SIP Analyzing & Troubleshooting Masterclass

Trace SIP transactions and dialogues to understand call flows, identify C-sequence changes with new requests, and define dialogues using Call-ID, From, and To tags for troubleshooting.

Explain how session description protocol negotiates media parameters in SIP via an offer-answer model, detailing SDP line types, codecs such as G711, RTP, and ptime in invites and 200 OK.

Master sip signaling with sngrep, a real-time command-line tool for monitoring and capturing SIP messages, with a graphical interface to visualize VoIP call flow for troubleshooting.

Explore Homer, an open-source real-time SIP capture and monitoring system with Grafana integration, guiding you through installation, configuration with Asterisk/FreePBX, and SIP call analysis and exports.

Compare tcpdump, sngrep, and homer for sip debugging and traffic analysis. Use sngrep for quick live debugging, tcpdump for full captures, and homer for long-term sip analytics.

Learn how the SIP options method, or SIP ping, queries capabilities and validates far-end state without a session, supporting heartbeat and NAT keep-alives with 200, 404, or 405 responses.

Learn how SIP registration binds a user address of record to a contact URI on a registrar, enabling multi-device presence, with expiring registrations, 401/407 challenges, and the role of proxies.

Explore the SIP subscribe and notify methods for managing event subscriptions, including expiration, renewal, and unsubscribe flows, and see how notify delivers message waiting indicators from voicemail and related events.

Explore PRAC, the provisional response acknowledgement in sip, and see how 180 ringing is reliably acknowledged with 100 REL, r-sequence, and rac headers in a pbx–ip phone call.

Analyze common VoIP issues like one-way and no audio by troubleshooting SIP signaling, RTP flows, and SDP configurations, including firewall rules, port ranges, and NAT considerations.

Analyze how NAT affects VoIP calls by showing how SDP and SIP-ALG issues misroute RTP streams, causing no audio or one-way audio.

Learn how jitter and packet loss affect VoIP audio and how RTCP monitors RTP to report jitter, packet loss, and MOS, with Wireshark demos.

Discover how sip over tls encrypts signaling with transport layer security, while media remains unencrypted, and learn default ports, decryption with a private key, and the difference from standard sip.

Describe how CA-signed and self-signed certificates bind public keys to server identities for SIP, enabling verification during the TLS handshake and secure key exchange with AES protecting the payload.

Analyze the tls handshake that secures sip traffic on port 5061, covering client hello, server hello, certificate exchange, key creation, and common failure causes.

Identify and fix common tls issues in production, including expired certificates, tls handshake problems, and untrusted authorities. Verify system time, daylight saving, trust stores, and open port 5061.