May 18, 2026
RTPENGINE
A telecom engineering note from the field.
SIP trace notes
INVITE, REGISTER, BYE, CANCEL, response codes, route headers, authentication, trunk behavior, and call-flow notes from real traces.
Reaz Telecom R&D
VoIP, SBC, SIP, WebRTC, Embedded Voice, Observability, SS7, AI Voice, RTPEngine
Telecom engineering, field notes, and research areas
Voice systems, signaling paths, media behavior, and carrier integrations.
This page is a technical map for writings and notes: SIP/SBC routing, WebRTC, embedded voice, observability, AI voice, call-center systems, Kubernetes signaling, FreeSWITCH, OpenSIPS, RTPEngine, SS7 learning, and practical case studies from call-flow debugging.
Field Notes
Notes, traces, experiments, and references organized clearly.
RTP and media notes
RTP direction, advertised media IPs, NAT behavior, codecs, jitter, SRTP/RTP boundaries, RTPEngine behavior, and one-way audio notes.
WebRTC notebook
Browser media behavior, ICE, STUN/TURN, SIP over WebSocket, WebRTC-to-SIP bridges, and debugging notes from browser voice experiments.
Embedded voice notes
SIP endpoints, ESP/audio experiments, constrained-device call flows, RTP send/receive, microphone/speaker paths, and device-level debugging.
Observability notes
sngrep traces, Wireshark captures, RTP checks, response-code patterns, logs, metrics, and operational notes for understanding voice systems.
Architecture notes
How proxies, SBCs, media servers, PBX systems, carriers, APIs, browser clients, and embedded endpoints fit together.
Latest Blog Posts
New notes and articles from the backend.
May 18, 2026
Tesy
A telecom engineering note from the field.
Case Studies
Problems worth documenting as field notes.
SIP 488 Not Acceptable Here
Codec list, SDP offer, RTP profile, and downstream carrier media requirements.
One-way Audio
Connection IP, NAT, advertised media address, RTP timeout, firewall path, and RTPEngine direction.
OpenSIPS Failover
Dispatcher and LCR routing strategy for carrier failure, timeout, route selection, and retry behavior.
WebRTC to SIP Bridge
Browser media, ICE, signaling gateway, RTP/SRTP conversion, and SIP trunk termination.
Recent Projects
Labs, builds, and implementation notes.
OpenSIPS routing lab
Dispatcher, route failover, and carrier trunk testing.
FreeSWITCH gateway notes
Registration, dialplan behavior, NAT mapping, and RTP checks.
Embedded SIP endpoint
Device audio path, SIP registration, RTP receive/transmit, and auto-answer flow.
WebRTC softphone test
SIP over WebSocket, ICE behavior, and browser audio handling.
R&D
Research tracks for deeper telecom architecture and signaling notes.
Protocol reading
RFC notes, SIP/SDP references, SS7 learning, XMPP signaling, RTPEngine behavior, and architecture reading notes.
Experiment log
Small tests, failed attempts, audio experiments, browser voice tests, embedded VoIP trials, and things worth documenting later.
Restore notes
Deployment notes, backup notes, config notes, Git-backed documentation, server layout, recovery steps, and production checklists.
Trace queue
Things to write from real traces: rejected calls, route loops, codec mismatches, registration problems, and media failures.
Topics