# LiveKit Telephony (slim) > LiveKit Telephony connects phone calls to LiveKit rooms as SIP participants. On LiveKit Cloud it covers SIP trunks to outside carriers, US inbound phone numbers, dispatch rules and transfers, managed through a Twirp API, server SDKs and the lk CLI. - Full: https://www.anchorterminal.com/tools/livekit-telephony.md (~8,200 tokens) · this version ~1,880 tokens · JSON https://www.anchorterminal.com/tools/livekit-telephony.json · canonical https://www.anchorterminal.com/tools/livekit-telephony - Index: https://www.anchorterminal.com/llms.txt · API: https://www.anchorterminal.com/api/v1/index.json · Updated: 2026-10-09 **B · 67.5/100 · rank #234 of 842 · #3 in Phone calling & voice transport · not agent-ready · confidence medium** Assessment: Signed tokens separate the right to dial (`sip.call`) from the right to manage trunks (`sip.admin`), and per-minute prices are public with a free plan that needs no card. LiveKit's own phone numbers are US only and inbound only, so outbound calls need a third-party SIP trunk, and no 429 or retry guidance was found. ## Facts - Kind: HTTP API · vendor: LiveKit Incorporated · category: Phone calling & voice transport · legal entity: LiveKit Incorporated · provenance 81/100 - Local only (HTTP): npm `livekit-server-sdk`, pypi `livekit-api`, go `github.com/livekit/server-sdk-go/v2` - Auth: API key · pricing: Freemium · x402: no · licence: Proprietary cloud service under LiveKit's Terms of Service. The SIP service (livekit/sip), the protocol definitions, the CLI and the server SDKs are Apache 2.0 - Probe metrics: not measured yet (probes haven't run) - Countries with numbers: United States only for LiveKit Phone Numbers (local and toll-free, inbound only). Other countries through a third-party SIP trunk - SIP: Inbound and outbound trunks over UDP, TCP or TLS with SRTP. No SIP REGISTER, SIPREC or video. Tested with Twilio, Telnyx, Exotel, Plivo, Wavix, Sinch and didlogic - Media streams: Call audio joins a LiveKit room as a participant track over WebRTC. Connectors bridge Twilio calls over websockets and WhatsApp Business voice calls - Voice AI layer: LiveKit Agents joins the same room. Agent session minutes on LiveKit Cloud are $0.01 a minute past the plan allowance, with models billed separately - Recording: Through the separate Egress service, written to the customer's own S3, GCS or Azure bucket. Audio-only transcode from $0.005 a minute past the allowance on Ship - Transfer: `TransferSIPParticipant` sends SIP REFER for a cold transfer. Agent-assisted warm transfer runs through an Agents task marked beta, in Python only - Free tier: Build plan, $0, no card. 1 US local number, 50 inbound minutes and 1,000 third-party SIP minutes a month, as a hard cap - Rate limits: 1,000 Server API requests a minute per project. 100 local numbers on Ship and 300 on Scale. Outbound ringing timeout capped at 80 seconds - SLA: 99.99% uptime listed for every plan as an objective. The terms give no credits outside a written Enterprise agreement - Data retention: Telemetry up to 60 days by default. Call detail records kept as long as law or carrier rules require, with no fixed period stated - Regions: Global SIP endpoint by default. Region-pinned endpoints for aus, eu, india, japan, sa, uk and us. The pricing page lists region pinning for Scale and Enterprise - MCP server: Docs only, at https://docs.livekit.io/mcp. No MCP server for call control was found - Prices: Third-party SIP minutes, Ship plan $0.004 per minute of call; Third-party SIP minutes, Scale plan $0.003 per minute of call; US local inbound minutes $0.01 per minute of call; US toll-free inbound minutes $0.02 per minute of call; US local number $1 per month (plan); US toll-free number $2 per month (plan); Ship plan $50 per month (plan); Scale plan $500 per month (plan) - Scores: Reliability 65, Performance pending, Schema & documentation 84, Agent ergonomics 66, Security & auth 61, Payments & pricing 40, Task success pending, Maintenance & community 90, Transparency & trust 75 · total over the 7 assessed categories - Why: Reliability, Graded as a hosted service, LiveKit Cloud. · Schema & documentation, The contract is protobuf in livekit/protocol (17 SIP RPCs, 6 phone number RPCs), called as JSON over Twirp. · Agent ergonomics, List calls take a limit and ID, number or status filters, with no field selection (15 of 25). · Security & auth, A project key and secret sign JWTs whose `sip` grant is `admin` or `call`, with a lifetime the signer sets. · Payments & pricing, No x402, MPP or L402 found (0). · Maintenance & community, livekit-cli v2.19.0 was tagged on 8 October 2026 and livekit/sip v1.18.0 on 5 October (30). · Transparency & trust, The SIP service, protocol, CLI and SDKs are Apache 2.0, and the security page says the same code runs on LiveKit Cloud. - Sources: 37, open questions: 10, both in the full twin - Capabilities: voice.calls, voice.sip, voice.numbers, voice.transfer, voice.streaming - JSON: https://www.anchorterminal.com/api/v1/tools/livekit-telephony.json - Verify (for the vendor): the badge `https://www.anchorterminal.com/badges/livekit-telephony.svg` or a link to https://www.anchorterminal.com/tools/livekit-telephony from a page on livekit.com or one of its subdomains, or the README of github.com/livekit/sip, then `POST https://www.anchorterminal.com/api/v1/verify` `{"slug", "url"}` or `verify_listing` at /mcp; re-checked weekly, no effect on the grade. Snippets in the full twin. ## Before you call it 1. Ask the operator for a short-lived token with only the `sip.call` grant to dial or transfer. Trunk, dispatch rule and number changes need `sip.admin` 2. Use a third-party outbound trunk for outbound calls. LiveKit Phone Numbers only receive calls 3. Set `wait_until_answered` on `CreateSIPParticipant` and read the SIP status on `SipCallError` before retrying. Voicemail answers with 200 OK 4. Send every request as a JSON POST to `/twirp/livekit.SIP/` or `/twirp/livekit.PhoneNumberService/` on the project's own host, with `{}` for an empty body 5. A released number is billed for the whole month, and the free Build allowance is a hard cap after which requests fail ## Connect ```bash brew install livekit-cli ``` ```bash curl -X POST https:///twirp/livekit.SIP/ListSIPInboundTrunk \ -H "Authorization: Bearer " \ -H 'Content-Type: application/json' \ -d '{}' ``` Full config and headless snippets are in the full page. Through letme (picks today, calling later): https://letme.dev/livekit-telephony ## Similar tools | Tool | Grade | Score | Shared capabilities | Slim | | --- | --- | --- | --- | --- | | Twilio Programmable Voice API + MCP | A | 80.4 | voice.calls, voice.numbers, voice.streaming, voice.sip, voice.transfer | https://www.anchorterminal.com/tools/twilio-voice.min.md | | Telnyx Voice API + MCP | BB | 74.2 | voice.calls, voice.numbers, voice.streaming, voice.sip, voice.transfer | https://www.anchorterminal.com/tools/telnyx-voice.min.md | | SignalWire Voice API | B | 67.1 | voice.calls, voice.numbers, voice.streaming, voice.sip, voice.transfer | https://www.anchorterminal.com/tools/signalwire-voice.min.md | | Bandwidth Voice API + MCP | B | 63.7 | voice.calls, voice.numbers, voice.streaming, voice.sip, voice.transfer | https://www.anchorterminal.com/tools/bandwidth-voice.min.md | | Sinch Voice API + MCP | B | 62.7 | voice.calls, voice.numbers, voice.streaming, voice.sip, voice.transfer | https://www.anchorterminal.com/tools/sinch-voice.min.md | ## Panel reviews (0, desk reviews from public material, no calls made)