📄 rfc1185.txt
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Network Working Group V. JacobsonRequest for Comments: 1185 LBL R. Braden ISI L. Zhang PARC October 1990 TCP Extension for High-Speed PathsStatus of This Memo This memo describes an Experimental Protocol extension to TCP for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "IAB Official Protocol Standards" for the standardization state and status of this protocol. Distribution of this memo is unlimited.Summary This memo describes a small extension to TCP to support reliable operation over very high-speed paths, using sender timestamps transmitted using the TCP Echo option proposed in RFC-1072.1. INTRODUCTION TCP uses positive acknowledgments and retransmissions to provide reliable end-to-end delivery over a full-duplex virtual circuit called a connection [Postel81]. A connection is defined by its two end points; each end point is a "socket", i.e., a (host,port) pair. To protect against data corruption, TCP uses an end-to-end checksum. Duplication and reordering are handled using a fine-grained sequence number space, with each octet receiving a distinct sequence number. The TCP protocol [Postel81] was designed to operate reliably over almost any transmission medium regardless of transmission rate, delay, corruption, duplication, or reordering of segments. In practice, proper TCP implementations have demonstrated remarkable robustness in adapting to a wide range of network characteristics. For example, TCP implementations currently adapt to transfer rates in the range of 100 bps to 10**7 bps and round-trip delays in the range 1 ms to 100 seconds. However, the introduction of fiber optics is resulting in ever-higher transmission speeds, and the fastest paths are moving out of the domain for which TCP was originally engineered. This memo and RFC- 1072 [Jacobson88] propose modest extensions to TCP to extend theJacobson, Braden & Zhang [Page 1]RFC 1185 TCP over High-Speed Paths October 1990 domain of its application to higher speeds. There is no one-line answer to the question: "How fast can TCP go?". The issues are reliability and performance, and these depend upon the round-trip delay and the maximum time that segments may be queued in the Internet, as well as upon the transmission speed. We must think through these relationships very carefully if we are to successfully extend TCP's domain. TCP performance depends not upon the transfer rate itself, but rather upon the product of the transfer rate and the round-trip delay. This "bandwidth*delay product" measures the amount of data that would "fill the pipe"; it is the buffer space required at sender and receiver to obtain maximum throughput on the TCP connection over the path. RFC-1072 proposed a set of TCP extensions to improve TCP efficiency for "LFNs" (long fat networks), i.e., networks with large bandwidth*delay products. On the other hand, high transfer rate can threaten TCP reliability by violating the assumptions behind the TCP mechanism for duplicate detection and sequencing. The present memo specifies a solution for this problem, extending TCP reliability to transfer rates well beyond the foreseeable upper limit of bandwidth. An especially serious kind of error may result from an accidental reuse of TCP sequence numbers in data segments. Suppose that an "old duplicate segment", e.g., a duplicate data segment that was delayed in Internet queues, was delivered to the receiver at the wrong moment so that its sequence numbers fell somewhere within the current window. There would be no checksum failure to warn of the error, and the result could be an undetected corruption of the data. Reception of an old duplicate ACK segment at the transmitter could be only slightly less serious: it is likely to lock up the connection so that no further progress can be made and a RST is required to resynchronize the two ends. Duplication of sequence numbers might happen in either of two ways: (1) Sequence number wrap-around on the current connection A TCP sequence number contains 32 bits. At a high enough transfer rate, the 32-bit sequence space may be "wrapped" (cycled) within the time that a segment may be delayed in queues. Section 2 discusses this case and proposes a mechanism to reject old duplicates on the current connection. (2) Segment from an earlier connection incarnationJacobson, Braden & Zhang [Page 2]RFC 1185 TCP over High-Speed Paths October 1990 Suppose a connection terminates, either by a proper close sequence or due to a host crash, and the same connection (i.e., using the same pair of sockets) is immediately reopened. A delayed segment from the terminated connection could fall within the current window for the new incarnation and be accepted as valid. This case is discussed in Section 3. TCP reliability depends upon the existence of a bound on the lifetime of a segment: the "Maximum Segment Lifetime" or MSL. An MSL is generally required by any reliable transport protocol, since every sequence number field must be finite, and therefore any sequence number may eventually be reused. In the Internet protocol suite, the MSL bound is enforced by an IP-layer mechanism, the "Time-to-Live" or TTL field. Watson's Delta-T protocol [Watson81] includes network-layer mechanisms for precise enforcement of an MSL. In contrast, the IP mechanism for MSL enforcement is loosely defined and even more loosely implemented in the Internet. Therefore, it is unwise to depend upon active enforcement of MSL for TCP connections, and it is unrealistic to imagine setting MSL's smaller than the current values (e.g., 120 seconds specified for TCP). The timestamp algorithm described in the following section gives a way out of this dilemma for high-speed networks.2. SEQUENCE NUMBER WRAP-AROUND 2.1 Background Avoiding reuse of sequence numbers within the same connection is simple in principle: enforce a segment lifetime shorter than the time it takes to cycle the sequence space, whose size is effectively 2**31. More specifically, if the maximum effective bandwidth at which TCP is able to transmit over a particular path is B bytes per second, then the following constraint must be satisfied for error-free operation: 2**31 / B > MSL (secs) [1] The following table shows the value for Twrap = 2**31/B in seconds, for some important values of the bandwidth B:Jacobson, Braden & Zhang [Page 3]RFC 1185 TCP over High-Speed Paths October 1990 Network B*8 B Twrap bits/sec bytes/sec secs _______ _______ ______ ______ ARPANET 56kbps 7KBps 3*10**5 (~3.6 days) DS1 1.5Mbps 190KBps 10**4 (~3 hours) Ethernet 10Mbps 1.25MBps 1700 (~30 mins) DS3 45Mbps 5.6MBps 380 FDDI 100Mbps 12.5MBps 170 Gigabit 1Gbps 125MBps 17 It is clear why wrap-around of the sequence space was not a problem for 56kbps packet switching or even 10Mbps Ethernets. On the other hand, at DS3 and FDDI speeds, Twrap is comparable to the 2 minute MSL assumed by the TCP specification [Postel81]. Moving towards gigabit speeds, Twrap becomes too small for reliable enforcement by the Internet TTL mechanism. The 16-bit window field of TCP limits the effective bandwidth B to 2**16/RTT, where RTT is the round-trip time in seconds [McKenzie89]. If the RTT is large enough, this limits B to a value that meets the constraint [1] for a large MSL value. For example, consider a transcontinental backbone with an RTT of 60ms (set by the laws of physics). With the bandwidth*delay product limited to 64KB by the TCP window size, B is then limited to 1.1MBps, no matter how high the theoretical transfer rate of the path. This corresponds to cycling the sequence number space in Twrap= 2000 secs, which is safe in today's Internet. Based on this reasoning, an earlier RFC [McKenzie89] has cautioned that expanding the TCP window space as proposed in RFC-1072 will lead to sequence wrap-around and hence to possible data corruption. We believe that this is mis-identifying the culprit, which is not the larger window but rather the high bandwidth. For example, consider a (very large) FDDI LAN with a diameter of 10km. Using the speed of light, we can compute the RTT across the ring as (2*10**4)/(3*10**8) = 67 microseconds, and the delay*bandwidth product is then 833 bytes. A TCP connection across this LAN using a window of only 833 bytes will run at the full 100mbps and can wrap the sequence space in about 3 minutes, very close to the MSL of TCP. Thus, highJacobson, Braden & Zhang [Page 4]RFC 1185 TCP over High-Speed Paths October 1990 speed alone can cause a reliability problem with sequence number wrap-around, even without extended windows. An "obvious" fix for the problem of cycling the sequence space is to increase the size of the TCP sequence number field. For example, the sequence number field (and also the acknowledgment field) could be expanded to 64 bits. However, the proposals for making such a change while maintaining compatibility with current TCP have tended towards complexity and ugliness. This memo proposes a simple solution to the problem, using the TCP echo options defined in RFC-1072. Section 2.2 which follows describes the original use of these options to carry timestamps in order to measure RTT accurately. Section 2.3 proposes a method of using these same timestamps to reject old duplicate segments that could corrupt an open TCP connection. Section 3 discusses the application of this mechanism to avoiding old duplicates from previous incarnations. 2.2 TCP Timestamps RFC-1072 defined two TCP options, Echo and Echo Reply. Echo carries a 32-bit number, and the receiver of the option must return this same value to the source host in an Echo Reply option. RFC-1072 furthermore describes the use of these options to contain 32-bit timestamps, for measuring the RTT. A TCP sending data would include Echo options containing the current clock value. The receiver would echo these timestamps in returning segments (generally, ACK segments). The difference between a timestamp from an Echo Reply option and the current time would then measure the RTT at the sender. This mechanism was designed to solve the following problem: almost all TCP implementations base their RTT measurements on a sample of only one packet per window. If we look at RTT estimation as a signal processing problem (which it is), a data signal at some frequency (the packet rate) is being sampled at a lower frequency (the window rate). Unfortunately, this lower sampling frequency violates Nyquist's criteria and may introduce "aliasing" artifacts into the estimated RTT [Hamming77]. A good RTT estimator with a conservative retransmission timeout calculation can tolerate the aliasing when the sampling frequency is "close" to the data frequency. For example, with a window of 8 packets, the sample rate is 1/8 the data frequency -- less than an order of magnitude different. However, when the window is tens or hundreds of packets, the RTT estimator may be seriously inJacobson, Braden & Zhang [Page 5]RFC 1185 TCP over High-Speed Paths October 1990 error, resulting in spurious retransmissions. A solution to the aliasing problem that actually simplifies the sender substantially (since the RTT code is typically the single biggest protocol cost for TCP) is as follows: the will sender place a timestamp in each segment and the receiver will reflect these timestamps back in ACK segments. Then a single subtract gives the sender an accurate RTT measurement for every ACK segment (which will correspond to every other data segment, with a
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