Reaz Telecom R&D

VoIP, SBC, SIP, WebRTC, Embedded Voice, Observability, SS7, AI Voice, RTPengine

Telecom engineering, field notes, and service areas

Voice systems, signaling paths, media behavior, and carrier integrations.

This page is the technical map for my telecom work: VoIP platforms, SIP/SBC routing, WebRTC, embedded voice, observability, AI voice, call center systems, Kubernetes signaling experiments, FreeSWITCH, OpenSIPS, RTPengine, SS7 learning, and practical case studies from call-flow debugging.

Expertise

Areas I want this telecom lab to organize and explain clearly.

VoIP Systems

SIP trunks, gateways, dialplans, softswitch behavior, registration, routing, call bridging, codec negotiation, and real call-flow troubleshooting.

SBC & SIP Signaling

SBC-style logic, topology control, SIP routing, header normalization, authentication, failover, trunk policies, and interconnect debugging.

WebRTC and Browser Voice

WebRTC-to-SIP flows, ICE/STUN/TURN concepts, browser media behavior, signaling gateways, and RTP/SRTP troubleshooting.

Embedded VoIP

Voice devices, SIP endpoints, ESP/audio experiments, constrained devices, RTP paths, microphone/speaker behavior, and practical embedded call flows.

Telecom Observability

Logs, SIP ladders, sngrep traces, RTP checks, response-code dashboards, carrier failure patterns, and operational visibility for voice systems.

Telecom Architectures

How proxies, SBCs, media servers, PBX systems, carriers, APIs, and customer-facing call applications fit together.

Services

Practical work around debugging, routing, deployment, and voice architecture.

SIP / VoIP Troubleshooting

Investigate failed calls, rejected trunks, registration issues, route loops, codec mismatches, and one-way/no-audio problems.

Softswitch and Routing Design

Design or clean up routing logic around FreeSWITCH, Asterisk, OpenSIPS, Kamailio, trunks, gateways, and carrier failover.

Call Center and AI Voice Systems

Work on voice workflows, IVR behavior, queue/campaign flow, SIP integration, call recording, and AI voice call architecture.

Self-hosted Telecom Stack

Plan self-hosted voice infrastructure with observability, backups, Git-backed documentation, deployment safety, and operational clarity.

Case Studies

Realistic telecom problems this site should document as posts.

SIP 488 Not Acceptable HereDebug the SDP offer, codec list, ptime, RTP profile, and carrier-supported media requirements.
One-way AudioTrace SIP and RTP together: connection IP, NAT, advertised media address, firewall path, and RTP packet direction.
OpenSIPS FailoverDispatcher and LCR routing strategy for carrier failure, timeout, route selection, and retry behavior.
FreeSWITCH Gateway RegistrationInspect credentials, realm, contact header, NAT mapping, registration interval, and provider response codes.
WebRTC to SIP BridgeUnderstand browser media, ICE, signaling gateway, RTP/SRTP conversion, and SIP trunk termination.
Kubernetes SignalingExplore what belongs in Kubernetes, what should stay close to media, and how SIP/RTP behaves in container networks.

Study Areas

Research tracks for deeper telecom architecture and signaling knowledge.

SS7

Learning map for legacy/mobile signaling, telecom-core concepts, and how it relates to modern VoIP/IMS work.

XMPP

Messaging/signaling concepts, presence, real-time communication, and how XMPP overlaps with telecom-style systems.

RTPengine

Media relay, NAT traversal, RTP/SRTP handling, WebRTC interop, and proxy/media separation.

RFCs

Reading protocol specs as engineering references: SIP, SDP, RTP, NAT traversal, and telecom interoperability.

IoT Telecom

Connected voice devices, embedded endpoints, signaling constraints, and device-to-network communication patterns.

Exploit Research

Security learning around SIP scanning, abuse patterns, weak authentication, exposed services, and defensive controls.