SIP3

SIP3 is a self-hosted VoIP monitoring and analytics platform focused on SIP signaling and RTP media. It captures traffic, reconstructs call flows, and calculates voice-quality and performance metrics to help operators and engineers troubleshoot and assure service quality.

Key Features

• SIP signaling capture and correlation into call legs/sessions with message-level details
• RTP/media analysis with common VoIP quality metrics (e.g., jitter, packet loss, latency; MOS where supported)
• Call flow visualization and drill-down troubleshooting (who-called-who, routing, failures)
• Search and filtering across calls/sessions for investigations and support workflows
• Time-series dashboards for QoS/QoE and signaling statistics
• Alerting/thresholding for degradations and abnormal behavior (implementation depends on stack)
• Horizontal scaling via collector/receiver components and backend storage

Use Cases

• NOC monitoring of VoIP/SIP trunks, SBCs, and PBX platforms
• Root-cause analysis of one-way audio, bad quality, failed calls, and routing issues
• Capacity planning and SLA reporting using aggregated QoS and signaling metrics

Limitations and Considerations

• Feature depth (dashboards, alerts, MOS support) depends on the deployed SIP3 stack and configuration; some advanced analytics may require additional components.

SIP3 is suited to telecom/VoIP environments needing packet-based visibility into SIP and RTP behavior. It provides the tooling to move from raw captures to actionable call-centric analytics and ongoing quality monitoring.