Adding peer-to-peer communications to an application is relatively straight-forward. Developers can leverage WebRTC APIs or a CPaaS service to quickly add real time voice and video to their web or mobile app. But, what if you want to hold a meeting with more than two people? How can you leverage powerful WebRTC APIs to build a multi party conferencing application?
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.
We are happy to announce that video calls that use H.264 video codec can now be recorded. Recorded video calls that use H.264 will be stored as mp4 files (calls with video in VP8 format are stored as webm files).
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.
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.
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