The present and the future of TCP/IP

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1 The present and the future of TCP/IP David Espina Project in Electronics Dariusz Baha Computer science ABSTRACT The Transport Control Protocol (TCP) and Internet Protocol (IP) are the two most important communication protocols used in computer network today. In this paper we describe the evolution and the basic functionality of the TCP\IP protocol. After reading this article the reader will have a brief knowledge of the origin and evolution of TCP and IP, as well as their structure, operational lifetime and philosophy. We expose the reasons why the first information networks needed to evolve to become what we know nowadays as the internet. Finally a future view is detailed of what the recent studies and tested technologies bring as the best solution to support the growing demands of the users and technological improvements. Multi-bit control feedback, are the key words to describe the future in network communications. Technologies like FAST TCP, delay-based systems and the 4G Mobile Networks based on IP are the most modern researches and the future for all networks communication. 1. INTRODUCTION The TCP protocol belongs to the Transport Layer in the OSI model which is an abstraction model for computer communication through networks. The task of the TCP protocol is to ensure a reliable communication between two hosts on an unreliable network. In one end it provides a service to the communicating application and in the other end, the IP protocol. In section II the birth of the protocols is explained both sociological and technological purposes. Since the 1960s the military in collaboration with several different universities in the U.S. started working in the implementation of a global network which purpose was connecting different locations working under different protocols and share information with several kinds of storage systems. In Section III we explain some basic functionalities of the TCP protocol and how it works. We will concentrate on connection establishment, connection release and the sliding window protocol. An expert reader may be interested in skipping the whole previous sections to section IV where is explained in detail the most recent discoveries and implementation about TCP and IP. There, some specifications about FAST TCP are detailed; a new estimation about the flux in the network is approached by the concept of delay-based systems instead of packet loss rate. Section V contains the conclusions reached after discussions about the statements in section IV always minding the history and evolution of the protocols. 2. History of TCP/IP IP was born to cover U.S. Department of Defense s communication needs. Last years of the 1960s the Advanced Research Projects Agency (ARPA), which is known nowadays as DARPA, started developing in common with some partner universities and the corporate research community the design of standard protocols and started building first multi-vendors networks. The result of working all together was ARPANET, the first packet switching network that was tested in 1969 with four nodes using Network Control Protocol. After the successful test the new born network turned into an operational network called ARPA Internet. In 1974 Vinton G.Cerf and Robert E.Kahn designed TCP/IP protocols. In January 1980 the Institute of Information Sciences at University of Southern California elaborated a reference document [5] describing the philosophy of the Internet Protocol. It was designed to be used in an environment of computer communication networks oriented to packet switched systems interconnected between them. In 1985 ARPANET started suffering from congestion and the National Science Foundation s developed NSFNET to support the previous net which was finally closed in The NSFNET was based on multiple regional networks and peer networks such as NASA Science Network. By 1986 there was a network architecture connecting campuses and research organizations connected also to super computer facilities. Over the years the speed of transmissions had to be increased and by 1991 the backbone was moved to a private company which started charging for connections and companies like IBM developed ANSNET in parallel which was nor aimed to enrich these companies. The structure of the information running through the network was designed as blocks of data split in small page 1 of 8

2 segments of bits called datagram s. Datagram s are packaged and sent from sources to destinations which are both hosts distinguished by a fixed length address. These datagram s are long enough to be considered a risk of loss of information due to the characteristics of the communications channel, fragmentation and reassembly. The internet protocol covered all needs to provide host-tohost delivery and its limitation such as reliability, flow control, sequencing among others, were not considered. The internet protocol was meant to interact with local network protocols to transport the required datagram to the next gateway or destination host. It implemented addressing and fragmentation as two basic functions. The datagram contains information about host source and host destination within its header. This information is coded as an address and in order to complete the delivery a path must be chosen. The operational structure isolates every host and its gateway that connects to the global network. These isolated modules take decisions based on common rules to interpret datagram s and make routing decisions as well as other functions. Each datagram is treated as an independent unit on its way through the network. Any trace of logical circuits or connections is banished. To provide service, four parameters were created: Type of Service, Time to Live, Options and Header Checksum. The first one provides information about the QoS (Quality of Service) available being eligible for example Real Time, Interactive or Bulk. This QoS responds to how gateways manage the transmission parameters for a specific network, the network used by the next node or the next gateway when routing a datagram. Time to Live is a countdown number that indicates how long a datagram is meant to last and after reaching zero it is self destroyed. Options parameter does not represent a vital part of the protocol because gives support to non regular situations in the communications process when errors occur or special routing is needed among others. The fourth parameter created was Header Checksum which gives feedback information about the correct transmission. If this parameter shows an error the whole datagram is discarded and a fundamental characteristic that comes with this error assumption is that IP does not have retransmissions, error control for data but the header checksum or flow control. As computer communication became more and more important, especially for the military at that time. It was realized that a robust communication standard is needed to replace the variety of different local network protocols that were used. A concept for the TCP was first described in [8] where several issues that would be solved where presented. The TCP was declared to be a reliable connection-oriented, end-to-end protocol. It was meant to operate on top of the IP protocol. TCP was first defined in [4]. 3. Present TCP/IP To accomplish its task, both the receiver and the sender must create communication end points, called sockets, or more precisely the communicating processes on each of the senders and receivers machines are creating the sockets. It is trough the sockets that the communicating processes can send and receive data. Each socket has an IP address and a port number. There are some well known port numbers like port 21 for the FTP protocol which with many others is included in the TCP/IP suite. Port numbers below 1024 are reserved for such well known services. A TCP connection is established between two hosts only, which mean that multicasting and broadcasting is not supported. Some features of the TCP include buffering of data, which means that the data passed to TCP may not be sent immediately instead the TCP may buffer it. Hence a larger chunk of data can be sent. This means that the data is sent as a byte stream and not as a message stream. 3.1 The TCP Segment Structure The TCP uses a segment structure to send data. Each segment contains information for the TCP/IP protocol, as well as the data itself. First we have the IP header with a fixed size of 20 bytes. The IP header contains IP specific information, like the IP address. After the IP header we have the TCP header, also with a fixed size of 20 bytes. Attached to the end of the TCP header is the data itself, the size of the data can be 0 65,495 bytes [7]. Figure 3.1 TCP header. [7] First we have the source and destination port fields, these are for the 16-bit port numbers that identify the sending and receiving applications. Sequence numbers are used to ensure that packets arrive in the right order, no packets are missing and to detect duplicates. We will explain this further in the following section. Acknowledgments are used to ensure that packets have arrived at their destination. The TCP header length field basically shows where the data field begins. This is because the options field is variable. After the reserved space, there are six 1-bit fields for different flags. page 2 of 8

3 TCP is using a flow control protocol called sliding window. The window size field specifies the number of bytes that may be sent to the receiver. The window size may change trough out the lifetime of a connection. A value of zero means that the receiver doesn t wish to receive more data. This is useful if the receiver becomes overloaded. In the checksum field, a checksum of the whole header including the data is calculated. The urgent data pointer is for the signaling mechanism in the TCP protocol, this field indicates the last byte of the urgent data. The options field is variable and allows adding extra functionality to the header. 3.2 TCP Operation A TCP connection undergoes three main stages throughout its lifetime. First a connection must be established, this is the connection establishment phase. Secondly the connection enters the establishment phase, it is in this phase that data is sent and received. And last the connection release phase where the connection is terminated and all resources are released Connection Establishment For two hosts to be able to communicate, they first need to establish a connection. There are some problems that need to be handled in order for a safe connection to be established. One such problem is the fact that the connecting host, sending a connection request message to another host cannot know if the connection request was received or not, as we know the internetwork is not reliable and packets can be lost or reappear after some time, these are called duplicates which is the second concern that has to be handled. How can the receiving host know that the connection request is actually a request and not an old duplicate. To solve these problems the TCP protocol is using a method called the three-way handshake [7], the basic idea is that the sending host sends a synchronization (SYN) package to the receiving host. The receiving host responds by sending back his own SYN packet together with an acknowledgement (ACK), forming an SYN-ACK. This is to show to the sending host that it has received the SYN. Lastly the sending host responds with an ACK. Totally three packets have been transmitted between the sending and receiving host, hence the name three-way handshake. The whole process is illustrated in Figure 3.2. Figure 3.2 Three-way handshake. [7] What is important to note is the content of the SYN and ACK packages. The SYN packet is basically a TCP header containing no data, with the SYN flag set and a sequence number, x. The sequence number is a randomly chosen 32- bit value. When Host 2 replies it also chooses its own randomly chosen sequence number, y and also in the Acknowledgement number field Host 2 sets an acknowledgement number which is the sequence number from the SYN message of Host 1 incremented by one (x+1). Note that the ACK number from Host 2 is the sequence number that Host 1 is expecting. This prevents duplicate messages from affecting the connection establishment process. Figure 3.3 shows a worst case scenario with a duplicate SYN message and an ACK reappearing at Host 2 s connection. Figure 3.3 scenario. [7] When Host 2 gets a duplicate SYN message it replies accordingly as it would be a real SYN. But Host 1 will not expect an SYN-ACK with an ACK-number of x, thus rejecting the connection attempt. When the other duplicate page 3 of 8

4 arrives at Host 2 with an ACK-number z, Host 2 knows that this to must be a duplicate because it is expecting the ACK number y and not z. Also there is the problem of lost packets, packets that don t arrive at all. This is solved by the use of timers, every time a SYN is sent the Host is starting a timer. If the timer reaches zero and there is no response from the other Host, the packet is considered to be lost, and resent Connection Release When a host decides to terminate a connection, it cannot do so without notifying the other host. This would result in a half open connection, which is the main concern in the release connection phase. To avoid half open connections we need a protocol that is similar to the three-way handshake. Let s say that Host 1 wants to terminate the connection with Host 2. Similar to the connection establishment phase, Host 1 sends a Disconnection Request to Host 2. Similar to a SYN, this is a packet with an FIN flag set in the TCP header with no data. When Host 2 receives the disconnection request from Host 1, it also sends an FIN packet back to Host 1. Host 1 will then respond with an ACK. This may appear to be simple at first, but there are some things that can go wrong here. What if Host 1 never receives a FIN from Host 2? Should Host 1 close the connection? Another possible scenario is when the last ACK from Host 1 is lost. Should Host 2 release the connection anyway? Again like lost SYN packets these are solved by using timers. But here we do not always retransmit a disconnection request after a timeout. Host 2 will release the connection after the FIN has timed out, this will solve the last lost ACK scenario and also the worst case where all messages after the first FIN was received are lost. Host 2 can close the connection after the timeout because we know that Host 1 will also close its connection after N retransmission attempts. The last lost ACK problem really doesn t matter because Host 1 has already released, and host 2 will do so after the time out. These scenarios are illustrated in the figure below. Figure 3.4 Connection release. [7] Sliding Window Sliding window is a protocol for efficient transferring of data between two connected hosts. Because always waiting for an acknowledgement before sending the next packet of data is not efficient, the sliding window lets a host send several packets of data before receiving an acknowledgement. The size of the window, tells the sender how many unacknowledged packets it can have. The size of the window is variable and can change during the connections lifetime, depending on the bandwidth and delays of the network. The point is to always have a window size big enough so that the sender can keep on sending until the first acknowledgement has arrived. There are two popular versions of the sliding window protocol, Go Back N and Selective Repeat [7]. The difference between them is how they handle lost and corrupt packets. Go Back N will simply discard all frames page 4 of 8

5 that are received after a lost packet. After the lost packets timer has expired the sender will retransmit it and all the discarded frames. A more efficient way is to use selective repeat, instead of discarding, the out of order packets are buffered. And only the lost packets will have to be retransmitted by the sender. For more efficiency selective repeat can use negative acknowledgements (NAK) when it detects that a packet must be lost. This is faster because the sender does not have to wait for a timeout. Figure below shows an example of Go Back N (a) and Selective Repeat (b). Figure 3.5 Sliding window. [7] 4. Future TCP/IP According to the metrics presented by Claffy [1], While almost 60% of packets are 44 bytes or less, constituting a total of 7% of the byte volume, over half of the bytes are carried in packets of size 1500 bytes or larger. These results mean that even during the young period of the internet (year 1998) the volume of packets was growing fast as the demand of the users and the amount of them was developing logarithmically. The demand is bigger and more exigent nowadays. Media, streaming, p2p and other services with an important load of traffic with heavy content are the user s needs to be covered. TCP will not be able to support an upcoming demand of larger packages as shown in Figure 4.1. Figure 4.1 Packet size demand. [3] There are a few assumptions from the design of the TCP protocol that are needed to be remembered in order to set the future [2]: TCP has been designed for a wired network but these days the dominion of wireless networks is irrefutable. The main problem with wireless networks is that they have a higher bit error rate (BER) than wired networks. This brings packet losses due to network congestion which is not anymore a problem of just bit-level corruption. TCP protocol understands that there is only one best way of routing since packet reordering is considered to occur in a minor scale. All the packets are expected to follow the same path in their way through the network. The network is supported by physical circuits with a limited bandwidth and TCP works under this rule. TCP considers the bandwidth constant and uses an end-to-end control loop for sending rates. If the bandwidth is changing in time TCP may take decisions with less risk for data loss. Buffers have first-in first-out (FIFO) philosophy and TCP is more effective if the buffer size that resolves connections within the switches is in the same order of magnitude of the bandwidth s delay, as a result of the associated link. Session duration makes a strong impact on TCP s efficiency which assumes that every TCP session has a specific number of time slots. The developing of short term sessions in applications such as short Web transfers makes more damage with an everyday more obsolete transport control protocol. page 5 of 8

6 Large payloads take place when there is a low bandwidth and the usual 40 bytes per packet overhead is assumed too inefficient. Interaction with other TCP sessions is a strategy planed to optimize the network. As it is planed every TCP session shares bandwidth with other connections but this interaction does not work properly with flow-control protocols which derives in unpredictable results and poor network efficiency. 4.1 Future TCP Modified TCP stacks Recently, several new protocols have made their presence in collaboration with Web server systems which stacks can interoperate with the regular TCP clients. This offers superior download ability over the traditional TCP. This is possible due to some changes in the flow control systems of TCP. According to [2] The modified implementation may use a lower initial RTT estimate to provide a more aggressive startup rate, and a more finely grained RTT timer system to allow the sender to react more quickly to network state changes. This kind of modification in the implementation as well as using a larger window size or changing the slow-start mode can produce favorable sending rate, up to triple in respect to previous version. This complex technique may force network buffers increasing pressure on them and forcing users to act more aggressive in their package transmissions. This aggressive behavior reduces traditional TCP session s sending rate, opens additional space in the network and increases selfsending rate. Despite this visible upgrade of the protocol into a dynamic equilibrium due to cooperation between sending systems the number of packet losses increase using these methods because forcing the network to saturation with its own packets produces less sensitivity to network congestion metrics. The consequence of large volume of packet transmissions from one session being lack of pressure from competitor sessions is a less efficiency of the modified TCP stack than the standard TCP. In the production environment the imposition of penalties to concurrent TCP sessions indicates that the modified TCP stacks should be preserved only for private use networks because its benefit to the public Internet is in doubt TCP Evolution As it has been shown in the comparison of traditional TCP and modified TCP stack there is a balance to keep between network congestion collapse and innovation avoiding competing protocols get into race under no control. Under tests with the more aggressive TCP optimized advances which result in congestion collapse of the network and the low network efficiency of the traditional TCP with the advantage of low jitter retransmission we have to take a compromise solution. Coherency needs to be preserved to keep the balance in the network, interaction between protocols and data flow between these different uses of similar architectures. Flexible changes may come to the transport protocols with a continuous innovation. The acceptable changes are driven under the trust between the competing user demands and not by the rigid rules of sharing networks FAST TCP In 2003 a group of scientist from different universities and private companies published their investigations and results about a new TCP congestion control algorithm for highspeed long-latency networks [6]. The work has its basis on the fact that the availability of resources and the set of competing users vary over the time unpredictably. It was necessary to use feedback control tools to adapt dynamically the rates of congestion. The solution adopted was using a delay-based congestion control because queuing delays provide more accurate results than loss probability and packet loss in networks since these phenomenon s are unusual. While packet loss provides only one bit of information plus noise, queuing delays have multi-bit information. This way an equation for stabilization at steady state is easier to formulate. Here are presented some problems that appear at large windows: packet and flow level modeling, equilibrium, dynamics. According to [3] The window of source i increases by 1 packet per RTT and decreases per unit time by packets where pkts/sec. being the round-trip time and is the (delayed) end-to-end loss probability, in period t. As long as the mathematical explanation develops two equations describe flow-level dynamics and equilibrium. There are currently other algorithms to improve congestion control such us HSTCP, STCP and Reno. Despite their different aspect they all have similarities concerning structure and dynamics at flow-level. Their congestion windows evolve all according to: The controllable parameters are: : gain, controls stability and responsiveness and does not affect equilibrium properties. page 6 of 8

7 : marginal utility function, controls equilibrium rate allocation and its fairness. : congestion measure, is limited to loss probability or queuing delay and its dynamics are set up inside the network. The equilibrium at large window is difficult to maintain in practice because the bandwidth-delay product increases. It has been tested that when a source increases its window too slowly decreases too drastically, which is a problem for equilibrium. This causes an oscillatory behavior at packet and flow level. At packet level, bottleneck queues get worse as bandwidth-delay product increases. On the other hand, at flow level, system dynamic are unstable at large bandwidth-delay products. To solve this there must be multi-bit feedback but some problems at flow level can be still distinguished: equation-based control which needs explicit estimation of end-to-end probability and the other one is flow dynamics. Flow dynamics can bring instability to feedback as feedback delay increases The Delay-based solution What determines the variations on the network as it increases its capacity, are the dynamics of the feedback system, and must be balanced in presence of delay. Another advantage over loss probability is that dynamics are scaled to fit network s capacity as it grows. The delay-based solution estimates the distance of the state from the equilibrium point Architecture According to [6] TCP congestion s control protocol is split into four independent functionally components: Data control: determines which packets to transmit. Window control: determines how many packets to transmit. Business control: determines when to transmit. Estimation component: takes decisions to control the other three components. This last component meant to rule the others handles two fragments of feedback information for every packet sent. One is a multi-bit queuing delay and the other one is a onebit loss-or-no-loss flag, understanding flag as an indicator. These two separate pieces of information are used by the other three components. Data control chooses the next packet to send from 3 different sources: new packets, packets with negative acknowledgement, and packets already transmitted but not acknowledged. Packet regulation at RTT time domain is done by window control. Finally business control makes the transmission fluent according to the available bandwidth. To fulfill FAST TCP two extra mechanisms are used one to supplement selfclocking in streaming out individual packets and the other to increase window size smoothly in smaller bursts. A detailed explanation can be found at [6] Most modern TCP works The latest reviews in TCP protocol [9] show that a compromise solution must be adopted between the different new TCP implementations as shown in Table 1. Table 1 TCP Most modern implementations. [9] 4.2 Future IP It is on some developers mind the future of mobile communications an IP based network that will be part of the 4G Mobile Networks. This forces to make improvements to change from the traditional packetswitched networks to a packet based wireless IP network and also is seen as the motor-breath for new communication systems and technologies development. The infrastructures of a wireless network are not so much physically dependent as wired networks, which brings economic and security issues to debate just to come up with a compromise relation between users and operators. As is explained by Kurtansky [3] in his paper, AAAC Arch. stands for Authentication, Authorization, Accounting and Charing Architecture. According to the paper this architecture has been implemented to facilitate the deployment of a ubiquitous mobile IPv6-based, QoS-aware infrastructure through a flexible and evolutionary AAAC Architecture. In order to give support to both AAAC client and AAAC server [3] focuses on both DIAMETER client and DIAMETER server. Client daemon is in charge of accepting registration requests and forwarding to the server daemon. It uses User Registration Protocol (URP) and DIAMETER protocol. The server daemon has the task of authentication, authorization, and accounting based on the user profile information as show in Figure 4.2. page 7 of 8

8 Figure 4.2. AAAC System. [3] 5. Results and Conclusion Since the giant step growing of the internet in the late 1990s it has been demonstrated that TCP protocol may evolve into a more flexible protocol. Since upcoming protocols such as modified TCP stacks appeared it has been tested that a balance must to be kept between network sharing with other competing protocols and innovations in these new protocols, in order not to get network congestion collapse. Fast TCP discards the old loss probability model and uses a delay-based model. A balanced zero-state is set at the delay queue which tries to control flow-level and bandwidth. A multi-bit feedback control system is used to be more accurate than the regular TCP. It is a fact the 4G Mobile Networks will be IP-based and this new concept of communications technology gives a new role to the internet protocol. The complexity of networks evolution grows in parallel to the digitalization era started in the 1990s. The only way to support the strong increase of users demand and the fast technological development lies on a twist on the concept of the future networks. To cover future needs we have to implement a cohesion point from improved actual tools. That is the reason why extreme changes are not accepted. Using IP as the protocol for 4G Mobile Networks is an example of technology improvement with improved actual tools. FAST TCP shows how a twist in the observer perspective, since a scientific point of view, can bring important developments without using new technological advances but a better knowledge of mathematics. 6. REFERENCES [1] Claffy, K., Miller, G., Thompson, K.,(1998), The Nature of the Beast: Recent Traffic Measurements from an Internet Backbone, INET'98 Proceedings, Internet Society. Available at: [2] Huston G., Telstra (2000), The future of TCP, The internet protocol Journal, Cisco Systems, Vol.3, Nº3. [3] Kurtansky, P. et al (2004), Extensions of AAA for Future IP Networks, IEEE Communications Society, Computer Engineering and Networks Lab TIK, Zürich, Switzerland; Institute FOKUS, Berlin, Germany; Universidad Carlos III Madrid, Spain; University of Stuttgart, Germany. [4] Postel, J. (1981), Transmission Control Protocol, RFC 793. [5] University of South California (1980), DOD Standard Internet Protocol, RFC 760. [6] Wei D. et al, (2004), Fast TCP, Engineering & Applied Sciences, Caltech. [7] Tanenbaum, Andrew S. ( ). Computer Networks (Fourth ed.). Prentice Hall. ISBN :Chapter 3.4 and 6. [8] Cerf, V., and R. Khan, A Protocol for Packet Network Intercommunication (1974) [9] B. Qureshi, M. Othman,(2009), Progress in Various TCP variants, IEEE, and N. A. W. Hamid. page 8 of 8

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