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?
Thanks to the WebRTC standard, your customers and colleagues can join a call without needing to install or download apps. See how innovative technology impacts the telecommunications world
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.
Victor Pascual from Quobis invited us to participate in WebRTC meetup that took place on March 4th in Barcelona, we accepted the invitation and I'm really happy that we did.
Mozilla recently released Firefox 34 and there were some changes in WebRTC stack that weren't compatible with our Web SDK. We have fixed most of them, p2p video calling will be fixed on Monday.
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
How does personalized customer communication drive business growth? Discover key strategies like data collection, segmentation, tailored messaging, technology, and customer-centric culture. Learn how to boost loyalty, retention, and sales with effective personalization!
Unlock the Full Power of Neural Text to Speech Sounds human-like. Power your applications with lifelike speech. Our low latency models are designed to enhance user interactions, making every conversation more engaging and realistic.
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.
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
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.