Sipflow
by sipflow.dev in Productivity & SaaS
SIP/VoIP/telecom grounding for AI agents: vendor docs, RFCs, STIR/SHAKEN, traces, configs.
https://mcp.sipflow.dev/mcp
Last 30 days
- Uptime
- 100%
- Response time
- 482 ms typical, 482 ms slowest 5%
- Last check
- 4 h ago
- Next check
- in 3 h
How to call it
Add it to any MCP client that supports remote servers.
{
"mcpServers": {
"sipflow": {
"type": "http",
"url": "https://mcp.sipflow.dev/mcp"
}
}
}22 tools
- search_sip_docs
[cost: rag (one embed + one vector search) | read-only, network: outbound to embed model only] Vector search over Sipflow's curated VoIP knowledge base: vendor docs (Asterisk, FreeSWITCH, Kamailio, OpenSIPS, Twilio, Cisco, etc.), SIP/SDP/WebRTC RFCs, STIR/SHAKEN material (RFC 82
- lookup_response_code
[cost: free (pure CPU, no network) | read-only] Instant static lookup of a SIP response code (100-699). Returns name, RFC anchor, category, description, common operator-flavored causes, and known vendor-specific reason-phrase variants (e.g. OpenSIPS emits 484 'Invalid FROM' on F
- lookup_sip_header
[cost: free (pure CPU, no network) | read-only] Instant lookup of a SIP header by canonical or compact form (e.g. "Via" / "v", "Diversion", "P-Asserted-Identity", "Identity", "Session-Expires"). Returns canonical form, compact alias, RFC anchor, where it appears (request / respo
- detect_sip_stack
[cost: free (pure CPU, no network) | read-only] Identify the SIP product behind a piece of input. Works on both: - a SIP trace (User-Agent / Server headers from PCAP/sngrep/syslog), and - a vendor config blob (kamailio.cfg, sip.conf, pjsip.conf, FreeSWITCH XML, opensips.cfg)
- detect_sip_vendor_from_config
[cost: free (pure CPU, no network) | read-only] Heuristic-only sibling of `detect_sip_stack`, scoped to vendor configs. Returns the matched vendor slug, a confidence level, and the structural signals that fired (loadmodule syntax, route blocks, profile elements, etc.). Use this
- minimize_sip_trace
[cost: free (pure CPU, no network) | read-only, no persistence] Reduce a raw SIP trace to a compact form suitable for sending to an LLM. Preserves SDP bodies and routing/auth/dialog headers; prunes well-known noise (User-Agent, Server, Allow, Accept-*, Date, P-* informational, e
- render_sip_ladder
[cost: free (pure CPU, no network) | read-only] Parse a raw SIP trace (PCAP-decoded text, sngrep export, syslog, or pasted INVITE/200 dialog) and emit a Mermaid `sequenceDiagram` block visualizing the call flow. Most chat hosts (Claude, ChatGPT, Cursor, GitHub) render Mermaid in
- sip_ladder_example
[cost: free (pure CPU, no network) | read-only] Return a hand-curated SIP scenario as a Mermaid `sequenceDiagram` plus a bullet list of step-by-step explanations with RFC references. Use this when the user asks 'show me what X looks like' and you don't have a real trace handy.
- lint_sip_request
[cost: free (pure CPU, no network) | read-only, no persistence] Run RFC 3261 / RFC 3325 / RFC 8224 / RFC 8225 / CTIA BCID compliance checks on a single raw SIP request (typically an INVITE) and return a list of findings. Catches the failure modes that silently break carrier int
- parse_sip_message
[cost: free (pure CPU, no network) | read-only, no persistence] Parse a single raw SIP message (request OR response) and return a structured view: start line (method/status), every header in order with line numbers, body, duplicate-header counts, and a list of structural flags t
- diff_sip_messages
[cost: free (pure CPU, no network) | read-only, no persistence] Take two SIP messages (typically the same request observed at two adjacent hops - e.g. the INVITE leaving FreeSWITCH and the INVITE arriving at Kamailio) and surface a structured per-header diff: `added`, `removed`,
- review_sip_config
[cost: free (pure CPU, no network) | read-only] Use this when the user asks 'review my config' or attaches a kamailio.cfg, sip.conf, pjsip.conf, FreeSWITCH XML profile, opensips.cfg, `res_fax.conf` / `udptl.conf` / `spandsp.conf` (fax-relay tuning), or a SIP-shaped source file f
- webrtc_sip_checklist
[cost: free (pure CPU, no network) | read-only] Return a curated checklist of WebRTC ↔ SIP requirements (WSS transport, ICE gathering, DTLS-SRTP fingerprint, rtcp-mux + BUNDLE, media relay / rtpengine, STUN/TURN, secure-context Origin allowlist, Opus codec, session-timer behavio
- parse_sdp
[cost: free (pure CPU, no network) | read-only] Parse a Session Description Protocol body and return a structured view: origin, session, timing, per-media codecs (rtpmap + fmtp), direction, DTLS setup + fingerprint, ICE credentials + candidates, rtcp-mux, BUNDLE groups, fax-rela
- compare_sdp_offer_answer
[cost: free (pure CPU, no network) | read-only] Diff a SIP/SDP offer and answer and surface the issues that actually break calls in practice: codec intersection per m-line, direction compatibility (sendrecv ↔ recvonly), DTLS setup-role conflicts (active+active / passive+passive)
- validate_stir_shaken_identity
[cost: external_io (HTTPS fetch of the x5u cert) | read-only] 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). The icon-has
- stir_attestation_explainer
[cost: free (pure CPU, no network) | read-only] Static explainer for STIR/SHAKEN: maps attestation levels (A / B / C per RFC 8588) to plain-English requirements + common scenarios, and SIP codes commonly emitted by signing/verification (428 / 436 / 437 / 438 / 608) to their RFC
- validate_e164_number
[cost: free (pure CPU, no network) | read-only] Parse a phone number, normalize to E.164, and classify it. International coverage is via libphonenumber-js (every country, line type when known). NANP numbers (CC=1) are additionally split into NPA (area code) / NXX (central office
- troubleshoot_response_code
[cost: rag (one embed + one vector search) | read-only, network: outbound to embed model only | rate-limited per IP] Like `lookup_response_code` but augmented: returns the static RFC entry PLUS the top vendor-specific RAG hits for the exact code (and any free-text context the us
- dns_diagnose_sip_target
[cost: external_io (DNS via Cloudflare + Google; TLS handshake + a SIP OPTIONS keepalive to public targets when applicable) | read-only | rate-limited per IP: 10/min, 200/day] Walk DNS the same way a SIP UA does (RFC 3263 §4.1): NAPTR → SRV → A/AAAA. Given a SIP URI ("sip:exampl
- fetch_sipflow_share
[cost: external_io (Mongo + S3 fetch on the Sipflow backend) | read-only, no persistence | rate limit: shared with the public share endpoint] Given a Sipflow share URL (https://sipflow.dev/share/<token>, or any sipflow.dev subdomain that serves /share/<token>), load the shared S
- submit_sipflow_feedback
[cost: write (single MongoDB row) | rate-limited per IP: 3/min, 20/day] Send the Sipflow team feedback when something doesn't work, a vendor or RFC isn't covered, or a tool produced a wrong/incomplete answer. Categories: - docs_gap: search_sip_docs returned nothing useful, ven
Security scan
- Unusually long description in tool:dns_diagnose_sip_target
dns_diagnose_sip_target [cost: external_io (DNS via Cloudflare + Google; TLS han… - Unusually long description in tool:fetch_sipflow_share
fetch_sipflow_share [cost: external_io (Mongo + S3 fetch on the Sipflow backend)…
Recent checks
| When | Result | HTTP | Time |
|---|---|---|---|
| 4 h ago | Passed | 200 | 499 ms |
| 12 h ago | Passed | 200 | 389 ms |
| 21 h ago | Passed | 200 | 482 ms |