Recent work on TCP performance has shown that TCP can work well over a variety of Internet paths, ranging from 800 Mbit/sec I/O channels to 300 bit/sec dial-up modems [Jacobson88]. However, there is still a fundamental TCP performance bottleneck for one transmission regime: paths with high bandwidth and long round-trip delays. The significant parameter is the product of bandwidth (bits per second) and round-trip delay (RTT in seconds); this product is the number of bits it takes to "fill the pipe", i.e., the amount of unacknowledged data that TCP must handle in order to keep the pipeline full. TCP performance problems arise when this product is large, e.g., significantly exceeds 10**5 bits. We will refer to an Internet path operating in this region as a "long, fat pipe", and a network containing this path as an "LFN" (pronounced "elephan(t)").
High-capacity packet satellite channels (e.g., DARPA's Wideband Net) are LFN's. For example, a T1-speed satellite channel has a bandwidth*delay product of 10**6 bits or more; this corresponds to 100 outstanding TCP segments of 1200 bytes each! Proposed future terrestrial fiber-optical paths will also fall into the LFN class; for example, a cross-country delay of 30 ms at a DS3 bandwidth (45Mbps) also exceeds 10**6 bits.
Clever algorithms alone will not give us good TCP performance over LFN's; it will be necessary to actually extend the protocol. This RFC proposes a set of TCP extensions for this purpose.
There are three fundamental problems with the current TCP over LFN paths:
The TCP header uses a 16 bit field to report the receive window size to the sender. Therefore, the largest window that can be used is 2**16 = 65K bytes. (In practice, some TCP implementations will "break" for windows exceeding 2**15, because of their failure to do unsigned arithmetic).
To circumvent this problem, we propose a new TCP option to allow windows larger than 2**16. This option will define an implicit scale factor, to be used to multiply the window size value found in a TCP header to obtain the true window size.
Any packet losses in an LFN can have a catastrophic effect on throughput. This effect is exaggerated by the simple cumulative acknowledgment of TCP. Whenever a segment is lost, the transmitting TCP will (eventually) time out and retransmit the missing segment. However, the sending TCP has no information about segments that may have reached the receiver and been queued because they were not at the left window edge, so it may be forced to retransmit these segments unnecessarily.
We propose a TCP extension to implement selective acknowledgements. By sending selective acknowledgments, the receiver of data can inform the sender about all segments that have arrived successfully, so the sender need retransmit only the segments that have actually been lost.
Selective acknowledgments have been included in a number of experimental Internet protocols -- VMTP [Cheriton88], NETBLT [Clark87], and RDP [Velten84]. There is some empirical evidence in favor of selective acknowledgments -- simple experiments with RDP have shown that disabling the selective acknowlegment facility greatly increases the number of retransmitted segments over a lossy, high-delay Internet path [Partridge87]. A simulation study of a simple form of selective acknowledgments added to the ISO transport protocol TP4 also showed promise of performance improvement [NBS85].
TCP implements reliable data delivery by measuring the RTT, i.e., the time interval between sending a segment and receiving an acknowledgment for it, and retransmitting any segments that are not acknowledged within some small multiple of the average RTT. Experience has shown that accurate, current RTT estimates are necessary to adapt to changing traffic conditions and, without them, a busy network is subject to an instability known as "congestion collapse" [Nagle84].
In part because TCP segments may be repacketized upon retransmission, and in part because of complications due to the cumulative TCP acknowledgement, measuring a segments's RTT may involve a non-trivial amount of computation in some implementations. To minimize this computation, some implementations time only one segment per window. While this yields an adequate approximation to the RTT for small windows (e.g., a 4 to 8 segment Arpanet window), for an LFN (e.g., 100 segment Wideband Network windows) it results in an unacceptably poor RTT estimate.
In the presence of errors, the problem becomes worse. Zhang [Zhang86], Jain [Jain86] and Karn [Karn87] have shown that it is not possible to accumulate reliable RTT estimates if retransmitted segments are included in the estimate. Since a full window of data will have been transmitted prior to a retransmission, all of the segments in that window will have to be ACKed before the next RTT sample can be taken. This means at least an additional window's worth of time between RTT measurements and, as the error rate approaches one per window of data (e.g., 10**-6 errors per bit for the Wideband Net), it becomes effectively impossible to obtain an RTT measurement.
We propose a TCP "echo" option that allows each segment to carry its own timestamp. This will allow every segment, including retransmissions, to be timed at negligible computational cost.
In designing new TCP options, we must pay careful attention to interoperability with existing implementations. The only TCP option defined to date is an "initial option", i.e., it may appear only on a SYN segment. It is likely that most implementations will properly ignore any options in the SYN segment that they do not understand, so new initial options should not cause a problem. On the other hand, we fear that receiving unexpected non-initial options may cause some TCP's to crash.
Therefore, in each of the extensions we propose, non-initial options may be sent only if an exchange of initial options has indicated that both sides understand the extension. This approach will also allow a TCP to determine when the connection opens how big a TCP header it will be sending.