1. Theoretical Basis
This project aims to address the inefficiency of the traditional TCP protocol in high-loss network environments by introducing a transformation based on network coding. The source code is modified by integrating network coding concepts into both the TCP and IP layers. A new network coding layer is introduced, which enables encoding at the sender, decoding at the receiver, and dynamically adjusting the redundancy of encoded segments based on real-time network loss rates. This helps to mitigate packet loss and improve the reliability of data transmission at the TCP layer. Additionally, we incorporate processor sharing technology to replace the traditional TCP slow start mechanism with a more efficient initial transmission rate. The ultimate goal is to enhance network throughput, increase reliability, and reduce the time required for data flows to complete.
2. Network Coding
Network coding was first introduced in 2000 by researchers from the Department of Information Engineering at the Chinese University of Hong Kong. It addresses the limitations of traditional routing by allowing intermediate nodes to combine and forward multiple packets instead of simply forwarding them as they are. At the receiving end, these combined packets are decoded to recover the original data. This approach significantly improves network efficiency, especially in multicast scenarios.
A simple way to understand network coding is through the butterfly topology shown in Figure 1-1:
Figure 1-1: Basic Principles of Network Coding
In this example, each link can carry only one bit per unit of time. If standard routing is used, the UV link becomes the bottleneck. Only one of the two messages (a or b) can be transmitted at a time. However, with network coding, the two messages are combined and sent over the UV link. As a result, both sink nodes P and Q can receive and decode both messages, achieving an average throughput of 2 bits per sink node, compared to 1.5 bits when using traditional routing.
This demonstrates that network coding can achieve the maximum capacity of a multicast network, even when traditional routing cannot. Since its introduction, network coding has evolved rapidly and has become a key area of research in information theory, communication systems, and computer networks. It has applications in various fields such as random coding, linear coding, wireless networks, sensor networks, and distributed storage systems.
Many leading universities, including Princeton, MIT, and EPFL, as well as companies like Microsoft Research, Bell Labs, and AT&T Shannon Lab, have invested heavily in both theoretical and applied research on network coding. In recent years, domestic institutions such as Tsinghua University, Xidian University, and the University of Science and Technology of China have also started to explore this promising field.
3. TCP Protocol
The Transmission Control Protocol (TCP) is a connection-oriented, reliable, and byte-stream-based transport layer protocol. It operates at the fourth layer of the OSI model and is responsible for ensuring accurate and ordered delivery of data between applications.
TCP uses port numbers to facilitate communication between processes and is stream-oriented, meaning it treats data as a continuous sequence of bytes. Each byte is assigned a sequence number, which is randomly generated at the start of the connection. When data is sent, each segment is given a sequence number corresponding to the first byte of data in that segment. Upon receiving data, the receiver sends an acknowledgment (ACK) with the next expected byte number, which is cumulative.
Figure 1-2: Flow Transmission
Figure 1-3: Sequence Number
In practice, to avoid congestion, TCP may send one byte at a time and wait for an ACK before sending the next. However, in long-distance transmissions, this can lead to underutilization of the channel. To address this, TCP employs the sliding window protocol, which allows multiple segments to be sent before requiring an acknowledgment, thereby improving efficiency and throughput.
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