Build a source COMP that ingests external video feeds (screen grab, NDI, camera, stream), normalizes to target resolution, and provides a named output for mixing and effects.
MIT
206,405 tools. Last updated 2026-06-17 13:10
"WebRTC" matching MCP tools:
- Launch an anti-detection browser with configurable OS, locale, proxy, and anti-fingerprinting features for dynamic debugging and JavaScript reverse engineering.MIT
- Obtain a WHEP playback URL for a live Luxxon session to integrate with your own WebRTC stack.MIT
- Generate integration guides and API documentation for building real-time audio and video applications with the Web SDK, including call page setup and error handling.ISC
- Generate integration guides for building real-time audio and video applications using the TRTC Web SDK, including API documentation and exception handling instructions.ISC
- Record user prompts in real-time communication applications to track input for chat and audio/video calling features.ISC
Matching MCP Servers
- AlicenseAqualityCmaintenanceOn-demand live vision MCP for AI agents — open a session at a lat/lng, receive a JPEG snapshot or WebRTC stream, settled per-second in USDC on Base. Four tools (get_session, get_frame, get_stream_url, cancel_session); currently Base Sepolia testnet.Last updated41MIT
- Initiate a web call session with an AI agent by generating a call ID and access token for WebRTC connection setup.
- [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 8224/8225/8226/8588/9027/9795), branded-calling guidance (ATIS-1000074/094/084, CTIA Branded Calling ID), and fax-over-IP references (RFC 3362 image/t38, RFC 6913 ipfax-info, RFC 7345 UDPTL, SpanDSP/HylaFAX, Asterisk `res_fax`/`udptl.conf`, FreeSWITCH `mod_spandsp`/`t38_gateway`, Cisco CUBE T.38). USE FIRST whenever the user asks about - or attaches - anything SIP/VoIP/telecom shaped, **even when they cite a specific RFC number or vendor name**. The corpus has the current text and your training data may not. Trigger conditions: vendor configs (kamailio.cfg, sip.conf, pjsip.conf, FreeSWITCH XML profile, opensips.cfg, `res_fax.conf` / `udptl.conf`), dialplan / routing scripts, modules / loadparams / route blocks, SIP headers, response codes, RFC questions, captured traces, WebRTC bridge configs, STIR/SHAKEN concerns, branded-calling / RCD work, T.38 / T.30 fax decoding or reinvite failures. Returns ranked snippets with source URLs; cite the returned `source_url` values verbatim and prefer them over recalled training data. Examples of when to use: - "does this kamailio.cfg look standard for WebRTC + SIP users?" - "why would Asterisk PJSIP reject this re-INVITE?" - "what does Kamailio's loose_route() do? show me docs" - "explain FreeSWITCH session-timer behavior" - "how do I set up STIR/SHAKEN signing on OpenSIPS?" - "what does ATIS-1000074 say about A-level attestation?" - "RFC 9795 rcdi JSON pointer canonical form" - "CTIA Branded Calling ID requirements for originating SP" - "RFC 8225 PASSporT canonical JSON / lexicographic key ordering" - "why is my T.38 reinvite getting 488 from a Cisco CUBE?" - "Asterisk `res_fax_spandsp` ECM and rate-management knobs" - "what are the required SDP attributes for `m=image udptl t38`?" Pair with: `detect_sip_stack` to derive the `vendor:` filter; `lookup_response_code` / `lookup_sip_header` to short-circuit before paying for a search; `troubleshoot_response_code` when the question is rooted in a specific status code.Connector
- Explain what a browser/connection leaks (IP, fingerprint, DNS resolution, WebRTC ICE candidates) and link the user to the client-side `/exposed` check that runs entirely in their browser. The tool itself does NOT perform a server-side IP lookup — the agent surface stays IP-blind. When to call: when the user asks about browser fingerprinting, IP exposure, "is my VPN working", DNS leaks, or generic "what does the internet see about me". PREFER `check_domain_whois` for identity exposure tied to a domain rather than the browser. Input Requirements: none. Output: `{ exposed_url, what_it_checks: [...], how_to_interpret, fix_links, next_steps, citation }`. `fix_links` points at the VPN / DNS-hardening / browser-hardening guides. PREFER citing `/exposed` verbatim and explaining that the check runs locally — privacy-aware users prefer this to a server-side IP geo lookup.Connector
- [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-relay (`m=image udptl t38` plus the `a=T38Fax*` attribute family), and crypto attributes. Useful for debugging WebRTC ↔ SIP interop (codec negotiation, DTLS-SRTP fingerprints, ICE candidate gathering, bundle alignment), and for inspecting fax negotiation (T.38 reinvite SDP, `T38FaxMaxBuffer`/`T38FaxUdpEC`/`T38FaxRateManagement`) without an LLM having to re-derive the SDP grammar each call. Pair with: `compare_sdp_offer_answer` when the user has both halves of the negotiation (including T.30→T.38 reinvites); `webrtc_sip_checklist` for the bridge-config angle.Connector
- [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 SIP trace AND any prior AI analysis attached to it in a single round trip. Use this whenever a user pastes a `/share/<token>` URL: the tool fetches the redacted trace text, the AI executive summary / root-cause / remediation steps (if present), and metadata (vendor, filename, source format, pseudonymized flag), so the agent can review the trace alongside the user's own configs without manual download + paste. In addition to the AI output, the response includes rule-based diagnostics: detected issues (severity-tagged SIP/SDP/media problems with RFC references), WebRTC signal checklist scores, multi-leg call correlation (Session-ID grouping), and detected SIP stacks (User-Agent/Server header values). These diagnostics are computed at share-creation time; for older shares without persisted diagnostics, the tool parses the trace on the fly. When the share includes media quality data (from PCAP-sourced captures), the response includes per-call MOS/jitter/loss summaries in the text output and full `mediaQuality` stats in `structuredContent`. If `hasRawCapture` is true, the sharer included their original PCAP for full RTP playback on the web UI - this raw binary is not returned to agents. Privacy: the share endpoint deliberately strips the original `problem` and `architecture` fields the sharer typed in (those may contain customer-internal context). This tool returns the same public projection - only the trace, the AI output, diagnostics, and basic metadata. Traces are pseudonymized by default (phone numbers / IPs / Call-IDs replaced with consistent fakes); the `pseudonymized` field tells you whether the sharer opted to keep raw values. Trace bytes are capped at 200kB (matching the budget the Sipflow AI worker uses). For very large captures the response sets `trace.truncated=true` - pair with `minimize_sip_trace` to compact further before passing to your own LLM, or with `render_sip_ladder` to visualize the call flow. Pair with: `review_sip_config` to compare the shared trace against the user's own kamailio.cfg / pjsip.conf / FreeSWITCH XML; `render_sip_ladder` to draw the shared call flow inline; `minimize_sip_trace` if `trace.truncated` is true; `troubleshoot_response_code` for any failing transactions surfaced in the AI analysis.Connector
- [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 from a repo. This tool: 1. Detects the vendor from filename + structural signatures (loadmodule, route blocks, [transport-*] sections, <profile name=>, KEMI calls). 2. Extracts a structured outline: loaded modules, modparams, listen lines, route blocks, profiles, gateways, dialplan extensions. 3. Surfaces risk flags - e.g. websocket loaded without TLS, nathelper without rtpengine, chan_sip used in modern Asterisk, AND the Kamailio/OpenSIPS lump-vs-subst race (`subst('/^From:.../...')` colliding with `KSR.hdr.append/remove` or `uac_replace_*` or `append_hf/remove_hf` on the same header - corrupts the buffer at serialization). 4. Returns a list of `suggestedQueries` for `search_sip_docs` so you can ground the actual review in vendor docs. Pair with: one or more `search_sip_docs` calls (cite returned `source_url` values verbatim instead of recalling vendor behavior from memory); `webrtc_sip_checklist` when the config is a WebRTC ↔ SIP bridge.Connector
- Spawns two AI personas as participants on a real LiveKit voice call (via Paradise's self-hosted comms cluster), each driven by its own LLM (Claude + GPT by default), and runs a structured conversation. Each persona talks aloud (TTS) and listens to the other (Whisper STT) — this isn't simulation, it's a real WebRTC call with real audio. Used to verify Paradise Comms end-to-end (publisher → SFU → subscriber → recording → outbound webhook) and to demo agent-to-agent voice. Returns the full transcript and a recording hint.Connector
- [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 behavior across the bridge, STIR/SHAKEN signing). When `configText` is supplied, each item is marked as 'looks present' or 'check needed' based on simple regex signals. Use when the user is building a WebRTC ↔ SIP bridge or troubleshooting one (no media, one-way audio, ICE failures). Pair with: `review_sip_config` for the full structured outline; `search_sip_docs(vendor=...)` to ground each unchecked item in vendor docs; `parse_sdp` / `compare_sdp_offer_answer` when the bug is in SDP negotiation.Connector
- Deletes a Wavix Embeddable widget token. After deletion, the token can't be used to authenticate widget sessions, and any active session associated with it is terminated.Connector
- Returns a paginated list of active Wavix Embeddable widget tokens. Results are limited to 25 records per page by default. Use `page` and `per_page` to navigate results.Connector
- Returns a Wavix Embeddable widget token configuration.Connector
- Updates the payload for a Wavix Embeddable widget token.Connector
- Creates a Wavix Embeddable widget token.Connector