SIP & VoIP knowledge in your editor.
The Sipflow MCP server exposes a curated VoIP knowledge base (vendor docs across 38 stacks, SIP/SDP RFCs, and STIR/SHAKEN material) plus four utilities for searching, validating Identity headers, minimizing traces, and detecting the SIP stack. Hook Cursor, VS Code, Claude, or Codex CLI up to it in seconds and ask SIP questions where you already work.
When to reach for it.
Three workflows where having vendor-specific SIP knowledge inside your editor changes the loop.
Debug a live SIP failure with a pcap
- Drop a pcap into your editor, ask the agent to detect the stack and look up vendor-specific guidance for the failing transaction (488 codec mismatch, 403 from carrier, missing ACK, NAT issues).
Configure Kamailio, FreeSWITCH, Asterisk, OpenSIPS
- Hand the agent your dialplan, kamailio.cfg, or sip.conf and ask why a re-INVITE is being rejected, why the registrar isn't matching, or how to wire up STIR/SHAKEN signing on this stack.
Look up RFC behavior without leaving your editor
- RFC 3261/3262/3263/3264, RFC 4566 (SDP), RFC 6028, RFC 8224 / 8225 / 8226 (STIR), RFC 9795 (RCD). Ranked snippets, cited source URLs, no context-switching to a browser.
What it ships.
Four tools and a resource template (sipflow://docs/{vendor}/{id}) for re-fetching any chunk returned by search.
search_sip_docs
Vector search over the curated VoIP corpus
Embed a question, search vendor docs, RFCs, and STIR/SHAKEN material across 38 vendor / stack slugs. Returns ranked snippets with source URLs. Cite source_url verbatim.
- query (required): natural-language question or keywords.
- vendor (optional): restrict to one of asterisk, freeswitch, kamailio, opensips, twilio, cisco-cube, …
- sourceType (optional): rfc | vendor-doc | stir-shaken | blog | wiki.
- limit (1–15, default 5).
validate_stir_shaken_identity
Verify a SIP Identity JWS (RFC 8224 / SHAKEN)
Fetches the x5u certificate, parses it, verifies the ES256 signature against the cert's public key, and optionally validates the RCD icon hash (RFC 9795). Useful for diagnosing 438 Invalid Identity Header rejections, expired certs, and tampered PASSporTs.
- headerB64, payloadB64, signatureB64 (required JWS segments).
- infoUrl (optional): cert URL. When omitted, signature verification is skipped.
- iconUrl, expectedIconHash (optional, RCD).
minimize_sip_trace
Prune noise headers from a raw trace
Pure CPU. No network, no AI cost. Reduces a raw SIP trace (PCAP-decoded text, sngrep export, syslog, or pasted INVITE/200 dialog) to a compact form for an LLM. Preserves SDP bodies and routing/auth/dialog headers; prunes well-known noise (User-Agent, Server, Allow, Accept-*, Date, P-* informational, etc.).
- text (required): raw SIP trace.
- maxBytes (1024–2,000,000, default 200,000): truncation cap.
detect_sip_stack
Identify the vendor / PBX from a trace
Scans User-Agent and Server headers and returns the deduped distinct values plus a best-effort vendor slug. The slug aligns with the vendor filter on search_sip_docs, so you can pipe the output of this tool directly into a follow-up search.
- text (required): raw SIP trace.
Install in seconds.
Pick your client, paste the snippet, restart the editor (or reload its MCP config). One-click install in Cursor.
~/.cursor/mcp.json
json
{
"mcpServers": {
"sipflow": { "url": "https://mcp.sipflow.dev/mcp" }
}
}Reload Cursor (Cmd-Shift-P → "MCP: Reload") to pick up changes.
A typical workflow.
The recommended pattern: detect the stack, search with a vendor filter, then minimize the trace before passing it into your own LLM.
You
Why is FreeSWITCH rejecting this re-INVITE? attached: re-invite.pcap
Agent
Detecting stack…
tool: detect_sip_stack args: { "text": "<pcap>" }Agent
→ vendor: freeswitch (Server: FreeSWITCH-mod_sofia/1.10.x)
Agent
Searching FreeSWITCH docs for re-INVITE rejection…
tool: search_sip_docs args: { "query": "freeswitch re-INVITE rejected SDP", "vendor": "freeswitch", "limit": 3 }Agent
Found 3 sources. The most likely cause is mod_sofia rejecting the re-INVITE because the SDP offers a codec that wasn't in the negotiated answer. Try setting `inbound-codec-negotiation = greedy` or extending `media_codec_array` for this profile. (See: signalwire.com/freeswitch/.../sofia.conf.xml)
Free, unauthenticated, private.
No account. No API key. We don't store the content of your queries. The endpoint is shared with sipflow.dev, so the same Next.js deployment serves both.
- No signup, no API key. Point your client at
https://mcp.sipflow.dev/mcpand you're live. - Query content is not logged or persisted.
- Pcaps and configs you pass to minimize_sip_trace are processed in-memory and never persisted.
- Open standards in, open standards out. Streamable HTTP, JSON-RPC, MCP.
FAQ.
What is MCP?
- The Model Context Protocol (MCP) is an open standard that lets AI editors and agents talk to external tools and data sources over a small JSON-RPC interface. Sipflow exposes its VoIP knowledge base and a few SIP utilities so any MCP-aware client (Cursor, VS Code, Claude Desktop, Codex CLI, Cline, Continue, Windsurf, Goose) can use them in your normal workflow.
Do I need an account or API key?
- No. The endpoint is unauthenticated. We don't log query content.
What does it cost?
- Nothing. The MCP server is free for everyone, no signup, no API key.
Is the corpus open source?
- The vendor docs we index are open source or publicly published. The curated index, the chunking pipeline, and the search are part of Sipflow. The MCP server lives at the same Next.js deployment that serves sipflow.dev. See the README on the project for details.
Does it send my pcap or config to the server?
- No. The server only receives whatever your editor explicitly sends, typically a search query string. The minimize_sip_trace tool processes text in-process and returns the minimized output; nothing is persisted.
Ship one MCP entry. Skip context-switching for the rest of the week.
Free. No signup. Works with every popular MCP client.