A self-paced, hands-on study site that takes a backend engineer from "what is a phone call" all the way to debugging a live customer SIP integration. Every concept is grounded in a runnable example or a real Freya deployment.
Signaling vs media, PSTN to VoIP, codecs, DTMF, Q.850 cause codes, and call-quality metrics.
Methods, headers, dialogs vs transactions, SDP, the INVITE handshake, status codes, registration.
RTP packet structure, codec negotiation, STUN/TURN/ICE, SBC, the famous 60-second timeout.
Modules, dialplan, pjsip.conf, contexts, transports, identify, and the Asterisk CLI.
chan_websocket, /telephony/ws, JSON frames, MixMonitor, end-to-end inbound/outbound flows.
STIR/SHAKEN, the 60-second drop, light proxies, Timer B, SRTP, concurrent caps, recording compliance.
Asterisk CLI, sngrep, tcpdump + Wireshark, the 6-hypothesis SIP-debug decision tree.
Compose layout, host-mounted configs, our PJSIP defaults, custom headers, campaign-worker outbound.
Allianz · Garanti · Anadolu Sigorta · KKB. The capstone: trace a live call across every layer.
Zoom out. The five layers of voice in 2026 (operator, trunk, SBC, PBX, contact center) plus BPO. Where each customer sits.