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.
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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.
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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
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