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In our rdt protocols, why did we need to introduce sequence numbers?

Short Answer

Expert verified
Sequence numbers help differentiate between new and duplicate packets in rdt protocols.

Step by step solution

01

Introduce Reliable Data Transfer (rdt) Protocols

Reliable Data Transfer protocols are essential in computer networks to ensure data is transmitted accurately and reliably from sender to receiver without being lost, duplicated, or corrupted during the process.
02

Understand the Problem of Duplicate Data

In unreliable communication channels, a receiver may receive the same packet multiple times. This can happen due to packet loss, which causes retransmissions that might arrive at different times. Without a mechanism to differentiate between new and old packets, the receiver cannot identify and discard duplicates.
03

Explain Sequence Numbers

Sequence numbers are used to uniquely identify each packet transmitted over the network. Each packet is assigned a sequence number that increases incrementally, ensuring that both the sender and receiver can keep track of the packets.
04

Sequence Numbers in Preventing Duplicates

When a receiver gets a packet, it checks the sequence number. If the sequence number matches a number of a previously received packet, the receiver knows the packet is a duplicate and can discard it. This ensures only new, unduplicated data is processed.

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Key Concepts

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

Sequence Numbers
Sequence numbers play a crucial role in ensuring reliable data transfer across networks. They act as unique identifiers for each packet sent over the network, enabling both the sender and receiver to keep track of data efficiently. When a data packet is transmitted, it includes a sequence number that is sequentially assigned, forming a numerical order.
Because of these sequence numbers, the receiver can determine the correct order of packets and detect any missing ones. If a packet appears with a previously used sequence number, the receiver identifies it as a duplicate. This mechanism allows the receiver to discard redundant packets, ensuring that all data is captured accurately and only once.
The use of sequence numbers not only organizes the data flow but also aids in maintaining the integrity of the communication process. By ensuring that packets are not misinterpreted or misplaced, sequence numbers greatly enhance the reliability of data transmission.
Duplicate Data Prevention
To avoid processing duplicate data, systems use a combination of techniques alongside sequence numbers. When a receiver obtains a packet with a sequence number it has already processed, it recognizes that packet as a duplicate. This recognition allows receivers to ensure each piece of data is processed only once.
Here are a few additional methods that enhance duplicate data prevention:
  • **Acknowledgment Protocols**: The sender receives confirmation when the receiver gets a packet. Without an acknowledgment, the sender might resend, risking duplicates. With proper acknowledgments, this is minimized.
  • **Buffer Checks**: Receivers use buffers to temporarily hold packet information. This helps them compare incoming packets to what has already been received, preventing duplication.
  • **Timers**: These track packet transmissions. If a timer expires before an acknowledgment, the sender decides on retransmission, relying on sequence numbers to identify duplicates at the receiver's end.
Acknowledgment protocols, buffer checks, and timers function together with sequence numbers to prevent duplicate data from clogging network systems, allowing for efficient data management and consistency.
Packet Loss in Networks
Packet loss is an everyday challenge in network communication. It occurs when data packets fail to reach their intended destination, often due to network congestion, signal interference, or errors in transmission protocols. Addressing packet loss is critical to maintain effective data transfer.
Reliable Data Transfer (rdt) protocols incorporate several strategies to handle packet loss, ensuring data integrity and continuity:
  • **Retransmissions**: When a packet doesn't arrive, the sender will resend it, triggered by either not receiving an acknowledgment or receiving a negative acknowledgment.
  • **Error Checking**: Techniques such as checksums are used to ensure data integrity. If errors are detected, packets can be discarded, and retransmissions can be requested.
  • **Flow Control**: Controls the rate of data transmission based on network capacity, minimizing loss due to congestion.
By using these techniques, networks can manage packet loss effectively. When combined with sequence numbers and duplicate data prevention strategies, these methods help maintain a robust and reliable communication environment.

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Most popular questions from this chapter

Suppose two TCP connections are present over some bottleneck link of rate \(R\) bps. Both connections have a huge file to send (in the same direction over the bottleneck link). The transmissions of the files start at the same time. What transmission rate would TCP like to give to each of the connections?

In Section 3.5.4, we saw that TCP waits until it has received three duplicate ACKs before performing a fast retransmit. Why do you think the TCP designers chose not to perform a fast retransmit after the first duplicate ACK for a segment is received?

Suppose Host A sends two TCP segments back to back to Host B over a TCP connection. The first segment has sequence number 90 ; the second has sequence number 110 . a. How much data is in the first segment? b. Suppose that the first segment is lost but the second segment arrives at B. In the acknowledgment that Host B sends to Host A, what will be the acknowledgment number?

Consider a simplified TCP's AIMD algorithm where the congestion window size is measured in number of segments, not in bytes. In additive increase, the congestion window size increases by one segment in each RTT. In multiplicative decrease, the congestion window size decreases by half (if the result is not an integer, round down to the nearest integer). Suppose that two TCP connections, \(C_{1}\) and \(C_{2}\), share a single congested link of speed 30 segments per second. Assume that both \(\mathrm{C}_{1}\) and \(\mathrm{C}_{2}\) are in the congestion avoidance phase. Connection \(\mathrm{C}_{1}\) 's RTT is \(50 \mathrm{msec}\) and connection \(\mathrm{C}_{2}\) 's RTT is \(100 \mathrm{msec}\). Assume that when the data rate in the link exceeds the link's speed, all TCP connections experience data segment loss. a. If both \(\mathrm{C}_{1}\) and \(\mathrm{C}_{2}\) at time \(\mathrm{t}_{0}\) have a congestion window of 10 segments, what are their congestion window sizes after 1000 msec? b. In the long run, will these two connections get the same share of the bandwidth of the congested link? Explain.

Consider the GBN protocol with a sender window size of 4 and a sequence number range of 1,024 . Suppose that at time \(t\), the next in-order packet that the receiver is expecting has a sequence number of \(k\). Assume that the medium does not reorder messages. Answer the following questions: a. What are the possible sets of sequence numbers inside the sender's window at time \(t\) ? Justify your answer. b. What are all possible values of the \(\mathrm{ACK}\) field in all possible messages cur rently propagating back to the sender at time \(t ?\) Justify your answer.

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