The relevance of remote business has grown rapidly due to changing conditions in world markets. Several companies are facing challenges because they are not set up for their employees to transition to remote work but situations like these call for immediate measures.
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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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 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 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.
Use Realtime API in VoxEngine to build speech-to-speech voice AI with data residency control. Set baseUrl to any OpenAI Realtime-compatible endpoint, including Azure EU deployments, and control where call audio is processed
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