
New billing for video calls running via servers
Video calls that go via Voximplant servers (not P2P ones) are billed per megabyte of video traffic.

Video calls that go via Voximplant servers (not P2P ones) are billed per megabyte of video traffic.

Blind transfer is when a person who transfers the call won't talk with the person to whom he transfers it.

We are happy to announce the high quality speech recognition for both audio call records transcription and real-time recognition scenarios.

The new version of Web SDK will help us to accelerate the development process and includes a lot of new features and improvements.

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.

New Features in Voximplant Kit: Update overview We are constantly working to improve our product to make it easier to use and more effective for you. In this update, we have added several useful features. Here’s what’s new:

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.

New Features in Voximplant Kit: Update overview. We are constantly working to improve our product to make it easier to use and more effective for you. In this update, we have added several useful features. Here’s what’s new:

New integrations for Voice AI have arrived: Google's Gemini 2.0 Flash model, featuring seamless voice-to-voice conversation capabilities and ElevenLabs low-latency streaming speech synthesis are now available for Voximplant developers

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 has new realtime speech generation for voice AI from Inworld, our latest Voice AI text-to-speech (TTS) partner. Together, we combine state-of-the-art TTS with carrier-grade connectivity so you can build voice agents that sound like your brand, not a generic robot.