In this paper, we discuss UE power saving (PS) techniques for eXtended Reality (XR) applications in 5G NR system. XR traffic is characterized by pseudo periodic packet arrival, high bit rate, and low latency requirement. It is challenging to support such traffic in a small device form factor with limited battery size, e.g., head-mounted display (HMD) or XR glasses. Therefore, it is desired for 5G NR to support effective UE PS techniques. We first discuss a few PS techniques defined in 5G NR specifications Release 15 and 16. Then, we introduce new PS techniques that are evaluated via system level simulations with the UE power consumption model defined in a 3GPP. Evaluation results show that UE power consumption can be substantially reduced by the new PS techniques, up to 40%.

We present system-level evaluation results to characterize the performance of eXtended Reality (XR) applications over a 5G NR system. A split-rendering framework is assumed, where the XR device uses 5G to offload computation associated with rendering and encoding of video frames to an edge server. The XR traffic model includes rendered video frames on the downlink and user pose and control updates on the uplink. We present results for both sub-6 GHz (FR1) and millimeter-wave (FR2) bands. Additionally, we also discuss potential enhancements to further improve user experience for XR applications over 5G.

Current approaches to the practical implementation of network coding are batch-based, and often do not use feedback, except possibly to signal completion of a file download. In this paper, the various benefits of using feedback in a network coded system are studied. It is shown that network coding can be performed in a completely online manner, without the need for batches or generations, and that such online operation does not affect the throughput. Although these ideas are presented in a single-hop packet erasure broadcast setting, they naturally extend to more general lossy networks, which employ network coding in the presence of feedback. The impact of feedback on sender-side queue size and receiver-side decoding delay is studied in an asymptotic sense as the traffic load approaches capacity. Different notions of decoding delay are considered, including an order-sensitive notion, which assumes that packets are useful only when delivered in order. Strategies for adaptive coding based on feedback are presented. Our scheme achieves throughput optimality and asymptotically optimal sender queue size and is conjectured to achieve asymptotically optimal in-order delivery delay for any number of receivers. This paper may be viewed as a natural extension of Automatic Repeat reQuest to coded networks.

The theory of network coding promises significant benefits in network performance, especially in lossy networks and in multicast and multipath scenarios. To realize these benefits in practice, we need to understand how coding across packets interacts with the acknowledgment (ACK)-based flow control mechanism that forms a central part of today's Internet protocols such as transmission control protocol (TCP). Current approaches such as rateless codes and batch-based coding are not compatible with TCP's retransmission and sliding-window mechanisms. In this paper, we propose a new mechanism called TCP/NC that incorporates network coding into TCP with only minor changes to the protocol stack, thereby allowing incremental deployment. In our scheme, the source transmits random linear combinations of packets currently in the congestion window. At the heart of our scheme is a new interpretation of ACKs-the sink acknowledges every degree of freedom (i.e., a linear combination that reveals one unit of new information) even if it does not reveal an original packet immediately. Thus, our new TCP ACK rule takes into account the network coding operations in the lower layer and enables a TCP-compatible sliding-window approach to network coding. Coding essentially masks losses from the congestion control algorithm and allows TCP/NC to react smoothly to losses, resulting in a novel and effective approach for congestion control over lossy networks such as wireless networks. An important feature of our solution is that it allows intermediate nodes to perform re-encoding of packets, which is known to provide significant throughput gains in lossy networks and multicast scenarios. Simulations show that our scheme, with or without re-encoding inside the network, achieves much higher throughput compared to TCP over lossy wireless links. We present a real-world implementation of this protocol that addresses the practical aspects of incorporating network coding and decoding with TCP's window management mechanism. We work with TCP-Reno, which is a widespread and practical variant of TCP. Our implementation significantly advances the goal of designing a deployable, general, TCP-compatible protocol that provides the benefits of network coding.

The problem of serving multicast flows in a crossbar switch is considered. Intraflow linear network coding is shown to achieve a larger rate region than the case without coding. A traffic pattern is presented which is achievable with coding but requires a switch speedup when coding is not allowed. The rate region with coding can be characterized in a simple graph-theoretic manner, in terms of the stable set polytope of the "enhanced conflict graph". No such graph-theoretic characterization is known for the case of fanout-splitting without coding. The minimum speedup needed to achieve 100% throughput with coding is shown to be upper bounded by the imperfection ratio of the enhanced conflict graph, where the imperfection ratio measures a certain graph theoretic property of the given graph. When applied to K × N switches with unicasts and broadcasts only, this gives a bound of min(2K-1/K, 2N/N+1) on the speedup. This shows that speedup, which is usually implemented in hardware, can often be substituted by network coding, which can be done in software. Computing an offline schedule (using prior knowledge of the flow rates) is reduced to fractional weighted graph coloring. A graph-theoretic online scheduling algorithm (using only queue occupancy information) is also proposed, that stabilizes the queues for all rates within the rate region.

We propose a mechanism that incorporates network coding into TCP with only minor changes to the protocol stack, thereby allowing incremental deployment. In our scheme, the source transmits random linear combinations of packets currently in the congestion window. At the heart of our scheme is a new interpretation of ACKs - the sink acknowledges every degree of freedom (i.e., a linear combination that reveals one unit of new information) even if it does not reveal an original packet immediately. Such ACKs enable a TCP-compatible sliding-window approach to network coding. Our scheme has the nice property that packet losses are essentially masked from the congestion control algorithm. Our algorithm therefore reacts to packet drops in a smooth manner, resulting in a novel and effective approach for congestion control over networks involving lossy links such as wireless links. Our scheme also allows intermediate nodes to perform re-encoding of the data packets. Our simulations show that our algorithm, with or without re-encoding inside the network, achieves much higher throughput compared to TCP over lossy wireless links. We also establish the soundness and fairness properties of our algorithm. Finally, we present queuing analysis for the case of intermediate node re-encoding.

A new coding and queue management algorithm is proposed for communication networks that employ linear network coding. The algorithm has the feature that the encoding process is truly online, as opposed to a block-by-block approach. The setup assumes a packet erasure broadcast channel with stochastic arrivals and full feedback, but the proposed scheme is potentially applicable to more general lossy networks with link-by-link feedback. The algorithm guarantees that the physical queue size at the sender tracks the backlog in degrees of freedom (also called the virtual queue size). The new notion of a node ldquoseeingrdquo a packet is introduced. In terms of this idea, our algorithm may be viewed as a natural extension of ARQ schemes to coded networks. Our approach, known as the drop-when-seen algorithm, is compared with a baseline queuing approach called drop-when-decoded. It is shown that the expected queue size for our approach is O[(1)/(1-rho)] as opposed to Omega[(1)/(1-rho)2] for the baseline approach, where rho is the load factor.

We consider the problem of serving multicast flows in a crossbar switch. We show that linear network coding across packets of a flow can sustain traffic patterns that cannot be served if network coding were not allowed. Thus, network coding leads to a larger rate region in a multicast crossbar switch. We demonstrate a traffic pattern which requires a switch speedup if coding is not allowed, whereas, with coding the speedup requirement is eliminated completely. In addition to throughput benefits, coding simplifies the characterization of the rate region. We give a graph-theoretic characterization of the rate region with fanout splitting and intra-flow coding, in terms of the stable set polytope of the "enhanced conflict graph" of the traffic pattern. Such a formulation is not known in the case of fanout splitting without coding. We show that computing the offline schedule (i.e. using prior knowledge of the flow arrival rates) can be reduced to certain graph coloring problems. Finally, we propose online algorithms (i.e. using only the current queue occupancy information) for multicast scheduling based on our graph-theoretic formulation. In particular, we show that a maximum weighted stable set algorithm stabilizes the queues for all rates within the rate region.

We present a graph theoretic upper bound on speedup needed to achieve 100% throughput in a multicast switch using network coding. By bounding speedup, we show the equivalence between network coding and speedup in multicast switches - i.e. network coding, which is usually implemented using software, can in many cases substitute speedup, which is often achieved by adding extra switch fabrics. This bound is based on an approach to network coding problems called the "enhanced conflict graph". We show that the "imperfection ratio" of the enhanced conflict graph gives an upper bound on speedup. In particular, we apply this result to K times N switches with traffic patterns consisting of unicasts and broadcasts only to obtain an upper bound of min(2K-1/K, 2N/N+1).

We propose a new queueing mechanism for coded networks with stochastic arrivals and/or lossy links. In this context, earlier work introduced the notion of "virtual queues" which represent the backlog in degrees of freedom. For instance, the work by Ho and Viswanathan defined the achievable rate region for which the virtual queue size is stabilized, using intra-session coding. The queueing scheme that we propose here forms a natural bridge between the virtual queue size and the physical queue size, and thus extends their result to the stability of the physical queues as well. Specifically, we show that the amount of memory used at the transmit buffer in our scheme is upper bounded by the total backlog in the number of linearly independent degrees of freedom. Moreover, our scheme gives an online algorithm for queue update and coding, in the sense that the coding does not happen block by block, but in a streaming manner. The main idea in our scheme is to ensure that the information stored at the sender excludes any knowledge that is common to all receivers. This requires the transmitting node to track the states of knowledge of its receivers. Therefore, if the links are lossy, some form of feedback may be necessary.

The Birkhoff-von Neumann (BVN) strategy for single-stage input-queued crossbar switches does not support multicast, as it considers only permutation-based switch configurations. This paper extends the BVN strategy to multicast switching, where an input can simultaneously transmit to multiple outputs. Knowledge of the average rates of flows is used to compute an offline schedule. We begin by considering a system in which the fanout of each flow is split in a predecided manner. We call this static splitting (as opposed to dynamic splitting where no such constraint is imposed), and we study the rate region of the switch under this restriction. We provide a graph-theoretic formulation of the rate region.

Multicasting is emerging as an important application in both cellular networks and wireless sensor networks. We consider several schemes for enhancing throughput and coverage for streaming multicast sessions by allowing intermediate nodes to relay information from the source. The schemes exploit the broadcast nature of the wireless medium, and techniques such as received signal combining and network coding to enhance performance while guaranteeing that a constant streaming delay is experienced by every user. The performance metric used for comparison is the throughput achieved at a given area coverage probability. Specifically, we study techniques such as decode-and-forward, amplify-and-forward, and pipelining and compare their performances through simulations in the context of a cellular network with relays and show that significant performance benefits are obtained through such schemes

In very large networks with heavy traffic, congestion control plays an important role in network resource management. One approach to this is the active queue management (AQM) algorithms. Many AQM algorithms have been proposed and analyzed but they mainly focus on single queued links. Recognizing the fact that input queued switches are limited in throughput and output queued switches require a large speedup factor, we direct our attention to combined input and output queued (CIOQ) switches. We propose a simple modification to the RED AQM algorithm in order to account for the presence of both input and output queues in the switch. Specifically we use the weighted sum of input and output queue lengths as the congestion measure instead of just the output queue length. Simulations show that with such a simple modification, the average backlog in the switch is significantly reduced in the low speedup region as compared to RED without this modification. Unlike the traditional dynamic of having the loss rate grow with the length of the queue, simulations show that for a loss rate in the modified RED slightly larger than that in RED, the output queue length in modified RED is tremendously reduced. The weighting factor used in the computation of the congestion measure provides a means to balance the reduction in the average backlog on the one hand, and the increase in the loss rate on the other hand. Finally, simulations show that the improvement gained in terms of the queue length does not compromise in any way the utilization of the switch as compared to RED and Droptail.