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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A practical guide to running a compliant voice AI stack on Voximplant: EU data residency, compliance documentation, and what to verify at each layer — telephony, LLM, STT, and TTS
Agentic VoiceDSP handles turn-taking for voice agents on your own STT, LLM and TTS: pauses, interruptions, silence and noisy lines, right inside Voximplant.
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 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