University of Oslo

PO Box 1080 Blindern N-0316 Oslo Norway +47 22 85 24 20 michawe@ifi.uio.no

Independent Submission Teleportation Entanglement 0-RTT The age of quantum networking is upon us, and with it comes "entanglement": a procedure in which a state (i.e., a bit) can be transferred instantly, with no measurable delay between peers. This will lead to a perceived round-trip time of zero seconds on some Internet paths, a capability which was not predicted and so not included as a possibility in many protocol specifications. Worse than the millennium bug, this unexpected value is bound to cause serious Internet failures unless the specifications are fixed in time.

Status of This Memo This document is not an Internet Standards Track specification; it is published for informational purposes. This is a contribution to the RFC Series, independently of any other RFC stream. The RFC Editor has chosen to publish this document at its discretion and makes no statement about its value for implementation or deployment. Documents approved for publication by the RFC Editor are not candidates for any level of Internet Standard; see Section 2 of RFC 7841. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at .

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Table of Contents

• .  Introduction
• .  Protocols and Protocol Mechanisms That Will Fail
  • .  LEDBAT
  • .  Multipath TCP (MPTCP)
  • .  RTP Circuit Breakers
• .  What can be done?
• .  Conclusion
• .  IANA Considerations
• .  Security Considerations
• .  References
  • .  Normative References
  • .  Informative References
• Author's Address

Introduction discusses faster-than-light communication, where packets arrive before they are sent. While it is amusing to entertain the possibility of time travel, we have to accept the cold facts: time travel will never work (or it would already have been used). Quantum networking, however, is an entirely different matter -- commercial products are already available, and quantum networks will without a doubt become the prevalent Internet link-layer technology across the globe within the next five to ten years. With the help of entanglement, implemented in quantum repeaters, quantum networks can transfer information faster than ever before: a state can be transmitted over a long distance instantly, with no delay. This is so cool that it is also called (and, by some, mistaken for) teleportation. If a path between a sender and a receiver is fully quantum-ized, the measured one-way delay (OWD) will be zero. What's more, assuming that there are blazing fast quantum computers involved on both ends, the processing time will be well below anything measurable; hence, even the round-trip time (RTT) will be zero in these scenarios. In today's Internet, only very few protocols are prepared for such "0-RTT" situations (e.g., TCP with "TCP Fast Open" (TFO) , TLS 1.3 , and QUIC ). Many others will fail in interesting ways; we coin the term "Quantum Bug" for such failures. In the following section, we will discuss some examples of Quantum Bugs.

Protocols and Protocol Mechanisms That Will Fail The number of protocols and protocol mechanisms that will fail in the face of a zero RTT is too large to report here; we are truly heading towards something close to an Internet meltdown. We can only provide some guidance to those who hunt for the Quantum Bug, by discussing examples of specification mistakes that will need to be fixed.

LEDBAT The Low Extra Delay Background Transfer (LEDBAT) congestion control mechanism is a very interesting failure case: designed to "get out of the way" of other traffic; it will end up sending as fast as possible. Specifically, when the algorithm described in obtains a delay sample, it updates a list of base delays that will all become 0 and current delays that will also all become 0. It calculates a queuing delay as the difference between the current delay and the base delay (resulting in 0) and keeps increasing the Congestion Window (cwnd) until the queuing delay reaches a predefined parameter value TARGET (100 milliseconds or less). A TARGET value of 100 milliseconds will never be reached, because the queuing delay does not grow when the sender increases its cwnd; this means that LEDBAT would endlessly increase its cwnd, limited only by the number of bits that are used to represent cwnd. However, given that TARGET=0 is also allowed, this parameter choice may seem to be a way out. Always staying at the target means that the sender would maintain its initial cwnd, which should be set to 2. This may seem like a small number, but remember that cwnd is the number of bytes that can be transmitted per RTT (which is 0). Thus, irrespective of the TARGET value, the sender will send data as fast as it can.

Multipath TCP (MPTCP) The coupled congestion control mechanism proposed for MPTCP in requires calculating a value called "alpha". Equation 2 in contains a term where a value called "cwnd_i" is divided by the square of the RTT, and another term where this value is divided by the RTT. Enough said.

RTP Circuit Breakers The RTP Circuit Breakers require calculation of a well-known equation which yields the throughput of a TCP connection: s X = ------------------------------------------------------------- Tr*sqrt(2*b*p/3)+(t_RTO * (3*sqrt(3*b*p/8) * p * (1+32*p*p))) where Tr is the RTT and t_RTO is the retransmission timeout of TCP (we don't need to care about the other variables). As we will discuss in , t_RTO is lower-bounded with 1 second; therefore, it saves us from a division by zero. However, there is also a simplified version of this equation: s X = ---------------- Tr*sqrt(2*b*p/3) Unfortunately, states: "It is RECOMMENDED that this simplified throughput equation be used since the reduction in accuracy is small, and it is much simpler to calculate than the full equation." Due to this simplification, many multimedia applications will crash.

What can be done? Fear not: when everything else fails, TCP will still work. Its retransmission timeout is lower-bounded by 1 second . Moreover, while its cwnd may grow up to the maximum storable number, data transmission is limited by the Receiver Window (rwnd). This means that flow control will save TCP from failing. From this, we can learn two simple rules: lower-bound any values calculated from the RTT (and, obviously, do not divide by the RTT), and use flow control. Specifications will need to be updated by fixing all RTT-based calculations and introducing flow control everywhere. For example, UDP will have to be extended with a receiver window, e.g., as a UDP option .

Conclusion We are in trouble, and there is only one way out: develop a comprehensive list of all RFCs containing "0-RTT" mistakes (taking as a guideline), and update all code. This needs to happen fast, the clock is ticking. Luckily, if we are too slow, we will still be able to use TCP to access the specifications. With DNS over TCP , name resolution to find the server containing the specifications should also work.

IANA Considerations This document has no IANA actions.

Security Considerations Flow control must be used on 0-RTT paths, or else an attacker can completely overwhelm a sender with data in a denial-of-service (DoS) attack within an instant. Flow control will need to be added to protocols that do not currently have it, such as UDP or ICMP. IPv6 will not save us.

References Normative References The Year 2000 Working Group (WG) has conducted an investigation into the millennium problem as it regards Internet related protocols. This investigation only targeted the protocols as documented in the Request For Comments Series (RFCs). This investigation discovered little reason for concern with regards to the functionality of the protocols. A few minor cases of older implementations still using two digit years (ala RFC 850) were discovered, but almost all Internet protocols were given a clean bill of health. Several cases of "period" problems were discovered, where a time field would "roll over" as the size of field was reached. In particular, there are several protocols, which have 32 bit, signed integer representations of the number of seconds since January 1, 1970 which will turn negative at Tue Jan 19 03:14:07 GMT 2038. Areas whose protocols will be effected by such problems have been notified so that new revisions will remove this limitation. This memo provides information for the Internet community. We are approaching the time when we will be able to communicate faster than the speed of light. It is well known that as we approach the speed of light, time slows down. Logically, it is reasonable to assume that as we go faster than the speed of light, time will reverse. The major consequence of this for Internet protocols is that packets will arrive before they are sent. This will have a major impact on the way we design Internet protocols. This paper outlines some of the issues and suggests some directions for additional analysis of these issues. Informative References This document defines the core of the QUIC transport protocol. Accompanying documents describe QUIC's loss detection and congestion control and the use of TLS for key negotiation. Note to Readers Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at <https://mailarchive.ietf.org/arch/search/?email_list=quic>. Working Group information can be found at <https://github.com/ quicwg>; source code and issues list for this draft can be found at <https://github.com/quicwg/base-drafts/labels/-transport>. Work in Progress This document defines the standard algorithm that Transmission Control Protocol (TCP) senders are required to use to compute and manage their retransmission timer. It expands on the discussion in Section 4.2.3.1 of RFC 1122 and upgrades the requirement of supporting the algorithm from a SHOULD to a MUST. This document obsoletes RFC 2988. [STANDARDS-TRACK] Often endpoints are connected by multiple paths, but communications are usually restricted to a single path per connection. Resource usage within the network would be more efficient were it possible for these multiple paths to be used concurrently. Multipath TCP is a proposal to achieve multipath transport in TCP. New congestion control algorithms are needed for multipath transport protocols such as Multipath TCP, as single path algorithms have a series of issues in the multipath context. One of the prominent problems is that running existing algorithms such as standard TCP independently on each path would give the multipath flow more than its fair share at a bottleneck link traversed by more than one of its subflows. Further, it is desirable that a source with multiple paths available will transfer more traffic using the least congested of the paths, achieving a property called "resource pooling" where a bundle of links effectively behaves like one shared link with bigger capacity. This would increase the overall efficiency of the network and also its robustness to failure. This document presents a congestion control algorithm that couples the congestion control algorithms running on different subflows by linking their increase functions, and dynamically controls the overall aggressiveness of the multipath flow. The result is a practical algorithm that is fair to TCP at bottlenecks while moving traffic away from congested links. This document defines an Experimental Protocol for the Internet community. Low Extra Delay Background Transport (LEDBAT) is an experimental delay-based congestion control algorithm that seeks to utilize the available bandwidth on an end-to-end path while limiting the consequent increase in queueing delay on that path. LEDBAT uses changes in one-way delay measurements to limit congestion that the flow itself induces in the network. LEDBAT is designed for use by background bulk-transfer applications to be no more aggressive than standard TCP congestion control (as specified in RFC 5681) and to yield in the presence of competing flows, thus limiting interference with the network performance of competing flows. This document defines an Experimental Protocol for the Internet community. This document describes an experimental TCP mechanism called TCP Fast Open (TFO). TFO allows data to be carried in the SYN and SYN-ACK packets and consumed by the receiving end during the initial connection handshake, and saves up to one full round-trip time (RTT) compared to the standard TCP, which requires a three-way handshake (3WHS) to complete before data can be exchanged. However, TFO deviates from the standard TCP semantics, since the data in the SYN could be replayed to an application in some rare circumstances. Applications should not use TFO unless they can tolerate this issue, as detailed in the Applicability section. This document specifies the requirement for support of TCP as a transport protocol for DNS implementations and provides guidelines towards DNS-over-TCP performance on par with that of DNS-over-UDP. This document obsoletes RFC 5966 and therefore updates RFC 1035 and RFC 1123. The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows. This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms. This document specifies version 1.3 of the Transport Layer Security (TLS) protocol. TLS allows client/server applications to communicate over the Internet in a way that is designed to prevent eavesdropping, tampering, and message forgery. This document updates RFCs 5705 and 6066, and obsoletes RFCs 5077, 5246, and 6961. This document also specifies new requirements for TLS 1.2 implementations. Transport protocols are extended through the use of transport header options. This document extends UDP by indicating the location, syntax, and semantics for UDP transport layer options. Work in Progress

Author's Address University of Oslo

PO Box 1080 Blindern N-0316 Oslo Norway +47 22 85 24 20 michawe@ifi.uio.no