Voice over IP (VOIP), SIP, Security, 5G and IoT Certification Course

Did you know that providers now centralize secure SIP trunking for enterprises using carrier-grade SBCs, cut telephony costs by about 40% while handling tens of thousands of daily calls over SIP, and invest more than 22,000 man‑days integrating VoLTE across IMS, RAN, and core before carrying around 170 million minutes in the first year?

Course Overview

The Voice over IP (VOIP), SIP, Security, 5G and IoT Certification Course by Rcademy is designed to equip VoIP engineers, telephony specialists, SIP/H.323 protocol engineers, security architects, network designers, telecommunications professionals, IoT platform engineers, and system integrators with comprehensive understanding of VoIP architecture, protocols, and call signaling procedures for unified communications and next-generation voice services. Participants gain expert knowledge of Session Initiation Protocol (SIP), H.323, MGCP, VoIP media transmission technologies, security threats and encryption mechanisms, and integration with LTE networks (VoLTE), Wi-Fi networks (VoWiFi), and 5G NR for seamless multimedia communications.​

Without specialized VoIP and SIP training, professionals may struggle to decode SIP call flows, configure Session Border Controllers for security, optimize voice quality using QoS mechanisms, integrate VoLTE with IMS networks, or prevent toll fraud attacks costing up to $300,000 per incident, limiting their ability to support enterprise UC deployments and carrier voice services. This comprehensive course provides a structured path to mastery across VoIP fundamentals, SIP signaling and call flows, RTP media transmission, voice codecs and compression, security architecture and fraud prevention, VoLTE/VoWiFi integration, and systematic protocol analysis and troubleshooting, preparing attendees to lead VoIP deployment and optimization initiatives.​

Real-world cases show how Axtel deployed AudioCodes Multi-Service Business Routers at customer premises to provide centralized SIP trunking, centralize traffic for different area codes into single SIP trunks, and deliver high-availability secure VoIP services with wide interoperability.

Studies also show that Kool Smiles deployed Ingate Systems’ secure SIP trunking solution to replace legacy PSTN connections, cutting phone costs by 40% while handling 35,000 calls per day across 1,800 handsets with up to 200 concurrent trunks during peak periods and maintaining HIPAA-suitable security.

Take charge of your VoIP expertise. Enroll now in the Rcademy Voice over IP (VOIP), SIP, Security, 5G and IoT Certification Course to master the protocol analysis and security skills that drive reliable voice communications.

Who Should Attend?

The Voice over IP (VOIP), SIP, Security, 5G and IoT Certification Course by Rcademy is ideal for:

• VoIP engineers and telephony specialists
• SIP/H.323 protocol engineers
• Security architects and security analysts
• Network designers and integration specialists
• Telecommunications professionals
• IoT platform engineers
• System integrators and solutions architects
• IP-PBX deployment engineers
• Session Border Controller administrators
• Unified communications specialists
• Carrier interconnect engineers
• VoLTE/VoWiFi implementation engineers
• Network operations center analysts
• Quality assurance engineers
• Anyone seeking comprehensive VoIP certification

What are the Training Goals?

The main objectives of the Voice over IP (VOIP), SIP, Security, 5G and IoT Certification Course are to enable professionals to:

• Master comprehensive VoIP architecture, protocols, and call signaling procedures for unified communications and next-generation voice services.
• Develop expert-level knowledge of Session Initiation Protocol (SIP), H.323, MGCP, and VoIP media transmission technologies.
• Understand VoIP security threats, encryption mechanisms, and implementing secure VoIP deployments across enterprise and carrier networks.
• Design and optimize voice quality mechanisms: QoS, codecs, RTP, jitter buffers, and end-to-end quality monitoring.
• Integrate VoIP with LTE networks (VoLTE), Wi-Fi networks (VoWiFi), and 5G NR for seamless multimedia communications.
• Apply VoIP and IoT convergence principles for mission-critical communications in smart cities, industrial IoT, and enterprise environments.
• Deploy and troubleshoot VoIP systems using protocol analyzers, performance monitoring tools, and systematic call flow analysis.
• Prepare for advanced professional certifications and roles in VoIP engineering, security, and network integration.

How Will This Training Course Be Presented?

At Rcademy, the extensive focus is laid on the relevance of the training content to the audience. Thus, content is reviewed and customised as per the professional backgrounds of the audience.

The training framework includes:

• Expert-led lectures by senior VoIP and unified communications professionals using audio-visual sessions
• Hands-on exercises with SIP message decoding and protocol analyzer tools
• Interactive workshops for VoLTE call setup signaling trace analysis and RTP stream quality assessment
• Case studies covering Axtel centralized SIP trunking deployment, Kool Smiles secure VoIP migration, and Three UK VoLTE integration
• Practical labs for fraud detection, security vulnerability assessment, and systematic call flow troubleshooting

The theoretical part of training is delivered by an experienced professional from the relevant domain, using audio-visual presentations. This VoIP-focused approach ensures professionals translate theory into practical workflows through SIP call flow analysis, codec selection, QoS configuration, security implementation, and systematic troubleshooting methodologies.​

This comprehensive certification model ensures participants gain both VoIP fundamentals and hands-on proficiency to immediately apply protocol analysis and security expertise in enterprise UC, carrier voice, VoLTE deployment, and troubleshooting roles.​

Register now to experience a rigorous, hands-on learning journey designed to equip you for leading VoIP deployment, security hardening, and protocol optimization projects.

Course Syllabus

Module 1: VoIP Fundamentals and Evolution

• From circuit-switched PSTN to packet-switched VoIP architecture.
• VoIP benefits: cost reduction, mobility, service flexibility, and unified communications.
• VoIP service categories: enterprise VoIP, residential broadband phone, mobile VoIP, carrier VoIP.
• VoIP vs traditional telephony: functional equivalence and service differentiation.
• Regulatory and emergency services (911/112) considerations for VoIP providers.

Module 2: VoIP Architecture and Components

• End-to-end VoIP system architecture: endpoints, gateways, controllers, and service platforms.
• VoIP endpoints: IP phones, softphones, SIP user agents, and mobile devices.
• Media gateways: converting between IP and circuit-switched voice streams.
• Call managers, SIP servers, and application servers orchestrating call setup and control.
• Interaction with legacy PBX, PSTN, and IMS networks for seamless interoperability.

Module 3: Voice Codecs and Media Compression

• Voice digitization: sampling, quantization, and PCM fundamentals (G.711).
• Codec families: G.7xx (narrowband), AMR/AMR-WB (wideband), OPUS, EVS (super-wideband).
• Codec parameters: bit rate, sampling rate, complexity, and mouth-to-ear delay.
• Codec selection for different network conditions and quality objectives.
• Packetization and jitter buffer management for smooth media playback.

Module 4: Real-Time Protocol (RTP) and Media Transmission

• RTP architecture, headers, and payload structures for voice media.
• RTP payload types, dynamic payload assignment, and payload identification.
• RTCP (RTP Control Protocol) for feedback, synchronization, and quality reporting.
• Jitter, latency, packet loss impacts on voice quality and mitigation techniques.
• SRTP (Secure RTP) for encrypted media transmission.

Module 5: Session Initiation Protocol (SIP) Fundamentals

• SIP architecture: user agents, proxies, registrars, and Back-to-Back User Agents (B2BUA).
• SIP methods and responses: REGISTER, INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, UPDATE.
• SIP message format: request/response line, headers, and message body.
• SIP transactions, dialogs, and call legs: lifecycle and state machines.
• SIP extensions for VoIP-specific features: reliability, session progress, preconditions.

Module 6: SIP Call Flows and Session Establishment

• SIP registration procedures: REGISTER method, contact management, subscription refresh.
• Basic SIP call setup: INVITE request, provisional responses, final response, ACK.
• Session Description Protocol (SDP) in SIP: offer/answer model and media negotiation.
• Session modification and re-INVITE procedures for call modifications.
• Call termination and BYE request handling.

Module 7: H.323 Protocol and VoIP Interoperability

• H.323 architecture and components: terminals, gatekeepers, MCUs, gateways.
• H.225.0 signaling for call setup and H.245 for capabilities negotiation.
• H.323 RAS (Registration, Admission, Status) protocol.
• H.323 vs SIP comparison: architecture, signaling complexity, and deployment scenarios.
• SIP-H.323 interworking gateways for legacy system integration.

Module 8: VoIP Quality of Service (QoS)

• QoS requirements for VoIP: latency, jitter, packet loss, and bandwidth.
• VoIP over IP networks: DSCP marking, priority queuing, and traffic shaping.
• IntServ (Integrated Services) and DiffServ (Differentiated Services) for VoIP QoS.
• VoIP adaptation to network conditions: adaptive jitter buffers and FEC (Forward Error Correction).
• Bandwidth provisioning and admission control for VoIP capacity management.

Module 9: VoIP Security Architecture

• VoIP security threats: eavesdropping, SPAM, DoS, protocol exploitation, registration hijacking.
• VoIP security domains: access, signaling, media, and application layer protection.
• Encryption for SIP signaling: TLS (Transport Layer Security) and mutual authentication.
• Media encryption: SRTP (Secure RTP) and end-to-end encryption.
• Authentication mechanisms: digest authentication, certificates, and identity verification.

Module 10: VoIP Fraud and Spam Prevention

• VoIP fraud types: toll fraud, subscription fraud, traffic pumping, and account takeover.
• SPIT (SPAM over Internet Telephony) detection and mitigation.
• Rate limiting, call duration monitoring, and suspicious pattern detection.
• DID (Direct Inward Dialing) security and number validation mechanisms.
• Fraud management systems and analytics for VoIP networks.

Module 11: VoLTE (Voice over LTE) Integration

• VoLTE architecture: IMS integration with LTE/EPS networks.
• VoLTE bearer establishment: default and dedicated bearer procedures.
• VoLTE call flows: registration, originating, and terminating call procedures.
• Quality of Service for VoLTE: guaranteed bit rate and latency management.
• VoLTE supplementary services: call forwarding, conference, call waiting.
• Circuit-Switched Fallback (CSFB) and Single Radio Voice Call Continuity (SRVCC).

Module 12: Voice over WiFi (VoWiFi) and Fixed Line Integration

• VoWiFi architecture and trusted/untrusted WiFi access modes.
• Evolved Packet Data Gateway (ePDG) for secure WiFi access.
• IKEv2 and IPsec tunnel establishment for VoWiFi security.
• VoWiFi call flows and handover between WiFi and LTE.
• Fixed wireless and broadband VoIP service integration.

Module 13: Voice and IoT Convergence in 5G

• 5G NR voice architecture and Vo5G/VoNR procedures.
• Network slicing for voice and multimedia services in 5G.
• IoT communication patterns and voice-triggered IoT commands.
• Mission-critical IoT communications with voice/video integration.
• Emergency services (E911/E112) and public safety broadband integration.

Module 14: VoIP for IoT and Industrial Communications

• Voice notifications and alerts in smart home and industrial IoT environments.
• Push-to-Talk (PTT) over LTE and 5G for mission-critical communications.
• Low-power VoIP clients for IoT devices and wearables.
• Group communication and conferencing in IoT networks.
• Voice-based device control and natural language interfaces in IoT.

Module 15: VoIP Network Management and Monitoring

• VoIP performance metrics: call setup time, call completion rate, call quality indices.
• Voice quality assessment: MOS (Mean Opinion Score) and objective quality metrics.
• Network monitoring and KPI tracking for VoIP networks.
• Provisioning and configuration management for VoIP systems.
• Troubleshooting tools: packet analyzers, real-time protocol analyzers, and monitoring platforms.

Module 16: Signaling Analysis and Troubleshooting

• Protocol analyzer tools for VoIP: Wireshark, NetScout, Acme SIP analyzer.
• Capturing and analyzing SIP call flows end-to-end.
• Decoding SDP offer/answer for media capability negotiation issues.
• RTP stream analysis: codec identification, jitter/latency measurement, packet loss assessment.
• Common VoIP issues: registration failures, call setup timeouts, media problems, quality degradation.

Module 17: Enterprise VoIP Deployment

• Enterprise voice architecture: IP-PBX, UC platforms, and integration with legacy systems.
• Cisco, Avaya, Polycom, and open-source VoIP solutions overview.
• VoIP endpoint provisioning and management.
• Unified communications: presence, IM, conferencing, and team collaboration.
• Network readiness assessment and VoIP deployment planning for enterprises.

Module 18: Carrier VoIP and Service Provider Infrastructure

• Carrier VoIP architecture: softswitches, session border controllers, and interconnection gateways.
• Wholesale VoIP services: origination, termination, and routing management.
• Number resource organization (NRO) and Direct Inward Dialing (DID) management.
• Peering and interconnection models for VoIP carriers.
• Billing and revenue assurance for carrier VoIP services.

Module 19: Regulatory, Compliance, and Emergency Services

• Regulatory requirements: FCC, ACMA, OFCOM, and regional VoIP regulations.
• Emergency services integration: E911/E112 location accuracy and call routing.
• Lawful intercept and government access requirements for VoIP providers.
• CALEA compliance and record-keeping for VoIP communications.
• Number portability and porting procedures for VoIP subscribers.

Module 20: Practical Labs and Case Studies

• Hands-on: SIP REGISTER and INVITE message analysis using Wireshark.
• Hands-on: VoLTE call setup signaling trace analysis.
• Hands-on: VoWiFi session establishment and mobility scenario analysis.
• Hands-on: RTP stream quality analysis and codec identification.
• Hands-on: Fraud detection and security vulnerability assessment in VoIP traces.
• Case studies: commercial VoIP deployments, migration scenarios, and real-world troubleshooting.

Training Impact

The impact of Voice over IP (VOIP), SIP, Security, 5G and IoT training is evident in how providers deliver centralized SIP trunking with secure connectivity, achieve substantial OPEX savings via SIP migrations, and accelerate VoLTE/VoWiFi rollouts using standardized IMS media resources.​

Axtel (Mexico) – Centralized, Secure SIP Trunking for Enterprise Customers

Implementation: Axtel, a leading Mexican provider of information and communication technology solutions for business, government, and residential markets with more than 42,000 km of trunk network, metropolitan rings, and fiber FTTx, sought a cost‑effective SIP trunking solution with wide interoperability and strong security that could be installed at customer premises while centralizing traffic from multiple area codes into a single SIP trunk. The company selected AudioCodes Mediant 500 and Mediant 800 Multi‑Service Business Routers deployed at each customer location to terminate SIP trunks, interconnect with existing PBXs or IP‑PBXs, provide firewall and NAT functionality, and offer survivability options to protect against downtime. The MSBRs’ multi‑core architecture enabled high‑performance data and voice services over converged access links, while integrated SBC capabilities ensured secure and interoperable SIP connectivity across heterogeneous customer environments.​

Results: Axtel successfully launched its new all‑IP service delivering secure, high‑quality VoIP services to enterprise customers throughout Mexico, with MSBRs providing versatility for SIP trunking, hosted PBX, and data cloud services, resilience and survivability to protect against outages, and robust security against malicious attacks and fraud. Different MSBR models allowed scalable deployments tailored to various business sizes and traffic profiles, enabling Axtel to centralize SIP connectivity, simplify customer edge infrastructure, and differentiate its enterprise offering with reliable, secure VoIP. This deployment illustrates how professionals trained in SIP signaling, SBC integration, QoS design, and secure VoIP architectures can design and operate centralized SIP trunking solutions that reduce complexity while maintaining high availability and security.​

Kool Smiles (USA) – 40% Telephony Cost Reduction with Secure SIP Trunking

Implementation: Kool Smiles, a U.S. children’s dental management practice with a nationwide network of clinics, operated a voice network with 1,800 handsets and approximately 35,000 calls per day, initially relying on traditional TDM systems that generated high telephony costs. To support an expanding footprint while controlling OPEX, Kool Smiles adopted SIP trunking and secure VoIP using Ingate Systems’ enterprise session border controller (e‑SBC) SIParator deployed at headquarters, connecting its IP‑PBX to an Internet telephony service provider’s SIP trunks and providing secure traversal of SIP through existing firewalls. The architecture enabled centralization of voice services, support for future Internet‑based applications, and robust security using SRTP and dedicated SIP security functions operating alongside the existing data firewall.​

Results: By shifting more than 35,000 calls per day onto SIP trunks and centralizing voice services, Kool Smiles cut communications costs by 40%, achieving full ROI on the SIP trunking project including a new IP‑PBX, phones, network upgrades, and Ingate’s solution within twelve months. The SIP trunks supported up to 200 concurrent trunks at peak usage, delivering excellent call quality and reliability across all clinics, while the security architecture provided HIPAA‑appropriate protection for medical communications. This case demonstrates how engineers skilled in SIP trunk design, SBC configuration, TLS/SRTP security, and QoS tuning can deliver significant cost savings and compliance‑ready VoIP solutions in healthcare and other regulated environments.

Be inspired by how Axtel and Kool Smiles cut costs and secured high-volume SIP trunking at scale. Join the Rcademy Voice over IP (VOIP), SIP, Security, 5G and IoT Certification Course to gain the signaling, security, and integration skills that drive modern voice networks.