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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`.
The original TCP specification proposed in :rfc:`793` to include two additional variables in the TCB :
`srtt` : the smoothed round-trip-time computed as :math:`srtt=(\alpha \times srtt)+( (1-\alpha) \times rtt)` where :math:`rtt` is the round-trip-time measured according to the above procedure and :math:`\alpha` a smoothing factor (e.g. 0.8 or 0.9)
`rto` : the retransmission timeout is computed as :math:`rto=\min(60,max(1,\beta \times srtt))` where :math:`\beta` is used to take into account the delay variance (value : 1.3 to 2.0). The `60` and `1` constants are used to ensure that the `rto` is not larger than one minute nor smaller than 1 second.
However, in practice, this computation for the retransmission timeout did not work well. The main problem was that the computed `rto` did not correctly take into account the variations in the measured round-trip-time. `Van Jacobson` proposed in his seminal paper [Jacobson1988]_ an improved algorithm to compute the `rto` and implemented it in the BSD Unix distribution. This algorithm is now part of the TCP standard :rfc:`2988`.
Jacobson's algorithm uses two state variables, `srtt` the smoothed `rtt` and `rttvar` the estimation of the variance of the `rtt` and two parameters : :math:`\alpha` and :math:`\beta`. When a TCP connection starts, the first `rto` is set to `3` seconds. When a first estimation of the `rtt` is available, the `srtt`, `rttvar` and `rto` are computed as follows :
Then, when other rtt measurements are collected, `srtt` and `rttvar` are updated as follows :
:math:`rttvar=(1-\beta) \times rttvar + \beta \times |srtt - rtt|`
:math:`srtt=(1-\alpha) \times srtt + \alpha \times rtt`
:math:`rto=srtt + 4 \times rttvar`
The proposed values for the parameters are :math:`\alpha=\frac{1}{8}` and :math:`\beta=\frac{1}{4}`. This allows a TCP implementation, implemented in the kernel, to perform the `rtt` computation by using shift operations instead of the more costly floating point operations [Jacobson1988]_. The figure below illustrates the computation of the `rto` upon `rtt` changes.
Example computation of the `rto`
Advanced retransmission strategies
The default go-back-n retransmission strategy was defined in :rfc:`793`. When the retransmission timer expires, TCP retransmits the first unacknowledged segment (i.e. the one having sequence number `snd.una`). After each expiration of the retransmission timeout, :rfc:`2988` recommends to double the value of the retransmission timeout. This is called an `exponential backoff`. This doubling of the retransmission timeout after a retransmission was included in TCP to deal with issues such as network/receiver overload and incorrect initial estimations of the retransmission timeout. If the same segment is retransmitted several times, the retransmission timeout is doubled after every retransmission until it reaches a configured maximum. :rfc:`2988` suggests a maximum retransmission timeout of at least 60 seconds. Once the retransmission timeout reaches this configured maximum, the remote host is considered to be unreachable and the TCP connection is closed.
This retransmission strategy has been refined based on the experience of using TCP on the Internet. The first refinement was a clarification of the strategy used to send acknowledgments. As TCP uses piggybacking, the easiest and less costly method to send acknowledgments is to place them in the data segments sent in the other direction. However, few application layer protocols exchange data in both directions at the same time and thus this method rarely works. For an application that is sending data segments in one direction only, the remote TCP entity returns empty TCP segments whose only useful information is their acknowledgment number. This may cause a large overhead in wide area network if a pure `ACK` segment is sent in response to each received data segment. Most TCP implementations use a `delayed acknowledgment` strategy. This strategy ensures that piggybacking is used whenever possible, otherwise pure `ACK` segments are sent for every second received data segments when there are no losses. When there are losses or reordering, `ACK` segments are more important for the sender and they are sent immediately :rfc:`813` :rfc:`1122`. This strategy relies on a new timer with a short delay (e.g. 50 milliseconds) and one additional flag in the TCB. It can be implemented as follows.
Due to this delayed acknowledgment strategy, during a bulk transfer, a TCP implementation usually acknowledges every second TCP segment received.
The default go-back-n retransmission strategy used by TCP has the advantage of being simple to implement, in particular on the receiver side, but when there are losses, a go-back-n strategy provides a lower performance than a selective repeat strategy. The TCP developers have designed several extensions to TCP to allow it to use a selective repeat strategy while maintaining backward compatibility with older TCP implementations. These TCP extensions assume that the receiver is able to buffer the segments that it receives out-of-sequence.
The first extension that was proposed is the fast retransmit heuristic. This extension can be implemented on TCP senders and thus does not require any change to the protocol. It only assumes that the TCP receiver is able to buffer out-of-sequence segments.
From a performance point of view, one issue with TCP's `retransmission timeout` is that when there are isolated segment losses, the TCP sender often remains idle waiting for the expiration of its retransmission timeouts. Such isolated losses are frequent in the global Internet [Paxson99]_. A heuristic to deal with isolated losses without waiting for the expiration of the retransmission timeout has been included in many TCP implementations since the early 1990s. To understand this heuristic, let us consider the figure below that shows the segments exchanged over a TCP connection when an isolated segment is lost.
Detecting isolated segment losses

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locale/pot/protocols/tcp.pot, string 139