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Modify the simplex-talk socket program so that it uses UDP as the transport protocol, rather than TCP. You will have to change SOCK_STREAM to SOCK_DGRAM in both client and server. Then, in the server, remove the calls to listen() and ac\(\operatorname{cept}()\), and replace the two nested loops at the end with a single loop that calls recv() with socket s. Finally, see what happens when two such UDP clients simultaneously connect to the same UDP server, and compare this to the TCP behavior.

Short Answer

Expert verified
Change `SOCK_STREAM` to `SOCK_DGRAM`, remove `listen()` and `accept()`, use `recv()` in a loop, then test with multiple clients.

Step by step solution

01

Change the transport protocol to UDP

Modify the initial socket creation in both the client and server programs. Replace the `SOCK_STREAM` parameter with `SOCK_DGRAM` in the `socket()` function call.
02

Modify Server - Remove listen() and accept()

In the server code, remove the calls to `listen()` and `accept()`. These functions are specific to TCP and are not required for UDP.
03

Modify Server - Adjust to use recv()

Replace the two nested loops in the server code with a single loop that continuously calls `recv()` using the socket `s`. This allows the server to receive messages from any client.
04

Test with multiple clients

Run the modified server and then start two UDP clients to see how the server handles simultaneous connections. Observe the behavior compared to when TCP was used.
05

Compare UDP and TCP behavior

Note how UDP behaves when multiple clients send data simultaneously. Unlike TCP, UDP does not guarantee order or delivery, so packets might arrive out of order or get lost.

Key Concepts

These are the key concepts you need to understand to accurately answer the question.

UDP Protocol
UDP, or User Datagram Protocol, is a communication protocol used across IP networks. Unlike TCP, UDP is connectionless which means it doesn't establish a connection before sending data. Instead, UDP sends datagrams directly to the recipient without ensuring they are received.

This makes UDP faster and more efficient for applications where speed is crucial, such as live broadcasts or online gaming. However, it also means data may be lost, duplicated, or delivered out of order.

Since UDP doesn't have built-in error checking or retransmission, applications using UDP must handle potential data issues themselves. This trade-off between speed and reliability is what defines UDP's unique utility in network communication.
Socket Programming
Socket programming is a way to connect two devices over a network to communicate with each other. It involves the use of sockets, which are endpoints for sending and receiving data between devices.

In Python, for example, you use the `socket` module to create a socket object. When you create the socket, you specify the socket type: `SOCK_STREAM` for TCP or `SOCK_DGRAM` for UDP. Once the socket is created, you can use various methods to connect, send, and receive data.

Here are some common socket methods:
  • `socket()` to create a new socket.
  • `bind()` to associate the socket with a specific network interface and port.
  • `sendto()` and `recvfrom()` for sending and receiving data using UDP.
  • `send()` and `recv()` for sending and receiving data using TCP.
Socket programming is a fundamental skill for network communication, enabling the creation of robust client-server applications.
TCP vs UDP Differences
TCP (Transmission Control Protocol) and UDP have several key differences:
  • Connection: TCP establishes a connection before data transfer, while UDP is connectionless.
  • Reliability: TCP guarantees data delivery and order, but UDP does not.
  • Speed: UDP is generally faster because it has less overhead.
  • Usage: TCP is used where reliability is key (e.g., web browsing, email), while UDP is used where speed is critical (e.g., live streaming).
These differences significantly affect their use cases.

For example, in TCP, `listen()` and `accept()` methods are used to establish a connection, but these are unnecessary in UDP. UDP uses `recvfrom()` to receive data from any client, which emphasizes its connectionless design.

Understanding these differences helps in choosing the right protocol based on the application's requirements.
Network Communication
Network communication involves the exchange of data between devices over networks. This can be within the same local area network (LAN) or across a wide area network (WAN) like the internet.

Protocols such as TCP and UDP, defined by the Internet Protocol (IP) suite, enable this communication. They specify how data is formatted, transmitted, and received, ensuring devices can communicate correctly.

Network communication includes several layers:
  • Application Layer: Where user applications operate.
  • Transport Layer: Manages data delivery (e.g., TCP/UDP).
  • Internet Layer: Handles logical addressing and routing (e.g., IP).
  • Link Layer: Manages physical connections (e.g., Ethernet).
Each layer has specific responsibilities, working together to provide comprehensive network communication.
Client-Server Model
The client-server model is a widely used structure in network communication. In this model, a server provides resources, services, or applications, while clients request them.

The server typically waits for client requests and responds accordingly. For instance, a web server serves web pages to browsers. The server's socket listens for incoming connections or data, while client sockets initiate communication by sending requests.

In the UDP version of a client-server program, multiple clients can send data to a single server simultaneously. Unlike TCP, where `listen()` and `accept()` manage connections, UDP's `recvfrom()` simply receives data from any sender.

This model allows scalable and manageable network applications, making it a cornerstone of modern networking.

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Most popular questions from this chapter

The Unix utility traceroute, or its Windows equivalent tracert, can be used to find the sequence of routers through which a message is routed. Use this to find the path from your site to some others. How well does the number of hops correlate with the RTT times from ping? How well does the number of hops correlate with geographical distance?

Suppose a host has a 1-MB file that is to be sent to another host. The file takes 1 second of CPU time to compress \(50 \%\), or 2 seconds to compress \(60 \%\). (a) Calculate the bandwidth at which each compression option takes the same total compression + transmission time. (b) Explain why latency does not affect your answer.

For the following, assume that no data compression is done; this would in practice almost never be the case. For (a)-(c), calculate the bandwidth necessary for transmitting in real time: (a) Video at a resolution of \(640 \times 480,3\) bytes/pixel, 30 frames/second. (b) \(160 \times 120\) video, 1 byte/pixel, 5 frames/second. (c) CD-ROM music, assuming one CD holds 75 minutes' worth and takes \(650 \mathrm{MB}\). (d) Assume a fax transmits an \(8 \times 10\)-inch black-and-white image at a resolution of 72 pixels per inch. How long would this take over a 14.4-Kbps modem?

Suppose hosts A and B are connected by a link. Host A continuously transmits the current time from a high-precision clock, at a regular rate, fast enough to consume all the available bandwidth. Host \(\mathrm{B}\) reads these time values and writes them each paired with its own time from a local clock synchronized with A's. Give qualitative examples of B's output assuming the link has (a) high bandwidth, high latency, low jitter (b) low bandwidth, high latency, high jitter (c) high bandwidth, low latency, low jitter, occasional lost data For example, a link with zero jitter, a bandwidth high enough to write on every other clock tick, and a latency of 1 tick might yield something like \((0000,0001)\), \((0002,0003),(0004,0005)\).

Assume you wish to transfer an \(n\)-byte file along a path composed of the source, destination, seven point-to-point links, and five switches. Suppose each link has a propagation delay of \(2 \mathrm{~ms}\), bandwidth of \(4 \mathrm{Mbps}\), and that the switches support both circuit and packet switching. Thus you can either break the file up into 1-KB packets, or set up a circuit through the switches and send the file as one contiguous bit stream. Suppose that packets have 24 bytes of packet header information and 1000 bytes of payload, that store-and-forward packet processing at each switch incurs a 1 -ms delay after the packet has been completely received, that packets may be sent continuously without waiting for acknowledgments, and that circuit setup requires a 1-KB message to make one round-trip on the path incurring a 1-ms delay at each switch after the message has been completely received. Assume switches introduce no delay to data traversing a circuit. You may also assume that file size is a multiple of 1000 bytes. (a) For what file size \(n\) bytes is the total number of bytes sent across the network less for circuits than for packets? (b) For what file size \(n\) bytes is the total latency incurred before the entire file arrives at the destination less for circuits than for packets? (c) How sensitive are these results to the number of switches along the path? To the bandwidth of the links? To the ratio of packet size to packet header size? (d) How accurate do you think this model of the relative merits of circuits and packets is? Does it ignore important considerations that discredit one or the other approach? If so, what are they?

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