Spanning the Globe: Geo-Routing and Scalable TURN Architectures

The Physics of Latency: Why "Good Enough" Fails at Scale In the realm of real-time communications (RTC), we are fundamentally fighting physics. While HTTP APIs can tolerate an extra 200ms of latency without the user noticing, real-time audio and video are unforgiving. WebRTC relies heavily on UDP for media transport, prioritizing timeliness over reliability. When a packet is lost in a TCP stream, the protocol pauses to retransmit, ensuring data integrity. In WebRTC, if a packet arrives late, it is useless. It is discarded, causing a glitch in audio or a freeze in video. The most critical metric in this equation is Round Trip Time (RTT). RTT dictates everything: the size of the jitter buffer, the aggressiveness of the congestion control algorithm (like Google GCC), and the user's perception of interactivity. Consider a concrete scenario: User A is in Tokyo, and User B is in Virginia. If they connect via a media server located in Virginia, User A's audio packets must travel across the Pa