CHAPTER 1 INTRODUCTION

In a world of fast changing technology, there is a rising requirement for people to communicate and get connected with each other and have appropriate and timely access to information regardless of the location of the each individuals or the information. The increasing demands and requirements for wireless communication systems ubiquity have led to the need for a better understanding of fundamental issues in communication theory and electromagnetic and their implications for the design of highly-capable wireless systems. In continuous development of mobile environments, the major service providers in the wireless market kept on monitoring the growths of 4th generation (4G) mobile technology. 2G and 3G are well-established as the mainstream mobile technology around the world. 3G is stumbling to obtain market share for a different reasons and 4G is achieving some confidence. In today's Internet, real-time applications such as VoIP, videoconferencing and on-line gaming mostly use RTP over UDP or UDP alone to transport data. Because these protocols are unresponsive to congestion events, the growing popularity of applications that use them endangers the stability of the Internet. So, to make it possible that real-time applications are widely adopted, common congestion control mechanisms suitable for real time multimedia are expected to be deployed. Also, a variety of wireless and wired technologies have been developed in the past years. The vision for the next generation of mobile communications networks consists in having these technologies integrated and handovers between them occurring seamlessly. These handovers may cause that during a connection the bandwidth available varies in one or more orders of magnitude. More volatile scenarios, such as ad hoc or sensor networks, are also expected. Most probably, next generation terminals will be multi-homed and will act as mobile routers. For these reasons, the control of real time flows in 4G networks is still an unsolved issue. New solutions are required so that the network stability is maintained even when conditions vary abruptly, and the quality perceived by interactive real-time applications is not degraded by the mechanisms controlling the flow

1.1 4G Network Architecture Figure shows the widely accepted 4G network structure with IP as the core network used for communication; integrating the 2G, 3G and 4G technologies using a convergence layer

Fig. Architecture of 4G Network

4G architecture will provide access through a collection of radio interfaces, seamless roaming/handover and the best-connected service, combining multiple radio access interfaces (such as WLAN, Bluetooth and GPRS) into a single network that subscribers may use. It allows any mobile device to seamlessly roam over different wireless technologies automatically, using the best connection available for the intended use. Users will have access to different services, increased coverage, the convenience of a single device, one bill with reduced total access cost, and more reliable wireless access even with the failure or loss of one or more networks.

In the 4G architecture, a single physical 4G communication device with multiple interfaces to access services on different wireless networks. The multimode device architecture may improve call completion and expand effective coverage area. The device itself incorporates most of the additional complexity without requiring wireless network modification or employing interworking devices. Each network can deploy a database that keeps track of user location, device capabilities, network conditions, and user preferences. It allow the social network user to connect the rest of the network members without any modification of his/her infrastructure, application, services and the architecture of communication system .

1.2 Issues in 4G Networks

Some of the issues in 4G Network are 1. Multimode user terminal: Multimode user terminal is a device working in different modes supporting a wide variety of 4G services and wireless networks by reconfiguring themselves to adapt to different wireless networks. They encounter several design issues such as limitations in the device size, cost, power consumption and backward compatibility to systems. 2. Wireless network discovery: Availing 4G services require the multimode user terminal to discover and select the preferred wireless network. Service discovery in 4G will be much more challenging then 3G because of the heterogeneity of the networks and their access protocols. 3. Wireless network selection 4G will provide the users a choice to select a wireless network providing optimized performance and high QoS for a particular place, time and desired service (communication, multimedia). But what parameters define high QoS and optimized performance at particular instant needs to be clearly defined to make the network selection procedure efficient and transparent to the end user.

Possible considerations may be available network resources, network supported service types and cost and user preference. 4. Terminal mobility Terminal mobility is an essential characteristic to fulfill the Anytime Anywhere promise of 4G. It allows the mobile users to roam across the geographic boundaries of wireless networks . Two main issues in terminal mobility are location and hand off management. Location management involves tracking the location of the mobile users and maintaining information such as the authentication data, QoS capabilities, and the original and the current cell location. Handoff management is maintaining the ongoing communication when the terminal roams. Handoff can be horizontal or vertical depending on whether the user moves from one cell to another within the same wireless systems or across different wireless systems (WLAN to GSM). Handoff process faces several challenges like maintaining the QoS and system performance across different systems, deciding the correct handoff time, designing the correct handoff mechanism, packet losses, handover latency and the increased system load. 5.Network infrastructure and QoS support Unlike previous generation networks (2G and 3G), 4G is an integration of IP and non-IP based system. Prior to 4G, QoS designs were made with a particular wireless system in mind. But in 4G networks QoS designs should consider the integration of different wireless networks to guarantee QoS for the end-to-end services end-to-end services. 6. Security Most of the security schemes and the encryption/decryption protocols of the current generation networks were designed only for specific services. They seem to be very inflexible to be used across the heterogeneous architecture of 4G which needs dynamically reconfigurable, adaptive and lightweight security mechanism. 7. Fault tolerance Wireless networks characterize a tree-like topology. Any failure in one of the levels can affect all

the network elements at the levels below. This problem can be further aggravated because of the multiple tree topologies. Adequate research work is required to devise a strategy for fault tolerance in wireless networks

8. Convergence services The idea of convergence means that the creation of the atmosphere that can eventually provide seamless and high-reliable and quality broadband mobile communication service and ubiquitous service through wired and wireless convergence networks without the space problem and terrestrial limitation, by means of ubiquitous connectivity. Convergence among industries is also accelerated by formation of alliances through participation in various projects to provide convergence services. 4G mobile systems will mainly be characterized by a horizontal communication model, where such different access technologies as cellular, cordless, wireless LAN type systems, short-range wireless connectivity, and wired systems will be combined on a common platform to complement each other in the best possible way for different service requirements and radio environments . The development is expected to inspire the trend of progressive information technologies a far from the current technical focus on fully mobile and widespread convergence of media. The trends from the service perspective include integration of services and convergence of service delivery mechanisms. In accordance with these trends, mobile network architecture will become flexible and versatile, and new services will be easy to deploy. 9. Broadband Services Broadband is a basis for the purpose of enabling multimedia communications including video service, which requires transmission of a large amount of data; it naturally calls media convergence aspect, based on packet transport, advocating the integration of various media on different qualities. The increasing position of broadband services like Asymmetric Digital Subscriber Line (ADSL) and optical fiber access systems and office or home LANs is expected to lead to a demand for similar services in the mobile communication environment. 4G service application characteristics will give broadband service its advantages;

 Low cost: To make broadband services available to the user to exchange various kinds of information, it is necessary to lower charges considerably in order to keep the cost at or below the cost of existing service. Coverage of Wide Area: One feature of mobile communications is that its availability and omnipresent. That advantage is important for future mobile communication as well. In particular, it is important to maintain the service area in which the terminals of the new system can be used during the transition from the existing system to a new system. Wide Variety of Services Capability: Mobile communication is for various types of users. In the future, we expect to make the advanced system performance and functionality to introduce a variety of services not only the ordinary telephone service. Those services must be made easier for anyone to use 10 .Interactive with Home-Networking, Telemetric, Sensor-network services Since technologies are becoming more collaborative and essential. Evolution of all network services based on All-IP network is needed for more converged services. IP-based unified network for far above the ground quality convergence services through active access is what broadband convergence network is all about. ALL-IP or Next Generation Network-IP based convergence of wired or wired backbone network, which may be the most rapidly deployed case of convergence. All-IP technology networking and IP multimedia services are the major trends in the wired and wireless network. The idea of the broadband convergence network (BcN) fit in the provision of a common, unified, and flexible service architecture that can support multiple types of services and management applications over multiple types of transport networks. The primary purpose of putting 4G service application into more interactive driven broadband convergence network is its applicability for home-networking, telemetric, and sensor-network service. Collaborative converged network will give a more beneficial service and application, especially if it is in broadband computing to the users and its providers. To give more emphasis on this service

application, one example is home networking as its applicability binds to give more advantage to the users and the society in terms of broadband connectivity. Far more than broadband convergence network application, telemetric application will put more tangible emphasis on the 4G mobile technology application.

11. Flexibility and Personalized Service The key concern in security designs for 4G networks is flexibility. 4G systems will support comprehensive and personalized services, providing stable system performance and quality of service. To support multimedia services, high-datarate services with good system reliability will be provided. At the same time, a low data rate transmission cost will be maintained. In order to meet the demands of these diverse users, service providers should design personal and customized services for them. Personal mobility is a concern in mobility management. Personal mobility concentrates on the movement of users instead of users terminals, and involves the provision of personal communications and personalized operating environments.

1.3.Congestion Control in 4G Heterogenous Network

In today's Internet, real-time applications such as VoIP, videoconferencing and on-line gaming mostly use RTP over UDP or UDP alone to transport data. Because these protocols are unresponsive to congestion events, the growing popularity of applications that use them endangers the stability of the Internet. So, to make it possible that real-time applications are widely adopted, common congestion control mechanisms suitable for real time multimedia are expected to be deployed. Also, a variety of wireless and wired technologies have been developed in the past years. The vision for the next generation of mobile communications networks consists in having these technologies integrated and handovers between them occurring seamlessly. These handovers may cause that during a connection the bandwidth available varies in one or more orders of magnitude. More volatile scenarios, such as ad hoc or sensor networks, are also expected. Most probably, next generation terminals will be multi-homed and will act as mobile

routers. For these reasons, the control of real time flows in 4G networks is still an unsolved issue. New solutions are required so that the network stability is maintained even when conditions vary abruptly, and the quality perceived by interactive real-time applications is not degraded by the mechanisms controlling the flow. Congestion control over network, for all types of media traffic, has been an active area of research in the last decade. This is due to the flourishing increase in the audiovisual traffic of digital convergence. There exists a variety of network applications built on its capability of streaming media either in real-time or on demand such as video streaming and conferencing, voice over IP (VoIP), and video on demand (VoD). The number of users for these network applications is continuously growing hence resulting in congestion. All the networks applications do not use TCP and therefore do not allow fair allocation with the available bandwidth. Thus, the result of the unfairness of the non-TCP applications did not have much impact because most of the traffic in the network uses TCP-based protocols. However, the quantity of audio/video streaming applications such as Internet audio and video players, video conferencing and analogous types of real-time applications is frequently increasing and it is soon expected that there will be an increase in the proportion of non-TCP traffic. In view of the fact that these applications commonly do not amalgamate TCP-compatible congestion control mechanisms, network applications treat challenging TCP-flows in an unreasonable manner. All TCP-flows reduce their data rates in an attempt to break up the congestion, where the non-TCP flows maintains to send at their original rate. This highly unfair condition will lead to starvation of TCP-traffic i.e.., congestion collapse, which describes the disagreeable situation where the accessible bandwidth in a network is almost entirely occupied by packets which are discarded because of the congestion before they reach their destination. For this reason, it is desirable to define suitable congestion control mechanisms for non-TCP traffic that are compatible with the rate-adaptation mechanism of TCP. These mechanisms should make non-TCP applications TCPfriendly, and thus lead to a fair distribution of bandwidth..

1.4 Problem Definition

Since IP was designed to be a protocol of integration, i.e. to interconnect various networks (which may or may not be using different transmission technologies), its essential concern has focused on robustness and scalability. So, regardless of the technology on which the network is built and the size of the growth, the protocol will be able to absorb them and keep on delivering packets in the best possible manner. The emergence of multimedia, real-time, and mission-critical applications, and the overwhelming presence of the Internet, has created the need to differ from the original all packets are created equal paradigm, and to look into some traffic differentiation and discrimination. Simply, while the best effort model should be preserved for most of the traffic flows, there are some applications, for instance voice and video that require special treatment or guarantees with respect to bandwidth, reliability, delay, the variation of the delay, and priority for processing at the routers. Congestion works against the stability and the efficiency of the networks. The more congested is the network, the less there is bandwidth for the flows, not to mention the effective throughput. Congestion control is a set of procedures and mechanisms whose primary function is to either prevent congestion or rectify its consequences. In general, congestion control schemes are used to maintain network operation at acceptable performance levels when Congestion occurs. Some of the main reasons behind congestion are: Slow links, End and intermediate systems limited processing power, and Shortage of buffer space.

Solving the congestion problem is not a simple case of just adding new resources or extending the capabilities of the old ones. For example, sending data at high rate through a high-speed LAN might be a problem for the gateway linking the network to the outside. Due to the high volume of data in a short time interval, or a burst, the buffer will be ultimately overflowed. In this case, having a larger buffer will most likely cause a larger accept loss, since burst are likely to challenge any reasonable buffer capacity. Occasionally, the complexity of TCP works against itself, viz. not all applications on the Internet have the same requirements concerning reliability, delay, or flow control. Reliability, which is based on redundancy or retransmissions of delayed or lost packets, is Counterproductive in real-time applications. In fact, the same is true for multimedia applications where the main concerns are the available bandwidth, small variations in the delay, and the guarantees that sustain the transmission quality over certain time interval. In order to avoid using TCP as a main vehicle for transport for all the applications on the Internet, a simpler protocol, termed as UDP, has been designed and implemented. It transports data at a high speed with a low overhead. Unlike TCP, UDP is not aware of congestion and thus does not care if it occurs. The protocol pumps data into the network, as much it is possible, and consequently, within a reasonable time, it induces congestion. The first sign is usually a dramatic drop in the performance of TCP, which in presence of congestion will slow down and eventually halt transporting segments.

Most of the applications on the Internet, or at least those that have been widely used so far, such as mail exchange, ftp, web browsing, employ TCP as a transport protocol. The initial procedures built into TCP to control congestion were rather elementary and restricted to preventing an overflow of the destination buffer. They did not deal with the routers at all. This problem was behind the series of congestion collapses at the end of eighties in the last century, and the surge of research into possible modifications and extensions of the protocol in order to meet the challenges of the new transmission technologies and the explosive growth of networking and the Internet. Indeed, the last fifteen years have witnessed quite an extensive and meritorious research in the nature of congestion and how to control it. The two types of mechanisms that address network congestion are congestion avoidance and congestion control. Congestion avoidance allows a network to operate in the optimal region of low delay and high throughput, thus preventing the network entering the state of congestion. Traditional congestion control facilitates network recovery from congestion, or high delay and low throughput, to a normal operating state. While trying to preserve end-to-end semantics that is inherent in the way TCP was conceived and operates, there are two ways to approach the congestion. The first venue is the host-centric one where the source host responds to congestion by reducing the load it injects in the network. The other venue, a router centric one, is to deal with the intermediate nodes by using queue scheduling and active queue management of the routers buffers. Finally, there is a blend of two, in essence a host-centric management that requires assistance from the network, and where the routers provide explicit information

about their own state in a form of a feedback to the host that consequently reduces the load. The number of TCP modifications and variants based on the host-centric and router centric schemas is substantial, yet each one of them has some limitations. The end-to-end congestion control schemes operate rather well, however they are limited to TCP flows. Some of them have problem with fairness, or the proportionate usage of the network resources by the majority of the flows. The problem with fairness may be somewhat fixed with the router-centric congestion control schemes. One of the problems that appear in this case is that the packet drop leads to low throughput and resource waste, since the packets have already reached the router and used some of the network resources along the way. Again, network-assisted schemes are less prone to packet loss than the router centric ones, but they only work with TCP. An additional concern is that both router centric and networkassisted congestion control schemes require modifications of the router architecture and sometimes imply a modification of the TCP packet structure. Moreover, the router itself does something with the packet, which is not part of its original functionality to route the packets in most efficient manner. Let us turn our attention to non-TCP or unresponsive flows (such as UDP) that do not recognize the state of congestion. As Floyd writes, the contribution of unresponsive flows is becoming increasingly present in creating congestion. One way to approach this problem is to move congestion control for unresponsive flows to the application layer. If all applications that use UDP have some kind of end-to-end congestion control mechanism then the problem may be resolved. This is hardly feasible. Namely, there are no standard mechanisms for congestion control on the application layer, and it is not pragmatic to expect that application designers will take care of the issues, which should not be their concern. Many multimedia applications do

not use end-to-end congestion control at all. They actually increase the sending rate in response to the increased loss to make up for the errors. Traffic on the Internet and networks in general is becoming intensive and mixed from both responsive and unresponsive flows. The primary research question they would like to answer is how they can make these different flows, socially responsible and irresponsible to work together, exhibit flavor for fairness and impose congestion control. The corollary is whether or not it is possible to come up with a mechanism that will do similar things to congestion when induced by non-responsive flows, as the one that works for TCP-flows.

1.5 Objectives of Congestion Control in 4G Heterogenous Network

For any congestion control mechanisms, the most fundamental design objectives are stability and scalability. However, achieving both properties are very challenging in such a heterogeneous environment as the Internet. From the end-users' perspective, heterogeneity is due to the fact that different flows have different routing paths and therefore different communication delays, which can significantly affect stability of the entire system. Congestion can be defined as a state or condition that occurs when network resources are overloaded resulting in impairments for network users as objectively measured by the probability of loss and/or of delay. Congestion control is a (typically distributed) algorithm to share network resources among competing traffic sources. The Internet encompasses a large variety of heterogeneous IP networks that are realized by a multitude of technologies, which result in a tremendous variety of link and path characteristics: capacity can be either scarce in very slow speed radio links (several kbps), or there may be an

abundant supply in high-speed optical links (several gigabit per second). Concerning latency, scenarios range from local interconnects (much less than a millisecond) to certain wireless and satellite links with very large latencies up to or over a second). Even higher latencies can occur in space communication. As a consequence, both the available bandwidth and the end-to-end delay in the Internet may vary over many orders of magnitude, and it is likely that the range of parameters will further increase in future.

1.6 Organisation of the report Chapter 2 deals with discussion of the papers that are referred. Chapter 3 deals with proposed work in which flowchart and algorithm of the project is discussed. Chapter 4 deals with simulation model . Chapter 5 deals with simulation results. Chapter 6 presents conclusion

CHAPTER 2 RELATED WORK

In the paper [1] authors have compared the feedback-based to the reservation-based congestion control approach and focus on the first one, by evaluating some mechanisms with respect to Media Friendliness, Scalability and Dynamic Behavior. They also present a set of requirements for the ideal congestion control mechanism of real-time flows in 4G networks. In this article authors have compared the feedback-based to the reservation-based congestion control approach and focus on the first one, by evaluating some mechanisms with respect to Media Friendliness, Scalability and Dynamic Behavior. They also present a set of requirements for the ideal congestion control mechanism of real-time flows in 4G networks. The paper [2] considers the potentially negative impacts of an increasing deployment of noncongestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, they argue that router mechanisms are needed to identify and restrict the bandwidth of selected high bandwidth best-effort flows in times of congestion. The authors have discussed several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, not TCP-friendly, or simply using disproportionate bandwidth. A flow that is not TCP-friendly is one whose long-term arrival rate exceeds that of any conformant TCP in

the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion. In the paper [3] author has described the main ideas behind some of the most important of such router congestion feedback (RCF) approaches based on network-information sharing (NIS). In addition, the properties, functionalities, and expected performance gain of these RCF approaches are compared and their applicability in the current Internet environment is investigated. The aim of this paper is to find potential RCF candidates that can be used to improve congestion control in the current Internet as well as in future IP based networks where diverse wired and wireless access technologies are used in parallel. The purpose of paper [4] is to analyze and compare the different congestion control and avoidance mechanisms which have been proposed for TCP/IP protocols, namely: Tahoe, Reno, New-Reno, TCP Vegas and SACK. TCPs robustness is as a result of its reactive behavior in the face of congestion, and fact that reliability is ensured by re-transmissions. All the above mentioned implementations suggest mechanisms for determining when a segment should be retransmitted and how should the sender behave when it encounters congestion and what pattern of transmissions should it follow to avoid congestion. In this paper they have discussed how the different mechanism affect the through put and efficiency of TCP and how they compare with TCP Vegas in terms of performance. The paper[5] uses simulations to explore the benefits of adding selective acknowledgments (SACK) and selective repeat to TCP. Authors have compared Tahoe and Reno TCP, the two most common reference implementations for TCP, with two modified versions of Reno TCP. The first version is New-Reno TCP, a modified version of TCP without SACK that avoids some of Reno TCP's performance problems when multiple packets are dropped from a window of data. The second version is SACK TCP, a conservative extension of Reno TCP modified to use the SACK option being proposed in the Internet Engineering Task Force (IETF). They described the congestion control algorithms in our simulated implementation of SACK TCP and show that

while selective acknowledgments are not required to solve Reno TCP's performance problems when multiple packets are dropped, the absence of selective acknowledgments does impose limits to TCP's ultimate performance. In particular, they showed that without selective acknowledgments, TCP implementations are constrained to either retransmit at most one dropped packet per round-trip time, or to retransmit packets that might have already been successfully delivered. Indirect TCP (or I-TCP), which is described in paper [6], is based on an indirect protocol model. In this approach, an end-to-end TCP connection between a fixed host and a mobile host is split into two separate connections: 1) a regular TCP connection between the fixed host and the mobility support router (base station) currently serving the mobile host and 2) a wireless TCP connection between the mobility support router and the mobile host. Use of mediation by the mobility support router (or indirection) at the transport layer allows special treatment of mobile hosts communicating over wireless links so as to address the problems mentioned earlier without sacrificing compatibility with existing fixed network protocols. In the paper [7], recent surge of interest towards congestion control that relies on single-router feedback suggests that such systems may offer certain benefits over traditional models of additive packet loss .Besides topology-independent stability and faster convergence to efficiency/fairness, it was recently shown that any stable single-router system with a symmetric Jacobian tolerates arbitrary fixed, as well as time-varying, feedback delays. Although delay-independence is an appealing characteristic, the EMKC system developed in exhibits undesirable equilibrium properties and slow convergence behavior. To overcome these drawbacks, authors proposed a new method called JetMax and show that it admits a low-overhead implementation inside routers (three additions per packet), overshoot-free transient and steady state, tunable link utilization, and delay-insensitive flow dynamics. The proposed framework also provides capacity-independent convergence time, where fairness and utilization are reached in the same number of RTT steps for a link of any bandwidth. Given a 1 mb/s, 10 gb/s, or googol (10100) bps link, the method converges to within 1% of the stationary

state in 6 control intervals. They have finished the paper by comparing JetMaxs performance to that of existing methods in ns2 simulations and discussing its Linux implementation. The paper [8] examines the problem of congestion control evaluation in dynamic networks. Authors have determined a source of deficiencies for existing metrics of congestion control performance the existing metrics are defined with respect to ideal allocations that do not represent short-term efficiency and fairness of network usage in dynamic environments. They have introduced the concept of an effair allocation, a dynamic ideal allocation that specifies optimal efficiency and fairness at every timescale. This concept has a general applicability; in particular, it applies to networks that provide both unicast and multicast services. Another desirable property of the effair allocation is its dependence on the communication needs and capabilities of applications. They have designed an algorithm that accounts for network delays and computes the effair allocation as a series of static ideal allocations. Using the notion of effair allocation as a foundation, they define a new metric of effairness that shows how closely the actual delivery times match the delivery times under the effair allocation. Authors have presented in paper [9] a new implementation of TCP that is better suited to todays Internet than TCP Reno or Tahoe. The implementation of TCP, which they call TCP Santa Cruz, is designed to work with path asymmetries, out-of-order packet delivery, and networks with lossy links, limited bandwidth and dynamic changes in delay. The new congestion-control and error-recovery mechanisms in TCP Santa Cruz are based on: using estimates of delay along the forward path, rather than the round-trip delay; reaching a target operating point for the number of packets in the bottleneck of the connection, without congesting the network; and making resilient use of any acknowledgments received over a window, rather than increasing the congestion window by counting the number of returned acknowledgments. They compared TCP Santa Cruz with the Reno and Vegas implementations using the ns2 simulator. The simulation experiments show that TCP Santa Cruz achieves significantly higher throughput, smaller delays, and smaller delay variances than Reno and Vegas. TCP Santa Cruz is also shown to prevent the swings in the size of the congestion window that typify TCP Reno and Tahoe traffic, and to determine the direction of congestion in the network and isolate the forward throughput from events on the

reverse path. In paper [10] authors describes heterogeneous congestion control protocols that react to different pricing signals share the same network, the resulting equilibrium may no longer be interpreted as a solution to the standard utility maximization problem. They prove the existence of equilibrium under mild assumptions. Then they show that multi-protocol networks whose equilibria are locally non-unique or infinite in number can only form a set of measure zero. Multiple locally unique equilibria can arise in two ways. First, unlike in the single-protocol case, the set of bottleneck links can be non-unique with heterogeneous protocols even when the routing matrix has full row rank. The equilibria associated with different sets of bottleneck links are necessarily distinct. Second, even when there is a unique set of bottleneck links, network equilibrium can still be non-unique, but is always finite and odd in number. They cannot all be locally stable unless it is globally unique. Finally, they provide various sufficient conditions for global uniqueness. Numerical examples are used throughout the paper to illustrate these results. In paper [11] TCP (transmission control protocol) is a feedback-based congestion control

algorithm and each TCP sending host determined its window size independently according to the timeouts and the receipt of the duplicate acknowledgments (ACKs). Since this blind rate adaptation mechanism led to multiple packet losses and a global synchronization problem, Floyd and Jacobson proposed the random early detection (RED) algorithm . RED tried to detect the beginning of the congestion by monitoring the average queue length at the router, and informed the sending hosts by dropping packets. An ECN (explicit congestion notification) algorithm has been proposed to avoid the throughput degradation due to unnecessary packet drops by the RED algorithm. The idea of ECN was to notify sending hosts explicitly of congestion occurrence in the network instead of packet drops. Since these congestion control mechanisms were based on an end-to-end fashion, it would be impossible to guarantee maxmin fair sharing of the bandwidth due to a lack of explicit information on the network states. To solve the TCP fairness problem, packet buffering and scheduling algorithms were proposed. However, these algorithms required per-connection state information at each router and they did not guarantee maxmin fair sharing of the bandwidth among the active connections. In they proposed an algorithm to eliminate the packet loss using IPv6 optional fields. In the congestion window control algorithms

for TCP with ECN were presented to achieve fairness and stability. However, they were limited to a single bottleneck link. In this paper, author has proposed a modified window control algorithm that guarantees TCP fairness. They use successive ECN congestion indications and obtain network information. Using the obtained network information and the modified RED algorithm, they develop a window control algorithm to achieve fair sharing of the available bandwidth in an ECN capable TCP network where each connection has a different propagation delay and traverses multiple bottleneck links. In paper [12], authors have discussed about Recent research has indicated that knowledge of Round Trip Time (RTT) and available bandwidth is crucial for efficient network control. In this contribution they discuss the problem of estimating these quantities. Based on a simple aggregated model of the network, an algorithm combining a Kalmanlter and a change detection algorithm (CUSUM) is proposed for RTT estimation. It is illustrated on real data that this algorithm provides estimates of significantly better accuracy as compared to the RTT estimator currently used in TCP, especially in scenarios where new cross-traffic flows cause bottle- neck queues to rapidly build up which in turn induces rapid changes of the RTT. They also analyze how wireless links affect the RTT distribution. It is well known that link re-transmissions induce delays which do not conform to the assumptions on which the transport protocol is based. This causes undesired TCP control actions which reduce through- put. A link layer solution is proposed to counter this problem. Carefully selected (artificial) delays are added to packets retransmitted on the link which makes the delay-distribution TCP-friendly. The information required for this algorithm is readily available at the link and consists of the actual delaydistribution induced by the link. The added delays are obtained from a non- convex program which due to its low complexity is easy to solve. In paper [13] authors paper presents and develops a novel delay-based AIMD congestion control algorithm. The main features of the proposed solution include: (1) low standing queues and delay in homogeneous environments (with delay-based flows only); (2) fair coexistence of delay- and loss-based flows in heterogeneous environments; (3) delay-based flows behave as loss-

based flows when loss-based flows are present in the network; otherwise they revert to delay-based operation. It is also shown that these properties can be achieved without any appreciable increase in network loss rate over that which would be present in a comparable network of standard TCP flows (loss-based AIMD). To demonstrate the potential of the presented algorithm both analytical and simulation results are provided in a range of different network scenarios. These include stability and convergence results in general multiple-bottleneck networks, and a number of simulation scenarios to demonstrate the utility of the proposed scheme. In particular, they show that networks employing our algorithm have the features of networks in which RED AQMs are deployed. Furthermore, in a wide range of situations (including high speed scenarios), they show that low delay is achieved irrespective of the queuing algorithm employed in the network, with only sender side modification to the basic AIMD algorithm. In this paper [14] authors have discussed about When heterogeneous congestion control protocols that react to different pricing signals (They could be different types of signals such as packet loss, queuing delay etc. or different values of the same type of signal such as different ECN marking values based on the same actual link congestion level) share the same network, the current theory based on utility maximization fails to predict the network behavior. Unlike in a homogeneous network, the bandwidth allocation now depends on router parameters and flow arrival patterns. It can be nonunique, suboptimal and unstable. In [36], existence and uniqueness of equilibrium of heterogeneous protocols are investigated. This paper extends the study with two objectives: analyze the optimality and stability of such networks and design control schemes to improve them. First, they demonstrate the intricate behavior of a heterogeneous network through simulations and present a framework to help understand its equilibrium properties. Second, they propose a simple source-based algorithm to decouple bandwidth allocation from router parameters and flow arrival patterns by only updating a linear parameter in the sources algorithms on a slow timescale. It is used to steer a network to the unique optimal equilibrium.

The scheme can be deployed incrementally as the existing protocol needs no change and only the new protocols need to adopt the slow timescale.

In paper [15] authors have said that the classical TCP lP layered protocol architecture is beginning to show signs of age. In order to cope with problems such as the poor manager performance of wireless links and mobile terminals, including the high error rate of wireless network interfaces, power saving requirements, quality of service, and an increasingly dynamic network environment, a protocol architecture that considers cross-layer interactions seems to he required. This article describes a framework for further enhancements of the traditional IP based protocol stack to meet current and future requirements. Known problems associated with the strictly layered protocol architecture are summarized and classified, and a first solution involving cross-layer design is proposed. In paper [16] authors have explained about 1design the multimedia transport protocol in heterogeneous wired-cum-wireless environment faces great challenges because of two contradictory objectives. On the one hand, the multimedia application requires smooth transfer rate, i.e., stability objective; on the other hand, vertical handoff in heterogeneous networks requires fast response at transfer rate, i.e., flexibility objective. To address this problem, this paper proposes to use passive bandwidth measurement at the receiver in the design of rate control algorithm for multimedia transport protocol. Moreover, a window based exponentially weighted moving average (EWMA) filter with two weights is introduced to achieve stability and flexibility at the same time. Based on these considerations, a multimedia transport protocol (MMTP) is proposed. Its stability and flexibility as well as its fairness are verified by simulations.

In this paper [17] authors have presented a new implementation of TCP that is better suited to todays Internet than TCP Reno or Tahoe. Our implementation of TCP, which they call TCP Santa Cruz, is designed to work with path asymmetries, out-of-order packet delivery, and

networks with lossy links, limited bandwidth and dynamic changes in delay. The new congestion-control and error-recovery mechanisms in TCP Santa Cruz are based on: using estimates of delay along the forward path, rather than the round-trip delay; reaching a target operating point for the number of packets in the bottleneck of the connection, without congesting the network; and making resilient use of any acknowledgments received over a window, rather than increasing the congestion window by counting the number of returned acknowledgments. They compare TCP Santa Cruz with the Reno and Vegas implementations using the ns2 simulator. The simulation experiments show that TCP Santa Cruz achieves significantly higher throughput, smaller delays, and smaller delay variances than Reno and Vegas. TCP Santa Cruz is also shown to prevent the swings in the size of the congestion window that typify TCP Reno and Tahoe traffic, and to determine the direction of congestion in the network and isolate the forward throughput from events on the reverse path.

In paper [18] authors have discussed about Today's wireless networks are highly heterogeneous, with mobile devices consisting of multiple wireless network interfaces (WNICs). Since battery lifetime is limited, power management of the interfaces has become essential with flexible and open architecture, capable of supporting various types of networks, terminals and applications. However how to integrate the protocols to meet the heterogeneous network environments becomes a significant challenge in the fourth generation wireless network. Adaptive protocols are proposed to solve heterogeneity problem in future wireless networks. This paper discusses two protocols RCP, and RCP and feasibility of RCP protocols applied to the manage power efficiently and adaptive Congestion control on heterogeneous wireless network.

In this paper [19] authors present new queue length based Internet congestion control protocol which is shown through simulations to work effectively. The control objective is to regulate the queue size at each link so that it tracks a reference queue size chosen by the designer. To achieve the latter, the protocol implements at each link a certainty equivalent proportional controller which utilizes estimates of the effective number of users utilizing the link. These estimates are generated online using a novel estimation algorithm which is based on online parameter

identification techniques. The protocol utilizes an explicit multibit feedback scheme and does not require maintenance of per flow states within the network. Extensive simulations indicate that the protocol is able to guide the network to a stable equilibrium which is characterized by maxmin fairness, high utilization, queue sizes close to the reference value and no observable packet drops. In addition, it is found to be scalable with respect to changing bandwidths, delays and number of users utilizing the network. The protocol also exhibits nice transient properties such as smooth responses with no oscillations and fast convergence In this paper [20] author describes the recent trend is that the mobile internet service has been offered in the integration of various wireless networks. In such heterogeneous networks, vertical handover is more common and important handover technologies. But during vertical handover, standard TCP has experienced many problems such as multiple packet losses, the packet reordering, the under-utilization due to the drastic change of the Bandwidth Delay Product (BDP) and the network transmission delay (Round Trip Time :RTT). In this paper, they propose Enhanced TCP congestion control scheme with RTT inflation and the measured-RTT of the new network for the seamless soft vertical handover and evaluate this by OPNET simulation. They assume the proposed scheme uses the cross-layer design in a TCP receiver and a TCP timestamp option. OPNET simulation results show that our proposed scheme improves better TCP performance than other handover congestion control schemes such as Freeze-TCP or SSTCP during the vertical handover.

In this paper [21], authors have described about how to develop a novel analytical framework for modeling and quantifying the performance of window controlled multimedia flows in a hybrid wireless/wired network. The framework captures the traffic characteristics of window controlled flows and is applicable to various wireless links and packet transmission schemes. They show analytically the relationship between the sender window size, the wireless link throughput distribution, and the delay distribution. They then substantiate the analysis by

demonstrating how to statistically bound the end-to-end delay of flows controlled by a TCP-like Datagram Congestion Control Protocol (DCCP) over an M-state Markovian wireless link. Simulation results validate the analysis and demonstrate the effectiveness and efficiency of the proposed delay control scheme. The scheme can also be applied to other window-based transport layer protocols. In this paper [22] authors present new proposed protocol to enhance the TCP/IP versatility as the main protocol for wireless data transmission. TCP/IP has shown its superiority in the selection of protocol for establishing wired networks. Unfortunately, its superiority cannot be extended to wireless networks. However, they believe that the integration of several types of networks would take place. The 4th Generation (4G) wireless mobile internet networks will merge the current existing cellular networks (i.e., CDMA2000, WCDMA and TD_SCDMA) and Wi-Fi networks (i.e., Wireless LAN) with the fixed internet to support wireless mobile internet. This integration would provide the same quality of service as fixed internet. Each of the networks has their own specified protocols, disparity frequency, and maximum data speed and cost characteristics. TCP/IP suite protocols were successful in web application of fixed internet, but exhibit limitation to work on the combined networks. Two research directions are available, which are replacement and improvement. Microsoft has issued a new protocol suite for replacement. In this paper, they propose a new protocol to improve TCP/IP suite protocols. This new protocol addresses the limitation of TCP/IP suite so that it can work on both cellular network and Wi-Fi network simultaneously; sending data requests through cellular network and getting reply from Wi-Fi network. Ns2 Java version (Java Network Simulator) was chosen to simulate the new protocol because of its feasibility. In this paper, they present the results and discussion of our simulation.

In paper [23] authors have discussed about various congestion control algorithms, using network awareness as a criterion to categorize different approaches. The first category (the box is black) consists of a group of algorithms that consider the network as a black box, assuming no knowledge of its state, other than the binary feedback upon congestion. The second category (the box is grey) groups approaches that use measurements to estimate available bandwidth,

level of contention or even the temporary characteristics of congestion. Due to the possibility of wrong estimations and measurements, the network is considered a grey box. The third category (the box is green) contains the bimodal congestion control, which calculates explicitly the fairshare, as well as the network-assisted control, where the network communicates its state to the transport layer; the box now is becoming green. They go beyond a description of the different approaches to discuss the tradeoffs of network parameters, the accuracy of congestion control models and the impact of network and application heterogeneity on congestion itself.

In paper [24] authors have explained about Modern Telecommunication, Computer Networks and both wired and wireless communications including the Internet, are being designed for fast transmission of large amounts of data, for which Congestion Control is very important. Without proper Congestion control mechanism the congestion collapse of such networks would become highly complex. Congestion control for streamed media traffic over network is a challenge due to the sensitivity of such traffic towards. This challenge has motivated the researchers over the last decade to develop a number of congestion control protocols and mechanisms that suit the traffic and provides fair maintenance for both unicast and multicast communications. This paper gives out a brief survey of major congestion control mechanisms, categorization characteristics, elaborates the TCP-friendliness concept and then a state-of-the-art for the congestion control mechanisms designed for network. The paper points the pros and cons of the congestion control mechanism, and evaluates their characteristics.

CHAPTER 4 SIMULATION

4.1 Simulation
In communication and computer network research, network simulation is a technique where a program models the behavior of a network either by calculating the interaction between the different network entities (hosts/routers, data links, packets, etc) using mathematical formulas, or actually capturing and playing back observations from a production network. The behavior of the network and the various applications and services it supports can then be observed in a test lab; various attributes of the environment can also be modified in a controlled manner to assess how the network would behave under different conditions.

4.2 Simulator
A network simulator is a software program that imitates the working of a computer network. In simulators, the computer network is typically modelled with devices, traffic etc and the performance is analysed. Typically, users can then customize the simulator to fulfill their specific analysis needs. Simulators typically come with support for the most popular protocols in use today, such as WLAN, Wi-Max, UDP, and TCP. We have used OMNET++ as a simulator for your project.

4.2.1 OMNET++

OMNeT++ is an object-oriented modular discrete event network simulator. The simulator can be used for: traffic modeling of telecommunication networks protocol modeling modeling queuing networks modeling multiprocessors and other distributed hardware systems validating hardware architectures evaluating performance aspects of complex software systems modeling any other system where the discrete event approach is suitable. An OMNeT++ model consists of hierarchically nested modules. The depth of module nesting is not limited, which allows the user to reflect the logical structure of the actual system in the model structure. Modules communicate through message passing. Messages can contain arbitrarily complex data structures. Module scan send messages either directly to their destination or along a predefined path, through gates and connections. Modules can have their own parameters. Parameters can be used to customize module behavior and to parameterize the models topology. Modules at the lowest level of the module hierarchy encapsulate behavior. These modules are termed simple modules, and they are programmed in C++ using the simulation library. OMNeT++ simulations can feature varying user interfaces for different purposes: debugging, demonstration and batch execution. Advanced user interfaces make the inside of the model visible to the user, allow control over simulation execution and to intervene by changing variables/objects inside the model. This is very useful in the development/debugging phase of the simulation project. User interfaces also facilitate demonstration of how a model works. The simulator as well as user interfaces and tools are portable: they are known to work on Windows and on several Unix flavors, using various C++ compilers. OMNeT++ also supports parallel distributed simulation. OMNeT++ can use several mechanisms for communication between partitions of a parallel distributed simulation, for example MPI or named pipes. The parallel simulation algorithm can easily be extended or new ones plugged in. Models do not need

any special instrumentation to be run in parallel it is just a matter of configuration. OMNeT++ can even be used for classroom presentation of parallel simulation algorithms, because simulations can be run in parallel even under the GUI which provides detailed feedback on what is going on .OMNEST is the commercially supported version of OMNeT++. OMNeT++ is only free for academic and non-profit use for commercial purposes one needs to obtain OMNEST licenses from Omnest Global, Inc.

4.2.1 Modeling concepts OMNeT++ provides efficient tools for the user to describe the structure of the actual system. Some of the main features are: hierarchically nested modules modules are instances of module types modules communicate with messages through channels flexible module parameters topology description language A. Hierarchical modules An OMNeT++ model consists of hierarchically nested modules, which communicate by passing messages to each another. OMNeT++ models are often referred to as networks. The top level module is the system module. The system module contains submodules, which can also contain submodules themselves (Fig. 2.1). The depth of module nesting is not limited; this allows the user to reflect the logical structure of the actual system in the model structure. Model structure is described in OMNeT++s NED language. Modules that contain submodules are termed compound modules, as opposed simple modules which are at the lowest level of the module hierarchy. Simple modules contain the algorithms in

the model. The user implements the simple modules in C++, using the OMNeT++ simulation class library.

Fig 4.1 Simple and Compound Modules

B. Module types Both simple and compound modules are instances of module types. While describing the model, the user defines module types; instances of these module types serve as components for more complex module types. Finally, the user creates the system module as an instance of a previously defined module type; all modules of the network are instantiated as submodules and subsubmodules of the system module. When a module type is used as a building block, there is no distinction whether it is a simple or a compound module. This allows the user to split a simple module into several simple modules embedded into a compound module, or vica versa, aggregate the functionality of a compound module into a single simple module, without affecting existing users of the module type. Module types can be stored in files separately from the place of their actual usage. This means that the user can group existing module types and create component libraries.

C.Messages, gates, links

Modules communicate by exchanging messages. In an actual simulation, messages can represent frames or packets in a computer network, jobs or customers in a queuing network or other types of mobile entities. Messages can contain arbitrarily complex data structures. Simple modules can

send messages either directly to their destination or along a predefined path, through gates and connections. The ``local simulation time'' of a module advances when the module receives a message. The message can arrive from another module or from the same module (self-messages are used to implement timers). Gates are the input and output interfaces of modules; messages are sent out through output gates and arrive through input gates. Each connection (also called link) is created within a single level of the module hierarchy: within a compound module, one can connect the corresponding gates of two submodules, or a gate of one submodule and a gate of the compound module (Fig. below).

Fig 4.2 Connections Due to the hierarchical structure of the model, messages typically travel through a series of connections, to start and arrive in simple modules. Such series of connections that go from simple module to simple module are called routes. Compound modules act as `cardboard boxes' in the model, transparently relaying messages between their inside and the outside world.

D.Modeling of packet transmissions Connections can be assigned three parameters, which facilitate the modeling of communication networks, but can be useful in other models too: propagation delay, bit error rate and data rate,

all three being optional. One can specify link parameters individually for each connection, or define link types and use them throughout the whole model. Propagation delay is the amount of time the arrival of the message is delayed by when it travels through the channel. Bit error rate speficifies the probability that a bit is incorrectly transmitted, and allows for simple noisy channel modelling. Data rate is specified in bits/second, and it is used for calculating transmission time of a packet. When data rates are in use, the sending of the message in the model corresponds to the transmission of the first bit, and the arrival of the message corresponds to the reception of the last bit. This model is not always applicable, for example protocols like Token Ring and FDDI do not wait for the frame to arrive in its entirety, but rather start repeating its first bits soon after they arrive -- in other words, frames ``flow through'' the stations, being delayed only a few bits. If you want to model such networks, the data rate modeling feature of OMNeT++ cannot be used.