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A simple BNF specification
The example above defines several terminals and two commands : `usercommand` and `passwordcommand`. The `ALPHA` terminal contains all letters in upper and lower case. In the `ALPHA` rule, `%x41` corresponds to ASCII character code 41 in hexadecimal, i.e. capital `A`. The `CR` and `LF` terminals correspond to the carriage return and linefeed control characters. The `CRLF` rule concatenates these two terminals to match the standard end of line termination. The `DIGIT` terminal contains all digits. The `SP` terminal corresponds to the white space characters. The `usercommand` is composed of two strings separated by white space. In the ABNF rules that define the messages used by Internet applications, the commands are case-insensitive. The rule `"user"` corresponds to all possible cases of the letters that compose the word between brackets, e.g. `user`, `uSeR`, `USER`, `usER`, ... A `username` contains at least one letter and up to 8 letters. User names are case-sensitive as they are not defined as a string between brackets. The `password` rule indicates that a password starts with a letter and can contain any number of letters or digits. The white space and the control characters cannot appear in a `password` defined by the above rule.
Besides character strings, some applications also need to exchange 16 bits and 32 bits fields such as integers. A naive solution would have been to send the 16- or 32-bits field as it is encoded in the host's memory. Unfortunately, there are different methods to store 16- or 32-bits fields in memory. Some CPUs store the most significant byte of a 16-bits field in the first address of the field while others store the least significant byte at this location. When networked applications running on different CPUs exchange 16 bits fields, there are two possibilities to transfer them over the transport service :
send the most significant byte followed by the least significant byte
send the least significant byte followed by the most significant byte
The first possibility was named `big-endian` in a note written by Cohen [Cohen1980]_ while the second was named `little-endian`. Vendors of CPUs that used `big-endian` in memory insisted on using `big-endian` encoding in networked applications while vendors of CPUs that used `little-endian` recommended the opposite. Several studies were written on the relative merits of each type of encoding, but the discussion became almost a religious issue [Cohen1980]_. Eventually, the Internet chose the `big-endian` encoding, i.e. multi-byte fields are always transmitted by sending the most significant byte first, :rfc:`791` refers to this encoding as the :term:`network-byte order`. Most libraries [#fhtonl]_ used to write networked applications contain functions to convert multi-byte fields from memory to the network byte order and the reverse.
Besides 16 and 32 bit words, some applications need to exchange data structures containing bit fields of various lengths. For example, a message may be composed of a 16 bits field followed by eight, one bit flags, a 24 bits field and two 8 bits bytes. Internet protocol specifications will define such a message by using a representation such as the one below. In this representation, each line corresponds to 32 bits and the vertical lines are used to delineate fields. The numbers above the lines indicate the bit positions in the 32-bits word, with the high order bit at position `0`.
Message format
The message mentioned above will be transmitted starting from the upper 32-bits word in network byte order. The first field is encoded in 16 bits. It is followed by eight one bit flags (`A-H`), a 24 bits field whose high order byte is shown in the first line and the two low order bytes appear in the second line followed by two one byte fields. This ASCII representation is frequently used when defining binary protocols. We will use it for all the binary protocols that are discussed in this book.
The peer-to-peer model emerged during the last ten years as another possible architecture for networked applications. In the traditional client-server model, hosts act either as servers or as clients and a server serves a large number of clients. In the peer-to-peer model, all hosts act as both servers and clients and they play both roles. The peer-to-peer model has been used to develop various networked applications, ranging from Internet telephony to file sharing or Internet-wide filesystems. A detailed description of peer-to-peer applications may be found in [BYL2008]_. Surveys of peer-to-peer protocols and applications may be found in [AS2004]_ and [LCP2005]_.
The transport layer
A network is always designed and built to enable applications running on hosts to exchange information. In a previous chapter, we have explained the principles of the `network layer` that enables hosts connected to different types of datalink layers to exchange information through routers. These routers act as relays in the network layer and ensure the delivery of packets between any pair of hosts attached to the network.
The network layer ensures the delivery of packets on a hop-by-hop basis through intermediate nodes. As such, it provides a service to the upper layer. In practice, this layer is usually the `transport layer` that improves the service provided by the `network layer` to make it usable by applications.
Most networks use a datagram organization and provide a simple service which is called the `connectionless service`.
The figure below provides a representation of the connectionless service as a `time-sequence diagram`. The user on the left, having address `S`, issues a `Data.request` primitive containing Service Data Unit (SDU) `M` that must be delivered by the service provider to destination `D`. The dashed line between the two primitives indicates that the `Data.indication` primitive that is delivered to the user on the right corresponds to the `Data.request` primitive sent by the user on the left.
There are several possible implementations of the connectionless service. Before studying these realizations, it is useful to discuss the possible characteristics of the connectionless service. A `reliable connectionless service` is a service where the service provider guarantees that all SDUs submitted in `Data.requests` by a user will eventually be delivered to their destination. Such a service would be very useful for users, but guaranteeing perfect delivery is difficult in practice. For this reason, network layers usually support an `unreliable connectionless service`.
An `unreliable connectionless` service may suffer from various types of problems compared to a `reliable connectionless service`. First of all, an `unreliable connectionless service` does not guarantee the delivery of all SDUs. This can be expressed graphically by using the time-sequence diagram below.
In practice, an `unreliable connectionless service` will usually deliver a large fraction of the SDUs. However, since the delivery of SDUs is not guaranteed, the user must be able to recover from the loss of any SDU.
A second imperfection that may affect an `unreliable connectionless service` is that it may duplicate SDUs. Some packets may be duplicated in a network and be delivered twice to their destination. This is illustrated by the time-sequence diagram below.
Finally, some unreliable connectionless service providers may deliver to a destination a different SDU than the one that was supplied in the `Data.request`. This is illustrated in the figure below.
As the transport layer is built on top of the network layer, it is important to know the key features of the network layer service. In this book, we only consider the `connectionless network layer service` which is the most widespread. Its main characteristics are :
the `connectionless network layer service` can only transfer SDUs of *limited size*
the `connectionless network layer service` may discard SDUs
the `connectionless network layer service` may corrupt SDUs
the `connectionless network layer service` may delay, reorder or even duplicate SDUs
These imperfections of the `connectionless network layer service` are caused by the operations of the `network layer`. This `layer` is able to deliver packets to their intended destination, but it cannot guarantee their delivery. The main cause of packet losses and errors are the buffers used on the network nodes. If the buffers of one of these nodes becomes full, all arriving packets must be discarded. This situation frequently happens in practice [#fqueuesize]_. Transmission errors can also affect packet transmissions on links where reliable transmission techniques are not enabled or because of errors in the buffers of the network nodes.
Transport layer services
When two applications need to communicate, they need to structure their exchange of information. Structuring this exchange of information requires solving two different problems. The first problem is how to represent the information being exchanged knowing that the two applications may be running on hosts that use different operating systems, different processors and have different conventions to store information. This requires a common syntax to transfer the information between the two applications. For this chapter, let us assume that this syntax exists and that the two applications simply need to exchange bytes. We will discuss later how more complex data can be encoded as sequences of bytes to be exchanged. The second problem is how to organize the interactions between the application and the underlying network. From the application's viewpoint, the `network` will appear as the `transport layer` service. This `transport layer` can provide three types of services to the applications :
the `connectionless service`
the `connection oriented service`
the `request-response service`

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Source string location
../../principles/transport.rst:179
String age
4 years ago
Source string age
4 years ago
Translation file
locale/pot/principles/transport.pot, string 52