What is LiveKit development?
LiveKit development involves building real-time audio/video and AI voice agent applications using the LiveKit open-source WebRTC infrastructure. CelloIP Technologies builds LiveKit Voice AI agents with GPT-4, Whisper ASR, and ElevenLabs TTS, plus LiveKit-to-SIP integrations connecting AI agents to PSTN telephone networks.
What can you build with LiveKit?
With LiveKit you can build AI voice agents, real-time transcription systems, multi-party video conferencing, SIP-to-WebRTC gateways, and voice-powered applications. CelloIP has delivered 25+ LiveKit Voice AI deployments with sub-1.5s response latency.
How much does LiveKit voice agent development cost?
A LiveKit Voice AI agent project typically costs $20,000–$100,000 depending on LLM complexity, ASR/TTS pipeline, SIP integration, and multi-language support. Dedicated LiveKit developers are available from $25/hr on monthly retainers.
Build Real-Time Voice AI Agents
with LiveKit
CelloIP delivers production-ready LiveKit solutions: voice AI agents, SIP trunk integration, Asterisk/FreeSWITCH bridge, self-hosted deployment, and full-stack WebRTC applications. 12+ years of VoIP expertise meets the world's leading real-time AI platform.
What is LiveKit?
LiveKit is an open-source, production-grade WebRTC infrastructure platform and AI agent framework. It powers real-time voice, video, and data applications at scale — from single-user consultations to 100,000-participant broadcasts.
The LiveKit Agents SDK (Python + Node.js) lets you build AI voice agents that join rooms as participants — receiving audio, processing it through STT → LLM → TTS pipelines, and speaking responses back in real-time. Integrated with SIP trunking, these agents can answer inbound PSTN calls.
CelloIP specialises in connecting LiveKit with your existing VoIP infrastructure — Asterisk, FreeSWITCH, Kamailio, and OpenSIPS — enabling AI capabilities without ripping out your existing telephony stack.
LiveKit Voice Agent Architecture
PSTN / SIP Carrier
Twilio · Telnyx · Your Asterisk/FS PBX
LiveKit SIP Bridge
Inbound + Outbound · Dispatch Rules · DTMF
LiveKit Server Room
WebRTC · SFU · Media routing
Voice Agent (Python)
VAD → Deepgram STT → GPT-4o → ElevenLabs TTS
Application Logic
CRM lookup · Tool calls · Human handoff · Webhooks
LiveKit Development Services
End-to-end LiveKit implementation — from first install to production AI voice agents handling thousands of calls per day.
LiveKit Voice AI Agent Development
Build production-ready voice AI agents using LiveKit Agents SDK. STT → LLM → TTS pipeline with streaming architecture for sub-500ms latency. Integrate with OpenAI GPT-4o, Claude, Gemini, Whisper, Deepgram, ElevenLabs, and Cartesia.
LiveKit Self-Hosted Installation
Complete LiveKit server installation on your infrastructure: Docker Compose single-node, Kubernetes Helm chart production cluster, TURN server setup, TLS/SSL configuration, Redis HA, and ingress/egress services with monitoring.
LiveKit SIP Trunk Integration
Connect LiveKit rooms with PSTN telephony via SIP trunking. Inbound + outbound call routing, DTMF support, dispatch rules, SIP REFER, trunk failover. Compatible with Twilio, Telnyx, Vonage, and custom SIP carriers.
LiveKit + Asterisk/FreeSWITCH Bridge
Seamlessly connect your existing Asterisk PBX or FreeSWITCH softswitch with LiveKit for AI capabilities, WebRTC browser calls, and real-time transcription. Custom SIP bridge with media passthrough, recording, and event hooks.
LiveKit WebRTC Application Development
Custom real-time communication applications using LiveKit JS/React SDK. Video conferencing rooms, broadcast/streaming platforms, virtual classrooms, telehealth calling platforms, and embedded communication widgets.
LiveKit Customisation & Integration
Custom LiveKit plugins, egress recording pipelines, ingress RTMP/WHIP ingest, participant metadata management, room webhooks, and integration with your existing authentication and data infrastructure.
LiveKit Technology Stack
Full coverage across every layer of the LiveKit ecosystem — server, agents, providers, SDKs, and infrastructure.
LiveKit Use Cases CelloIP Delivers
From AI customer service bots to HIPAA telehealth platforms — production-proven LiveKit implementations.
AI Customer Service Voice Bot
Deploy conversational voice agents that handle customer enquiries, qualify leads, schedule appointments, and escalate to human agents — with sub-800ms response latency.
AI-Powered IVR System
Replace touch-tone IVR with natural language understanding. LiveKit agent on inbound SIP calls: Deepgram STT → GPT-4o → ElevenLabs TTS → Asterisk routing action.
Telehealth Video + AI Notes
HIPAA-aware telehealth platform with LiveKit video rooms, real-time transcription, AI clinical note generation, and integration with Epic/HL7 FHIR.
Real-Time Contact Centre AI
Empower agents with real-time AI coaching — live transcription, sentiment analysis, suggested responses, and post-call summaries. Integrates with your existing PBX.
AI Outbound Sales Dialler
Automated outbound campaigns with intelligent voice agents that qualify prospects, leave contextual voicemails, and hand off warm leads to human reps.
Live Broadcast & Streaming
Low-latency live streaming platform with RTMP ingest, LiveKit SFU distribution, recording egress, and viewer chat — for education, fitness, and events.
LiveKit + VoIP Platform Integrations
Connect LiveKit AI capabilities to your existing telephony infrastructure without replacing it.
LiveKit + Asterisk
MediumSIP bridge from Asterisk PBX to LiveKit rooms. chan_pjsip trunk, media passthrough, call recording via LiveKit egress.
Integration detailsLiveKit + FreeSWITCH
MediumFreeSWITCH mod_sofia → LiveKit SIP bridge. ESL event integration for real-time call control and LiveKit participant management.
Integration detailsLiveKit + Kamailio
AdvancedKamailio as SIP proxy routing calls to LiveKit SIP bridge. Load balancing, topology hiding, fraud detection before LiveKit agent pickup.
Integration detailsLiveKit + OpenSIPS
AdvancedOpenSIPS dispatcher module routing inbound SIP traffic to LiveKit SIP service. LCR, failover, and scripted routing logic.
Integration detailsLiveKit + Twilio / Telnyx
EasyDirect SIP trunk configuration with Twilio Elastic SIP Trunking or Telnyx as carrier for LiveKit inbound/outbound calls.
Integration detailsLiveKit + OpenAI Realtime API
EasyLiveKit MultimodalAgent with OpenAI Realtime API for speech-to-speech without separate STT/TTS — ultra-low latency conversational AI.
Integration detailsWhy LiveKit for Voice AI in 2026
Sub-500ms Voice Latency
Streaming STT → LLM → TTS pipeline. Tokens stream from LLM to TTS before full transcription — first audio byte under 500ms in optimal conditions.
Open Source + Self-Hostable
LiveKit server, SIP bridge, and Agents SDK are fully open source (Apache 2.0). Run entirely on your infrastructure with no vendor lock-in. Or use LiveKit Cloud.
WebRTC at Scale
300,000+ developers, 3+ billion calls annually. Kubernetes-native, horizontal scaling. Proven at carrier scale — not just startup demos.
Universal SIP Compatibility
LiveKit SIP bridge works with any RFC-3261-compliant SIP carrier. Tested with Twilio, Telnyx, Vonage, and connects to any Asterisk/FreeSWITCH/Kamailio SIP endpoint.
AI Provider Agnostic
Mix and match any STT, LLM, and TTS provider. OpenAI + Deepgram + ElevenLabs today, Anthropic + AssemblyAI + Cartesia tomorrow — without rewriting your agent.
Production-Grade SDK
Python and Node.js Agent SDK with VAD, turn detection, interruption handling, function tools, multi-modal input, and built-in test framework.
LiveKit Development FAQ
Q:What is LiveKit and what does CelloIP develop on it?
LiveKit is an open-source, scalable WebRTC platform that also provides a framework for building voice and video AI agents. CelloIP develops custom voice AI agents, installs and configures self-hosted LiveKit infrastructure, builds SIP integrations with existing Asterisk/FreeSWITCH systems, and creates real-time communication applications (telehealth, contact centre AI, virtual classrooms) on top of LiveKit.
Q:Can LiveKit connect to our existing Asterisk or FreeSWITCH PBX?
Yes. CelloIP builds SIP bridges that connect your existing Asterisk or FreeSWITCH system to LiveKit. Inbound calls from Asterisk are forwarded as SIP to the LiveKit SIP service, which routes them to LiveKit rooms where AI agents can answer. Outbound calls originate from LiveKit via SIP trunk to Asterisk/FreeSWITCH for PSTN termination. The bridge preserves existing call logic, recording, and CDR.
Q:What is the difference between LiveKit Cloud and self-hosted LiveKit?
LiveKit Cloud is a managed SaaS service — no infrastructure to manage, pay-per-use pricing, built-in global TURN network. Self-hosted LiveKit runs on your servers (bare metal, VMs, or Kubernetes) — full data control, predictable cost at high call volumes, GDPR/HIPAA data residency compliance, and zero per-minute fees. CelloIP implements both. For enterprise clients with data residency requirements, self-hosted Kubernetes deployment is typically recommended.
Q:How does the LiveKit voice agent pipeline work?
A LiveKit voice agent runs as a server-side participant in a LiveKit room. When a user speaks, VAD (Voice Activity Detection) detects the end of turn. The audio is streamed to STT (e.g. Deepgram Nova-2) which returns partial transcripts. Those stream into an LLM (e.g. GPT-4o) which generates a response token-by-token. Each token streams to TTS (e.g. ElevenLabs) which synthesises audio and streams it back into the room. With streaming across all stages, response latency is typically 400–900ms.
Q:Can LiveKit agents handle inbound phone calls from PSTN?
Yes. LiveKit's SIP service receives inbound calls from any SIP trunk provider (Twilio, Telnyx, Vonage, or your own Asterisk/Kamailio). Incoming calls are matched to dispatch rules that route them to specific LiveKit rooms where voice agents are waiting. The agent answers, speaks to the caller, and can route to a human agent, play recordings, collect DTMF, or transfer the call.
Q:What is the typical project timeline for a LiveKit voice agent deployment?
A basic LiveKit voice agent (single intent, fixed dialogue flow, SIP trunk, hosted on LiveKit Cloud) can be deployed in 2–3 weeks. A production-grade system with self-hosted Kubernetes infrastructure, custom STT/LLM/TTS providers, CRM integration, Asterisk bridge, multi-intent NLU, and fallback human handoff typically takes 8–16 weeks depending on complexity.
Explore LiveKit Solutions
LiveKit Voice Agents
Build AI voice agents with STT/LLM/TTS
LiveKit SIP Integration
Connect PSTN & PBX to LiveKit
Asterisk Development
Asterisk PBX & IVR solutions
FreeSWITCH Development
High-scale softswitch platform
AI VoIP Integration
LLM IVR, transcription, analytics
WebRTC Development
Real-time communication platforms
Mobile App Development
iOS & Android voice/video apps
Hire LiveKit Developers
Dedicated LiveKit engineers
Developer FAQs & Pricing
Rates, timelines & engagement models
Flexible Engagement
Need a Dedicated LiveKit Developer?
Scale your engineering team with a vetted LiveKit specialist from CelloIP. NDA Day 1, full IP ownership, onboard in 48 hours. From $25/hr.
Ready to Build Your LiveKit Solution?
Whether you need a voice AI agent, SIP integration with your Asterisk PBX, or a self-hosted enterprise deployment — CelloIP delivers it with 12+ years of VoIP expertise.