Real-time audio is not something every Android developer works with, but it quickly becomes interesting once you do. Packets can arrive late or out of order, networks can change during a call, and even small delays can make a conversation feel unnatural. In this session, I'll share what I learned from working on a real-world calling system and explain the foundations Android developers should know when dealing with real-time audio. We'll look at how audio travels from microphone to network to speaker, how latency, jitter, packet loss, and network changes affect call quality, and why there is rarely one perfect solution. You'll leave with a practical mental model for debugging call-quality issues, working more effectively with backend and networking teams, and feeling more confident when approaching WebRTC or other real-time technologies.