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.
Voximplant serverless architecture provides a simpler alternative to deploying traditional VoIP servers in a datacenter. Voximplant APIs offer developers scale and on-demand communications services. Learn how easily you can create VoIP services in the form of Voximplant-based apps.
Each country has established an independent phone number format valid only for local communications. To reach customers worldwide, you should apply the E.164 phone number format.
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.
When a user connected to an enterprise PBX makes a VoIP phone call to a customer’s landline or mobile phone, the communications protocol has to be converted. In this article, we explain how gateways and SIP trunks enable interoperability between a PBX and the PSTN.
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.
You will need а microphone and a PC or mobile device for a quick start. This how-to will guide you through a selection of hardware and software as well as making the right recordings.
Analyzing sources of customer requests is a very important business task. When you’re dealing with web traffic, it’s fairly easy to do — businesses can get behavioral customer data with the help of services like Google Analytics. However, when it comes to calls, it might get difficult. Call tracking is a simple solution.
Use Realtime API in VoxEngine to build speech-to-speech voice AI with data residency control. Set baseUrl to any OpenAI Realtime-compatible endpoint, including Azure EU deployments, and control where call audio is processed
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 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 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.
A practical guide to running a compliant voice AI stack on Voximplant: EU data residency, compliance documentation, and what to verify at each layer — telephony, LLM, STT, and TTS
Voximplant now includes a native Deepgram module that connects any Voximplant call to Deepgram’s Voice Agent API for real-time, speech‑to‑speech conversations. You can stream audio from phone numbers, SIP trunks, WhatsApp, or WebRTC into Deepgram’s unified agent environment—combining STT, LLM reasoning, and TTS—and play responses via Voximplant’s serverless runtime with minimal latency.