This study assesses the Quality of Service (QoS) performance of a Web Real-Time Communication (WebRTC)-based video streaming system operating on a local network utilizing a broadcaster-viewer model. The system was constructed with Node.js as the local server, accompanied by broadcaster and viewer web pages that facilitate laptop camera and microphone streaming, video conferencing, camera effects, and MP4 media playback. The research strategy utilized experimental performance evaluation through a scenario-based load testing approach and Quality of Service benchmarking. Testing was performed across three scenarios: one broadcaster with one viewer, one broadcaster with two viewers, and one broadcaster with three viewers. The examined QoS characteristics encompassed throughput, packet loss, delay, and jitter, assessed utilizing Wireshark. The test findings indicated that throughput escalated with the increase in viewers, rising from 2.525 Mbps in Scenario 1 to 5.026 Mbps in Scenario 3. Packet loss remained minimal throughout all scenarios, however it increased to 0.2% in Scenario 3. Scenario 2 had the largest average delay at 751.92 ms, whilst Scenario 3 demonstrated the lowest average delay at 349.06 ms and the minimal average jitter at 1.645 ms. The findings demonstrate that WebRTC can provide multi-viewer local streaming with steady performance; yet, a rise in watchers necessitates meticulous management of bandwidth and network stability to preserve video content quality.