
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.

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

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

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

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.

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 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.

Voximplant now includes a native MCP Client for VoxEngine, giving developers direct connectivity to any MCP server and full control over every tool call

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.