The Internet today is what is called a packet switched network. Information is broken into small packets. Each packet has a destination address. Each router / packet switch along the path looks up the destination address, and sends the packet along in the right direction. Is this the only way to build networks? No. There is also circuit switching.
Circuit-swtching (CS) vs packet switching (PS). These are two fundamantal ways of transporting data through a network. In circuit switching, before starting a flow, a setup phase happens. A setup message goes through the network, resources at all links are reserved for the flow. That is, a dedicated circuit is established from one end point to the other, through which you can push data. The links along the path are shared by time/frequency division multiplexing (TDM/FDM) between various circuits. With TDM for example, you have a timeslot reserved for every established flow on all links. Telephone networks work this way. Why? Lower delay and jitter (variance of delay). But CS may potentially waste resources (e.g., when voice call is idle).
Packet switching is different from CS. There is no explicit setup phase for a flow. Whenever a source has a packet for a destination, the packet is sent along the first link. The node at the end of that link receives the packet and forwards it along as soon as it can. And so on. No explicit resources are reserved at any intermediate node for any flow. So if a link is busy, incoming packets may have to be temporarily stored in a PS network. Thus the intermediate nodes are called “store-and-forward” switches.
The links in PS are shared not by explicitly reserving resources, but by “statistical multiplexing”. That is, all packets that arrive at a node are sent out on the outgoing link in a first-come-first-serve or some other order. Intermediate nodes have buffers to store packets till they are forwarded. Packets may incur a “queueing delay” if there are many packets arriving at a node and waiting to be forwarded. Some packets may also be dropped if the buffer at an intermediate node overflows. Thus you cannot provide any guaranteed Quality of service (QoS), it is only “best effort”. Packet switched networks incur longer delays and jitter, but lead to better utilization of resources.
The concept of packet swtching started in the 1960s. Until then, all networks (telephone) were all circuit switched. PS was seen as a better option for data transfer (not voice) due to the burstiness and long idle periods in data transfer (as opposed to voice). The Internet is packet switched. Recently, cellular data networks (4G etc) are also moving towards packet switching, but with better scheduling algorithms to provide QoS.
Problem: statistical multiplexing vs TDMA.
Consider a link of rate 1 Mbps. Users generating data at 100 kbps when busy, but are busy with probability 0.1. With TDMA, we can support only 10 users, and 90% of slots are wasted. If M users in total, probability that N users are active = choose(M,N) * p^N * (1-p)^{M-N}. So, for 35 users, prob that 11 or more active is 0.0004. So most of the time it is okay as 10 or fewer users are active. When more than 10 users active, queueing takes place, and queue drains when fewer than 10 users are active. Statistical mux can support 3 times as many users. Also, leads to better utilization of the link.
Note the notion of a packet: the unit at which you transmit information. If you have a big application message, you split it into smaller packets, put a header on each one, and send each one independently through the packet switched network. Headers are not so important for circuit switching because a circuit and timeslot schedule is established. Headers are a must for packet switching. How big should a packet be? Small packets - relatively greater overhead due to repeating header information in every packet, so larger packets are better this way. But with large packets, there is a long delay for the packet to traverse every hop. With smaller packets, the first packet can do down the second link while the first packet arrives. Also, even if one bit is corrupted, the full packet must be retransmitted (at link layer or transport layer). So choosing optimum packet size is a fine balancing act.
History of networking: The only networks were telephone networks for a long time. Packet switching concepts and theory developed in the 1960s, mainly to transfer data. Initially, it wasn’t clear if the queueing delays and uncertainty of PS (as compared to CS) would be manageable or not, but several important theoretical results proved that it would all work fine.
Several proprietary packet switched networks started in the 1970s beginning with ARPANET. The need came to up to integrate these networks using a few top-level store-and-forward gateways. The primitive TCP/IP protocols came into place, but TCP and IP were tightly coupled were integrated. What lead to the split into layers we see now?
One of the main goals of the internet - survivability, fault tolerance. The intermediate gateways have a lot of state. How should this state be preserved across failures? It was decided that the core would store soft state (periodically refreshed, nothing permanent, okay to lose state once in a while), and would only forward datagrams, in a connection-less manner. The end hosts would maintain any connection state needed.
People also realized that TCP provides only one type of transport, and people may want to use different transport protocols for different types of applications. For example, not all applications will want the reliability provided by TCP. So, TCP and IP were split. UDP was provided as another alternative simpler transport.
Finally, we have IP that runs on any network that can deliver a datagram from one IP address to another. No assumptions made on the underlying network (in order, reliable etc). TCP and other transport run on top of IP and customize the transport protocol to the application needs.
The split of TCP and IP is also referred to as the end-to-end argument. All application and transport “intelligence” (reliability, security etc) should reside at the end hosts. The core of the network should be kept “dumb”, with only the job of delivering datagrams as well as it can. It may be possible to spend a lot of energy in engineering the network core to do some of the jobs of the higher layers (e.g., provide reliability guarantees), but it was decided that the effort to realize it in the core was too high to be worth it, and the end points would be doing it anyways. End-to-end checks, retries, recovery at end-to-end level etc makes more sense. The tradeoff of where to place a certain functionality - end hosts or network - is a common debate in several aspects of networking.
Continuing the story, the 1980s saw a refining of the TCP congestion control algorithms. The internet grows. US government lets go of the backbone routers and decides to commercialize it. ISPs come up in 1990s. Late 1990s, early 2000s - dotcom bubble, explosion of usage and applications. Late 2000s - stabilization, growth of cellular/mobile data and convergence with the Internet.
Demo: show wireshark trace of a browsing session