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Prerequisites

Before you begin, ensure you have:
  1. An active phone number on your Wavix account. If you need to purchase one, see our guide on how to buy a phone number.
  2. A SIP trunk (How to create a SIP trunk).

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Receive calls with Wavix and LiveKit

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Create LiveKit inbound trunk

  1. Sign in to your LiveKit project.
  2. In the left-hand side menu, select Telephony → Configuration.
  3. Select Create new and choose Trunk.
  4. Enter a comma‑separated list of Wavix numbers to associate with this trunk.
Configure LiveKit telephony screenshot
For instructions on creating a LiveKit inbound trunk using the CLI or code, see the LiveKit documentation.

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Create a dispatch rule

Dispatch rules determine how inbound calls are routed to LiveKit rooms.
  1. Click Create new and select Dispatch rule.
  2. (Optional) Under Match trunks, associate the rule with a specific trunk; otherwise, it applies to all inbound calls in the project.
LiveKit dispatch rule screenshot
For more information about dispatch rules and instructions for creating them using the LiveKit CLI or code, see the LiveKit documentation.

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Set up inbound call routing on Wavix

Wavix can route inbound calls to a SIP trunk on the platform, SIP URI, or forward them to a phone number. LiveKit requires calls to be routed to a SIP URI.
  1. In your LiveKit project, go to Settings → Project.
  2. Copy the value from SIP URI (e.g., sip:2mpbww13htk.sip.livekit.cloud).
LiveKit SIP URI screenshot To route calls from your Wavix number to the LiveKit platform:
  1. In Wavix, open Numbers & trunks → My numbers.
  2. Select your number by clicking the ⋯ menu → Edit number.
  3. Set the inbound call destination type to SIP URI, and enter the destination in the format of:
[did]@[LIVEKIT_SIP_URI]
In the example above, it would be: 
[did]@2mpbww13htk.sip.livekit.cloud
  1. Save your changes.
Make sure you entered the correct LiveKit SIP URI.
Now, all calls to your Wavix number are routed to your LiveKit AI project. Deploy a LiveKit agent to answer the calls. Check the LiveKit recipes for voice AI and LiveKit Agent use cases.
You can choose to encrypt call media. To enable encryption on your Wavix numbers, contact support@wavix.com and provide a list of numbers.

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Place calls with Wavix and LiveKit

To place outbound calls, you need both a Wavix SIP trunk and a LiveKit outbound trunk.

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Create an outbound trunk on LiveKit

  1. Sign in to your LiveKit account.
  2. Go to Telephony → Configuration.
  3. Select Create new and replace the following variables:
    • Replace YOUR_SIP_TRUNK_ID and YOUR_SIP_TRUNK_PASSWORD with the SIP trunk credentials configured at Wavix.
    • Replace 61290597521 with the phone number purchased at Wavix.
LiveKit new trunk screenshot
For instructions on creating a LiveKit outbound trunk using the CLI or code, see the LiveKit documentation.
You can encrypt call media on your Wavix trunk. Contact support@wavix.com and provide your SIP trunk ID.
You’ve connected LiveKit with Wavix and now can place outbound calls. Check the LiveKit recipes for voice AI and LiveKit Agent use cases.

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Transfer a call

Wavix supports using the SIP REFER command. To transfer a call, you need two Wavix numbers, one for an active call and one to receive the transferred call. Make sure inbound call routing is set up on the second number. To transfer an active LiveKit call, use the TransferSIPParticipant server API: 
import { SipClient } from 'livekit-server-sdk';
// ...
async function transferParticipant(participant) {
  console.log("transfer participant initiated");
  const sipTransferOptions = {
    playDialtone: false
  };

  const sipClient = new SipClient(process.env.LIVEKIT_URL,
                                  process.env.LIVEKIT_API_KEY,
                                  process.env.LIVEKIT_API_SECRET);

  const transferTo = "sip:+<YOUR_WAVIX_NUMBER>@<WAVIX_SIP_GATEWAY>"; // e.g., sip:+61290597520@au.wavix.net

  await sipClient.transferSipParticipant('open-room', participant.identity,
                                          transferTo, sipTransferOptions);
  console.log('transfer participant');
}
Replace transfer_destination with your Wavix number. Use the following format: sip:+[YOUR_WAVIX_NUMBER]@[WAVIX_SIP_GATEWAY] . Example: sip:+61290597520@au.wavix.net
Choose the primary Wavix gateway with the lowest ping from your location. A list of regional gateways is shown at the bottom of the Trunks page in the Wavix portal.
Learn more about LiveKit call transfer in the LiveKit documentation.

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Troubleshooting

  • 603 Declined can occur if the destination’s per‑minute rate is higher than your account’s Max call rate (check the value on the Trunks page). Contact support@wavix.com to request a change if needed.
  • 603 Declined can also occur if there is no active registration on your SIP trunk. Verify registration status as described above.
  • Wrong number format. Dial the full international E.164 number, e.g. 19085594899 (US) or 4408001218915 (UK). Do not dial local formats like 9085594899. Strip prefixes like 0, 00, or 011 before the dialed number.