
HD Audio Conferencing
In HD mode audio is being mixed at 48KHz, all audio sources with lower sample rate will be resampled to 48KHz.

In HD mode audio is being mixed at 48KHz, all audio sources with lower sample rate will be resampled to 48KHz.

We chose 48 KHz as the base sample rate for HD audio recorder, since WebRTC/Opus can offer this quality, audio from endpoints with lower sample rate will be re-sampled.

Full Featured Instant Messaging

If a call is made in non-P2P mode then its media stream goes via our media servers and we can record it if required.

We've started with audio, then we've added video calls and now it's time to let our developers use instant messaging and presence - two very important features of UC stack.

The new version of our mobile SDK uses WebRTC engine for audio/video processing and supports all features available for WebSDK.

Now there is a way to restrict access to VoxImplant HTTP API and only allow it for certain IP addresses or networks when api_key is being used.

Now developers can get phone numbers connected to VoxImplant in more than 40 countries, including United States, United Kingdom, Russia, Argentina, Australia, Austria etc.

All VoxImplant developers can now create/edit/delete Queues and Skills remotely using HTTP API.

We introduced a lot of new features in 2014, but we have even more planned for 2015.

Mozilla recently released Firefox 34 and there were some changes in WebRTC stack that weren't compatible with our Web SDK. We have fixed most of them, p2p video calling will be fixed on Monday.

We are going to provide ready-to-use VoxEngine scenarios, How To's and screencasts.

Voximplant has added Secrets, a dedicated credential store for API keys, tokens, and other sensitive values that VoxEngine scenarios need at runtime

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 supports Inworld's Realtime API, so you can bring Inworld's expressive, conversation-aware agents into real phone calls, SIP, and WhatsApp without custom media infrastructure

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.

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.

Connect any OpenAI-compatible text LLM to your voice AI pipeline in VoxEngine via Chat Completions or Responses API. Set baseUrl to any provider or your own deployment, including EU endpoints for data residency

Voximplant now lets developers build full-cascade voice AI pipelines in VoxEngine without sacrificing turn-taking quality.

Voximplant AI Agent Skills let your coding agent build and ship voice applications without switching tools