FALL 2013 ASSIGNMENT MC0087-INTERNETWORKING WITH TCP/IP 1. What is application layer?

Discuss any four protocols with its role that are used for exchange of information Answer: Application Layer The Application layer allows applications to access the services of the other layers, and it defines the protocols that applications use to exchange date. The Application layer contains may protocols, and more are always being developed. The most widely known Application layer protocols help users exchange information: The Hypertext Transfer Protocol (HTTP) transfers files that make up pages on the Word Wide Web The Simple Mail Transfer Protocol protocol resolves a host name, such www.cisco.com, to an IP address and copies name information between DNS servers. The Routing Information Protocol (RIP) is a protocol that routers use to exchange routing information on an IP network. The Simple Network Management Protocol (SNMP) collects and exchanges network management information between an network management console and network devices such as routers, bridges, and server. Windows Sockets and NetBIOS are examples of Application layer interfaces for TCP/IP application. (SMTP) transfers mail messages and attachements. Additionally, the following Application layer protocols help you use and manage TCP/IP networks: The Domain Name System (DNS) protocol resolves a host name, such www.cisco.com, to an IP address and copies name information between DNS servers. The Routing Information Protocol (RIP) is a protocol that routers use to exchange routing information on an IP network. The Simple Network Management Protocol (SNMP) collects and exchanges network management information between an network management console and network devices such as routers, bridges, and server. Windows Sockets and NetBIOS are examples of Application layer interfaces for TCP/IP application 2.Brief the following a. SLIP b. PPP

A) Answer: A) SLIP
The Serial Line Internet Protocol (SLIP) is an encapsulation of the Internet Protocol designed to work over serial ports and modem connections. It is documented in RFC 1055. On personal computers, SLIP has been largely replaced by the Point-to-Point Protocol (PPP), which is better engineered, has more features and does not require its IP address configuration to be set before it is established. On microcontrollers, however, SLIP is still the preferred way of encapsulating IP packets due to its very small overhead. SLIP modifies a standard TCP/IP datagram by appending a special "SLIP END" character to it, which distinguishes datagram boundaries in the byte stream. SLIP requires a serial port configuration of 8 data bits, no parity, and either EIA hardware flow control, or CLOCAL mode (3-wire null-modem) UART operation settings. SLIP does not provide error detection, being reliant on upper layer protocols for this. Therefore SLIP on its own is not satisfactory over an error-prone dial-up connection. It is however still useful for testing operating systems' response capabilities under load (by looking at flood-pingstatistics). SLIP is also currently used in the Blue Core Serial Protocol for communication between Bluetooth modules and host computers.[1]

B) PPP

In networking, the Point-to-Point Protocol (PPP) is a data link protocol commonly used in establishing a direct connection between two networking nodes. It can provide connection authentication, transmission encryption, and compression. PPP is used over many types of physical networks including serial cable, phone line, line, cellular, specialized radio links, and fiber optic links such as SONET. PPP is also used over Internet access connections (now marketed as "broadband"). Internet service providers (ISPs) have used PPP for customer dial-up access to the Internet, since IP packets cannot be transmitted over a modem line on their own, without some data link protocol. Two encapsulated forms of PPP, Point-to-Point Protocol over Ethernet (Pope) and Point-to-Point Protocol over ATM (Poppa) are used most commonly by Internet Service Providers (ISPs) to establish digital (DSL) Internet service connection with customers. PPP is commonly used as a data link layer protocol for connection over synchronous and asynchronous circuits, where it has largely superseded the older Serial Line Internet Protocol(SLIP) and telephone company mandated standards (such as Link Access Protocol, Balanced(LAPB) in the X.25 protocol suite). PPP was designed to work with numerous layer protocols, including Internet Protocol (IP), TRILL, Novell's Internetwork Packet Exchange (IPX), NBF and AppleTalk.

3. What are congestion? .Mention two algorithms to overcome Congestion Answer: TCP is the popular transport protocol for best-effort trafc in Internet. However, TCP is not well suited for many applications such as streaming multimedia, because TCP congestion control algorithms introduce large variations in the congestion window size (and corresponding large variations in the sending rate). Such variability in the sending rate is not acceptable to many multimedia applications. Hence, many multimedia applications are built over UDP and use no congestion control at all. The absence of congestion control in applications built over UDP may lead to congestion collapse on the Internet. In addition, the UDP ows may starve any competing TCP ows. To overcome these adverse effects, congestion control needs to be incorporated into all applications using the Internet, whether at the transport layer or provided by the application itself. Furthermore, the congestion control algorithms must be TCP-friendly, i.e. the TCP-friendly ows should not gain more throughput than competing TCP ows in the long run. Thus, in recent years, many researchers have focussed on developing TCP-friendly transport protocols which are suitable for many applications that currently use UDP. In this direction, IETF is currently working on developing a new protocol called, Datagram Congestion Control Protocol (DCCP), that provides an unreliable datagram service with congestion control. DCCP is designed to use any suitable TCP friendly congestion control algorithm. With a multitude of TCPfriendly congestion control algo- rithms available, some important questions that need to be answered are: What are the strengths and weakness of the various TCP-friendly algorithms? Is there a single algorithm which is uniformly superior over other algorithms. The rst step in answering the se questions is to study the short-term and long-term behavior of these algorithms. Although the goal of all TCP-friendly algorithms is to emulate the behavior of TCP in the long term, these algorithms may have an adverse impact in the short-term on competing TCP ows. Since TCP-friendly algorithms are designed for smoother sending rates than TCP, these algorithms may react slowly to new connections that share a common bottleneck link. Such a slower response may have a deleterious effect on TCP ows. For example, a TCP connection suffering losses in its slow start phase may enter the congestion avoidance phase with a small window, and consequently obtain lesser throughput than other competing ows. Hence, it is clear that a detailed study is required on the short-term (transient)behavior of TCP-friendly ows in addition to their long-term behavior. In this paper, we study the transient behavior of three TCP-friendly congestion control algorithms: general AIMD congestion control, TFRC and binomial congestion control algorithm . Prior work has studied the transient behavior of these algorithms when RED queues are used at the bottleneck link. However, as droptail queues are still widely used in practice, in this paper we study the transient behavior of these algo rithms with droptail queues. Past work has also identied certain unfairness of AIMD and binomial congestion control algorithms to TCP with droptail queues, but has not identied the reasons for this unfairness. In this paper, we analyze the reasons for th is unfairness, and validate the analysis by simulations. The rest of the paper is organized as follows. In Section II, we briey overview the various TCP-friendly congestioncontrol algorithms proposed in literature. In Section III, we dene the transient behaviors studied in this paper, and analyze the expected transient behaviors of the various TCP-friendly congestion control algorithms. Section IV analyzes in detail the reasons for unfairness of AIMD and binomial

congestion control algorithms with droptail queues. We present our sim- ulation results in Section V, and we conclude in Section VI. few algorithms to overcome congestion A. Transient behaviors evaluated in the paper B. Equation-Based Congestion Control Algorithm C. General AIMD-Based Congestion Control Algorithms D. Binomial Congestion Control Algorithm 4.What do you mean by OPTION NEGOTIATION? Explain with an Example Answer: Option Negotiation Using internal commands, TELNET in each host is able to negotiate options. The starting base of negotiation is the NVT capability: each host to be connected must agree to this minimum. Every option can be negotiated by the use of the four command codes WILL, WON'T, DO, DON'T described above. In addition, some options have sub-options: if both parties agree to the option, they use the SB and SE commands to manage the sub-negotiation. Here is a simplified example of how option negotiation works. n To use an option, the client and server must negotiate and agree to use it. The tools for negotiatio n are the commands we've already talked about. One side - usually, but not always, the client - sends a "WILL X" packet (WILL is decimal value 251), where X is the option it wants to use (numeric values for X are given on p. 373). The other side will respond with a DO X or a DON'T X, depending on whether it is willing to support the option. Alternatively, the first side could send a "DO X" packet, in which case the response is either "WILL X" or "WON'T X". TELNET is one of the programs that requires the TCP Urgent Data function, because buffers may fill up (for example, if a program being executed is in an infinite loop), and the server's program will stop reading data - including the "IP" command the user sent after he realized what was happening. The packet with the "Terminate" command can be sent as "urgent data" at the TCP level; that will bypass the standard TCP flow controls and enable an out-of-control process to be stopped. 5.What is domain name resolution? Discuss the domain name resolution process Answer: The Domain Name System (DNS) is a way to resolve meaningful and easy-to-remember names to IP addresses. Because millions of sites are connected to the Internet, maintaining one central list of the name to-

IP-address relationships across the Internet is unrealistic. The DNS system was designed to coordinate and distribute the resolution load. The two major tasks that DNS provides are: 1. IP address resolution to hosts on the Internet, for local hosts 2. IP address resolution to hosts on the local network, for other hosts on the Internet Domain Name Resolution The domain name resolution process can be summarized in the following steps: 1. A user program issues a request such as the gethostbyname() system call (this particular call asks for the IP address of a host by passing the host name) or the gethostname() system call (which asks for a host name of a host by passing the IP address). 2. The resolver formulates a query to the name server. (Full resolvers have a local name cache to consult first; stub revolvers do not.) 3. The name server checks to see if the answer is in its local authoritative database or cache, and if so, returns it to the client. Otherwise, it queries other available name servers, starting down from the root of the DNS tree or as high up the tree as possible. 4. The user program is finally given a corresponding IP address (or host name, depending on the query) or an error if the query could not be answered. Normally, the program will not be given a list of all the name servers that have been consulted to process the query. Domain name resolution is a client/server process. The client function (called the resolver or name resolver) is transparent to the user and is called by an application to resolve symbolic high-level names into real IP addresses or vice versa. The name server (also called a domain name server) is the server application providing the translation between high-level machine names and the IP addresses. The query/reply messages can be transported by either UDP or TCP. 6.Discuss the importance of SIP technology. List and brief any four primary functions of session initiation protocol Answer: The Session Initiation Protocol (SIP), defined in RFC 3261 [6], is an application level signaling protocol for setting up, modifying, and terminating real-time sessions between participants over an IP data network. SIP can support any type of single-media or multi-media session, including teleconferencing. SIP is just one component in the set of protocols and services needed to support multimedia exchanges over the Internet. SIP is the signaling protocol that enables one party to place a call to another party and to negotiate the parameters of a multimedia session. The actual audio, video, or other multimedia content is exchanged between session participants using an appropriate transport protocol. In many cases, the transport protocol to use is the Real-Time Transport Protocol (RTP). Directory access and lookup protocols are also needed. The key driving force behind SIP is to enable Internet telephony, also referred to as Voice over IP (VoIP). There is wide industry acceptance that SIP will be the standard IP signaling mechanism for voice and multimedia calling services. Further, as older Private Branch Exchanges (PBXs) and network switches are phased out, industry is moving toward a voice networking model that is SIP signaled, IP based, and packet switched, not only in the wide area but also on the customer premises [2, 3]. SIP supports five facets of establishing and terminating multimedia communications: User location: Users can move to other locations and access their telephony or other application features from remote locations. User availability: This step involves determination of the willingness of the called party to engage in communications. User capabilities: In this step, the media and media parameters to be used are determined. Session setup: Point-to-point and multiparty calls are set up, with agreed session parameters. Session management: This step includes transfer and termination of sessions, modifying session parameters, and invoking services.

SIP employs design elements developed for earlier protocols. SIP is based on an HTTP-like request/response transaction model. Each transaction consists of a client request that invokes a particular method, or function, on

the server and at least one response. SIP uses most of the header fields, encoding rules, and status codes of HTTP. This provides a readable text-based format for displaying information. SIP incorporates the use of a Session Description Protocol (SDP), which defines session content using a set of types similar to those used in Multipurpose Internet Mail Extensions (MIME). SIP Components and Protocols A system using SIP can be viewed as consisting of components defined on two dimensions: client/server and individual network elements. RFC 3261 defines client and server as follows: Client: A client is any network element that sends SIP requests and receives SIP responses. Clients may or may not interact directly with a human user. User agent clients and proxies are clients. Server: A server is a network element that receives requests in order to service them and sends back responses to those requests. Examples of servers are proxies, user agent servers, redirect servers, and registrars. The individual elements of a standard SIP configuration include the following: User Agent: The user agent resides in every SIP end station. It acts in two roles: User Agent Client (UAC): Issues SIP requests User Agent Server (UAS): Receives SIP requests and generates a response that accepts, rejects, or redirects the request Redirect Server: The redirect server is used during session initiation to determine the address of the called device. The redirect server returns this information to the calling device, directing the UAC to contact an alternate Universal Resource Identifier (URI). A URI is a generic identifier used to name any resource on the Internet. The URL used for Web addresses is a type of URI. See RFC 2396 [1] for more detail. Proxy Server: The proxy server is an intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, meaning that its job is to ensure that a request is sent to another entity closer to the targeted user. Proxies are also useful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it. Registrar: A registrar is a server that accepts REGISTER requests and places the information it receives (the SIP address and associated IP address of the registering device) in those requests into the location service for the domain it handles. Location Service: A location service is used by a SIP redirect or proxy server to obtain information about a callee's possible location(s). For this purpose, the location service maintains a database of SIP-address/ IPaddress mappings.

The various servers are defined in RFC 3261 as logical devices. They may be implemented as separate servers configured on the Internet or they may be combined into a single application that resides in a physical server.