Secure trunking | LiveKit Docs

LiveKit SIP supports secure trunking using Transport Layer Security (TLS) to encrypt signaling traffic, and Secure Real-time Transport (SRTP) to encrypt media traffic. Encryption ensures that an Internet Service Provider (ISP) or a eavesdropping attacker (man-in-the-middle) cannot listen in on the conversation.

SIP REFER is not supported when using TLS.

Transferring calls with SIP REFER is not currently supported over TLS.

Configure secure trunking for SIP calls

Setting up secure trunking requires multiple steps and includes enabling SRTP and TLS on your SIP trunking provider side, and enabling media encryption on your LiveKit trunks or on a per-call basis. The following sections provide instructions for enabling secure trunking with Twilio and Telnyx and setting up your LiveKit SIP trunks.

To secure calls you must complete all of the following steps:

Enable secure trunking with your SIP trunking provider.
Update your SIP URIs to use TLS for transport.
Enable media encryption for your LiveKit SIP trunks.

Step 1: Enable secure trunking with your SIP trunking provider

Depending on your SIP trunking provider, you might need to explicitly enable secure trunking. The following instructions provide steps for Twilio and Telnyx. If you're using a different provider, check with them to see if you need to enable secure trunking.

Enable secure trunking with Twilio and Telnyx

The following instructions assume you have already configured trunking with your SIP provider. If you haven't, see the SIP trunk setup quickstart or select your provider-specific instructions from the navigation menu.

Sign in to the Twilio Console.
Select Develop tab → Elastic SIP Trunking → Manage → Trunks.
Select the trunk you want to edit.
On the General Settings page, under Features, enable Secure Trunking.
Save your changes.

Sign in to the Telnyx Portal.
Select Real-Time Communications → Voice → SIP Trunking.
Select the trunk you want to edit.
Select the Inbound tab.
For SIP transport protocol, select TLS.
For Encrypted media, select SRTP.
Save your changes.

Step 2: Update your SIP URIs to use TLS

The following instructions apply to inbound calls for Twilio or Telnyx. For other providers, set the origination URI to your SIP URI with ;transport=tls appended to it. For example, if your SIP URI is:

sip:bwwn08a2m4o.sip.livekit.cloud

Set the origination URI to:

sip:bwwn08a2m4o.sip.livekit.cloud;transport=tls.

You can find your SIP URI on your LiveKit Cloud project settings page.

If your provider doesn't support a SIP URI with URI parameters, you must enable TLS another way:

Enable TLS in the trunk settings (required).
If supported, set the port to 5061, the default port for SIP over TLS.
TLS must be enabled
Changing only the port number without enabling TLS is not enough. Some providers might treat port 5061 as a non-standard port for insecure UDP or TCP traffic.

Check your provider's documentation for exact steps.

For outbound calls, you must set the transport protocol to TLS in the outbound trunk settings in the following section.

Sign in to the Twilio Console.
Select the Develop tab → Elastic SIP Trunking → Manage → Trunks.
Select the trunk you want to edit → Origination.
Update the Origination URI to include ;transport=tls.
Save your changes.

Sign in to the Telnyx Portal.
Select Real-Time Communications → Voice → SIP Trunking.
Select the edit icon for your trunk → Inbound settings.
Select Authentication and routing.
In the FQDN section, select Add FQDN.
Add your SIP domain and port 5061 for TLS and save.
In the Inbound calls routing section, select the option you just added with port 5061.
Save your changes.

Step 3: Enable media encryption for your SIP trunks

Set the media_encryption parameter for your inbound or outbound trunks to either allow or require encryption. Valid values are as follows:

SIP_MEDIA_ENCRYPT_ALLOW: Use media encryption if available.
SIP_MEDIA_ENCRYPT_REQUIRE: Require media encryption.

By default, media encryption is turned off. To see all options, see the API reference.

Create an inbound trunk

Create an inbound trunk with media encryption enabled. To edit a trunk instead, see Edit an existing trunk.

Sign in to your Telephony configuration dashboard.
Select Create new → Trunk.

Select the JSON editor tab and copy and paste the following contents. Replace the phone number with the one purchased from your SIP trunking provider.

{
    "name": "My trunk",
    "numbers": [
      "+15105550100"
    ],
    "krispEnabled": true,
    "mediaEncryption": "SIP_MEDIA_ENCRYPT_ALLOW"
}

Select Create.

Create an outbound trunk

For outbound calls, create an outbound trunk with media encryption enabled and transport protocol set to SIP_TRANSPORT_TLS. All calls made using this trunk use TLS and SRTP.

You can also enable media encryption on a call-by-call basis by setting the media_encryption parameter in the CreateSIPParticipant request. However, you should still enable TLS for calls on the outbound trunk.

Use the following instructions to create a new wildcard outbound trunk with SRTP and TLSenabled. The wildcard allows all calls to be routed to the same trunk. To edit a trunk instead, see Edit an existing trunk.

Sign in to your Telephony configuration dashboard.
Select Create new → Trunk.
Select the JSON editor → select Outbound for Trunk direction.

Copy and paste the following contents. Replace the SIP trunking provider endpoint, and username and password for authentication.

{
"name": "My outbound trunk",
"address": "<sip-trunking-provider-endpoint>",
"transport": "SIP_TRANSPORT_TLS",
"numbers": [
   "*"
],
"authUsername": "<username>",
"authPassword": "<password>",
"mediaEncryption": "SIP_MEDIA_ENCRYPT_ALLOW"
}

Select Create.

Edit an existing trunk

Edit an existing inbound or outbound trunk to enable media encryption using the LiveKit Cloud dashboard.

Sign in to your Telephony configuration dashboard.
Navigate to the Inbound or Outbound section on the page.
Select the more menu (⋮) next to the trunk you want to edit → Configure trunk.
For outbound trunks, for Transport select TLS.
Expand the Optional settings section. Select either Media encryption enabled or Media encryption required.
Select Update.

Enable media encryption per call

You can enable media encryption on a per-call basis by setting the media_encryption parameter in the CreateSIPParticipant request.

SRTP must be enabled

You must also enable SRTP on the SIP trunking provider side. If you haven't already enabled this, see Step 1: Enable secure trunking with your SIP trunking provider.

Create a sip-participant.json file with the following participant details:

{
  "sip_trunk_id": "<your-outbound-trunk-id>",
  "sip_call_to": "<phone-number-to-dial>",
  "room_name": "my-sip-room",
  "participant_identity": "sip-test",
  "participant_name": "Test Caller",
  "krisp_enabled": true,
  "wait_until_answered": true,
  "media_encryption": "SIP_MEDIA_ENCRYPT_ALLOW"
}

Create the SIP Participant using the CLI. After you run this command, a call is made to the <phone-number-to-dial> number from the number configured in your outbound trunk.
```
lk sip participant create sip-participant.json
```