StarTrinity SIP Tester: Complete Guide to Features & Setup

Step-by-Step StarTrinity SIP Tester Tutorial for Network EngineersThis guide walks network engineers through using the StarTrinity SIP Tester to test SIP-based VoIP systems. It covers installation, configuration, test scenarios, interpreting results, and practical tips for troubleshooting SIP call quality and signaling issues.

What the StarTrinity SIP Tester does (brief)

The StarTrinity SIP Tester is a traffic generator and analyzer for SIP (Session Initiation Protocol). It can simulate SIP endpoints, generate call traffic, measure call setup and teardown behavior, record call metrics (latency, jitter, packet loss), and help validate SIP servers, PBXs, SBCs, gateways, and network QoS.

Before you begin — requirements

• A machine (Windows or Linux) with sufficient CPU and network capacity to generate desired call volume.
• StarTrinity SIP Tester software (download from StarTrinity).
• Access to the SIP server(s) or PBX you will test, including credentials and dialplan info.
• Test media files (optional) for RTP audio.
• Network visibility and permissions to send/receive SIP/RTP traffic (open UDP/TCP ports as needed).
• Basic familiarity with SIP, RTP, codecs (G.711, G.729, etc.), NAT, and QoS.

Installation and initial setup

1. Download and install the StarTrinity SIP Tester on your test machine following vendor instructions.
2. Start the application and open the main configuration dashboard.
3. License: enter license key if required, or run in demo mode for basic testing.
4. Network interface: select the correct NIC that will send/receive SIP/RTP traffic. If you have multiple NICs, pick the one on the same network as the SIP server.
5. Time sync: ensure the test machine clock is synced (NTP) — accurate timestamps are important for measurements.

Creating a basic test scenario

1. New scenario: create a new test scenario and give it a descriptive name (e.g., “100-calls-G711-to-PBX”).
2. SIP accounts/endpoints: define caller and callee accounts. Provide SIP URIs, authentication credentials (username/password), and registration options if needed.
3. Dialplan or call routing: configure dialed numbers and mapping rules so the tester knows how to reach the destination endpoints.
4. Call generation parameters: set the number of concurrent calls, call rate (calls per second), call duration (mean and variance), ramp-up time, and test duration.
5. Media settings: choose codec(s) (e.g., G.711 for uncompressed audio, G.729 if licensed), RTP port ranges, and whether to use real media files or silent RTP.
6. NAT & keepalive: configure STUN, SIP ALG workarounds, or periodic keepalive options if the test crosses NAT.

Advanced SIP options

• Transport: choose UDP, TCP, or TLS depending on your server configuration.
• SIP headers and custom SIP messages: add or modify headers like User-Agent, P-Asserted-Identity, or custom headers required by the SIP server.
• Forking and forking behavior: simulate scenarios where a single INVITE forks to multiple devices.
• Early media: configure to handle 183 Session Progress + SDP and receive early RTP.
• Re-INVITE and session refresh: test mid-call codec changes and session timer behavior.

Running the test

1. Validate configuration: use built-in validation to check SIP URIs, credentials, and port availability.
2. Start a short smoke test (1–5 calls) to confirm basic registration and media flow. Verify audio paths and that RTP packets are flowing both directions.
3. Increase to planned load gradually — use ramp-up to avoid sudden overload.
4. Monitor real-time graphs for call attempts, successes, call setup time (call setup delay), packet loss, jitter, and MOS (if provided).
5. Capture logs: enable SIP trace (pcap or internal logs) for failed calls and signaling details.

Interpreting key metrics

• Call Setup Time (CST): time from INVITE to 200 OK — high values indicate signaling delays or overloaded SIP servers.
• Call Completion Rate (CCR): percentage of successfully established calls — low values suggest authentication, routing, or capacity issues.
• MOS (Mean Opinion Score): subjective audio quality estimate derived from packet loss and jitter — values below 3.5 indicate poor user experience.
• Jitter: variation in packet inter-arrival time — high jitter can cause audio breakup unless jitter buffers compensate.
• Packet Loss: percentage of RTP packets lost — even 1–2% can noticeably affect audio quality for certain codecs.
• RTP path checks: confirm symmetric RTP paths (NAT issues cause one-way audio).

Analyzing failures and common troubleshooting steps

• Authentication failures: check SIP credentials, realm, and authentication method (MD5). Enable SIP logs to view ⁄₄₀₇ responses.
• One-way audio: inspect SDP, RTP ports, and NAT behavior. Use STUN or configure media proxying on SBCs.
• High call setup times: verify DNS (SRV/TXT) resolution, network latency, and server capacity. Check for forking or complex call routing.
• Codec negotiation failures: ensure both sides support the same codec and that codec payload types match.
• Call drops: look for mid-call BYE, session timeout, or SIP BYE due to QoS or inactivity.
• Intermittent failures under load: increase server resources, check thread pools, SIP transaction queues, and concurrent call limits.

Collecting evidence — logs & captures

• SIP traces: collect full SIP logs for failed and successful calls. Use the tester’s export or capture to PCAP for Wireshark analysis.
• RTP captures: capture RTP flow for MOS and packet loss verification. Consider using rtpdump or Wireshark to analyze streams.
• Metrics export: export CSV or JSON results (call-by-call) for offline analysis and trending.

Example test scenario (practical)

• Objective: validate PBX capacity for 300 concurrent G.711 calls.
  • Concurrency: 300 calls.
  • Call rate: ramp up over 5 minutes.
  • Call duration: mean 180 seconds (±30s).
  • Codec: G.711 u-law.
  • Transport: UDP.
  • Registration: 50 SIP accounts rotating caller IDs.
  • Monitoring: MOS, packet loss, jitter, CCR, CST.
• Run test, collect PCAPs from both tester and PBX, and compare metrics. Look for resource limits (CPU, threads) on PBX and network bottlenecks.

Best practices

• Use realistic call durations and patterns (not continuous 1-second calls).
• Test with both silent RTP and real audio files to validate media handling.
• Run tests during maintenance windows for production systems.
• Use separate VLANs or dedicated test networks to avoid impacting production voice traffic.
• Automate repeated tests and collect historical metrics to detect regressions.

Wrapping up

After tests, summarize findings with key metrics, PCAP evidence, and recommended remediation (codec changes, NAT/STUN configurations, server sizing). Use the StarTrinity SIP Tester logs and captures to verify fixes and re-run tests for confirmation.