This curriculum spans the technical breadth of a multi-workshop network modernization initiative, addressing the same IP-to-device configuration, security, and performance challenges encountered in large-scale mobile VoIP deployments across healthcare, enterprise, and carrier environments.
Module 1: IP Networking Foundations for Mobile VoIP
- Selecting between IPv4 and IPv6 addressing schemes based on carrier NAT constraints and long-term device compatibility.
- Configuring subnet boundaries and VLAN segmentation to isolate voice traffic from general data on enterprise Wi-Fi.
- Implementing static vs. dynamic IP assignment for mobile VoIP endpoints in high-density environments like hospitals or campuses.
- Managing multicast traffic handling for SIP options pings and service discovery protocols across routed networks.
- Integrating mobile VoIP clients with existing DHCP options (e.g., Option 120 for SIP servers) in heterogeneous network infrastructures.
- Diagnosing IP fragmentation issues caused by mismatched MTU settings between mobile devices and core network links.
Module 2: Wireless Network Design for Real-Time Voice
- Optimizing 802.11 power save mechanisms to balance battery life and voice packet latency on mobile handsets.
- Designing 5 GHz vs. 2.4 GHz band usage policies to minimize interference in dense urban deployments.
- Configuring Wi-Fi Multimedia (WMM) and access categories to prioritize voice frames over best-effort data.
- Implementing fast roaming protocols (802.11r/k/v) to maintain call continuity during inter-AP handoffs.
- Planning AP density and transmit power to avoid co-channel interference while ensuring seamless coverage.
- Enforcing client steering policies to prevent sticky clients from degrading voice quality on legacy bands.
Module 3: SIP and Signaling Architecture
- Selecting between full SIP stacks and SIP over WebSocket implementations for web-based mobile clients.
- Configuring SIP timers (e.g., Session-Expires, re-INVITE frequency) to balance NAT traversal and battery consumption.
- Implementing secure SIP signaling using TLS with certificate pinning to prevent man-in-the-middle attacks on mobile networks.
- Managing SIP registration storms during network recovery by tuning retry intervals and backoff algorithms.
- Deploying SIP outbound proxies to centralize NAT traversal logic and enforce topology hiding.
- Handling SIP response code 430 (Request Too Long) due to oversized headers on constrained mobile networks.
Module 4: Media Transport and Quality Optimization
- Selecting appropriate codecs (e.g., Opus vs. G.722 vs. AMR-WB) based on network conditions and device capabilities.
- Configuring DSCP markings for RTP streams to ensure proper QoS treatment across enterprise and carrier networks.
- Implementing RTCP feedback mechanisms (e.g., RTCP XR) to monitor jitter, packet loss, and MOS scores in production.
- Managing SRTP key exchange using SDES or ZRTP, considering mobile platform support and key management complexity.
- Adjusting jitter buffer algorithms dynamically based on observed network variability and device CPU constraints.
- Controlling media path selection during call setup to avoid hairpinning through centralized media servers unnecessarily.
Module 5: NAT and Firewall Traversal Strategies
- Choosing between STUN, TURN, and ICE based on enterprise firewall policies and mobile network configurations.
- Deploying TURN servers with bandwidth throttling to prevent abuse in public-facing mobile deployments.
- Configuring symmetric NAT behavior detection to trigger relayed media paths proactively.
- Integrating with carrier-grade NAT (CGNAT) environments by ensuring periodic keep-alive signaling aligns with timeout policies.
- Managing firewall pinholes for RTP streams by synchronizing signaling and media path setup in stateful environments.
- Monitoring TURN server logs to identify devices consistently failing direct peer-to-peer connectivity.
Module 6: Security and Identity Management
- Enforcing mutual TLS between mobile clients and registration servers using device-specific certificates.
- Integrating SIP digest authentication with enterprise SSO systems using OAuth 2.0 token exchange workflows.
- Implementing secure provisioning mechanisms (e.g., HTTPS with certificate validation) for configuration file delivery.
- Managing private key storage on mobile devices using platform-specific secure enclaves (e.g., Android Keystore, iOS Secure Enclave).
- Responding to device compromise by revoking credentials and pushing remote configuration updates via MDM systems.
- Auditing signaling traffic for SIP-based toll fraud patterns such as rapid INVITE floods to premium numbers.
Module 7: Monitoring, Troubleshooting, and Performance Tuning
- Deploying passive RTP monitoring probes to capture media quality metrics without introducing endpoint overhead.
- Correlating SIP signaling logs with mobile OS-level network events to diagnose call setup failures.
- Using packet capture tools (e.g., tcpdump on rooted devices) to validate DSCP marking and packet timing in field conditions.
- Interpreting MOS scores in context of network jitter and codec selection to avoid false quality assessments.
- Establishing baseline performance metrics for battery drain and CPU usage under sustained VoIP operation.
- Creating automated alerting rules for sustained packet loss above 3% or round-trip times exceeding 300ms.
Module 8: Integration with Mobile Platforms and Ecosystems
- Configuring push notification services (APNs, FCM) to wake dormant VoIP apps for incoming call delivery.
- Managing background execution limits on iOS and Android to maintain registration without excessive battery drain.
- Implementing CallKit (iOS) and ConnectionService (Android) integrations for native dialer appearance and call handling.
- Negotiating carrier VoLTE interoperability requirements when coexisting with native telephony services.
- Handling SIM-based authentication conflicts when multiple VoIP accounts are provisioned on dual-SIM devices.
- Testing app behavior under network switching events (e.g., Wi-Fi to cellular handover) to ensure call preservation.