With the increasing reliance on real-time communication platforms such as Zoom, Teams, and WhatsApp, maintaining low latency, jitter, and packet loss for VoIP (Voice over IP) has become critical. Traditional Linux networking stacks handle all traffic uniformly, often leading to degraded Quality of Service (QoS) for VoIP under heavy congestion.
Conventional QoS mechanisms like tc filters or static traffic shaping lack the adaptability and efficiency required for modern, high-throughput environments. There is a need for a programmable, kernel-level solution that can dynamically prioritize VoIP traffic while ensuring fair sharing of bandwidth.
We brought together XDP, Traffic Control (TC), and AQM to build a low-latency path specifically for WebRTC voice traffic, helping maintain a stable bitrate and minimal jitter even under heavy network congestion.
Looks good to me, although it might have been nice to a more in-depth explanation.
My only concern is if the audience will understand this talk, considering it requires a primer on networking etc.