apolo-next

Use this skill for any task involving VoIP infrastructure with Kamailio, FreeSWITCH, RTPEngine, Asterisk, or related SIP/RTP components. Triggers include configuring SBCs, debugging SIP signaling (REGISTER, INVITE, 100rel/PRACK, 302 redirects), troubleshooting RTP/media flows, designing dialplans, building IVRs, integrating CGRateS or custom billing engines, analyzing SIP traces (pcap, sngrep, sipgrep), tuning timers (fr_timer, fr_inv_timer), bypass_media configurations, dual-interface RTPEngine setups, FreeSWITCH B2BUA interworking, Asterisk-to-FreeSWITCH migrations, dSIPRouter, concurrent call limiting, air-gapped/containerized VoIP deployments on RHEL/Debian, observability and monitoring of voice stacks (MOS, jitter, packet loss, RTT, call counters via Prometheus/Loki/Grafana/Telegraf), and modifying or extending the operational scripts in scripts/ (notably voip-doctor.sh for end-to-end diagnostics and voip-stats-collector.sh for JSON metrics export). Do NOT use for general networking questions unrelated t