Future 5G systems will need to support ultra-reliable low-latency communications scenarios. From a latency-reliability viewpoint, it is inefficient to rely on average utility-based system design. Therefore, we introduce the notion of guaranteeable delay which is the average delay plus three standard deviations of the mean. We investigate the trade-off between guaranteeable delay and throughput for point-to-point wireless erasure links with unreliable and delayed feedback, by bringing together signal flow techniques to the area of coding. We use tiny codes, i.e. sliding window by coding with just 2 packets, and design three variations of selective-repeat ARQ protocols, by building on the baseline scheme, i.e. uncoded ARQ, developed by Ausavapattanakun and Nosratinia: (i) Hybrid ARQ with soft combining at the receiver; (ii) cumulative feedback-based ARQ without rate adaptation; and (iii) Coded ARQ with rate adaptation based on the cumulative feedback. Contrasting the performance of these protocols with uncoded ARQ, we demonstrate that HARQ performs only slightly better, cumulative feedback-based ARQ does not provide significant throughput while it has better average delay, and Coded ARQ can provide gains up to about 40 in terms of throughput. Coded ARQ also provides delay guarantees, and is robust to various challenges such as imperfect and delayed feedback, burst erasures, and round-trip time fluctuations. This feature may be preferable for meeting the strict end-to-end latency and reliability requirements of future use cases of ultra-reliable low-latency communications in 5G, such as mission-critical communications and industrial control for critical control messaging.
Network RTT can be estimated using delay measurements reported from the receiver's acknowledgments @cite_32 . RTT can also be estimated using traffic already flowing between two points without requiring precise time synchronization between each point @cite_33 . However, measuring the time-varying RTT actually experienced by an application is a substantial challenge, particularly when the RTT fluctuates more frequently than one can sample the path. The unpredictability of RTT might have massive effects on upper layer protocols. Therefore, it is required to design robust protocols which are more predictable across statistics, hence are more stable when RTT is unreliable.
@cite_32 @cite_33
2050600889 2070712461
Legacy congestion controls including TCP and its variants are known to perform poorly over cellular networks due to highly variable capacities over short time scales, self-inflicted packet delays, and packet losses unrelated to congestion. To cope with these challenges, we present Verus, an end-to-end congestion control protocol that uses delay measurements to react quickly to the capacity changes in cellular networks without explicitly attempting to predict the cellular channel dynamics. The key idea of Verus is to continuously learn a delay profile that captures the relationship between end-to-end packet delay and outstanding window size over short epochs and uses this relationship to increment or decrement the window size based on the observed short-term packet delay variations. While the delay-based control is primarily for congestion avoidance, Verus uses standard TCP features including multiplicative decrease upon packet loss and slow start. Through a combination of simulations, empirical evaluations using cellular network traces, and real-world evaluations against standard TCP flavors and state of the art protocols like Sprout, we show that Verus outperforms these protocols in cellular channels. In comparison to TCP Cubic, Verus achieves an order of magnitude (> 10x) reduction in delay over 3G and LTE networks while achieving comparable throughput (sometimes marginally higher). In comparison to Sprout, Verus achieves up to 30 higher throughput in rapidly changing cellular networks. Accurate and frequent round trip time (RTT) measurements are important in testbeds and operational networks. Active measurement techniques inject probe packets that may modify the behaviour of the observed network and may produce misleading RTT estimates if the network handles probe packets differently to regular packets. Previous passive measurement techniques address these issues, but require precise time synchronisation or are limited to certain traffic types. We introduce Synthetic Packet-Pairs (SPP), a novel passive technique for RTT measurement. SPP provides frequently updated RTT measurements using any network traffic already present in the network without the need for time synchronisation. SPP accurately measures the RTT experienced by any application's traffic, even applications that do not exhibit symmetric client-server packet exchanges. We experimentally demonstrate the advantages of SPP.