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
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
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:
Check out the latest useful Voximplant Kit updates — we developed chat analytics, improved call history, added new tools for supervisors, expanded scenario capabilities, and updated the softphone. Below is a brief overview of the essential enhancements.
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 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.