jigasi  by jitsi

Server-side gateway for Jitsi Meet, linking SIP clients to conferences

Created 11 years ago
582 stars

Top 55.6% on SourcePulse

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Project Summary

Jigasi is a server-side application that bridges SIP clients into Jitsi Meet conferences and provides transcription services. It targets developers and administrators of Jitsi Meet deployments looking to integrate traditional telephony or add real-time speech-to-text capabilities to their video conferences.

How It Works

Jigasi acts as a SIP user agent, registering with a SIP server to enable incoming and outgoing calls to Jitsi Meet conferences. It handles XMPP signaling, ICE negotiation, DTLS/SRTP encryption, and multi-SSRC management for SIP clients. For transcription, it processes audio streams from conferences and forwards them to various speech-to-text services.

Quick Start & Requirements

  • Install: Clone the repository (git clone https://github.com/jitsi/jigasi.git), build with Maven (mvn install -Dassembly.skipAssembly=false), and extract the appropriate zip archive.
  • Prerequisites: Java Development Kit (JDK), Maven, a SIP server, and an XMPP server (e.g., Prosody).
  • Configuration: Requires configuring SIP and XMPP accounts, including MUC components for call control.
  • Docs: https://github.com/jitsi/jigasi

Highlighted Details

  • Supports SIP clients joining Jitsi Meet conferences.
  • Enables real-time transcription and translation using services like Google Cloud Speech-to-Text, Vosk, Whisper, and Oracle Cloud AI Speech.
  • Facilitates LibreTranslate integration for real-time translation.
  • Configurable via properties files and runtime REST API calls.

Maintenance & Community

Jigasi is part of the Jitsi project, a widely used open-source conferencing suite. Community support is available through Jitsi's channels.

Licensing & Compatibility

Jigasi is released under the Apache License 2.0, allowing for commercial use and integration into closed-source projects.

Limitations & Caveats

The setup process involves complex configuration of both SIP and XMPP servers. Transcription service integration requires separate setup and potential API costs. The project is primarily documented through its README, with limited separate documentation.

Health Check
Last Commit

1 week ago

Responsiveness

1 day

Pull Requests (30d)
2
Issues (30d)
2
Star History
4 stars in the last 30 days

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