
Promise support has arrived to VoxEngine
Now developers can use Promise in their VoxEngine scenarios and we also added Net.httpRequestAsync and Net.sendMailAsync functions.

Now developers can use Promise in their VoxEngine scenarios and we also added Net.httpRequestAsync and Net.sendMailAsync functions.

Your mp3 or ogg files played on VoxEngine scenario level with call.startPlayback or using Player will be played on the Web or Mobile SDK side in HD quality (48KHz), or on SIP side if it does support wideband audio codecs (Speex or Opus).

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.

Agentic VoiceDSP handles turn-taking for voice agents on your own STT, LLM and TTS: pauses, interruptions, silence and noisy lines, right inside Voximplant.

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 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

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.

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