De-ossifying the Internet Transport Layer : A Survey and Future Perspectives

ACKNOWLEDGMENT The authors would like to thank the anonymous reviewers for their useful suggestions and comments.Peer reviewedPublisher PD

Congestion Control using FEC for Conversational Multimedia Communication

In this paper, we propose a new rate control algorithm for conversational multimedia flows. In our approach, along with Real-time Transport Protocol (RTP) media packets, we propose sending redundant packets to probe for available bandwidth. These redundant packets are Forward Error Correction (FEC) encoded RTP packets. A straightforward interpretation is that if no losses occur, the sender can increase the sending rate to include the FEC bit rate, and in the case of losses due to congestion the redundant packets help in recovering the lost packets. We also show that by varying the FEC bit rate, the sender is able to conservatively or aggressively probe for available bandwidth. We evaluate our FEC-based Rate Adaptation (FBRA) algorithm in a network simulator and in the real-world and compare it to other congestion control algorithms

End-to-end mobility for the internet using ILNP

This work was partially funded by the Government of Thailand through a PhD scholarship for Dr Phoomikiattisak.As the use of mobile devices and methods of wireless connectivity continue to increase, seamless mobility becomes more desirable and important. The current IETF Mobile IP standard relies on additional network entities for mobility management, can have poor performance, and has seen little deployment in real networks. We present a host-based mobility solution with a true end-to-end architecture using the Identifier-Locator Network Protocol (ILNP). We show how the TCP code in the Linux kernel can be extended allowing legacy TCP applications that use the standard C sockets API to operate over ILNP without requiring changes or recompilation. Our direct testbed performance comparison shows that ILNP provides better host mobility support than Mobile IPv6 in terms of session continuity, packet loss, and handoff delay for TCP.Publisher PDFPeer reviewe

Managing Network Delay for Browser Multiplayer Games

Latency is one of the key performance elements affecting the quality of experience (QoE) in computer games. Latency in the context of games can be defined as the time between the user input and the result on the screen. In order for the QoE to be satisfactory the game needs to be able to react fast enough to player input. In networked multiplayer games, latency is composed of network delay and local delays. Some major sources of network delay are queuing delay and head-of-line (HOL) blocking delay. Network delay in the Internet can be even in the order of seconds. In this thesis we discuss what feasible networking solutions exist for browser multiplayer games. We conduct a literature study to analyze the Differentiated Services architecture, some salient Active Queue Management (AQM) algorithms (RED, PIE, CoDel and FQ-CoDel), the Explicit Congestion Notification (ECN) concept and network protocols for web browser (WebSocket, QUIC and WebRTC). RED, PIE and CoDel as single-queue implementations would be sub-optimal for providing low latency to game traffic. FQ-CoDel is a multi-queue AQM and provides flow separation that is able to prevent queue-building bulk transfers from notably hampering latency-sensitive flows. WebRTC Data-Channel seems promising for games since it can be used for sending arbitrary application data and it can avoid HOL blocking. None of the network protocols, however, provide completely satisfactory support for the transport needs of multiplayer games: WebRTC is not designed for client-server connections, QUIC is not designed for traffic patterns typical for multiplayer games and WebSocket would require parallel connections to mitigate the effects of HOL blocking

End-to-end security in active networks

Active network solutions have been proposed to many of the problems caused by the increasing heterogeneity of the Internet. These ystems allow nodes within the network to process data passing through in several ways. Allowing code from various sources to run on routers introduces numerous security concerns that have been addressed by research into safe languages, restricted execution environments, and other related areas. But little attention has been paid to an even more critical question: the effect on end-to-end security of active flow manipulation. This thesis first examines the threat model implicit in active networks. It develops a framework of security protocols in use at various layers of the networking stack, and their utility to multimedia transport and flow processing, and asks if it is reasonable to give active routers access to the plaintext of these flows. After considering the various security problem introduced, such as vulnerability to attacks on intermediaries or coercion, it concludes not. We then ask if active network systems can be built that maintain end-to-end security without seriously degrading the functionality they provide. We describe the design and analysis of three such protocols: a distributed packet filtering system that can be used to adjust multimedia bandwidth requirements and defend against denial-of-service attacks; an efficient composition of link and transport-layer reliability mechanisms that increases the performance of TCP over lossy wireless links; and a distributed watermarking servicethat can efficiently deliver media flows marked with the identity of their recipients. In all three cases, similar functionality is provided to designs that do not maintain end-to-end security. Finally, we reconsider traditional end-to-end arguments in both networking and security, and show that they have continuing importance for Internet design. Our watermarking work adds the concept of splitting trust throughout a network to that model; we suggest further applications of this idea

Protocols and Algorithms for Adaptive Multimedia Systems

The deployment of WebRTC and telepresence systems is going to start a wide-scale adoption of high quality real-time communication. Delivering high quality video usually corresponds to an increase in required network capacity and also requires an assurance of network stability. A real-time multimedia application that uses the Real-time Transport Protocol (RTP) over UDP needs to implement congestion control since UDP does not implement any such mechanism. This thesis is about enabling congestion control for real-time communication, and deploying it on the public Internet containing a mixture of wired and wireless links. A congestion control algorithm relies on congestion cues, such as RTT and loss. Hence, in this thesis, we first propose a framework for classifying congestion cues. We classify the congestion cues as a combination of: where they are measured or observed? And, how is the sending endpoint notified? For each there are two options, i.e., the cues are either observed and reported by an in-path or by an off-path source, and, the cue is either reported in-band or out-of-band, which results in four combinations. Hence, the framework provides options to look at congestion cues beyond those reported by the receiver. We propose a sender-driven, a receiver-driven and a hybrid congestion control algorithm. The hybrid algorithm relies on both the sender and receiver co-operating to perform congestion control. Lastly, we compare the performance of these different algorithms. We also explore the idea of using capacity notifications from middleboxes (e.g., 3G/LTE base stations) along the path as cues for a congestion control algorithm. Further, we look at the interaction between error-resilience mechanisms and show that FEC can be used in a congestion control algorithm for probing for additional capacity. We propose Multipath RTP (MPRTP), an extension to RTP, which uses multiple paths for either aggregating capacity or for increasing error-resilience. We show that our proposed scheduling algorithm works in diverse scenarios (e.g., 3G and WLAN, 3G and 3G, etc.) with paths with varying latencies. Lastly, we propose a network coverage map service (NCMS), which aggregates throughput measurements from mobile users consuming multimedia services. The NCMS sends notifications to its subscribers about the upcoming network conditions, which take these notifications into account when performing congestion control. In order to test and refine the ideas presented in this thesis, we have implemented most of them in proof-of-concept prototypes, and conducted experiments and simulations to validate our assumptions and gain new insights.