First of all, in the project, I cannot use any transport protocol libraries, such as TCP. Therefore, I construct the packets and acknowledgements, and interpret the incoming packets by myself. A datagram is a basic transfer unit associated with a packet-switched network. Datagrams are typically structured in header and payload sections. Datagrams provide a connectionless communication service across a packet-switched network. The delivery, arrival time, and order of arrival of datagrams need not be guaranteed by the network. \subsection{Design the packet format} Like in Project 1 I need to design fields for packet type, ACK number, window size, data etc. Each byte of data sent as in a TCP connection has an associated sequence number. …show more content…
if send is called twice with data sizes 1000 and 2000 then two packets will be sent of sizes 1000 and 2000 (assuming the MTU size for the network is large enough). If send is called with a data size larger than the MTU value, the packet will be fragmented. When the remote host calls receive twice with sizes 100 and 150, the first call will return the first 100 bytes of the 1000 byte packet and throw away the last 900 bytes. The second call will return the first 150 bytes of the 2000 byte packet and throw away the last 1850 bytes. The client on either side of a TCP session maintains a 32-bit sequence number it uses to keep track of how much data it has sent. This sequence number is included on each transmitted packet, and acknowledged by the opposite host as an acknowledgement number to inform the sending host that the transmitted data was received successfully. When a host initiates a TCP session, its initial sequence number is effectively random; it may be any value between 0 and 4,294,967,295, inclusive. However, protocol analyzers like Wireshark will typically display relative sequence and acknowledgement numbers in place of the actual values. These numbers are relative to the initial sequence number of that stream. This is handy, as it is much easier to keep track of relatively small, predictable numbers rather than the actual numbers sent on the wire. \section{ different configurations of netem} Network emulation
TCP/IP is a protocol which transfers data across a network. It allows two different computers to communicate well even if they use different codes. Putting both computers into a new common code language which both computers
Handshake: - This sub protocol is used to negotiate session information between the client and server. The session information consists of a session ID, peer certificates, the cipher spec to be used, the compression algorithm to be used, and a shared secret that is used to generate keys.
pieces. The data packet is sent to a gateway computer which either recognizes the address
The User Datagram Protocol (UDP) is defined for use with IP network layer protocol in the transport layer protocol. It provides a best-effort datagram service to an End System (IP host). It uses a simple connectionless transmission model with a minimum of protocol mechanism. It uses no handshaking dialogues, and thus the user's program is exposed to any unreliability of the underlying network protocol. There is no guarantee of delivery, ordering, or duplicate protection. UDP provides checksums for data integrity, and port numbers for addressing different functions at the source and destination of the datagram.
Module 1 establishes the framework of networking standards and protocols for the rest of the class. The module introduces data communications and defines their components and the types of data exchanged. You will learn how different types of data are represented and how data is flowed through the network.
12. In the encrypted handshake record, the session will generate a MAC of the concatenation that includes all the previous handshake messages sent from the client, and then send this concatenation to the server.
The messages are M1,M2,M3,M4 and after many send and receive the process finally reached the host1.
3. Maximum packet lifetime needs to be large enough so that the packet and its acknowledgement have disappeared
In general, on the internet, every packet follows 5 layers of structures (Application, Transport, Network, Data Link, and Physical). Since I have to talk about Connection-Oriented and Connectionless as a service, I will focus on the Transport layer because they are part of this layer. It allows multiple applications to use one network connection simultaneously, the layer will create ports on your computer allowing them to be reserved and used by the Application layer. Let's say that port 3600 will be used to send a message to port 80 of some other computer, whenever the port is ready to send a message, it passed on to Transport layer. In this layer, it wraps the message inside the Segment which contains some additional
RFC 1180: This RFC is based on a tutorial of TCP/IP protocol. The focus of this tutorial is to tell how the datagram is forwarded from source to destination. The distribution of this memo is unlimited. This RFC defines the key points of TCP/IP, it completely skips the history and development of TCP/IP. It just gives a minimum information related to TCP/IP which is only understandable by technical
Internet Protocol (IP) datagrams may arrive in a seemingly random order of chunks that the receiving IP entity must continuously collect until it can reconstruct the original datagram. Consider that the receiving IP entity possesses a buffer for assembling the original datagram's data field. The buffer will comprise of chunks of data and "holes" between them corresponding to data not yet received.
The TCP/IP protocols are the heart and soul of the Internet, and they describe the fundamental rules that govern all communications in the network. The original address system of the Internet is Internet Protocol version 4 (IPv4). The Internet Engineering Task Force (IETF) developed IPv6 to deal with the long-anticipated problem of IPv4 address exhaustion. However, IPv6 is not foreseen to supplant IPv4 instantaneously.
Packet switched systems utilize a customary technique by exchanging each single packet to the destination address at nonstop interims however the greatest result for every interim will be the same one, this will include a few different routes or paths . This is one method to ensure the nature of the methodology by permitting one and only single packet at a solitary interim to the destination address. In another way it can be seen as moving information in divided little blocks to the destination address which are again assembled in a exact sequence to produce a message.
The ECN mechanism for TCP/IP protocol \cite{floyd1994tcp}\cite{ramakrishnan1998proposal}, TCP-ECN for short, is an end-to-end congestion control scheme deployed on layer 3 and layer 4. TCP-ECN employs Random Early Detection (RED) mechanism to mark the congested packet with \textit{Congestion Experienced} (CE) bit, which is carried in packet's IP header. And the end-host enables ECN-Capable TCP (ECT) bit in the packet header to support the ECN mechanism. Fig. \ref{TCP-ECN} illustrates the working process of TCP-ECN. When the queue length exceeds a fixed threshold, the router will use RED to set the CE bit in the packet header instead of dropping the packet, and
Well-known communications protocols are Ethernet, a hardware and Link standard that is ubiquitous in local area networks, and the Internet Protocol Suite, which defines a set of protocols for internetworking, i.e. for data communication between multiple networks, as well as host-to-host data transfer, and application-specific data transmission formats.