--- 1/draft-ietf-taps-transports-06.txt 2015-10-07 04:15:33.365660474 -0700 +++ 2/draft-ietf-taps-transports-07.txt 2015-10-07 04:15:33.493663565 -0700 @@ -1,21 +1,21 @@ Network Working Group G. Fairhurst, Ed. Internet-Draft University of Aberdeen Intended status: Informational B. Trammell, Ed. -Expires: January 7, 2016 M. Kuehlewind, Ed. +Expires: April 9, 2016 M. Kuehlewind, Ed. ETH Zurich - July 06, 2015 + October 07, 2015 Services provided by IETF transport protocols and congestion control mechanisms - draft-ietf-taps-transports-06 + draft-ietf-taps-transports-07 Abstract This document describes services provided by existing IETF protocols and congestion control mechanisms. It is designed to help application and network stack programmers and to inform the work of the IETF TAPS Working Group. Status of This Memo @@ -25,140 +25,155 @@ Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on December 14, 2015. + This Internet-Draft will expire on April 9, 2016. Copyright Notice Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 - 3. Existing Transport Protocols . . . . . . . . . . . . . . . . 4 - 3.1. Transport Control Protocol (TCP) . . . . . . . . . . . . 4 + 3. Existing Transport Protocols . . . . . . . . . . . . . . . . 5 + 3.1. Transport Control Protocol (TCP) . . . . . . . . . . . . 5 3.1.1. Protocol Description . . . . . . . . . . . . . . . . 5 3.1.2. Interface description . . . . . . . . . . . . . . . . 6 - 3.1.3. Transport Protocol Components . . . . . . . . . . . . 6 - 3.2. Multipath TCP (MPTCP) . . . . . . . . . . . . . . . . . . 7 + 3.1.3. Transport Features . . . . . . . . . . . . . . . . . 7 + 3.2. Multipath TCP (MPTCP) . . . . . . . . . . . . . . . . . . 8 3.2.1. Protocol Description . . . . . . . . . . . . . . . . 8 3.2.2. Interface Description . . . . . . . . . . . . . . . . 8 - 3.2.3. Transport Protocol Components . . . . . . . . . . . . 8 + 3.2.3. Transport features . . . . . . . . . . . . . . . . . 8 3.3. Stream Control Transmission Protocol (SCTP) . . . . . . . 9 3.3.1. Protocol Description . . . . . . . . . . . . . . . . 9 - 3.3.2. Interface Description . . . . . . . . . . . . . . . . 11 - 3.3.3. Transport Protocol Components . . . . . . . . . . . . 13 - 3.4. User Datagram Protocol (UDP) . . . . . . . . . . . . . . 13 - 3.4.1. Protocol Description . . . . . . . . . . . . . . . . 14 - 3.4.2. Interface Description . . . . . . . . . . . . . . . . 14 - 3.4.3. Transport Protocol Components . . . . . . . . . . . . 15 - 3.5. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 15 - 3.5.1. Protocol Description . . . . . . . . . . . . . . . . 15 - 3.5.2. Interface Description . . . . . . . . . . . . . . . . 16 - 3.5.3. Transport Protocol Components . . . . . . . . . . . . 16 - 3.6. Datagram Congestion Control Protocol (DCCP) . . . . . . . 17 - 3.6.1. Protocol Description . . . . . . . . . . . . . . . . 17 - 3.6.2. Interface Description . . . . . . . . . . . . . . . . 19 - 3.6.3. Transport Protocol Components . . . . . . . . . . . . 19 - 3.7. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 19 - 3.8. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . . 20 - 3.8.1. Protocol Description . . . . . . . . . . . . . . . . 20 - 3.8.2. Interface Description . . . . . . . . . . . . . . . . 21 - 3.8.3. Transport Protocol Components . . . . . . . . . . . . 21 - 3.9. Transport Layer Security (TLS) and Datagram TLS (DTLS) as - a pseudotransport . . . . . . . . . . . . . . . . . . . . 22 - 3.9.1. Protocol Description . . . . . . . . . . . . . . . . 23 - 3.9.2. Interface Description . . . . . . . . . . . . . . . . 24 - 3.9.3. Transport Protocol Components . . . . . . . . . . . . 24 - 3.10. Hypertext Transport Protocol (HTTP) over TCP as a - pseudotransport . . . . . . . . . . . . . . . . . . . . . 25 - 3.10.1. Protocol Description . . . . . . . . . . . . . . . . 25 - 3.10.2. Interface Description . . . . . . . . . . . . . . . 26 - 3.10.3. Transport Protocol Components . . . . . . . . . . . 27 - 3.11. WebSockets . . . . . . . . . . . . . . . . . . . . . . . 27 - 3.11.1. Protocol Description . . . . . . . . . . . . . . . . 27 - 3.11.2. Interface Description . . . . . . . . . . . . . . . 27 - 3.11.3. Transport Protocol Components . . . . . . . . . . . 28 - 4. Transport Service Features . . . . . . . . . . . . . . . . . 28 - 4.1. Complete Protocol Feature Matrix . . . . . . . . . . . . 30 - 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31 - 6. Security Considerations . . . . . . . . . . . . . . . . . . . 31 - 7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 32 - 8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 32 - 9. References . . . . . . . . . . . . . . . . . . . . . . . . . 32 - 9.1. Normative References . . . . . . . . . . . . . . . . . . 33 - 9.2. Informative References . . . . . . . . . . . . . . . . . 33 - Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 39 + 3.3.2. Interface Description . . . . . . . . . . . . . . . . 12 + 3.3.3. Transport Features . . . . . . . . . . . . . . . . . 14 + 3.4. User Datagram Protocol (UDP) . . . . . . . . . . . . . . 15 + 3.4.1. Protocol Description . . . . . . . . . . . . . . . . 15 + 3.4.2. Interface Description . . . . . . . . . . . . . . . . 16 + 3.4.3. Transport Features . . . . . . . . . . . . . . . . . 16 + 3.5. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 17 + 3.5.1. Protocol Description . . . . . . . . . . . . . . . . 17 + 3.5.2. Interface Description . . . . . . . . . . . . . . . . 18 + 3.5.3. Transport Features . . . . . . . . . . . . . . . . . 18 + 3.6. Datagram Congestion Control Protocol (DCCP) . . . . . . . 19 + 3.6.1. Protocol Description . . . . . . . . . . . . . . . . 19 + 3.6.2. Interface Description . . . . . . . . . . . . . . . . 20 + 3.6.3. Transport Features . . . . . . . . . . . . . . . . . 21 + 3.7. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 21 + 3.7.1. Protocol Description . . . . . . . . . . . . . . . . 21 + 3.7.2. Interface Description . . . . . . . . . . . . . . . . 22 + 3.7.3. Transport Features . . . . . . . . . . . . . . . . . 22 + 3.8. Internet Control Message Protocol (ICMP) . . . . . . . . 23 + 3.8.1. Protocol Description . . . . . . . . . . . . . . . . 23 + 3.8.2. Interface Description . . . . . . . . . . . . . . . . 24 + 3.8.3. Transport Features . . . . . . . . . . . . . . . . . 24 + 3.9. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 25 + 3.9.1. Protocol Description . . . . . . . . . . . . . . . . 25 + 3.9.2. Interface Description . . . . . . . . . . . . . . . . 26 + 3.9.3. Transport Features . . . . . . . . . . . . . . . . . 26 + 3.10. File Delivery over Unidirectional Transport/Asynchronous + Layered Coding Reliable Multicast (FLUTE/ALC) . . . . . . 26 + 3.10.1. Protocol Description . . . . . . . . . . . . . . . . 27 + 3.10.2. Interface Description . . . . . . . . . . . . . . . 29 + 3.10.3. Transport Features . . . . . . . . . . . . . . . . . 29 + 3.11. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . . 30 + 3.11.1. Protocol Description . . . . . . . . . . . . . . . . 30 + 3.11.2. Interface Description . . . . . . . . . . . . . . . 31 + 3.11.3. Transport Features . . . . . . . . . . . . . . . . . 32 + 3.12. Transport Layer Security (TLS) and Datagram TLS (DTLS) as + a pseudotransport . . . . . . . . . . . . . . . . . . . . 32 + 3.12.1. Protocol Description . . . . . . . . . . . . . . . . 33 + 3.12.2. Interface Description . . . . . . . . . . . . . . . 34 + 3.12.3. Transport Features . . . . . . . . . . . . . . . . . 34 + 3.13. Hypertext Transport Protocol (HTTP) over TCP as a + pseudotransport . . . . . . . . . . . . . . . . . . . . . 35 + 3.13.1. Protocol Description . . . . . . . . . . . . . . . . 36 + 3.13.2. Interface Description . . . . . . . . . . . . . . . 37 + 3.13.3. Transport features . . . . . . . . . . . . . . . . . 37 + 4. Transport Service Features . . . . . . . . . . . . . . . . . 38 + 4.1. Complete Protocol Feature Matrix . . . . . . . . . . . . 40 + 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 42 + 6. Security Considerations . . . . . . . . . . . . . . . . . . . 42 + 7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 42 + 8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 43 + 9. Informative References . . . . . . . . . . . . . . . . . . . 43 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 52 1. Introduction Most Internet applications make use of the Transport Services provided by TCP (a reliable, in-order stream protocol) or UDP (an unreliable datagram protocol). We use the term "Transport Service" to mean the end-to-end service provided to an application by the transport layer. That service can only be provided correctly if information about the intended usage is supplied from the application. The application may determine this information at design time, compile time, or run time, and may include guidance on whether a feature is required, a preference by the application, or something in between. Examples of features of Transport Services are reliable delivery, ordered delivery, content privacy to in-path - devices, integrity protection, and minimal latency. + devices, and integrity protection. The IETF has defined a wide variety of transport protocols beyond TCP and UDP, including SCTP, DCCP, MP-TCP, and UDP-Lite. Transport services may be provided directly by these transport protocols, or layered on top of them using protocols such as WebSockets (which runs over TCP), RTP (over TCP or UDP) or WebRTC data channels (which run over SCTP over DTLS over UDP or TCP). Services built on top of UDP or UDP-Lite typically also need to specify additional mechanisms, - including a congestion control mechanism (such as a windowed - congestion control, TFRC or LEDBAT congestion control mechanism). - This extends the set of available Transport Services beyond those - provided to applications by TCP and UDP. + including a congestion control mechanism (such as NewReno, TFRC or + LEDBAT). This extends the set of available Transport Services beyond + those provided to applications by TCP and UDP. + + [GF: Ledbat is a mechanism, not protocol - hence use the work + "support" in para below.] Transport protocols can also be differentiated by the features of the services they provide: for instance, SCTP offers a message-based service providing full or partial reliability and allowing to minimize the head of line blocking due to the support of unordered - and unordered message delivery within multiple streams, UDP-Lite - provides partial integrity protection, and LEDBAT can provide low- - priority "scavenger" communication. + and unordered message delivery within multiple streams, UDP-Lite and + DCCP provide partial integrity protection, and LEDBAT can support + low-priority "scavenger" communication. 2. Terminology The following terms are defined throughout this document, and in subsequent documents produced by TAPS describing the composition and decomposition of transport services. [EDITOR'S NOTE: we may want to add definitions for the different kinds of interfaces that are important here.] + [GF: Interfaces may be covered by Micahel Welzl's companion + document?] + Transport Service Feature: a specific end-to-end feature that a transport service provides to its clients. Examples include confidentiality, reliable delivery, ordered delivery, message- versus-stream orientation, etc. Transport Service: a set of transport service features, without an association to any given framing protocol, which provides a complete service to an application. Transport Protocol: an implementation that provides one or more @@ -175,169 +190,161 @@ Application: an entity that uses the transport layer for end-to-end delivery data across the network (this may also be an upper layer protocol or tunnel encapsulation). 3. Existing Transport Protocols This section provides a list of known IETF transport protocol and transport protocol frameworks. - [EDITOR'S NOTE: Contributions to the subsections below are welcome] - 3.1. Transport Control Protocol (TCP) TCP is an IETF standards track transport protocol. [RFC0793] introduces TCP as follows: "The Transmission Control Protocol (TCP) is intended for use as a highly reliable host-to-host protocol between hosts in packet-switched computer communication networks, and in interconnected systems of such networks." Since its introduction, TCP has become the default connection-oriented, stream-based transport protocol in the Internet. It is widely implemented by endpoints and widely used by common applications. 3.1.1. Protocol Description TCP is a connection-oriented protocol, providing a three way - handshake to allow a client and server to set up a connection, and - mechanisms for orderly completion and immediate teardown of a - connection. TCP is defined by a family of RFCs [RFC4614]. + handshake to allow a client and server to set up a connection and + negotiate features, and mechanisms for orderly completion and + immediate teardown of a connection. TCP is defined by a family of + RFCs [RFC4614]. TCP provides multiplexing to multiple sockets on each host using port - numbers. An active TCP session is identified by its four-tuple of + numbers.] A similar approach is adopted by other IETF-defined + transports. An active TCP session is identified by its four-tuple of local and remote IP addresses and local port and remote port numbers. - The destination port during connection setup has a different role as - it is often used to indicate the requested service. + The destination port during connection setup is often used to + indicate the requested service. TCP partitions a continuous stream of bytes into segments, sized to fit in IP packets. ICMP-based PathMTU discovery [RFC1191][RFC1981] as well as Packetization Layer Path MTU Discovery (PMTUD) [RFC4821] are supported. Each byte in the stream is identified by a sequence number. The sequence number is used to order segments on receipt, to identify segments in acknowledgments, and to detect unacknowledged segments - for retransmission. This is the basis of TCP's reliable, ordered - delivery of data in a stream. TCP Selective Acknowledgment [RFC2018] - extends this mechanism by making it possible to identify missing - segments more precisely, reducing spurious retransmission. + for retransmission. This is the basis of the reliable, ordered + delivery of data in a TCP stream. TCP Selective Acknowledgment + [RFC2018] extends this mechanism by making it possible to identify + missing segments more precisely, reducing spurious retransmission. Receiver flow control is provided by a sliding window: limiting the amount of unacknowledged data that can be outstanding at a given time. The window scale option [RFC7323] allows a receiver to use windows greater than 64KB. - All TCP senders provide Congestion Control: This uses a separate - window, where each time congestion is detected, this congestion - window is reduced. A receiver detects congestion using one of three - mechanisms: A retransmission timer, detection of loss (interpreted as - a congestion signal), or Explicit Congestion Notification (ECN) - [RFC3168] to provide early signaling (see - [I-D.ietf-aqm-ecn-benefits]) + All TCP senders provide Congestion Control [RFC5681]: This uses a + separate window, where each time congestion is detected, this + congestion window is reduced. Most of the used congestion control + mechanisms use one of three mechanisms to detect congestion: A + retransmission timer (with exponential back-up), detection of loss + (interpreted as a congestion signal), or Explicit Congestion + Notification (ECN) [RFC3168] to provide early signaling (see + [I-D.ietf-aqm-ecn-benefits]). In addition, a congestion control + mechanism may react to changes in delay as an early indication for + congestion. A TCP protocol instance can be extended [RFC4614] and tuned. Some features are sender-side only, requiring no negotiation with the receiver; some are receiver-side only, some are explicitly negotiated during connection setup. By default, TCP segment partitioning uses Nagle's algorithm [RFC0896] to buffer data at the sender into large segments, potentially incurring sender-side buffering delay; this algorithm can be disabled - by the sender to transmit more immediately, e.g. to enable smoother - interactive sessions. + by the sender to transmit more immediately, e.g., to reduce latency + for interactive sessions. - [EDITOR'S NOTE: add URGENT and PUSH flag (note [RFC6093] says SHOULD - NOT use due to the range of TCP implementations that process TCP - urgent indications differently.) ] + TCP provides a push and a urgent function to enable data to be + directly accessed by the receiver wihout having to wait for in-order + delivery of the data. However, [RFC6093] does not recommend the use + of the urgent flag due to the range of TCP implementations that + process TCP urgent indications differently. A checksum provides an Integrity Check and is mandatory across the - entire packet. The TCP checksum does not support partial corruption - protection as in DCCP/UDP-Lite). This check protects from - misdelivery of data corrupted data, but is relatively weak, and - applications that require end to end integrity of data are - recommended to include a stronger integrity check of their payload - data. + entire packet. This check protects from delivery of corrupted data + and miselivery of packets to the wrong endpoint. This check is + relatively weak, applications that require end to end integrity of + data are recommended to include a stronger integrity check of their + payload data. The TCP checksum does not support partial corruption + protection (as in DCCP/UDP-Lite). - A TCP service is unicast. + TCP only supports unicast connections. 3.1.2. Interface description A User/TCP Interface is defined in [RFC0793] providing six user commands: Open, Send, Receive, Close, Status. This interface does not describe configuration of TCP options or parameters beside use of the PUSH and URGENT flags. + [RFC1122] describes extensions of the TCP/application layer interface + for 1) reporting soft errors such as reception fo ICMP error + messages, extensive retransmission or urgent pointer advance, 2) + providing a possibility to specify the Type-of-Service (TOS) for + segments, 3) providing a fush call to empty the TCP send queue, and + 4) multihoming support. + In API implementations derived from the BSD Sockets API, TCP sockets - are created using the "SOCK_STREAM" socket type. + are created using the "SOCK_STREAM" socket type as described in the + IEEE Portable Operating System Interface (POSIX) Base Specifications + [POSIX]. The features used by a protocol instance may be set and + tuned via this API. However, there is no RFC that documents this + interface. - The features used by a protocol instance may be set and tuned via - this API. +3.1.3. Transport Features - (more on the API goes here) + The transport features provided by TCP are: -3.1.3. Transport Protocol Components + [EDITOR'S NOTE: expand each of these slightly] - The transport protocol components provided by TCP (new version) are: + o unicast transport - [EDITOR'S NOTE: discussion of how to map this to features and TAPS: - what does the higher layer need to decide? what can the transport - layer decide based on global settings? what must the transport layer - decide based on network characteristics?] + o connection setup with feature negotiation and application-to-port + mapping, implemented using SYN segments and the TCP option field + to negotiate features. - o Connection-oriented bidirectional communication using three-way - handshake connection setup with feature negotiation and an - explicit distinction between passive and active open: This implies - both unicast addressing and a guarantee of return routability. + o port multiplexing: each TCP session is uniquely identified by a + combination of the ports and IP address fields. - o Single stream-oriented transmission: The stream abstraction atop - the datagram service provided by IP is implemented by dividing the - stream into segments. + o Uni-or bidirectional communication - o Limited control over segment transmission scheduling (Nagle's - algorithm): This allows for delay minimization in interactive - applications by preventing the transport to add additional delays - (by deactivating Nagle's algorithm). + o stream-oriented delivery in a single stream - o Port multiplexing, with application-to-port mapping during - connection setup: Note that in the presence of network address and - port translation (NAPT), TCP ports are in effect part of the - endpoint address for forwarding purposes. + o fully reliable delivery, implemented using ACKs sent from the + receiver to confirm delivery. - o Full reliability using (S)ACK- and RTO-based loss detection and - retransmissions: Loss is sensed using duplicated ACKs ("fast - retransmit"), which places a lower bound on the delay inherent in - this approach to reliability. The retransmission timeout - determines the upper bound on the delay (expect if also - exponential back-off is performed). The use of selective - acknowlegdements further reduces the latency for retransmissions - if multiple packets are lost during one congestion event. + o error detection: a segment checksum verifies delivery to the + correct endpoint and integrity of the data and options. - o Error detection based on a checksum covering the network and - transport headers as well as payload: Packets that are detected as - corrupted are dropped, relying on the reliability mechanism to - retransmit them. + o segmentation: packets are fragmented to a negotiated maximum + segment size, further constrained by the effective MTU from PMTUD. - o Window-based flow control, with receiver-side window management - and signaling of available window: Scaling the flow control window - beyond 64kB requires the use of an optional feature, which has - performance implications in environments where this option is not - supported; this can be the case either if the receiver does not - implement window scaling or if a network node on the path strips - the window scaling option. + o data bundling, an optional mechanism that uses Nagle's algorithm + to coalesce data sent within the same RTT into full-sized + segments. - o Window-based congestion control reacting to loss, delay, - retransmission timeout, or an explicit congestion signal (ECN): - Most commonly used is a loss signal from the reliability - component's retransmission mechanism. TCP reacts to a congestion - signal by reducing the size of the congestion window; - retransmission timeout is generally handled with a larger reaction - than other signals. + o flow control using a window-based mechanism, where the receiver + advertises the window that it is willing to buffer. + + o congestion control: a window-based method that uses AIMD to + control the sending rate and to conservatively choose a rate after + congestion is detected. 3.2. Multipath TCP (MPTCP) Multipath TCP [RFC6824] is an extension for TCP to support multi- homing. It is designed to be as transparent as possible to middle- boxes. It does so by establishing regular TCP flows between a pair of source/destination endpoints, and multiplexing the application's stream over these flows. 3.2.1. Protocol Description @@ -346,455 +353,525 @@ signal multipath capabilities, as well as to negotiate data sequence numbers, and advertise other available IP addresses and establish new sessions between pairs of endpoints. 3.2.2. Interface Description By default, MPTCP exposes the same interface as TCP to the application. [RFC6897] however describes a richer API for MPTCP- aware applications. - This Basic API describes how an application can - enable or disable - MPTCP; - bind a socket to one or more selected local endpoints; - - query local and remote endpoint addresses; - get a unique connection - identifier (similar to an address-port pair for TCP). + This Basic API describes how an application can - The document also recommend the use of extensions defined for SCTP - [RFC6458] (see next section) to deal with multihoming. + o enable or disable MPTCP; - [AUTHOR'S NOTE: research work, and some implementation, also suggest - that the scheduling algorithm, as well as the path manager, are - configurable options that should be exposed to higher layer. Should - this be discussed here?] + o bind a socket to one or more selected local endpoints; -3.2.3. Transport Protocol Components + o query local and remote endpoint addresses; - [AUTHOR'S NOTE: shouldn't it be "service feature"?] + o get a unique connection identifier (similar to an address-port + pair for TCP). - As an extension to TCP, MPTCP provides mostly the same components. - By establishing multiple sessions between available endpoints, it can + The document also recommends the use of extensions defined for SCTP + [RFC6458] (see next section) to support multihoming. + +3.2.3. Transport features + + As an extension to TCP, MPTCP provides mostly the same features. By + establishing multiple sessions between available endpoints, it can additionally provide soft failover solutions should one of the paths become unusable. In addition, by multiplexing one byte stream over separate paths, it can achieve a higher throughput than TCP in certain situations (note however that coupled congestion control [RFC6356] might limit this benefit to maintain fairness to other flows at the bottleneck). When aggregating capacity over multiple paths, and depending on the way packets are scheduled on each TCP subflow, an additional delay and higher jitter might be observed observed before in-order delivery of data to the applications. - The transport protocol components provided by MPTCP in addition to - TCP therefore are: + The transport features provided by MPTCP in addition to TCP therefore + are: - o congestion control with load balancing over mutiple connections + o congestion control with load balancing over mutiple connections. - o endpoint multiplexing of a single byte stream (higher throughput) + o endpoint multiplexing of a single byte stream (higher throughput). - o resilience to network failure and/or handoverss + o address family multiplexing: sub-flows can be started over IPv4 or + IPv6 for the same session. + + o resilience to network failure and/or handover. [AUTHOR'S NOTE: it is unclear whether MPTCP has to provide data - bundling.] [AUTHOR'S NOTE: AF muliplexing? sub-flows can be started - over IPv4 or IPv6 for the same session] + bundling.] 3.3. Stream Control Transmission Protocol (SCTP) - SCTP is a message oriented standards track transport protocol and the + SCTP is a message-oriented standards track transport protocol. The base protocol is specified in [RFC4960]. It supports multi-homing to - handle path failures. An SCTP association has multiple - unidirectional streams in each direction and provides in-sequence - delivery of user messages only within each stream. This allows to - minimize head of line blocking. SCTP is extensible and the currently - defined extensions include mechanisms for dynamic re-configurations - of streams [RFC6525] and IP-addresses [RFC5061]. Furthermore, the - extension specified in [RFC3758] introduces the concept of partial - reliability for user messages. + handle path failures. It also optionally supports path failover to + provide resilliance to path failures. An SCTP association has + multiple unidirectional streams in each direction and provides in- + sequence delivery of user messages only within each stream. This + allows it to minimize head of line blocking. SCTP is extensible and + the currently defined extensions include mechanisms for dynamic re- + configurations of streams [RFC6525] and IP-addresses [RFC5061]. + Furthermore, the extension specified in [RFC3758] introduces the + concept of partial reliability for user messages. SCTP was originally developed for transporting telephony signalling messages and is deployed in telephony signalling networks, especially - in mobile telephony networks. Additionally, it is used in the WebRTC - framework for data channels and is therefore deployed in all WEB- - browsers supporting WebRTC. + in mobile telephony networks. It can also be used for other + services, for example in the WebRTC framework for data channels and + is therefore deployed in all WEB-browsers supporting WebRTC. 3.3.1. Protocol Description - SCTP is a connection oriented protocol using a four way handshake to + SCTP is a connection-oriented protocol using a four way handshake to establish an SCTP association and a three way message exchange to gracefully shut it down. It uses the same port number concept as - DCCP, TCP, UDP, and UDP-Lite do and only supports unicast. + DCCP, TCP, UDP, and UDP-Lite, and only supports unicast. SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit - errors. This is stronger than the 16-bit checksums used by TCP or - UDP. However, a partial checksum coverage as provided by DCCP or - UDP-Lite is not supported. + errors and miselivery of packets to the wrong endpoint. This is + stronger than the 16-bit checksums used by TCP or UDP. However, a + partial checksum coverage, as provided by DCCP or UDP-Lite is not + supported. SCTP has been designed with extensibility in mind. Each SCTP packet starts with a single common header containing the port numbers, a verification tag and the CRC32c checksum. This common header is followed by a sequence of chunks. Each chunk consists of a type field, flags, a length field and a value. [RFC4960] defines how a receiver processes chunks with an unknown chunk type. The support of extensions can be negotiated during the SCTP handshake. SCTP provides a message-oriented service. Multiple small user messages can be bundled into a single SCTP packet to improve the efficiency. For example, this bundling may be done by delaying user - messages at the sender side similar to the Nagle algorithm used by - TCP. User messages which would result in IP packets larger than the - MTU will be fragmented at the sender side and reassembled at the - receiver side. There is no protocol limit on the user message size. - ICMP-based path MTU discovery as specified for IPv4 in [RFC1191] and - for IPv6 in [RFC1981] as well as packetization layer path MTU - discovery as specified in [RFC4821] with probe packets using the - padding chunks defined the [RFC4820] are supported. + messages at the sender similar to the Nagle algorithm used by TCP. + User messages which would result in IP packets larger than the MTU + will be fragmented at the sender and reassembled at the receiver. + There is no protocol limit on the user message size. ICMP-based path + MTU discovery as specified for IPv4 in [RFC1191] and for IPv6 in + [RFC1981] as well as packetization layer path MTU discovery as + specified in [RFC4821] with probe packets using the padding chunks + defined the [RFC4820] are supported. [RFC4960] specifies a TCP friendly congestion control to protect the network against overload. SCTP also uses a sliding window flow - control to protect receivers against overflow. + control to protect receivers against overflow. Similar to TCP, SCTP + also supports delaying acknowledgements. [RFC7053] provides a way + for the sender of user messages to request the immediate sending of + the corresponding acknowledgements. Each SCTP association has between 1 and 65536 uni-directional streams in each direction. The number of streams can be different in each direction. Every user-message is sent on a particular stream. User messages can be sent un-ordered or ordered upon request by the upper layer. Un-ordered messages can be delivered as soon as they are - completely received. Only all ordered messages sent on the same - stream are delivered at the receiver in the same order as sent by the - sender. For user messages not requiring fragmentation, this - minimises head of line blocking. The base protocol defined in - [RFC4960] doesn't allow interleaving of user-messages, which results - in sending a large message on one stream can block the sending of - user messages on other streams. [I-D.ietf-tsvwg-sctp-ndata] - overcomes this limitation. Furthermore, [I-D.ietf-tsvwg-sctp-ndata] - specifies multiple algorithms for the sender side selection of which - streams to send data from supporting a variety of scheduling - algorithms including priority based ones. The stream re- - configuration extension defined in [RFC6525] allows to reset streams - during the lifetime of an association and to increase the number of - streams, if the number of streams negotiated in the SCTP handshake is - not sufficient. + completely received. Ordered messages sent on the same stream are + delivered at the receiver in the same order as sent by the sender. + For user messages not requiring fragmentation, this minimises head of + line blocking. - According to [RFC4960], each user message sent is either delivered to - the receiver or, in case of excessive retransmissions, the - association is terminated in a non-graceful way, similar to the TCP - behaviour. In addition to this reliable transfer, the partial - reliability extension defined in [RFC3758] allows the sender to - abandon user messages. The application can specify the policy for - abandoning user messages. Examples for these policies include: + The base protocol defined in [RFC4960] does not allow interleaving of + user-messages, which results in sending a large message on one stream + can block the sending of user messages on other streams. + [I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation. Furthermore, + [I-D.ietf-tsvwg-sctp-ndata] specifies multiple algorithms for the + sender side selection of which streams to send data from supporting a + variety of scheduling algorithms including priority based methods. + The stream re-configuration extension defined in [RFC6525] allows + streams to be reset during the lifetime of an association and to + increase the number of streams, if the number of streams negotiated + in the SCTP handshake becomes insufficient. + + Each user message sent is either delivered to the receiver or, in + case of excessive retransmissions, the association is terminated in a + non-graceful way [RFC4960], similar to TCP behaviour. In addition to + this reliable transfer, the partial reliability extension [RFC3758] + allows a sender to abandon user messages. The application can + specify the policy for abandoning user messages. Examples for these + policies defined in [RFC3758] and [RFC7496] are: o Limiting the time a user message is dealt with by the sender. o Limiting the number of retransmissions for each fragment of a user message. If the number of retransmissions is limited to 0, one gets a service similar to UDP. o Abandoning messages of lower priority in case of a send buffer shortage. - SCTP supports multi-homing. Each SCTP end-point uses a list of IP- + SCTP supports multi-homing. Each SCTP endpoint uses a list of IP- addresses and a single port number. These addresses can be any mixture of IPv4 and IPv6 addresses. These addresses are negotiated during the handshake and the address re-configuration extension specified in [RFC5061] in combination with [RFC4895] can be used to change these addresses in an authenticated way during the livetime of an SCTP association. This allows for transport layer mobility. Multiple addresses are used for improved resilience. If a remote address becomes unreachable, the traffic is switched over to a reachable one, if one exists. Each SCTP end-point supervises continuously the reachability of all peer addresses using a heartbeat mechanism. For securing user messages, the use of TLS over SCTP has been specified in [RFC3436]. However, this solution does not support all services provided by SCTP (for example un-ordered delivery or partial reliability), and therefore the use of DTLS over SCTP has been specified in [RFC6083] to overcome these limitations. When using DTLS over SCTP, the application can use almost all services provided by SCTP. - [I-D.ietf-tsvwg-natsupp] defines a methods for end-hosts and - middleboxes to provide for NAT support for SCTP over IPv4. For - legacy NAT traversal, [RFC6951] defines the UDP encapsulation of - SCTP-packets. Alternatively, SCTP packets can be encapsulated in - DTLS packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The + [I-D.ietf-tsvwg-natsupp] defines methods for endpoints and + middleboxes to provide support NAT for SCTP over IPv4. For legacy + NAT traversal, [RFC6951] defines the UDP encapsulation of SCTP- + packets. Alternatively, SCTP packets can be encapsulated in DTLS + packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The latter encapsulation is used with in the WebRTC context. - Having a well defined API is also a feature provided by SCTP as - described in the next subsection. + SCTP has a well-defined API, described in the next subsection. 3.3.2. Interface Description - [RFC4960] defines an abstract API for the base protocol. An - extension to the BSD Sockets API is defined in [RFC6458] and covers: + [RFC4960] defines an abstract API for the base protocol. This API + describes the following functions callable by the upper layer of + SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message, + Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status, + Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure + Threshold, Set Protocol Parameters, and Destroy. The following + notifications are provided by the SCTP stack to the upper layer: + COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST, + COMMUNICATION ERROR, RESTART, SEND FAILURE, NETWORK STATUS CHANGE. - o the base protocol defined in [RFC4960]. + An extension to the BSD Sockets API is defined in [RFC6458] and + covers: - o the SCTP Partial Reliability extension defined in [RFC3758]. + o the base protocol defined in [RFC4960]. The API allows to control + the local addresses and port numbers and the primary path. + Furthermore the application has fine control about parameters like + retransmission thresholds, the path supervision parameters, the + delayed acknowledgement timeout, and the fragmentation point. The + API provides a mechanism to allow the SCTP stack to notify the + application about event if the application has requested them. + These notifications provide Information about status changes of + the association and each of the peer addresses. In case of send + failures that application can also be notified and user messages + can be returned to the application. When sending user messages, + the stream id, a payload protocol identifier, an indication + whether ordered delivery is requested or not. These parameters + can also be provided on message reception. Additionally a context + can be provided when sending, which can be use in case of send + failures. The sending of arbitrary large user messages is + supported. - o the SCTP Authentication extension defined in [RFC4895]. + o the SCTP Partial Reliability extension defined in [RFC3758] to + specify for a user message the PR-SCTP policy and the policy + specific parameter. + + o the SCTP Authentication extension defined in [RFC4895] allowing to + manage the shared keys, the HMAC to use, set the chunk types which + are only accepted in an authenticated way, and get the list of + chunks which are accepted by the local and remote end point in an + authenticated way. o the SCTP Dynamic Address Reconfiguration extension defined in - [RFC5061]. + [RFC5061]. It allows to manually add and delete local addresses + for SCTP associations and the enabling of automatic address + addition and deletion. Furthermore the peer can be given a hint + for choosing its primary path. For the following SCTP protocol extensions the BSD Sockets API extension is defined in the document specifying the protocol extensions: - o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053]. - o the SCTP Stream Reconfiguration extension defined in [RFC6525]. + The API allows to trigger the reset operation for incoming and + outgoing streams and the whole association. It provides also a + way to notify the association about the corresponding events. + Furthermore the application can increase the number of streams. o the UDP Encapsulation of SCTP packets extension defined in - [RFC6951]. + [RFC6951]. The API allows the management of the remote UDP + encapsulation port. - o the additional PR-SCTP policies defined in - [I-D.ietf-tsvwg-sctp-prpolicies]. + o the SCTP SACK-IMMEDIATELY extension defined in [RFC7053]. The API + allows the sender of a user message to request the receiver to + send the corresponding acknowledgement immediately. + + o the additional PR-SCTP policies defined in [RFC7496]. The API + allows to enable/disable the PR-SCTP extension, choose the PR-SCTP + policies defined in the document and provide statistical + information about abandoned messages. Future documents describing SCTP protocol extensions are expected to describe the corresponding BSD Sockets API extension in a "Socket API Considerations" section. The SCTP socket API supports two kinds of sockets: o one-to-one style sockets (by using the socket type "SOCK_STREAM"). o one-to-many style socket (by using the socket type "SOCK_SEQPACKET"). One-to-one style sockets are similar to TCP sockets, there is a 1:1 relationship between the sockets and the SCTP associations (except for listening sockets). One-to-many style SCTP sockets are similar - to unconnected UDP sockets as there is a 1:n relationship between the - sockets and the SCTP associations. + to unconnected UDP sockets, where there is a 1:n relationship between + the sockets and the SCTP associations. The SCTP stack can provide information to the applications about state changes of the individual paths and the association whenever they occur. These events are delivered similar to user messages but are specifically marked as notifications. - A couple of new functions have been introduced to support the use of - multiple local and remote addresses. Additional SCTP-specific send - and receive calls have been defined to allow dealing with the SCTP - specific information without using ancillary data in the form of - additional cmsgs, which are also defined. These functions provide - support for detecting partial delivery of user messages and - notifications. + New functions have been introduced to support the use of multiple + local and remote addresses. Additional SCTP-specific send and + receive calls have been defined to permit SCTP-specific information + to be snet without using ancillary data in the form of additional + cmsgs. These functions provide support for detecting partial + delivery of user messages and notifications. The SCTP socket API allows a fine-grained control of the protocol behaviour through an extensive set of socket options. The SCTP kernel implementations of FreeBSD, Linux and Solaris follow mostly the specified extension to the BSD Sockets API for the base protocol and the corresponding supported protocol extensions. -3.3.3. Transport Protocol Components +3.3.3. Transport Features - The transport protocol components provided by SCTP are: + The transport features provided by SCTP are: - o unicast + [GF: This needs to be harmonised with the components for TCP] - o connection setup with feature negotiation and application-to-port - mapping + o unicast. - o port multiplexing + o connection setup with feature negotiation and application-to-port + mapping. - o reliable or partially reliable delivery + o port multiplexing. - o ordered and unordered delivery within a stream + o message-oriented delivery. - o support for multiple concurrent streams + o fully reliable or partially reliable delivery. - o support for stream scheduling prioritization + o ordered and unordered delivery within a stream. - o flow control + o support for multiple concurrent streams. - o message-oriented delivery + o support for stream scheduling prioritization. - o congestion control + o flow control. - o user message bundling + o congestion control. - o user message fragmentation and reassembly + o user message bundling. - o strong error detection (CRC32C) + o user message fragmentation and reassembly. - o transport layer multihoming for resilience + o strong error/misdelivery detection (CRC32c). - o transport layer mobility + o transport layer multihoming for resilience. - [EDITOR'S NOTE: update this list.] + o transport layer mobility. 3.4. User Datagram Protocol (UDP) The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF - standards track transport protocol. It provides a uni-directional, - datagram protocol which preserves message boundaries. It provides + standards track transport protocol. It provides a unidirectional, + datagram protocol that preserves message boundaries. It provides none of the following transport features: error correction, congestion control, or flow control. It can be used to send broadcast datagrams (IPv4) or multicast datagrams (IPv4 and IPv6), in addition to unicast (and anycast) datagrams. IETF guidance on the - use of UDP is provided in[RFC5405]. UDP is widely implemented and - widely used by common applications, especially DNS. + use of UDP is provided in[I-D.ietf-tsvwg-rfc5405bis]. UDP is widely + implemented and widely used by common applications, including DNS. 3.4.1. Protocol Description - UDP is a connection-less protocol which maintains message boundaries, + UDP is a connection-less protocol that maintains message boundaries, with no connection setup or feature negotiation. The protocol uses - independent messages, ordinarily called datagrams. The lack of error - control and flow control implies messages may be damaged, re-ordered, - lost, or duplicated in transit. A receiving application unable to - run sufficiently fast or frequently may miss messages. The lack of - congestion handling implies UDP traffic may cause the loss of - messages from other protocols (e.g., TCP) when sharing the same - network paths. UDP traffic can also cause the loss of other UDP - traffic in the same or other flows for the same reasons. + independent messages, ordinarily called datagrams. Each stream of + messages is independently managed, therefore retransmission does not + hold back data sent using other logical streams. It provides + detection of payload errors and misdelivery of packets to the wrong + endpoint, either of which result in discard of received datagrams. - Messages with bit errors are ordinarily detected by an invalid end- - to-end checksum and are discarded before being delivered to an - application. There are some exceptions to this general rule, - however. UDP-Lite (see [RFC3828], and below) provides the ability - for portions of the message contents to be exempt from checksum - coverage. It is also possible to create UDP datagrams with no - checksum, and while this is generally discouraged [RFC1122] - [RFC5405], certain special cases permit its use [RFC6935]. The - checksum support considerations for omitting the checksum are defined - in [RFC6936]. Note that due to the relatively weak form of checksum - used by UDP, applications that require end to end integrity of data - are recommended to include a stronger integrity check of their - payload data. + It is possible to create IPv4 UDP datagrams with no checksum, and + while this is generally discouraged [RFC1122] + [I-D.ietf-tsvwg-rfc5405bis], certain special cases permit its use. + These datagrams relie on the IPv4 header checksum to protect from + misdelivery to the wrong endpoint. IPv6 does not by permit UDP + datagrams with no checksum, although in certain cases this rule may + be relaxed [RFC6935]. The checksum support considerations for + omitting the checksum are defined in [RFC6936]. Note that due to the + relatively weak form of checksum used by UDP, applications that + require end to end integrity of data are recommended to include a + stronger integrity check of their payload data. + + It does not provide reliability and does not provide retransmission. + This implies messages may be re-ordered, lost, or duplicated in + transit. + + A receiving application that is unable to run sufficiently fast, or + frequently, may miss messages since there is no flow control. The + lack of congestion handling implies UDP traffic may experience loss + when using an overlaoded path and may cause the loss of messages from + other protocols (e.g., TCP) when sharing the same network path. + + [GF: This para isn't needed": Messages with payload errors are + ordinarily detected by an invalid end- to-end checksum and are + discarded before being delivered to an application. UDP-Lite (see + [RFC3828], and below) provides the ability for portions of the + message contents to be exempt from checksum coverage.] On transmission, UDP encapsulates each datagram into an IP packet, - which may in turn be fragmented by IP. Applications concerned with - fragmentation or that have other requirements such as receiver flow - control, congestion control, PathMTU discovery/PLPMTUD, support for - ECN, etc need to be provided by protocols other than UDP [RFC5405]. + which may in turn be fragmented by IP and are reassembled before + delivery to the UDP receiver. + + Applications that need to provide fragmentation or that have other + requirements such as receiver flow control, congestion control, + PathMTU discovery/PLPMTUD, support for ECN, etc need these to be + provided by protocols operating over UDP [I-D.ietf-tsvwg-rfc5405bis]. 3.4.2. Interface Description [RFC0768] describes basic requirements for an API for UDP. Guidance - on use of common APIs is provided in [RFC5405]. + on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis]. A UDP endpoint consists of a tuple of (IP address, port number). Demultiplexing using multiple abstract endpoints (sockets) on the same IP address are supported. The same socket may be used by a single server to interact with multiple clients (note: this behavior differs from TCP, which uses a pair of tuples to identify a - connection). Multiple server instances (processes) binding the same - socket can cooperate to service multiple clients- the socket + connection). Multiple server instances (processes) that bind the + same socket can cooperate to service multiple clients- the socket implementation arranges to not duplicate the same received unicast message to multiple server processes. Many operating systems also allow a UDP socket to be "connected", i.e., to bind a UDP socket to a specific (remote) UDP endpoint. Unlike TCP's connect primitive, for UDP, this is only a local operation that serves to simplify the local send/receive functions and to filter the traffic for the specified addresses and ports - [RFC5405]. + [I-D.ietf-tsvwg-rfc5405bis]. -3.4.3. Transport Protocol Components +3.4.3. Transport Features - The transport protocol components provided by UDP are: + The transport features provided by UDP are: - o unidirectional + o unicast. - o port multiplexing + o multicast, anycast, or IPv4 broadcast. - o 2-tuple endpoints + o port multiplexing. A receiving port can be configured to receive + datagrams from multiple senders. - o IPv4 broadcast, multicast and anycast + o message-oriented delivery. - o IPv6 multicast and anycast + o unidirectional or bidirectional. Transmission in each direction + is independent. - o IPv6 jumbograms + o non-reliable delivery. - o message-oriented delivery + o non-ordered delivery. - o error detection (checksum) + o IPv6 jumbograms. - o checksum optional + o error and misdelivery detection (checksum). + + o optional checksum. All or none of the payload data is protected. 3.5. Lightweight User Datagram Protocol (UDP-Lite) The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an - IETF standards track transport protocol. UDP-Lite provides a - bidirectional set of logical unicast or multicast message streams - over a datagram protocol. IETF guidance on the use of UDP-Lite is - provided in [RFC5405]. + IETF standards track transport protocol. It provides a + unidirectional, datagram protocol that preserves message boundaries. + IETF guidance on the use of UDP-Lite is provided in + [I-D.ietf-tsvwg-rfc5405bis]. 3.5.1. Protocol Description + UDP-Lite is a connection-less datagram protocol, with no connection setup or feature negotiation. The protocol use messages, rather than a byte-stream. Each stream of messages is independently managed, therefore retransmission does not hold back data sent using other logical streams. It provides multiplexing to multiple sockets on each host using port - numbers. An active UDP-Lite session is identified by its four-tuple - of local and remote IP addresses and local port and remote port - numbers. - - UDP-Lite fragments packets into IP packets, constrained by the - maximum size of IP packet. + numbers, and its operation follows that for UDP. An active UDP-Lite + session is identified by its four-tuple of local and remote IP + addresses and local port and remote port numbers. UDP-Lite changes the semantics of the UDP "payload length" field to - that of a "checksum coverage length" field. Otherwise, UDP-Lite is + that of a "checksum coverage length" field, and is identified by a + different IP protocol/next-header value. Otherwise, UDP-Lite is semantically identical to UDP. Applications using UDP-Lite therefore can not make assumptions regarding the correctness of the data received in the insensitive part of the UDP-Lite payload. As for UDP, mechanisms for receiver flow control, congestion control, PMTU or PLPMTU discovery, support for ECN, etc need to be provided by - upper layer protocols [RFC5405]. + upper layer protocols [I-D.ietf-tsvwg-rfc5405bis]. Examples of use include a class of applications that can derive benefit from having partially-damaged payloads delivered, rather than discarded. One use is to support error tolerate payload corruption when used over paths that include error-prone links, another application is when header integrity checks are required, but payload - integrity is provided by some other mechanism (e.g. [RFC6936]. + integrity is provided by some other mechanism (e.g., [RFC6936]. A UDP-Lite service may support IPv4 broadcast, multicast, anycast and - unicast. + unicast, and IPv6 multicast, anycast and unicast. 3.5.2. Interface Description There is no current API specified in the RFC Series, but guidance on - use of common APIs is provided in [RFC5405]. + use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis]. The interface of UDP-Lite differs from that of UDP by the addition of a single (socket) option that communicates a checksum coverage length value: at the sender, this specifies the intended checksum coverage, with the remaining unprotected part of the payload called the "error- insensitive part". The checksum coverage may also be made visible to the application via the UDP-Lite MIB module [RFC5097]. -3.5.3. Transport Protocol Components - The transport protocol components provided by UDP-Lite are: +3.5.3. Transport Features - o unicast + The transport features provided by UDP-Lite are: - o IPv4 broadcast, multicast and anycast + o unicast. - o port multiplexing + o multicast, anycast, or IPv4 broadcast. - o non-reliable, non-ordered delivery + o port multiplexing (as for UDP). - o message-oriented delivery + o message-oriented delivery (as for UDP). - o partial integrity protection + o non-reliable delivery (as for UDP). + + o non-ordered delivery (as for UDP). + + o error and misdelivery detection (checksum). + + o partialor full integrity protection. The checksum coverage field + indicates the size of the payload data covered by the checksum. 3.6. Datagram Congestion Control Protocol (DCCP) Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF standards track bidirectional transport protocol that provides - unicast connections of congestion-controlled unreliable messages. - - [EDITOR'S NOTE: Gorry Fairhurst signed up as a contributor for this - section.] + unicast connections of congestion-controlled messages without + providing reliability. The DCCP Problem Statement describes the goals that DCCP sought to address [RFC4336]. It is suitable for applications that transfer fairly large amounts of data and that can benefit from control over the trade off between timeliness and reliability [RFC4336]. It offers low overhead, and many characteristics common to UDP, but can avoid "Re-inventing the wheel" each time a new multimedia application emerges. Specifically it includes core functions (feature negotiation, path state management, RTT calculation, PMTUD, @@ -803,139 +880,542 @@ to manage their functions. Examples of suitable applications include interactive applications, streaming media or on-line games [RFC4336]. 3.6.1. Protocol Description DCCP is a connection-oriented datagram protocol, providing a three way handshake to allow a client and server to set up a connection, and mechanisms for orderly completion and immediate teardown of a connection. The protocol is defined by a family of RFCs. - It provides multiplexing to multiple sockets on each host using port - numbers. An active DCCP session is identified by its four-tuple of - local and remote IP addresses and local port and remote port numbers. - At connection setup, DCCP also exchanges the the service code - - [RFC5595] mechanism to allow transport instantiations to indicate the - service treatment that is expected from the network. + It provides multiplexing to multiple sockets at each endpoint using + port numbers. An active DCCP session is identified by its four-tuple + of local and remote IP addresses and local port and remote port + numbers. At connection setup, DCCP also exchanges the service code + [RFC5595], a mechanism that allows transport instantiations to + indicate the service treatment that is expected from the network. The protocol segments data into messages, typically sized to fit in IP packets, but which may be fragmented providing they are less than - the A DCCP interface MAY allow applications to request fragmentation - for packets larger than PMTU, but not larger than the maximum packet - size allowed by the current congestion control mechanism (CCMPS) - [RFC4340]. + the maximum packet size. A DCCP interface allows applications to + request fragmentation for packets larger than PMTU, but not larger + than the maximum packet size allowed by the current congestion + control mechanism (CCMPS) [RFC4340]. Each message is identified by a sequence number. The sequence number is used to identify segments in acknowledgments, to detect unacknowledged segments, to measure RTT, etc. The protocol may support ordered or unordered delivery of data, and does not itself - provide retransmission. There is a Data Checksum option, which - contains a strong CRC, lets endpoints detect application data - corruption. It also supports reduced checksum coverage, a partial - integrity mechanisms similar to UDP-lIte. + provide retransmission. DCCP supports reduced checksum coverage, a + partial integrity mechanisms similar to UDP-lIte. There is also a + Data Checksum option that when enabled, contains a strong CRC, to + enable endpoints to detect application data corruption. Receiver flow control is supported: limiting the amount of unacknowledged data that can be outstanding at a given time. - A DCCP protocol instance can be extended [RFC4340] and tuned. Some - features are sender-side only, requiring no negotiation with the - receiver; some are receiver-side only, some are explicitly negotiated - during connection setup. + A DCCP protocol instance can be extended [RFC4340] and tuned using + features. Some features are sender-side only, requiring no + negotiation with the receiver; some are receiver-side only, some are + explicitly negotiated during connection setup. + + A DCCP service is unicast. DCCP supports negotiation of the congestion control profile, to - provide Plug and Play congestion control mechanisms. examples of + provide Plug and Play congestion control mechanisms. Examples of specified profiles include [RFC4341] [RFC4342] [RFC5662]. All IETF- defined methods provide Congestion Control. - DCCP use a Connect packet to start a session, and permits half- - connections that allow each client to choose features it wishes to - support. Simultaneous open [RFC5596], as in TCP, can enable + DCCP use a Connect packet to initiate a session, and permits half- + connections that allow each client to choose the features it wishes + to support. Simultaneous open [RFC5596], as in TCP, can enable interoperability in the presence of middleboxes. The Connect packet includes a Service Code field [RFC5595] designed to allow middle boxes and endpoints to identify the characteristics required by a - session. A lightweight UDP-based encapsulation (DCCP-UDP) has been - defined [RFC6773] that permits DCCP to be used over paths where it is - not natively supported. Support in NAPT/NATs is defined in [RFC4340] - and [RFC5595]. + session. + + A lightweight UDP-based encapsulation (DCCP-UDP) has been defined + [RFC6773] that permits DCCP to be used over paths where it is not + natively supported. Support in NAPT/NATs is defined in [RFC4340] and + [RFC5595]. Upper layer protocols specified on top of DCCP include: DTLS [RFC5595], RTP [RFC5672], ICE/SDP [RFC6773]. - A DCCP service is unicast. - A common packet format has allowed tools to evolve that can read and interpret DCCP packets (e.g. Wireshark). 3.6.2. Interface Description API characteristics include: - Datagram transmission. - Notification of the current maximum packet size. - Send and reception of zero- - length payloads. - Set the Slow Receiver flow control at a receiver. - - Detect a Slow receiver at the sender. + length payloads. - Slow Receiver flow control at a receiver. - + Detect a Slow receiver at the sender. - There is no current API specified in the RFC Series. + There is no current API curremntly specified in the RFC Series. -3.6.3. Transport Protocol Components +3.6.3. Transport Features - The transport protocol components provided by DCCP are: + The transport features provided by DCCP are: - o unicast + o unicast. o connection setup with feature negotiation and application-to-port - mapping + mapping. - o Service Codes + o Service Codes. Identifies the upper layer service to the endpoint + and network. - o port multiplexing + o port multiplexing. - o non-reliable, ordered delivery + o message-oriented delivery. - o flow control (slow receiver function) + o non-reliable delivery. - o drop notification + o ordered delivery. - o timestamps + o flow control. The slow receiver function allows a receiver to + control the rate of the sender. - o message-oriented delivery + o drop notification. Allows a receiver to notify which datagrams + were not delivered to the peer upper layer protocol. - o partial integrity protection + o timestamps. -3.7. Realtime Transport Protocol (RTP) + o partial and full integrity protection (with optional strong + integrity check). + +3.7. Lightweight User Datagram Protocol (UDP-Lite) + + The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an + IETF standards track transport protocol. It provides a + unidirectional, datagram protocol that preserves message boundaries. + IETF guidance on the use of UDP-Lite is provided in + [I-D.ietf-tsvwg-rfc5405bis]. + +3.7.1. Protocol Description + + UDP-Lite is a connection-less datagram protocol, with no connection + setup or feature negotiation. The protocol use messages, rather than + a byte-stream. Each stream of messages is independently managed, + therefore retransmission does not hold back data sent using other + logical streams. + + It provides multiplexing to multiple sockets on each host using port + numbers, and its operation follows that for UDP. An active UDP-Lite + session is identified by its four-tuple of local and remote IP + addresses and local port and remote port numbers. + + UDP-Lite changes the semantics of the UDP "payload length" field to + that of a "checksum coverage length" field, and is identified by a + different IP protocol/next-header value. Otherwise, UDP-Lite is + semantically identical to UDP. Applications using UDP-Lite therefore + can not make assumptions regarding the correctness of the data + received in the insensitive part of the UDP-Lite payload. + + As for UDP, mechanisms for receiver flow control, congestion control, + PMTU or PLPMTU discovery, support for ECN, etc need to be provided by + upper layer protocols [I-D.ietf-tsvwg-rfc5405bis]. + + Examples of use include a class of applications that can derive + benefit from having partially-damaged payloads delivered, rather than + discarded. One use is to support error tolerate payload corruption + when used over paths that include error-prone links, another + application is when header integrity checks are required, but payload + integrity is provided by some other mechanism (e.g., [RFC6936]. + + A UDP-Lite service may support IPv4 broadcast, multicast, anycast and + unicast, and IPv6 multicast, anycast and unicast. + +3.7.2. Interface Description + + There is no current API specified in the RFC Series, but guidance on + use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis]. + + The interface of UDP-Lite differs from that of UDP by the addition of + a single (socket) option that communicates a checksum coverage length + value: at the sender, this specifies the intended checksum coverage, + with the remaining unprotected part of the payload called the "error- + insensitive part". The checksum coverage may also be made visible to + the application via the UDP-Lite MIB module [RFC5097]. + +3.7.3. Transport Features + + The transport features provided by UDP-Lite are: + + o unicast + + o multicast, anycast, or IPv4 broadcast. + + o port multiplexing (as for UDP). + + o message-oriented delivery (as for UDP). + + o non-reliable delivery(as for UDP). + + o non-ordered delivery (as for UDP). + + o partial or full integrity protection. + +3.8. Internet Control Message Protocol (ICMP) + + The Internet Control Message Protocol (ICMP) [RFC0792] for IPv4 and + [RFC4433] for IPv6 are IETF standards track protocols. + + It provides a conection-less unidirectional protocol that delivers + individual messages. It provides none of the following transport + features: error correction, congestion control, or flow control. + Some messages may be sent as broadcast datagrams (IPv4) or multicast + datagrams (IPv4 and IPv6), in addition to unicast (and anycast) + datagrams. + +3.8.1. Protocol Description + + ICMP is a conection-less unidirectional protocol that delivers + individual messages. The protocol uses independent messages, + ordinarily called datagrams. Each message is required to carry a + checksum as an integrity check and to protect from misdelivery to the + wrong endpoint. + + ICMP messages typically relay diagnostic information from an endpoint + [RFC1122] or network device [RFC1716] addressed to the sender of a + flow. This usually contains the network protocol header of a packet + that encountered the reported issue. Some formats of messages may + also carry other payload data. Each message carries an integrity + check calculated in the same way as UDP. + + The RFC series defines additional IPv6 message formats to support a + range of uses. In the case of IPv6 the protocol incorporates + neighbour discovery [RFC2461] [RFC3971]} (provided by ARP for IPv4) + and the Multicast Listener Discovery (MLD) [RFC2710] group management + functions (provided by IGMP for IPv4). + + Reliable transmission can not be assumed. A receiving application + that is unable to run sufficiently fast, or frequently, may miss + messages since there is no flow or congestion control. In addition + some network devices rate-limit ICMP messages. + + Transport Protocols and upper layer protocols can use ICMP messages + to help them take appropriate decisions when network or endpoint + errors are reported. For example to implement, ICMP-based PathMTU + discovery [RFC1191][RFC1981] or assist in Packetization Layer Path + MTU Discovery (PMTUD) [RFC4821]. Such reactions to received messages + needs to protects from off-path data injection + [I-D.ietf-tsvwg-rfc5405bis], avoiding an application receiving + packets that were created by an unauthorized third party. An + application therefore needs to ensure that aLL messaged are + appropriately validated, by checking the payload of the messages to + ensure these are received in response to actually transmitted traffic + (e.g., a reported error condition that corresponds to a UDP datagram + or TCP segment was actually sent by the application). This requires + context [RFC6056], such as local state about communication instances + to each destination (e.g., in the TCP, DCCP, or SCTP protocols). + This state is not always maintained by UDP-based applications + [I-D.ietf-tsvwg-rfc5405bis]. + + Any response to ICMP error messages ought to be robust to temporary + routing failures (sometimes called "soft errors"), e.g., transient + ICMP "unreachable" messages ought to not normally cause a + communication abort [RFC5461] [I-D.ietf-tsvwg-rfc5405bis]. + +3.8.2. Interface Description + + ICMP processing is integrated into many connection-oriented + transports, but like other functions needs to be provided by an + upper-layer protocol when using UDP and UDP-Lite. On some stacks, a + bound socket also allows a UDP application to be notified when ICMP + error messages are received for its transmissions + [I-D.ietf-tsvwg-rfc5405bis]. + +3.8.3. Transport Features + + The transport features provided by ICMP are: + + o unidirectional. + + o multicast, anycast and IP4 broadcast. + + o message-oriented delivery. + + o non-reliable delivery. + + o non-ordered delivery. + + o error and misdelivery detection (checksum). + +3.9. Realtime Transport Protocol (RTP) RTP provides an end-to-end network transport service, suitable for applications transmitting real-time data, such as audio, video or data, over multicast or unicast network services, including TCP, UDP, - UDP-Lite, DCCP. + UDP-Lite, or DCCP. [EDITOR'S NOTE: Varun Singh signed up as contributor for this section. Given the complexity of RTP, suggest to have an abbreviated section here contrasting RTP with other transports, and focusing on - those features that are RTP-unique.] + those features that are RTP-unique. Gorry Fairhurst contributed this + stub section] -3.8. NACK-Oriented Reliable Multicast (NORM) +3.9.1. Protocol Description + + The RTP standard [RFC3550] defines a pair of protocols, RTP and the + Real Time Control Protocol, RTCP. The transport does not provide + connection setup, but relies on out-of-band techniques or associated + control protocols to setup, negotiate parameters or tear-down a + session. + + An RTP sender encapsulates audio/video data into RTP packets to + transport media streams. The RFC-series specifies RTP media formats + allow packets to carry a wide range of media, and specifies a wide + range of mulriplexing, error control and other support mechanisms. + + If a frame of media data is large, it will be fragment this into + several RTP packets. If small, several frames may be bundled into a + single RTP packet. RTP may runs over a congestion-controlled or non- + congestion-controlled transport protocol. + + An RTP receiver collects RTP packets from network, validates them for + correctness, and sends them to the media decoder input-queue. + Missing packet detection is performed by the channel decoder. The + play-out buffer is ordered by time stamp and is used to reorder + packets. Damaged frames may be repaired before the media payloads + are decompressed to display or store the data. + + RTCP is an associated control protocol that works with RTP. Both the + RTP sender and receiver can send RTCP report packets. This is used + to periodically send control information and report performance. + Based on received RTCP feedback, an RTP sender can adjust the + transmission, e.g., perform rate adaptation at the application layer + in the case of congestion. + + An RTCP receiver report (RTCP RR) is returned to the sender + periodically to report key parameters (e.g, the fraction of packets + lost in the last reporting interval, the cumulative number of packets + lost, the highest sequence number received, and the inter-arrival + jitter). The RTCP RR packets also contain timing information that + allows the sender to estimate the network round trip time (RTT) to + the receivers. + + The interval between reports sent from each receiver tends to be on + the order of a few seconds on average, although this varies with the + session rate, and sub-second reporting intervals are possible for + high rate sessions. The interval is randomised to avoid + synchronization of reports from multiple receivers. + +3.9.2. Interface Description + + [EDITOR'S NOTE: to do] + +3.9.3. Transport Features + + The transport features provided by RTP are: + + o unicast. + + o multicast, anycast or IPv4 broadcast. + + o port multiplexing. + + o message-oriented delivery. + + o associated protocols for connection setup with feature negotiation + and application-to-port mapping. + + o support for media types and other extensions. + + o segmentation and aggregation. + + o performance reporting. + + o drop notification. + + o timestamps. + +3.10. File Delivery over Unidirectional Transport/Asynchronous Layered + Coding Reliable Multicast (FLUTE/ALC) + + FLUTE/ALC is an IETF standards track protocol specified in [RFC6726] + and [RFC5775],. ALC provides an underlying reliable transport service + and FLUTE a file-oriented specialization of the ALC service (e.g., to + carry associated metadata). The [RFC6726] and [RFC5775] protocols + are non-backward-compatible updates of the [RFC3926] and [RFC3450] + experimental protocols; these experimental protocols are currently + largely deployed in the 3GPP Multimedia Broadcast and Multicast + Services (MBMS) (see [MBMS], section 7) and similar contexts (e.g., + the Japanese ISDB-Tmm standard). + + The FLUTE/ALC protocol has been designed to support massively + scalable reliable bulk data dissemination to receiver groups of + arbitrary size using IP Multicast over any type of delivery network, + including unidirectional networks (e.g., broadcast wireless + channels). However, the FLUTE/ALC protocol also supports point-to- + point unicast transmissions. + + FLUTE/ALC bulk data dissemination has been designed for discrete file + or memory-based "objects". Transmissions happen either in push mode, + where content is sent once, or in on-demand mode, where content is + continuously sent during periods of time that can largely exceed the + average time required to download the session objects (see [RFC5651], + section 4.2). + + Altough FLUTE/ALC is not well adapted to byte- and message-streaming, + there is an exception: FLUTE/ALC is used to carry 3GPP Dynamic + Adaptive Streaming over HTTP (DASH) when scalability is a requirement + (see [MBMS], section 5.6). In that case, each Audio/Video segment is + transmitted as a distinct FLUTE/ALC object in push mode. FLUTE/ALC + uses packet erasure coding (also known as Application-Level Forward + Erasure Correction, or AL-FEC) in a proactive way. The goal of using + AL-FEC is both to increase the robustness in front of packet erasures + and to improve the efficiency of the on-demand service. FLUTE/ALC + transmissions can be governed by a congestion control mechanism such + as the "Wave and Equation Based Rate Control" (WEBRC) [RFC3738] when + FLUTE/ALC is used in a layered transmission manner, with several + session channels over which ALC packets are sent. However many + FLUTE/ALC deployments involve only Constant Bit Rate (CBR) channels + with no competing flows, for which a sender-based rate control + mechanism is sufficient. In any case, FLUTE/ALC's reliability, + delivery mode, congestion control, and flow/rate control mechanisms + are distinct components that can be separately controlled to meet + different application needs. + +3.10.1. Protocol Description + + The FLUTE/ALC protocol works on top of UDP (though it could work on + top of any datagram delivery transport protocol), without requiring + any connectivity from receivers to the sender. Purely unidirectional + networks are therefore supported by FLUTE/ALC. This guarantees + scalability to an unlimited number of receivers in a session, since + the sender behaves exactly the same regardness of the number of + receivers. + + FLUTE/ALC supports the transfer of bulk objects such as file or in- + memory content, using either a push or an on-demand mode. in push + mode, content is sent once to the receivers, while in on-demand mode, + content is sent continuously during periods of time that can greatly + exceed the average time required to download the session objects. + + This enables receivers to join a session asynchronously, at their own + discretion, receive the content and leave the session. In this case, + data content is typically sent continuously, in loops (also known as + "carousels"). FLUTE/ALC also supports the transfer of an object + stream, with loose real-time constraints. This is particularly + useful to carry 3GPP DASH when scalability is a requirement and + unicast transmissions over HTTP cannot be used ([MBMS], section 5.6). + In this case, packets are sent in sequence using push mode. FLUTE/ + ALC is not well adapted to byte- and message-streaming and other + solutions could be preferred (e.g., FECFRAME [RFC6363] with real-time + flows). + + The FLUTE file delivery instantiation of ALC provides a metadata + delivery service. Each object of the FLUTE/ALC session is described + in a dedicated entry of a File Delivery Table (FDT), using an XML + format (see [RFC6726], section 3.2). This metadata can include, but + is not restricted to, a URI attribute (to identify and locate the + object), a media type attribute, a size attribute, an encoding + attribute, or a message digest attribute. Since the set of objects + sent within a session can be dynamic, with new objects being added + and old ones removed, several instances of the FDT can be sent and a + mechanism is provided to identify a new FDT Instance. + + To provide robustness against packet loss and improve the efficiency + of the on-demand mode, FLUTE/ALC relies on packet erasure coding (AL- + FEC). AL-FEC encoding is proactive (since there is no feedback and + therefore no (N)ACK-based retransmission) and ALC packets containing + repair data are sent along with ALC packets containing source data. + Several FEC Schemes have been standardized; FLUTE/ALC does not + mandate the use of any particular one. Several strategies concerning + the transmission order of ALC source and repair packets are possible, + in particular in on-demand mode where it can deeply impact the + service provided (e.g., to favor the recovery of objects in sequence, + or at the other extreme, to favor the recovery of all objects in + parallel), and FLUTE/ALC does not mandate nor recommend the use of + any particular one. + + A FLUTE/ALC session is composed of one or more channels, associated + to different destination unicast and/or multicast IP addresses. ALC + packets are sent in those channels at a certain transmission rate, + with a rate that often differs depending on the channel. FLUTE/ALC + does not mandate nor recommend any strategy to select which ALC + packet to send on which channel. FLUTE/ALC can use a multiple rate + congestion control building block (e.g., WEBRC) to provide congestion + control that is feedback free, where receivers adjust their reception + rates individually by joining and leaving channels associated with + the session. To that purpose, the ALC header provides a specific + field to carry congestion control specific information. However + FLUTE/ALC does not mandate the use of a particular congestion control + mechanism although WEBRC is mandatory to support in case of Internet + ([RFC6726], section 1.1.4). FLUTE/ALC is often used over a network + path with pre-provisoned capacity [RFC5404] whete theres are no flows + competing for capacity. In this case, a sender-based rate control + mechanism and a single channel is sufficient. + + [RFC6584] provides per-packet authentication, integrity, and anti- + replay protection in the context of the ALC and NORM protocols. + Several mechanisms are proposed that seamlessly integrate into these + protocols using the ALC and NORM header extension mechanisms. + +3.10.2. Interface Description + + The FLUTE/ALC specification does not describe a specific application + programming interface (API) to control protocol operation. + Open source reference implementations of FLUTE/ALC are available at + http://planete-bcast.inrialpes.fr/ (no longer maintained) and + http://mad.cs.tut.fi/ (no longer maintained), and these + implementations specify and document their own APIs. Commercial + versions are also available, some derived from the above + implementations, with their own API. + +3.10.3. Transport Features + + The transport features provided by FLUTE/ALC are: + + o unicast + + o multicast, anycast or IPv4 broadcast. + + o per-object dynamic meta-data delivery. + + o push delivery or on-demand delivery service. + + o fully reliable or partially reliable delivery (of file or in- + memory objects). + + o ordered or unordered delivery (of file or in-memory objects). + + o per-packet authentication, integrity, and anti-replay services. + + o proactive packet erasure coding (AL-FEC) to recover from packet + erasures and improve the on-demand delivery service, + + o error detection (through UDP and lower level checksums). + + o congestion control for layered flows (e.g., with WEBRC). + + o rate control transmission in a given channel. + +3.11. NACK-Oriented Reliable Multicast (NORM) NORM is an IETF standards track protocol specified in [RFC5740]. The protocol was designed to support reliable bulk data dissemination to receiver groups using IP Multicast but also provides for point-to- point unicast operation. Its support for bulk data dissemination includes discrete file or computer memory-based "objects" as well as byte- and message-streaming. NORM is designed to incorporate packet erasure coding as an inherent part of its selective ARQ in response to receiver negative acknowledgements. The packet erasure coding can also be proactively applied for forward protection from packet loss. - NORM transmissions are governed by TCP-friendly congestion control. - NORM's reliability, congestion control, and flow control mechanism - are distinct components and can be separately controlled to meet - different application needs. + NORM transmissions are governed by the TCP-friendly congestion + control. NORM's reliability, congestion control, and flow control + mechanism are distinct components and can be separately controlled to + meet different application needs. -3.8.1. Protocol Description +3.11.1. Protocol Description [EDITOR'S NOTE: needs to be more clear about the application of FEC and packet erasure coding; expand ARQ.] The NORM protocol is encapsulated in UDP datagrams and thus provides multiplexing for multiple sockets on hosts using port numbers. For purposes of loosely coordinated IP Multicast, NORM is not strictly connection-oriented although per-sender state is maintained by receivers for protocol operation. [RFC5740] does not specify a handshake protocol for connection establishment and separate session @@ -982,89 +1462,87 @@ While NORM is NACK-based for reliability transfer, it also supports a positive acknowledgment (ACK) mechanism that can be used for receiver flow control. Again, since this mechanism is decoupled from the reliability and congestion control, applications that have different needs in this aspect can use the protocol differently. One example is the use of NORM for quasi-reliable delivery where timely delivery of newer content may be favored over completely reliable delivery of older content within buffering and RTT constraints. -3.8.2. Interface Description +3.11.2. Interface Description The NORM specification does not describe a specific application programming interface (API) to control protocol operation. A freely- available, open source reference implementation of NORM is available at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented API is provided for this implementation. While a sockets-like API is not currently documented, the existing API supports the necessary functions for that to be implemented. -3.8.3. Transport Protocol Components - - The transport protocol components provided by NORM are: +3.11.3. Transport Features - o unicast + The transport features provided by NORM are: - o multicast + o unicast or multicast. - o port multiplexing (UDP ports) - o reliable delivery + o stream-oriented delivery in a single stream. - o unordered delivery of in-memory data or file bulk content objects + o object-oriented delivery of discrete data or file items. - o error detection (UDP checksum) + o reliable delivery. - o segmentation + o unordered unidirectional delivery (of in-memory data or file bulk + content objects). - o stream-oriented delivery in a single stream + o error detection (UDP checksum). - o object-oriented delivery of discrete data or file items + o segmentation. - o data bundling (Nagle's algorithm) + o data bundling (Nagle's algorithm). - o flow control (timer-based and/or ack-based) + o flow control (timer-based and/or ack-based). - o congestion control + o congestion control. - o packet erasure coding (both proactively and as part of ARQ) + o packet erasure coding (both proactively and as part of ARQ). -3.9. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a +3.12. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a pseudotransport Transport Layer Security (TLS) and Datagram TLS (DTLS) are IETF protocols that provide several security-related features to applications. TLS is designed to run on top of a reliable streaming transport protocol (usually TCP), while DTLS is designed to run on - top of a best-effort datagram protocol (usually UDP). At the time of - writing, the current version of TLS is 1.2; it is defined in - [RFC5246]. DTLS provides nearly identical functionality to + top of a best-effort datagram protocol (UDP or DCCP [RFC5238]). At + the time of writing, the current version of TLS is 1.2; it is defined + in [RFC5246]. DTLS provides nearly identical functionality to applications; it is defined in [RFC6347] and its current version is also 1.2. The TLS protocol evolved from the Secure Sockets Layer (SSL) protocols developed in the mid 90s to support protection of HTTP traffic. While older versions of TLS and DTLS are still in use, they provide weaker security guarantees. [RFC7457] outlines important attacks on TLS and DTLS. [RFC7525] is a Best Current Practices (BCP) document that describes secure configurations for TLS and DTLS to counter these attacks. The recommendations are applicable for the vast majority of use cases. [NOTE: The Logjam authors (weakdh.org) give (inconclusive) evidence that one of the recommendations of [RFC7525], namely the use of DHE-1024 as a fallback, may not be sufficient in all cases to counter an attacker with the resources of a nation-state. It is unclear at this time if the RFC is going to be updated as a result, or whether there will be an RFC7525bis.] -3.9.1. Protocol Description +3.12.1. Protocol Description Both TLS and DTLS provide the same security features and can thus be discussed together. The features they provide are: o Confidentiality o Data integrity o Peer authentication (optional) @@ -1078,45 +1556,45 @@ should be interpreted. For example, in the common use case of authentication by means of an X.509 certificate, it is the application's decision whether the certificate of the peering entity is acceptable for authorization decisions. Perfect forward secrecy, if enabled and supported by the selected algorithms, ensures that traffic encrypted and captured during a session at time t0 cannot be later decrypted at time t1 (t1 > t0), even if the long-term secrets of the communicating peers are later compromised. As DTLS is generally used over an unreliable datagram transport such - as TCP, applications will need to tolerate loss, re-ordered, or + as UDP, applications will need to tolerate loss, re-ordered, or duplicated datagrams. Like TLS, DTLS conveys application data in a sequence of independent records. However, because records are mapped to unreliable datagrams, there are several features unique to DTLS that are not applicable to TLS: - o Record replay detection (optional) + o Record replay detection (optional). o Record size negotiation (estimates of PMTU and record size - expansion factor) + expansion factor). - o Coveyance of IP don't fragment (DF) bit settings by application + o Coveyance of IP don't fragment (DF) bit settings by application. - o An anti-DoS stateless cookie mechanism (optional) + o An anti-DoS stateless cookie mechanism (optional). Generally, DTLS follows the TLS design as closely as possible. To operate over datagrams, DTLS includes a sequence number and limited forms of retransmission and fragmentation for its internal operations. The sequence number may be used for detecting replayed information, according to the windowing procedure described in - Section 4.1.2.6 of [RFC6347]. Note also that DTLS bans the use of + Section 4.1.2.6 of [RFC6347]. Note also that DTLS forbids the use of stream ciphers, which are essentially incompatible when operating on independent encrypted records. -3.9.2. Interface Description +3.12.2. Interface Description TLS is commonly invoked using an API provided by packages such as OpenSSL, wolfSSL, or GnuTLS. Using such APIs entails the manipulation of several important abstractions, which fall into the following categories: long-term keys and algorithms, session state, and communications/connections. There may also be special APIs required to deal with time and/or random numbers, both of which are needed by a variety of encryption algorithms and protocols. Considerable care is required in the use of TLS APIs in order to @@ -1129,68 +1607,78 @@ As an example, in the case of OpenSSL, the primary abstractions are the library itself and method (protocol), session, context, cipher and connection. After initializing the library and setting the method, a cipher suite is chosen and used to configure a context object. Session objects may then be minted according to the parameters present in a context object and associated with individual connections. Depending on how precisely the programmer wishes to select different algorithmic or protocol options, various levels of details may be required. -3.9.3. Transport Protocol Components +3.12.3. Transport Features Both TLS and DTLS employ a layered architecture. The lower layer is - commonly called the record protocol. It is responsible for - fragmenting messages, applying message authentication codes (MACs), - encrypting data, and invoking transmission from the underlying - transport protocol. DTLS augments the TLS record protocol with - sequence numbers used for ordering and replay detection. + commonly called the record protocol. It is responsible for: + + o message fragmentation + + o authentication and integrity via message authentication codes + (MAC) + + o data encryption + + o scheduling transmission using the underlying transport protocol + + DTLS augments the TLS record protocol with: + + o ordering and replay protection, implemented using sequence + numbers. Several protocols are layered on top of the record protocol. These include the handshake, alert, and change cipher spec protocols. There is also the data protocol, used to carry application traffic. The handshake protocol is used to establish cryptographic and compression parameters when a connection is first set up. In DTLS, this protocol also has a basic fragmentation and retransmission capability and a cookie-like mechanism to resist DoS attacks. (TLS compression is not recommended at present). The alert protocol is used to inform the peer of various conditions, most of which are terminal for the connection. The change cipher spec protocol is used to synchronize changes in cryptographic parameters for each peer. -3.10. Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport +3.13. Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport Hypertext Transfer Protocol (HTTP) is an application-level protocol widely used on the Internet. Version 1.1 of the protocol is specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234] [RFC7235], and version 2 in [RFC7540]. Furthermore, HTTP is used as a substrate for other application-layer protocols. There are various reasons for this practice listed in [RFC3205]; these include being a well-known and well-understood protocol, reusability of existing servers and client libraries, easy use of existing security mechanisms such as HTTP digest authentication [RFC2617] and TLS [RFC5246], the ability of HTTP to traverse firewalls which makes it work with a lot of infrastructure, and cases where a application server often needs to support HTTP anyway. Depending on application's needs, the use of HTTP as a substrate protocol may add complexity and overhead in comparison to a special- - purpose protocol (e.g. HTTP headers, suitability of the HTTP security - model etc.). [RFC3205] address this issues and provides some - guidelines and concerns about the use of HTTP standard port 80 and - 443, the use of HTTP URL scheme and interaction with existing + purpose protocol (e.g. HTTP headers, suitability of the HTTP + security model etc.). [RFC3205] address this issues and provides + some guidelines and concerns about the use of HTTP standard port 80 + and 443, the use of HTTP URL scheme and interaction with existing firewalls, proxies and NATs. Though not strictly bound to TCP, HTTP is almost exclusively run over TCP, and therefore inherits its properties when used in this way. -3.10.1. Protocol Description +3.13.1. Protocol Description Hypertext Transfer Protocol (HTTP) is a request/response protocol. A client sends a request containing a request method, URI and protocol version followed by a MIME-like message (see [RFC7231] for the differences between an HTTP object and a MIME message), containing information about the client and request modifiers. The message can contain a message body carrying application data as well. The server responds with a status or error code followed by a MIME-like message containing information about the server and information about carried data and it can include a message body. It is possible to specify a @@ -1224,21 +1712,21 @@ eliminates addition round-trip. Arbitrary cookie strings, included as part of the MIME headers, are often used as bearer tokens in HTTP. Application layer protocols using HTTP as substrate may use existing method and data formats, or specify new methods and data formats. Furthermore some protocols may not fit a request/response paradigm and instead rely on HTTP to send messages (e.g. [RFC6546]). Because HTTP is working in many restricted infrastructures, it is also used to tunnel other application-layer protocols. -3.10.2. Interface Description +3.13.2. Interface Description There are many HTTP libraries available exposing different APIs. The APIs provide a way to specify a request by providing a URI, a method, request modifiers and optionally a request body. For the response, callbacks can be registered that will be invoked when the response is received. If TLS is used, API expose a registration of callbacks in case a server requests client authentication and when certificate verification is needed. World Wide Web Consortium (W3C) standardized the XMLHttpRequest API @@ -1247,81 +1735,67 @@ response data format can also be JSON, HTML and plain text. Specifically JavaScript and XMLHttpRequest are a ubiquitous programming model for websites, and more general applications, where native code is less attractive. Representational State Transfer (REST) [REST] is another example how applications can use HTTP as transport protocol. REST is an architecture style for building application on the Internet. It uses HTTP as a communication protocol. -3.10.3. Transport Protocol Components +3.13.3. Transport features - The transport protocol components provided by HTTP, when used as a + The transport features provided by HTTP, when used as a pseudotransport, are: - o unicast - - o reliable delivery - - o ordered delivery - - o message and stream-oriented - - o object range request + o unicast. - o message content type negotiation + o message and stream-oriented transfer. - o congestion control + o bi- or unidirectional transmission. - HTTPS (HTTP over TLS) additionally provides the following components: + o ordered delivery. - o authentication (of one or both ends of a connection) + o fully reliable delivery. - o confidentiality + o object range request. - o integrity protection + o message content type negotiation. -3.11. WebSockets + o flow control. - [RFC6455] + HTTPS (HTTP over TLS) additionally provides the following components: - [EDITOR'S NOTE: Salvatore Loreto will contribute text for this - section.] + o authentication (of one or both ends of a connection). -3.11.1. Protocol Description + o confidentiality. -3.11.2. Interface Description -3.11.3. Transport Protocol Components + o integrity protection. 4. Transport Service Features [EDITOR'S NOTE: This section is still work-in-progress. This list is probably not complete and/or too detailed.] The transport protocol components analyzed in this document which can be used as a basis for defining common transport service features, normalized and separated into categories, are as follows: o Control Functions * Addressing + unicast - + broadcast (IPv4 only) - - + multicast - - + anycast + + multicast, anycast and IPv4 broadcast - + something on ports and NAT + + use of NAPT-compatible port numbers * Multihoming support + multihoming for resilience + multihoming for mobility - specify handover latency? + multihoming for load-balancing @@ -1331,34 +1805,34 @@ * Multiplexing + application to port mapping + single vs. multiple streaming o Delivery * reliability - + reliable delivery - + partially reliable delivery + + fully reliable delivery + + partially reliable delivery - packet erasure coding + unreliable delivery - drop notification - Integrity protection o checksum for error detection - o partial checksum protection + o partial payload checksum protection o checksum optional * ordering + ordered delivery + unordered delivery - unordered delivery of in-memory data @@ -1379,39 +1853,39 @@ o Transmission control * rate control + timer-based + ACK-based * congestion control - * flow control + * flow control * segmentation * data/message bundling (Nagle's algorithm) * stream scheduling prioritization o Security * authentication of one end of a connection * authentication of both ends of a connection * confidentiality * cryptographic integrity protection - The next revision of this document will define transport service + A future revision of this document will define transport service features based upon this list. [EDITOR'S NOTE: this section will drawn from the candidate features provided by protocol components in the previous section - please discuss on taps@ietf.org list] 4.1. Complete Protocol Feature Matrix [EDITOR'S NOTE: Dave Thaler has signed up as a contributor for this section. Michael Welzl also has a beginning of a matrix which could @@ -1495,340 +1970,495 @@ o Section 3.2 on MPTCP was contributed by Simone Ferlin-Oliviera (ferlin@simula.no) and Olivier Mehani (olivier.mehani@nicta.com.au) o Section 3.4 on UDP was contributed by Kevin Fall (kfall@kfall.com) o Section 3.3 on SCTP was contributed by Michael Tuexen (tuexen@fh- muenster.de) - o Section 3.8 on NORM was contributed by Brian Adamson + o Section 3.10 on FLUTE/ALC was contributed by Vincent Roca + (vincent.roca@inria.fr) + + o Section 3.11 on NORM was contributed by Brian Adamson (brian.adamson@nrl.navy.mil) - o Section 3.9 on MPTCP was contributed by Ralph Holz + o Section 3.12 on TLS and DTLS was contributed by Ralph Holz (ralph.holz@nicta.com.au) and Olivier Mehani (olivier.mehani@nicta.com.au) - o Section 3.10 on HTTP was contributed by Dragana Damjanovic + o Section 3.13 on HTTP was contributed by Dragana Damjanovic (ddamjanovic@mozilla.com) 8. Acknowledgments Thanks to Karen Nielsen, Joe Touch, and Michael Welzl for the comments, feedback, and discussion. This work is partially supported - by the European Commission under grant agreement FP7-ICT-318627 - mPlane; support does not imply endorsement. - - [EDITOR'S NOTE: add H2020-NEAT ack]. - -9. References -9.1. Normative References + by the European Commission under grant agreements FP7-ICT-318627 + mPlane and from the Horizon 2020 research and innovation program + under grant agreement No. 644334 (NEAT); support does not imply + endorsement. - [RFC0791] Postel, J., "Internet Protocol", STD 5, RFC 791, September - 1981. +9. Informative References -9.2. Informative References + [RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, DOI + 10.17487/RFC0768, August 1980, + . - [RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, - August 1980. + [RFC0792] Postel, J., "Internet Control Message Protocol", STD 5, + RFC 792, DOI 10.17487/RFC0792, September 1981, + . [RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC - 793, September 1981. + 793, DOI 10.17487/RFC0793, September 1981, + . - [RFC0896] Nagle, J., "Congestion control in IP/TCP internetworks", - RFC 896, January 1984. + [RFC0896] Nagle, J., "Congestion Control in IP/TCP Internetworks", + RFC 896, DOI 10.17487/RFC0896, January 1984, + . - [RFC1122] Braden, R., "Requirements for Internet Hosts - - Communication Layers", STD 3, RFC 1122, October 1989. + [RFC1122] Braden, R., Ed., "Requirements for Internet Hosts - + Communication Layers", STD 3, RFC 1122, DOI 10.17487/ + RFC1122, October 1989, + . [RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191, - November 1990. + DOI 10.17487/RFC1191, November 1990, + . + + [RFC1716] Almquist, P. and F. Kastenholz, "Towards Requirements for + IP Routers", RFC 1716, DOI 10.17487/RFC1716, November + 1994, . [RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery - for IP version 6", RFC 1981, August 1996. + for IP version 6", RFC 1981, DOI 10.17487/RFC1981, August + 1996, . [RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP - Selective Acknowledgment Options", RFC 2018, October 1996. + Selective Acknowledgment Options", RFC 2018, DOI 10.17487/ + RFC2018, October 1996, + . [RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail Extensions (MIME) Part One: Format of Internet Message - Bodies", RFC 2045, November 1996. + Bodies", RFC 2045, DOI 10.17487/RFC2045, November 1996, + . [RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6 - (IPv6) Specification", RFC 2460, December 1998. + (IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460, + December 1998, . + + [RFC2461] Narten, T., Nordmark, E., and W. Simpson, "Neighbor + Discovery for IP Version 6 (IPv6)", RFC 2461, DOI + 10.17487/RFC2461, December 1998, + . [RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., Leach, P., Luotonen, A., and L. Stewart, "HTTP Authentication: Basic and Digest Access Authentication", - RFC 2617, June 1999. + RFC 2617, DOI 10.17487/RFC2617, June 1999, + . + + [RFC2710] Deering, S., Fenner, W., and B. Haberman, "Multicast + Listener Discovery (MLD) for IPv6", RFC 2710, DOI + 10.17487/RFC2710, October 1999, + . [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC - 3168, September 2001. + 3168, DOI 10.17487/RFC3168, September 2001, + . [RFC3205] Moore, K., "On the use of HTTP as a Substrate", BCP 56, - RFC 3205, February 2002. - - [RFC3390] Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's - Initial Window", RFC 3390, October 2002. + RFC 3205, DOI 10.17487/RFC3205, February 2002, + . [RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport Layer Security over Stream Control Transmission Protocol", - RFC 3436, December 2002. + RFC 3436, DOI 10.17487/RFC3436, December 2002, + . + + [RFC3450] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J. + Crowcroft, "Asynchronous Layered Coding (ALC) Protocol + Instantiation", RFC 3450, DOI 10.17487/RFC3450, December + 2002, . [RFC3452] Luby, M., Vicisano, L., Gemmell, J., Rizzo, L., Handley, M., and J. Crowcroft, "Forward Error Correction (FEC) - Building Block", RFC 3452, December 2002. + Building Block", RFC 3452, DOI 10.17487/RFC3452, December + 2002, . + + [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. + Jacobson, "RTP: A Transport Protocol for Real-Time + Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, + July 2003, . + + [RFC3738] Luby, M. and V. Goyal, "Wave and Equation Based Rate + Control (WEBRC) Building Block", RFC 3738, DOI 10.17487/ + RFC3738, April 2004, + . [RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. Conrad, "Stream Control Transmission Protocol (SCTP) - Partial Reliability Extension", RFC 3758, May 2004. + Partial Reliability Extension", RFC 3758, DOI 10.17487/ + RFC3758, May 2004, + . - [RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and - G. Fairhurst, "The Lightweight User Datagram Protocol - (UDP-Lite)", RFC 3828, July 2004. + [RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed., + and G. Fairhurst, Ed., "The Lightweight User Datagram + Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July + 2004, . + + [RFC3926] Paila, T., Luby, M., Lehtonen, R., Roca, V., and R. Walsh, + "FLUTE - File Delivery over Unidirectional Transport", RFC + 3926, DOI 10.17487/RFC3926, October 2004, + . + + [RFC3971] Arkko, J., Ed., Kempf, J., Zill, B., and P. Nikander, + "SEcure Neighbor Discovery (SEND)", RFC 3971, DOI + 10.17487/RFC3971, March 2005, + . [RFC4324] Royer, D., Babics, G., and S. Mansour, "Calendar Access - Protocol (CAP)", RFC 4324, December 2005. + Protocol (CAP)", RFC 4324, DOI 10.17487/RFC4324, December + 2005, . [RFC4336] Floyd, S., Handley, M., and E. Kohler, "Problem Statement for the Datagram Congestion Control Protocol (DCCP)", RFC - 4336, March 2006. + 4336, DOI 10.17487/RFC4336, March 2006, + . [RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram - Congestion Control Protocol (DCCP)", RFC 4340, March 2006. + Congestion Control Protocol (DCCP)", RFC 4340, DOI + 10.17487/RFC4340, March 2006, + . [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 2: TCP-like - Congestion Control", RFC 4341, March 2006. + Congestion Control", RFC 4341, DOI 10.17487/RFC4341, March + 2006, . [RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342, - March 2006. + DOI 10.17487/RFC4342, March 2006, + . + + [RFC4433] Kulkarni, M., Patel, A., and K. Leung, "Mobile IPv4 + Dynamic Home Agent (HA) Assignment", RFC 4433, DOI + 10.17487/RFC4433, March 2006, + . [RFC4614] Duke, M., Braden, R., Eddy, W., and E. Blanton, "A Roadmap for Transmission Control Protocol (TCP) Specification - Documents", RFC 4614, September 2006. + Documents", RFC 4614, DOI 10.17487/RFC4614, September + 2006, . [RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast Congestion Control (TFMCC): Protocol Specification", RFC - 4654, August 2006. + 4654, DOI 10.17487/RFC4654, August 2006, + . [RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and Parameter for the Stream Control Transmission Protocol - (SCTP)", RFC 4820, March 2007. + (SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007, + . [RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU - Discovery", RFC 4821, March 2007. + Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007, + . [RFC4895] Tuexen, M., Stewart, R., Lei, P., and E. Rescorla, "Authenticated Chunks for the Stream Control Transmission - Protocol (SCTP)", RFC 4895, August 2007. + Protocol (SCTP)", RFC 4895, DOI 10.17487/RFC4895, August + 2007, . - [RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC - 4960, September 2007. + [RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol", + RFC 4960, DOI 10.17487/RFC4960, September 2007, + . [RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. Kozuka, "Stream Control Transmission Protocol (SCTP) - Dynamic Address Reconfiguration", RFC 5061, September - 2007. + Dynamic Address Reconfiguration", RFC 5061, DOI 10.17487/ + RFC5061, September 2007, + . [RFC5097] Renker, G. and G. Fairhurst, "MIB for the UDP-Lite - protocol", RFC 5097, January 2008. + protocol", RFC 5097, DOI 10.17487/RFC5097, January 2008, + . [RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security - (TLS) Protocol Version 1.2", RFC 5246, August 2008. + (TLS) Protocol Version 1.2", RFC 5246, DOI 10.17487/ + RFC5246, August 2008, + . - [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP - Friendly Rate Control (TFRC): Protocol Specification", RFC - 5348, September 2008. + [RFC5238] Phelan, T., "Datagram Transport Layer Security (DTLS) over + the Datagram Congestion Control Protocol (DCCP)", RFC + 5238, DOI 10.17487/RFC5238, May 2008, + . - [RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines - for Application Designers", BCP 145, RFC 5405, November - 2008. + [RFC5404] Westerlund, M. and I. Johansson, "RTP Payload Format for + G.719", RFC 5404, DOI 10.17487/RFC5404, January 2009, + . + + [RFC5461] Gont, F., "TCP's Reaction to Soft Errors", RFC 5461, DOI + 10.17487/RFC5461, February 2009, + . [RFC5595] Fairhurst, G., "The Datagram Congestion Control Protocol - (DCCP) Service Codes", RFC 5595, September 2009. + (DCCP) Service Codes", RFC 5595, DOI 10.17487/RFC5595, + September 2009, . [RFC5596] Fairhurst, G., "Datagram Congestion Control Protocol (DCCP) Simultaneous-Open Technique to Facilitate NAT/ - Middlebox Traversal", RFC 5596, September 2009. + Middlebox Traversal", RFC 5596, DOI 10.17487/RFC5596, + September 2009, . - [RFC5662] Shepler, S., Eisler, M., and D. Noveck, "Network File - System (NFS) Version 4 Minor Version 1 External Data - Representation Standard (XDR) Description", RFC 5662, - January 2010. + [RFC5651] Luby, M., Watson, M., and L. Vicisano, "Layered Coding + Transport (LCT) Building Block", RFC 5651, DOI 10.17487/ + RFC5651, October 2009, + . - [RFC5672] Crocker, D., "RFC 4871 DomainKeys Identified Mail (DKIM) - Signatures -- Update", RFC 5672, August 2009. + [RFC5662] Shepler, S., Ed., Eisler, M., Ed., and D. Noveck, Ed., + "Network File System (NFS) Version 4 Minor Version 1 + External Data Representation Standard (XDR) Description", + RFC 5662, DOI 10.17487/RFC5662, January 2010, + . + + [RFC5672] Crocker, D., Ed., "RFC 4871 DomainKeys Identified Mail + (DKIM) Signatures -- Update", RFC 5672, DOI 10.17487/ + RFC5672, August 2009, + . [RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker, "NACK-Oriented Reliable Multicast (NORM) Transport - Protocol", RFC 5740, November 2009. - - [RFC6773] Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A - Datagram Congestion Control Protocol UDP Encapsulation for - NAT Traversal", RFC 6773, November 2012. + Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009, + . - [RFC5925] Touch, J., Mankin, A., and R. Bonica, "The TCP - Authentication Option", RFC 5925, June 2010. + [RFC5775] Luby, M., Watson, M., and L. Vicisano, "Asynchronous + Layered Coding (ALC) Protocol Instantiation", RFC 5775, + DOI 10.17487/RFC5775, April 2010, + . [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion - Control", RFC 5681, September 2009. + Control", RFC 5681, DOI 10.17487/RFC5681, September 2009, + . + + [RFC6056] Larsen, M. and F. Gont, "Recommendations for Transport- + Protocol Port Randomization", BCP 156, RFC 6056, DOI + 10.17487/RFC6056, January 2011, + . [RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram Transport Layer Security (DTLS) for Stream Control - Transmission Protocol (SCTP)", RFC 6083, January 2011. + Transmission Protocol (SCTP)", RFC 6083, DOI 10.17487/ + RFC6083, January 2011, + . [RFC6093] Gont, F. and A. Yourtchenko, "On the Implementation of the - TCP Urgent Mechanism", RFC 6093, January 2011. + TCP Urgent Mechanism", RFC 6093, DOI 10.17487/RFC6093, + January 2011, . [RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control Transmission Protocol (SCTP) Stream Reconfiguration", RFC - 6525, February 2012. + 6525, DOI 10.17487/RFC6525, February 2012, + . [RFC6546] Trammell, B., "Transport of Real-time Inter-network - Defense (RID) Messages over HTTP/TLS", RFC 6546, April - 2012. - - [RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent, - "Computing TCP's Retransmission Timer", RFC 6298, June - 2011. + Defense (RID) Messages over HTTP/TLS", RFC 6546, DOI + 10.17487/RFC6546, April 2012, + . [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer - Security Version 1.2", RFC 6347, January 2012. + Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347, + January 2012, . [RFC6356] Raiciu, C., Handley, M., and D. Wischik, "Coupled Congestion Control for Multipath Transport Protocols", RFC - 6356, October 2011. + 6356, DOI 10.17487/RFC6356, October 2011, + . + + [RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error + Correction (FEC) Framework", RFC 6363, DOI 10.17487/ + RFC6363, October 2011, + . [RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC - 6455, December 2011. + 6455, DOI 10.17487/RFC6455, December 2011, + . [RFC6458] Stewart, R., Tuexen, M., Poon, K., Lei, P., and V. Yasevich, "Sockets API Extensions for the Stream Control - Transmission Protocol (SCTP)", RFC 6458, December 2011. + Transmission Protocol (SCTP)", RFC 6458, DOI 10.17487/ + RFC6458, December 2011, + . - [RFC6691] Borman, D., "TCP Options and Maximum Segment Size (MSS)", - RFC 6691, July 2012. + [RFC6584] Roca, V., "Simple Authentication Schemes for the + Asynchronous Layered Coding (ALC) and NACK-Oriented + Reliable Multicast (NORM) Protocols", RFC 6584, DOI + 10.17487/RFC6584, April 2012, + . + + [RFC6726] Paila, T., Walsh, R., Luby, M., Roca, V., and R. Lehtonen, + "FLUTE - File Delivery over Unidirectional Transport", RFC + 6726, DOI 10.17487/RFC6726, November 2012, + . + + [RFC6773] Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A + Datagram Congestion Control Protocol UDP Encapsulation for + NAT Traversal", RFC 6773, DOI 10.17487/RFC6773, November + 2012, . [RFC6824] Ford, A., Raiciu, C., Handley, M., and O. Bonaventure, "TCP Extensions for Multipath Operation with Multiple - Addresses", RFC 6824, January 2013. + Addresses", RFC 6824, DOI 10.17487/RFC6824, January 2013, + . [RFC6897] Scharf, M. and A. Ford, "Multipath TCP (MPTCP) Application - Interface Considerations", RFC 6897, March 2013. + Interface Considerations", RFC 6897, DOI 10.17487/RFC6897, + March 2013, . [RFC6935] Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and - UDP Checksums for Tunneled Packets", RFC 6935, April 2013. + UDP Checksums for Tunneled Packets", RFC 6935, DOI + 10.17487/RFC6935, April 2013, + . [RFC6936] Fairhurst, G. and M. Westerlund, "Applicability Statement for the Use of IPv6 UDP Datagrams with Zero Checksums", - RFC 6936, April 2013. + RFC 6936, DOI 10.17487/RFC6936, April 2013, + . [RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream Control Transmission Protocol (SCTP) Packets for End-Host - to End-Host Communication", RFC 6951, May 2013. + to End-Host Communication", RFC 6951, DOI 10.17487/ + RFC6951, May 2013, + . [RFC7053] Tuexen, M., Ruengeler, I., and R. Stewart, "SACK- IMMEDIATELY Extension for the Stream Control Transmission - Protocol", RFC 7053, November 2013. + Protocol", RFC 7053, DOI 10.17487/RFC7053, November 2013, + . - [RFC7230] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol - (HTTP/1.1): Message Syntax and Routing", RFC 7230, June - 2014. + [RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer + Protocol (HTTP/1.1): Message Syntax and Routing", RFC + 7230, DOI 10.17487/RFC7230, June 2014, + . - [RFC7231] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol - (HTTP/1.1): Semantics and Content", RFC 7231, June 2014. + [RFC7231] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer + Protocol (HTTP/1.1): Semantics and Content", RFC 7231, DOI + 10.17487/RFC7231, June 2014, + . - [RFC7232] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol - (HTTP/1.1): Conditional Requests", RFC 7232, June 2014. + [RFC7232] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer + Protocol (HTTP/1.1): Conditional Requests", RFC 7232, DOI + 10.17487/RFC7232, June 2014, + . - [RFC7233] Fielding, R., Lafon, Y., and J. Reschke, "Hypertext - Transfer Protocol (HTTP/1.1): Range Requests", RFC 7233, - June 2014. + [RFC7233] Fielding, R., Ed., Lafon, Y., Ed., and J. Reschke, Ed., + "Hypertext Transfer Protocol (HTTP/1.1): Range Requests", + RFC 7233, DOI 10.17487/RFC7233, June 2014, + . - [RFC7234] Fielding, R., Nottingham, M., and J. Reschke, "Hypertext - Transfer Protocol (HTTP/1.1): Caching", RFC 7234, June - 2014. + [RFC7234] Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke, + Ed., "Hypertext Transfer Protocol (HTTP/1.1): Caching", + RFC 7234, DOI 10.17487/RFC7234, June 2014, + . - [RFC7235] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol - (HTTP/1.1): Authentication", RFC 7235, June 2014. + [RFC7235] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer + Protocol (HTTP/1.1): Authentication", RFC 7235, DOI + 10.17487/RFC7235, June 2014, + . [RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol - Negotiation Extension", RFC 7301, July 2014. + Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, + July 2014, . [RFC7323] Borman, D., Braden, B., Jacobson, V., and R. - Scheffenegger, "TCP Extensions for High Performance", RFC - 7323, September 2014. + Scheffenegger, Ed., "TCP Extensions for High Performance", + RFC 7323, DOI 10.17487/RFC7323, September 2014, + . [RFC7457] Sheffer, Y., Holz, R., and P. Saint-Andre, "Summarizing Known Attacks on Transport Layer Security (TLS) and - Datagram TLS (DTLS)", RFC 7457, February 2015. + Datagram TLS (DTLS)", RFC 7457, DOI 10.17487/RFC7457, + February 2015, . + + [RFC7496] Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, + "Additional Policies for the Partially Reliable Stream + Control Transmission Protocol Extension", RFC 7496, DOI + 10.17487/RFC7496, April 2015, + . [RFC7525] Sheffer, Y., Holz, R., and P. Saint-Andre, "Recommendations for Secure Use of Transport Layer Security (TLS) and Datagram Transport Layer Security - (DTLS)", BCP 195, RFC 7525, May 2015. + (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May + 2015, . - [RFC7540] Belshe, M., Peon, R., and M. Thomson, "Hypertext Transfer - Protocol Version 2 (HTTP/2)", RFC 7540, May 2015. + [RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext + Transfer Protocol Version 2 (HTTP/2)", RFC 7540, DOI + 10.17487/RFC7540, May 2015, + . + + [I-D.ietf-tsvwg-rfc5405bis] + Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage + Guidelines", draft-ietf-tsvwg-rfc5405bis-05 (work in + progress), August 2015. [I-D.ietf-aqm-ecn-benefits] Fairhurst, G. and M. Welzl, "The Benefits of using Explicit Congestion Notification (ECN)", draft-ietf-aqm- - ecn-benefits-05 (work in progress), June 2015. + ecn-benefits-06 (work in progress), July 2015. [I-D.ietf-tsvwg-sctp-dtls-encaps] Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- dtls-encaps-09 (work in progress), January 2015. - [I-D.ietf-tsvwg-sctp-prpolicies] - Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, - "Additional Policies for the Partial Reliability Extension - of the Stream Control Transmission Protocol", draft-ietf- - tsvwg-sctp-prpolicies-07 (work in progress), February - 2015. - [I-D.ietf-tsvwg-sctp-ndata] Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "Stream Schedulers and User Message Interleaving for the Stream Control Transmission Protocol", draft-ietf-tsvwg- - sctp-ndata-03 (work in progress), March 2015. + sctp-ndata-04 (work in progress), July 2015. [I-D.ietf-tsvwg-natsupp] Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control Transmission Protocol (SCTP) Network Address Translation - Support", draft-ietf-tsvwg-natsupp-07 (work in progress), - February 2015. + Support", draft-ietf-tsvwg-natsupp-08 (work in progress), + July 2015. [XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen, "XMLHttpRequest working draft (http://www.w3.org/TR/XMLHttpRequest/)", 2000. [REST] Fielding, R., "Architectural Styles and the Design of - Network-based Software Architectures, Ph. D. (UC Irvune), + Network-based Software Architectures, Ph. D. (UC Irvine), Chapter 5: Representational State Transfer", 2000. + [POSIX] 1-2008, IEEE., "IEEE Standard for Information Technology + -- Portable Operating System Interface (POSIX) Base + Specifications, Issue 7", n.d.. + + [MBMS] 3GPP TSG WS S4, ., "3GPP TS 26.346: Multimedia Broadcast/ + Multicast Service (MBMS); Protocols and codecs, release 13 + (http://www.3gpp.org/DynaReport/26346.htm).", 2015. + Authors' Addresses Godred Fairhurst (editor) University of Aberdeen School of Engineering, Fraser Noble Building Aberdeen AB24 3UE Email: gorry@erg.abdn.ac.uk - Brian Trammell (editor) ETH Zurich Gloriastrasse 35 8092 Zurich Switzerland Email: ietf@trammell.ch Mirja Kuehlewind (editor) ETH Zurich