Certified: The CompTIA CloudNetX Audio Course

Voice and video protocols appear in CloudNetX scenarios as clues about traffic behavior and performance sensitivity, and this episode teaches how to interpret those clues without getting lost in implementation detail. It defines SIP as a signaling protocol that establishes voice and video sessions, WebRTC as a framework for real-time communication in browsers and applications using encrypted media transport, RTSP as a control protocol commonly used for streaming and camera feeds, and H.323 as a legacy conferencing suite still found in some enterprise environments. The first paragraph focuses on the key implication shared by these workloads: they are sensitive to latency, jitter, and packet loss in ways that basic web browsing is not, and they often require consistent paths and appropriate prioritization. The episode explains that protocol names in a scenario are signals that you should think about QoS, capacity planning, and inspection impacts rather than treating the traffic as generic TCP data.