Skip to content

Voice Calls

Ilias Koukovinis edited this page Jul 31, 2025 · 4 revisions

Voice Calls

I am thrilled to announce, Ermis finally supports voice call functionality, powered by WebRTC.

What is WebRTC?

WebRTC (Web Real-Time Communication), in short, is a free and open-source technology - originally developed by Google -, which facilitates voice/video streaming, enabling real-time communication (RTC) for web browsers and mobile applications. It enables audio and video communication and streaming by allowing direct peer-to-peer and end-to-end-encrypted communication, eliminating the need for an intermediary server - though signaling and TURN servers may be used to facilitate connections, the communication itself is peer-to-peer.

What This Means

With WebRTC, you benefit from:

  • Lower Latency Calls: Direct Peer-to-Peer Communication implies calls with shorter delays!
  • Enhanced Security: With end-to-end encryption, only you and the person you're calling can hear them!
  • Better Scalability: Thanks to WebRTC's P2P communication, a heavy load is taken off the server!

By extension, these advantages all translate into a smoother, more secure calling experience on Ermis!

Availability

Unfortunately, though, at the present moment, voice call functionality is only available on mobile devices. It is highly unlikely that desktop support will ever be available, but nevertheless stay tuned for future updates!

Clone this wiki locally