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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 :
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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.
How to measure the round-trip-time ?
However, when a data segment is lost, as illustrated in the bottom part of the figure, the measurement is ambiguous as the sender cannot determine whether the received acknowledgment was triggered by the first transmission of segment `123` or its retransmission. Using incorrect round-trip-time estimations could lead to incorrect values of the retransmission timeout. For this reason, Phil Karn and Craig Partridge proposed, in [KP91]_, to ignore the round-trip-time measurements performed during retransmissions.
To avoid this ambiguity in the estimation of the round-trip-time when segments are retransmitted, recent TCP implementations rely on the `timestamp option` defined in :rfc:`1323`. This option allows a TCP sender to place two 32 bit timestamps in each TCP segment that it sends. The first timestamp, TS Value (`TSval`) is chosen by the sender of the segment. It could for example be the current value of its real-time clock [#ftimestamp]_. The second value, TS Echo Reply (`TSecr`), is the last `TSval` that was received from the remote host and stored in the :term:`TCB`. The figure below shows how the utilization of this timestamp option allows for the disambiguation of the round-trip-time measurement when there are retransmissions.
Disambiguating round-trip-time measurements with the :rfc:`1323` timestamp option
Once the round-trip-time measurements have been collected for a given TCP connection, the TCP entity must compute the retransmission timeout. As the round-trip-time measurements may change during the lifetime of a connection, the retransmission timeout may also change. At the beginning of a connection [#ftcbtouch]_, the TCP entity that sends a `SYN` segment does not know the round-trip-time to reach the remote host and the initial retransmission timeout is usually set to 3 seconds :rfc:`2988`.

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