Multiple Transmission Control Protocol Connections Research Articles

Data sequence signal manipulation in multipath TCP (MPTCP): The vulnerability, attack and its detection

Multipath Transmission Control Protocol (MPTCP) is an innovative next-generation transport protocol standardized by the Internet Engineering Task Force (IETF) to overcome the single path limitation of the Transmission Control Protocol (TCP). MPTCP augments TCP with a new set of signaling options for seamless transmission and reception of application data across multiple interlinked TCP connections called subflows. In this paper, we focus on a new security concern associated with the signal exchanging process of MPTCP. To the best of our knowledge, for the first time, this paper exposes that MPTCP signal exchange scheme is vulnerable to a sophisticated packet spoofing technique, which we name as Data Sequence Signal (DSS) manipulation. We implement the vulnerability, create attack scenarios in Linux Kernel and conduct experiments over emulated testbed to demonstrate the existence of the vulnerability and means of exploiting it for powerful attacks. Our results show that DSS manipulation can be tactically exploited, on top of TCP optimistic ACKing, to generate non-responsive traffic like Denial-of-Service (DoS) attack flood. Particularly, we demonstrate two new adverse scenarios, where a MPTCP sender is forced to: (a) transmit at a rate significantly higher than the bottleneck link bandwidth, and (b) induce high intensity and harmful packet bursts at line-rate called Maliciously-induced-Bursts (MiBs). We also show that the non-responsive traffic resulting from the attack can suppress genuine congestion controlled traffic to the extent of causing DoS attack. We capture and analyze the dynamics of important MPTCP parameters, like send buffer occupancy of meta and subflow sockets, congestion window and flightsize to highlight the attack impact. DSS manipulation originates from a fundamental protocol design limitation rather than from any implementation flaw. We also propose a novel technique called data sequence map skipping for detection and countermeasure against DSS manipulation based attacks.

BATCP: Bandwidth-Aggregation Transmission Control Protocol

The Transmission Control Protocol (TCP) is the most used transport protocol to exchange reliable data between network devices. A considerable number of extensions have been implemented into TCP to achieve better performance. In this paper, we will present, describe, implement, and analyze a new protocol extension called Bandwidth-Aggregation TCP (BATCP), which enables the concurrent use of network interfaces, to improve network performance on multi-homed nodes. BATCP allows the use of multiple TCP connections to accept multiple IP addresses from a multi-homed node, scheduling segments among them based on a scheduling algorithm. Our results show that BATCP achieves full exploitation of each network interface, achieving up to 100 % network utilization using two ADSL connections in real-world scenarios. MultiPath TCP (MPTCP) is currently being standardized, and achieves up to 96 % of network utilization when in ideal conditions. BATCP and MPTCP are the only protocols tested on real-world scenarios. Related work such as the Proxy Inverse Multiplexer, called PRISM, and bandwidth aggregation with Stream Control Transmission Protocol (SCTP) achieve 80 % utilization or less with network simulators.

Parallel TCP and scalable video coding for jitter free video transmission over MIMO wireless networks

There is a significant rise in demand for video transmission over 3G and 4G wireless networks due to the rising popularity of video streaming websites such as YouTube. The market for video streaming over wireless networks is expected to increase sharply in the future. Both of the two basic transport layer protocols without modifications are not suited for video transmission over wireless networks. UDP (user datagram protocol) suffers from inherent unreliability, resulting in corrupted video due to frequent corruption of packets. Inherent features of wireless networks such as noise, interference, etc. result in packet corruption. On the other hand, the performance of TCP (transmission control protocol) is worse than UDP (Thangaraj et al. in Telecommun Syst 45(4):303---312, 2010) because of frequently corrupted packets. Due to its reliable data transfer feature, TCP continuously retransmits the corrupted packet until successful reception at the receiver. This leads to jitter in video playback and poor end user quality of experience. Multiple TCP connections with appropriate optimization can lead to an increased efficiency of bandwidth utilization in comparison to single TCP based video transmission over wireless networks. It has been shown that multiple TCP connections enhance the video transmission and playback experience by providing reliable communication. The parallel TCP scheme proposed in this paper enhances the quality of video transmission and playback experience over MIMO wireless networks employing scalable hierarchical wavelet decomposition based video encoding with multiple TCP connections.

Congestion-aware quality-adaptive streaming of scalable video

Internet video streaming is a hot topic in multimedia systems. A large variety of devices (computers, mobile phones, TVs, etc.) are connected to the Internet via wired or wireless networks and are capable of receiving and decoding HD video content. To enable new services like HD video streaming (e.g., online video rental), the Internet’s infrastructure was enhanced. But the Internet is still a best-effort network, which does not implement quality-of-service or admission control, resulting in time-varying bandwidth and packet delay, packet loss and network congestion. Because video streaming accounts for a considerable amount of the Internet’s traffic, video streaming needs additionally to be congestion-aware, to avoid a congestion collapse of the Internet. The Transmission Control Protocol (TCP) can adapt to changing network conditions and is currently the de facto standard protocol for congestion-aware and reliable data transmission in the Internet. This fact gave TCP-based video streaming a huge momentum. Consequently, this thesis investigates TCP-based adaptive video streaming for the Internet. The main goal is to provide a solution for congestion-aware video streaming, while still being able to achieve a reasonable performance in error-prone networks. To complement existing work on congestion-aware adaptive streaming, this thesis makes six contributions. (1) The baseline performance of TCP-based adaptive streaming is identified by means of an evaluation of different adaptive streaming approaches. The results represent a reference for further investigations. (2) An investigation on the influence of TCP’s behavior in presence of packet loss on the video streaming performance. (3) To overcome the shortcomings of TCP-based video streaming (single TCP connections fail to deliver a good performance in case of packet loss), a new approach to video streaming based on multiple request-response streams was introduced. The novelty of this system is that it is able to make use of multiple HTTP-based request-response streams while still providing TCP-friendliness. (4) A performance model of the HTTP-based request-response streams was developed, to estimate the influence of the system parameters and the network characteristics on the throughput performance. (5) A comprehensive evaluation of the HTTP-based request-response streams under diverse network conditions was conducted, to validate the model’s estimations. Additionally, the TCP-friendliness was evaluated, showing that request-response streaming systems can be configured to achieve TCP-friendliness. (6) A cellular network with high bandwidth fluctuations and RTTs was used to investigate the performance of the request-response streaming system in a mobile video streaming scenario. The results indicate that the streaming system can make good use of the available bandwidth, while the number of quality switches is kept low. While aggregating multiple TCP connections to improve the TCP streaming performance is quite common, usually the improvement comes at the cost of high deployment effort. By placing the streaming logic at the client, request-response streams can avoid this complexity. Additionally, this client-driven approach responds faster to changing network conditions and enables easy recovery from connection stalls or aborts, because the control loop is at the client. To improve the network efficiency and the scalability in terms of number of clients served, HTTP-based request-response streams can utilize HTTP proxies and caches.

Adaptive Transmission Control Protocol‐trunking flow control mechanism for supporting proxy‐assisted video on demand system

SUMMARYVideo streaming has been challenged by the best‐effort service inherently provided by the Internet. Transmission Control Protocol (TCP) is sensitive to traffic congestion, and current TCP is insufficient to offer stable and high throughput bandwidth for video transmissions. Although many investigations have been proposed to deal with the inefficient problem of TCP protocol, such as Stream Control Transmission Protocol, these mechanisms and protocols are still not well employed by current Internet and used by Hypertext Transfer Protocol (HTTP). Adaptive TCP‐trunking flow control (ATCP+), which is a segment‐based flow control scheme built upon TCP protocol, is proposed in the paper. ATCP+ dynamically adjusts the transmitted segment size and makes multiple TCP connections be trunked together based on network condition, in order to optimize all TCP connection throughputs efficiently. The proposed scheme is implemented through HTTP 1.1. Experiments show that the proposed method can achieve stable throughput for smooth multimedia presentation on numerous network states, and the scheme can be handily applied on general web servers. Lastly, ATCP+ presents a cost‐effective video streaming solution in real deployment. Copyright © 2011 John Wiley & Sons, Ltd.

TCP-ROME: A transport-layer parallel streaming protocol for real-time online multimedia environments

Real-time multimedia streaming over the Internet is rapidly increasing with the popularity of user-created contents, Web 2.0 trends, and P2P (peer-to-peer) delivery support. While many homes today are broadband-enabled, the quality of experience (QoE) of a user is still limited due to frequent interruption of media playout. The vulnerability of TCP (transmission control protocol), the popular transport-layer protocol for streaming in practice, to the packet losses, retransmissions, and timeouts makes it hard to deliver a timely and persistent flow of packets for online multimedia contents. This paper presents TCP-real-time online multimedia environment (ROME), a novel transport-layer framework that allows the establishment and coordination of multiple many-to-one TCP connections. Between one client with multiple home addresses and multiple co-located or distributed servers, TCP-ROME increases the total throughput by aggregating the re- sources of multiple TCP connections. It also overcomes the band- width fluctuations of network bottlenecks by dynamically coordinating the streams of contents from multiple servers and by adapting the streaming rate of all connections to match the bandwidth requirement of the target video.

An adaptive mechanism to guarantee the bandwidth fairness of TCP flows

End-to-end TCP (transmission control protocol) congestion control can cause unfairness among multiple TCP connections with different RTT (Round Trip Time). The throughput of TCP connection is inversely proportional to its RTT. To resolve this problem, researchers have proposed many methods. The existing proposals for RTT-aware conditioner work well when congestion level is low. However, they over-protect long RTT flows and starve short RTT flows when congestion level is high. Due to this reason, an improved method based on adaptive thought is proposed. According to the congestion level of networks, the mechanism can adaptively adjust the degree of the protection to long RTT flows. Extensive simulation experiments showed that the proposed mechanism can guarantee the bandwidth fairness of TCP flows effectively and outperforms the existing methods.