De Fre, MatthiasMatthiasDe Frevan der Hooft, JeroenJeroenvan der HooftChang, Chia-YuChia-YuChangDe Schepper, KoenKoenDe SchepperAlface, Patrice RondaoPatrice RondaoAlfaceDe Vleeschauwer, DannyDannyDe VleeschauwerWauters, TimTimWautersSteenkiste, PeterPeterSteenkisteDe Turck, FilipFilipDe Turck2025-06-022025-05-302025-06-022025https://imec-publications.be/handle/20.500.12860/45738Current networking solutions are unable to satisfy the low-latency requirements of real-time volumetric video conferencing when faced with heavy congestion scenarios. Traditional congestion controllers use packet loss or the change in round-trip time (RTT) to estimate the bandwidth. Commonly, this method is too slow as congestion has already occurred and the receiving user has already experienced a latency spike. Low latency, low loss and scalable throughput (L4S), recently published as RFC 9330, wants to alleviate this problem by aiming for sub 1 ms queuing delay for low-latency traffic by using accurate explicit congestion notification (AccECN) packet marking to notify applications of early congestion. We propose an L4S-based pipeline for volumetric video delivery, which achieves a more consistent latency under congestion compared to web real-time communication (WebRTC). In addition, L4S bandwidth estimation achieves a 45% faster convergence compared to Google congestion control (GCC) estimation, commonly used in WebRTC. Furthermore, in our detailed evaluation setup the L4S application experiences no packet loss, while the WebRTC-based version suffers from irrecoverable packet loss, resulting in 3% of frames being undecodable.Proceedings paper10.1145/3712676.3714443979-8-4007-1467-2WOS:001480944300011REAL-TIME