<p>One of the most parts of VOIP management is fault management and, in having a good fault management, finding good mechanisms to detect faults in the network have to be considered.</p><p>The main focus of this project is to implement different types of fast fault recovery protocols in networks<strong>,</strong> especially networks that carry IP telephony. Having a complete understanding of some common link failure detection and fault recovery protocols<strong>,</strong> such as spanning tree protocol (STP), rapid spanning tree protocol (RSTP) and per-VLAN spanning tree protocol (PVSTP)<strong>,</strong> and also having a complete understanding of three other common techniques for fault detection and fault recovery, such as hot standby routing protocol (HSRP), virtual router redundancy protocol (VRRP) and gateway load balancing protocol (GLBP) will be regarded in the project. We are going to test some fault recovery protocols which can be used in IP telephony networks and choose the best. We intend to focus on this issue in LAN environment in theoretical descriptions and practical implementations.</p><p>The final outcome of the thesis is implementation in the Halmstad University’s lab environment to obtain the final result. For doing our thesis, we are going to use some technical tools as hardware tools (Cisco L3 and L2 switches, Routers, IP Phones) and tools which are used for network performance monitoring<strong>,</strong> like as CommVeiw.</p>

Over the past 15 years, the Internet has proven itself to be one of the most influential inventions that humankind has ever conceived. The success of the Internet can be largely attributed to its stability and ease of access. Among the various pieces of technologies that constitute the Internet, TCP/IP can be regarded as the cornerstone to the Internet’s impressive scalability and stability. Many researchers have been and are currently actively engaged in the studies on the optimization of TCP’s performance in various network environments. This thesis presents an alternative transport layer protocol called RRTP, which is designed to provide reliable transport layer services to software applications. The motivation for this work comes from the fact that the most commonly used versions of TCP perform unsatisfactorily when they are deployed over non-conventional network platforms such as cellular/wireless, satellite, and long fat pipe networks. These non-conventional networks usually have higher network latency and link failure rate as compared with the conventional wired networks and the classic versions of TCP are unable to adapt to these characteristics. This thesis attempts to address this problem by introducing a user-level, reliable, and reconfigurable transport layer protocol that runs on top of UDP and appropriately tends to the characteristics of non-conventional networks that TCP by default ignores. A novel aspect of RRTP lies in identifying three key characteristic parameters of a network to optimize its performance. The single most important contribution of this work is its empirical demonstration of the fact that parameter-based, user-configurable, flow-control and congestion-control algorithms are highly effective at adapting to and fully utilizing various networks. This fact is demonstrated through experiments designed to benchmark the performance of RRTP against that of TCP on simulated as well as real-life networks. The experimental results indicate that the performance of RRTP consistently match and exceed TCP’s performance on all major network platforms. This leads to the conclusion that a user-level, reliable, and reconfigurable transport-layer protocol, which possesses the essential characteristics of RRTP, would serve as a viable replacement for TCP over today’s heterogeneous network platforms.

Over the past 15 years, the Internet has proven itself to be one of the most influential inventions that humankind has ever conceived. The success of the Internet can be largely attributed to its stability and ease of access. Among the various pieces of technologies that constitute the Internet, TCP/IP can be regarded as the cornerstone to the Internet’s impressive scalability and stability. Many researchers have been and are currently actively engaged in the studies on the optimization of TCP’s performance in various network environments. This thesis presents an alternative transport layer protocol called RRTP, which is designed to provide reliable transport layer services to software applications. The motivation for this work comes from the fact that the most commonly used versions of TCP perform unsatisfactorily when they are deployed over non-conventional network platforms such as cellular/wireless, satellite, and long fat pipe networks. These non-conventional networks usually have higher network latency and link failure rate as compared with the conventional wired networks and the classic versions of TCP are unable to adapt to these characteristics. This thesis attempts to address this problem by introducing a user-level, reliable, and reconfigurable transport layer protocol that runs on top of UDP and appropriately tends to the characteristics of non-conventional networks that TCP by default ignores. A novel aspect of RRTP lies in identifying three key characteristic parameters of a network to optimize its performance. The single most important contribution of this work is its empirical demonstration of the fact that parameter-based, user-configurable, flow-control and congestion-control algorithms are highly effective at adapting to and fully utilizing various networks. This fact is demonstrated through experiments designed to benchmark the performance of RRTP against that of TCP on simulated as well as real-life networks. The experimental results indicate that the performance of RRTP consistently match and exceed TCP’s performance on all major network platforms. This leads to the conclusion that a user-level, reliable, and reconfigurable transport-layer protocol, which possesses the essential characteristics of RRTP, would serve as a viable replacement for TCP over today’s heterogeneous network platforms.

One of the most parts of VOIP management is fault management and, in having a good fault management, finding good mechanisms to detect faults in the network have to be considered. The main focus of this project is to implement different types of fast fault recovery protocols in networks, especially networks that carry IP telephony. Having a complete understanding of some common link failure detection and fault recovery protocols, such as spanning tree protocol (STP), rapid spanning tree protocol (RSTP) and per-VLAN spanning tree protocol (PVSTP), and also having a complete understanding of three other common techniques for fault detection and fault recovery, such as hot standby routing protocol (HSRP), virtual router redundancy protocol (VRRP) and gateway load balancing protocol (GLBP) will be regarded in the project. We are going to test some fault recovery protocols which can be used in IP telephony networks and choose the best. We intend to focus on this issue in LAN environment in theoretical descriptions and practical implementations. The final outcome of the thesis is implementation in the Halmstad University’s lab environment to obtain the final result. For doing our thesis, we are going to use some technical tools as hardware tools (Cisco L3 and L2 switches, Routers, IP Phones) and tools which are used for network performance monitoring, like as CommVeiw.

Zone Routing Protocol (ZRP) is a typical hybrid routing protocol used in Mobile Ad Hoc Networks (MANETs). Hybrid routing protocols are especially suitable for dynamic environments because they combine the best features of proactive and reactive routing protocols. The Gossip-based Zone Routing Protocol (GZRP) uses a gossip scheme, in which the node forwards a packet to some nodes instead of all nodes to further reduce the control overhead. However, GZRP does not perform well when the network includes articulation points since packets will be lost if an articulation node happens not to forward the packet or nodes happen not to forward packets to the articulation point. To raise the packet delivery ratio, the gossip probability of articulation points must be set to 1 and the packets to be forwarded must be sent to the articulation points in peripheral nodes. Accordingly, how to identify articulation nodes in the network becomes a critical issue. This paper proposes an effective scheme, called articulation point detection (APD), to find the articulation points. Simulation results show that the proposed APD-GZRP (GZRP with articulation point detection) can improve the packet delivery ratio and reduce both the control overhead and power consumption.

The objective of this research is focused on two of the most important privacy-preserving techniques: privacy-preserving element matching protocols and privacy-preserving credential authentication protocols, where an element represents the information generated by users themselves and a credential represents a group membership assigned from an independent central authority (CA). The former is also known as private set intersection (PSI) protocol and the latter is also known as secret handshake (SH) protocol. In this dissertation, I present a general framework for design of efficient and secure PSI and SH protocols based on similar message exchange and computing procedures to confirm “commonality” of their exchanged information, while protecting the information from each other when the commonalty test fails. I propose to use the homomorphic randomization function (HRF) to meet the privacy-preserving requirements, i.e., common element/credential can be computed efficiently based on homomorphism of the function and uncommon element/credential are difficult to derive because of the randomization of the same function. Based on the general framework two new PSI protocols with linear computing and communication cost are proposed. The first protocol uses full homomorphic randomization function as the cryptographic basis and the second one uses partial homomorphic randomization function. Both of them achieve element confidentiality and private set intersection. A new SH protocol is also designed based on the framework, which achieves unlinkability with a reusable pair of credential and pseudonym and least number of bilinear mapping operations. I also propose to interlock the proposed PSI protocols and SH protocol to design new protocols with new security properties. When a PSI protocol is executed first and the matched elements are associated with the credentials in a following SH protocol, authenticity is guaranteed on matched elements. When a SH protocol is executed first and the verified credentials is used in a following PSI protocol, detection resistance and impersonation attack resistance are guaranteed on matching elements. The proposed PSI and SH protocols are implemented to provide privacy-preserving inquiry matching service (PPIM) for social networking applications and privacy-preserving correlation service (PAC) of network security alerts. PPIM allows online social consumers to find partners with matched inquiries and verified group memberships without exposing any information to unmatched parties. PAC allows independent network alert sources to find the common alerts without unveiling their local network information to each other.

Many applications in modern science need to transmit extremely massive amount of data over wide area networks. These data usually do not need stringent real-time requirements but require large bandwidth to finish transmission with unreasonable time. High-energy physics experiments and climate modeling and analysis are typical examples of such applications. As TCP is known to perform inefficiently over networks of large delay-bandwidth product, efficient transmission of this kind of massive, non-real-time data has been heavily studied in the past. The previous results work well in dedicated networks but will compete for fair share of bandwidth with normal TCP connections if they operate in the public networks. The objective of this thesis is to design a new transmission protocol for the above applications that can operate in the public networks without affecting normal TCP connections. The new protocol is called Rate Control Transmission Protocol (RCTP). The idea is to apply the packet-pair measurement technique to measure the bandwidth share in the network for the transmission. The sending rate is based on that measurement and is precisely compensated by the RTT variance measurement. Due to the RTT compensation, RCTP can efficiently utilize the unused bandwidth in the network while not affecting the normal TCP transmissions, making it perfect for transmitting massive, non-real-time data in the public networks.

The congestion control mechanisms of Transmission Control Protocol (TCP) are very effective in providing best effort service in wired networks, where packet losses are mainly due to congestion in the network. In wireless mobile networks, more often than not, loss of packets is because of corruption of data on the wireless link. The TCP sender responds to these losses as if they are due to congestion, by reducing its congestion window, thereby reducing the rate of flow of packets. The reduction in congestion window is a necessity when network is experiencing congestion to avoid congestion collapse but it is not required if packet losses occur due to corruption of data on the wireless link. This unnecessary reduction in congestion window for corruption losses is the main reason for poor throughput of data transfer in wireless networks. The reduction in congestion window for corruption losses can be avoided if TCP can successfully differentiate between packet losses due to congestion and corruption. We suggest enhancements to TCP that, if implemented, will help the TCP receiver in separately identifying corruption losses and congestion losses. The enhancements are suggested over and above standard TCP NewReno and we call this new scheme as "NewRenoEln (NewReno with Explicit Loss Notification)". We suggest that the TCP sender attach a separate checksum for the TCP header with the packet. Since the length of the TCP header is much smaller as compared to the length of the TCP packet, there is a large probability that the TCP receiver will receive the header portion of the TCP packet without error even if the data portion of the packet is corrupted. Once the header information is found to be correct for a corrupted packet, the receiver can generate reliable Explicit 5oss Notification (ELN) for the sender. We derive an expression for the probability of a receiver generating successful Explicit Loss Notification, assuming a generic link layer protocol that is used for data transfer over wireless link. With this analysis, we show that there is large probability that receiver will generate successful ELN for various channel conditions We also suggest modifications to the sender behavior on receiving successful Explicit Loss Notification from the receiver. With these modifications, the TCP sender will recover from corruption losses without any reduction in congestion window. There is also a need to develop a unified analytical approach for the evaluation of TCP performance. We develop an analytical approach for the performance evaluation of NewRenoEln scheme. We compare the throughput results obtained by analytical calculations with results obtained by simulation and find them to be very close to each other. We also compare the performance of the proposed scheme NewRenoEln and the standard NewReno TCP via simulation as well as analytical approach, and find considerable improvement in throughput over wireless links.

Thesis (M.S. in Computer Science)--Naval Postgraduate School, March 2003. / Thesis advisor(s): Ted Lewis, Geoffrey Xie, Gurminder Singh. Includes bibliographical references (p. 89). Also available online.

Thesis (M.S.)--Worcester Polytechnic Institute. / Keywords: TCP; RED; satellite; PEP; STP; performance enhancing proxy; segment caching; IP-ABR; Internet; bandwidth reservation; IP-VBR; congestion avoidance; bandwidth sharing. Includes bibliographical references (p. 98-99).