KubeVoIP Docs Quickstart

KubeVoIP Docs

KubeVoIP documentation

Install KubeVoIP, understand the SIP platform resources, and operate Kamailio, RTPengine, Asterisk, and PostgreSQL on Kubernetes.

Start the quickstart Explore concepts
Getting Started

Two-phone quickstart

Install the Helm chart, create users and routes with the CLI, register Alice and Bob, and place the first SIP call.

Concepts

Platform resources

Learn how SIPGateway, MediaRelay, SIPUser, SIPTrunk, DialPolicy, CallScope, and CallRoute fit together.

Networking

SIP and RTP flow

Understand external addresses, LoadBalancer mode, HostNetwork mode, port preservation, and media relay behavior.

Operations

PostgreSQL runtime data

See what KubeVoIP stores in PostgreSQL, why HA1 values matter, and how migrations are handled.

Security

Secrets and credentials

Keep SIP passwords, trunk credentials, caller ID values, and database access out of ConfigMaps and Git.

Reference

Generated API reference

Browse current Kubernetes resource fields, validation constraints, names, scopes, and API versions.

Suggested reading path

  1. Install KubeVoIP into a namespace-scoped operator release.
  2. Run the two-phone quickstart with the CLI and confirm internal SIP calling.
  3. Read the LoadBalancer networking guide before exposing SIP or RTP externally.
  4. Prepare a production setup with real PostgreSQL, SIP device Secrets, and dial policy.
  5. Add provider-neutral SIP trunks when you are ready for external calls.