SIPFlow - SIP & VoIP Call Protocol Analyzer Bridge

Tracking since Apr 25, 2026.

Changelog

• May 13, 2026
  description

  SIP Flow is a free, private SIP/VoIP call flow analyzer at sipflow.dev — drop a PCAP, paste a SIP log, see your call ladder in seconds.
  
  This companion extension powers SIP Flow's live feed. It bridges sipflow.dev (HTTPS) to your internal Homer 10 / qryn instance (typically plain HTTP on a private network), bypassing the browser's CORS and mixed-content restrictions.
  
  Privacy • The bridge is a transport relay only — it does not parse SIP and stores no data. • Nothing is sent off your machine; traffic only flows between your browser and your Homer endpoint. • No host permissions by default; access is requested per-origin, only when you connect. • Open source.

  SIP Flow - Bridge is the companion extension for sipflow.dev, the
  free, private SIP / VoIP call-flow analyzer (a browser-side
  alternative to Wireshark + sngrep for SIP).
  
  WHAT SIPFLOW.DEV DOES (the web app this extension connects to):
  • Import and visualize PCAP / PCAPNG packet captures
  • Parse sngrep text dumps, raw SIP messages, and tshark / tcpdump exports
  • Interactive sequence / ladder diagram with sngrep-like keyboard navigation
  • SDP media and codec negotiation detail (offer/answer diff)
  • Message comparison and retransmission detection
  • RFC-grade issue detection — missing 200 OK / ACK, codec mismatch,
    NAT mismatch, auth loops, one-way audio
  • STIR/SHAKEN Identity-header decoding (PASSporT, RCD, cert chain)
  • Share-link generation for tickets — no signup, no upload required
  • MCP server for Cursor, VS Code, Claude Desktop, Codex CLI
  • curl-based PCAP upload for headless servers
  
  It plugs the live workspace at sipflow.dev/live into multiple local SIP sources
  without ever sending traffic off your machine:
  
  1. HOMER / qryn relay
     • Lets sipflow.dev (HTTPS) reach your internal Homer 10 / qryn instance
       over plain HTTP and WebSocket — bypassing the browser's CORS and
       mixed-content restrictions.
     • Proxies HTTP fetch calls to qryn's Loki API.
     • Opens a /loki/api/v1/tail WebSocket on your behalf and streams frames
       back to the page over a chrome.runtime.Port.
  
  2. Browser SIP capture
     • Click the extension icon on any dialer page (JsSIP, SIP.js, Janus,
       OpenSIPS WebRTC, …) to detect SIP/WebRTC and enable capture with a
       single click — no need to visit sipflow.dev first.
     • Observes SIP-over-WebSocket traffic on origins you allow, and hooks
       RTCPeerConnection getStats() to surface real-time audio diagnostics
       — one-way audio, symmetric-NAT TURN failures, codec runtime drift,
       high packet loss, high jitter, ICE flap — directly in the Live
       workspace's issues panel.
     • The toolbar icon changes color: default when idle, grey when SIP is
       detected but not yet capturing, green with "REC" badge when actively
       capturing.
  
  The extension does not parse SIP. All parsing happens in the SipFlow web
  app. The extension keeps captured frames in memory only (a small ring
  buffer, cleared on reload) and never sends them anywhere except to the
  sipflow.dev tab you're already looking at.
  
  WHAT IT DOES NOT DO:
  • It does not collect, transmit, or sell any data.
  • It does not run analytics or telemetry.
  • It does not request broad host permissions by default — every endpoint
    and every dialer origin is opted into individually.
  • It does not modify pages it has not been allowlisted on.
  
  PERMISSIONS YOU'LL SEE:
  • "storage" — remembers your dialer-origin allowlist.
  • "scripting" — registers the content scripts that observe SIP frames on
    the dialer origins you allow.
  • "tabs" — labels captured traffic with the source tab in the SipFlow live
    status pill, and lets the toolbar badge reflect capture state.
  • "activeTab" — when you click the extension icon, the extension runs a
    one-shot script on the active tab to detect whether the page uses
    SIP-over-WebSocket or RTCPeerConnection, and shows the result in the
    popup. The script does not modify the page.
  • "host permissions: <all_urls>" — OPTIONAL. Never granted by default.
    Each origin (your Homer endpoint or your dialer page) is requested
    individually — either from the popup's "Enable capture" button or from
    the SipFlow settings drawer — and you can revoke at any time from
    chrome://extensions.
  
  WHO IT'S FOR:
  • VoIP / SIP engineers debugging in-browser dialers and softphones.
  • Carrier / SBC / PSAP teams correlating live traffic with Homer captures.
  • WebRTC operators who need real-time getStats() insight without writing
    custom test pages.
  • Network engineers who reach for Wireshark, tshark, tcpdump, or sngrep
    and want a faster ladder view in the browser.
  • NOC / helpdesk teams who receive PCAPs and need a quick visual without
    installing Wireshark.
  • Anyone replacing the deprecated original "SIP Flow" Chrome extension —
    this is its supported successor.
  
  PRIVACY POLICY:
  https://sipflow.dev/privacy#chrome-extension

• May 13, 2026
  short_description

  Companion add-on for SIP Flow's live feed. Bridges sipflow.dev to your internal Homer 10 / qryn over HTTP and WebSocket.

  Companion for sipflow.dev — the free SIP/VoIP analyzer. Bridge Homer, capture browser dialers, debug with PCAPs.

• May 13, 2026
  name

  SIP Flow — Homer Bridge

  SIPFlow - SIP & VoIP Call Protocol Analyzer Bridge

• May 13, 2026
  permissions

  (empty)

  storage, scripting, tabs, activeTab

Permissions & access

Permissions

storagescriptingtabsactiveTab

Host access

None declared

Screenshots