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Telephony on the Telnyx LiveKit platform is one vendor, one portal. Buy a number, connect it to your agent, and start receiving calls. On other platforms, you pay third-party SIP fees on top of your usage. On Telnyx, those are gone entirely.

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Connect a phone number to your agent

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1. Buy a phone number

  1. Log in to the Telnyx Portal
  2. Go to Real Time Communications → Numbers → Buy Numbers
  3. Purchase a number in your preferred country
You can also manage numbers via the Telnyx API.

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2. Create a SIP connection

  1. Go to Real Time Communications → Voice → SIP Connections
  2. Create a new connection with type FQDN
  3. Point it at your region’s SIP endpoint:
RegionSIP Endpoint
New Yorknyc1.sip.livekit-telnyx.com
San Franciscosfo3.sip.livekit-telnyx.com
Atlantaatl1.sip.livekit-telnyx.com
Sydneysyd1.sip.livekit-telnyx.com
  1. Assign your phone number to this SIP connection

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3. Create an inbound trunk

lk sip inbound create \
  --name "telnyx-inbound" \
  --numbers "+15551234567"

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4. Create a dispatch rule

Map the phone number to your agent: 
lk sip dispatch create \
  --name "route-to-my-agent" \
  --trunks "<trunk-id-from-step-3>" \
  --individual "my-agent"
Now when someone calls your number, the platform dispatches the call to your agent.

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What’s different from LiveKit Cloud

The setup steps are similar — you still configure SIP and dispatch rules. The difference is it’s one vendor, one account, no third-party SIP fees. On other platforms, you pay third-party SIP fees on top of your usage. On Telnyx, those are gone entirely — your phone numbers, SIP connections, and agent deployments are all in the same portal under the same bill.

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Outbound calls

Use CreateSIPParticipant to place outbound calls from your agent: 
from livekit.api import LiveKitAPI

api = LiveKitAPI()
await api.sip.create_sip_participant(
    room_name="my-room",
    sip_trunk_id="your-trunk-id",
    sip_call_to="+15551234567",
    participant_identity="outbound-caller",
)

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Features

  • HD voice — G.722 codec support for higher quality audio
  • DTMF — Supported via RFC 2833/4733
  • Cold transfer — REFER-based transfer to another number or agent
  • Warm transfer — Agent-assisted handoff with context