
High quality Speech Recognition is now available
We are happy to announce the high quality speech recognition for both audio call records transcription and real-time recognition scenarios.

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.

Victor Pascual from Quobis invited us to participate in WebRTC meetup that took place on March 4th in Barcelona, we accepted the invitation and I'm really happy that we did.

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.

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

The new integration enables instant connection of any Voximplant call to an Ultravox agent, delivering seamless voice-to-voice conversations.

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

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 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 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.

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.