Adding peer-to-peer communications to an application is relatively straight-forward. Developers can leverage WebRTC APIs or a CPaaS service to quickly add real time voice and video to their web or mobile app. But, what if you want to hold a meeting with more than two people? How can you leverage powerful WebRTC APIs to build a multi party conferencing application?
A REST API is a simple, standardized method of communication between web clients and servers. The main building blocks of the REST API are the request and the response. Learn about the REST API and how to issue requests and receive response data.
With webhooks, your app always knows what happens on the server-side in real time. This makes webhooks ideal for integrating communications apps with events and data from other systems.
Recently, we published a blog post describing why WebSockets are great for real-time services. In this article, we describe the process of establishing, maintaining and closing the WebSockets connection.
Voice Recognition API captures human speech in real-time, transcribes it, and returns it via text. By converting speech to text, you can process live or prerecorded audio, and receive transcriptions and summaries/interpretations with high speed and precision.
Voximplant now includes a native Cartesia module for streaming, low-latency text-to-speech (TTS). You can use a single VoxEngine API to synthesize speech in real time, connect it to any call (PSTN, SIP, WebRTC, WhatsApp) and control playback from a Large Language Model (LLM) or other source, all inside VoxEngine.
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New integrations for Voice AI have arrived: Google's Gemini 2.0 Flash model, featuring seamless voice-to-voice conversation capabilities and ElevenLabs low-latency streaming speech synthesis are now available for Voximplant developers
Today Ultravox announced they are directly integrating Voximplant into their platform to provide SIP capabilities. The integration builds on Voximplant’s deep telephony and Voice AI tooling
Voximplant has added a WebSocket privacy option that redacts message payloads from logs across all WebSocket-based services – Voice AI connectors and external speech system – and speech control modules
Voximplant now includes a native Cartesia Line / Agents connector that connects any Voximplant call to a Cartesia Line voice agent for real-time, speech-to-speech conversations—over PSTN, SIP, WebRTC, or WhatsApp Business Calling—without building custom media gateways or WebSocket streaming infrastructure.
Learn how a Voice AI Orchestration Platform connects LLMs, STT/TTS, turn‑taking, and telephony (PSTN, SIP, WebRTC) to build reliable real‑time voice agents. See benefits, architecture, and how Voximplant helps.