SIP setup

The sip target needs two things you bring: a LiveKit server with the SIP service and a SIP trunk with a number. tamesu does not run a carrier of its own.

The LiveKit SIP service

• LiveKit Cloud includes SIP; use your project’s wss:// URL and API key and secret.
• Self-hosted: run livekit-server with Redis and the livekit/sip service pointed at the same Redis and API key. examples/compose.yaml is a complete local setup (with Asterisk standing in for the carrier) that you can adapt.

Put the URL in sip.livekit_url and the keys in secrets: as LIVEKIT_API_KEY and LIVEKIT_API_SECRET.

The trunk

Any SIP trunk works: Twilio Elastic SIP Trunking, Telnyx, Plivo, or your own PBX (Asterisk, FreeSWITCH). Describe it once in tamesu.yaml:

sip:
  livekit_url: ${LIVEKIT_URL}
  inbound_number: "+34910000000"             # a number of the trunk routed to tamesu (inbound tests)
  allowed_numbers: ["+1800555*", "+34*"]     # what tamesu may dial
  trunk:
    address: sip.twilio.com                  # the provider's SIP host (host:port for a PBX)
    numbers: ["+34910000000"]                # the trunk's numbers: outbound caller IDs, inbound numbers
    auth_username: "{{secrets.SIP_USER}}"    # outbound authentication
    auth_password: "{{secrets.SIP_PASSWORD}}"
    transport: auto                          # auto | udp | tcp | tls
    allowed_addresses: ["54.172.60.0/23"]    # who may call the inbound trunk (the provider's IPs); optional
secrets:
  LIVEKIT_API_KEY: env:LIVEKIT_API_KEY
  LIVEKIT_API_SECRET: env:LIVEKIT_API_SECRET
  SIP_USER: env:SIP_USER
  SIP_PASSWORD: op://CI/twilio-trunk/password

Then:

tamesu sip setup

creates, or reuses by name, the outbound trunk tamesu-outbound, the inbound trunk tamesu-inbound and the dispatch rule tamesu-inbound (which routes calls to the trunk’s numbers into the room sip.inbound_room, tamesu-sip-inbound by default), and prints:

sip:
  outbound_trunk_id: ST_abc123
  inbound_trunk_id: ST_def456
  dispatch_rule_id: SDR_789xyz

Paste the ids into tamesu.yaml (only outbound_trunk_id is read at run time; the others document what exists). tamesu sip setup --outbound or --inbound sets up one side.

On the provider’s side, point the trunk’s origination (inbound) at your LiveKit SIP service URI (sip:<your-sip-host>:5060, or the Cloud project’s SIP URI) so calls to inbound_number reach tamesu, and allow termination (outbound) from it with the username and password above.

Checking

tamesu doctor reports sip: livekit_url, the trunk id, inbound_number and the keys for the sip targets your suites declare, and livekit probes sip.livekit_url over HTTP. A first outbound test against your own phone number is the end-to-end check: you should hear the persona speak.

Cost and safety

Real calls cost real money. allowed_numbers is the fence: a to that comes from a variable or secret is refused unless it matches a pattern, and tamesu run exits 2 before dialing. Keep sip.max_concurrent_calls low (the default is 2) and max_call_duration short; the SIP service ends calls that run past it.