TCP's Retransmission Timeout

Follow the path in this section to see how the TCP retransmission timeout improves transport performance.

In a go-back-n transport protocol such as TCP, the retransmission timeout must be correctly set in order to achieve good performance. If the retransmission timeout expires too early, then bandwidth is wasted by retransmitting segments that have already been correctly received, whereas 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.

Figure 4.44: 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 acknowledgement 21. As illustrated in the figure below, this measurement works well when there are no segment losses.

Figure 4.45: 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 22. The second value, TS Echo Reply (TSecr), is the last TSval that was received from the remote host and stored in the 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.

Figure 4.46: 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 23 , 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 srrt = (α × srtt) + ((1 α) × rtt) where rtt is the round-trip-time measured according to the above procedure and α a smoothing factor (e.g. 0.8 or 0.9)
  • rto: the retransmission timeout is computed as rto = min(60, max(1, β × srtt)) where β 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 Ja- cobson 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: α and β. 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

srtt=rtt
rttvar=rtt/2
rto=srtt+4*rttvar

Then, when other rtt measurements are collected, srtt and rttvar are updated as follows:

rttvar = (1 β) × rttvar + β × |srttrtt|

srtt = (1 α) × srtt + α × rtt

rto = srtt + 4 × rttvar

The proposed values for the parameters are \alpha = \frac{1}{8} and \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.

Figure 4.47: Example computation of the rto


Source: Olivier Bonaventure, https://s3.amazonaws.com/saylordotorg-resources/wwwresources/site/wp-content/uploads/2012/02/Computer-Networking-Principles-Bonaventure-1-30-31-OTC1.pdf
Creative Commons License This work is licensed under a Creative Commons Attribution 3.0 License.

Last modified: Thursday, November 9, 2023, 2:17 PM