Video conferencing applications require low latency and high bandwidth. Standard TCP is not suitable for video conferencing since its reliability and in order delivery mechanisms induce large latency. Recently the idea of using the delay gradient to infer congestion is appearing again and is gaining momentum. In this paper we present an algorithm that is based on estimating through a Kalman filter the end-to-end one way delay variation which is experienced by packets traveling from a sender to a destination. This estimate is compared to an adaptive threshold to dynamically throttle the sending rate. The control algorithm has been implemented over the RTP/RTCP protocol and is currently used in Google Hangouts and in the Chrome WebRTC stack. Experiments have been carried out to evaluate the algorithm performance in the case of variable link capacity, presence of heterogeneous or homogeneous concurrent traffic, and backward path traffic.
|Titolo:||Analysis and design of the google congestion control for web real-time communication (WebRTC)|
|Data di pubblicazione:||2016|
|Nome del convegno:||7th ACM International Conference on Multimedia Systems, MMSys 2016|
|Digital Object Identifier (DOI):||10.1145/2910017.2910605|
|Appare nelle tipologie:||4.1 Contributo in Atti di convegno|