This paper presents an end-to-end algorithm for detecting a shared congested router in the internet when communication is established using multiple paths between wireless devices endowed with multiple interfaces (e.g., LTE, and WiFi). The proposed technique is based on one way end-to- end delay observation of packets received through multiple paths at the end hosts to infer whether or not the flows share the same congested router. With very different statistics on the wireless access delays encountered across different interfaces in practical scenarios, this paper describes techniques for estimating queuing delay variations for each path from one way end-to-end delay observations. The proposed scheme does not make use of exclusive probing packets, but instead uses the data packets that are transmitted and received through multiple paths to infer the path's one way delay characteristics. Simulation results assessing the performance of our proposed detection algorithm are provided for scenarios where wireless devices exchange traffic through its LTE and WiFi interface. With the recently proposed protocols such as multi path transmission control protocol (MPTCP), and multi path real time protocol (MPRTP) beginning to be embraced with increasing interests, this study becomes important as it enables the end hosts to perform efficient congestion control.

This paper presents an Active Queue Management (AQM) algorithm for improving the Quality of Experience (QoE) of video telephony over packet switched networks. The algorithm exploits the characteristics of the video coding structure, and builds on the Controlled Delay (Codel) active queue management algorithm recently proposed by Nichols and Jacobson to address the prevalent ‘bufferbloat’ problem in the current Internet. The proposed algorithm, Loss Concentration based controlled Delay (LC-Codel), maintains low queuing delay which is essential for video telephony, while using loss concentration to improve video QoE. Simulation results show significant gains in QoE with negligible impact on cross traffic.

The ultimate goal of network resource allocation for video teleconferencing is to optimize the Quality of Experience (QoE) of the video. We consider the IPPP video coding structure with macroblock intra refresh, which is widely used for video teleconferencing. Generally, the loss of a current frame causes error propagation to subsequent frames due to the video coding structure. Therefore, to optimize the QoE, a communication network needs to be able to accurately predict the consequence of each of its resource allocation decisions. We propose a QoE prediction scheme by considering QoE models that use the per-frame PSNR time series as the input, thus reducing the QoE prediction problem to a per-frame PSNR prediction problem. The QoE prediction scheme is jointly implemented by the video sender (orMCU) and the communication network. Simulation results show that the proposed per-frame PSNR prediction method is fairly accurate, with an average error well below 1dB.

This paper considers a downlink LTE system where a basestation (eNodeB) is serving video traffic (generated possibly due to video teleconferencing applications) to many users in a cell. The video traffic is separated into multiple sub-streams (logical channels) based on the coding structure or priority of video packets. The objective is to maximize video quality by servicing appropriate video sub-streams and users while taking into account the resource constrained wireless channel. It is shown that one can obtain significant gains in video quality by determining the quality of service (QoS) parameters on a per sub-stream basis, rather than in `user-based' approach where all the sub-streams are lumped into one single stream. Motivated by video teleconferencing applications, we provide simulation results for the case where video traffic is separated into logical channels based on a Hierarchical-P encoding structure. Furthermore, we demonstrate the gains in video quality that can be obtained by allocating resources across logical channels. With the exploding growth in mobile video traffic expected to happen in 4G systems, this study suggests potential benefits of a logical channel based approach as compared to conventional user-based schemes.

Symmetrical multilevel diversity coding (SMDC) is a network compression problem introduced by Roche (1992) and Yeung (1995). In this setting, a simple separate encoding strategy known as superposition coding was shown to be optimal in terms of achieving the minimum sum rate (Roche-Yeung-Hau 1997) and the entire admissible rate region (Yeung-Zhang 1999) of the general problem. This paper considers a natural generalization of SMDC to the secure communication setting with an additional eavesdropper. It is required that all sources need to be kept perfectly secret from the eavesdropper as long as the number of encoder outputs available at the eavesdropper is no more than a given threshold. First, the problem of encoding individual sources is studied. A precise characterization of the entire admissible rate region is established via a connection to the problem of ramp-type secret sharing (Yamamoto 1985 and Blakley-Meadows 1985) and utilizing some basic polyhedral structure of the admissible rate region. Building on this result, it is then shown that superposition coding remains optimal in terms of achieving the minimum sum rate for the general secure SMDC problem.

A wireless communication system where a common base station (BS) is scheduled to transmit information to mobile users on a time division (TD) basis, under quality of service (QoS) constraints is considered. The quality of service requirement is specified in terms of the asymptotic decay rate of the buffer occupancy. The effective capacity of this system is characterized and the optimal scheduling scheme which achieves the boundary of the effective capacity region is found. It is further shown that the optimal scheduling scheme without quality of service constraints and the opportunistic scheduling schemes fall out as special cases of a more generalized class of scheduling schemes derived in this paper.

A wireless communication system where a common base station is scheduled to transmit information to multiple mobile users on a time division (TD) basis is considered. The capacity region of this system is found for the two user case and the optimal scheduling scheme is proposed which achieves the boundary of the capacity region. Furthermore, the optimal scheduling scheme is found for the case when the remote mobile users can perform downlink cooperation and the achievable rate region of the downlink cooperative system is characterized. Finally, a simple iterative algorithm is proposed for finding the resource allocation parameters and the scheduling scheme for the cooperative system.