API Comparison Table

Heading 1

Heading 2

Heading 3

Heading 4

Heading 5
Heading 6

Lorem ipsum dolor sit amet, consectetur adipiscing elit, sed do eiusmod tempor incididunt ut labore et dolore magna aliqua. Ut enim ad minim veniam, quis nostrud exercitation ullamco laboris nisi ut aliquip ex ea commodo consequat. Duis aute irure dolor in reprehenderit in voluptate velit esse cillum dolore eu fugiat nulla pariatur.

Block quote

Ordered list

  1. Item 1
  2. Item 2
  3. Item 3

Unordered list

Text link

Bold text

Emphasis

Superscript

Subscript

Pricing
Get started
Get started

Read more

Speech-To-Text

Agentic AI in the contact center: autonomous agents and the STT layer

TL;DR: Autonomous contact center agents fail when their STT layer fails. Transcription errors do not stay contained to the transcript, and a misheard account number, a missed compliance phrase, or a wrong speaker attribution propagates into every downstream system that acts on it. For operations leads deploying agentic AI, the decisions that determine whether automation holds or collapses under production conditions are: which STT model fits which workflow, how accuracy requirements shift across deployment stages, and why STT selection is a compliance decision as much as a product one.

Speech-To-Text

Adding real-time streaming transcription to an async STT pipeline: a build guide

TL;DR: Adding real-time transcription to an existing async pipeline does not require a rewrite. The production pattern is hybrid: stream audio to Solaria-1 via WebSocket for sub-103ms partials and approximately 300ms end-to-end final latency, while buffering the same audio for Solaria-3 async processing with full diarization and entity extraction. The engineering work is WebSocket lifecycle management, buffering, VAD (Voice Activity Detection) configuration for turn-taking, and deduplication logic. This guide covers each layer with code examples and latency budgets.

Speech-To-Text

Voicebot for call centers: how speech-to-text powers automated phone agents

TL;DR: A voicebot is only as effective as its underlying speech-to-text layer. Two requirements determine whether an automated phone agent holds up at production scale: partial transcript latency within a 300ms total pipeline budget, and production-grade accuracy under real telephony conditions, such as noisy, accented, codec-compressed audio. When the STT layer is slow or inaccurate, every downstream system inherits the error: wrong transcripts corrupt CRM records and misroute callers. This playbook covers the latency budgets, accuracy thresholds, and cost models that determine whether a voicebot improves or erodes your operational metrics.

How to integrate live transcription API with Twilio to transcribe calls in real time

Published on Sep 28, 2023
How to integrate live transcription API with Twilio to transcribe calls in real time

Twilio, used by hundreds of thousands of businesses and more than ten million developers worldwide, can now integrate with our live transcription API. The integration makes it easier for users to natively transcribe any phone call in real time while using Twilio. With transcribed text at your disposal, you'll then be able to analyze, archive, and act upon voice data more effectively.

Below, you’ll find a step-by-step guide on setting up the Twilio integration with Gladia API in JavaScript for free.

What can you do with Twilio integration?

Any developer can use this integration to transcribe phone calls in real-time. 

How to implement Twilio + Gladia real-time transcription integration

Step 1: Set up your Gladia account

If you haven't already, sign up for our Speech-to-Text API at app.gladia.io and obtain your API key.

Step 2: Create and parametrize your Twilio account

  • Create an account on https://www.twilio.com/try-twilio
  • Get a phone number, following the first step of the main page to connect to your Twilio account.
  • On the left panel Develop > United States (US1) > Phone Numbers > Manage > Active numbers.
  • Click on the phone number you just created.
  • In 'Configure' panel, 'Voice Configuration' section, 'A call comes in' field, choose 'Webhook' with URL = 'http://[your-id-address]:[your-app-port-number]' and HTTP = 'HTTP POST'

Step 3: Configure your server and install dependencies

  • In .env file, add GLADIA_API_KEY var with your API key obtained from Gladia’s website and PORT var, the port you used to configure your phone number in above section (default is 8080)
  • Install dependencies:

npm i

Step 4: Make it work

  • Launch the websocket server:

npm run start

Voila! The transcription should appear in the server logs now.

🔗 Source GitHub repository is available here.

Feel free to check out the video version of the tutorial for a step-by-step walkthrough with one of our software engineers, Antoine.

We hope you enjoyed this how-to tutorial! Given how much audio data still goes to wasted, we’re always curious to explore the many ways in which transcription tech can be used to remedy that. Let us know if you end up using our API with Twilio, Discord, or other, we’d love to hear from you.

About Gladia

At Gladia, we built an optimized version of Whisper in the form of an API, adapted to real-life professional use cases and distinguished by exceptional accuracy, speed, extended multilingual capabilities and state-of-the-art features, including speaker diarization and word-level timestamps.

Contact us

280
Your request has been registered
A problem occurred while submitting the form.

Read more