CS 176B -- Network Computing
Homework Assignment #2
Due by 11:59pm on Wednesday, February 6, 2002


Assignment Background

While TCP is a very good protocol, it is not useful for all applications. For example, TCP is not good at all for real-time streaming audio (or video). Consider the scenario where a user has a 56 kbps connection and they want to receive an audio file. Either that file can be delivered in its entierty before the user starts listening or it can be streamed at whatever rate is necessary to sustain playback. The first option has the drawback that it has a long initial delay (you cannot start playing the file until all or most of it has been downloaded). The second option has the drawback that if the path between the server and the client cannot sustain the streaming bandwidth, the client might lose a significant portion of the streamed data.

As greater bandwidth becomes more available in the Internet, the use of the streaming option is becoming more viable. When audio (or video) is streamed, UDP is used instead of TCP. The reason is that TCP does re-transmission of data. It is not the re-transmission that is bad, but the fact that when TCP discovers a loss, it stops and handles the loss before continuing with the initial transmission. This does not work at all for streaming. Instead of stopping the loss should just be ignored. A little bit of static is better than stopping the stream completely. Therefore, UDP is often used.

The problem with UDP though is that it does not have congestion control. While we have not yet studied the need for congestion control or techniques to handle it, understanding the problem is straightforward enough. The problem is that if a server sends data too quickly it will cause congestion in the network. A result of congestion is lost data. This creates a delicate situation for streaming UDP: send too slow and not enough data gets to the client. Send too quickly and a lot of data might be lost.

The goal of this assignment is to understand the challenges of using UDP to stream data from a server to a client. A really hard assignment would be to implement both full TCP-style congestion control and 100% reliability on top of UDP. Since we only have two weeks, this assignment will just give you a taste of some of these functions.


Operational Overview

In this assignment you will write a client and server. The function of the client and server is essentially to have the server stream a set of packets to the client and then have the client figure out if any of the packets were lost. Initially the client will send a request to the server requesting a specified bit rate. The server will send at this bit rate and the client will figure out if it is too slow or too fast based on the amount of loss. The client will then send a new bit rate to the server and will repeat the process for a specified number of rounds.

Now, breaking this functionality into a deeper level of detail, the operation starts with the server running and waiting for a request from the client. When the client starts, it will first send a request to the server for a stream of packets at a given rate. In this request, the client will specify a number of parameters including the number of packets to send, the transmission time between packets, and the packet size. The values of these parameters can be used to calculate the bandwidth of the stream sent by the server. Upon receiving this request, the server will send a stream of packets (each packet will have an application layer sequence number) to the client. The client will obvserve the number of packets lost and compute the actual bandwidth used.

The above set of actions constitutes a ``round''. The client will run a number of rounds. At the start of each round the client will send a new request to the server and will either increase or decrease the rate at which the server sends packets. The change in rate will depend on whether any packets were lost from the server.


Operational Details

The things that are important for this assignment include the following:

1. Server command line parameters

This is relatively straightforward. It is just the port number that the server is going to listen on. See the Operational Examples section for sample input.

2. Client command line parameters

All of the parameters that the client needs are given as command line parameters. There is no input needed once the program starts running. This makes testing easier. The set of parameters include:

See the Operational Examples section for sample client input.

3. Format of the message from the client to the server

The message should be sent as a 4 integers and the four values represented by these fields should be: <num_rounds> <num_packets> <pkt_size> <xmit_time>

4. Format of the data packets sent from the server to the client

The data sent between the server and client is pretty much irrelevant. Instead of adding additional complexity by having the server open a file and read data, you can simply create random data to send to the client. The client will not care what is sent. However, while the data itself is not important, what is important is whether the packet carrying the data is received or not. In order for the client to make this distinction, the server will send data packets with sequence numbers. The sequence number will be an integer prepended to each packet. Therefore, the size of the packet actually sent will be the pkt_size plus 4 bytes. The first sequence number will always be zero (0) and will increment up to INT_MAX and will then wrap to zero again. Your program should be able to handle this sequence number wrap.

5. Operation and output of the client

Client operation is relatively straightforward. After starting and receiving the command line parameters, it should send a request to the server. Then it should start waiting to receive data packets.

From here, the operation turns into a bunch of conditionals. First, if the client receives num_packets then everything works very simply. Second, if any but the last packet are lost, the client can use the sequence number and immediately tell which and how many packets were lost. Now it becomes a little trickier...

If the last packet is lost the client will not know whether to continue waiting or assume the packet is lost. The problem is actually worse than this. What if the last two packets are lost, or even the last three! Therefore, you will need to use a timer after each packet is received. If no additional packets are received within two seconds, you should assume all the remaining packets have been lost and continue to the end-of-round processing step.

Be aware that after a request is sent, it is entirely possible that NO packets are received. After the request is sent, you should wait 10 seconds to receive the first packet. If no packet is received within the first 10 seconds, you should assume 100% loss. (While this scenario is pretty unlike (at least the first packet should make it), your program should be able to handle this possibility.)

Once a round ends, your program should print the following information:

End of Round AA
Packets Expected: BB
Requested Bandwidth: CC bps
Packets Received: DD
Loss Rate:  EE.E%
Actual Bandwidth: FF bps
Adjusting Bandwidth from XX to YY (new inter-transmission time is ZZ ms)

This should continue for as many rounds as specified in client's command line.

6. Ending the test and closing the programs

Once the client has finished the number of tests specified in the command line, it should send a final request to the server with all values as "-1". Therefore, the message sent should be "-1 -1 -1 -1". After sending this message the client should exit.

For the server, once it receives the "-1 -1 -1 -1" message, it too should exit. Of course it is possible that the server never receives this message. Therefore, the server should use the following procedure to avoid hanging and waiting forever. When the first request is received, the server should satisfy the request. Once the last packet is sent to the client, the server should start a timer. If a new request or a final request is not received within 10 seconds, the server should simply exit.

Operational Examples

The server is straightforward. It essentially returns no feedback. It runs and then it exits.
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Here is an example from the client's perspective:
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Beyond This Assignment

There are a number of things to keep in mind when doing this assignment:


Assignment Turnin

You should turn in the source code for your client and server. Remember that you can do this assignment in either Java or C. In either case, you should turn in only two programs: the client and server. The names of these two files should either be server.c and client.c or server.java and client.java. There is no hard copy turnin for this assignment. Be sure to include your name in each program that you turnin.

To turn in assignments, use the following command from the Computer Science CSIL lab:

NOTE: The final "hw2" is a local directory containing the 2 and only 2 programs you are turning in. Be certain to name this directory exactly "hw2".

ANOTHER NOTE: It is highly recommended that you use the CSIL machines to do this assignment. All of the tools you will need are available there; it significantly improves our ability to help you if you have problems; and it ensures that if your programs work there, they will work when we grade them.


Assignment Grading

Your client and server will be tested against a separately written client and server. The goal of this testing is to check your two programs for operational correctness against the specfication as described above. The assignment will be graded out of a maximum of 100 points.

In addition to correctness, part of your grade will depend on how well your code is written and documented. NOTE: good code/documentation does not imply that more is better. The goal is to be efficient, elegant and succinct!