Computer Networks Assignment 4 Solution

Description

Preamble

This handout starts out by giving a brief introduction to TCP in Section 1 before moving onto the problem statement. So, don’t get scared if it seems very long at first. Also, note that unlike the previous assignments in this course, this assignment is relatively more open ended and as a result you have more freedom in your implementation details as long as you adhere to the protocol specified in section 2.1.

Updates

• Any data packet correspondence after the initial handshake should always have ACK flag set and the correct ACK number.

• Please make sure your Makefile places compiled *.class files under src as well.

Learning Outcomes

After completing this programming assignment, students should be able to:

• Write code that builds reliable data transfer for application running atop unreliable UDP sockets

• Understand and implement retransmission timeouts

• Understand and implement one’s complement checksum for data integrity

1 Background

1.1 Introduction

The Transmission Control Protocol (TCP) standard is defined in the Request For Comment (RFC) standards document number 793 by the Internet Engineering Task Force (IETF). The original specification written in 1981 was based on earlier research and experimentation in the original ARPANET. The design of TCP was heavily influenced by what has come to be known as the “end-to-end argument”.

As it applies to the Internet, the end-to-end argument says that by putting excessive intelligence in physical and link layers to handle error control, encryption or flow control you unnecessarily complicate the system. This is because these functions will usually need to be implemented at the endpoints anyways, so duplication of this functionality in the intermediate points can be a waste. The result of an end-to-end network then, is to provide minimal functionality on a hop-by-hop basis and maximal control between end- to-end commu-nicating systems. The end-to-end argument helped determine the design of various components of TCP’s reliability, flow control, and congestion control algorithms. The following are a few important characteristics of TCP.

1.2 Byte Stream Delivery

TCP interfaces between the application layer above and the network layer below. When an application sends data to TCP, it does so in 8-bit byte streams. It is then up to the sending TCP to segment or delineate the byte stream in order to transmit data in manageable pieces to the receiver. It is this lack of record boundaries which give it the name byte stream delivery service.

1.3 Connection-oriented Approach

Before two communicating TCP endpoints can exchange data, they must first agree upon the willingness to communicate. Analogous to a telephone call, a connection must first be made before two parties exchange information.

1.4 Reliability

A number of mechanisms help provide the reliability TCP guarantees. Each of these is described briefly below.

• Checksums: All TCP segments carry a checksum, which is used by the receiver to detect errors with either the TCP header or data.

• Duplicate data detection: It is possible for packets to be duplicated in packet switched network; therefore TCP keeps track of bytes received in order to discard duplicate copies of data that has already been received.

• Retransmissions: In order to guarantee delivery of data, TCP must implement retransmission schemes for data that may be lost or damaged. The use of positive acknowledgements by the re-ceiver to the sender confirms successful reception of data. The lack of positive acknowledgements, coupled with a timeout period (see timers below) calls for a retransmission.

• Sequence numbers: In packet switched networks, it is possible for packets to be delivered out of order. It is TCP’s job to properly sequence segments it receives so it can deliver the byte stream data to an application in order.

• Timers: TCP maintains various static and dynamic timers on data sent. The sending TCP waits for the receiver to reply with an acknowledgement within a bounded length of time. If the timer expires before receiving an acknowledgement, the sender can retransmit the segment.

1.5 Connection Establishment and Termination

TCP provides a connection-oriented service over packet switched networks. Connection-oriented implies that there is a virtual connection between two endpoints. There are three phases in any virtual connection. These are the connection establishment, data transfer and connection termination phases.

1.6 3-way handshake

In order for two hosts to communicate using TCP they must first establish a connection by exchanging messages in what is known as the three-way handshake. The diagram below depicts the process of the three-way handshake. From Figure 1, it can be seen that there are three TCP segments exchanged between two hosts, Host A and Host B. Reading down the diagram depicts events in time.

To start, Host A initiates the connection by sending a TCP segment with the SYN control bit set and an initial sequence number (ISN) we represent as the variable x in the sequence number field.

At some moment later in time, Host B receives this SYN segment, processes it and responds with a TCP segment of its own. The response from Host B contains the SYN control bit set and its own ISN represented as variable y. Host B also sets the ACK control bit to indicate the next expected byte from Host A should contain data starting with sequence number x + 1.

When Host A receives Host B’s ISN and ACK, it finishes the connection establishment phase by sending a final acknowledgement segment to Host B. In this case, Host A sets the ACK control bit and indicates the next expected byte from Host B by placing acknowledgement number y + 1 in the acknowledgement field.

In addition to the information shown in the diagram above, an exchange of source and destination ports to use for this connection are also included in each senders’ segments.

Host A Host B

1.10.1 Slow Start

Slow Start, a requirement for TCP software implementations is a mechanism used by the sender to control the transmission rate, otherwise known as sender-based flow control. This is accomplished through the return rate of acknowledgements from the receiver. In other words, the rate of acknowledgements from the receiver determines the rate at which the sender can transmit data.

Initially, the Slow Start algorithm sets the congestion window to one segment, which is the maximum segment size (MSS) initialized by the receiver during the connection establishment phase. When acknowledgements are returned by the receiver, the congestion window increases by one segment for each acknowledgement returned. Thus, the sender can transmit the minimum of the congestion window and the advertised window of the receiver, which is simply called the transmission window.

Slow Start is actually not very slow when the network is not congested and network response time is good. For example, the first successful transmission and acknowledgement of a TCP segment increases the window to two segments. After successful transmission of these two segments and acknowledgements completes, the window is increased to four segments. Then eight segments, then sixteen segments and so on, doubling from there on out up to the maximum window size advertised by the receiver or until congestion finally does occur.

1.10.2 Congestion Avoidance

During the initial data transfer phase of a TCP connection the Slow Start algorithm is used. However, there may be a point during Slow Start that the network is forced to drop one or more packets due to overload or congestion. If this happens, Congestion Avoidance is used to slow the transmission rate. However, Slow Start is used in conjunction with Congestion Avoidance as the means to get the data transfer going again so it doesn’t slow down and stay slow.

In the Congestion Avoidance algorithm retransmission timer expiring or the reception of duplicate ACKs can implicitly signal the sender that a network congestion situation is occurring. The sender immediately sets its transmission window to one half of the current window size (the minimum of the congestion window and the receiver’s advertised window size), but to at least two segments. If congestion was indicated by a timeout, the congestion window is reset to one segment, which automatically puts the sender into Slow Start mode. If congestion was indicated by duplicate ACKs, the Fast Retransmit and Fast Recovery algorithms are invoked (see below).

As data is received during Congestion Avoidance, the congestion window is increased. However, Slow Start is only used up to the halfway point where congestion originally occurred. This halfway point was recorded earlier as the new transmission window. After this halfway point, the congestion window is increased by one segment for all segments in the transmission window that are acknowledged. This mechanism will force the sender to more slowly grow its transmission rate, as it will approach the point where congestion had previously been detected.

1.11 Fast Retransmit

When a duplicate ACK is received, the sender does not know if it is because a TCP segment was lost or simply that a segment was delayed and received out of order at the receiver. If the receiver can reorder segments, it should not be long before the receiver sends the latest expected acknowledgement. Typically no more than one or two duplicate ACKs should be received when simple out of order conditions exist. If however more than two duplicate ACKs are received by the sender, it is a strong indication that at least one segment has been lost. The TCP sender will assume enough time has lapsed for all segments to be properly re-ordered by the fact that the receiver had enough time to send three duplicate ACKs.

When three or more duplicate ACKs are received, the sender does not even wait for a retransmission timer to expire before retransmitting the segment (as indicated by the position of the duplicate ACK in the byte stream). This process is called the Fast Retransmit algorithm.

2 Problem Statement

For this assignment, you have to implement a Transmission Control Protocol which should incorporate only the following features on top of the unreliable UDP sockets:

• Reliability (with proper re-transmissions in the event of packet losses / corruption)

• Data Integrity (with checksums)

• Connection Management (SYN and FIN)

• Optimizations (fast retransmit on 3 or more duplicate ACKs)

Then you are required to transfer a file from the client to the server using the new reliable transport protocol you just developed.

2.1 Protocol Specification

The various components of the protocol are explained step by step. Please strictly adhere to the specifications.

2.1.1 Message Format

All data are sent using UDP. You have to communicate all the transport layer information in the data portion of a UDP packet. To support reliability, hosts will implement a variant of the Go-Back-N protocol. The sender will tag each outgoing message with an increasing sequence number. The receiver will use the sequence numbers to ensure that all messages have been received in the correct order. If a message is received out of order it will be stored in the receiver’s buffer (but not delivered to an application) and an acknowledgment corresponding to the last successful contiguous byte received will be retransmitted. The sender will maintain an acknowledgment timeout based on the round trip time of the link between the end hosts. If a duplicate acknowledgment is received or if the acknowledgment timeout expires the sender will resend the unacknowledged segment destined for the receiver. A one’s complement checksum (The detailed explanation is in later section) on the data part is used to enforce message integrity. See figure 4 for details.

		Figure 4: Binary packet format
0		16		293031
		Byte Sequence Number
		Acknowledgment
		Timestamp
		Length			S	F	A
		All Zeros		Checksum
		Data

• Byte Sequence Number is incremented according to the bytes sent. It indicates the position of the first byte of the data in this segment.

• Acknowledgment indicates the next byte expected in the reverse direction

• Timestamp is derived from the System.nanoTime() function and is the time of data transmission (in nanoSeconds) which is 64 bits (or 8 bytes) long in size.

Length is the length of the data portion (in bytes) and please pay attention that the least three significant bits are used by flags. That means valid number of bits for the length field are only 29 bits (You will need to do bit manipulation to access and set these fields).

• Three flags: S for SYN, A for ACK, and F for FIN.

• Checksum is the one’s complement checksum computed over the entire packet with the checksum assumed zero in the input. See appendix A on how to calculate one’s complement checksum.

2.1.2 Maximum Number of Retransmissions

If unacknowledged messages remain in a host’s send buffer and no response from the destination has been received after multiple retransmission attempts, the sending host will stop trying to send the messages and report an error. This maximum is set to 16 by default.

2.1.3 Maximum Transmission Unit

Maximum Transmission Unit (MTU) is the maximum size of IP packet you may transmit. This value is required to derive the maximum size of payload in your protocol. So in order to transfer a huge file, it needs to be divided into smaller chunks the maximum size of each chunk being (MTU – size of IP header (usually 20B) – size of UDP header (8B) – your header (24B)) It is passed as a command line argument to the client and server during startup. Normal ethernet links use 1500B MTU – actual MTU used can be inspected with ‘ip link‘ command.

On the safe size, don’t specify anything larger than 1500 in MTU unless your test environment can support Ethernet Jumbo Frame. If you do not understand what this means, your network is unlikely to support an Ethernet frame larger than 1518 bytes.

2.1.4 Connection State and Peer Actions

Suppose host A wants to send a message to host B. Assume, for now, that A has never sent a message to B (as will be clear, it does not matter if A has rebooted or has sent a message a long time ago, etc.).

• Data Send Actions: A will send a data segment as governed by the size of the sliding congestion window. In this assignment, the congestion window is a configured parameter. A will include a monotonically increasing timestamp (See Section 2.2 on how to compute this timestamp) on the packet. If A thinks this is a new connection (because it has never communicated with B before, or because it has somehow lost the connection state to B), it will set the sequence number to 0. For each subsequent message, A will include a new timestamp on the packet and increment the sequence number by the number of bytes sent.

• Receiver Actions

Connection start and data transfer: If this is the first time B is communicating with A and has just received a segment with sequence number equal to zero, it creates a new connection state for A. For each segment received, B will send an acknowledgment to A. The acknowledgment packet has sequence number corresponding to the next expected byte. If no data is sent along with the acknowledgment, the length of this segment will be zero. Note data packets cannot have zero length. B also copies the timestamp field from the data segment into the corresponding ACK segment. Thus, A can use this timestamp field in the acknowledgment to calculate the round trip time for segments (refer to our discussion in class on calculating round trip times). Retransmissions: For each packet sent, the sender must maintain a retransmission timer (the computation of the timer is described in the next section). Whenever a packet has not been ACKed before its retransmission timer goes off, it must be re-sent. There is a timer set for the re-transmissions as well — thus, if the re-transmission is not ACKed, the packet will be re-re-transmitted. Apart from timeout based retransmissions, the sender also uses three duplicate acknowledgments for the same sequence number as an indicator of loss and retransmits the corresponding lost segment. This is the “fast retransmit” approach.

2.2 Timeout Computation

The sender places the current time in the packet timestamp field of the message header. When a packet is acknowledged by the receiving client it will copy the packet timestamp into the acknowledged timestamp field. When the sender receives the acknowledgment it can subtract the acknowledgment timestamp from the current time to calculate the round trip time. If a client is sending a cumulative acknowledgment of several packets, the timestamp from the latest received packet which is causing this acknowledgment should be copied into the reply. We will use a simple exponentially weighted average to compute the timeout.

Before sending the first packet, the timeout value is, arbitrarily, set to 5 seconds. Assume the sender has just received an ack with sequence S and timestamp T. Let C be the current time at the sender and TO be the timeout time.

if (S = 0)

ERTT := (C – T)

EDEV := 0

TO := 2*ERTT

else

SRTT := (C – T)

SDEV := |SRTT – ERTT|

ERTT := a*ERTT + (1-a)*SRTT

EDEV := b*EDEV + (1-b)*SDEV

TO := ERTT + 4*EDEV

The value of a is set to 0.875 and b is 0.75.

Computing the sender timestamp: In the 64-bit timestamp field, the sender includes a monotonically in-creasing timestamp with granularity of one nanosecond. This can be obtained from the System.nanoTime() function.

2.3 Host Commands/Output Format

You will implement a java executable called TCPend. Command line arguments will indicate which host is the initiator of a TCP transfer. The sender TCPend must support the following options at startup:

java TCPend -p <port> -s <remote IP> -a <remote port> –f <file name> -m <mtu> -c <sws>

• port: port number at which the client will run

• remote IP: the IP address of the remote peer (i.e. receiver). With the -s flag your program should operate in sending mode.

• remote port: the port at which the remote receiver is running

• file name: the file to be sent

• mtu: maximum transmission unit in bytes

• sws: sliding window size in number of segments

The remote receiver uses the following set of arguments:

java TCPend -p <port> -m <mtu> -c <sws> -f <file name>

• port: port number at which the receiver will listen at

• file name: the path where the incoming file should be written

• mtu: maximum transmission unit in bytes

• sws: sliding window size in number of segments

Host Output

Each host should output the information about each segment that it sends and receives in the following format:

<snd/rcv> <time> <flag-list> <seq-number> <number of bytes> <ack number>

where flag list includes S for SYN, A for ACK, F for FIN and D for data. The following are valid output lines for a connection initiator that sends 112 bytes of data:

snd 34.335 S – – – 0 0 0

rcv 34.8 S A – – 0 0 1

snd 34.81 – A – – 1 0 1

snd 35.5 – A – D 1 56 1

snd 35.6 – A – D 57 56 1

rcv 36.2 – A – – 1 0 113

snd 36.65 – A F – 113 0 1

rcv 37.2 – A F – 1 0 114

snd 37.3 – A – – 114 0 2

Note that every packet past the initial SYN should always carry a ACK flag and the correct acknowledgement number.

At the end of the transfer, i.e., once the connection has been closed you should print the following statistics:

• Amount of Data transferred/received

• Number of packets sent/received

• Number of out-of-sequence packets discarded

• Number of packets discarded due to incorrect checksum

• Number of retransmissions

• Number of duplicate acknowledgements

3 Testing

To test your implementation you could use a mininet virtual topology like you have been doing in the earlier assignments but modify the router implementation to drop packets with a given probability. E.g., drop 5% of the IP packets received. Then you could try sending the file from one mininet host to another (use xterm to open terminal windows on different mininet hosts) connected via the modified router and see if the file gets transferred correctly. Here (requires UW login) is a sample router implementation of your assignment2 which drops roughly 5% of the packets. Look at the end of forwardIPPacket() method inside Router.java. You can modify the drop rates for your testing. You can use the similar commands from assignment 2 to run it.

To run it first start mininet using the following command:

sudo ./run_mininet.py topos/single_rt.topo -a

Then start pox:

./run_pox.sh

Then start your virtual router that you’ve modified that will drop 5

java -jar VirtualNetwork.jar -v r1 -r rtable.r1 -a arp_cache

Now open terminal windows on hosts h1 and h2 by typing xterm h1 h2 in mininet console. Navigate to the directory where with your TCP implementation and send a file from h1 to h2. If your implementation is correct, the file should be successfully transferred to h2 i.e., content should exactly match. In addition to visual inspection of the received file, you may also use checksum utilities such as sha256sum to verify that the files are indeed identical.

Note that this assignment is more open ended than your previous assignments in this course. You have more freedom for your implementation methodology as long as you correctly meet the protocol defined above. To test, the TAs would try to transfer files of different sizes (small, medium, large) with different packet drop rates and would check if data transfer was correct. I.e., file received by the receiver should be identical to the one present at the sender.

Submission Instructions

Create a gzipped tar file with tgz extension, containing the following:

• The source code for your TCP implementation. Include all the java files in a src directory.

• A makefile that compiles the java source code in place, i.e. your class files should be in src as well.

• A README file with the names and CS usernames of both group members.

tar -czvf username1_username2.tgz src Makefile README

As usual, upload the tar file to the Assignment 4 dropbox on canvas. Please submit only one tar file per group.

• Appendix A: One’s complement checksum

• 1. Divide the data you want to calculate a checksum over into 16bit segments. Think of it as unsigned 16-bit integers (or uint16_t in C terminology). If the data has an extra byte at the end, pad it with

0x00.

• 2. Add all such segments up. Whenever you produce a 17th bit carry-over, add it back to the least significant bit (i.e. add 1). Do this for all the segments.

  3. Do a bitwise not (i.e. flipping all 16 bits). This is the resulting checksum.

Example

Say we want to checksum the following data:

1000 0110 0101 1110

1010 1100 0110 0000

0111 0001 0010 1010

1000 0001 1011 0101

The algorithm works like:

1000 0110 0101 1110 first uint16

• 1010 1100 0110 0000 second uint16

========================

1 0011 0010 1011 1110

• • 17-th bit carry over

========================

0011 0010 1011 1111

• 0111 0001 0010 1010 third uint16

========================

0 1010 0011 1110 1001

• 1000 0001 1011 0101 fourth uint16

========================

1 0010 0101 1001 1110

• • 17-th bit carry over

========================

0010 0101 1001 1111 “one’s complement sum”

~~~~~~~~~~~~~~~~~~~~~~~~

1101 1010 0110 0000 “one’s complement checksum”

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