SIP & VoIP utilities.
Small, focused tools for VoIP engineers. They run in your browser or your editor - no signup required. Pick the one that matches what you're trying to do.
- New
Test numbers
Free SIP/VoIP test phone numbers
Dial-in test DIDs for VoIP engineers: caller-ID readback IVR, 1-second echo, milliwatt 1004 Hz, real-time DTMF playback, STIR/SHAKEN attestation readback, time announcement, and T.38 faxback.
See the numbers
- Browser-side
STIR/SHAKEN
STIR/SHAKEN Identity decoder
Paste a SIP Identity header (or share link) to decode the PASSporT, verify the certificate chain, and inspect Rich Call Data - RFC 8224 / 8225 / 8226 / ATIS-1000074 / RFC 9795.
Open the decoder
- New
DNS & TLS
SIP DNS & TLS diagnostics
RFC 3263 NAPTR/SRV/A walk with a 3-way consensus matrix across Sipflow, Cloudflare DoH, and Google DoH, plus sips TLS handshake with RFC 5922 §7.2 cert checks. Shareable links.
Diagnose a target
- Browser-side
WebRTC
WebRTC diagnostics
One-click browser and network readiness check: API support, media devices, codec detection, WSS SIP handshake, STUN/TURN ICE gather, and local audio loopback - all from your browser.
Run diagnostics
- New
Softphone
WebRTC SIP softphone
Browser dialer built on JsSIP. Multiple profiles, custom SIP headers, STUN/TURN, raw WebSocket log, live diagnostics, STIR/SHAKEN attestation badge, and one-click handoff into the SIP Flow analyzer. Everything stays on this device.
Open the dialer
- New
Fax over IP
T.38 & T.30 fax decoder
Drop a PCAP that carries fax: the analyzer classifies T.38 (UDPTL) or T.30 G.711 pass-through, surfaces failed reinvites (488 / 415 / 606), and decodes the pages to TIFF in your browser via spandsp compiled to WebAssembly.
Decode a fax PCAP
- Browser-side
IP Lookup
IP & client diagnostics
See your public IP with geolocation, ISP, ASN, and VPN/proxy/TOR signals. Includes browser, device, and connection details plus a Cloudflare-based speed test. Shareable links.
Check your IP
- Editor
MCP server
Sipflow MCP - SIP & VoIP knowledge in your editor
Hook Cursor, VS Code, Claude, or Codex CLI up to a curated VoIP knowledge base (Asterisk, FreeSWITCH, Kamailio, OpenSIPS, Twilio, Cisco, RFCs, STIR/SHAKEN) plus 20+ utilities: search, validate Identity, minimize a trace, detect the stack, and more.
Install in seconds