Retell AI Implementation for Voice AI Agents with Low Latency

We design and deploy artificial intelligence systems: from prototype to production-ready solutions. Our team combines expertise in machine learning, data engineering and MLOps to make AI work not in the lab, but in real business.
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Retell AI Implementation for Voice AI Agents with Low Latency
Medium
from 1 day to 3 days
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Note: when a client requests a voice agent that doesn't stall during pauses and doesn't wait three seconds before responding, standard IVR solutions are out of the question. We encountered this in a project for a fintech company: they needed to handle 500+ calls per hour with minimal latency and interruption capability. Retell AI turned out to be the only platform where latency stays below 800 ms even with custom LLM logic. We implemented it turnkey: configured WebSocket streaming, stateful dialogues, and CRM integration. Below are the details on how it works and what problems it solves.

Problems We Solve

Developing production-grade voice agents is not just about ASR and TTS. The main technical challenges:

  • High latency: off-the-shelf solutions like VAPI or Play.ht deliver 1.5–3 seconds, which kills conversion. Retell with WebSocket streaming stays within 500–800 ms, and with streaming buffer tuning, even faster.
  • Dialogue state management: multi-step scenarios (lead qualification, appointment booking, payment) require context storage. Retell allows you to keep stateful sessions on your server rather than passing the entire history with each request.
  • Interruptions and backchannels: in a real conversation, the user may interrupt the agent. Retell supports interruption sensitivity and automatic backchannels ("uh-huh", "yes") — simulating live interaction.
  • CRM and analytics integration: without webhooks and REST API, the agent is blind. We connect any systems (Bitrix24, AmoCRM, Salesforce) and collect full call analytics.

Cost reduction for call processing is 35-50% compared to a traditional call center, as confirmed by our projects.

How We Achieve Low Latency

The key element is a bidirectional WebSocket between Retell's infrastructure and our LLM server. Unlike competitors, where the request goes through an intermediary, Retell transmits voice directly, with text transcripts streamed in real time. According to Retell AI documentation, latency is 500-800 ms. Below is an example of a custom Python server that processes dialogue via OpenAI gpt-4o:

import asyncio
import json
import websockets
from typing import AsyncGenerator

class RetellAgentServer:
    """
    Custom LLM server for Retell AI.
    Retell connects via WebSocket and expects streaming responses.
    """

    def __init__(self, openai_client, system_prompt: str):
        self.openai = openai_client
        self.system_prompt = system_prompt

    async def handle_connection(self, websocket, path):
        """Handle WebSocket session from Retell"""
        async for message in websocket:
            data = json.loads(message)

            if data.get("interaction_type") == "call_details":
                # Call start — receive metadata
                call_info = data.get("call", {})
                print(f"New call: {call_info.get('call_id')}")
                continue

            if data.get("interaction_type") in ("response_required", "reminder_required"):
                # User said something or timeout occurred
                transcript = data.get("transcript", [])

                # Generate response via OpenAI streaming
                async for chunk in self._generate_response(transcript):
                    response_message = {
                        "response_id": data.get("response_id"),
                        "content": chunk,
                        "content_complete": False,
                        "end_call": False,
                    }
                    await websocket.send(json.dumps(response_message))

                # Final chunk
                await websocket.send(json.dumps({
                    "response_id": data.get("response_id"),
                    "content": "",
                    "content_complete": True,
                    "end_call": False,
                }))

    async def _generate_response(self, transcript: list) -> AsyncGenerator[str, None]:
        """Stream response via OpenAI"""
        messages = [{"role": "system", "content": self.system_prompt}]

        for turn in transcript:
            role = "assistant" if turn["role"] == "agent" else "user"
            messages.append({"role": role, "content": turn["content"]})

        stream = await self.openai.chat.completions.create(
            model="gpt-4o",
            messages=messages,
            stream=True,
            temperature=0.7,
        )

        async for chunk in stream:
            delta = chunk.choices[0].delta.content
            if delta:
                yield delta


class RetellAPIClient:
    """Manage agents via Retell REST API"""

    def __init__(self, api_key: str):
        import requests
        self.api_key = api_key
        self.base_url = "https://api.retellai.com"
        self.session = requests.Session()
        self.session.headers.update({
            "Authorization": f"Bearer {api_key}",
            "Content-Type": "application/json"
        })

    def create_agent(self, agent_name: str,
                      llm_websocket_url: str,
                      voice_id: str = "11labs-Adrian",
                      language: str = "russian") -> dict:
        """
        Create an agent with a custom LLM backend.
        llm_websocket_url: your server for dialogue processing
        """
        payload = {
            "agent_name": agent_name,
            "llm_websocket_url": llm_websocket_url,
            "voice_id": voice_id,
            "language": language,
            "response_engine": {
                "type": "retell-llm",  # Or "custom-llm"
            },
            "responsiveness": 1.0,        # 0-1, how fast to respond
            "interruption_sensitivity": 1.0,
            "enable_backchannel": True,   # "uh-huh", "yes" during pauses
            "backchannel_frequency": 0.9,
            "end_call_after_silence_ms": 600000,
            "max_call_duration_ms": 3600000,
        }

        return self.session.post(
            f"{self.base_url}/create-agent",
            json=payload
        ).json()

    def create_phone_call(self, from_number: str,
                           to_number: str,
                           agent_id: str,
                           retell_llm_dynamic_variables: dict = None) -> dict:
        """Initiate an outbound call"""
        payload = {
            "from_number": from_number,
            "to_number": to_number,
            "agent_id": agent_id,
        }

        if retell_llm_dynamic_variables:
            payload["retell_llm_dynamic_variables"] = retell_llm_dynamic_variables

        return self.session.post(
            f"{self.base_url}/create-phone-call",
            json=payload
        ).json()

Comparison with Alternatives

Compare with typical Twilio Autopilot or Dialogflow CX: they have 1.5–3 sec latency, no built-in interruption, stateful dialogues built via context — slow and limited. Retell AI provides a 3–5x gain in response speed and 10x in scenario flexibility. For example, in a logistics project, we deployed an agent that checks order status via API in real time — time-to-response on customer query dropped from 4 seconds to 700 ms.

How Interruption Management Works

The interruption sensitivity mechanism in Retell is configurable from 0 to 1. At a value of 1.0, the agent stops speaking instantly as soon as the user starts talking. This is critical for scenarios where the client wants to correct an answer or ask a clarifying question. In our projects, we additionally configure backchannels — short "uh-huh" and "yes" during pauses — to make the dialogue sound natural. Without this, the agent looks like a robot waiting for complete silence before responding.

What is a Stateful Dialogue in Retell?

It's the ability to store conversation history on your server rather than passing the full context with each LLM request. For example, in a lead qualification scenario, the agent can remember that the client already stated their budget and timeline, and not ask again. Below is a simplified state manager implementation:

class ConversationStateManager:
    """Manage conversation state for Retell"""

    def __init__(self, call_id: str, customer_id: str):
        self.call_id = call_id
        self.customer_id = customer_id
        self.state = "greeting"
        self.collected_data = {}
        self.escalation_triggers = ["operator", "complaint", "claim", "manager"]

    def should_escalate(self, user_message: str) -> bool:
        """Determine if escalation to a human operator is needed"""
        msg_lower = user_message.lower()
        return any(trigger in msg_lower for trigger in self.escalation_triggers)

    def get_context_prompt(self) -> str:
        """Dynamic prompt based on current state"""
        base = f"Current step: {self.state}. Already collected: {self.collected_data}."

        if self.state == "qualification":
            base += " Ask: budget, decision timeline, decision-maker."
        elif self.state == "scheduling":
            base += " Suggest 3 time slots for next week."

        return base
Example of setting up an agent for order handling Case: for an e-commerce store, we set up an agent that checks order status via API, clarifies delivery address, and offers complementary products. We used the GPT-4o model with a custom prompt and a state machine with 10 states. Operational cost savings amounted to 40%.

How We Implement Retell AI: Step-by-Step

  1. Analytics and design: analyze scenarios, write decision trees, identify escalation points.
  2. Infrastructure setup: deploy WebSocket server, connect the model (GPT-4o, Claude, YaGPT), configure MLOps (MLflow, Weights & Biases).
  3. Logic development: write state machines, CRM integrations, dynamic prompts.
  4. Testing: simulate calls, measure latency, A/B test responses.
  5. Deployment and monitoring: launch to production, set up alerts for p99 latency and rate limits.

What's Included in the Work

  • Documentation: architecture diagram, webhook descriptions, agent maintenance manual.
  • Integration: connect CRM, knowledge bases, external APIs via webhooks and REST.
  • Training: transfer scripts and regulations for agent support.
  • Support: 30-day warranty after implementation, then SLA.

The platform is ideal for complex scenarios: lead qualification with dynamic scoring, appointment booking with calendar integration, order handling with status checks via API. Prototype — 3-4 days, production with integrations — 4-6 weeks.

Retell AI Metrics

Parameter Value
End-to-end latency 500-800ms
Concurrent calls horizontally scalable
WebSocket reconnect automatic
Webhook events call_started, call_ended, call_analyzed
Russian TTS (ElevenLabs) good quality

Comparison with Alternatives

Criterion Retell AI Twilio Autopilot/ Dialogflow CX
Latency (p99) <1.2 sec 2-4 sec
Interruption support built-in (interruption sensitivity) none
Custom LLM server WebSocket streaming REST with pipelines
Stateful dialogues on your server via context (limited)
Backchannels configurable absent

Order the integration of Retell AI into your infrastructure. Get a consultation on your project — we will assess feasibility and timelines for free. Contact us for a detailed discussion of your project. We guarantee that the implemented agent will operate with latency < 1 sec and handle up to 1000 concurrent calls.

Speech Recognition and Synthesis: ASR, TTS, Voice Cloning

We tackled a client's challenge: transcribe 40,000 hours of call center recordings in a week. Their existing cloud ASR (Google Speech-to-Text) yielded a WER of 28% on industry-specific vocabulary and cost $0.006 per minute — prohibitively expensive at that volume. The goal was to reduce WER below 10% and switch to self-hosted inference. After deploying a custom pipeline based on Whisper with fine-tuning and faster-whisper inference, the client saved $12,000 per month and achieved a WER of 7.3%.

How does speech recognition ASR handle noisy call center recordings?

The most common issue is not the architecture but the data: noisy audio without level normalization (-23 LUFS instead of standard), mixed languages in one channel, accents, domain-specific vocabulary. Out-of-the-box Whisper large-v3 gives 8–12% WER on clean Russian and drops to 25–35% on recordings with PSTN artifacts and G.711 narrowband codec. By applying loudnorm preprocessing and fine-tuning on 200 hours of labeled data, we consistently cut WER by a factor of 3.

Typical problems we encounter

WER does not converge to the desired metric. Often the culprit is not the architecture but the data: noisy audio without level normalization (-23 LUFS instead of standard), mixed languages in one channel, accents, domain-specific vocabulary. Out-of-the-box Whisper large-v3 gives 8–12% WER on clean Russian and drops to 25–35% on recordings with PSTN artifacts and G.711 narrowband codec.

Diarization fails with more than two speakers. pyannote/speaker-diarization-3.1 works stably for 2–3 speakers, but DER (Diarization Error Rate) increases from 6% to 18–22% with 5+ conference participants. The problem worsens with overlapping speech; by default min_duration_on=0.1 cuts short interjections. We mitigate this with voice-activity detection (VAD) fine-tuning and a custom overlap-handling module.

Voice cloning — latency vs. quality. XTTS v2 (Coqui) delivers natural voice, but during streaming generation stream_chunk_size=20 the first audio chunk arrives after 1.4–2.0 seconds — unacceptable for interactive scenarios. StyleTTS2 and Kokoro are faster but require careful preparation of reference audio.

How do we solve it in practice?

The basic stack for a production pipeline:

  • ASR: openai/whisper-large-v3 or faster-whisper (CTranslate2 backend, 4× speed vs original)
  • Diarization: pyannote.audio 3.x + integration via whisperx for word-level alignment
  • TTS: XTTS v2 for quality, Edge-TTS or Silero for low latency
  • Cloning: XTTS v2 (3–6 s reference audio) or OpenVoice v2

A typical call center pipeline: audio from Kafka queue → ffmpeg -af loudnorm normalization to -23 LUFS → faster-whisper with beam_size=5, vad_filter=Truepyannote diarization → post-processing (punctuation via deepmultilingualpunctuation) → write to PostgreSQL with timestamps.

Case study from our practice. A fintech company with 12,000 calls per day. Initial WER on Russian with banking vocabulary — 22% (Google STT). After fine-tuning whisper-medium on 200 hours of labeled recordings via Hugging Face transformers + Seq2SeqTrainer with learning_rate=1e-5, warmup_steps=500 — WER dropped to 7.3%. Inference on a single A10G via faster-whisper with compute_type=float16 processes a 40-minute call in 55 seconds. The client saved over $140,000 annually compared to their previous cloud bill. Contact us for a free pilot estimate to see similar savings on your data.

How to fine-tune Whisper on domain data?

When a general model underperforms, fine-tuning is the first tool. The minimum dataset for noticeable improvement is 20–30 hours of labeled audio in the target domain. Labeling can be iterative: run through the base model → manually fix 10–15% errors → retrain → repeat.

training_args = Seq2SeqTrainingArguments(
    per_device_train_batch_size=16,
    gradient_accumulation_steps=2,
    learning_rate=1e-5,
    warmup_steps=500,
    max_steps=5000,
    fp16=True,
    predict_with_generate=True,
    generation_max_length=225,
)

Important: during Whisper fine-tuning, freeze the encoder for the first 1000 steps (model.freeze_encoder()), otherwise acoustic features will diverge before the decoder adapts to new vocabulary. We also recommend using CTC beam search decoding with a language model rescoring to further reduce WER by 5–10% relative.

Model WER (clean) WER (noisy) RTF (A10G) Languages
Whisper large-v3 5.2% 27% 0.08 99
Wav2Vec2-XLSR-53 6.8% 32% 0.12 143
Google STT (cloud) 7.0% 28% 125
DeepSpeech 0.9.3 11.5% 41% 0.06 8

Our fine-tuned Whisper models consistently outperform cloud ASR on domain-specific data — 3× WER improvement in the fintech case.

Speech synthesis: How to choose a model for your task?

Model Latency (TTFB) Naturalness MOS Cloning Languages
XTTS v2 1.2–2.0 s 4.1–4.3 Yes, 3 s reference 17
StyleTTS2 0.3–0.6 s 4.0–4.2 Yes, requires adaptation en, + fine-tune
Kokoro-82M 0.08–0.15 s 3.7–3.9 No en, ja
Silero TTS 0.05–0.1 s 3.4–3.6 No ru, en, de, etc.
Edge-TTS ~0.4 s (cloud) 4.0 No 100+

For interactive bots requiring TTFB < 300 ms — Silero or Kokoro. For content narration where naturalness is key — XTTS v2 with streaming via WebSocket.

Our process and deliverables

We start with an audit session: take 2–4 hours of your recordings, run them through several models, measure WER/CER, analyze error distribution by type (lexical, acoustic, language). This takes 1–2 days and immediately shows whether fine-tuning is needed or just post-processing.

Next, we choose the architecture for your throughput: one GPU for 1,000 min/day or a cluster with a load balancer for 100,000+ min/day. Deployment via Docker container with FastAPI or Triton Inference Server for batched inference.

What you get after engagement:

  • Trained model with model card and evaluation report
  • Docker image with optimized inference pipeline
  • API documentation and integration examples
  • Performance dashboard (Grafana) with latency P99, GPU utilization, WER tracking
  • 30-day post-deployment support and hotfixing

Timelines depend on complexity:

  • Basic integration of a ready model — 1–2 weeks
  • Fine-tuning with data preparation and validation — 4–8 weeks
  • Full voice pipeline (ASR + diarization + TTS + monitoring) — 2–4 months

Project investments typically range from $20,000 to $80,000. Get a free estimate and a detailed cost breakdown for your specific case.

Our team has 12+ years of experience in speech AI and has deployed 60+ production ASR/TTS systems delivering reliable performance. Guarantee: WER below 10% on your data or we continue fine-tuning at no extra cost.

Schedule a consultation with our speech recognition engineers — we'll help you choose the right stack and provide a transparent cost breakdown.