Zenko VoIP Server is a fully open-source, sovereign SIP/VoIP solution developed as part of the Zenth Cloud ecosystem by Sky Genesis Enterprise. It provides a modular, secure, and extensible platform for real-time voice communication using open standards like SIP and RTP.
- ✅ Full SIP/VoIP stack (INVITE, REGISTER, BYE, etc.)
- ✅ Native support for audio call routing and SIP trunking
- ✅ LDAP authentication and user provisioning
- ✅ Integrated with Zenth API (
api-server) and Zenth OS - ✅ Built-in NAT traversal (STUN/TURN/ICE)
- ✅ Support for TLS/SRTP for secure communication
- ✅ Fully customizable dial plans and extensions
- ✅ Modular backend compatible with Asterisk or Kamailio
Zenko VoIP Server is part of the sovereign and ethical Zenth Cloud suite, and integrates seamlessly with:
ldap-server– User directorypanel-server– Web administration interfacestatus-server– Real-time service monitoringapi-server– Central unified API interfacefirewall-server– Secure VoIP network perimeterdns-server– SIP SRV and ENUM support
- SIP stack based on Asterisk
- Containerized for deployment in Proxmox, Docker, or Kubernetes
- AGPLv3 licensed for maximum openness and transparency
Full documentation is available in the /docs folder or on Documentations.
This project is licensed under the GNU Affero General Public License v3 (AGPLv3). See LICENSE for more details.
For contributions, issues, or forks, feel free to open a pull request or start a discussion on Github Repo.