Inbound LiveKit SIP trunks receive calls from a provider, but a separate dispatch rule decides which room the caller joins. Outbound calls start in LiveKit and use either a reusable outbound trunk or configuration supplied with each call. Create trunks and dispatch rules for reuse—not once per call—unless outbound settings genuinely need to vary per call.
What’s the difference between inbound and outbound SIP trunks?
The distinction is call direction and responsibility. An inbound trunk handles calls traveling from a SIP provider or the public telephone network (PSTN) into LiveKit. A dispatch rule then matches the incoming call and routes its SIP participant to a room. An outbound trunk holds configuration for calls traveling from LiveKit to a provider or PSTN; a CreateSIPParticipant request initiates the call.
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| Configuration question | Inbound | Outbound |
|---|---|---|
| Call direction | Provider/PSTN to LiveKit | LiveKit to provider/PSTN |
| Main LiveKit configuration | Inbound trunk | Stored outbound trunk or inline outbound configuration |
| What directs or starts the call? | A dispatch rule selects the room; LiveKit can create the room if needed. | A CreateSIPParticipant request starts the call. |
| Typical persistence | Create and reuse the trunk and dispatch rule. | Store and reuse a trunk, or provide settings inline when they vary by call. |
How does inbound trunk configuration work?
Use the trunk to admit and describe incoming calls
For a third-party SIP provider, configure the provider side and then create a LiveKit inbound trunk associated with the provider number or numbers. The inbound trunk can define allowed-number restrictions and other incoming-call properties. LiveKit describes trunks as long-lived, cached configuration objects and recommends reusing them; its documented pattern is typically one inbound trunk per phone number. LiveKit’s inbound trunk documentation explains the setup and provider considerations.
Use a dispatch rule to choose the room
The inbound trunk does not by itself determine the room for a caller. A dispatch rule matches the call and controls how LiveKit adds the caller as a SIP participant. Depending on the rule type and configuration, a call can join an existing room, a separate room for each caller, or a room selected using the called destination. Rules are reusable objects too: create the needed routing rules once and apply them to incoming calls rather than making a rule per call. See LiveKit’s dispatch-rule documentation.
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A rule without trunk_ids can match calls from all inbound trunks. Rules can also constrain caller numbers, hide phone numbers, set metadata and attributes, and specify room configuration. The exact field names and supported options are documented in the SIP API reference.
Do I need to create a new LiveKit trunk for every call?
No. LiveKit recommends creating trunks once and reusing them. Its documentation warns that creating trunks for each call bypasses caching and can degrade reliability at scale. The same reuse principle applies to dispatch rules: they are intended to be long-lived routing configuration, not per-call objects.
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Separate configuration only where the underlying integration calls for it. For example, LiveKit’s inbound guidance describes one trunk per phone number as a typical pattern. For outbound calls where only the caller ID changes, set sip_number on the participant request rather than creating a separate trunk for each caller ID.
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Both approaches are supported. A stored outbound trunk contains provider connection details and authorization data, and a call can reference that reusable configuration. Alternatively, provide outbound trunk configuration inline in each CreateSIPParticipant request. Inline settings are useful for a quick setup or when requirements differ per call, such as a multi-tenant service using different providers or credentials. LiveKit says a stored outbound trunk is not required. For the supported patterns and provider examples, see the telephony introduction.
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The outbound configuration needs the right provider endpoint and authorization details. The SIP API reference lists fields such as provider address, destination country, caller ID numbers, transport, optional authentication, and SIP headers. Its address field takes a hostname or IP address, not a value prefixed with sip:. Check the current reference for exact request fields, and ensure your SIP API requests have the required sip:admin permission unless an operation says otherwise.
Can the same provider trunk support both directions?
Yes. LiveKit documents that one SIP provider trunk can be associated with both an inbound and an outbound LiveKit trunk object. That shared provider relationship does not combine the LiveKit roles: inbound still receives calls and uses dispatch rules for room routing, while outbound still uses a trunk configuration and a participant request to place calls.
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What should I check when a call fails?
- For inbound calls, verify the path end to end. Check the provider trunk and phone number, the LiveKit inbound trunk, and that at least one dispatch rule matches the call.
- For outbound calls, confirm which configuration the request uses. Verify the stored outbound trunk reference or the inline settings, including the provider endpoint and authorization details.
- Place a test call and inspect what LiveKit created. Confirm that the expected room and SIP participant appear, then inspect logs to narrow down a routing, provider, or authorization problem. Follow LiveKit’s telephony testing guidance.
- Confirm provider authentication compatibility. Supported methods vary by provider. LiveKit’s inbound documentation says, for example, that Twilio Elastic SIP Trunking does not support inbound username/password authentication, while TwiML can use username/password authentication. Check current provider guidance before relying on a method.
- For geographically constrained deployments, review region behavior. LiveKit says its default inbound endpoint routes toward the region closest to call origination and that region pinning can restrict the region. Check the current inbound documentation and local telephony requirements for your deployment.
How do trunk settings and media defaults interact?
LiveKit’s SIP API reference describes configuration precedence separately for each direction: inbound calls resolve media settings from the dispatch rule, then the inbound trunk, then defaults; outbound calls resolve them from the CreateSIPParticipant request, then the outbound trunk, then defaults. Encryption and media timeout fall through independently, while codecs are inherited as a pair. This matters when the call connects but uses unexpected media settings: check the highest-precedence configuration first, then the next applicable level.
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