First Telecom Field Note

This is a simple test post for the Telecom R&D notebook.

The goal is to confirm the publishing workflow:

• write a post from terminal
• keep the post in GitHub
• deploy through CI/CD
• show the post on the website

What this note is for

This note is a placeholder for future SIP, RTP, WebRTC, FreeSWITCH, OpenSIPS, and embedded VoIP debugging notes.

Example trace checklist

SIP REGISTER: check authentication and contact header
SIP INVITE: check route decision and SDP offer
RTP path: check advertised media IP and port
Audio issue: check codec, NAT, firewall, and RTPEngine direction
Next notes to write

Future posts can document real call-flow problems such as SIP 488, one-way audio, trunk failover, RTP path issues, or WebRTC-to-SIP bridge behavior.