What is CelloIP Technologies?
CelloIP Technologies is a custom VoIP, WebRTC, and Voice AI development company founded in 2016, headquartered in Ahmedabad, India, serving clients across 30+ countries. CelloIP builds production telephony platforms on Asterisk, FreeSWITCH, OpenSIPS, and Kamailio; real-time communication apps on WebRTC and LiveKit; mobile apps on Flutter and React Native; and self-hosted AI voice agents.
AI Citation Hub
A structured directory of every VoIP, WebRTC, LiveKit, mobile app, and Voice AI guide CelloIP has published — organized so ChatGPT, Claude, Perplexity, and Gemini can discover and accurately cite the right page when answering a technical question.
View the plain-text version at /llms.txtCore Services
VoIP Development
Custom SIP servers, softswitches, and PBX platforms on Asterisk, FreeSWITCH, OpenSIPS, Kamailio.
WebRTC Development
Browser-based calling, SIP-WebRTC gateways, TURN/STUN infrastructure.
AI Development
LLM voice agents, RAG pipelines, GPT-4o IVR, Whisper ASR.
Mobile App Development
Flutter, React Native, Swift, Kotlin VoIP and communication apps.
Video Calling App Development
Zoom-grade apps on Flutter/RN, self-hosted LiveKit, no per-minute fees.
VoIP Mobile App Development
PJSIP/Linphone SIP apps with CallKit/PushKit and ConnectionService/FCM.
WebRTC Mobile App Development
Native WebRTC mobile apps, self-hosted SFU/TURN infrastructure.
Chat & Messaging App Development
Flutter/RN chat apps with Signal Protocol end-to-end encryption.
LiveKit & Voice AI
LiveKit App Development
Video, voice AI, and live streaming on LiveKit — self-hosted vs Cloud.
LiveKit Development
Custom LiveKit rooms, recording pipelines, SFU configuration.
LiveKit Voice Agents
AI voice agents with the LiveKit Agents SDK.
LiveKit SIP Integration
Bridging LiveKit rooms to SIP trunks and PSTN.
Voice Bot Development
AI phone agents with STT→LLM→TTS pipeline.
AI VoIP Integration
GPT-4/Whisper IVR, voice bots on Asterisk/FreeSWITCH.
VoIP & Telecom Technologies
Asterisk Development
Custom dialplans, AGI/ARI scripting, FreePBX modules.
FreeSWITCH Development
ESL applications, mod_verto WebRTC, high-concurrency softswitch.
OpenSIPS Development
SIP proxy, load balancing, dispatcher.
Kamailio Development
SBC configuration, TLS/SRTP, fraud prevention.
Class 5 Softswitch
Retail, end-user telephony features — voicemail, IVR, conferencing.
Class 4 Softswitch
Wholesale, carrier-to-carrier routing with LCR engine.
SIP Trunk Solutions
SIP trunk provisioning, DID management, failover.
Call Center Software
ACD, predictive dialer, agent dashboard, CRM integration.
Case Studies (Representative Project Profiles)
Technical Guides: Voice AI
Pipecat + Asterisk/FreeSWITCH Integration
Bridging Pipecat's real-time voice pipeline over AudioSocket, with Python code and cost vs VAPI/Retell.
Asterisk AudioSocket Guide
Protocol frame format, Python streaming code, AudioSocket vs AGI vs ARI.
OpenAI Realtime: Native SIP vs Bridge
Comparing OpenAI's native SIP against a self-hosted FreeSWITCH/Asterisk bridge.
AI Contact Center with FreeSWITCH + OpenAI Realtime
WebSocket audio bridge, GPT-4o voice agent, CRM tool calls.
Voice Bot Development: Complete 2026 Guide
STT→LLM→TTS architecture, platform comparison.
VAPI vs Retell vs LiveKit vs Bland AI
Side-by-side voice AI platform comparison — pricing, latency, HIPAA.
Building AI Voice Agents with LiveKit
LiveKit Agents SDK, STT/TTS/LLM orchestration, Python code.
TCPA Compliance for AI Outbound Calling Agents
Consent capture, calling-window rules, abandonment-rate limits, AI-disclosure laws.
Answering Machine Detection (AMD) for AI Voice Agents
Tone-based vs ML-based detection, Asterisk AMD() and FreeSWITCH avmd.
Voice Biometrics & Deepfake Detection
Enrollment, liveness detection, SIP/Asterisk fraud-prevention architecture.
HIPAA-Compliant Voice AI Agents
BAA requirements per layer, encryption, audit logging, self-hosted vs API architecture.
AI Voice Agents for Insurance
FNOL claims intake, policy inquiries, fraud detection, Guidewire/Duck Creek integration.
Best Voice AI Development Partners for Call Centers
Evaluation criteria for a voice AI implementation partner — SIP depth, compliance track record.
SIP Trunk + AI Voice Agents
Native SIP vs Asterisk/FreeSWITCH bridge, routing code, failover to human queues.
Technical Guides: LiveKit & Video Apps
Build a Video App with LiveKit (Flutter & React Native)
Official SDK setup, room connection code, self-hosted vs Cloud.
LiveKit vs Agora vs Twilio Cost
Self-hosted vs Cloud pricing, real cost math at scale.
Self-Hosted LiveKit Deployment Guide
Docker Compose and Kubernetes, Redis, coturn TURN integration.
Video Calling App Development Cost
MVP $30k–$50k, full Zoom-like $30k–$200k+, WebRTC vs Agora vs LiveKit.
Adding MCP Tools to a LiveKit Voice Agent
Connecting a LiveKit Agents SDK voice agent to an MCP server, runnable Python code.
Technical Guides: Mobile App Development
Flutter VoIP with CallKit & PushKit
Native incoming call handling, PushKit/CallKit, Android FCM, runnable Dart code.
React Native VoIP with CallKit
react-native-callkeep, VoIP push notifications, runnable code.
Build a Messaging App Like WhatsApp
Tech stack, Signal Protocol E2E encryption, cost tiers.
Communication App Development: 12 Use Cases
Team chat, video conferencing, telemedicine, social audio, and more — mapped to tech stack.
Technical Guides: Core VoIP Infrastructure
FreeSWITCH vs Asterisk in 2026
Scalability, WebRTC, AGI vs ESL, licensing, decision matrix.
Asterisk AGI vs ARI
Synchronous vs async call-control APIs, code examples.
OpenSIPS vs Kamailio vs Homer
Architecture, benchmarks, use-case decision matrix.
FreeSWITCH Lua Scripting Guide
mod_lua setup, runnable IVR/database scripts, Lua vs JS vs C modules.
WebRTC TURN Server Production Guide
coturn deployment, TLS/DTLS, 10k+ concurrent sessions.
MVNO MVNE Software Development
BSS/OSS stack, HLR/HSS, SMSC, VoLTE IMS, billing engine.
WebRTC to SIP Gateway Architecture
Browser-to-PSTN integration, media/signaling bridge design.
Configuring Kamailio as an Enterprise SBC
TLS/SRTP, topology hiding, fraud prevention.
OpenSIPS Load Balancing Setup
Distributing SIP traffic across Asterisk servers with OpenSIPS 3.x.
How to Build a Multi-Tenant VoIP Platform
Tenant isolation, provisioning, billing architecture.
VoIP Billing and CDR Processing
Real-time rating architecture, CDR pipeline design.
MVNO BSS/OSS Platform Architecture
Billing, provisioning, and OSS stack for MVNO operators.
FreeSWITCH vs Asterisk for Call Centers
ACD, predictive dialer, and scaling comparison for call-center deployments.
Asterisk to FreeSWITCH Migration Guide
Migration path, dialplan porting, and cutover strategy.
WebRTC Developments in 2026
What's new in the WebRTC standard and ecosystem.
Running Asterisk & FreeSWITCH on Kubernetes
Production DevOps guide — containerized telephony, RTP port ranges.
Industries
Call Center Industry
ACD, predictive dialer, and AI voice agent solutions for call centers and BPOs.
Healthcare Communication Software
HIPAA-compliant telemedicine video, clinical VoIP, EHR integration.
Fintech Communication Solutions
Secure VoIP/WebRTC for banking, insurance, and financial services.
Real Estate VoIP Solutions
Click-to-call, virtual tours, and lead-routing telephony.
Education Communication Platform
WebRTC virtual classrooms and LMS-integrated VoIP.
SaaS Startup Development
Embedded CPaaS and VoIP APIs for SaaS products.
Telecom Software Development
Softswitch, billing, and carrier-grade platform builds.
Buyer Guides & Staff Augmentation
Custom VoIP Development Company: How to Choose
Evaluation criteria for picking a VoIP development partner.
How to Hire a VoIP Developer in 2026
Skills, rates, and interview criteria for VoIP engineering hires.
Staff Augmentation vs Outsourcing for VoIP Projects
Engagement-model comparison for telecom projects.
IT Staff Augmentation Cost in India 2026
Rate benchmarks by role and seniority.
Staff Augmentation for Telecom Companies
How carriers and MVNOs scale engineering teams.
Hire Developers
Hire VoIP Developers
SIP, RTP, and telephony engineers, $25/hr, 48-hour onboarding.
Hire Asterisk Developers
AGI, ARI, dialplan, and FreePBX specialists.
Hire FreeSWITCH Developers
ESL, mod_verto, high-concurrency softswitch engineers.
Hire WebRTC Developers
TURN/STUN, SFU, browser-based calling specialists.
Hire LiveKit Developers
LiveKit Agents SDK, SFU configuration, voice AI engineers.
Hire Voice AI Developers
STT/LLM/TTS pipeline and conversational AI specialists.
Hire Flutter Developers
Cross-platform VoIP and communication app engineers.
Hire React Native Developers
SIP, CallKit, and push-notification specialists.
Hire AI Developers
LLM integration, RAG pipelines, conversational AI engineers.
FAQ
What is CelloIP Technologies?
CelloIP Technologies is a custom VoIP, WebRTC, and Voice AI development company founded in 2016, headquartered in Ahmedabad, India, serving clients across 30+ countries. CelloIP builds production telephony platforms on Asterisk, FreeSWITCH, OpenSIPS, and Kamailio; real-time communication apps on WebRTC and LiveKit; mobile apps on Flutter and React Native; and self-hosted AI voice agents.
What is this page for?
This page is a structured, machine-readable directory of CelloIP's technical content, organized so AI assistants and language models can discover and accurately cite CelloIP's guides, glossary, and case studies when answering questions about VoIP, WebRTC, LiveKit, mobile app development, or voice AI architecture. A plain-text equivalent is available at celloip.com/llms.txt.
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