Low-latency, interactive media applications (eg, video conferencing) present a unique set of
challenges for congestion control. Unlike TCP, the transport mechanism for interactive …

One of the important use cases for WebRTC is multi-party audio/video conferencing, and
there are multiple services that offer that functionality. Two aspects that the maintainers of …

Due to the presence of buffers in the inner network nodes, each congestion event leads to
buffer queueing and thus to an increasing end-to-end delay. In the case of delay sensitive …

A method for adjusting media stream transmission bandwidth and a related apparatus are
provided. A media stream sending end acquires scores that are given, according to …

Despite the fact that non-interactive Internet multimedia applications are prevalently
deployed via HTTP/TCP, dedicated real-time protocols remain the essential basis for …

WebRTC is a promising standard for real-time communication in the browser. It supports
transfer of delay sensitive real-time video and audio material using SRTP, as well as transfer …

The majority of real-time multimedia traffic on the Internet is transported using the
Transmission Control Protocol (TCP) or the User Datagram Protocol (UDP). Multimedia …

IP telephony has evolved following a structure inherited from traditional telephony, that is, a
relay-based, station to station model. This philosophy has deeply influenced the evolution of …

This paper proposes a novel congestion control algorithm for low-delay communication over
best effort packet switched networks. Due to the presence of buffers in the internal network …