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

Short Answer

Expert verified
Timers are needed to detect packet loss and trigger retransmission.

Step by step solution

01

Understanding Reliable Data Transfer (rdt) Protocols

Reliable data transfer protocols are designed to ensure that data is transmitted accurately from sender to receiver over a network. In real-world networks, several issues can corrupt or lose data packets, so protocols are needed to handle these concerns.
02

Analyzing the Problem of Packet Loss

One of the main challenges in data transmission is packet loss. This can occur for various reasons, such as network congestion or signal interference. If packets are lost, the receiver won't receive the correct data, making it necessary for the protocol to account for this issue.
03

Introduction of Timers

Timers are a solution to the problem of packet loss. By introducing timers, we allow the sender to determine if an acknowledgement (ACK) for a sent packet has not been received within a predetermined time frame. If the timer expires without receiving an ACK, it is assumed that the packet was lost, prompting the sender to retransmit the packet.
04

Retransmission Strategy

Once a timer expires, the rdt protocol initiates a retransmission of the lost packet. This ensures that even if packets are lost in transmission, they will be resent until successfully acknowledged by the receiver, maintaining the reliability of data transfer.

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Key Concepts

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

Packet Loss
Packet loss is one of the key challenges in computer networking that affects reliable data transfer. Imagine sending a letter, but somewhere along the way, it gets lost. In networking, a packet, which is a small unit of data, may vanish in transit due to various reasons. Common causes include:
  • Network Congestion: Too much data traveling on the same path can lead to overflow, causing packets to drop.
  • Signal Interference: Environmental factors, like walls or electronic devices, can disrupt wireless signals, leading to packet loss.
  • Hardware Failures: Faulty routers, switches, or cables can result in packets getting lost.
Packet loss can degrade the performance and reliability of data transmission. For critical applications, like video conferencing or financial transactions, it's essential to minimize packet loss to ensure seamless communication.
Timers in Networking
Timers are a pivotal tool in networking protocols, especially when ensuring reliable data transfer. They function much like an alarm clock, which tells you when it's time to take action. In the context of networking, timers help determine if a packet has reached its destination correctly.

A timer starts when the sender transmits a packet. If an "acknowledgement packet" or ACK isn't received before the timer runs out, this signals a problem. The absence of an ACK typically indicates packet loss. To combat this, the sender will:
  • Retransmit the packet, assuming it didn't reach the receiver.
  • Adjust the timer depending on network conditions, ensuring efficiency and reliability.
Timers help networks avoid unnecessary waits, ensuring data is sent swiftly and accurately even in the face of unforeseen issues.
Acknowledgement Packets
Acknowledgement packets, known as ACKs, are crucial in ensuring data is sent accurately over a network. Think of ACKs as a "receipt" that tells the sender, "message received." When the receiver gets a packet, it sends back an ACK to confirm successful delivery. This process ensures:
  • Reliability: With ACKs, senders know which packets were successfully delivered, and which need retransmission due to loss.
  • Flow Control: ACKs help manage the rate of data transmission, adjusting to prevent overload or congestion in the network.
  • Error Correction: By detecting errors early, ACKs enable corrective measures, like packet resending, maintaining data integrity.
This method makes networks more reliable and efficient, as errors can be swiftly corrected, ensuring seamless data delivery.

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Most popular questions from this chapter

In this problem, we consider the delay introduced by the TCP slow-start phase. Consider a client and a Web server directly connected by one link of rate \(R\). Suppose the client wants to retrieve an object whose size is exactly equal to \(15 S\), where \(S\) is the maximum segment size (MSS). Denote the round-trip time between client and server as RTT (assumed to be constant). Ignoring protocol headers, determine the time to retrieve the object (including TCP connection establishment) when a. \(4 S / R>S / R+R T T>2 S / R\) b. \(S / R+R T T>4 S / R\) c. \(S / R>R T T\).

Consider that only a single TCP (Reno) connection uses one \(10 \mathrm{Mbps}\) link which does not buffer any data. Suppose that this link is the only congested link between the sending and receiving hosts. Assume that the TCP sender has a huge file to send to the receiver, and the receiver's receive buffer is much larger than the congestion window. We also make the following assumptions: each TCP segment size is 1,500 bytes; the two-way propagation delay of this connection is \(150 \mathrm{msec}\); and this TCP connection is always in congestion avoidance phase, that is, ignore slow start. a. What is the maximum window size (in segments) that this TCP connection can achieve? b. What is the average window size (in segments) and average throughput (in bps) of this TCP connection? c. How long would it take for this TCP connection to reach its maximum window again after recovering from a packet loss?

Consider a modification to TCP's congestion control algorithm. Instead of additive increase, we can use multiplicative increase. A TCP sender increases its window size by a small positive constant \(a(0

Consider two network entities, \(\mathrm{A}\) and \(\mathrm{B}\), which are connected by a perfect bidirectional channel (i.e., any message sent will be received correctly; the channel will not corrupt, lose, or re-order packets). A and B are to deliver data messages to each other in an alternating manner: First, A must deliver a message to \(\mathrm{B}\), then \(\mathrm{B}\) must deliver a message to \(\mathrm{A}\), then \(\mathrm{A}\) must deliver a message to \(\mathrm{B}\) and so on. If an entity is in a state where it should not attempt to deliver a message to the other side, and there is an event like rdt_send (data) call from above that attempts to pass data down for transmission to the other side, this call from above can simply be ignored with a call to rdt_unable_to_send (data), which informs the higher layer that it is currently not able to send data. [Note: This simplifying assumption is made so you don't have to worry about buffering data.] Draw a FSM specification for this protocol (one FSM for A, and one FSM for B!). Note that you do not have to worry about a reliability mechanism here; the main point of this question is to create a FSM specification that reflects the synchronized behavior of the two entities. You should use the following events and actions that have the same meaning as protocol rdt \(1.0\) in Figure 3.9: rdt_send(data), packet = make_pkt(data), udt_send (packet), rdt_rcv (packet), extract (packet, data), deliver_data (data). Make sure your protocol reflects the strict alternation of sending between \(\mathrm{A}\) and \(\mathrm{B}\). Also, make sure to indicate the initial states for A and B in your FSM descriptions.

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

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