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Expert-Led · Hands-On · Market-Ready

Learn the Skills
Companies Pay For

Professional training in AI Agents, Voice & Video Bots, SIP, VoIP, WebRTC, FreeSWITCH, OpenSIPS, Kamailio, RTPEngine, and Linphone — taught by practitioners with 18+ years of real production experience.

Individual · University · Corporate · Enterprise · Research
Online & On-Site worldwide.

🚀
Market Ready
Every track ends with a deployable project you can show employers or clients — not just a certificate.
💰
Start Earning
We teach you what companies pay for. VoIP engineers and AI agent developers are in high demand globally.
🛠️
Hands-On First
No death-by-slides. Every session has labs, real configs, and something running by the end.
🎓
Expert Trainers
Trainers with 18+ years of production deployments — not educators reading from a textbook.
📡
Latest Curriculum
Updated constantly. Pipecat, MCP, Agentic AI — we teach what's shipping in production today.
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Offline & Secure
Air-gapped lab environments available for defence, government, and classified research training.
Course Tracks

8 Specialised Training Tracks

Deep, practical programmes covering every technology we ship in production. Pick one track or combine them into a complete learning path.

Beginner → Advanced

Agentic AI & LLMs

Build autonomous agents that work for you

  • What are AI agents vs chatbots — real differences
  • Prompt engineering and LLM fundamentals (Claude, GPT-4o, Llama)
  • Building multi-step agentic workflows
  • Tool use, function calling, and MCP (Model Context Protocol)
  • RAG: grounding agents in your business knowledge
  • Deploying agents: cloud, on-premise, and air-gapped
  • Monetising agents — freelance, product, or SaaS
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Intermediate

Voice & Video Bots with Pipecat

Ship production voice agents from scratch

  • Pipecat architecture: STT → LLM → TTS pipeline
  • Speech-to-Text: Deepgram, AssemblyAI, Whisper
  • LLM integration: Claude, GPT-4o, Groq, Ollama (offline)
  • Text-to-Speech: ElevenLabs, Cartesia, Kokoro (local)
  • WebRTC real-time streaming with Daily.co
  • RAG + MCP: connecting agents to CRMs and APIs
  • On-premise offline stack — zero cloud dependency
  • Deploying and monitoring production voice agents
Enquire about this track
Beginner → Advanced

SIP, VoIP & RTP Fundamentals

Master the protocol stack powering real-time comms

  • SIP protocol deep dive: REGISTER, INVITE, BYE, OPTIONS
  • SIP message structure, headers, and call flows
  • RTP / RTCP — media transport and quality metrics
  • Codecs: G.711, G.729, Opus — trade-offs and selection
  • NAT traversal: STUN, TURN, ICE
  • SIP security: TLS, SRTP, DTLS
  • Troubleshooting with Wireshark and Homer SIP capture
  • Designing fault-tolerant VoIP architectures
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Intermediate → Advanced

FreeSWITCH Mastery

From installation to enterprise-grade deployments

  • FreeSWITCH architecture: core, modules, event system
  • Installation and configuration on Linux
  • Dialplan XML: conditions, extensions, applications
  • ESL (Event Socket Library) scripting in Python/Node
  • IVR design, call recording, and voicemail
  • Conference bridges, call queues, and ACD
  • High-availability clustering and failover
  • AI integration: connecting Pipecat to FreeSWITCH via SIP
Enquire about this track
Intermediate → Advanced

OpenSIPS & Kamailio

Build carrier-grade SIP proxy and routing platforms

  • OpenSIPS vs Kamailio — when to use which
  • Routing script fundamentals: request_route, reply_route
  • Load balancing, failover, and dispatcher module
  • Registration and location services at scale
  • OpenSIPS dialplan for LCR (Least Cost Routing)
  • CDR generation and billing integration
  • Security: anti-flood, SIP scanner detection, ACL
  • Kamailio htable, permissions, and custom modules
Enquire about this track
Beginner → Advanced

WebRTC Development

Build real-time voice & video apps for the browser

  • WebRTC API fundamentals: getUserMedia, RTCPeerConnection
  • Signalling servers: WebSocket, SIP-over-WebSocket
  • ICE, STUN, TURN — NAT traversal explained
  • Coturn setup for production-grade TURN servers
  • Janus WebRTC Gateway — VideoRoom, SIPre, AudioBridge
  • mediasoup — building scalable SFU architectures
  • Browser + mobile WebRTC app development
  • WebRTC ↔ SIP gateway bridging
Enquire about this track
Advanced

RTPEngine & Media Proxying

Scale media handling to millions of concurrent sessions

  • RTPEngine architecture and integration with OpenSIPS/Kamailio
  • Media proxying: why, when, and how
  • Transcoding: codec conversion at scale
  • SRTP ↔ RTP bridging for encrypted media
  • Redis-based clustering for high availability
  • Performance tuning for high concurrent sessions
  • DTMF handling and RFC 2833
  • Monitoring RTPEngine with Prometheus + Grafana
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Intermediate

Linphone & Mobile VoIP

Build white-label SIP apps for iOS and Android

  • Linphone SDK architecture: core, Java, Swift wrappers
  • Configuring SIP accounts, codecs, and STUN in SDK
  • Push notifications: APNs (iOS) and FCM (Android)
  • Flexisip server: push-compatible SIP proxy
  • Battery-optimised background SIP in mobile apps
  • White-label customisation: UI, branding, features
  • Video calls, SRTP encryption, and conference in SDK
  • End-to-end encrypted messaging over SIP/LIME
Enquire about this track
Audiences

We Train Everyone

From a solo developer building their first voice agent to an enterprise onboarding 100 engineers — we customise every engagement.

Individual / Freelancer

Upskill, Get Market-Ready, and Start Earning.

Whether you're a developer switching careers, a student building expertise, or a freelancer expanding your service offerings — our training turns skills into income.

  • Job-ready curriculum aligned with what companies actually hire for
  • Practical, project-based — ship something real by the end
  • Flexible scheduling: weekends, evenings, self-paced online
  • 1-on-1 mentor sessions with expert trainers
  • Certificate of completion for LinkedIn and résumés
  • Guidance on freelance positioning and finding clients
Available Tracks
Agentic AI & LLMsVoice & Video BotsSIP & VoIPFreeSWITCHOpenSIPS & KamailioWebRTCRTPEngineLinphone & Mobile VoIP
Delivery
Online LiveOn-SiteHybrid
How We Deliver

Train Your Way — Online or On-Site

Our trainers travel globally for on-site engagements. For distributed teams, live online sessions with real labs work just as well.

Online Live

Instructor-led sessions over Zoom or Google Meet. Scheduled batches or 1-on-1. Recorded for later review.

  • Live Q&A with trainer
  • Screen-share labs
  • Session recordings
  • LMS access

On-Site / Offline

Trainer travels to your office, campus, or facility. Ideal for corporate teams, universities, and enterprise engagements.

  • Your premises, your schedule
  • Hands-on lab setup
  • Team workshops
  • Whiteboard deep dives

Hybrid

Mix of online and in-person. Theory online, hands-on labs on-site. Best of both worlds for distributed teams.

  • Flexible scheduling
  • On-site for key sessions
  • Remote for theory
  • Custom pace
Suggested Learning Path

From Zero to Deployable in One Learning Path

SIP & VoIP Basics
FreeSWITCH / OpenSIPS
WebRTC Dev
Agentic AI & LLMs
Pipecat Voice Agents
Ship & Earn 🚀

Each track is standalone — or combine them into a structured path. We'll help you plan the right sequence.

Ready to Start Learning?

Tell us who you are, what you want to learn, and your preferred format. We'll come back with a curriculum, schedule, and pricing — fast.

Individual enrolments · University partnerships · Corporate bootcamps · Enterprise enablement · Research programmes

✓ Online live & on-site✓ 18+ years of trainer experience✓ Project-based, hands-on labs✓ Worldwide delivery