What is LiveKit SIP Integration?
LiveKit SIP integration connects the LiveKit real-time media infrastructure to standard telephone networks using Session Initiation Protocol (SIP). It allows AI voice agents, WebRTC applications, and media servers built on LiveKit to place and receive real phone calls through SIP trunks from carriers like Twilio, Vonage, Bandwidth, or self-hosted OpenSIPS/Kamailio. CelloIP Technologies provides end-to-end LiveKit SIP integration connecting your AI voice agent to any telephony network.
How to Connect LiveKit to a SIP Trunk
LiveKit SIP trunk configuration requires: (1) provisioning a SIP trunk with an inbound DID number, (2) configuring LiveKit SIP participant rules to dispatch incoming calls to an Agent worker, (3) setting outbound SIP transport credentials for the LiveKit SIP bridge, and (4) testing end-to-end audio quality with rtpengine or SRTP. CelloIP handles the full SIP trunk setup and LiveKit integration in 1–2 weeks.
LiveKit SIP vs Asterisk vs FreeSWITCH for AI Agents
LiveKit SIP is purpose-built for AI agent pipelines with minimal latency. Asterisk and FreeSWITCH provide richer telephony feature sets (IVR, conferencing, call recording, CDR) but require more configuration. CelloIP's preferred architecture uses LiveKit Agents for the AI pipeline and Asterisk or FreeSWITCH as the SIP gateway — combining the strengths of both. This hybrid approach handles complex PBX routing while keeping AI agent latency below 800ms.
CelloIP LiveKit SIP Integration Services
CelloIP delivers: LiveKit SIP trunk provisioning and configuration, Asterisk/FreeSWITCH to LiveKit bridge development, carrier SIP trunk validation and number porting, SRTP/DTLS media encryption, SIP load balancer setup with OpenSIPS or Kamailio, and full end-to-end QA testing. Services start from $25/hr with dedicated engineers available in 48 hours.
LiveKit SIP Integration: Connect Your VoIP Infrastructure
CelloIP provides expert LiveKit SIP trunk integration. Connect Asterisk, FreeSWITCH, Kamailio, OpenSIPS, and any RFC-3261 SIP carrier (Twilio, Telnyx, Vonage) directly to LiveKit rooms. Support for inbound and outbound PSTN calls, dispatch rules to route calls by DID, DTMF pass-through, SIP REFER call transfers, TLS/SRTP security, and Prometheus metrics for production monitoring.
Connect Your VoIP Infrastructure to LiveKit
Bridge Asterisk, FreeSWITCH, Kamailio, OpenSIPS — and any SIP carrier — with LiveKit AI rooms. Handle inbound/outbound PSTN calls through your existing telephony stack.
How LiveKit SIP Works
SIP Trunks
LiveKit creates inbound/outbound SIP trunks via REST API. Any RFC-3261 SIP carrier can connect.
Dispatch Rules
Rules decide which LiveKit room a call joins. Match by DID number, caller ID, or custom SIP headers.
SIP Participants
Each phone call appears as a participant in the LiveKit room — audio tracks exchanged bidirectionally.
DTMF & Transfer
Full DTMF pass-through, SIP REFER for call transfer, hold/resume support.
Integration Compatibility Matrix
| Platform | Integration Method | Complexity | Inbound | Outbound | Notes |
|---|---|---|---|---|---|
| Twilio Elastic SIP | Direct SIP trunk | Easy | ✓ | ✓ | Best for cloud-first setups |
| Telnyx | Direct SIP trunk | Easy | ✓ | ✓ | Lower cost than Twilio |
| Vonage | Direct SIP trunk | Easy | ✓ | ✓ | Good EU coverage |
| Asterisk 20/21 | chan_pjsip SIP trunk | Medium | ✓ | ✓ | Route specific DIDs to LiveKit |
| FreeSWITCH | mod_sofia SIP gateway | Medium | ✓ | ✓ | ESL event integration |
| Kamailio | dispatcher module | Advanced | ✓ | ✓ | LCR + failover before LiveKit |
| OpenSIPS | SIP proxy routing | Advanced | ✓ | ✓ | Multi-tenant routing logic |
| Generic SIP PBX | SIP trunk | Easy-Med | ✓ | ✓ | Any RFC-3261 compatible |
Integration Architectures
Asterisk + LiveKit Architecture
Asterisk receives PSTN call on DID → Dialplan routes to SIP trunk pointing at LiveKit SIP service → LiveKit dispatches to room based on dispatch rule → Voice Agent joins and greets caller → Agent can transfer back to Asterisk extension via SIP REFER.
FreeSWITCH + LiveKit Integration
FreeSWITCH mod_sofia receives call → ESL event fires to Node.js controller → Controller decides if AI agent needed → bridges to LiveKit via SIP → ESL monitors for disconnect → Full recording via LiveKit egress.
Kamailio/OpenSIPS Advanced Routing
All DIDs hit Kamailio (fraud detection, rate limiting) → Dispatcher routes AI-destined calls to LiveKit SIP, human-destined to Asterisk/FreeSWITCH → LiveKit agent answers, optionally transfers back via SIP REFER.
LiveKit SIP Configuration
// Create inbound SIP trunk via LiveKit API
const sipTrunk = await sipClient.createSIPInboundTrunk({
name: 'Asterisk-to-LiveKit',
numbers: ['+15551234567', '+15559876543'],
allowedAddresses: ['203.0.113.10'], // Asterisk server IP
authUsername: 'livekit-sip',
authPassword: 'secure-password-here',
});
// Create dispatch rule: route calls to AI agent room
const rule = await sipClient.createSIPDispatchRule({
trunkIds: [sipTrunk.sipTrunkId],
rule: {
dispatchRuleIndividual: {
roomPrefix: 'ai-agent-',
},
},
});
// Enable TLS and SRTP for production security
const trunk = await sipClient.updateSIPInboundTrunk(sipTrunk.sipTrunkId, {
tlsCertificateRequest: true,
enableSRTP: true,
});Security & Production Considerations
IP Allowlisting
Restrict SIP signalling to known carrier/PBX IPs only.
SIP Authentication
Digest auth username + password per trunk required.
TLS/SRTP
Enforce encrypted media and signalling in production.
Rate Limiting
Dispatch rules with max concurrent calls per trunk.
Monitoring
LiveKit webhooks + Prometheus metrics for observability.
Frequently Asked Questions
How does LiveKit SIP receive calls from Asterisk?
Configure chan_pjsip trunk pointing to LiveKit SIP service IP:5060. Set dispatch rule for room routing based on DID. Asterisk will route matching calls to the SIP trunk.
Can LiveKit transfer calls back to Asterisk?
Yes, via SIP REFER. The agent calls sip_client.transfer_sip_participant() pointing to the Asterisk extension. The caller is transferred seamlessly back to the PBX.
Does LiveKit SIP support SRTP encryption?
Yes. TLS signalling and SRTP media encryption are enforced in production trunks. Configure enableSRTP: true when creating or updating trunks.
What SIP carriers work with LiveKit?
Any RFC-3261 compliant carrier. Tested: Twilio Elastic SIP, Telnyx, Vonage, plus self-hosted Kamailio and OpenSIPS proxies.
Can different DIDs route to different AI agents?
Yes. Dispatch rules match by DID number, directing each to a different room prefix and agent configuration. One trunk can support many DIDs with independent routing.
Integration Use Cases
Asterisk PBX Enhancement
Bridge existing Asterisk PBX to LiveKit rooms for AI-powered IVR, voice transcription, and lead qualification.
Contact Center
Route incoming calls from SIP carriers through LiveKit for AI pre-screening, skill-based routing, and agent assist.
Carrier Integration
Connect Twilio, Telnyx, or Vonage SIP trunks directly to LiveKit for PSTN-to-AI-room bridges.
Multi-Tenant SIP Proxy
Deploy Kamailio/OpenSIPS with intelligent routing: send some calls to LiveKit AI, others to Asterisk extensions.
Ready to Bridge Your VoIP Stack?
CelloIP's SIP integration experts will connect your Asterisk, FreeSWITCH, or SIP carrier to LiveKit in days, not weeks.
Questions? Email our team — we respond within 24 hours. Developer FAQs & Pricing