Розробка адаптивного механізму керування перевантаженням для мультимедіа-стримінгу в реальному часі за змінних умовах мережі

This research focuses on real-time multimedia streaming using RTP and RTCP protocols. The main issue addressed is that standard RTP/RTCP congestion control is inadequately adapted to changing and unstable network conditions, resulting in increased packet loss, end-to-end latency, unstable bitrates, and poor video quality. A dynamic bandwidth-adaptive congestion control mechanism was developed for RTP streaming, which utilizes RTCP feedback to dynamically change the bitrate and framerate in real time during the streaming session. Controlled experiment results show that average packet loss decreases from 8.2% to 3.4%; end-to-end latency decreases from an average of 220 ms to 135 ms; and provides a more stable average bitrate than standard RTP/RTCP systems. Furthermore, this system also provides a more stable average framerate than standard RTP/RTCP systems and a higher average framerate under poor network conditions. This result can be attributed to the ability of the adaptive mechanism to continuously monitor packet loss, interference, and delays in addition to reacting immediately to conditions instead of waiting for RTCP reports to appear at fixed time intervals. A key point regarding the proposed design is the integration of bitrate and framerate to ensure smooth playback and user enjoyment with reduced risk of interruption and improved stability in dynamic and unpredictable network environments. This contribution can be practically applied in real-time applications, such as video conferencing, telemedicine, or live streaming while traversing mobile or wireless networks where conditions are always dynamic and unpredictable. The proposed method can be practically applied under unfavorable internet network conditions, which is an advantage of this method