异构移动网络中基于可达性的多路径QUIC分组调度

Riki Okunishi, R. Teng, Kenya Sato
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引用次数: 0

摘要

近年来,我们需要更多的带宽来享受视频流、音乐和在线游戏等娱乐内容。为了获得足够的带宽,需要同时使用多个接口来组合带宽的技术。多路径传输协议是一种很有前途的技术,它在传输层将多条路径组合在一起进行分组传输。这样的协议有一种机制,叫做“包调度程序”,用来选择发送包的接口。然而,现有的分组调度研究并没有明确地考虑移动性与带宽绑定的兼容性。因此,当智能手机用户离开无线局域网(LAN)和长期演进(LTE)等通信网络的覆盖范围时,就会出现丢包现象,从而导致吞吐量下降。在本研究中,我们提出了一个数据包调度程序,它可以选择合适的通信路径,以便数据包在移动到覆盖范围之外之前到达对等端。根据车辆的路线和接入点的位置和通信范围,预测通信路径丢失的时间。此外,我们采用MPQUIC (Multipath QUIC (Quick UDP Internet Connections))作为QUIC协议的扩展而提出的多路径传输协议,以减少多路径通信中ACK数据包的丢失,并缩短重新开始传输的时间。我们使用模拟器评估了丢包数、吞吐量和重新开始传输的时间,并证明了所提出方法的优越性。
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Reachability-Based Packet Scheduler of Multipath QUIC for Heterogeneous Mobile Networks
In recent years, we need more bandwidth to enjoy entertainment contents such as video streaming, music and online gaming. To gain enough bandwidth, technologies that combine bandwidth by using multiple interfaces at same time are desired. Multipath transport protocols which combine multiple paths for packet delivery at the transport layer are a promising technology. Such protocols have a mechanism, called “packet scheduler”, to select the interface to send a packet. However, existing studies of the packet scheduler have not explicitly considered the compatibility of mobility with bonding of bandwidth. Therefore, when smartphone users move out of coverage of communication networks such as wireless Local Area Network (LAN) and Long Term Evolution (LTE) by vehicle, packet loss occurs, leading to a decrease of throughput. In this study, we propose a packet scheduler that selects an appropriate communication path so that packets can reach the peer before it moves out of the coverage. Based on routes of a vehicle and the position and communication range of the access point, the time at which a communication path will be lost is predicted. In addition, we employ MPQUIC (Multipath QUIC (Quick UDP Internet Connections)), which is a multipath transport protocol proposed as the extension of QUIC protocol, to reduce the ACK packet loss in multipath communication, and to reduce the time until the starts of retransmission. We evaluated the number of packet losses, the throughput and the time until starts of retransmission using a simulator and show the superiority of proposed method.
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