This paper proposes a controlled load service that provides a network state with bounded and well known worst-case behavior. The service is primarily developed for real time applications. The full system for achieving quality of service to the application consists of an admission control combined with forward-error correction. The admission control is used to limit the packet-loss probability to a known value; the error-control coding (i.e., FEC) is then used to raise the quality above the level enforced by the admission control. The basic idea for the admission control is that a host must probe the path to the receiver before sending actual data. It accepts the session if the probe is received with no or at most a moderate amount of loss. The performance evaluation shows clearly that the proposed scheme avoids network congestion and high packet losses even over short time scales.

Admission control based on end-to-end measurements

In differentiated services (DiffServ) domains, where services are provisioned on a per-class basis, admission control is an essential control factor in order to ensure that congestion is avoided and that the quality of service (QoS) requirements of individual flows are met. We consider traffic-engineered and provisioned IP differentiated services domains able to support real-time traffic. We present a new measurement-based admission control (MBAC) scheme that uses measurements of aggregate bandwidth only, without keeping the state of any per-flow information. In our scheme there is no assumption made on the nature of the traffic characteristics of the real-time sources, which can be of any heterogeneous nature. Through simulations we show that the admission control scheme is robust with respect to traffic heterogeneity and measurement errors. We also show that our approach compares favorably against other admission control schemes found in the literature.

/pdf/heterogeneous-real-time-traffic-admission-control-in-rz401ylpnu.pdf

Heterogeneous real-time traffic admission control in differentiated services domains

Wireless multimedia synchronization is concerned with distributed multimedia packets such as video, audio, text and graphics being played-out onto the mobile clients via a base station (BS) that services the mobile client with the multimedia packets. Our focus is on improving the Quality of Service (QoS) of the mobile client's on-time-arrival of distributed multimedia packets through network multimedia synchronization. We describe a media synchronization scheme for wireless networks, and we investigate the multimedia packet scheduling algorithms at the base station to accomplish our goal. In this paper, we extend the media synchronization algorithm by investigating four packet scheduling algorithms: First-In-First-Out (FIFO), Highest-Priority-First (PQ), Weighted Fair-Queuing (WFQ) and Round-Robin (RR). We analyze the effect of the four packet scheduling algorithms in terms of multimedia packet delivery time and the delay between concurrent multimedia data streams. We show that the play-out of multimedia units on the mobile clients by the base station plays an important role in enhancing the mobile client's quality of service in terms of intra-stream synchronization and inter-stream synchronization. Our results show that the Round-Robin (RR) packet scheduling algorithm is, by far, the best of the four packet scheduling algorithms in terms of mobile client buffer usage. We analyze the four packet scheduling algorithms and make a correlation between play-out of multimedia packets, by the base station, onto the mobile clients and wireless network multimedia synchronization. We clarify the meaning of buffer usage, buffer overflow, buffer underflow, message complexity and multimedia packet delay in terms of synchronization between distributed multimedia servers, base stations and mobile clients.

Media synchronization and QoS packet scheduling algorithms for wireless systems

One of the most interesting improvements of the Internet today is the provisioning of services for interactive audio-visual applications. Such applications have quality requirements which place limits on both transfer delays and information losses. Our goal is to support interactive video with only small modifications to network routers and control systems. The approach we favor is to ensure the quality for a session by forward-error correction. The code strength is dynamically tuned to meet a user’s quality expectation, given the experienced loss process of the transfer. To make the tuning feasible, the network state must be predictable. We accomplish this by regulation of the load through sender-based admission control. It uses end-to-end probes of the network state and a self-imposed blocking if then sender determines that a sensible transfer cannot be made. The paper outlines this procedure along with a review of source coding considerations and a description of tunable error-control coding.

Providing quality for Internet video services

Place reservation: Delay analysis of a novel scheduling mechanism

We introduce a new scheduling algorithm for multimedia traffic using capacity reservation. We compare it with other algorithms in the literature. It has been implemented and its worst-case performance has been analysed. It appears to give a noticeably improved quality of service to delay-sensitive traffic.

http://iopscience.iop.org/article/10.1088/0967-1846/5/1/002/pdf

Torsten Köhler

Papers

Capacity reservation for multimedia traffic