracy and conversational experience.
I'd separate deterministic metrics from LLM-judged metrics.
Deterministic:
LLM/audio judge:
Keep task success separate from conversation quality. A call can complete the task while still being a terrible voice interaction. EVA makes a similar distinction between accuracy and experience.
I'd allocate something like:
| Test population | Purpose |
|---|---|
| 20% | Normal/happy paths |
| 20% | Natural variations in wording |
| 20% | Difficult personas |
| 15% | Tool/API failures |
| 10% | Interruptions & barge-in |
| 5% | Noise/accent/audio degradation |
| 5% | Safety/policy/compliance |
| 5% | Completely unexpected behavior |
Then oversample failures. If a scenario fails, generate 100 variations around that failure rather than merely generating another 100 random calls.
This is probably the most important part.
Synthetic callers are excellent for coverage, but real callers are better for realism. Take anonymized production conversations and extract:
Turn those into scenario templates, then continuously feed production failures back into your simulation suite.
That's the basic production → scenario → simulation → regression test → production loop.
These are related but different.
Quality simulation:
"Can the agent successfully handle 5,000 different callers?"
Load test:
"Can my infrastructure handle 500 simultaneous calls without latency exploding?"
For the latter, use realistic stateful multi-turn calls and ramp concurrency rather than simply hammering an HTTP endpoint. Voice load-testing guidance specifically recommends tracking latency percentiles, call setup, queue depth, error rates, and task completion under load.
If you're building this yourself, the core system can be surprisingly small:
Scenario DB
↓
Scenario sampler
↓
Caller simulator × N
↓
WebRTC/SIP/WebSocket
↓
Real voice agent
↓
Recorder
↓
┌──────────────┬───────────────┐
│ Rule evaluator│ LLM/audio judge│
└──────────────┴───────────────┘
↓
Results DB
↓
Dashboard / regression report
Each run should produce a record like:
{
"scenario": "cancel_subscription",
"agent_version": "2026-08-14.3",
"seed": 48192,
"success": false,
"failure": "agent_claimed_cancellation_without_tool_call",
"turns": 11,
"duration_sec": 94,
"p95_latency_ms": 1420,
"tool_errors": 0,
"scores": {
"task_completion": 0,
"groundedness": 0.92,
"conversation_quality": 0.71,
"turn_taking": 0.63
}
}
Then make agent version + scenario seed deterministic, so every prompt/model/tool change can be compared against the same test population.
There are now several systems specifically designed around this workflow. EVA — open-source voice-agent evaluation framework is particularly interesting if you want to build your own infrastructure; Microsoft Foundry and MLflow also provide conversation simulation/evaluation workflows.
My recommendation: start with 500 carefully designed scenarios × 5–10 stochastic callers = 2,500–5,000 calls, run them against real audio, automatically validate the tool/state outcomes, and have an evaluator score the conversation. Once that's reliable, scale it to tens or hundreds of thousands.
The key isn't actually "thousands of conversations." It's making the simulated callers realistic enough that a 5,000-call test predicts what your real callers will experience.
Simulating thousands of conversations to test a voice agent requires moving away from manual testing and toward automated, scalable agent-to-agent (or LLM-to-agent) simulation . The modern approach treats your voice agent like an API, pairing it with a synthetic "user" powered by a Large Language Model.
Here is the best, most structured way to set this up:
Step-by-Step Implementation Framework
Industry Tools and Platforms
asyncio alongside OpenAI/Anthropic APIs for the user simulator and WebRTC/SIP python libraries for the voice connection.If you'd like to dive deeper, let me know:
I can tailor a specific architecture or script structure for your setup.