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In a transport protocol such as TCP that offers a bytestream, a practical issue that was left as an implementation choice in :rfc:`793` is to decide when a new TCP segment containing data must be sent. There are two simple and extreme implementation choices. The first implementation choice is to send a TCP segment as soon as the user has requested the transmission of some data. This allows TCP to provide a low delay service. However, if the user is sending data one byte at a time, TCP would place each user byte in a segment containing 20 bytes of TCP header [#fnagleip]_. This is a huge overhead that is not acceptable in wide area networks. A second simple solution would be to only transmit a new TCP segment once the user has produced MSS bytes of data. This solution reduces the overhead, but at the cost of a potentially very high delay.
An elegant solution to this problem was proposed by John Nagle in :rfc:`896`. John Nagle observed that the overhead caused by the TCP header was a problem in wide area connections, but less in local area connections where the available bandwidth is usually higher. He proposed the following rules to decide to send a new data segment when a new data has been produced by the user or a new `ack` segment has been received.
The first rule ensures that a TCP connection used for bulk data transfer always sends full TCP segments. The second rule sends one partially filled TCP segment every round-trip-time.
This algorithm, called the Nagle algorithm, takes a few lines of code in all TCP implementations. These lines of code have a huge impact on the packets that are exchanged in TCP/IP networks. Researchers have analyzed the distribution of the packet sizes by capturing and analyzing all the packets passing through a given link. These studies have shown several important results :
in TCP/IP networks, a large fraction of the packets are TCP segments that contain only an acknowledgment. These packets usually account for 40-50% of the packets passing through the studied link
in TCP/IP networks, most of the bytes are exchanged in long packets, usually packets containing about 1440 bytes of payload which is the default MSS for hosts attached to an Ethernet network, the most popular type of LAN
`Recent measurements <http://www.caida.org/research/traffic-analysis/pkt_size_distribution/graphs.xml>`_ indicate that these packet size distributions are still valid in today's Internet, although the packet distribution tends to become bi-modal with small packets corresponding to TCP pure acknowledgments and large 1440-bytes packets carrying most of the user data [SMASU2012]_.
TCP windows
From a performance point of view, one of the main limitations of the original TCP specification is the 16 bits `window` field in the TCP header. As this field indicates the current size of the receive window in bytes, it limits the TCP receive window at 65535 bytes. This limitation was not a severe problem when TCP was designed since at that time high-speed wide area networks offered a maximum bandwidth of 56 kbps. However, in today's network, this limitation is not acceptable anymore. The table below provides the rough [#faveragebandwidth]_ maximum throughput that can be achieved by a TCP connection with a 64 KBytes window in function of the connection's round-trip-time
RTT
Maximum Throughput
1 msec
524 Mbps
10 msec
52.4 Mbps
100 msec
5.24 Mbps
500 msec
1.05 Mbps
To solve this problem, a backward compatible extension that allows TCP to use larger receive windows was proposed in :rfc:`1323`. Today, most TCP implementations support this option. The basic idea is that instead of storing `snd.wnd` and `rcv.wnd` as 16 bits integers in the :term:`TCB`, they should be stored as 32 bits integers. As the TCP segment header only contains 16 bits to place the window field, it is impossible to copy the value of `snd.wnd` in each sent TCP segment. Instead the header contains `snd.wnd >> S` where `S` is the scaling factor ( :math:`0 \le S \le 14`) negotiated during connection establishment. The client adds its proposed scaling factor as a TCP option in the `SYN` segment. If the server supports :rfc:`1323`, it places in the `SYN+ACK` segment the scaling factor that it uses when advertising its own receive window. The local and remote scaling factors are included in the :term:`TCB`. If the server does not support :rfc:`1323`, it ignores the received option and no scaling is applied.
By using the window scaling extensions defined in :rfc:`1323`, TCP implementations can use a receive buffer of up to 1 GByte. With such a receive buffer, the maximum throughput that can be achieved by a single TCP connection becomes :
8590 Gbps
859 Gbps
86 Gbps
17 Gbps
These throughputs are acceptable in today's networks. However, there are already servers having 10 Gbps interfaces... Early TCP implementations had fixed receiving and sending buffers [#ftcphosts]_. Today's high performance implementations are able to automatically adjust the size of the sending and receiving buffer to better support high bandwidth flows [SMM1998]_.
TCP's retransmission timeout
In a go-back-n transport protocol such as TCP, the retransmission timeout must be correctly set in order to achieve good performance. On one hand, if the retransmission timeout expires too early, then bandwidth is wasted by retransmitting segments that have already been correctly received. On the other hand, if the retransmission timeout expires too late, then bandwidth is wasted because the sender is idle waiting for the expiration of its retransmission timeout.
A good setting of the retransmission timeout clearly depends on an accurate estimation of the round-trip-time of each TCP connection. The round-trip-time differs between TCP connections, but may also change during the lifetime of a single connection. For example, the figure below shows the evolution of the round-trip-time between two hosts during a period of 45 seconds.
Evolution of the round-trip-time between two hosts
The easiest solution to measure the round-trip-time on a TCP connection is to measure the delay between the transmission of a data segment and the reception of a corresponding acknowledgment [#frttmes]_. As illustrated in the figure below, this measurement works well when there are no segment losses.

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../../protocols/tcp.rst:336 • ../../protocols/tcp.rst:349
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locale/fr/LC_MESSAGES/protocols/tcp.po, string 115