Translation components API.

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            "timestamp": "2022-09-17T01:14:37.607293+02:00",
            "action": 59,
            "target": "This is an unpolished draft of the third edition of this e-book. If you find any error or have suggestions to improve the text, please create an issue via https://github.com/CNP3/ebook/issues?milestone=4 or help us by providing pull requests to close the existing issues.",
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            "timestamp": "2022-09-17T01:14:37.607380+02:00",
            "action": 59,
            "target": "The first and more important resource inside a network is the link bandwidth. There are two situations where link bandwidth needs to be shared between different users. The first situation is when several hosts are attached to the same physical link. This situation mainly occurs in Local Area Networks (LAN). A LAN is a network that efficiently interconnects several hosts (usually a few dozens to a few hundreds) in the same room, building or campus. Consider for example a network with five hosts. Any of these hosts needs to be able to exchange information with any of the other hosts. A first organization for this LAN is the full-mesh.",
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            "target": "The full-mesh is the most reliable and highest performing network to interconnect these five hosts. However, this network organization has two important drawbacks. First, if a network contains `n` hosts, then :math:`\\frac{n\\times(n-1)}{2}` links are required. If the network contains more than a few hosts, it becomes impossible to lay down the required physical links. Second, if the network contains `n` hosts, then each host must have :math:`n-1` interfaces to terminate :math:`n-1` links. This is beyond the capabilities of most hosts. Furthermore, if a new host is added to the network, new links have to be laid down and one interface has to be added to each participating host. However, full-mesh has the advantage of providing the lowest delay between the hosts and the best resiliency against link failures. In practice, full-mesh networks are rarely used except when there are few network nodes and resiliency is key.",
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            "timestamp": "2022-09-17T01:14:37.607511+02:00",
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            "target": "Sharing resources is important to ensure that the network efficiently serves its user. In practice, there are many ways to share resources. Some resource sharing schemes consider that some users are more important than others and should obtain more resources. For example, on the roads, police cars and ambulances have priority. In some cities, traffic lanes are reserved for buses to promote public services, ... In computer networks, the same problem arise. Given that resources are limited, the network needs to enable users to efficiently share them. Before designing an efficient resource sharing scheme, one needs to first formalize its objectives. In computer networks, the most popular objective for resource sharing schemes is that they must be `fair`. In a simple situation, for example two hosts using a shared 2 Mbps link, the sharing scheme should allocate the same bandwidth to each user, in this case 1 Mbps. However, in a large networks, simply dividing the available resources by the number of users is not sufficient. Consider the network shown in the figure below where `A1` sends data to `A2`, `B1` to `B2`, ... In this network, how should we divide the bandwidth among the different flows ? A first approach would be to allocate the same bandwidth to each flow. In this case, each flow would obtain 5 Mbps and the link between `R2` and `R3` would not be fully loaded. Another approach would be to allocate 10 Mbps to `A1-A2`, 20 Mbps to `C1-C2` and nothing to `B1-B2`. This is clearly unfair.",
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            "timestamp": "2022-09-17T01:14:37.607569+02:00",
            "action": 59,
            "target": "Sharing bandwidth among the hosts directly attached to a link is not the only sharing problem that occurs in computer networks. To understand the general problem, let us consider a very simple network which contains only point-to-point links. This network contains three hosts and two routers. All the links inside the network have the same capacity. For example, let us assume that all links have a bandwidth of 1000 bits per second and that the hosts send packets containing exactly one thousand bits.",
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            "target": "If `R1` has enough buffers, it will be able to absorb the load without having to discard packets. The packets sent by hosts `A` and `B` will reach their final destination `C`, but will experience a longer delay than when they are transmitting alone. The amount of buffering on the network node is the first parameter that a network operator can tune to control congestion inside his network. Given the decreasing cost of memory, one could be tempted to put as many buffers [#fbufferbloat]_ as possible on the network nodes. Let us consider this case in the network above and assume that `R1` has infinite buffers. Assume now that hosts `A` and `B` try to transmit a file that corresponds to one thousand packets each. Both are using a reliable protocol that relies on go-back-n to recover from transmission errors. The transmission starts and packets start to accumulate in `R1`'s buffers. The presence of these packets in the buffers increases the delay between the transmission of a packet by `A` and the return of the corresponding acknowledgment. Given the increasing delay, host `A` (and `B` as well) will consider that some of the packets that it sent have been lost. These packets will be retransmitted and will enter the buffers of `R1`. The occupancy of the buffers of `R1` will continue to increase and the delays as well. This will cause new retransmissions, ... In the end, only one file will be delivered (very slowly) to the destination, but the link `R1-R2` will transfer much more bytes than the size of the file due to the multiple copies of the same packets. This is known as the `congestion collapse` problem :rfc:`896`. Congestion collapse is the nightmare for network operators. When it happens, the network carries packets without delivering useful data to the end users.",
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            "target": "Besides bandwidth and memory, a third resource that needs to be shared inside a network is the (packet) processing capacity. To forward a packet, a router needs bandwidth on the outgoing link, but it also needs to analyze the packet header to perform a lookup inside its forwarding table. Performing these lookup operations require resources such as CPU cycles or memory accesses. Routers are usually designed to be able to sustain a given packet processing rate, measured in packets per second [#fpps]_.",
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            "timestamp": "2022-09-17T01:14:37.607726+02:00",
            "action": 59,
            "target": "The performance of network nodes (either routers or switches) can be characterized by two key metrics :",
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            "action": 59,
            "target": "Let us now try to present a broad overview of the congestion problem in networks. We will assume that the network is composed of dedicated links having a fixed bandwidth [#fadjust]_. A network contains hosts that generate and receive packets and nodes (routers and switches) that forward packets. Assuming that each host is connected via a single link to the network, the largest demand is :math:`\\sum{Access Links}`. In practice, this largest demand is never reached and the network will be engineered to sustain a much lower traffic demand. The difference between the worst-case traffic demand and the sustainable traffic demand can be large, up to several orders of magnitude. Fortunately, the hosts are not completely dumb and they can adapt their traffic demand to the current state of the network and the available bandwidth. For this, the hosts need to `sense` the current level of congestion and adjust their own traffic demand based on the estimated congestion. Network nodes can react in different ways to network congestion and hosts can sense the level of congestion in different ways.",
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            "timestamp": "2022-09-17T01:14:37.607847+02:00",
            "action": 59,
            "target": "`What triggers a packet to be discarded ?` What are the conditions that lead a network node to decide to discard a packet? The simplest answer to this question is : `When the buffer is full`. Although this is a good congestion indication, it is probably not the best one from a performance viewpoint. An alternative is to discard packets when the buffer occupancy grows too much. In this case, it is likely that the buffer will become full shortly. Since packet discarding is an information that allows hosts to adapt their transmission rate, discarding packets early could allow hosts to react earlier and thus prevent congestion from happening.",
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            "target": "`Tail drop` is the simplest packet discard technique. When a buffer is full, the arriving packet is discarded. `Tail drop` can be easily implemented. This is, by far, the most widely used packet discard mechanism. However, it suffers from two important drawbacks. First, since `tail drop` discards packets only when the buffer is full, buffers tend to be congested and real-time applications may suffer from increased delays. Second, `tail drop` is blind when it discards a packet. It may discard a packet from a low bandwidth interactive flow while most of the buffer is used by large file transfers.",
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            "timestamp": "2022-09-17T01:14:37.607950+02:00",
            "action": 59,
            "target": "`Drop from front` is an alternative packet discard technique. Instead of removing the arriving packet, it removes the packet that was at the head of the queue. Discarding this packet instead of the arriving one can have two advantages. First, it already stayed a long time in the buffer. Second, hosts should be able to detect the loss (and thus the congestion) earlier.",
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