The QUIC Fix for Optimal Video Streaming

Within a few years of its introduction, QUIC has gained traction: a significant chunk of traffic is now delivered over QUIC. The networking community is actively engaged in debating the fairness, performance, and applicability of QUIC for various use cases, but these debates are centered around a narrow, common theme: how does the new reliable transport built on top of UDP fare in different scenarios? Support for unreliable delivery in QUIC remains largely unexplored. The option for delivering content unreliably, as in a best-effort model, deserves the QUIC designers' and community's attention. We propose extending QUIC to support unreliable streams and present a simple approach for implementation. We discuss a simple use case of video streaming---an application that dominates the overall Internet traffic---that can leverage the unreliable streams and potentially bring immense benefits to network operators and content providers. To this end, we present a prototype implementation that, by using both the reliable and unreliable streams in QUIC, outperforms both TCP and QUIC in our evaluations.Comment: Published to ACM CoNEXT Workshop on the Evolution, Performance, and Interoperability of QUIC (EPIQ

Post Sockets: Towards an Evolvable Network Transport Interface

The traditional Sockets API is showing its age, and no longer provides effective support for modern networked applications. This has led to a proliferation of non-standard extensions, alternative APIs, and workarounds that enable new features and allow applications to make good use of the network, but are difficult to use, and require expert knowledge that is not widespread. In this paper, we present Post Sockets, a proposed new standard network API, that is designed to support modern network transport protocols and features, while raising the level of abstraction and enhancing usability. Specifically, Post Sockets aims to give portable applications the ability to use a clear, messages based, interface to multi-path and multi-stream transports, rendezvous and connection racing, and fast connection re-establishment

Using Transcoding for Hidden Communication in IP Telephony

The paper presents a new steganographic method for IP telephony called TranSteg (Transcoding Steganography). Typically, in steganographic communication it is advised for covert data to be compressed in order to limit its size. In TranSteg it is the overt data that is compressed to make space for the steganogram. The main innovation of TranSteg is to, for a chosen voice stream, find a codec that will result in a similar voice quality but smaller voice payload size than the originally selected. Then, the voice stream is transcoded. At this step the original voice payload size is intentionally unaltered and the change of the codec is not indicated. Instead, after placing the transcoded voice payload, the remaining free space is filled with hidden data. TranSteg proof of concept implementation was designed and developed. The obtained experimental results are enclosed in this paper. They prove that the proposed method is feasible and offers a high steganographic bandwidth. TranSteg detection is difficult to perform when performing inspection in a single network localisation.Comment: 17 pages, 16 figures, 4 table

Throughput of ADSL modems

This paper considers the throughput of ADSL (Asymmetric Digital Subscriber Line) modems, used for high-speed data transmission over relatively unreliable connections, e.g. copper telephone wires. The modem technique uses an error correcting code and interleaving. The settings include a grouping factor $S$ which affects the amount of data per code word, the number $R$ of redundant bytes per code word and the interleave depth $D$. The influence of these parameters on both the effective data transmission rate and the resulting error rate in the received signal are determined for two error situations: random errors and bursts of errors. An approximate analysis for the random error case of the throughput of a TCP (Transport Control Protocol) connection using an ADSL modem shows that maximum throughput is obtained for the highest values of $S$ and $R$

The Xpress Transfer Protocol (XTP): A tutorial (short version)

The Xpress Transfer Protocol (XTP) is a reliable, light weight transfer layer protocol. Current transport layer protocols such as DoD's Transmission Control Protocol (TCP) and ISO's Transport Protocol (TP) were not designed for the next generation of high speed, interconnected reliable networks such as fiber distributed data interface (FDDI) and the gigabit/second wide area networks. Unlike all previous transport layer protocols, XTP is being designed to be implemented in hardware as a VLSI chip set. By streamlining the protocol, combining the transport and network layers, and utilizing the increased speed and parallelization possible with a VLSI implementation, XTP will be able to provide the end-to-end data transmission rates demanded in the high speed networks without compromising reliability and functionality. This tutorial briefly describes the operation of the XTP protocol and in particular, its error, flow and rate control; inter-networking addressing mechanisms; and multicast support features, as defined in the XTP Protocol Definition Revision 3.4