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For teams evaluating STT providers, Telnyx offers a unique advantage: speech-to-text built directly into carrier-grade telephony infrastructure. Unlike providers that bolt on transcription as an afterthought, Telnyx colocates AI compute with global voice PoPs, delivering low latency that makes conversational AI feel natural while eliminating the complexity of coordinating multiple vendors.

Core evaluation criteria for production STT

When evaluating speech-to-text APIs, focus on metrics that directly impact your use case. For contact centers and voice AI applications, these five criteria determine success:

Accuracy and word error rate

Accuracy remains the foundation of any STT deployment. Look for providers that publish Word Error Rate (WER) benchmarks across different audio conditions: clean speech, background noise, accented speakers, and domain-specific vocabulary. Telnyx's in-house engine delivers consistent accuracy across diverse audio conditions, with HD STT testing guides that help validate performance with your actual audio codecs and network conditions before committing.

Latency and streaming capabilities

Real-time applications demand sub-second response times. The best STT APIs offer WebSocket streaming for continuous transcription, with initial response times under 300ms. Telnyx achieves industry-leading latency by colocating compute with telephony infrastructure, placing GPUs adjacent to voice network PoPs minimizes the physical distance data travels, solving the latency problem that physics would otherwise impose.

Feature comparison across providers

Not all STT APIs offer the same capabilities. Telnyx is the only provider delivering all five essential features through both WebSocket and SIP streaming:

Note: Telnyx's dual WebSocket/SIP infrastructure enables direct telephony integration, a key differentiator for production voice applications.

Integration complexity and SDKs

Production deployments require more than raw transcription. Evaluate how STT integrates with your existing stack, whether that's SIP trunks, WebRTC applications, or contact center platforms. Native integration with call control APIs eliminates the complexity of coordinating multiple vendors for telephony and transcription.

Pricing transparency and scalability

Volume-based pricing with clear tiers helps predict costs as you scale. Watch for hidden fees around features like diarization, timestamps, or custom models that can inflate per-minute rates.

The healthcare industry alone is estimated to reach $493.3 million in STT spending by 2025, according to Fortune Business Insights. Volume-based pricing with clear tiers helps predict costs as you scale. Telnyx provides transparent, per-minute pricing without hidden fees for features like diarization or timestamps, unlike competitors that inflate costs with add-on charges. With volume discounts built in, Telnyx scales economically as your usage grows.

Integrating STT into your voice stack

Moving from evaluation to implementation requires understanding your integration path. Modern STT APIs support multiple approaches depending on your architecture.

Real-time streaming for live calls

For contact centers processing live calls, WebSocket streaming provides the lowest latency. Here's a basic Node.js implementation using Telnyx's Voice API with real-time transcription:

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const WebSocket = require('ws');
const ws = new WebSocket('wss://api.telnyx.com/v2/transcriptions');

ws.on('open', () => {
  ws.send(JSON.stringify({
    event: 'start',
    streamSid: callSid,
    start: {
      tracks: ['inbound', 'outbound'],
      mediaFormat: 'audio/x-mulaw'
    }
  }));
});

ws.on('message', (data) => {
  const transcript = JSON.parse(data);
  if (transcript.event === 'transcript') {
    console.log(`Speaker ${transcript.speaker}: ${transcript.text}`);
    // Feed to your LLM or routing logic
  }
});

This approach captures both sides of the conversation with speaker labels, essential for compliance recording and quality monitoring.

Batch processing with REST APIs

For recorded calls or audio files, REST endpoints offer simpler integration. The OpenAI-compatible transcription endpoint works with existing SDKs:

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import requests

response = requests.post(
    'https://api.telnyx.com/v2/ai/transcribe',
    headers={'Authorization': f'Bearer {api_key}'},
    files={'file': open('call_recording.mp3', 'rb')},
    data={
        'model': 'whisper-large',
        'response_format': 'json',
        'timestamp_granularities': ['word']
    }
)

transcript = response.json()

Connecting STT to telephony infrastructure

The gap between STT and telephony often creates integration headaches. When your STT provider also handles SIP trunking and number provisioning, you eliminate network hops and vendor coordination. Media streams flow directly from PSTN to transcription without third-party handoffs.

Security and compliance considerations

With North America controlling 32.27% of the global STT market, regulatory compliance has become non-negotiable for enterprise deployments. Telnyx meets all enterprise compliance requirements, providing the security foundation teams need.

Data residency and processing location

GDPR and data sovereignty requirements demand control over where audio and transcripts are processed. Telnyx offers complete data residency control with regional processing options, keeping European calls within EU data centers, for instance. This becomes especially critical as markets like China grow at 11.7% CAGR, each with distinct regulatory frameworks.

Encryption and retention policies

Audio recordings and transcripts require encryption both in transit and at rest. Telnyx provides end-to-end encryption with configurable retention policies that balance compliance requirements against storage costs. Our SOC 2 Type II and HIPAA compliance certifications provide third-party validation of these security controls, giving enterprises the assurance they need.

PII detection and redaction

Automatic detection and redaction of personally identifiable information protects customer data. Telnyx automatically masks credit card numbers, social security numbers, and other sensitive information in transcripts while maintaining readability for agent review, built into the platform with no additional configuration required.

Building production-ready voice AI

As the global STT market approaches $21 billion by 2034, the focus has shifted from basic transcription to enabling intelligent voice experiences. Modern STT APIs must feed conversational AI systems that understand context, maintain dialogue state, and respond naturally.

Telnyx's unified platform advantage becomes clear here: STT, TTS, LLM inference, and voice infrastructure on the same network. When these components share the same infrastructure, end-to-end latency stays under the 300ms threshold where conversations feel natural. No other provider offers this level of integration.

Teams deploying STT should also consider the broader ecosystem, how transcription connects to analytics platforms, CRM systems, and quality monitoring tools. Telnyx's APIs provide webhooks, event streams, and flexible output formats that simplify these integrations.

Start transcribing with production-grade infrastructure

Selecting the right speech-to-text API requires balancing accuracy, latency, features, and cost against your specific use case. For teams building real-time voice applications, the integration with telephony infrastructure often matters more than raw transcription performance.

Telnyx combines carrier-grade telephony with colocated AI infrastructure to deliver sub-200ms transcription latency. With STT, TTS, and call control on the same private network, you eliminate the complexity of coordinating multiple vendors. Our in-house speech-to-text engine offers transparent per-minute pricing with volume discounts, while SOC 2 and HIPAA compliance meet enterprise security requirements.

Get started with Telnyx's Speech-to-Text API in minutes, test with your own audio files and experience the difference unified infrastructure makes.