achrafka

achrafka / sipstress

Public

Python SIP call generator and diagnostics with INVITE/RTP, health checks, and JSON/HTML/PDF reports

12
2
89% credibility
Found May 17, 2026 at 12 stars -- GitGems finds repos before they trend. Get early access to the next one.
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AI Analysis
Python
AI Summary

sipstress is a Python tool that helps engineers and quality assurance teams test their Voice over IP phone systems. It works like an automated caller that places real phone calls through your VoIP infrastructure, measures how well everything performs, and produces detailed reports. You can simulate a single test call to check basic connectivity, or run stress tests with many simultaneous calls to see how your system handles heavy traffic. The tool measures important things like how quickly calls connect, whether audio comes through clearly, and how much network delay or packet loss occurs. It generates easy-to-read reports showing pass/fail results, response time charts, and audio quality metrics. This is a legitimate testing tool for VoIP infrastructure, similar to how web developers use load-testing tools to check if their websites can handle high traffic.

How It Works

1
📞 You have a phone system to test

You've set up a VoIP server, a phone gateway, or a call center system, and you want to make sure it works well under pressure.

2
⚙️ You connect your testing tool

You point the tool at your phone system's address and tell it which phone numbers to dial, like setting up a test call.

3
🚀 You launch your test calls

With one command, you send out dozens or hundreds of simultaneous calls through your system, watching them ring and connect in real time.

4
Choose your test style
📻
Simple call test

Make calls that just ring and stay open for a set time, measuring how well your system handles the basic connection.

🎛️
Interactive IVR test

Navigate through voice menus by sending keypad tones, testing automated phone systems like customer service bots.

5
📊 You watch the live dashboard

As calls happen, you see a real-time view showing how many succeeded, how fast they connected, and whether audio came through clearly.

6
📄 You get detailed reports

After the test, you receive comprehensive reports showing response times, audio quality, and any problems your system encountered.

✅ You know exactly how your system performs

You now have clear answers: Can your phone system handle 50 calls at once? Where does it slow down? Is the audio quality good? Everything is documented and ready to share.

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AI-Generated Review

What is sipstress?

sipstress is a Python SIP call generator and diagnostic tool for load testing and benchmarking VoIP infrastructure. It places real INVITE dialogs through an SBC or OpenSIPS server, opens RTP streams for a configurable duration, and measures everything: response codes, setup latency, answer latency, jitter, packet loss, and per-call traces. After each run it produces console summaries, JSON reports, interactive HTML dashboards with Plotly charts, and PDF exports. It ships with a rich CLI supporting load shaping via `--calls`, `--cps`, and `--concurrency` flags, plus WAV recording and optional microphone input into outbound RTP.

Why is it gaining traction?

SIPp is the old standard for SIP testing, but it requires XML scenarios and is notoriously opaque to maintain. sipstress offers a SIPp-like workflow in readable Python, with YAML configs, a proper CLI, and multiple output formats that fit naturally into CI pipelines and team workflows. The ability to emit HTML dashboards and PDF reports from a single command makes sharing results with non-technical stakeholders trivial. Health checks with configurable thresholds give you a clear PASS/FAIL without manual analysis, and the RTP quality metrics surface network problems that SIP response codes alone cannot.

Who should use this?

VoIP engineers evaluating SBC or OpenSIPS configurations will find the most value here. Platform developers running IVR integrations can use the health checks to validate call quality before releasing changes. QA teams building automated regression suites for SIP infrastructure can wire this into CI pipelines using the JSON output and YAML config support. Teams testing PSTN bridges, call routing logic, or media relay behavior can use the WAV recording and DTMF support to validate end-to-end call flows.

Verdict

The project is functional and well-structured for its purpose, but with 12 stars it remains early-stage. The credibility score sits at 0.8999999761581421%, which is respectable for a niche tool but signals you should review the test coverage and documentation carefully before committing to production use. If you need SIP load testing with human-readable output and Python flexibility, sipstress is worth evaluating now. Watch the repository for maturity signals like expanded test suites and more real-world usage before treating it as a mission-critical dependency.

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