Episode 63: SIP and VoIP Protocols

This episode introduces Session Initiation Protocol (SIP) and other Voice over IP (VoIP) protocols that support modern telephony systems. SIP establishes, manages, and terminates calls, while protocols such as RTP (Real-Time Transport Protocol) handle the actual media streams. The exam tests these because voice traffic has specific requirements for quality and reliability, making it distinct from typical data services.

The discussion expands with examples, such as how SIP messages initiate calls between softphones or desk phones and how QoS mechanisms prioritize RTP traffic to reduce jitter and latency. Troubleshooting scenarios include diagnosing dropped calls, one-way audio, or firewall rules blocking SIP traffic. By mastering these protocols, you’ll be prepared to interpret exam questions and support real-world VoIP deployments effectively. Produced by BareMetalCyber.com, where you’ll find more cyber audio courses, books, and information to strengthen your certification path.