SIP3

SIP3 is a self‑hosted, end‑to‑end platform for real‑time monitoring, analysis, and troubleshooting of Voice over IP (VoIP) and Real‑Time Communication (RTC) traffic. From a developer’s standpoint, it exposes a rich OpenAPI surface that allows programmatic ingestion of call logs, metrics, and diagnostic events. The core engine parses SIP/SDP packets in real time, aggregates call state transitions, and feeds them into a time‑series store that powers dashboards, alerts, and historical trend analysis. This architecture makes SIP3 ideal for environments where latency, data retention, and compliance are critical.

Key Features

• Real‑time analytics dashboard – visualizes live call quality metrics (RTP jitter, packet loss, MOS) and triggers instant alerts.
• OpenAPI integration – RESTful endpoints for querying call records, configuring filters, and pushing custom telemetry.
• Custom flow definition – developers can script non‑standard call flows (e.g., multi‑leg scenarios) using a JSON‑based rule engine.
• Historical trend analysis – time‑series queries enable SLA reporting and capacity planning.

Technical Stack

• Language & Runtime – The core is written in Go, chosen for its low memory footprint and excellent concurrency primitives.
• Frameworks – Uses the Gin web framework for the HTTP API and WebSocket support.
• Database Layer – Ingested packets are stored in PostgreSQL for relational metadata and ClickHouse (or InfluxDB) for high‑velocity time‑series data.
• Message Bus – NATS or Kafka is optional for scaling the packet capture pipeline across multiple nodes.
• Containerization – Docker images are provided; Kubernetes manifests (Helm charts) support rolling upgrades and horizontal scaling.

Deployment & Infrastructure

SIP3 is designed to run on a private network behind your firewall. A single‑node deployment can handle thousands of concurrent calls, while a multi‑pod Kubernetes cluster scales horizontally by adding more packet‑capture workers. The system can be deployed on bare metal, virtual machines, or cloud VMs; it requires only standard TCP/UDP ports (SIP 5060/5061, RTP ranges). Storage persistence is achieved via persistent volumes; the time‑series database can be backed by SSDs for low latency.

Integration & Extensibility

The OpenAPI spec is fully documented with Swagger UI, allowing developers to generate SDKs in multiple languages. Webhooks can be configured for event‑driven integrations (e.g., Slack alerts, PagerDuty incidents). A plugin interface exposes hooks for custom metrics collectors or external authentication providers. The configuration is expressed in YAML, making it compatible with infrastructure‑as‑code tools such as Terraform or Ansible.

Developer Experience

SIP3’s configuration is declarative and version‑controlled, which aligns well with DevOps practices. The documentation covers API endpoints, data models, and deployment patterns in depth. Community support is active on GitHub Issues and a dedicated Slack channel where developers share custom scripts and troubleshooting tips. Licensing is permissive (MIT), eliminating vendor lock‑in.

Use Cases

• Telecom operators monitoring PBX traffic for QoS and fraud detection.
• Enterprise IT ensuring compliance with regulatory requirements by archiving call records.
• SaaS providers offering built‑in diagnostics to their customers without exposing raw packet data.
• Research labs studying real‑time communication protocols at scale.

Advantages

SIP3’s Go‑based core delivers low CPU overhead, making it suitable for high‑traffic environments where traditional Java or Node.js solutions struggle. The dual database strategy balances relational integrity with rapid time‑series querying, giving developers both auditability and performance. Its open API and extensible architecture allow rapid integration into existing monitoring stacks, while the permissive license removes licensing costs that plague commercial VoIP analytics tools.