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).
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.
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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
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:
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 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 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.
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.