- 1Runs onLinux, self-hosted
- 2MOS scoringYes
- 3Jitter trackingYes
- 4Packet-loss trackingYes
- 5Synthetic call testsNo
Overview
sipmon is a free, open-source Linux tool for passively monitoring SIP signaling and RTP media quality. It runs as a standalone Rust executable and does not require a running PBX. You can feed it live interface captures, pcap or pcapng files, tcpdump output from standard input, or recorded event logs. Calls and SIP transactions are correlated by Call-ID, with SIP-over-TCP reassembly supported. Reports include RFC3550 jitter and packet loss, RTCP round-trip time, one-way delay estimates, and estimated E-model MOS. More than 20 diagnostic rules address Contact reachability, Record-Route, SDP/RTP consistency, one-way media, and TURN allocation or refresh. sipmon can learn TURN servers and identify client and peer relay legs. Its terminal interface has overview, call, statistics, stream, event, and IP pages. JSONL export and Call-ID queries support scripting. Raw RTP payload is not stored.
Who it is for
It suits Linux operators investigating SIP and RTP behavior from live or recorded captures. Self-hosted deployments, mirrored capture ports, and headless or replay workflows are documented.
What is good
- Free and open source
- Measures jitter, loss, delay, and MOS
- Supports live, offline, and stdin captures
- More than 20 diagnostic rules
- JSONL export and Call-ID queries
What to know first
- Linux and self-hosted only
- Capture can compete with rtpengine CPU
- 100,000-call memory cap
- No synthetic call tests
Verdict
sipmon combines signaling correlation, media measurements, diagnostics, and terminal reporting in one Linux executable. Its documented capture limits and CPU caveat matter for busy relay hosts.
Compared on VoIP monitoring software
- MOS scoring
- Yesgithub.com
- Jitter tracking
- Yesgithub.com
- Packet-loss tracking
- Yesgithub.com
- Synthetic call tests
- Nogithub.com
Facts
- Purpose
- sipmon is a passive SIP/RTP signaling and media quality monitor that runs as a standalone Rust executable without depending on a running PBX.github.com · 4 Oct 2026
- Capture inputs
- It accepts live interface captures, offline pcap/pcapng files, tcpdump streams from stdin, and recorded event logs.github.com · 4 Oct 2026
- SIP analysis
- It correlates SIP calls and transactions by Call-ID and supports SIP-over-TCP reassembly.github.com · 4 Oct 2026
- Media metrics
- It reports RFC3550 jitter and loss, RTCP round-trip time, one-way delay estimates, and estimated E-model MOS.github.com · 4 Oct 2026
- Diagnostics
- The project lists more than 20 diagnostic rules covering Contact reachability, Record-Route, SDP/RTP consistency, one-way media, and TURN allocation or refresh.github.com · 4 Oct 2026
- TURN support
- sipmon can automatically learn TURN servers and label TURN client and peer relay legs.github.com · 4 Oct 2026
- Interface and exports
- It provides a terminal UI with Overview, Call Detail, SIP Stats, Streams, Event Log, and IP Stats pages, plus JSONL exports and Call-ID queries.github.com · 4 Oct 2026
- Integrations
- The README documents piping a live tcpdump capture into sipmon and describes capture setups involving Kamailio and rtpengine.github.com · 4 Oct 2026
- Privacy and storage
- The event log stores analysis summaries and truncated raw SIP messages, and the README states that raw RTP payload is not stored.github.com · 4 Oct 2026
- Linux downloads
- The README says prebuilt static binaries are available for Linux x86_64 and aarch64, with installation from GitHub Releases or Cargo.github.com · 4 Oct 2026
- Capture caveat
- On busy relay hosts capture competes with rtpengine for CPU; the README recommends filtering traffic or using a dedicated mirrored capture box at high packet rates.github.com · 4 Oct 2026
- Call analysis
- It correlates SIP transactions and calls by Call-ID and supports SIP-over-TCP reassembly.github.com · 8 Oct 2026
- Interfaces and exports
- It provides a terminal UI, JSONL exports, and a query command that retrieves a Call-ID flow for scripting.github.com · 8 Oct 2026
- Deployment
- The README describes deployment on a mirrored port or packet-capture box, with live, record, replay, and headless modes.github.com · 8 Oct 2026
- Linux binaries
- Prebuilt static Linux binaries are offered for x86_64 and aarch64, and the README also gives a Cargo installation option.github.com · 8 Oct 2026
- Storage limits
- The documented defaults include a 100,000-call memory cap, a 15-minute call TTL, and 50,000-entry caps for RTP streams and diagnostics.github.com · 8 Oct 2026
- Packet data
- The event log stores analysis summaries and truncated raw SIP messages, and the README says raw RTP payload is not stored.github.com · 8 Oct 2026
- Integration
- The README documents capturing Kamailio SIP and rtpengine media together, including a topology where SIP is mirrored to the rtpengine host.github.com · 8 Oct 2026
- Maker
- The repository links to miuda.ai, whose linked About page says the company was founded by a team of telecommunications and AI experts; it does not state a founding year or headquarters.miuda.ai · 8 Oct 2026
- Support
- The maker's site lists [email protected] as a contact email.miuda.ai · 8 Oct 2026
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Sources
- github.com/miuda-ai/sipmon· checked 4 Oct 2026
- miuda.ai/about· checked 8 Oct 2026
- miuda.ai· checked 8 Oct 2026