Voice

WebRTC

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WebRTC (Web Real-Time Communication) is a free, open-source project that adds real-time voice and video capabilities to web browsers and mobile applications. It lets web apps stream audio and video media directly between peers without plugins or third-party software.

WebRTC combines a set of W3C APIs and IETF protocols that enable peer-to-peer data sharing, audio/video capture, and network traversal. SignalWire uses WebRTC as the transport layer for browser-based voice and video calling.

Key benefits

  • No plugins required — works natively in all major browsers.
  • Cross-platform — connects devices regardless of operating system.
  • Encrypted by default — all media is secured with SRTP (Secure Real-Time Transport Protocol).
  • Adaptive quality — adjusts bitrate and resolution based on network conditions.
  • Interoperable — works alongside SIP, PSTN, and other voice/video protocols.

Browser support

All major browsers support WebRTC natively:

  • Google Chrome
  • Mozilla Firefox
  • Safari (macOS and iOS)
  • Microsoft Edge
  • Opera
  • Brave
  • Vivaldi

No plugins or extensions are needed — users grant microphone/camera permissions and connect immediately.