Internet Engineering Task Force (IETF)                 G. Fairhurst, Ed.
Request for Comments: 8095                        University of Aberdeen
Category: Informational                                 B. Trammell, Ed.
ISSN: 2070-1721                                       M. Kuehlewind, Ed.
                                                              ETH Zurich
                                                              March 2017
Internet Engineering Task Force (IETF)                 G. Fairhurst, Ed.
Request for Comments: 8095                        University of Aberdeen
Category: Informational                                 B. Trammell, Ed.
ISSN: 2070-1721                                       M. Kuehlewind, Ed.
                                                              ETH Zurich
                                                              March 2017

Services Provided by IETF Transport Protocols and Congestion Control Mechanisms




This document describes, surveys, and classifies the protocol mechanisms provided by existing IETF protocols, as background for determining a common set of transport services. It examines the Transmission Control Protocol (TCP), Multipath TCP, the Stream Control Transmission Protocol (SCTP), the User Datagram Protocol (UDP), UDP-Lite, the Datagram Congestion Control Protocol (DCCP), the Internet Control Message Protocol (ICMP), the Real-Time Transport Protocol (RTP), File Delivery over Unidirectional Transport / Asynchronous Layered Coding (FLUTE/ALC) for Reliable Multicast, NACK-Oriented Reliable Multicast (NORM), Transport Layer Security (TLS), Datagram TLS (DTLS), and the Hypertext Transport Protocol (HTTP), when HTTP is used as a pseudotransport. This survey provides background for the definition of transport services within the TAPS working group.

本文档描述、调查和分类现有IETF协议提供的协议机制,作为确定公共传输服务集的背景。它检查了传输控制协议(TCP)、多路径TCP、流控制传输协议(SCTP)、用户数据报协议(UDP)、UDP Lite、数据报拥塞控制协议(DCCP)、Internet控制消息协议(ICMP)、实时传输协议(RTP),当HTTP用作伪传输时,通过单向传输/异步分层编码(FLUTE/ALC)进行文件传输,用于可靠多播、面向NACK的可靠多播(NORM)、传输层安全(TLS)、数据报TLS(DTLS)和超文本传输协议(HTTP)。这项调查为TAPS工作组内运输服务的定义提供了背景资料。

Status of This Memo


This document is not an Internet Standards Track specification; it is published for informational purposes.


This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 7841.

本文件是互联网工程任务组(IETF)的产品。它代表了IETF社区的共识。它已经接受了公众审查,并已被互联网工程指导小组(IESG)批准出版。并非IESG批准的所有文件都适用于任何级别的互联网标准;见RFC 7841第2节。

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at


Copyright Notice


Copyright (c) 2017 IETF Trust and the persons identified as the document authors. All rights reserved.

版权所有(c)2017 IETF信托基金和确定为文件作者的人员。版权所有。

This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents ( 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.

本文件受BCP 78和IETF信托有关IETF文件的法律规定的约束(自本文件出版之日起生效。请仔细阅读这些文件,因为它们描述了您对本文件的权利和限制。从本文件中提取的代码组件必须包括信托法律条款第4.e节中所述的简化BSD许可证文本,并提供简化BSD许可证中所述的无担保。

Table of Contents


   1. Introduction ....................................................4
      1.1. Overview of Transport Features .............................4
   2. Terminology .....................................................5
   3. Existing Transport Protocols ....................................6
      3.1. Transport Control Protocol (TCP) ...........................6
           3.1.1. Protocol Description ................................6
           3.1.2. Interface Description ...............................8
           3.1.3. Transport Features ..................................9
      3.2. Multipath TCP (MPTCP) .....................................10
           3.2.1. Protocol Description ...............................10
           3.2.2. Interface Description ..............................10
           3.2.3. Transport Features .................................11
      3.3. User Datagram Protocol (UDP) ..............................11
           3.3.1. Protocol Description ...............................11
           3.3.2. Interface Description ..............................12
           3.3.3. Transport Features .................................13
      3.4. Lightweight User Datagram Protocol (UDP-Lite) .............13
           3.4.1. Protocol Description ...............................13
           3.4.2. Interface Description ..............................14
           3.4.3. Transport Features .................................14
      3.5. Stream Control Transmission Protocol (SCTP) ...............14
           3.5.1. Protocol Description ...............................15
           3.5.2. Interface Description ..............................17
           3.5.3. Transport Features .................................19
      3.6. Datagram Congestion Control Protocol (DCCP) ...............20
           3.6.1. Protocol Description ...............................21
           3.6.2. Interface Description ..............................22
           3.6.3. Transport Features .................................22
   1. Introduction ....................................................4
      1.1. Overview of Transport Features .............................4
   2. Terminology .....................................................5
   3. Existing Transport Protocols ....................................6
      3.1. Transport Control Protocol (TCP) ...........................6
           3.1.1. Protocol Description ................................6
           3.1.2. Interface Description ...............................8
           3.1.3. Transport Features ..................................9
      3.2. Multipath TCP (MPTCP) .....................................10
           3.2.1. Protocol Description ...............................10
           3.2.2. Interface Description ..............................10
           3.2.3. Transport Features .................................11
      3.3. User Datagram Protocol (UDP) ..............................11
           3.3.1. Protocol Description ...............................11
           3.3.2. Interface Description ..............................12
           3.3.3. Transport Features .................................13
      3.4. Lightweight User Datagram Protocol (UDP-Lite) .............13
           3.4.1. Protocol Description ...............................13
           3.4.2. Interface Description ..............................14
           3.4.3. Transport Features .................................14
      3.5. Stream Control Transmission Protocol (SCTP) ...............14
           3.5.1. Protocol Description ...............................15
           3.5.2. Interface Description ..............................17
           3.5.3. Transport Features .................................19
      3.6. Datagram Congestion Control Protocol (DCCP) ...............20
           3.6.1. Protocol Description ...............................21
           3.6.2. Interface Description ..............................22
           3.6.3. Transport Features .................................22
      3.7. Transport Layer Security (TLS) and Datagram TLS
           (DTLS) as a Pseudotransport ...............................23
           3.7.1. Protocol Description ...............................23
           3.7.2. Interface Description ..............................24
           3.7.3. Transport Features .................................25
      3.8. Real-Time Transport Protocol (RTP) ........................26
           3.8.1. Protocol Description ...............................26
           3.8.2. Interface Description ..............................27
           3.8.3. Transport Features .................................27
      3.9. Hypertext Transport Protocol (HTTP) over TCP as a
           Pseudotransport ...........................................28
           3.9.1. Protocol Description ...............................28
           3.9.2. Interface Description ..............................29
           3.9.3. Transport Features .................................30
      3.10. File Delivery over Unidirectional Transport /
            Asynchronous Layered Coding (FLUTE/ALC) for
            Reliable Multicast .......................................31
           3.10.1. Protocol Description ..............................31
           3.10.2. Interface Description .............................33
           3.10.3. Transport Features ................................33
      3.11. NACK-Oriented Reliable Multicast (NORM) ..................34
           3.11.1. Protocol Description ..............................34
           3.11.2. Interface Description .............................35
           3.11.3. Transport Features ................................36
      3.12. Internet Control Message Protocol (ICMP) .................36
           3.12.1. Protocol Description ..............................37
           3.12.2. Interface Description .............................37
           3.12.3. Transport Features ................................38
   4. Congestion Control .............................................38
   5. Transport Features .............................................39
   6. IANA Considerations ............................................42
   7. Security Considerations ........................................42
   8. Informative References .........................................42
   Acknowledgments ...................................................53
   Contributors ......................................................53
   Authors' Addresses ................................................54
      3.7. Transport Layer Security (TLS) and Datagram TLS
           (DTLS) as a Pseudotransport ...............................23
           3.7.1. Protocol Description ...............................23
           3.7.2. Interface Description ..............................24
           3.7.3. Transport Features .................................25
      3.8. Real-Time Transport Protocol (RTP) ........................26
           3.8.1. Protocol Description ...............................26
           3.8.2. Interface Description ..............................27
           3.8.3. Transport Features .................................27
      3.9. Hypertext Transport Protocol (HTTP) over TCP as a
           Pseudotransport ...........................................28
           3.9.1. Protocol Description ...............................28
           3.9.2. Interface Description ..............................29
           3.9.3. Transport Features .................................30
      3.10. File Delivery over Unidirectional Transport /
            Asynchronous Layered Coding (FLUTE/ALC) for
            Reliable Multicast .......................................31
           3.10.1. Protocol Description ..............................31
           3.10.2. Interface Description .............................33
           3.10.3. Transport Features ................................33
      3.11. NACK-Oriented Reliable Multicast (NORM) ..................34
           3.11.1. Protocol Description ..............................34
           3.11.2. Interface Description .............................35
           3.11.3. Transport Features ................................36
      3.12. Internet Control Message Protocol (ICMP) .................36
           3.12.1. Protocol Description ..............................37
           3.12.2. Interface Description .............................37
           3.12.3. Transport Features ................................38
   4. Congestion Control .............................................38
   5. Transport Features .............................................39
   6. IANA Considerations ............................................42
   7. Security Considerations ........................................42
   8. Informative References .........................................42
   Acknowledgments ...................................................53
   Contributors ......................................................53
   Authors' Addresses ................................................54
1. Introduction
1. 介绍

Internet applications make use of the services provided by a transport protocol, such as 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, and integrity protection.


The IETF has defined a wide variety of transport protocols beyond TCP and UDP, including SCTP, DCCP, MPTCP, 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 NewReno [RFC6582], TCP-Friendly Rate Control (TFRC) [RFC5348], or Low Extra Delay Background Transport (LEDBAT) [RFC6817]). This extends the set of available transport services beyond those provided to applications by TCP and UDP.

IETF定义了TCP和UDP之外的多种传输协议,包括SCTP、DCCP、MPTCP和UDP Lite。传输服务可以由这些传输协议直接提供,也可以使用诸如WebSockets(通过TCP运行)、RTP(通过TCP或UDP运行)或WebRTC数据通道(通过UDP或TCP通过DTL通过SCTP运行)等协议在传输协议之上分层。构建在UDP或UDP Lite之上的服务通常还需要指定其他机制,包括拥塞控制机制(如NewReno[RFC6582]、TCP友好速率控制(TFRC)[RFC5348]或低额外延迟后台传输(LEDBAT)[RFC6817])。这将扩展可用传输服务集,使其超出TCP和UDP提供给应用程序的传输服务集。

The transport protocols described in this document provide a basis for the definition of transport services provided by common protocols, as background for the TAPS working group. The protocols listed here were chosen to help expose as many potential transport services as possible and are not meant to be a comprehensive survey or classification of all transport protocols.


1.1. Overview of Transport Features
1.1. 运输功能概述

Transport protocols can be differentiated by the features of the services they provide.


Some of these provided features are closely related to basic control function that a protocol needs to work over a network path, such as addressing. The number of participants in a given association also determines its applicability: a connection can be between endpoints (unicast), to one of multiple endpoints (anycast), or simultaneously to multiple endpoints (multicast). Unicast protocols usually support bidirectional communication, while multicast is generally


unidirectional. Another feature is whether a transport requires a control exchange across the network at setup (e.g., TCP) or whether it is connectionless (e.g., UDP).


For packet delivery itself, reliability and integrity protection, ordering, and framing are basic features. However, these features are implemented with different levels of assurance in different protocols. As an example, a transport service may provide full reliability, with detection of loss and retransmission (e.g., TCP). SCTP offers a message-based service that can provide full or partial reliability and allows the protocol to minimize the head-of-line blocking due to the support of ordered and unordered message delivery within multiple streams. UDP-Lite and DCCP can provide partial integrity protection to enable corruption tolerance.

对于数据包交付本身,可靠性和完整性保护、排序和帧是基本特性。但是,这些功能在不同协议中以不同的保证级别实现。例如,传输服务可以提供完全的可靠性,并检测丢失和重传(例如,TCP)。SCTP提供了一种基于消息的服务,它可以提供完全或部分可靠性,并允许协议由于支持多个流中的有序和无序消息传递而最小化行首阻塞。UDP Lite和DCCP可以提供部分完整性保护,以实现容错性。

Usually, a protocol has been designed to support one specific type of delivery/framing: either data needs to be divided into transmission units based on network packets (datagram service) or a data stream is segmented and re-combined across multiple packets (stream service). Whole objects such as files are handled accordingly. This decision strongly influences the interface that is provided to the upper layer.


In addition, transport protocols offer a certain support for transmission control. For example, a transport service can provide flow control to allow a receiver to regulate the transmission rate of a sender. Further, a transport service can provide congestion control (see Section 4). As an example, TCP and SCTP provide congestion control for use in the Internet, whereas UDP leaves this function to the upper-layer protocol that uses UDP.


Security features are often provided independently of the transport protocol, via Transport Layer Security (TLS) (see Section 3.7) or by the application-layer protocol itself. The security properties TLS provides to the application (such as confidentiality, integrity, and authenticity) are also features of the transport layer, even though they are often presently implemented in a separate protocol.


2. Terminology
2. 术语

The following terms are used throughout this document and in subsequent documents produced by the TAPS working group that describe the composition and decomposition of transport services.


Transport Feature: a specific end-to-end feature that the transport layer provides to an application. Examples include confidentiality, reliable delivery, ordered delivery, message-versus-stream orientation, etc.


Transport Service: a set of transport features, without an association to any given framing protocol, that provides a complete service to an application.


Transport Protocol: an implementation that provides one or more different transport services using a specific framing and header format on the wire.


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
3. 现有的传输协议

This section provides a list of known IETF transport protocols and transport protocol frameworks. It does not make an assessment about whether specific implementations of protocols are fully compliant to current IETF specifications.


3.1. Transport Control Protocol (TCP)
3.1. 传输控制协议(TCP)

TCP is an IETF Standards Track transport protocol. [RFC793] 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
3.1.1. 协议描述

TCP is a connection-oriented protocol that provides a three-way handshake to allow a client and server to set up a connection and negotiate features and provides mechanisms for orderly completion and immediate teardown of a connection [RFC793] [TCP-SPEC]. TCP is defined by a family of RFCs (see [RFC7414]).


TCP provides multiplexing to multiple sockets on each host using port 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 and remote port numbers. 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 based on a negotiated maximum segment size and further constrained by the effective Maximum Transmission Unit (MTU) from Path MTU Discovery (PMTUD). ICMP-based PMTUD [RFC1191] [RFC1981] as well as Packetization Layer PMTUD (PLPMTUD) [RFC4821] have been defined by the IETF.


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 the reliable, ordered delivery of data in a TCP stream. TCP Selective Acknowledgment (SACK) [RFC2018] extends this mechanism by making it possible to provide earlier identification of which segments are missing, allowing faster retransmission. SACK-based methods (e.g., Duplicate Selective ACK) can also result in less spurious retransmission.


Receiver flow control is provided by a sliding window, which limits 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 64 KB.

接收器流量控制由滑动窗口提供,该窗口限制在给定时间未确认的数据量。窗口缩放选项[RFC7323]允许接收器使用大于64 KB的窗口。

All TCP senders provide congestion control, such as that described in [RFC5681]. TCP uses a sequence number with a sliding receiver window for flow control. The TCP congestion control mechanism also utilizes this TCP sequence number to manage a separate congestion window [RFC5681]. The sending window at a given point in time is the minimum of the receiver window and the congestion window. The congestion window is increased in the absence of congestion and decreased if congestion is detected. Often, loss is implicitly handled as a congestion indication, which is detected in TCP (also as input for retransmission handling) based on two mechanisms: a retransmission timer with exponential back-off or the reception of three acknowledgments for the same segment, so called "duplicated ACKs" (fast retransmit). In addition, Explicit Congestion Notification (ECN) [RFC3168] can be used in TCP and, if supported by both endpoints, allows a network node to signal congestion without inducing loss. Alternatively, a delay-based congestion control scheme that reacts to changes in delay as an early indication of congestion can be used in TCP. This is further described in Section 4. Examples of different kinds of congestion control schemes are provided in Section 4.


TCP protocol instances can be extended (see [RFC7414]). Some protocol features may also be tuned to optimize for a specific deployment scenario. Some features are sender-side only, requiring no negotiation with the receiver; some are receiver-side only; and some are explicitly negotiated during connection setup.


TCP may buffer data, e.g., to optimize processing or capacity usage. TCP therefore provides mechanisms to control this, including an optional "PUSH" function [RFC793] that explicitly requests the transport service not to delay data. By default, TCP segment partitioning uses Nagle's algorithm [TCP-SPEC] 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 reduce latency for interactive sessions.


TCP provides an "urgent data" function for limited out-of-order delivery of the data. This function is deprecated [RFC6093].


A TCP Reset (RST) control message may be used to force a TCP endpoint to close a session [RFC793], aborting the connection.


A mandatory checksum provides a basic integrity check against misdelivery and data corruption over the entire packet. Applications that require end-to-end integrity of data are recommended to include a stronger integrity check of their payload data. The TCP checksum [RFC1071] [RFC2460] does not support partial payload protection (as in DCCP/UDP-Lite).

强制校验和提供了基本的完整性检查,以防止整个数据包上的错误传递和数据损坏。建议需要端到端数据完整性的应用程序对其有效负载数据进行更严格的完整性检查。TCP校验和[RFC1071][RFC2460]不支持部分有效负载保护(如在DCCP/UDP Lite中)。

TCP supports only unicast connections.


3.1.2. Interface Description
3.1.2. 接口描述

The User/TCP Interface defined in [RFC793] provides six user commands: Open, Send, Receive, Close, Status, and Abort. This interface does not describe configuration of TCP options or parameters aside from the use of the PUSH and URGENT flags.


[RFC1122] describes extensions of the TCP/application-layer interface for:


o reporting soft errors such as reception of ICMP error messages, extensive retransmission, or urgent pointer advance,

o 报告软错误,例如接收ICMP错误消息、大量重新传输或紧急指针提前,

o providing a possibility to specify the Differentiated Services Code Point (DSCP) [RFC3260] (formerly, the Type-of-Service (TOS)) for segments,

o 提供了为段指定区分服务代码点(DSCP)[RFC3260](以前称为服务类型(TOS))的可能性,

o providing a flush call to empty the TCP send queue, and

o 提供刷新调用以清空TCP发送队列,以及

o multihoming support.

o 多归宿支持。

In API implementations derived from the BSD Sockets API, TCP sockets 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. There are currently no documents in the RFC Series that describe this interface.


3.1.3. Transport Features
3.1.3. 运输特征

The transport features provided by TCP are:


o connection-oriented transport with feature negotiation and application-to-port mapping (implemented using SYN segments and the TCP Option field to negotiate features),

o 具有功能协商和应用程序到端口映射(使用SYN段和TCP选项字段协商功能)的面向连接的传输,

o unicast transport (though anycast TCP is implemented, at risk of instability due to rerouting),

o 单播传输(尽管实现了选播TCP,但存在因重新路由而导致不稳定的风险),

o port multiplexing,

o 端口多路复用,

o unidirectional or bidirectional communication,

o 单向或双向通信,

o stream-oriented delivery in a single stream,

o 在单个流中面向流的交付,

o fully reliable delivery (implemented using ACKs sent from the receiver to confirm delivery),

o 完全可靠的交付(使用从接收方发送的确认交付的ACK实施),

o error detection (implemented using a segment checksum to verify delivery to the correct endpoint and integrity of the data and options),

o 错误检测(使用段校验和来验证向正确端点的传递以及数据和选项的完整性),

o segmentation,

o 分段,

o data bundling (optional; uses Nagle's algorithm to coalesce data sent within the same RTT into full-sized segments),

o 数据绑定(可选;使用Nagle算法将同一RTT内发送的数据合并为全尺寸段),

o flow control (implemented using a window-based mechanism where the receiver advertises the window that it is willing to buffer), and

o 流量控制(使用基于窗口的机制实现,其中接收方播发其愿意缓冲的窗口),以及

o congestion control (usually implemented using a window-based mechanism and four algorithms for different phases of the transmission: slow start, congestion avoidance, fast retransmit, and fast recovery [RFC5681]).

o 拥塞控制(通常使用基于窗口的机制和四种算法来实现传输的不同阶段:慢启动、拥塞避免、快速重传和快速恢复[RFC5681])。

3.2. Multipath TCP (MPTCP)
3.2. 多路径TCP(MPTCP)

Multipath TCP [RFC6824] is an extension for TCP to support multihoming for resilience, mobility, and load balancing. It is designed to be as indistinguishable to middleboxes from non-multipath TCP as possible. It does so by establishing regular TCP flows between a pair of source/destination endpoints and multiplexing the application's stream over these flows. Sub-flows can be started over IPv4 or IPv6 for the same session.


3.2.1. Protocol Description
3.2.1. 协议描述

MPTCP uses TCP options for its control plane. They are used to signal multipath capabilities, as well as to negotiate data sequence numbers, advertise other available IP addresses, and establish new sessions between pairs of endpoints.


By multiplexing one byte stream over separate paths, MPTCP can achieve a higher throughput than TCP in certain situations. However, if coupled congestion control [RFC6356] is used, it 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, additional delay and higher jitter might be observed before in-order delivery of data to the applications.


3.2.2. Interface Description
3.2.2. 接口描述

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:


o enable or disable MPTCP.

o 启用或禁用MPTCP。

o bind a socket to one or more selected local endpoints.

o 将套接字绑定到一个或多个选定的本地端点。

o query local and remote endpoint addresses.

o 查询本地和远程端点地址。

o get a unique connection identifier (similar to an address-port pair for TCP).

o 获取唯一的连接标识符(类似于TCP的地址端口对)。

The document also recommends the use of extensions defined for SCTP [RFC6458] (see Section 3.5) to support multihoming for resilience and mobility.


3.2.3. Transport Features
3.2.3. 运输特征

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 in the case that one of the paths becomes unusable.


Therefore, the transport features provided by MPTCP in addition to TCP are:


o multihoming for load balancing, with endpoint multiplexing of a single byte stream, using either coupled congestion control or throughput maximization,

o 使用耦合拥塞控制或吞吐量最大化,通过单字节流的端点多路复用实现负载平衡的多宿主,

o address family multiplexing (using IPv4 and IPv6 for the same session), and

o 地址族多路复用(在同一会话中使用IPv4和IPv6),以及

o resilience to network failure and/or handover.

o 对网络故障和/或切换的恢复能力。

3.3. User Datagram Protocol (UDP)
3.3. 用户数据报协议(UDP)

The User Datagram Protocol (UDP) [RFC768] [RFC2460] is an IETF Standards Track transport protocol. It provides a unidirectional datagram protocol that preserves message boundaries. It provides no 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 [RFC8085]. UDP is widely implemented and widely used by common applications, including DNS.


3.3.1. Protocol Description
3.3.1. 协议描述

UDP is a connectionless protocol that maintains message boundaries, with no connection setup or feature negotiation. The protocol uses independent messages, ordinarily called "datagrams". It provides detection of payload errors and misdelivery of packets to an unintended endpoint, both of which result in discard of received datagrams, with no indication to the user of the service.


It is possible to create IPv4 UDP datagrams with no checksum, and while this is generally discouraged [RFC1122] [RFC8085], certain special cases permit this use. These datagrams rely on the IPv4 header checksum to protect from misdelivery to an unintended endpoint. IPv6 does not permit UDP datagrams with no checksum, although in certain cases [RFC6936], this rule may be relaxed [RFC6935].

可以创建不带校验和的IPv4 UDP数据报,虽然通常不建议这样做[RFC1122][RFC8085],但某些特殊情况允许这样做。这些数据报依靠IPv4报头校验和来防止误发到非预期的端点。IPv6不允许没有校验和的UDP数据报,尽管在某些情况下[RFC6936],此规则可能会放宽[RFC6935]。

UDP does not provide reliability and does not provide retransmission. Messages may be reordered, lost, or duplicated in transit. 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.


Because UDP provides no flow control, a receiving application that is unable to run sufficiently fast, or frequently, may miss messages. The lack of congestion handling implies UDP traffic may experience loss when using an overloaded path and may cause the loss of messages from other protocols (e.g., TCP) when sharing the same network path.


On transmission, UDP encapsulates each datagram into a single IP packet or several IP packet fragments. This allows a datagram to be larger than the effective path MTU. Fragments are reassembled before delivery to the UDP receiver, making this transparent to the user of the transport service. When jumbograms are supported, larger messages may be sent without performing fragmentation.


UDP on its own does not provide support for segmentation, receiver flow control, congestion control, PMTUD/PLPMTUD, or ECN. Applications that require these features need to provide them on their own or use a protocol over UDP that provides them [RFC8085].


3.3.2. Interface Description
3.3.2. 接口描述

[RFC768] describes basic requirements for an API for UDP. Guidance on the use of common APIs is provided in [RFC8085].

[RFC768]描述了UDP API的基本要求。[RFC8085]中提供了通用API的使用指南。

A UDP endpoint consists of a tuple of (IP address, port number). De-multiplexing using multiple abstract endpoints (sockets) on the same IP address is 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) that bind to 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 [RFC8085].


3.3.3. Transport Features
3.3.3. 运输特征

The transport features provided by UDP are:


o unicast, multicast, anycast, or IPv4 broadcast transport,

o 单播、多播、选播或IPv4广播传输,

o port multiplexing (where a receiving port can be configured to receive datagrams from multiple senders),

o 端口多路复用(接收端口可配置为从多个发送方接收数据报),

o message-oriented delivery,

o 以信息为导向的交付,

o unidirectional or bidirectional communication where the transmissions in each direction are independent,

o 单向或双向通信,其中每个方向上的传输是独立的,

o non-reliable delivery,

o 不可靠的交付,

o unordered delivery, and

o 无序交货,以及

o error detection (implemented using a segment checksum to verify delivery to the correct endpoint and integrity of the data; optional for IPv4 and optional under specific conditions for IPv6 where all or none of the payload data is protected).

o 错误检测(使用段校验和来验证到正确端点的传递和数据的完整性;对于IPv4是可选的,在所有或没有有效负载数据受到保护的IPv6的特定条件下是可选的)。

3.4. Lightweight User Datagram Protocol (UDP-Lite)
3.4. 轻量级用户数据报协议(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 [RFC8085]. A UDP-Lite service may support IPv4 broadcast, multicast, anycast, and unicast, as well as IPv6 multicast, anycast, and unicast.

轻量级用户数据报协议(UDP Lite)[RFC3828]是IETF标准的轨道传输协议。它提供了一个单向的数据报协议,可以保留消息边界。[RFC8085]中提供了有关UDP Lite使用的IETF指南。UDP Lite服务可以支持IPv4广播、多播、选播和单播,以及IPv6多播、选播和单播。

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 provide header integrity checks but allow delivery of corrupted payloads to error-tolerant applications or to applications that use some other mechanism to provide payload integrity (see [RFC6936]).


3.4.1. Protocol Description
3.4.1. 协议描述

Like UDP, UDP-Lite is a connectionless datagram protocol, with no connection setup or feature negotiation. It 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. The Checksum Coverage Length field specifies the intended checksum coverage, with the remaining unprotected part of the payload called

与UDP一样,UDP Lite是一种无连接的数据报协议,没有连接设置或功能协商。它将UDP有效负载长度字段的语义更改为校验和覆盖长度字段的语义,并由不同的IP协议/下一个报头值标识。校验和覆盖范围长度字段指定预期的校验和覆盖范围,有效负载的剩余未受保护部分称为

the "error-insensitive part". Therefore, applications using UDP-Lite cannot make assumptions regarding the correctness of the data received in the insensitive part of the UDP-Lite payload.

“错误不敏感部分”。因此,使用UDP Lite的应用程序无法对UDP Lite有效负载的不敏感部分中接收的数据的正确性做出假设。

Otherwise, UDP-Lite is semantically identical to UDP. In the same way 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 [RFC8085].

否则,UDP Lite在语义上与UDP相同。与UDP相同,上层协议需要提供接收方流量控制、拥塞控制、PMTU或PLPMTU发现、ECN支持等机制[RFC8085]。

3.4.2. Interface Description
3.4.2. 接口描述

There is no API currently specified in the RFC Series, but guidance on use of common APIs is provided in [RFC8085].


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. The checksum coverage may also be made visible to the application via the UDP-Lite MIB module [RFC5097].

UDP Lite的接口与UDP的接口不同,它增加了一个(套接字)选项,用于传递校验和覆盖长度值。校验和覆盖范围也可以通过UDP Lite MIB模块[RFC5097]使应用程序可见。

3.4.3. Transport Features
3.4.3. 运输特征

The transport features provided by UDP-Lite are:

UDP Lite提供的传输功能包括:

o unicast, multicast, anycast, or IPv4 broadcast transport (same as for UDP),

o 单播、多播、选播或IPv4广播传输(与UDP相同),

o port multiplexing (same as for UDP),

o 端口多路复用(与UDP相同),

o message-oriented delivery (same as for UDP),

o 面向消息的传递(与UDP相同),

o unidirectional or bidirectional communication where the transmissions in each direction are independent (same as for UDP),

o 单向或双向通信,其中每个方向上的传输是独立的(与UDP相同),

o non-reliable delivery (same as for UDP),

o 不可靠传递(与UDP相同),

o non-ordered delivery (same as for UDP), and

o 非订单交付(与UDP相同),以及

o partial or full payload error detection (where the Checksum Coverage field indicates the size of the payload data covered by the checksum).

o 部分或全部有效负载错误检测(其中校验和覆盖率字段指示校验和覆盖的有效负载数据的大小)。

3.5. Stream Control Transmission Protocol (SCTP)
3.5. 流控制传输协议(SCTP)

SCTP is a message-oriented IETF Standards Track transport protocol. The base protocol is specified in [RFC4960]. It supports multihoming and path failover to provide resilience to path failures. An SCTP association has multiple streams in each direction, providing in-sequence delivery of user messages within each stream. This


allows it to minimize head-of-line blocking. SCTP supports multiple stream- scheduling schemes controlling stream multiplexing, including priority and fair weighting schemes.


SCTP was originally developed for transporting telephony signaling messages and is deployed in telephony signaling networks, especially in mobile telephony networks. It can also be used for other services, for example, in the WebRTC framework for data channels.


3.5.1. Protocol Description
3.5.1. 协议描述

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. SCTP only supports unicast.

SCTP是一种面向连接的协议,使用四路握手建立SCTP关联,并使用三路消息交换正常关闭SCTP。它使用与DCCP、TCP、UDP和UDP Lite相同的端口号概念。SCTP仅支持单播。

SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit errors and misdelivery of packets to an unintended endpoint. This is stronger than the 16-bit checksums used by TCP or UDP. However, partial payload checksum coverage as provided by DCCP or UDP-Lite is not supported.

SCTP使用32位CRC32c保护SCTP数据包,防止数据包误码和误发到非预期端点。这比TCP或UDP使用的16位校验和更强。但是,不支持DCCP或UDP Lite提供的部分有效负载校验和覆盖。

SCTP has been designed with extensibility in mind. A common header is followed by a sequence of chunks. [RFC4960] defines how a receiver processes chunks with an unknown chunk type. The support of extensions can be negotiated during the SCTP handshake. Currently defined extensions include mechanisms for dynamic reconfiguration of streams [RFC6525] and IP addresses [RFC5061]. Furthermore, the extension specified in [RFC3758] introduces the concept of partial reliability for user messages.


SCTP provides a message-oriented service. Multiple small user messages can be bundled into a single SCTP packet to improve efficiency. For example, this bundling may be done by delaying user messages at the sender, similar to Nagle's algorithm used by TCP. User messages that 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. For MTU discovery, the same mechanism as for TCP can be used [RFC1981] [RFC4821], as well as utilization of probe packets with padding chunks, as defined in [RFC4820].


[RFC4960] specifies TCP-friendly congestion control to protect the network against overload. SCTP also uses sliding window flow control to protect receivers against overflow. Similar to TCP, SCTP also supports delaying acknowledgments. [RFC7053] provides a way for the sender of user messages to request immediate sending of the corresponding acknowledgments.


Each SCTP association has between 1 and 65536 unidirectional 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 unordered or ordered upon request by the upper layer. Unordered messages can be delivered as soon as they are completely received. For user messages not requiring fragmentation, this minimizes head-of-line blocking. On the other hand, ordered messages sent on the same stream are delivered at the receiver in the same order as sent by the sender.


The base protocol defined in [RFC4960] does not allow interleaving of user messages. Large messages on one stream can therefore block the sending of user messages on other streams. [SCTP-NDATA] describes a method to overcome this limitation. This document also 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 reconfiguration 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 delivered to the receiver or, in case of excessive retransmissions, the association is terminated in a non-graceful way [RFC4960], similar to TCP behavior. 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.


SCTP supports multihoming. 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 reconfiguration extension specified in [RFC5061] in combination with [RFC4895] can be used to change these addresses in an authenticated way during the lifetime 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.


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, such as unordered delivery or partial reliability. 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.


[NAT-SUPP] defines methods for endpoints and middleboxes to provide NAT traversal 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 [SCTP-DTLS-ENCAPS]. The latter encapsulation is used within the WebRTC [WEBRTC-TRANS] context.


An SCTP ABORT chunk may be used to force a SCTP endpoint to close a session [RFC4960], aborting the connection.


SCTP has a well-defined API, described in the next subsection.


3.5.2. Interface Description
3.5.2. 接口描述

[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, and NETWORK STATUS CHANGE.


An extension to the BSD Sockets API is defined in [RFC6458] and covers:


o the base protocol defined in [RFC4960]. The API allows control over local addresses and port numbers and the primary path. Furthermore, the application has fine control of parameters like retransmission thresholds, the path supervision, the delayed acknowledgment timeout, and the fragmentation point. The API provides a mechanism to allow the SCTP stack to notify the application about events 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, including drop of messages sent unreliably, the application can also be notified, and user messages can be returned to the application. When sending user messages, the application can indicate a stream id, a payload protocol identifier, and an indication of whether ordered delivery is requested. These parameters can also be provided on message reception. Additionally, a context can be provided when sending, which can be used in case of send failures. The sending of arbitrarily large user messages is supported.

o [RFC4960]中定义的基本协议。API允许控制本地地址、端口号和主路径。此外,应用程序可以很好地控制参数,如重传阈值、路径监控、延迟确认超时和分段点。API提供了一种机制,允许SCTP堆栈在应用程序请求事件时通知应用程序事件。这些通知提供有关关联和每个对等地址的状态更改的信息。如果发送失败,包括丢失不可靠发送的消息,还可以通知应用程序,并将用户消息返回给应用程序。当发送用户消息时,应用程序可以指示流id、有效负载协议标识符以及是否请求有序交付的指示。这些参数也可在信息接收时提供。此外,发送时可以提供上下文,在发送失败时可以使用上下文。支持发送任意大的用户消息。

o the SCTP Partial Reliability extension defined in [RFC3758] to specify for a user message the Partially Reliable SCTP (PR-SCTP) policy and the policy-specific parameter. Examples of these policies defined in [RFC3758] and [RFC7496] are:

o [RFC3758]中定义的SCTP部分可靠性扩展,用于为用户消息指定部分可靠SCTP(PR-SCTP)策略和策略特定参数。[RFC3758]和[RFC7496]中定义的这些策略的示例如下:

* limiting the time a user message is dealt with by the sender.

* 限制发件人处理用户消息的时间。

* 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.

* 限制用户消息每个片段的重新传输次数。如果重新传输的次数限制为0,则可以获得类似于UDP的服务。

* abandoning messages of lower priority in case of a send buffer shortage.

* 在发送缓冲区不足的情况下放弃优先级较低的消息。

o the SCTP Authentication extension defined in [RFC4895] allowing management of the shared keys and allowing the HMAC to use and set the chunk types (which are only accepted in an authenticated way) and get the list of chunks that are accepted by the local and remote endpoints in an authenticated way.

o [RFC4895]中定义的SCTP认证扩展,允许管理共享密钥,允许HMAC使用和设置区块类型(仅以认证方式接受),并获取本地和远程端点以认证方式接受的区块列表。

o the SCTP Dynamic Address Reconfiguration extension defined in [RFC5061]. It allows the manual addition and deletion of local addresses for SCTP associations, as well as the enabling of automatic address addition and deletion. Furthermore, the peer can be given a hint for choosing its primary path.

o [RFC5061]中定义的SCTP动态地址重新配置扩展。它允许手动添加和删除SCTP关联的本地地址,以及启用自动添加和删除地址。此外,可以向对等方提供选择其主路径的提示。

A BSD Sockets API extension has been defined in the documents that specify the following SCTP extensions:


o the SCTP Stream Reconfiguration extension defined in [RFC6525]. The API allows triggering of the reset operation for incoming and outgoing streams and the whole association. It also provides a way to notify the association about the corresponding events. Furthermore, the application can increase the number of streams.

o [RFC6525]中定义的SCTP流重新配置扩展。API允许触发传入和传出流以及整个关联的重置操作。它还提供了一种将相应事件通知关联的方法。此外,应用程序可以增加流的数量。

o the UDP Encapsulation of SCTP packets extension defined in [RFC6951]. The API allows the management of the remote UDP encapsulation port.

o [RFC6951]中定义的SCTP数据包扩展的UDP封装。API允许管理远程UDP封装端口。

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 acknowledgment immediately.

o [RFC7053]中定义的SCTP SACK-REQUEST扩展。API允许用户消息的发送方请求接收方立即发送相应的确认。

o the additional PR-SCTP policies defined in [RFC7496]. The API allows enabling/disabling the PR-SCTP extension, choosing the PR-SCTP policies defined in the document, and providing statistical information about abandoned messages.

o [RFC7496]中定义的其他PR-SCTP策略。API允许启用/禁用PR-SCTP扩展,选择文档中定义的PR-SCTP策略,并提供有关已放弃消息的统计信息。

Future documents describing SCTP 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 一对一样式的套接字(通过使用套接字类型“SOCK_STREAM”)。

o one-to-many style socket (by using the socket type "SOCK_SEQPACKET").

o 一对多样式套接字(通过使用套接字类型“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, 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 similarly to user messages but are specifically marked as 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 sent without using ancillary data in the form of additional Control Message (cmsg) calls. 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 behavior 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.


3.5.3. Transport Features
3.5.3. 运输特征

The transport features provided by SCTP are:


o connection-oriented transport with feature negotiation and application-to-port mapping,

o 具有功能协商和应用到端口映射的面向连接的传输,

o unicast transport,

o 单播传输,

o port multiplexing,

o 端口多路复用,

o unidirectional or bidirectional communication,

o 单向或双向通信,

o message-oriented delivery with durable message framing supporting multiple concurrent streams,

o 面向消息的交付,具有支持多个并发流的持久消息框架,

o fully reliable, partially reliable, or unreliable delivery (based on user-specified policy to handle abandoned user messages) with drop notification,

o 具有丢弃通知的完全可靠、部分可靠或不可靠的传递(基于用户指定的策略来处理放弃的用户消息),

o ordered and unordered delivery within a stream,

o 流中的有序和无序交付,

o support for stream scheduling prioritization,

o 支持流调度优先级,

o segmentation,

o 分段,

o user message bundling,

o 用户消息绑定,

o flow control using a window-based mechanism,

o 使用基于窗口的机制进行流量控制,

o congestion control using methods similar to TCP,

o 使用类似于TCP的方法进行拥塞控制,

o strong error detection (CRC32c), and

o 强错误检测(CRC32c),以及

o transport-layer multihoming for resilience and mobility.

o 传输层多归属,实现弹性和移动性。

3.6. Datagram Congestion Control Protocol (DCCP)
3.6. 数据报拥塞控制协议(DCCP)

The Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF Standards Track bidirectional transport protocol that provides unicast connections of congestion-controlled messages without providing reliability.


The DCCP Problem Statement [RFC4336] describes the goals that DCCP sought to address. 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].


DCCP 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 transport functions (feature negotiation, path state management, RTT calculation, PMTUD, etc.): DCCP applications select how they send packets and, where suitable, choose common algorithms to manage their functions. Examples of applications that can benefit from such transport services include interactive applications, streaming media, or on-line games [RFC4336].


3.6.1. Protocol Description
3.6.1. 协议描述

DCCP is a connection-oriented datagram protocol that provides a three-way handshake to allow a client and server to set up a connection and provides mechanisms for orderly completion and immediate teardown of a connection.


A DCCP protocol instance can be extended [RFC4340] and tuned using additional features. Some features are sender-side only, requiring no negotiation with the receiver; some are receiver-side only; and some are explicitly negotiated during connection setup.


DCCP uses a Connect packet to initiate a session and permits each endpoint 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 [RFC5595] that identifies the application or protocol using DCCP, providing middleboxes with information about the intended use of a connection.


The DCCP service is unicast-only.


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 and remote port numbers.


The protocol segments data into messages that are typically sized to fit in IP packets but may be fragmented if they are smaller than 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 (Congestion Control Maximum Packet Size (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 unordered delivery of data and does not itself provide retransmission. DCCP supports reduced checksum coverage, a partial payload protection mechanism similar to UDP-Lite. There is also a Data Checksum option, which when enabled, contains a strong Cyclic Redundancy Check (CRC), to enable endpoints to detect application data corruption.

每条消息由一个序列号标识。序列号用于识别确认中的段、检测未确认的段、测量RTT等。协议可能支持无序传输数据,并且自身不提供重传。DCCP支持减少校验和覆盖,这是一种类似于UDP Lite的部分有效负载保护机制。还有一个数据校验和选项,启用时,该选项包含一个强循环冗余校验(CRC),以使端点能够检测应用程序数据损坏。

Receiver flow control is supported, which limits the amount of unacknowledged data that can be outstanding at a given time.


A DCCP Reset packet may be used to force a DCCP endpoint to close a session [RFC4340], aborting the connection.


DCCP supports negotiation of the congestion control profile between endpoints, to provide plug-and-play congestion control mechanisms. Examples of specified profiles include "TCP-like" [RFC4341], "TCP-friendly" [RFC4342], and "TCP-friendly for small packets" [RFC5622]. Additional mechanisms are recorded in an IANA registry (see <>).

DCCP支持端点之间的拥塞控制配置文件协商,以提供即插即用的拥塞控制机制。指定配置文件的示例包括“TCP-like”[RFC4341]、“TCP-friendly”[RFC4342]和“TCP-friendly for small packets”[RFC5622]。IANA注册表中记录了其他机制(参见<>).

   A lightweight UDP-based encapsulation (DCCP-UDP) has been defined
   [RFC6773] that permits DCCP to be used over paths where DCCP is not
   natively supported.  Support for DCCP in NAPT/NATs is defined in
   [RFC4340] and [RFC5595].  Upper-layer protocols specified on top of
   DCCP include DTLS [RFC5238], RTP [RFC5762], and Interactive
   Connectivity Establishment / Session Description Protocol (ICE/SDP)
   A lightweight UDP-based encapsulation (DCCP-UDP) has been defined
   [RFC6773] that permits DCCP to be used over paths where DCCP is not
   natively supported.  Support for DCCP in NAPT/NATs is defined in
   [RFC4340] and [RFC5595].  Upper-layer protocols specified on top of
   DCCP include DTLS [RFC5238], RTP [RFC5762], and Interactive
   Connectivity Establishment / Session Description Protocol (ICE/SDP)
3.6.2. Interface Description
3.6.2. 接口描述

Functions expected for a DCCP API include: Open, Close, and Management of the progress a DCCP connection. The Open function provides feature negotiation, selection of an appropriate Congestion Control Identifier (CCID) for congestion control, and other parameters associated with the DCCP connection. A function allows an application to send DCCP datagrams, including setting the required checksum coverage and any required options. (DCCP permits sending datagrams with a zero-length payload.) A function allows reception of data, including indicating if the data was used or dropped. Functions can also make the status of a connection visible to an application, including detection of the maximum packet size and the ability to perform flow control by detecting a slow receiver at the sender.

DCCP API的预期功能包括:打开、关闭和管理DCCP连接的进度。开放功能提供功能协商、为拥塞控制选择适当的拥塞控制标识符(CCID)以及与DCCP连接相关的其他参数。函数允许应用程序发送DCCP数据报,包括设置所需的校验和覆盖率和任何所需选项。(DCCP允许发送具有零长度有效负载的数据报。)功能允许接收数据,包括指示数据是否被使用或丢弃。功能还可以使连接的状态对应用程序可见,包括检测最大数据包大小以及通过检测发送方的慢速接收器来执行流控制的能力。

There is no API currently specified in the RFC Series.


3.6.3. Transport Features
3.6.3. 运输特征

The transport features provided by DCCP are:


o unicast transport,

o 单播传输,

o connection-oriented communication with feature negotiation and application-to-port mapping,

o 面向连接的通信,具有功能协商和端口映射应用,

o signaling of application class for middlebox support (implemented using Service Codes),

o 中间箱支持应用程序类的信令(使用服务代码实现),

o port multiplexing,

o 端口多路复用,

o unidirectional or bidirectional communication,

o 单向或双向通信,

o message-oriented delivery,

o 以信息为导向的交付,

o unreliable delivery with drop notification,

o 带丢弃通知的不可靠传递,

o unordered delivery,

o 无序交货,

o flow control (implemented using the slow receiver function), and

o 流量控制(使用慢速接收器功能实现),以及

o partial and full payload error detection (with optional strong integrity check).

o 部分和全部有效负载错误检测(可选的强完整性检查)。

3.7. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a Pseudotransport

3.7. 作为伪传输的传输层安全性(TLS)和数据报TLS(DTLS)

Transport Layer Security (TLS) [RFC5246] and Datagram TLS (DTLS) [RFC6347] 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 (UDP or DCCP [RFC5238]). At the time of writing, the current version of TLS is 1.2, defined in [RFC5246]; work on TLS version is 1.3 [TLS-1.3] nearing completion. 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) [RFC6101] protocols developed in the mid-1990s 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.


3.7.1. Protocol Description
3.7.1. 协议描述

Both TLS and DTLS provide the same security features and can thus be discussed together. The features they provide are:


o Confidentiality

o 保密性

o Data integrity

o 数据完整性

o Peer authentication (optional)

o 对等身份验证(可选)

o Perfect forward secrecy (optional)

o 完美前向保密(可选)

The authentication of the peer entity can be omitted; a common web use case is where the server is authenticated and the client is not. TLS also provides a completely anonymous operation mode in which neither peer's identity is authenticated. It is important to note that TLS itself does not specify how a peering entity's identity 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 UDP, applications will need to tolerate lost, reordered, 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 录制重播检测(可选)。

o Record size negotiation (estimates of PMTU and record size expansion factor).

o 记录大小协商(估计PMTU和记录大小扩展系数)。

o Conveyance of IP don't fragment (DF) bit settings by application.

o 通过应用程序传输IP不分段(DF)位设置。

o An anti-DoS stateless cookie mechanism (optional).

o 反DoS无状态cookie机制(可选)。

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 of [RFC6347]. DTLS forbids the use of stream ciphers, which are essentially incompatible when operating on independent encrypted records.


3.7.2. Interface Description
3.7.2. 接口描述

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.


Considerable care is required in the use of TLS APIs to ensure creation of a secure application. The programmer should have at least a basic understanding of encryption and digital signature algorithms and their strengths, public key infrastructure (including X.509 certificates and certificate revocation), and the Sockets API. See [RFC7525] and [RFC7457], as mentioned above.

在使用TLS API时需要相当谨慎,以确保创建安全的应用程序。程序员应至少对加密和数字签名算法及其优势、公钥基础设施(包括X.509证书和证书吊销)以及套接字API有基本了解。如上所述,参见[RFC7525]和[RFC7457]。

As an example, in the case of OpenSSL, the primary abstractions are the library itself, 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.


3.7.3. Transport Features
3.7.3. 运输特征

Both TLS and DTLS employ a layered architecture. The lower layer is commonly called the "record protocol". It is responsible for:


o message fragmentation,

o 消息碎片,

o authentication and integrity via message authentication codes (MACs),

o 通过消息身份验证码(MAC)进行身份验证和完整性,

o data encryption, and

o 数据加密,以及

o scheduling transmission using the underlying transport protocol.

o 使用基础传输协议安排传输。

DTLS augments the TLS record protocol with:


o ordering and replay protection, implemented using sequence numbers.

o 排序和重播保护,使用序列号实现。

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.

在记录协议之上有几个协议分层。其中包括握手、警报和更改密码规范协议。还有数据协议,用于承载应用程序流量。握手协议用于在首次建立连接时建立加密和压缩参数。在DTLS中,该协议还具有基本的分段和重传能力,并具有类似cookie的机制来抵抗DoS攻击。(目前不建议使用TLS压缩)。警报协议用于通知对等方各种情况,其中大多数是用于连接的终端。change cipher spec协议用于同步每个对等方的加密参数更改。

The data protocol, when used with an appropriate cipher, provides:


o authentication of one end or both ends of a connection,

o 连接一端或两端的身份验证,

o confidentiality, and

o 保密性,以及

o cryptographic integrity protection.

o 密码完整性保护。

Both TLS and DTLS are unicast-only.


3.8. Real-Time Transport Protocol (RTP)
3.8. 实时传输协议(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 transport services, including TCP, UDP, UDP-Lite, DCCP, TLS, and DTLS.

RTP提供端到端网络传输服务,适用于通过多播或单播传输服务(包括TCP、UDP、UDP Lite、DCCP、TLS和DTL)传输实时数据(如音频、视频或数据)的应用程序。

3.8.1. Protocol Description
3.8.1. 协议描述

The RTP standard [RFC3550] defines a pair of protocols: RTP and the RTP Control Protocol (RTCP). The transport does not provide connection setup, instead relying 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 payload formats that allow packets to carry a wide range of media and specifies a wide range of multiplexing, error control, and other support mechanisms.


If a frame of media data is large, it will be fragmented into several RTP packets. Likewise, several small frames may be bundled into a single RTP packet.


An RTP receiver collects RTP packets from the network, validates them for correctness, and sends them to the media decoder input queue. Missing packet detection is performed by the channel decoder. The playout 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. Some uses of RTP are able to exploit the partial payload protection features offered by DCCP and UDP-Lite.

RTP接收器从网络收集RTP数据包,验证其正确性,并将其发送到媒体解码器输入队列。丢失分组检测由信道解码器执行。播放缓冲区按时间戳排序,用于重新排序数据包。在解压媒体有效载荷以显示或存储数据之前,可以修复损坏的帧。RTP的某些用途能够利用DCCP和UDP Lite提供的部分有效负载保护功能。

RTCP is a control protocol that works alongside an RTP flow. Both the RTP sender and receiver will 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.

RTCP接收器报告(RTCP RR)定期返回给发送方,以报告关键参数(例如,上次报告间隔内丢失的数据包分数、丢失的数据包累计数量、接收到的最高序列号和到达间抖动)。RTCP RR数据包还包含允许发送方估计到接收方的网络往返时间(RTT)的定时信息。

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 randomized to avoid synchronization of reports from multiple receivers.


3.8.2. Interface Description
3.8.2. 接口描述

There is no standard API defined for RTP or RTCP. Implementations are typically tightly integrated with a particular application and closely follow the principles of application-level framing and integrated layer processing [ClarkArch] in media processing [RFC2736], error recovery and concealment, rate adaptation, and security [RFC7202]. Accordingly, RTP implementations tend to be targeted at particular application domains (e.g., voice-over-IP, IPTV, or video conferencing), with a feature set optimized for that domain, rather than being general purpose implementations of the protocol.


3.8.3. Transport Features
3.8.3. 运输特征

The transport features provided by RTP are:


o unicast, multicast, or IPv4 broadcast (provided by lower-layer protocol),

o 单播、多播或IPv4广播(由低层协议提供),

o port multiplexing (provided by lower-layer protocol),

o 端口多路复用(由下层协议提供),

o unidirectional or bidirectional communication (provided by lower-layer protocol),

o 单向或双向通信(由下层协议提供),

o message-oriented delivery with support for media types and other extensions,

o 面向消息的交付,支持媒体类型和其他扩展,

o reliable delivery when using erasure coding or unreliable delivery with drop notification (if supported by lower-layer protocol),

o 使用擦除编码时的可靠传递或带有丢弃通知的不可靠传递(如果下层协议支持),

o connection setup with feature negotiation (using associated protocols) and application-to-port mapping (provided by lower-layer protocol),

o 通过功能协商(使用相关协议)和应用程序到端口映射(由下层协议提供)建立连接,

o segmentation, and

o 细分,以及

o performance metric reporting (using associated protocols).

o 性能指标报告(使用相关协议)。

3.9. Hypertext Transport Protocol (HTTP) over TCP as a Pseudotransport
3.9. TCP上的超文本传输协议(HTTP)作为伪传输

The Hypertext Transfer Protocol (HTTP) is an application-level protocol widely used on the Internet. It provides object-oriented delivery of discrete data or files. Version 1.1 of the protocol is specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234] [RFC7235], and version 2 is specified in [RFC7540]. HTTP is usually transported over TCP using ports 80 and 443, although it can be used with other transports. When used over TCP, it inherits TCP's properties.


Application-layer protocols may use HTTP as a substrate with an existing method and data formats, or specify new methods and data formats. 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 [RFC7235] and TLS [RFC5246], and the ability of HTTP to traverse firewalls, which allows it to work over many types of infrastructure and in cases where an application server often needs to support HTTP anyway.


Depending on application need, 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] addresses this issue, provides some guidelines, and identifies concerns about the use of HTTP standard ports 80 and 443, the use of the HTTP URL scheme, and interaction with existing firewalls, proxies, and NATs.

根据应用需要,与专用协议相比,将HTTP用作底层协议可能会增加复杂性和开销(例如,HTTP头、HTTP安全模型的适用性等)。[RFC3205]解决了这个问题,提供了一些指导原则,并确定了有关HTTP标准端口80和443的使用、HTTP URL方案的使用以及与现有防火墙、代理和NAT的交互的问题。

Representational State Transfer (REST) [REST] is another example of how applications can use HTTP as a transport protocol. REST is an architecture style that may be used to build applications using HTTP as a communication protocol.

Representational State Transfer(REST)[REST]是应用程序如何使用HTTP作为传输协议的另一个示例。REST是一种架构风格,可用于构建使用HTTP作为通信协议的应用程序。

3.9.1. Protocol Description
3.9.1. 协议描述

The Hypertext Transfer Protocol (HTTP) is a request/response protocol. A client sends a request containing a request method, URI, and protocol version followed by message whose design is inspired by


MIME (see [RFC7231] for the differences between an HTTP object and a MIME message), containing information about the client and request modifiers. The message can also contain a message body carrying application data. The server responds with a status or error code followed by a message containing information about the server and information about the data. This may include a message body. It is possible to specify a data format for the message body using MIME media types [RFC2045]. The protocol has additional features; some relevant to pseudotransport are described below.


Content negotiation, specified in [RFC7231], is a mechanism provided by HTTP to allow selection of a representation for a requested resource. The client and server negotiate acceptable data formats, character sets, and data encoding (e.g., data can be transferred compressed using gzip). HTTP can accommodate exchange of messages as well as data streaming (using chunked transfer encoding [RFC7230]). It is also possible to request a part of a resource using an object range request [RFC7233]. The protocol provides powerful cache control signaling defined in [RFC7234].


The persistent connections of HTTP 1.1 and HTTP 2.0 allow multiple request/response transactions (streams) during the lifetime of a single HTTP connection. This reduces overhead during connection establishment and mitigates transport-layer slow-start that would have otherwise been incurred for each transaction. HTTP 2.0 connections can multiplex many request/response pairs in parallel on a single transport connection. Both are important to reduce latency for HTTP's primary use case.

HTTP 1.1和HTTP 2.0的持久连接允许在单个HTTP连接的生命周期内进行多个请求/响应事务(流)。这减少了连接建立期间的开销,并减轻了传输层的慢启动,否则每个事务都会发生这种情况。HTTP 2.0连接可以在单个传输连接上并行多路复用多个请求/响应对。这两者对于减少HTTP的主要用例的延迟都很重要。

HTTP can be combined with security mechanisms, such as TLS (denoted by HTTPS). This adds protocol properties provided by such a mechanism (e.g., authentication and encryption). The TLS Application-Layer Protocol Negotiation (ALPN) extension [RFC7301] can be used to negotiate the HTTP version within the TLS handshake, eliminating the latency incurred by additional round-trip exchanges. Arbitrary cookie strings, included as part of the request headers, are often used as bearer tokens in HTTP.


3.9.2. Interface Description
3.9.2. 接口描述

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 HTTPS is used, the API exposes a registration of callbacks when a server requests client authentication and when certificate verification is needed.


The World Wide Web Consortium (W3C) has standardized the XMLHttpRequest API [XHR]. This API can be used for sending HTTP/ HTTPS requests and receiving server responses. Besides the XML data format, the request and response data format can also be JSON, HTML, and plain text. JavaScript and XMLHttpRequest are ubiquitous programming models for websites and more general applications where native code is less attractive.


3.9.3. Transport Features
3.9.3. 运输特征

The transport features provided by HTTP, when used as a pseudotransport, are:


o unicast transport (provided by the lower-layer protocol, usually TCP),

o 单播传输(由下层协议提供,通常为TCP),

o unidirectional or bidirectional communication,

o 单向或双向通信,

o transfer of objects in multiple streams with object content type negotiation, supporting partial transmission of object ranges,

o 通过对象内容类型协商在多个流中传输对象,支持对象范围的部分传输,

o ordered delivery (provided by the lower-layer protocol, usually TCP),

o 有序交付(由下层协议提供,通常为TCP),

o fully reliable delivery (provided by the lower-layer protocol, usually TCP),

o 完全可靠的交付(由下层协议提供,通常为TCP),

o flow control (provided by the lower-layer protocol, usually TCP), and

o 流控制(由下层协议提供,通常为TCP),以及

o congestion control (provided by the lower-layer protocol, usually TCP).

o 拥塞控制(由下层协议提供,通常是TCP)。

HTTPS (HTTP over TLS) additionally provides the following features (as provided by TLS):

HTTPS(HTTP over TLS)还提供以下功能(由TLS提供):

o authentication (of one or both ends of a connection),

o 认证(连接的一端或两端),

o confidentiality, and

o 保密性,以及

o integrity protection.

o 完整性保护。

3.10. File Delivery over Unidirectional Transport / Asynchronous Layered Coding (FLUTE/ALC) for Reliable Multicast

3.10. 用于可靠多播的单向传输/异步分层编码(FLUTE/ALC)上的文件传递

FLUTE/ALC is an IETF Standards Track protocol specified in [RFC6726] and [RFC5775]. It provides object-oriented delivery of discrete data or files. Asynchronous Layer Coding (ALC) provides an underlying reliable transport service and FLUTE a file-oriented specialization of the ALC service (e.g., to carry associated metadata). [RFC6726] and [RFC5775] are non-backward-compatible updates of [RFC3926] and [RFC3450], which are Experimental protocols; these Experimental protocols are currently largely deployed in the 3GPP Multimedia Broadcast / Multicast Service (MBMS) (see [MBMS], Section 7) and similar contexts (e.g., the Japanese ISDB-Tmm standard).

FLUTE/ALC是[RFC6726]和[RFC5775]中规定的IETF标准跟踪协议。它提供离散数据或文件的面向对象交付。异步层编码(ALC)提供了一个底层的可靠传输服务,并实现了ALC服务面向文件的专门化(例如,携带相关的元数据)。[RFC6726]和[RFC5775]是[RFC3926]和[RFC3450]的非向后兼容更新,这是实验协议;这些实验协议目前主要部署在3GPP多媒体广播/多播服务(MBMS)(参见[MBMS],第7节)和类似环境(例如,日本ISDB Tmm标准)中。

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". Although 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).


FLUTE/ALC's reliability, delivery mode, congestion control, and flow/ rate control mechanisms can be separately controlled to meet different application needs. Section 4.1 of [RFC8085] describes multicast congestion control requirements for UDP.


3.10.1. Protocol Description
3.10.1. 协议描述

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 regardless 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 (see [RFC5651], Section 4.2).


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).

这使接收者能够自行决定异步加入会话,接收内容,然后离开会话。在这种情况下,数据内容通常以循环(也称为“转盘”)的形式连续发送。FLUTE/ALC还支持对象流的传输,具有松散的实时约束。当需要可伸缩性且无法使用HTTP上的单播传输时,这对于携带3GPP DASH特别有用([MBMS],第5.6节)。在这种情况下,使用推送模式按顺序发送数据包。FLUTE/ALC不能很好地适应字节流和消息流,可以首选其他解决方案(例如,具有实时流的FECFRAME[RFC6363])。

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.


Error detection and verification of the protocol control information relies on the underlying transport (e.g., UDP checksum).


To provide robustness against packet loss and improve the efficiency of the on-demand mode, FLUTE/ALC relies on packet erasure coding (Application-Layer Forward Error Correction (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., Wave and Equation Based Rate


Control (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 for the Internet ([RFC6726], Section 1.1.4). FLUTE/ALC is often used over a network path with pre-provisioned capacity [RFC8085] where there are no flows competing for capacity. In this case, a sender-based rate control mechanism and a single channel are 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
3.10.2. 接口描述

The FLUTE/ALC specification does not describe a specific API to control protocol operation. Although open source and commercial implementations have specified APIs, there is no IETF-specified API for FLUTE/ALC.


3.10.3. Transport Features
3.10.3. 运输特征

The transport features provided by FLUTE/ALC are:


o unicast, multicast, anycast, or IPv4 broadcast transmission,

o 单播、多播、选播或IPv4广播传输,

o object-oriented delivery of discrete data or files and associated metadata,

o 离散数据或文件及相关元数据的面向对象交付,

o fully reliable or partially reliable delivery (of file or in-memory objects), using proactive packet erasure coding (AL-FEC) to recover from packet erasures,

o 使用主动数据包擦除编码(AL-FEC)从数据包擦除中恢复(文件或内存对象)的完全可靠或部分可靠传输,

o ordered or unordered delivery (of file or in-memory objects),

o 有序或无序交付(文件或内存对象),

o error detection (based on the UDP checksum),

o 错误检测(基于UDP校验和),

o per-packet authentication,

o 每包身份验证,

o per-packet integrity,

o 每包完整性,

o per-packet replay protection, and

o 每包重播保护,以及

o congestion control for layered flows (e.g., with WEBRC).

o 分层流的拥塞控制(例如,使用WEBRC)。

3.11. NACK-Oriented Reliable Multicast (NORM)
3.11. 面向NACK的可靠组播(NORM)

NORM is an IETF Standards Track protocol specified in [RFC5740]. It provides object-oriented delivery of discrete data or files.


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. Support for bulk data dissemination includes discrete file or computer memory-based "objects" as well as byte and message streaming.


NORM can incorporate packet erasure coding as a part of its selective Automatic Repeat reQuest (ARQ) in response to negative acknowledgments from the receiver. The packet erasure coding can also be proactively applied for forward protection from packet loss. NORM transmissions are governed by TCP-Friendly Multicast Congestion Control (TFMCC) [RFC4654]. The reliability, congestion control, and flow control mechanisms can be separately controlled to meet different application needs.


3.11.1. Protocol Description
3.11.1. 协议描述

The NORM protocol is encapsulated in UDP datagrams and thus provides multiplexing for multiple sockets on hosts using port numbers. For 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. Separate session initiation can be used to coordinate port numbers. However, in-band "client-server" style connection establishment can be accomplished with the NORM congestion control signaling messages using port binding techniques like those for TCP client-server connections.


NORM supports bulk "objects" such as file or in-memory content but also can treat a stream of data as a logical bulk object for purposes of packet erasure coding. In the case of stream transport, NORM can support either byte streams or message streams where application-defined message boundary information is carried in the NORM protocol messages. This allows the receiver(s) to join/rejoin and recover message boundaries mid-stream as needed. Application content is carried and identified by the NORM protocol with encoding symbol identifiers depending upon the Forward Error Correction (FEC) Scheme [RFC5052] configured. NORM uses NACK-based selective ARQ to reliably deliver the application content to the receiver(s). NORM proactively measures round-trip timing information to scale ARQ timers appropriately and to support congestion control. For multicast


operation, timer-based feedback suppression is used to achieve group size scaling with low feedback traffic levels. The feedback suppression is not applied for unicast operation.


NORM uses rate-based congestion control based upon the TCP-Friendly Rate Control (TFRC) [RFC5348] principles that are also used in DCCP [RFC4340]. NORM uses control messages to measure RTT and collect congestion event information (e.g., reflecting a loss event or ECN event) from the receiver(s) to support dynamic adjustment or the rate. TCP-Friendly Multicast Congestion Control (TFMCC) [RFC4654] provides extra features to support multicast but is functionally equivalent to TFRC for unicast.


Error detection and verification of the protocol control information relies on the on the underlying transport (e.g., UDP checksum).


The reliability mechanism is decoupled from congestion control. This allows invocation of alternative arrangements of transport services, for example, to support, fixed-rate reliable delivery or unreliable delivery (that may optionally be "better than best effort" via packet erasure coding) using TFRC. Alternative congestion control techniques may be applied, for example, TFRC with congestion event detection based on ECN.


While NORM provides NACK-based reliability, it also supports a positive acknowledgment (ACK) mechanism that can be used for receiver flow control. This mechanism is decoupled from the reliability and congestion control, supporting applications with different needs. One example is 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.


3.11.2. Interface Description
3.11.2. 接口描述

The NORM specification does not describe a specific API to control protocol operation. A freely available, open-source reference implementation of NORM is available at <>, 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.


3.11.3. Transport Features
3.11.3. 运输特征

The transport features provided by NORM are:


o unicast or multicast transport,

o 单播或多播传输,

o unidirectional communication,

o 单向通信,

o stream-oriented delivery in a single stream or object-oriented delivery of in-memory data or file bulk content objects,

o 单流中的面向流交付或内存数据或文件批量内容对象的面向对象交付,

o fully reliable (NACK-based) or partially reliable (using erasure coding both proactively and as part of ARQ) delivery,

o 完全可靠(基于NACK)或部分可靠(主动或作为ARQ的一部分使用擦除编码)交付,

o unordered delivery,

o 无序交货,

o error detection (relies on UDP checksum),

o 错误检测(依赖UDP校验和),

o segmentation,

o 分段,

o data bundling (using Nagle's algorithm),

o 数据绑定(使用Nagle算法),

o flow control (timer-based and/or ACK-based), and

o 流量控制(基于定时器和/或基于ACK),以及

o congestion control (also supporting fixed-rate reliable or unreliable delivery).

o 拥塞控制(也支持固定速率可靠或不可靠的传输)。

3.12. Internet Control Message Protocol (ICMP)
3.12. 因特网控制消息协议(ICMP)

The Internet Control Message Protocol (ICMP) [RFC792] for IPv4 and ICMP for IPv6 [RFC4443] are IETF Standards Track protocols. It is a connectionless unidirectional protocol that delivers individual messages, without error correction, congestion control, or flow control. Messages may be sent as unicast, IPv4 broadcast, or multicast datagrams (IPv4 and IPv6), in addition to anycast datagrams.


While ICMP is not typically described as a transport protocol, it does position itself over the network layer, and the operation of other transport protocols can be closely linked to the functions provided by ICMP.


Transport protocols and upper-layer protocols can use received ICMP messages to help them make appropriate decisions when network or endpoint errors are reported, for example, to implement ICMP-based Path MTU Discovery (PMTUD) [RFC1191] [RFC1981] or assist in Packetization Layer PMTUD (PLPMTUD) [RFC4821]. Such reactions to received messages need to protect from off-path data injection


[RFC8085] to avoid an application receiving packets created by an unauthorized third party. An application therefore needs to ensure that all messages are appropriately validated by checking the payload of the messages to ensure they 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 TCP, DCCP, or SCTP). This state is not always maintained by UDP-based applications [RFC8085].


3.12.1. Protocol Description
3.12.1. 协议描述

ICMP is a connectionless unidirectional protocol. It delivers independent messages, called "datagrams". Each message is required to carry a checksum as an integrity check and to protect from misdelivery to an unintended endpoint.


ICMP messages typically relay diagnostic information from an endpoint [RFC1122] or network device [RFC1812] addressed to the sender of a flow. This usually contains the network protocol header of a packet that encountered a reported issue. Some formats of messages can also carry other payload data. Each message carries an integrity check calculated in the same way as for UDP; this checksum is not optional.


The RFC Series defines additional IPv6 message formats to support a range of uses. In the case of IPv6, the protocol incorporates neighbor discovery [RFC4861] [RFC3971] (provided by ARP for IPv4) and Multicast Listener Discovery (MLD) [RFC2710] group management functions (provided by IGMP for IPv4).


Reliable transmission cannot 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.


3.12.2. Interface Description
3.12.2. 接口描述

ICMP processing is integrated in many connection-oriented transports but, like other functions, needs to be provided by an upper-layer protocol when using UDP and UDP-Lite.

ICMP处理集成在许多面向连接的传输中,但与其他功能一样,在使用UDP和UDP Lite时,需要由上层协议提供。

On some stacks, a bound socket also allows a UDP application to be notified when ICMP error messages are received for its transmissions [RFC8085].


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] [RFC8085].


3.12.3. Transport Features
3.12.3. 运输特征

ICMP does not provide any transport service directly to applications. Used together with other transport protocols, it provides transmission of control, error, and measurement data between endpoints or from devices along the path to one endpoint.


4. Congestion Control
4. 拥塞控制

Congestion control is critical to the stable operation of the Internet. A variety of mechanisms are used to provide the congestion control needed by many Internet transport protocols. Congestion is detected based on sensing of network conditions, whether through explicit or implicit feedback. The congestion control mechanisms that can be applied by different transport protocols are largely orthogonal to the choice of transport protocol. This section provides an overview of the congestion control mechanisms available to the protocols described in Section 3.


Many protocols use a separate window to determine the maximum sending rate that is allowed by the congestion control. The used congestion control mechanism will increase the congestion window if feedback is received that indicates that the currently used network path is not congested and will reduce the window otherwise. Window-based mechanisms often increase their window slowing over multiple RTTs, while decreasing strongly when the first indication of congestion is received. One example is an Additive Increase Multiplicative Decrease (AIMD) scheme, where the window is increased by a certain number of packets/bytes for each data segment that has been successfully transmitted, while the window decreases multiplicatively on the occurrence of a congestion event. This can lead to a rather unstable, oscillating sending rate but will resolve a congestion situation quickly. Examples of window-based AIMD schemes include TCP NewReno [RFC5681], TCP Cubic [CUBIC] (the default mechanism for TCP in Linux), and CCID 2 specified for DCCP [RFC4341].

许多协议使用单独的窗口来确定拥塞控制允许的最大发送速率。如果收到指示当前使用的网络路径未拥塞的反馈,则使用的拥塞控制机制将增加拥塞窗口,否则将减少窗口。基于窗口的机制通常会在多个RTT上增加其窗口速度,而在收到拥塞的第一个指示时,其速度会显著降低。一个示例是加法-增加-乘法-减少(AIMD)方案,其中对于已成功传输的每个数据段,窗口增加一定数量的分组/字节,而在发生拥塞事件时,窗口则乘性地减小。这可能导致相当不稳定、振荡的发送速率,但会很快解决拥塞情况。基于窗口的AIMD方案的示例包括TCP NewReno[RFC5681]、TCP Cubic[Cubic](Linux中TCP的默认机制)和为DCCP[RFC4341]指定的CCID 2。

Some classes of applications prefer to use a transport service that allows sending at a more stable rate that is slowly varied in response to congestion. Rate-based methods offer this type of congestion control and have been defined based on the loss ratio and observed round-trip time, such as TFRC [RFC5348] and TFRC-SP


[RFC4828]. These methods utilize a throughput equation to determine the maximum acceptable rate. Such methods are used with DCCP CCID 3 [RFC4342], CCID 4 [RFC5622], WEBRC [RFC3738], and other applications.

[RFC4828]。这些方法利用吞吐量方程来确定最大可接受速率。这些方法用于DCCP CCID 3[RFC4342]、CCID 4[RFC5622]、WEBRC[RFC3738]和其他应用。

Another class of applications prefers a transport service that yields to other (higher-priority) traffic, such as interactive transmissions. While most traffic in the Internet uses loss-based congestion control and therefore tends to fill the network buffers (to a certain level if Active Queue Management (AQM) is used), low-priority congestion control methods often react to changes in delay as an earlier indication of congestion. This approach tends to induce less loss than a loss-based method but does generally not compete well with loss-based traffic across shared bottleneck links. Therefore, methods such as LEDBAT [RFC6817] are deployed in the Internet for scavenger traffic that aims to only utilize otherwise unused capacity.


5. Transport Features
5. 运输特征

The transport protocol features described in this document can be used as a basis for defining common transport features. These are listed below with the protocols supporting them:


o Control Functions

o 控制功能

* Addressing

* 寻址



+ multicast (UDP, UDP-Lite, RTP, ICMP, FLUTE/ALC, NORM). Note that, as TLS and DTLS are unicast-only, there is no widely deployed mechanism for supporting the features listed under the Security bullet (below) when using multicast addressing.

+ 多播(UDP、UDP Lite、RTP、ICMP、FLUTE/ALC、NORM)。请注意,由于TLS和DTL仅为单播,因此在使用多播寻址时,没有广泛部署的机制来支持安全项目符号(以下)下列出的功能。

+ IPv4 broadcast (UDP, UDP-Lite, ICMP)

+ IPv4广播(UDP、UDP Lite、ICMP)

+ anycast (UDP, UDP-Lite). Connection-oriented protocols such as TCP and DCCP have also been deployed using anycast addressing, with the risk that routing changes may cause connection failure.

+ 选播(UDP、UDP-Lite)。面向连接的协议(如TCP和DCCP)也已使用选播寻址进行部署,存在路由更改可能导致连接失败的风险。

* Association type

* 关联类型

+ connection-oriented (TCP, MPTCP, DCCP, SCTP, TLS, RTP, HTTP, NORM)


+ connectionless (UDP, UDP-Lite, FLUTE/ALC)

+ 无连接(UDP、UDP Lite、长笛/ALC)

* Multihoming support

* 多归宿支援

+ resilience and mobility (MPTCP, SCTP)

+ 弹性和流动性(MPTCP、SCTP)

+ load balancing (MPTCP)

+ 负载平衡(MPTCP)

+ address family multiplexing (MPTCP, SCTP)

+ 地址族多路复用(MPTCP、SCTP)

* Middlebox cooperation

* 中间商合作

+ application-class signaling to middleboxes (DCCP)

+ 到中间盒(DCCP)的应用级信令

+ error condition signaling from middleboxes and routers to endpoints (ICMP)

+ 从中间盒和路由器到端点的错误条件信令(ICMP)

* Signaling

* 信号

+ control information and error signaling (ICMP)

+ 控制信息和错误信号(ICMP)

+ application performance reporting (RTP)

+ 应用程序性能报告(RTP)

o Delivery

o 传送

* Reliability

* 可靠性

+ fully reliable delivery (TCP, MPTCP, SCTP, TLS, HTTP, FLUTE/ ALC, NORM)


+ partially reliable delivery (SCTP, NORM)

+ 部分可靠交付(SCTP,标准)

- using packet erasure coding (RTP, FLUTE/ALC, NORM)

- 使用数据包擦除编码(RTP、FLUTE/ALC、NORM)

- with specified policy for dropped messages (SCTP)

- 具有指定的丢弃消息策略(SCTP)

+ unreliable delivery (SCTP, UDP, UDP-Lite, DCCP, RTP)

+ 不可靠的传递(SCTP、UDP、UDP-Lite、DCCP、RTP)

- with drop notification to sender (SCTP, DCCP, RTP)

- 向发送方发送丢弃通知(SCTP、DCCP、RTP)

+ error detection

+ 错误检测

- checksum for error detection (TCP, MPTCP, UDP, UDP-Lite, SCTP, DCCP, TLS, DTLS, FLUTE/ALC, NORM, ICMP)


- partial payload checksum protection (UDP-Lite, DCCP). Some uses of RTP can exploit partial payload checksum protection feature to provide a corruption-tolerant transport service.

- 部分有效负载校验和保护(UDP Lite、DCCP)。RTP的某些用途可以利用部分有效负载校验和保护功能来提供容错传输服务。

- checksum optional (UDP). Possible with IPv4 and, in certain cases, with IPv6.

- 校验和可选(UDP)。IPv4和IPv6(在某些情况下)都有可能。

* Ordering

* 订购

+ ordered delivery (TCP, MPTCP, SCTP, TLS, RTP, HTTP, FLUTE)


+ unordered delivery permitted (UDP, UDP-Lite, SCTP, DCCP, RTP, NORM)


* Type/framing

* 类型/框架

+ stream-oriented delivery (TCP, MPTCP, SCTP, TLS, HTTP)


- with multiple streams per association (SCTP, HTTP2)

- 每个关联具有多个流(SCTP、HTTP2)

+ message-oriented delivery (UDP, UDP-Lite, SCTP, DCCP, DTLS, RTP)


+ object-oriented delivery of discrete data or files and associated metadata (HTTP, FLUTE/ALC, NORM)

+ 离散数据或文件及相关元数据的面向对象交付(HTTP、FLUTE/ALC、NORM)

- with partial delivery of object ranges (HTTP)

- 部分交付对象范围(HTTP)

* Directionality

* 方向性

+ unidirectional (UDP, UDP-Lite, DCCP, RTP, FLUTE/ALC, NORM)


+ bidirectional (TCP, MPTCP, SCTP, TLS, HTTP)


o Transmission control

o 传输控制

* flow control (TCP, MPTCP, SCTP, DCCP, TLS, RTP, HTTP)


* congestion control (TCP, MPTCP, SCTP, DCCP, RTP, FLUTE/ALC, NORM). Congestion control can also provided by the transport supporting an upper-layer transport (e.g., TLS, RTP, HTTP).

* 拥塞控制(TCP、MPTCP、SCTP、DCCP、RTP、FLUTE/ALC、NORM)。拥塞控制也可以由支持上层传输的传输提供(例如,TLS、RTP、HTTP)。



* data/message bundling (TCP, MPTCP, SCTP, TLS, HTTP)


* stream scheduling prioritization (SCTP, HTTP2)

* 流调度优先级(SCTP、HTTP2)

* endpoint multiplexing (MPTCP)

* 端点复用(MPTCP)

o Security

o 安全

* authentication of one end of a connection (TLS, DTLS, FLUTE/ ALC)

* 连接一端的身份验证(TLS、DTLS、FLUTE/ALC)

* authentication of both ends of a connection (TLS, DTLS)

* 连接两端的身份验证(TLS、DTL)

* confidentiality (TLS, DTLS)

* 保密性(TLS、DTL)

* cryptographic integrity protection (TLS, DTLS)

* 加密完整性保护(TLS、DTLS)

* replay protection (TLS, DTLS, FLUTE/ALC)

* 重放保护(TLS、DTLS、长笛/自动高度控制)

6. IANA Considerations
6. IANA考虑

This document does not require any IANA actions.


7. Security Considerations
7. 安全考虑

This document surveys existing transport protocols and protocols providing transport-like services. Confidentiality, integrity, and authenticity are among the features provided by those services. This document does not specify any new features or mechanisms for providing these features. Each RFC referenced by this document discusses the security considerations of the specification it contains.


8. Informative References
8. 资料性引用

[ClarkArch] Clark, D. and D. Tennenhouse, "Architectural Considerations for a New Generation of Protocols", Proceedings of ACM SIGCOMM, DOI 10.1145/99517.99553, 1990.

[ClarkArch]Clark,D.和D.Tennenhouse,“新一代协议的架构考虑”,ACM SIGCOMM会议录,DOI 10.1145/99517.99553,1990年。

[CUBIC] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and R. Scheffenegger, "CUBIC for Fast Long-Distance Networks", Work in Progress, draft-ietf-tcpm-cubic-04, February 2017.


[MBMS] 3GPP, "Multimedia Broadcast/Multicast Service (MBMS); Protocols and codecs", 3GPP TS 26.346, 2015, <>.

[MBMS]3GPP,“多媒体广播/多播服务(MBMS);协议和编解码器”,3GPP TS 26.3462015<>.

[NAT-SUPP] Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control Transmission Protocol (SCTP) Network Address Translation Support", Work in Progress, draft-ietf-tsvwg-natsupp-09, May 2016.


[POSIX] IEEE, "Standard for Information Technology -- Portable Operating System Interface (POSIX(R)) Base Specifications, Issue 7", IEEE 1003.1, DOI 10.1109/ieeestd.2016.7582338, <>.

[POSIX]IEEE,“信息技术标准——便携式操作系统接口(POSIX(R))基本规范,第7期”,IEEE 1003.1,DOI 10.1109/ieeestd.2016.7582338<>.

[REST] Fielding, R., "Architectural Styles and the Design of Network-based Software Architectures, Chapter 5: Representational State Transfer", Ph.D. Dissertation, University of California, Irvine, 2000.


[RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, DOI 10.17487/RFC0768, August 1980, <>.

[RFC768]Postel,J.,“用户数据报协议”,STD 6,RFC 768,DOI 10.17487/RFC0768,1980年8月<>.

[RFC792] Postel, J., "Internet Control Message Protocol", STD 5, RFC 792, DOI 10.17487/RFC0792, September 1981, <>.

[RFC792]Postel,J.,“互联网控制消息协议”,STD 5,RFC 792,DOI 10.17487/RFC0792,1981年9月<>.

[RFC793] Postel, J., "Transmission Control Protocol", STD 7, RFC 793, DOI 10.17487/RFC0793, September 1981, <>.

[RFC793]Postel,J.,“传输控制协议”,标准7,RFC 793,DOI 10.17487/RFC0793,1981年9月<>.

[RFC1071] Braden, R., Borman, D., and C. Partridge, "Computing the Internet checksum", RFC 1071, DOI 10.17487/RFC1071, September 1988, <>.

[RFC1071]Braden,R.,Borman,D.,和C.Partridge,“计算互联网校验和”,RFC 1071,DOI 10.17487/RFC10711988年9月<>.

[RFC1122] Braden, R., Ed., "Requirements for Internet Hosts - Communication Layers", STD 3, RFC 1122, DOI 10.17487/RFC1122, October 1989, <>.

[RFC1122]Braden,R.,Ed.“互联网主机的要求-通信层”,STD 3,RFC 1122,DOI 10.17487/RFC1122,1989年10月<>.

[RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191, DOI 10.17487/RFC1191, November 1990, <>.

[RFC1191]Mogul,J.和S.Deering,“MTU发现路径”,RFC 1191,DOI 10.17487/RFC1191,1990年11月<>.

[RFC1812] Baker, F., Ed., "Requirements for IP Version 4 Routers", RFC 1812, DOI 10.17487/RFC1812, June 1995, <>.

[RFC1812]Baker,F.,Ed.,“IP版本4路由器的要求”,RFC 1812,DOI 10.17487/RFC1812,1995年6月<>.

[RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery for IP version 6", RFC 1981, DOI 10.17487/RFC1981, August 1996, <>.

[RFC1981]McCann,J.,Deering,S.,和J.Mogul,“IP版本6的路径MTU发现”,RFC 1981,DOI 10.17487/RFC19811996年8月<>.

[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP Selective Acknowledgment Options", RFC 2018, DOI 10.17487/RFC2018, October 1996, <>.

[RFC2018]Mathis,M.,Mahdavi,J.,Floyd,S.,和A.Romanow,“TCP选择性确认选项”,RFC 2018,DOI 10.17487/RFC2018,1996年10月<>.

[RFC2045] Freed, N. and N. Borenstein, "Multipurpose Internet Mail Extensions (MIME) Part One: Format of Internet Message Bodies", RFC 2045, DOI 10.17487/RFC2045, November 1996, <>.

[RFC2045]Freed,N.和N.Borenstein,“多用途互联网邮件扩展(MIME)第一部分:互联网邮件正文格式”,RFC 2045,DOI 10.17487/RFC20451996年11月<>.

[RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460, December 1998, <>.

[RFC2460]Deering,S.和R.Hinden,“互联网协议,第6版(IPv6)规范”,RFC 2460,DOI 10.17487/RFC2460,1998年12月<>.

[RFC2710] Deering, S., Fenner, W., and B. Haberman, "Multicast Listener Discovery (MLD) for IPv6", RFC 2710, DOI 10.17487/RFC2710, October 1999, <>.

[RFC2710]Deering,S.,Fenner,W.,和B.Haberman,“IPv6的多播侦听器发现(MLD)”,RFC 2710,DOI 10.17487/RFC2710,1999年10月<>.

[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP Payload Format Specifications", BCP 36, RFC 2736, DOI 10.17487/RFC2736, December 1999, <>.

[RFC2736]Handley,M.和C.Perkins,“RTP有效载荷格式规范编写者指南”,BCP 36,RFC 2736,DOI 10.17487/RFC2736,1999年12月<>.

[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC 3168, DOI 10.17487/RFC3168, September 2001, <>.

[RFC3168]Ramakrishnan,K.,Floyd,S.,和D.Black,“向IP添加显式拥塞通知(ECN)”,RFC 3168,DOI 10.17487/RFC3168,2001年9月<>.

[RFC3205] Moore, K., "On the use of HTTP as a Substrate", BCP 56, RFC 3205, DOI 10.17487/RFC3205, February 2002, <>.

[RFC3205]Moore,K.,“关于HTTP作为底物的使用”,BCP 56,RFC 3205,DOI 10.17487/RFC3205,2002年2月<>.

[RFC3260] Grossman, D., "New Terminology and Clarifications for Diffserv", RFC 3260, DOI 10.17487/RFC3260, April 2002, <>.

[RFC3260]Grossman,D.,“区分服务的新术语和澄清”,RFC 3260,DOI 10.17487/RFC3260,2002年4月<>.

[RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport Layer Security over Stream Control Transmission Protocol", RFC 3436, DOI 10.17487/RFC3436, December 2002, <>.

[RFC3436]Jungmaier,A.,Rescorla,E.,和M.Tuexen,“流控制传输协议上的传输层安全”,RFC 3436,DOI 10.17487/RFC3436,2002年12月<>.

[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, <>.

[RFC3450]Luby,M.,Gemmell,J.,Vicisano,L.,Rizzo,L.,和J.Crowcroft,“异步分层编码(ALC)协议实例化”,RFC 3450,DOI 10.17487/RFC3450,2002年12月<>.

[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, <>.

[RFC3550]Schulzrinne,H.,Casner,S.,Frederick,R.,和V.Jacobson,“RTP:实时应用的传输协议”,STD 64,RFC 3550,DOI 10.17487/RFC3550,2003年7月<>.

[RFC3738] Luby, M. and V. Goyal, "Wave and Equation Based Rate Control (WEBRC) Building Block", RFC 3738, DOI 10.17487/RFC3738, April 2004, <>.

[RFC3738]Luby,M.和V.Goyal,“基于波动和方程的速率控制(WEBRC)构造块”,RFC 3738,DOI 10.17487/RFC3738,2004年4月<>.

[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. Conrad, "Stream Control Transmission Protocol (SCTP) Partial Reliability Extension", RFC 3758, DOI 10.17487/RFC3758, May 2004, <>.

[RFC3758]Stewart,R.,Ramalho,M.,Xie,Q.,Tuexen,M.,和P.Conrad,“流控制传输协议(SCTP)部分可靠性扩展”,RFC 3758,DOI 10.17487/RFC3758,2004年5月<>.

[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, <>.

[RFC3828]Larzon,L-A.,Degermark,M.,Pink,S.,Jonsson,L-E.,Ed.,和G.Fairhurst,Ed.,“轻量级用户数据报协议(UDP Lite)”,RFC 3828,DOI 10.17487/RFC3828,2004年7月<>.

[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, <>.

[RFC3926]Paila,T.,Luby,M.,Lehtonen,R.,Roca,V.,和R.Walsh,“长笛-单向传输上的文件交付”,RFC 3926,DOI 10.17487/RFC3926,2004年10月<>.

[RFC3971] Arkko, J., Ed., Kempf, J., Zill, B., and P. Nikander, "SEcure Neighbor Discovery (SEND)", RFC 3971, DOI 10.17487/RFC3971, March 2005, <>.

[RFC3971]Arkko,J.,Ed.,Kempf,J.,Zill,B.,和P.Nikander,“安全邻居发现(SEND)”,RFC 3971,DOI 10.17487/RFC3971,2005年3月<>.

[RFC4336] Floyd, S., Handley, M., and E. Kohler, "Problem Statement for the Datagram Congestion Control Protocol (DCCP)", RFC 4336, DOI 10.17487/RFC4336, March 2006, <>.

[RFC4336]Floyd,S.,Handley,M.,和E.Kohler,“数据报拥塞控制协议(DCCP)的问题陈述”,RFC 4336,DOI 10.17487/RFC4336,2006年3月<>.

[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram Congestion Control Protocol (DCCP)", RFC 4340, DOI 10.17487/RFC4340, March 2006, <>.

[RFC4340]Kohler,E.,Handley,M.和S.Floyd,“数据报拥塞控制协议(DCCP)”,RFC 4340,DOI 10.17487/RFC4340,2006年3月<>.

[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 2: TCP-like Congestion Control", RFC 4341, DOI 10.17487/RFC4341, March 2006, <>.

[RFC4341]Floyd,S.和E.Kohler,“数据报拥塞控制协议(DCCP)拥塞控制ID 2的配置文件:类似TCP的拥塞控制”,RFC 4341,DOI 10.17487/RFC4341,2006年3月<>.

[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, DOI 10.17487/RFC4342, March 2006, <>.

[RFC4342]Floyd,S.,Kohler,E.,和J.Padhye,“数据报拥塞控制协议(DCCP)拥塞控制ID 3的配置文件:TCP友好速率控制(TFRC)”,RFC 4342,DOI 10.17487/RFC4342,2006年3月<>.

[RFC4443] Conta, A., Deering, S., and M. Gupta, Ed., "Internet Control Message Protocol (ICMPv6) for the Internet Protocol Version 6 (IPv6) Specification", RFC 4443, DOI 10.17487/RFC4443, March 2006, <>.

[RFC4443]Conta,A.,Deering,S.,和M.Gupta,Ed.,“互联网协议版本6(IPv6)规范的互联网控制消息协议(ICMPv6)”,RFC 4443,DOI 10.17487/RFC4443,2006年3月<>.

[RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast Congestion Control (TFMCC): Protocol Specification", RFC 4654, DOI 10.17487/RFC4654, August 2006, <>.

[RFC4654]Widmer,J.和M.Handley,“TCP友好多播拥塞控制(TFMCC):协议规范”,RFC 4654,DOI 10.17487/RFC4654,2006年8月<>.

[RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and Parameter for the Stream Control Transmission Protocol (SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007, <>.

[RFC4820]Tuexen,M.,Stewart,R.,和P.Lei,“流控制传输协议(SCTP)的填充块和参数”,RFC 4820,DOI 10.17487/RFC4820,2007年3月<>.

[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007, <>.

[RFC4821]Mathis,M.和J.Heffner,“打包层路径MTU发现”,RFC 4821,DOI 10.17487/RFC4821,2007年3月<>.

[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control (TFRC): The Small-Packet (SP) Variant", RFC 4828, DOI 10.17487/RFC4828, April 2007, <>.

[RFC4828]Floyd,S.和E.Kohler,“TCP友好速率控制(TFRC):小数据包(SP)变体”,RFC 4828,DOI 10.17487/RFC4828,2007年4月<>.

[RFC4861] Narten, T., Nordmark, E., Simpson, W., and H. Soliman, "Neighbor Discovery for IP version 6 (IPv6)", RFC 4861, DOI 10.17487/RFC4861, September 2007, <>.

[RFC4861]Narten,T.,Nordmark,E.,Simpson,W.,和H.Soliman,“IP版本6(IPv6)的邻居发现”,RFC 4861,DOI 10.17487/RFC48612007年9月<>.

[RFC4895] Tuexen, M., Stewart, R., Lei, P., and E. Rescorla, "Authenticated Chunks for the Stream Control Transmission Protocol (SCTP)", RFC 4895, DOI 10.17487/RFC4895, August 2007, <>.

[RFC4895]Tuexen,M.,Stewart,R.,Lei,P.,和E.Rescorla,“流控制传输协议(SCTP)的认证块”,RFC 4895,DOI 10.17487/RFC4895,2007年8月<>.

[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol", RFC 4960, DOI 10.17487/RFC4960, September 2007, <>.

[RFC4960]Stewart,R.,Ed.“流控制传输协议”,RFC 4960,DOI 10.17487/RFC4960,2007年9月<>.

[RFC5052] Watson, M., Luby, M., and L. Vicisano, "Forward Error Correction (FEC) Building Block", RFC 5052, DOI 10.17487/RFC5052, August 2007, <>.

[RFC5052]Watson,M.,Luby,M.,和L.Vicisano,“前向纠错(FEC)构造块”,RFC 5052,DOI 10.17487/RFC5052,2007年8月<>.

[RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. Kozuka, "Stream Control Transmission Protocol (SCTP) Dynamic Address Reconfiguration", RFC 5061, DOI 10.17487/RFC5061, September 2007, <>.

[RFC5061]Stewart,R.,Xie,Q.,Tuexen,M.,Maruyama,S.,和M.Kozuka,“流控制传输协议(SCTP)动态地址重新配置”,RFC 5061,DOI 10.17487/RFC5061,2007年9月<>.

[RFC5097] Renker, G. and G. Fairhurst, "MIB for the UDP-Lite protocol", RFC 5097, DOI 10.17487/RFC5097, January 2008, <>.

[RFC5097]Renker,G.和G.Fairhurst,“UDP Lite协议的MIB”,RFC 5097,DOI 10.17487/RFC5097,2008年1月<>.

[RFC5238] Phelan, T., "Datagram Transport Layer Security (DTLS) over the Datagram Congestion Control Protocol (DCCP)", RFC 5238, DOI 10.17487/RFC5238, May 2008, <>.

[RFC5238]Phelan,T.,“数据报拥塞控制协议(DCCP)上的数据报传输层安全性(DTLS)”,RFC 5238,DOI 10.17487/RFC5238,2008年5月<>.

[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS) Protocol Version 1.2", RFC 5246, DOI 10.17487/RFC5246, August 2008, <>.

[RFC5246]Dierks,T.和E.Rescorla,“传输层安全(TLS)协议版本1.2”,RFC 5246,DOI 10.17487/RFC5246,2008年8月<>.

[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP Friendly Rate Control (TFRC): Protocol Specification", RFC 5348, DOI 10.17487/RFC5348, September 2008, <>.

[RFC5348]Floyd,S.,Handley,M.,Padhye,J.,和J.Widmer,“TCP友好速率控制(TFRC):协议规范”,RFC 5348,DOI 10.17487/RFC5348,2008年9月<>.

[RFC5461] Gont, F., "TCP's Reaction to Soft Errors", RFC 5461, DOI 10.17487/RFC5461, February 2009, <>.

[RFC5461]Gont,F.,“TCP对软错误的反应”,RFC 5461,DOI 10.17487/RFC5461,2009年2月<>.

[RFC5595] Fairhurst, G., "The Datagram Congestion Control Protocol (DCCP) Service Codes", RFC 5595, DOI 10.17487/RFC5595, September 2009, <>.

[RFC5595]Fairhurst,G.“数据报拥塞控制协议(DCCP)服务代码”,RFC 5595,DOI 10.17487/RFC5595,2009年9月<>.

[RFC5596] Fairhurst, G., "Datagram Congestion Control Protocol (DCCP) Simultaneous-Open Technique to Facilitate NAT/ Middlebox Traversal", RFC 5596, DOI 10.17487/RFC5596, September 2009, <>.

[RFC5596]Fairhurst,G.,“数据报拥塞控制协议(DCCP)同时开放技术促进NAT/中间盒遍历”,RFC 5596,DOI 10.17487/RFC5596,2009年9月<>.

[RFC5622] Floyd, S. and E. Kohler, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate Control for Small Packets (TFRC-SP)", RFC 5622, DOI 10.17487/RFC5622, August 2009, <>.

[RFC5622]Floyd,S.和E.Kohler,“数据报拥塞控制协议(DCCP)拥塞ID 4的配置文件:小数据包的TCP友好速率控制(TFRC-SP)”,RFC 5622,DOI 10.17487/RFC5622,2009年8月<>.

[RFC5651] Luby, M., Watson, M., and L. Vicisano, "Layered Coding Transport (LCT) Building Block", RFC 5651, DOI 10.17487/RFC5651, October 2009, <>.

[RFC5651]Luby,M.,Watson,M.,和L.Vicisano,“分层编码传输(LCT)构建块”,RFC 5651,DOI 10.17487/RFC5651,2009年10月<>.

[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion Control", RFC 5681, DOI 10.17487/RFC5681, September 2009, <>.

[RFC5681]Allman,M.,Paxson,V.和E.Blanton,“TCP拥塞控制”,RFC 5681,DOI 10.17487/RFC56812009年9月<>.

[RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker, "NACK-Oriented Reliable Multicast (NORM) Transport Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009, <>.

[RFC5740]Adamson,B.,Bormann,C.,Handley,M.,和J.Macker,“面向NACK的可靠多播(NORM)传输协议”,RFC 5740,DOI 10.17487/RFC5740,2009年11月<>.

[RFC5762] Perkins, C., "RTP and the Datagram Congestion Control Protocol (DCCP)", RFC 5762, DOI 10.17487/RFC5762, April 2010, <>.

[RFC5762]Perkins,C.,“RTP和数据报拥塞控制协议(DCCP)”,RFC 5762,DOI 10.17487/RFC5762,2010年4月<>.

[RFC5775] Luby, M., Watson, M., and L. Vicisano, "Asynchronous Layered Coding (ALC) Protocol Instantiation", RFC 5775, DOI 10.17487/RFC5775, April 2010, <>.

[RFC5775]Luby,M.,Watson,M.,和L.Vicisano,“异步分层编码(ALC)协议实例化”,RFC 5775,DOI 10.17487/RFC5775,2010年4月<>.

[RFC6056] Larsen, M. and F. Gont, "Recommendations for Transport-Protocol Port Randomization", BCP 156, RFC 6056, DOI 10.17487/RFC6056, January 2011, <>.

[RFC6056]Larsen,M.和F.Gont,“运输协议端口随机化建议”,BCP 156,RFC 6056,DOI 10.17487/RFC6056,2011年1月<>.

[RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram Transport Layer Security (DTLS) for Stream Control Transmission Protocol (SCTP)", RFC 6083, DOI 10.17487/RFC6083, January 2011, <>.

[RFC6083]Tuexen,M.,Seggelmann,R.,和E.Rescorla,“流控制传输协议(SCTP)的数据报传输层安全性(DTLS)”,RFC 6083,DOI 10.17487/RFC6083,2011年1月<>.

[RFC6093] Gont, F. and A. Yourtchenko, "On the Implementation of the TCP Urgent Mechanism", RFC 6093, DOI 10.17487/RFC6093, January 2011, <>.

[RFC6093]Gont,F.和A.Yourtchenko,“关于TCP紧急机制的实施”,RFC 6093,DOI 10.17487/RFC6093,2011年1月<>.

[RFC6101] Freier, A., Karlton, P., and P. Kocher, "The Secure Sockets Layer (SSL) Protocol Version 3.0", RFC 6101, DOI 10.17487/RFC6101, August 2011, <>.

[RFC6101]Freier,A.,Karlton,P.,和P.Kocher,“安全套接字层(SSL)协议版本3.0”,RFC 6101,DOI 10.17487/RFC6101,2011年8月<>.

[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347, January 2012, <>.

[RFC6347]Rescorla,E.和N.Modadugu,“数据报传输层安全版本1.2”,RFC 6347,DOI 10.17487/RFC6347,2012年1月<>.

[RFC6356] Raiciu, C., Handley, M., and D. Wischik, "Coupled Congestion Control for Multipath Transport Protocols", RFC 6356, DOI 10.17487/RFC6356, October 2011, <>.

[RFC6356]Raiciu,C.,Handley,M.,和D.Wischik,“多路径传输协议的耦合拥塞控制”,RFC 6356,DOI 10.17487/RFC6356,2011年10月<>.

[RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error Correction (FEC) Framework", RFC 6363, DOI 10.17487/RFC6363, October 2011, <>.

[RFC6363]Watson,M.,Begen,A.和V.Roca,“前向纠错(FEC)框架”,RFC 6363,DOI 10.17487/RFC6363,2011年10月<>.

[RFC6458] Stewart, R., Tuexen, M., Poon, K., Lei, P., and V. Yasevich, "Sockets API Extensions for the Stream Control Transmission Protocol (SCTP)", RFC 6458, DOI 10.17487/RFC6458, December 2011, <>.

[RFC6458]Stewart,R.,Tuexen,M.,Poon,K.,Lei,P.,和V.Yasevich,“流控制传输协议(SCTP)的套接字API扩展”,RFC 6458,DOI 10.17487/RFC6458,2011年12月<>.

[RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control Transmission Protocol (SCTP) Stream Reconfiguration", RFC 6525, DOI 10.17487/RFC6525, February 2012, <>.

[RFC6525]Stewart,R.,Tuexen,M.,和P.Lei,“流控制传输协议(SCTP)流重新配置”,RFC 6525,DOI 10.17487/RFC6525,2012年2月<>.

[RFC6582] Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The NewReno Modification to TCP's Fast Recovery Algorithm", RFC 6582, DOI 10.17487/RFC6582, April 2012, <>.

[RFC6582]Henderson,T.,Floyd,S.,Gurtov,A.,和Y.Nishida,“TCP快速恢复算法的NewReno修改”,RFC 6582,DOI 10.17487/RFC6582,2012年4月<>.

[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, <>.

[RFC6584]Roca,V.“异步分层编码(ALC)和面向NACK的可靠多播(NORM)协议的简单认证方案”,RFC 6584,DOI 10.17487/RFC6584,2012年4月<>.

[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, <>.

[RFC6726]Paila,T.,Walsh,R.,Luby,M.,Roca,V.,和R.Lehtonen,“长笛-单向传输上的文件交付”,RFC 6726,DOI 10.17487/RFC6726,2012年11月<>.

[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, <>.

[RFC6773]Phelan,T.,Fairhurst,G.,和C.Perkins,“DCCP-UDP:NAT穿越的数据报拥塞控制协议UDP封装”,RFC 6773,DOI 10.17487/RFC6773,2012年11月<>.

[RFC6817] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, "Low Extra Delay Background Transport (LEDBAT)", RFC 6817, DOI 10.17487/RFC6817, December 2012, <>.

[RFC6817]Shalunov,S.,Hazel,G.,Iyengar,J.,和M.Kuehlewind,“低额外延迟背景传输(LEDBAT)”,RFC 6817,DOI 10.17487/RFC6817,2012年12月<>.

[RFC6824] Ford, A., Raiciu, C., Handley, M., and O. Bonaventure, "TCP Extensions for Multipath Operation with Multiple Addresses", RFC 6824, DOI 10.17487/RFC6824, January 2013, <>.

[RFC6824]Ford,A.,Raiciu,C.,Handley,M.,和O.Bonaventure,“多地址多路径操作的TCP扩展”,RFC 6824DOI 10.17487/RFC68242013年1月<>.

[RFC6897] Scharf, M. and A. Ford, "Multipath TCP (MPTCP) Application Interface Considerations", RFC 6897, DOI 10.17487/RFC6897, March 2013, <>.

[RFC6897]Scharf,M.和A.Ford,“多路径TCP(MPTCP)应用程序接口注意事项”,RFC 6897,DOI 10.17487/RFC6897,2013年3月<>.

[RFC6935] Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and UDP Checksums for Tunneled Packets", RFC 6935, DOI 10.17487/RFC6935, April 2013, <>.

[RFC6935]Eubanks,M.,Chimento,P.,和M.Westerlund,“隧道数据包的IPv6和UDP校验和”,RFC 6935,DOI 10.17487/RFC6935,2013年4月<>.

[RFC6936] Fairhurst, G. and M. Westerlund, "Applicability Statement for the Use of IPv6 UDP Datagrams with Zero Checksums", RFC 6936, DOI 10.17487/RFC6936, April 2013, <>.

[RFC6936]Fairhurst,G.和M.Westerlund,“使用具有零校验和的IPv6 UDP数据报的适用性声明”,RFC 6936,DOI 10.17487/RFC6936,2013年4月<>.

[RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream Control Transmission Protocol (SCTP) Packets for End-Host to End-Host Communication", RFC 6951, DOI 10.17487/RFC6951, May 2013, <>.

[RFC6951]Tuexen,M.和R.Stewart,“用于端主机到端主机通信的流控制传输协议(SCTP)数据包的UDP封装”,RFC 6951,DOI 10.17487/RFC6951,2013年5月<>.

[RFC7053] Tuexen, M., Ruengeler, I., and R. Stewart, "SACK-IMMEDIATELY Extension for the Stream Control Transmission Protocol", RFC 7053, DOI 10.17487/RFC7053, November 2013, <>.

[RFC7053]Tuexen,M.,Ruengeler,I.,和R.Stewart,“流控制传输协议的SACK-立即扩展”,RFC 7053,DOI 10.17487/RFC7053,2013年11月<>.

[RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security Solution", RFC 7202, DOI 10.17487/RFC7202, April 2014, <>.

[RFC7202]Perkins,C.和M.Westerlund,“保护RTP框架:为什么RTP不要求单一媒体安全解决方案”,RFC 7202,DOI 10.17487/RFC7202,2014年4月<>.

[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, <>.

[RFC7230]Fielding,R.,Ed.和J.Reschke,Ed.,“超文本传输协议(HTTP/1.1):消息语法和路由”,RFC 7230,DOI 10.17487/RFC7230,2014年6月<>.

[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, <>.

[RFC7231]Fielding,R.,Ed.和J.Reschke,Ed.,“超文本传输协议(HTTP/1.1):语义和内容”,RFC 7231,DOI 10.17487/RFC72312014年6月<>.

[RFC7232] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer Protocol (HTTP/1.1): Conditional Requests", RFC 7232, DOI 10.17487/RFC7232, June 2014, <>.

[RFC7232]Fielding,R.,Ed.和J.Reschke,Ed.,“超文本传输协议(HTTP/1.1):有条件请求”,RFC 7232,DOI 10.17487/RFC72322014年6月<>.

[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, <>.

[RFC7233]Fielding,R.,Ed.,Lafon,Y.,Ed.,和J.Reschke,Ed.,“超文本传输协议(HTTP/1.1):范围请求”,RFC 7233,DOI 10.17487/RFC7233,2014年6月<>.

[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, <>.

[RFC7234]Fielding,R.,Ed.,Nottingham,M.,Ed.,和J.Reschke,Ed.,“超文本传输协议(HTTP/1.1):缓存”,RFC 7234,DOI 10.17487/RFC72342014年6月<>.

[RFC7235] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer Protocol (HTTP/1.1): Authentication", RFC 7235, DOI 10.17487/RFC7235, June 2014, <>.

[RFC7235]Fielding,R.,Ed.和J.Reschke,Ed.,“超文本传输协议(HTTP/1.1):认证”,RFC 7235,DOI 10.17487/RFC7235,2014年6月<>.

[RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan, "Transport Layer Security (TLS) Application-Layer Protocol Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, July 2014, <>.

[RFC7301]Friedl,S.,Popov,A.,Langley,A.,和E.Stephan,“传输层安全(TLS)应用层协议协商扩展”,RFC 7301,DOI 10.17487/RFC7301,2014年7月<>.

[RFC7323] Borman, D., Braden, B., Jacobson, V., and R. Scheffenegger, Ed., "TCP Extensions for High Performance", RFC 7323, DOI 10.17487/RFC7323, September 2014, <>.

[RFC7323]Borman,D.,Braden,B.,Jacobson,V.,和R.Scheffenegger,编辑,“高性能TCP扩展”,RFC 7323,DOI 10.17487/RFC73232014年9月<>.

[RFC7414] Duke, M., Braden, R., Eddy, W., Blanton, E., and A. Zimmermann, "A Roadmap for Transmission Control Protocol (TCP) Specification Documents", RFC 7414, DOI 10.17487/RFC7414, February 2015, <>.

[RFC7414]杜克,M.,布拉登,R.,艾迪,W.,布兰顿,E.,和A.齐默尔曼,“传输控制协议(TCP)规范文件路线图”,RFC 7414,DOI 10.17487/RFC7414,2015年2月<>.

[RFC7457] Sheffer, Y., Holz, R., and P. Saint-Andre, "Summarizing Known Attacks on Transport Layer Security (TLS) and Datagram TLS (DTLS)", RFC 7457, DOI 10.17487/RFC7457, February 2015, <>.

[RFC7457]Sheffer,Y.,Holz,R.,和P.Saint Andre,“总结对传输层安全(TLS)和数据报TLS(DTLS)的已知攻击”,RFC 7457,DOI 10.17487/RFC7457,2015年2月<>.

[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, <>.

[RFC7496]Tuexen,M.,Seggelmann,R.,Stewart,R.,和S.Loreto,“部分可靠流控制传输协议扩展的附加策略”,RFC 7496,DOI 10.17487/RFC7496,2015年4月<>.

[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, DOI 10.17487/RFC7525, May 2015, <>.

[RFC7525]Sheffer,Y.,Holz,R.,和P.Saint Andre,“安全使用传输层安全性(TLS)和数据报传输层安全性(DTLS)的建议”,BCP 195,RFC 7525,DOI 10.17487/RFC7525,2015年5月<>.

[RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext Transfer Protocol Version 2 (HTTP/2)", RFC 7540, DOI 10.17487/RFC7540, May 2015, <>.

[RFC7540]Belshe,M.,Paon,R.,和M.Thomson,编辑,“超文本传输协议版本2(HTTP/2)”,RFC 7540,DOI 10.17487/RFC7540,2015年5月<>.

[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085, March 2017, <>.

[RFC8085]Eggert,L.,Fairhurst,G.和G.Shepherd,“UDP使用指南”,BCP 145,RFC 8085,DOI 10.17487/RFC8085,2017年3月<>.

[SCTP-DTLS-ENCAPS] Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS Encapsulation of SCTP Packets", Work in Progress, draft-ietf-tsvwg-sctp-dtls-encaps-09, January 2015.


[SCTP-NDATA] Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "Stream Schedulers and User Message Interleaving for the Stream Control Transmission Protocol", Work in Progress, draft-ietf-tsvwg-sctp-ndata-08, October 2016.


[TCP-SPEC] Eddy, W., Ed., "Transmission Control Protocol Specification", Work in Progress, draft-ietf-tcpm-rfc793bis-04, December 2016.


[TLS-1.3] Rescorla, E., "The Transport Layer Security (TLS) Protocol Version 1.3", Work in Progress, draft-ietf-tls-tls13-18, October 2016.


[WEBRTC-TRANS] Alvestrand, H., "Transports for WebRTC", Work in Progress, draft-ietf-rtcweb-transports-17, October 2016.


[XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen, "XMLHttpRequest Level 1", World Wide Web Consortium NOTE-XMLHttpRequest-20161006, October 2016, <>.

[XHR]van Kesteren,A.,Aubourg,J.,Song,J.,和H.Steen,“XMLHttpRequest 1级”,万维网联盟NOTE-XMLHttpRequest-20161006,2016年10月<>.



Thanks to Joe Touch, Michael Welzl, Spencer Dawkins, and the TAPS working group for the comments, feedback, and discussion. This work is supported by the European Commission under grant agreement No. 318627 mPlane and from the Horizon 2020 research and innovation program under grant agreements No. 644334 (NEAT) and No. 688421 (MAMI). This support does not imply endorsement.

感谢Joe Touch、Michael Welzl、Spencer Dawkins和TAPS工作组的评论、反馈和讨论。这项工作由欧盟委员会根据第318627号mPlane赠款协议以及地平线2020研究与创新计划根据第644334号(NEAT)和第688421号(MAMI)赠款协议提供支持。这种支持并不意味着认可。



In addition to the editors, this document is the work of Brian Adamson, Dragana Damjanovic, Kevin Fall, Simone Ferlin-Oliviera, Ralph Holz, Olivier Mehani, Karen Nielsen, Colin Perkins, Vincent Roca, and Michael Tuexen.


o Section 3.2 on MPTCP was contributed by Simone Ferlin-Oliviera ( and Olivier Mehani (

o 关于MPTCP的第3.2节由Simone Ferlin Oliviera提供(和奥利维尔·梅哈尼(奥利维尔。

o Section 3.3 on UDP was contributed by Kevin Fall (

o 关于UDP的第3.3节由Kevin Fall提供(

o Section 3.5 on SCTP was contributed by Michael Tuexen ( and Karen Nielsen (

o 关于SCTP的第3.5节由Michael Tuexen提供(和Karen Nielsen(Karen。

o Section 3.7 on TLS and DTLS was contributed by Ralph Holz ( and Olivier Mehani (

o 关于TLS和DTL的第3.7节由Ralph Holz(Ralph。和奥利维尔·梅哈尼(奥利维尔。

o Section 3.8 on RTP contains contributions from Colin Perkins (

o 关于RTP的第3.8节包含Colin Perkins的贡献(

o Section 3.9 on HTTP was contributed by Dragana Damjanovic (

o 关于HTTP的第3.9节由Dragana Damjanovic提供(

o Section 3.10 on FLUTE/ALC was contributed by Vincent Roca (

o 关于长笛/ALC的第3.10节由Vincent Roca(Vincent。

o Section 3.11 on NORM was contributed by Brian Adamson (

o 关于规范的第3.11节由Brian Adamson(Brian。

Authors' Addresses


Godred Fairhurst (editor) University of Aberdeen School of Engineering, Fraser Noble Building Aberdeen AB24 3UE

Godred Fairhurst(编辑)阿伯丁大学工程学院,弗雷泽贵族大厦阿伯丁AB24 3UE


Brian Trammell (editor) ETH Zurich Gloriastrasse 35 8092 Zurich Switzerland

布莱恩·特拉梅尔(编辑)ETH苏黎世Gloriastrasse 35 8092苏黎世瑞士


Mirja Kuehlewind (editor) ETH Zurich Gloriastrasse 35 8092 Zurich Switzerland

Mirja Kuehlewind(编辑)ETH苏黎世Gloriastrasse 35 8092苏黎世瑞士