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.
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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 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.
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.
Voximplant now includes a native Grok module that connects any Voximplant call to xAI’s Grok Voice Agent API for real-time, speech-to-speech conversations. With a single VoxEngine scenario, you can interact via audio with Grok over phone numbers, SIP trunks and infrastructure, WhatsApp Business, or WebRTC into Grok — all without building custom media gateways or WebSocket streaming infrastructure.
Check out the latest useful Voximplant Kit updates — we developed chat analytics, improved call history, added new tools for supervisors, expanded scenario capabilities, and updated the softphone. Below is a brief overview of the essential enhancements.