Custom telephony and SIP trunking

Tried answers and places calls over SIP through LiveKit SIP. Any carrier or PBX that can send SIP to a hostname can bring its numbers: Twilio Elastic SIP Trunking, Telnyx, your own carrier, or an on-premise PBX. You keep the number with your carrier; Tried only needs to receive its calls and to send outbound calls back through your trunk.

There are three ways to connect:

The two directions#

Inbound (origination). Your carrier sends calls for the number to Tried. Point the number's route at:

sip:<number>@<LIVEKIT_SIP_URI>

GET /sip-origination returns the values for your deployment: sip_host, origination_uri (sip:<host>;transport=tcp) and ip_ranges, the address ranges our SIP traffic comes from, for your trunk's allow list.

Outbound (termination). Tried sends calls to your carrier's termination URI, authenticated with a SIP username and password if your carrier needs them. The caller ID is the number itself.

Import a number#

Import with the API#

curl -X POST https://api.tried.com.au/import-phone-number \
  -H "Authorization: Bearer $TRIED_API_KEY" \
  -H "Content-Type: application/json" \
  -d '{
    "phone_number": "+61290001234",
    "termination_uri": "sip.carrier.example.com",
    "sip_trunk_auth_username": "tried",
    "sip_trunk_auth_password": "a-strong-password",
    "transport": "TLS",
    "media_encryption": "required",
    "allowed_inbound_addresses": ["203.0.113.10/32"],
    "inbound_agents": [{ "agent_id": "agent_8e0c...", "weight": 1 }]
  }'

The response is HTTP 201 with the number. Imported numbers have provider: "custom" (or "twilio").

What Tried provisions#

Importing, buying, updating or deleting a number creates, updates or deletes its LiveKit objects through the LiveKit server SDK. You never run lk sip commands.

The number records the IDs in livekit_inbound_trunk_id, livekit_outbound_trunk_id and livekit_dispatch_rule_id, and the result in provisioning_status:

POST /provision-phone-number/:n re-applies everything. Changing termination_uri, credentials, transport, media_encryption, allowed_inbound_addresses or sip_trunk_id with PATCH /update-phone-number/:n replaces the trunks in place.

Shared SIP trunks#

When many numbers use the same carrier trunk, create one shared trunk and attach the numbers to it. One inbound trunk, one outbound trunk and one dispatch rule serve all of them. This is API only; the dashboard imports one number at a time.

curl -X POST https://api.tried.com.au/create-sip-trunk \
  -H "Authorization: Bearer $TRIED_API_KEY" \
  -H "Content-Type: application/json" \
  -d '{
    "name": "Head office PBX",
    "termination_uri": "pbx.example.com:5061",
    "transport": "TLS",
    "media_encryption": "allowed",
    "allowed_inbound_addresses": ["203.0.113.10/32"],
    "inbound_auth_username": "pbx",
    "inbound_auth_password": "another-strong-password"
  }'

Then import each number with sip_trunk_id and no termination_uri. The trunk's number list stays in sync. When the last number leaves the trunk, its LiveKit objects are removed, because an inbound trunk with no numbers would catch every call. Other endpoints: GET /list-sip-trunks, GET /get-sip-trunk/:id, PATCH /update-sip-trunk/:id, DELETE /delete-sip-trunk/:id.

Test the SIP connection#

• POST /test-sip-connection with { termination_uri, transport } tests a trunk before you import.
• POST /test-phone-number-connection/:n tests a saved number (its trunk, its shared trunk, or sip.telnyx.com for Telnyx numbers).

Both send one SIP OPTIONS over UDP, TCP or TLS and return { reachable, status_code, reason, latency_ms }. In mock mode the result is simulated (200 OK (mock)) and the dashboard says (simulated, no SIP server configured). The same test is on each number's Routing and SIP card.

Codecs and media#

LiveKit SIP negotiates PCMU, PCMA and G.722. Caller keypad presses arrive as RFC 2833 telephone events, and the agent's press_digit tool sends RFC 2833 events.

Local testing without keys#

scripts/sip-smoke.sh runs livekit-server and livekit-sip locally with the API and the worker, and places real SIP calls with a small SIP client (PCMU and RFC 4733 DTMF). No vendor keys or root access needed.

Related#

• Twilio
• Telnyx
• Contact centre platforms
• SIP security
• Import phone number API