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If all the links in the Internet were to provide reliable delivery service, would the TCP reliable delivery service be redundant? Why or why not?

Short Answer

Expert verified
No, TCP reliable service is not redundant because it ensures end-to-end data integrity and connection management, beyond what reliable links can provide.

Step by step solution

01

Understanding TCP's Role

TCP (Transmission Control Protocol) provides reliable delivery service by ensuring that data sent from sender to receiver arrives completely and in the correct order. It handles packet loss, errors, duplication, and ensures data integrity by incorporating error checking and acknowledgment features.
02

Defining 'Reliable Delivery' at the Link Layer

If all links provided reliable service, this means that each link between devices could guarantee perfect transmission without losing packets, duplicating, or delivering them out of order. In this context, individual links behave accurately, without errors, and handle congestion control effectively.
03

Evaluating the End-to-End Argument

The end-to-end argument in network design suggests that certain functions (like reliability) are best implemented at the endpoints of a communication system rather than by the individual links. This is because links only see a portion of the entire path, while the endpoints have the broader context necessary to ensure overall system reliability.
04

Differentiating Between Link and End-to-End Reliability

Even if individual links offer reliability, TCP still provides important end-to-end services that are essential for applications. TCP manages data flow, controls congestion across the entire communication path, ensures complete data transfer by re-transmitting lost packets, and maintains data integrity and ordering.
05

Conclusion: TCP's Continued Importance

While reliable link-layer services would improve certain network aspects, TCP's role remains essential for ensuring end-to-end transport reliability. It addresses complexities that arise over multi-hop paths, such as variable link performance, network congestion, and differences in error rates among links.

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Key Concepts

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

End-to-End Argument
The end-to-end argument is a principle in system design that proposes key functions should be implemented at the endpoints of a system rather than within the intermediate links.
This concept significantly impacts how network protocols are engineered, especially in the context of reliability. When we talk about delivering data reliably across a network, the end-to-end argument suggests that this reliability should be managed by the sending and receiving machines rather than expecting all individual network links to handle it.

The idea is simple but powerful. Each link can only see a fraction of the entire network path, and therefore may lack the necessary context to fully ensure overall reliability.
  • Efficiency: By having endpoints manage errors and recovery, we can avoid duplicating efforts at multiple points in the network.
  • Flexibility: Endpoints can implement various reliability methods tailored to specific applications.
In essence, even if links were reliable, endpoints need to ensure a holistic reliability strategy across potentially diverse and complex network paths.
Link Layer
The link layer is a crucial component in the networking model that serves to transfer data between adjacent network nodes within the same network segment.
This layer handles the physical transmission of data and can encompass technologies like Ethernet and Wi-Fi. A key responsibility of the link layer is to ensure that data frames are properly sent to and received from the directly connected nodes.

If all links were to provide perfect reliability, they would:
  • Deliver packets without losses.
  • Maintain the correct order of delivery.
  • Avoid duplications.
However, despite these capabilities, the link layer operates independently for each hop in the network. This means that while a single link might perform reliably, a network path made up of several links might still experience failures in aggregate. The link layer lacks awareness of network conditions beyond its immediate connection, highlighting why end-to-end reliability is critical.
This is where the Transmission Control Protocol (TCP) comes into play.
Congestion Control
Congestion control is an essential aspect of TCP that helps manage and prevent network congestion, ensuring efficient use of network resources.
In a network, congestion occurs when demand for bandwidth exceeds available capacity, leading to packet loss and delays.

TCP tackles congestion with several mechanisms:
  • Slow Start: Gradually increases data transmission until it finds the network's capacity limit.
  • Congestion Avoidance: Changes data flow rates to prevent congestion before it occurs.
  • Fast Retransmit: Quickly re-sends data packets presumed lost.
By managing the flow of data across the entire path from sender to receiver, TCP ensures that congestion is minimized and that bandwidth is fairly allocated among users.
TCP's congestion control is integral to maintaining end-to-end performance on the internet, even when individual links are reliable.
Data Integrity
Data integrity involves ensuring that information has not been altered or corrupted during transmission.
For network communications, maintaining data integrity is critical, as even minor errors can lead to significant misunderstandings or system inefficiencies.

TCP ensures data integrity by using checksums, which are mathematical formulations accompanying each packet.
  • When a packet is received, the destination uses the checksum to verify that the data has not been tampered with.
  • If the checksum does not match, TCP knows the data is suspect and may request a retransmission.
Thus, TCP provides a comprehensive method to maintain data integrity across various network conditions.
Even if a network's links were completely reliable, only end-to-end mechanisms like TCP can robustly handle the complexities that arise during data transfer between endpoints, ensuring that data remains complete and unaltered throughout the journey.

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Most popular questions from this chapter

Suppose the information portion of a packet ( \(D\) in Figure \(5.3\) ) contains 10 bytes consisting of the 8-bit unsigned binary ASCII representation of string "Networking." Compute the Internet checksum for this data.

Suppose nodes \(\mathrm{A}\) and \(\mathrm{B}\) are on the same \(10 \mathrm{Mbps}\) broadcast channel, and the propagation delay between the two nodes is 325 bit times. Suppose CSMA/CD and Ethernet packets are used for this broadcast channel. Suppose node A begins transmitting a frame and, before it finishes, node B begins transmitting a frame. Can A finish transmitting before it detects that B has transmitted? Why or why not? If the answer is yes, then A incorrectly believes that its frame was successfully transmitted without a collision. Hint: Suppose at time \(t=0\) bits, A begins transmitting a frame. In the worst case, A transmits a minimum-sized frame of \(512+64\) bit times. So A would finish transmitting the frame at \(t=512+64\) bit times. Thus, the answer is no, if B's signal reaches A before bit time \(t=512+64\) bits. In the worst case, when does B's signal reach A?

Consider a broadcast channel with \(N\) nodes and a transmission rate of \(R\) bps. Suppose the broadcast channel uses polling (with an additional polling node) for multiple access. Suppose the amount of time from when a node completes transmission until the subsequent node is permitted to transmit (that is, the polling delay) is \(d_{\text {poll }}\). Suppose that within a polling round, a given node is allowed to transmit at most \(Q\) bits. What is the maximum throughput of the broadcast channel?

In this problem, we explore the use of small packets for Voice-over-IP applications. One of the drawbacks of a small packet size is that a large fraction of link bandwidth is consumed by overhead bytes. To this end, suppose that the packet consists of \(P\) bytes and 5 bytes of header. a. Consider sending a digitally encoded voice source directly. Suppose the source is encoded at a constant rate of \(128 \mathrm{kbps}\). Assume each packet is entirely filled before the source sends the packet into the network. The time required to fill a packet is the packetization delay. In terms of \(L\), determine the packetization delay in milliseconds. b. Packetization delays greater than 20 msec can cause a noticeable and unpleasant echo. Determine the packetization delay for \(L=1,500\) bytes (roughly corresponding to a maximum-sized Ethernet packet) and for \(L=50\) (corresponding to an ATM packet). c. Calculate the store-and-forward delay at a single switch for a link rate of \(R=622 \mathrm{Mbps}\) for \(L=1,500\) bytes, and for \(L=50\) bytes. d. Comment on the advantages of using a small packet size.

Suppose four active nodes - nodes A, B, C and D-are competing for access to a channel using slotted ALOHA. Assume each node has an infinite number of packets to send. Each node attempts to transmit in each slot with probability \(p\). The first slot is numbered slot 1 , the second slot is numbered slot 2 , and so on. a. What is the probability that node A succeeds for the first time in slot \(5 ?\) b. What is the probability that some node (either A, B, C or D) succeeds in slot 4? c. What is the probability that the first success occurs in slot 3 ? d. What is the efficiency of this four-node system?

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