Abstract
Video streaming services account for a substantial proportion of global internet traffic, but the quality of user experience highly depends on the performance of the transport protocol used. This study comparatively analyzes the performance of the Transmission Control Protocol (TCP) and User Datagram Protocol (UDP) in Full HD 1080p video streaming under normal and congested network conditions. The main issue addressed is how the flow control mechanism in TCP affects data delivery stability compared to the connectionless best-effort transmission mechanism of UDP during network congestion. This study employed an experimental research method by capturing network traffic using Wireshark version 4.4.2 under two testing scenarios: normal network conditions and congested network conditions. The experimental procedure consisted of network topology design, traffic generation, packet capture, QoS parameter extraction, and comparative performance evaluation of the TCP and UDP protocols. The novelty of this study lies in the direct analysis of the impact of TCP retransmission mechanisms on latency spikes in a congested local network environment. Experimental results show that UDP achieves higher throughput than TCP under both normal and congested network conditions. TCP maintains stable latency at around 14–15 ms with reliable data delivery, while UDP exhibits more fluctuating latency ranging from 10 ms to 35 ms under congested conditions. However, UDP maintains real-time video transmission despite experiencing 2.4% packet loss. Both protocols remain well below the 150 ms one-way latency threshold recommended by ITU-T G.1010 for acceptable interactive multimedia experience. Consequently, UDP is recommended for delay-sensitive live streaming because it preserves real-time transmission continuity despite moderate packet loss, whereas TCP is recommended for Video on Demand (VoD) applications because its acknowledgment and retransmission mechanisms ensure superior data integrity and visual stability.
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