In this project, you will build a simple reliable transport protocol, RTP, on top of UDP. Your RTP implementation must provide in order, reliable delivery of UDP datagrams in the presence of events like packet loss, delay, corruption, duplication, and reordering.
This assignment can be done individually or in groups of 2 students.
There are a variety of ways to ensure a message is reliably delivered from a sender to a receiver. You are to implement a sender (sender
) and a receiver (receiver
) that follows the following RTP specification.
RTP sends data in the format of a header, followed by a chunk of data.
RTP has four header types: START
, END
, DATA
, and ACK
, all following the same format:
PacketHeader:
int type; // 0: START; 1: END; 2: DATA; 3: ACK
int seq_num; // Described below
int length; // Length of data; 0 for ACK, START and END packets
int checksum; // 32-bit CRC
To initiate a connection, sender
starts with a START
message along with a random seq_num
value, and wait for an ACK for this START
message. After sending the START
message, additional packets in the same connection are sent using the DATA
message type, adjusting seq_num
appropriately. After everything has been transferred, the connection should be terminated with sender
sending an END
message, and waiting for the corresponding ACK for this message.
The ACK seq_num
values for START
and END
messages should both be set to whatever the seq_num
values are that were sent by sender
.
sender
will use 0 as the initial sequence number for data packets in that connection.
An important limitation is the maximum size of your packets. The UDP protocol has an 8 byte header, and the IP protocol underneath it has a header of 20 bytes. Because we will be using Ethernet networks, which have a maximum frame size of 1500 bytes, this leaves 1472 bytes for your entire packet
structure (including both the header and the chunk of data).
Overall, this assignment has the following components:
- Part 1: Implement
sender
- Part 2: Implement
receiver
- Part 3: Optimizations
- Important notes
- Submission Instructions
Similar to assignement 1, we provide scaffolding code in sender_reciver
. We use the same VM as assignment 1.
- Use
vagrant up
to boot the VM. - Use
vagrant ssh
to log into the VM. - Use
vagrant suspend
to save the state of the VM and stop it. - Use
vagrant halt
to gracefully shutdown the VM operating system and power down the VM. - Use
sudo pip install scapy
in the VM to installscapy
package required by this assignment.
After completing this programming assignment, students should be able to:
- Explain the mechanisms required to reliably transfer data
- Describe how different sliding window protocols work
sender
should read an input message and transmit it to a specified receiver using UDP sockets following the RTP protocol. It should split the input message into appropriately sized chunks of data, and append a checksum
to each packet. seq_num
should increment by one for each additional packet in a connection. Please use the 32-bit CRC header we provide in sender_receiver/util.py
, in order to add a checksum to your packet.
You will implement reliable transport using a sliding window mechanism. The size of the window (window-size
) will be specified in the command line. sender
must accept cumulative ACK
packets from receiver
.
After transferring the entire message, you should send an END
message to mark the end of connection.
sender
must ensure reliable data transfer under the following network conditions:
- Loss of arbitrary levels;
- Reordering of ACK messages;
- Duplication of any amount for any packet;
- Delay in the arrivals of ACKs.
To handle cases where ACK
packets are lost, you should implement a 500 milliseconds retransmission timer to automatically retransmit packets that were never acknowledged.
Whenever the window moves forward (i.e., some ACK(s) are received and some new packets are sent out), you reset the timer. If after 500ms the window still has not advanced, you retransmit all packets in the window because they are all never acknowledged.
sender
should be invoked as follows:
python sender.py [Receiver IP] [Receiver Port] [Window Size] < [Message]
Receiver IP
: The IP address of the host thatreceiver
is running on.Receiver Port
: The port number on whichreceiver
is listening.Window Size
: Maximum number of outstanding packets.Message
: The message to be transferred. It can be a text as well as a binary message.
receiver
needs to handle only one sender
at a time and should ignore START
messages while in the middle of an existing connection. It must receive and store the message sent by the sender on disk completely and correctly.
receiver
should also calculate the checksum value for the data in each packet
it receives using the header mentioned in part 1. If the calculated checksum value does not match the checksum
provided in the header, it should drop the packet (i.e. not send an ACK back to the sender).
For each packet received, it sends a cumulative ACK
with the seq_num
it expects to receive next. If it expects a packet of sequence number N
, the following two scenarios may occur:
- If it receives a packet with
seq_num
not equal toN
, it will send back anACK
withseq_num=N
. Note that this is slightly different from the Go-Back-N (GBN) mechanism discussed in class. GBN totally discards out-of-order packets, while herereceiver
buffers out-of-order packets. The mechanism here is more efficient than GBN. - If it receives a packet with
seq_num=N
, it will check for the highest sequence number (sayM
) of the inorder packets it has already received and sendACK
withseq_num=M+1
.
If the next expected seq_num
is N
, receiver
will drop all packets with seq_num
greater than or equal to N + window_size
to maintain a window_size
window.
Put the programs written in parts 1 and 2 of this assignment into a folder called RTP-base
.
receiver
should be invoked as follows:
python receiver.py [Receiver Port] [Window Size] > Message
Receiver Port
: The port number on whichreceiver
is listening for data.Window Size
: Maximum number of outstanding packets.Message
: The received message received.
For this part of the assignment, you will be making a few modifications to the programs written in the previous two sections. Consider how the programs written in the previous sections would behave for the following case where there is a window of size 3:
In this case receiver
would send back two ACKs both with the sequence number set to 0 (as this is the next packet it is expecting). This will result in a timeout in sender
and a retransmission of packets 0, 1 and 2. However, since receiver
has already received and buffered packets 1 and 2. Thus, there is an unnecessary retransmission of these packets.
In order to account for situations like this, you will be modifying your receiver
and sender
accordingly (save these different versions of the program in a folder called RTP-opt
):
receiver
will not send cumulative ACKs anymore; instead, it will send back an ACK withseq_num
set to whatever it was in the data packet (i.e., if a sender sends a data packet withseq_num
set to 2,receiver
will also send back an ACK withseq_num
set to 2). It should still drop all packets withseq_num
greater than or equal toN + window_size
, whereN
is the next expectedseq_num
.sender
must maintain information about all the ACKs it has received in its current window and maintain an individual timer for each packet. So, for example, packet 0 having a timeout would not necessarily result in a retransmission of packets 1 and 2.
For a more concrete example, here is how your improved sender
and receiver
should behave for the case described at the beginning of this section:
receiver
individually ACKs both packet 1 and 2.
sender
receives these ACKs and denotes in its buffer that packets 1 and 2 have been received. Then, the it waits for the 500 ms timeout and only retransmits packet 0 again.
The command line parameters passed to these new sender
and receiver
are the same as the previous two sections.
- Please closely follow updates on Piazza. All further clarifications will be posted on Piazza via pinned Instructor Notes. We recommend you follow these notes to receive updates in time.
- You MUST NOT use TCP sockets.
- We are NOT providing any test cases. You can take a look at the test script in assignment 1, and write your own test script.
You must submit:
- The source code for
sender
andreceiver
from parts 1 and 2: all source files should be in a folder calledRTP-base
. - The source code for
sender
andreceiver
from part 3: all source files should be in a folder calledRTP-opt
. - Submit the assignment by uploading your files to Gradescope. Join the course with entry code 94BWPW.
For students that find this assignment not challenging enough, you can
- Complete it individually.
- Complete it with C.
- Allow the receiver to receive messages from multiple concurrent senders.
- Add congestion control to the sender and receiver, so that the sender and receiver can figure out the sizes of their sliding windows by themselves.
- Implement different congestion control algorithms, and compare their pros and cons.
- Design your own congestion control algorithms. You can try crazy ideas like using deep learning to learn good congestion control algorithms.
This programming assignment is based on UC Berkeley's Project 2 from EE 122: Introduction to Communication Networks.