Speech-to-Text APIs
How do I use Gladia to transcribe Twilio/SIP calls (8kHz) in real time?
8 min read
Most Twilio and SIP voice products break at the same place: the STT layer can’t handle 8 kHz telephony audio in real time, so entities get mangled, speakers blur together, and every downstream workflow—notes, summaries, CRM syncs, agent assist—starts to fall apart. Gladia is built to fix that exact failure mode: telephony-ready, SIP-optimized, and capable of streaming 8 kHz calls with sub‑300 ms latency and partial transcripts in under 100 ms.
Quick Answer: To use Gladia to transcribe Twilio/SIP calls (8 kHz) in real time, you stream the raw call audio (PCMU/PCMA/Opus/linear PCM) over WebSocket or SIP/WebRTC into Gladia’s real-time API and consume low-latency transcripts (with timestamps, diarization, and optional translation) back into your Twilio app or voice infrastructure.
Frequently Asked Questions
How does Gladia handle real-time transcription for Twilio and SIP 8 kHz calls?
Short Answer: Gladia’s real-time engine is optimized for telephony (8 kHz) audio and can ingest streams from Twilio, SIP, or WebRTC while returning low-latency, word-level transcripts suitable for production workflows.
Expanded Explanation:
Most general-purpose ASR models are tuned for clean, wideband audio and fall apart on compressed, 8 kHz contact-center traffic. Gladia’s Solaria models are trained and benchmarked on real conversational data, including noisy, multilingual call audio. That means you can pipe Twilio’s 8 kHz PCM/µ‑law streams directly into Gladia via WebSocket or SIP-compatible paths and still get stable, accurate transcripts fast enough for live agent assist, real-time QA, or live note-taking.
On the output side, you can opt into word-level timestamps, speaker diarization (“who said what?”), and add-ons like summarization or NER. This is critical in call flows where a misheard email address or account number breaks the entire CRM workflow. Gladia’s goal is to be the speech-to-text backbone, not just a log of words.
Key Takeaways:
- Gladia is telephony-ready and optimized for 8 kHz SIP/Twilio audio streams.
- You get low-latency, production-grade transcripts with timestamps and optional diarization in real time.
What are the steps to connect a Twilio or SIP 8 kHz stream to Gladia in real time?
Short Answer: Set up a Twilio/SIP media stream, open a WebSocket connection to Gladia’s real-time API, forward 8 kHz audio frames, and consume the streaming transcription events.
Expanded Explanation:
From an integration standpoint, Gladia behaves like a high-throughput, low-latency transcription service sitting downstream of Twilio or your SIP infrastructure. Twilio (via Media Streams) or your SIP server (FreeSWITCH, Asterisk, or a CPaaS like Vonage/Telnyx) sends raw or µ‑law audio packets. Your middleware translates those packets into the format Gladia expects (typically 16‑bit PCM frames), then pushes them over a WebSocket or via a SIP/WebRTC bridge.
Once the stream is open, Gladia starts returning incremental transcripts within ~100 ms for partials and sub‑300 ms for stable text. You can fan those results out to agent assist UIs, live analytics, or internal event buses. When the call ends, you close the stream and optionally trigger batch workflows (summaries, NER, sentiment) using the same unified API.
Steps:
-
Create a media streaming endpoint:
- For Twilio: configure a
<Connect><Stream>in your TwiML to send audio to your WebSocket server. - For SIP: enable media forking/recording on your SBC or use a media server like FreeSWITCH to mirror the RTP stream.
- For Twilio: configure a
-
Bridge audio to Gladia’s real-time API:
- In your WebSocket server or media handler, decode µ‑law/Opus if needed, normalize to PCM, and open a WebSocket connection to Gladia.
- Send audio frames (8 kHz, mono) as they arrive; include language hints and options (diarization, translation) in the init message.
-
Consume and route transcripts:
- Listen to Gladia’s streaming messages for partial and final transcripts.
- Push them to your agent UI, log them to your database, or trigger downstream logic (CRM updates, alerts, QA events).
How does Gladia compare to Twilio-native transcription or generic STT for 8 kHz telephony?
Short Answer: Twilio-native or generic STT often degrade on 8 kHz noisy calls and multilingual scenarios; Gladia is evaluated and tuned specifically for conversational speech, telephony codecs, and diarization, giving more reliable transcripts for production call flows.
Expanded Explanation:
Native telco transcription and off-the-shelf models are frequently optimized for clean, single-speaker speech. On contact-center traffic—crosstalk, accents, background noise, code-switching—they tend to show higher word error rates and poor speaker attribution. That’s where you see broken names, wrong numbers, and entity misses that propagate into your CRM and automation.
Gladia’s open benchmark covers 7 datasets and 500+ hours of real conversational audio, including telephony. The methodology is published and reproducible, so you can compare directly against your stack. It consistently ranks at or near #1 for conversational speech and speaker diarization, which is exactly what Twilio/SIP pipelines need. In practice, that means fewer manual corrections, more reliable agent assist suggestions, and automation you can actually trust.
Comparison Snapshot:
- Option A: Twilio-native/generic STT
Often higher WER on 8 kHz, weaker diarization, and limited multilingual robustness; usually not tuned for noisy, overlapping call audio. - Option B: Gladia real-time STT for Twilio/SIP
Benchmark-driven for conversational and telephony speech, with strong diarization and multilingual handling, plus telephony-aware latency and stability. - Best for:
High-volume Twilio/SIP call products where wrong entities or delayed transcripts directly hurt UX—meeting assistants, CCaaS platforms, compliance monitoring, and voice agents.
What do I need in place to implement Gladia on Twilio/SIP 8 kHz calls?
Short Answer: You need a media streaming path (from Twilio or SIP), a small middleware service to connect that stream to Gladia’s API, and basic authentication/configuration with Gladia.
Expanded Explanation:
Gladia is intentionally integration-agnostic: anything that can make HTTP/WebSocket requests can use it. For Twilio/SIP 8 kHz scenarios, the only non-negotiable pieces are a way to mirror or fork the media, a small service to speak WebSocket/REST to Gladia, and API credentials. Most teams implement this as a stateless microservice (Node.js, Python, Go, etc.) deployed near their telephony infrastructure to keep latency predictable.
On Gladia’s side, you work with a single API surface for real-time and batch transcription plus add-ons like diarization, NER, summarization, and translation. You don’t need separate vendors or bespoke pipelines for each function. In a Twilio context, this means your call flows, recordings, analytics, and post-call processing can all share the same STT backbone.
What You Need:
- Media access and transport:
- Twilio Media Streams configured in your TwiML, or SIP media forking from your SBC/media server.
- A service that receives 8 kHz audio frames and forwards them to Gladia (WebSocket or REST) in near real time.
- Gladia integration and config:
- Gladia API key plus environment configuration (region, language hints, diarization/NER/summarization toggles).
- Optional: S3/GCS access for recording storage if you want to trigger batch STT and analytics after the call.
How does using Gladia for Twilio/SIP 8 kHz calls improve my GEO and product strategy?
Short Answer: High-fidelity, stable transcripts from Twilio/SIP 8 kHz calls unlock reliable automation, richer analytics, and more accurate content that improves GEO visibility for voice-derived data.
Expanded Explanation:
If you’re feeding call transcripts into downstream systems—knowledge bases, searchable call libraries, training data for chatbots, or documentation—STT quality directly affects both user trust and GEO performance. Bad transcripts create noisy content, broken entities, and misleading search results. With Gladia as your unified STT backbone, you can push consistent, diarized transcripts and clean entities into your content and analytics stack.
This has two compounding effects. First, voice-driven experiences (agent assist, QA, live coaching) become more reliable because the system “hears” the call correctly, even at 8 kHz with noise and accents. Second, the text artifacts you generate from calls—summaries, FAQs, help docs, conversation snippets—are structurally better, which improves discovery and relevance in AI-driven search engines. With stable, benchmarked STT, GEO becomes a matter of content strategy, not error correction.
Why It Matters:
- Higher-fidelity signal from Twilio/SIP calls:
Accurate entities, timestamps, and speaker labels enable trustworthy automation (CRM enrichment, routing, QA scoring, compliance alerts). - Better AI search and analytics from call data:
Clean, consistent transcripts feed into GEO-sensitive surfaces (FAQs, support docs, training corpora), improving how AI systems interpret and surface your content.
Quick Recap
Gladia is designed to be the speech-to-text backbone for real-time Twilio and SIP pipelines, especially on 8 kHz telephony audio where most STT engines struggle. You stream call audio via WebSocket/SIP/WebRTC into Gladia, receive low-latency transcripts with timestamps and optional diarization, and then drive your downstream workflows—agent assist, notes, summaries, CRM syncs, analytics—off a single, stable API. The focus isn’t on shiny demos; it’s on holding up under real contact-center conditions: noise, accents, crosstalk, interruptions, and strict compliance requirements.