UDP

This document presents two YANG 1.1 modules with reusable groupings for managing UDP clients and UDP servers.

This document addresses the problem of protocol support for measuring One-Way IP Capacity metrics specified by RFC 9097. The Method of Measurement discussed in that RFC requires a feedback channel from the receiver to control the sender's transmission rate in near real-time. This document defines the UDP Speed Test Protocol (UDPSTP) for conducting measurements as described in RFC 9097 and other related measurements.

Transport protocols are extended through the use of transport header options. This document updates RFC 768 (UDP) by indicating the location, syntax, and semantics for UDP transport layer options within the surplus area after the end of the UDP user data but before the end of the IP datagram.

This document specifies how a UDP Options sender implements Datagram Packetization Layer Path MTU Discovery (DPLPMTUD) as a robust method for Path MTU Discovery (PMTUD). This method uses the UDP Options packetization layer. It allows an application to discover the largest size of datagram that can be sent across a network path. It also provides a way to allow the application to periodically verify the current Maximum Packet Size (MPS) supported by a path and to update this when required.

This document specifies new IP Flow Information Export (IPFIX) Information Elements for UDP Options.

The widely deployed Extension Mechanisms for DNS (EDNS(0)) feature in the DNS enables a DNS receiver to indicate its received UDP message size capacity, which supports the sending of large UDP responses by a DNS server. Large DNS/UDP messages are more likely to be fragmented, and IP fragmentation has exposed weaknesses in application protocols. It is possible to avoid IP fragmentation in DNS by limiting the response size where possible and signaling the need to upgrade from UDP to TCP transport where necessary. This document describes techniques to avoid IP fragmentation in DNS.

This document describes how the DetNet IP data plane can support the Packet Replication, Elimination, and Ordering Functions (PREOF) built on the existing MPLS PREOF solution defined for the DetNet MPLS data plane and the mechanisms defined by MPLS-over-UDP technology.

This document describes how to proxy IP packets in HTTP. This protocol is similar to UDP proxying in HTTP but allows transmitting arbitrary IP packets. More specifically, this document defines a protocol that allows an HTTP client to create an IP tunnel through an HTTP server that acts as an IP proxy. This document updates RFC 9298.

This document describes a YANG data model for the Static Context Header Compression (SCHC) compression and fragmentation Rules.

This document formalizes the description of the Rules for better interoperability between SCHC instances either to exchange a set of Rules or to modify the parameters of some Rules.

This document describes how to proxy UDP in HTTP, similar to how the HTTP CONNECT method allows proxying TCP in HTTP. More specifically, this document defines a protocol that allows an HTTP client to create a tunnel for UDP communications through an HTTP server that acts as a proxy.

This document defines the properties of the QUIC transport protocol that are common to all versions of the protocol.

This document defines the core of the QUIC transport protocol. QUIC provides applications with flow-controlled streams for structured communication, low-latency connection establishment, and network path migration. QUIC includes security measures that ensure confidentiality, integrity, and availability in a range of deployment circumstances. Accompanying documents describe the integration of TLS for key negotiation, loss detection, and an exemplary congestion control algorithm.

This document specifies the MPLS Deterministic Networking (DetNet) data plane operation and encapsulation over an IP network. The approach is based on the operation of MPLS-over-UDP technology.

The Static Context Header Compression and fragmentation (SCHC) specification (RFC 8724) describes generic header compression and fragmentation techniques for Low-Power Wide Area Network (LPWAN) technologies. SCHC is a generic mechanism designed for great flexibility so that it can be adapted for any of the LPWAN technologies.

This document defines a profile of SCHC (RFC 8724) for use in LoRaWAN networks and provides elements such as efficient parameterization and modes of operation.

This document specifies Datagram Packetization Layer Path MTU Discovery (DPLPMTUD). This is a robust method for Path MTU Discovery (PMTUD) for datagram Packetization Layers (PLs). It allows a PL, or a datagram application that uses a PL, to discover whether a network path can support the current size of datagram. This can be used to detect and reduce the message size when a sender encounters a packet black hole. It can also probe a network path to discover whether the maximum packet size can be increased. This provides functionality for datagram transports that is equivalent to the PLPMTUD specification for TCP, specified in RFC 4821, which it updates. It also updates the UDP Usage Guidelines to refer to this method for use with UDP datagrams and updates SCTP.

The document provides implementation notes for incorporating Datagram PMTUD into IETF datagram transports or applications that use datagram transports.

This specification updates RFC 4960, RFC 4821, RFC 6951, RFC 8085, and RFC 8261.

This document defines the Static Context Header Compression and fragmentation (SCHC) framework, which provides both a header compression mechanism and an optional fragmentation mechanism. SCHC has been designed with Low-Power Wide Area Networks (LPWANs) in mind.

SCHC compression is based on a common static context stored both in the LPWAN device and in the network infrastructure side. This document defines a generic header compression mechanism and its application to compress IPv6/UDP headers.

This document also specifies an optional fragmentation and reassembly mechanism. It can be used to support the IPv6 MTU requirement over the LPWAN technologies. Fragmentation is needed for IPv6 datagrams that, after SCHC compression or when such compression was not possible, still exceed the Layer 2 maximum payload size.

The SCHC header compression and fragmentation mechanisms are independent of the specific LPWAN technology over which they are used. This document defines generic functionalities and offers flexibility with regard to parameter settings and mechanism choices. This document standardizes the exchange over the LPWAN between two SCHC entities. Settings and choices specific to a technology or a product are expected to be grouped into profiles, which are specified in other documents. Data models for the context and profiles are out of scope.

MPLS Segment Routing (SR-MPLS) is a method of source routing a packet through an MPLS data plane by imposing a stack of MPLS labels on the packet to specify the path together with any packet-specific instructions to be executed on it. SR-MPLS can be leveraged to realize a source-routing mechanism across MPLS, IPv4, and IPv6 data planes by using an MPLS label stack as a source-routing instruction set while making no changes to SR-MPLS specifications and interworking with SR-MPLS implementations.

This document describes how SR-MPLS-capable routers and IP-only routers can seamlessly coexist and interoperate through the use of SR-MPLS label stacks and IP encapsulation/tunneling such as MPLS-over-UDP as defined in RFC 7510.

This document summarizes an operator's perception of the benefits that may be provided by intermediary devices that execute functions beyond normal IP forwarding. Such intermediary devices are often called "middleboxes".

RFC 3234 defines a taxonomy of middleboxes and issues in the Internet. Most of those middleboxes utilize or modify application- layer data. This document primarily focuses on devices that observe and act on information carried in the transport layer, and especially information carried in TCP packets.

This document defines an extension to the DHCP protocols that allows a relay agent to use any available source port for upstream communications. The extension also allows inclusion of a DHCP option that can be used to statelessly route responses back to the appropriate source port on downstream communications.

This document explains what is meant by the term "network transport Circuit Breaker". It describes the need for Circuit Breakers (CBs) for network tunnels and applications when using non-congestion- controlled traffic and explains where CBs are, and are not, needed. It also defines requirements for building a CB and the expected outcomes of using a CB within the Internet.

The User Datagram Protocol (UDP) provides a minimal message-passing transport that has no inherent congestion control mechanisms. This document provides guidelines on the use of UDP for the designers of applications, tunnels, and other protocols that use UDP. Congestion control guidelines are a primary focus, but the document also provides guidance on other topics, including message sizes, reliability, checksums, middlebox traversal, the use of Explicit Congestion Notification (ECN), Differentiated Services Code Points (DSCPs), and ports.

Because congestion control is critical to the stable operation of the Internet, applications and other protocols that choose to use UDP as an Internet transport must employ mechanisms to prevent congestion collapse and to establish some degree of fairness with concurrent traffic. They may also need to implement additional mechanisms, depending on how they use UDP.

Some guidance is also applicable to the design of other protocols (e.g., protocols layered directly on IP or via IP-based tunnels), especially when these protocols do not themselves provide congestion control.

This document obsoletes RFC 5405 and adds guidelines for multicast UDP usage.

This document specifies a method of encapsulating network protocol packets within GRE and UDP headers. This GRE-in-UDP encapsulation allows the UDP source port field to be used as an entropy field. This may be used for load-balancing of GRE traffic in transit networks using existing Equal-Cost Multipath (ECMP) mechanisms. There are two applicability scenarios for GRE-in-UDP with different requirements: (1) general Internet and (2) a traffic-managed controlled environment. The controlled environment has less restrictive requirements than the general Internet.

RFC 6374 defines a protocol for Packet Loss and Delay Measurement for MPLS networks (MPLS-PLDM). This document specifies the procedures to be used when sending and processing out-of-band MPLS performance management Responses over an UDP/IP return path.

The One-Way Active Measurement Protocol (OWAMP) and the Two-Way Active Measurement Protocol (TWAMP) are used for performance monitoring in IP networks. Delay measurement is performed in these protocols by using timestamped test packets. Some implementations use hardware-based timestamping engines that integrate the accurate transmission time into every outgoing OWAMP/TWAMP test packet during transmission. Since these packets are transported over UDP, the UDP Checksum field is then updated to reflect this modification. This document proposes to use the last 2 octets of every test packet as a Checksum Complement, allowing timestamping engines to reflect the checksum modification in the last 2 octets rather than in the UDP Checksum field. The behavior defined in this document is completely interoperable with existing OWAMP/TWAMP implementations.

The Network Time Protocol (NTP) allows clients to synchronize to a time server using timestamped protocol messages. To facilitate accurate timestamping, some implementations use hardware-based timestamping engines that integrate the accurate transmission time into every outgoing NTP packet during transmission. Since these packets are transported over UDP, the UDP Checksum field is then updated to reflect this modification. This document proposes an extension field that includes a 2-octet Checksum Complement, allowing timestamping engines to reflect the checksum modification in the last 2 octets of the packet rather than in the UDP Checksum field. The behavior defined in this document is interoperable with existing NTP implementations.

The Internet Architecture Board (IAB) through its IP Stack Evolution program, the Internet Society, and the Swiss Federal Institute of Technology (ETH) Zurich hosted the Stack Evolution in a Middlebox Internet (SEMI) workshop in Zurich on 26-27 January 2015 to explore the ability to evolve the transport layer in the presence of middlebox- and interface-related ossification of the stack. The goal of the workshop was to produce architectural and engineering guidance on future work to break the logjam, focusing on incrementally deployable approaches with clear incentives to deployment both on the endpoints (in new transport layers and applications) as well as on middleboxes (run by network operators). This document summarizes the contributions to the workshop and provides an overview of the discussion at the workshop, as well as the outcomes and next steps identified by the workshop. The views and positions documented in this report are those of the workshop participants and do not necessarily reflect IAB views and positions.

This document describes several Network Address Translator (NAT) traversal techniques that were considered to be used for establishing the RTP media flows controlled by the Real-Time Streaming Protocol (RTSP). Each technique includes a description of how it would be used, the security implications of using it, and any other deployment considerations it has. There are also discussions on how NAT traversal techniques relate to firewalls and how each technique can be applied in different use cases. These findings were used when selecting the NAT traversal for RTSP 2.0, which is specified in a separate document.

This document provides recommendations to designers of application and service protocols on how to use the transport protocol port number space and when to request a port assignment from IANA. It provides designer guidance to requesters or users of port numbers on how to interact with IANA using the processes defined in RFC 6335; thus, this document complements (but does not update) that document.

This document specifies an IP-based encapsulation for MPLS, called MPLS-in-UDP for situations where UDP (User Datagram Protocol) encapsulation is preferred to direct use of MPLS, e.g., to enable UDP-based ECMP (Equal-Cost Multipath) or link aggregation. The MPLS- in-UDP encapsulation technology must only be deployed within a single network (with a single network operator) or networks of an adjacent set of cooperating network operators where traffic is managed to avoid congestion, rather than over the Internet where congestion control is required. Usage restrictions apply to MPLS-in-UDP usage for traffic that is not congestion controlled and to UDP zero checksum usage with IPv6.

This document specifies how the UDP Transport Layer (UDPTL) protocol, the predominant transport protocol for T.38 fax, can be transported over the Datagram Transport Layer Security (DTLS) protocol, how the usage of UDPTL over DTLS is indicated in the Session Description Protocol (SDP), and how UDPTL over DTLS is negotiated in a session established using the Session Initiation Protocol (SIP).

This document describes a simple method of encapsulating Stream Control Transmission Protocol (SCTP) packets into UDP packets and its limitations. This allows the usage of SCTP in networks with legacy NATs that do not support SCTP. It can also be used to implement SCTP on hosts without directly accessing the IP layer, for example, implementing it as part of the application without requiring special privileges.

Please note that this document only describes the functionality required within an SCTP stack to add on UDP encapsulation, providing only those mechanisms for two end-hosts to communicate with each other over UDP ports. In particular, it does not provide mechanisms to determine whether UDP encapsulation is being used by the peer, nor the mechanisms for determining which remote UDP port number can be used. These functions are out of scope for this document.

This document covers only end-hosts and not tunneling (egress or ingress) endpoints.

This document updates the IPv6 specification (RFC 2460) to improve performance when a tunnel protocol uses UDP with IPv6 to tunnel packets. The performance improvement is obtained by relaxing the IPv6 UDP checksum requirement for tunnel protocols whose header information is protected on the "inner" packet being carried. Relaxing this requirement removes the overhead associated with the computation of UDP checksums on IPv6 packets that carry the tunnel protocol packets. This specification describes how the IPv6 UDP checksum requirement can be relaxed when the encapsulated packet itself contains a checksum. It also describes the limitations and risks of this approach and discusses the restrictions on the use of this method.

This document provides an applicability statement for the use of UDP transport checksums with IPv6. It defines recommendations and requirements for the use of IPv6 UDP datagrams with a zero UDP checksum. It describes the issues and design principles that need to be considered when UDP is used with IPv6 to support tunnel encapsulations, and it examines the role of the IPv6 UDP transport checksum. The document also identifies issues and constraints for deployment on network paths that include middleboxes. An appendix presents a summary of the trade-offs that were considered in evaluating the safety of the update to RFC 2460 that changes the use of the UDP checksum with IPv6.

Cisco's Service-Level Assurance Protocol (Cisco's SLA Protocol) is a Performance Measurement protocol that has been widely deployed. The protocol is used to measure service-level parameters such as network latency, delay variation, and packet/frame loss. This document describes the Cisco SLA Protocol Measurement-Type UDP-Measurement, to enable vendor interoperability. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document specifies an alternative encapsulation of the Datagram Congestion Control Protocol (DCCP), referred to as DCCP-UDP. This encapsulation allows DCCP to be carried through the current generation of Network Address Translation (NAT) middleboxes without modification of those middleboxes. This document also updates the Session Description Protocol (SDP) information for DCCP defined in RFC 5762. [STANDARDS-TRACK]

This memo specifies how Explicit Congestion Notification (ECN) can be used with the Real-time Transport Protocol (RTP) running over UDP, using the RTP Control Protocol (RTCP) as a feedback mechanism. It defines a new RTCP Extended Report (XR) block for periodic ECN feedback, a new RTCP transport feedback message for timely reporting of congestion events, and a Session Traversal Utilities for NAT (STUN) extension used in the optional initialisation method using Interactive Connectivity Establishment (ICE). Signalling and procedures for negotiation of capabilities and initialisation methods are also defined. [STANDARDS-TRACK]

This document formally deprecates the use of ICMP Source Quench messages by transport protocols, formally updating RFC 792, RFC 1122, and RFC 1812. [STANDARDS-TRACK]

This document describes IAX, the Inter-Asterisk eXchange protocol, an application-layer control and media protocol for creating, modifying, and terminating multimedia sessions over Internet Protocol (IP) networks. IAX was developed by the open source community for the Asterisk Private Branch Exchange (PBX) and is targeted primarily at Voice over Internet Protocol (VoIP) call control, but it can be used with streaming video or any other type of multimedia.

IAX is an "all in one" protocol for handling multimedia in IP networks. It combines both control and media services in the same protocol. In addition, IAX uses a single UDP data stream on a static port greatly simplifying Network Address Translation (NAT) gateway traversal, eliminating the need for other protocols to work around NAT, and simplifying network and firewall management. IAX employs a compact encoding that decreases bandwidth usage and is well suited for Internet telephony service. In addition, its open nature permits new payload type additions needed to support additional services. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document describes the transport for syslog messages over UDP/ IPv4 or UDP/IPv6. The syslog protocol layered architecture provides for support of any number of transport mappings. However, for interoperability purposes, syslog protocol implementers are required to support this transport mapping. [STANDARDS-TRACK]

This document describes a lightweight UDP transfer protocol for the Internet Registry Information Service (IRIS). This transfer protocol uses a single packet for every request and response, and optionally employs compression over the contents of the packet. [STANDARDS-TRACK]

RFC 3095 defines the RObust Header Compression (ROHC) framework and profiles for IP (Internet Protocol), UDP (User Datagram Protocol), RTP (Real-Time Transport Protocol), and ESP (Encapsulating Security Payload). Some parts of the specification are unclear or contain errors that may lead to misinterpretations that may impair interoperability between different implementations. This document provides corrections, additions, and clarifications to RFC 3095; this document thus updates RFC 3095. In addition, other clarifications related to RFC 3241 (ROHC over PPP), RFC 3843 (ROHC IP profile) and RFC 4109 (ROHC UDP-Lite profiles) are also provided. [STANDARDS-TRACK]

This document defines basic terminology for describing different types of Network Address Translation (NAT) behavior when handling Unicast UDP and also defines a set of requirements that would allow many applications, such as multimedia communications or online gaming, to work consistently. Developing NATs that meet this set of requirements will greatly increase the likelihood that these applications will function properly. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.

When experimenting with or extending protocols, it is often necessary to use some sort of protocol number or constant in order to actually test or experiment with the new function, even when testing in a closed environment. This document reserves some ranges of numbers for experimentation purposes in specific protocols where the need to support experimentation has been identified, and it describes the numbers that have already been reserved by other documents. [STANDARDS-TRACK]

This memo defines a Media Delivery Index (MDI) measurement that can be used as a diagnostic tool or a quality indicator for monitoring a network intended to deliver applications such as streaming media, MPEG video, Voice over IP, or other information sensitive to arrival time and packet loss. It provides an indication of traffic jitter, a measure of deviation from nominal flow rates, and a data loss at-a-glance measure for a particular flow. For instance, the MDI may be used as a reference in characterizing and comparing networks carrying UDP streaming media. This memo provides information for the Internet community.

We propose here a service that enables nodes located behind one or more IPv4 Network Address Translations (NATs) to obtain IPv6 connectivity by tunneling packets over UDP; we call this the Teredo service. Running the service requires the help of "Teredo servers" and "Teredo relays". The Teredo servers are stateless, and only have to manage a small fraction of the traffic between Teredo clients; the Teredo relays act as IPv6 routers between the Teredo service and the "native" IPv6 Internet. The relays can also provide interoperability with hosts using other transition mechanisms such as "6to4". [STANDARDS-TRACK]

This document defines a ROHC (Robust Header Compression) profile for compression of IP/UDP/RTP (Internet Protocol/User Datagram Protocol/Real-Time Transport Protocol) packets, utilizing functionality provided by the lower layers to increase compression efficiency by completely eliminating the header for most packets during optimal operation. The profile is built as an extension to the ROHC RTP profile. It defines additional mechanisms needed in ROHC, states requirements on the assisting layer to guarantee transparency, and specifies general logic for compression and decompression related to the usage of the header-free packet format. This document is a replacement for RFC 3242, which it obsoletes. [STANDARDS-TRACK]

This memo defines a portion of the Management Information Base (MIB) for use with network management protocols in the Internet community. In particular, it describes managed objects used for implementations of the User Datagram Protocol (UDP) in an IP version independent manner. This memo obsoletes RFCs 2013 and 2454. [STANDARDS-TRACK]

This protocol specification defines methods to encapsulate and decapsulate IP Encapsulating Security Payload (ESP) packets inside UDP packets for traversing Network Address Translators. ESP encapsulation, as defined in this document, can be used in both IPv4 and IPv6 scenarios. Whenever negotiated, encapsulation is used with Internet Key Exchange (IKE). [STANDARDS-TRACK]

Mobile IP's datagram tunnelling is incompatible with Network Address Translation (NAT). This document presents extensions to the Mobile IP protocol and a tunnelling method which permits mobile nodes using Mobile IP to operate in private address networks which are separated from the public internet by NAT devices. The NAT traversal is based on using the Mobile IP Home Agent UDP port for encapsulated data traffic. [STANDARDS-TRACK]

This document describes lower layer guidelines for robust header compression (ROHC) and the requirements ROHC puts on lower layers. The purpose of this document is to support the incorporation of robust header compression algorithms, as specified in the ROHC working group, into different systems such as those specified by Third Generation Partnership Project (3GPP), 3GPP Project 2 (3GPP2), European Technical Standards Institute (ETSI), etc. This document covers only lower layer guidelines for compression of RTP/UDP/IP and UDP/IP headers as specified in [RFC3095]. Both general guidelines and guidelines specific for cellular systems are discussed in this document. This memo provides information for the Internet community.

This document contains requirements for the 0-byte IP/UDP/RTP (Internet Protocol/User Datagram Protocol/Real-Time Transport Protocol) header compression scheme to be developed by the Robust Header Compression (ROHC) Working Group. It also includes the basic assumptions for the typical link layers over which 0-byte compression may be implemented, and assumptions about its usage in general.

This document describes the applicability of the Stream Control Transmission Protocol (SCTP). It also contrasts SCTP with the two dominant transport protocols, User Datagram Protocol (UDP) & Transmission Control Protocol (TCP), and gives some guidelines for when best to use SCTP and when not best to use SCTP. This memo provides information for the Internet community.

This document specifies a highly robust and efficient header compression scheme for RTP/UDP/IP (Real-Time Transport Protocol, User Datagram Protocol, Internet Protocol), UDP/IP, and ESP/IP (Encapsulating Security Payload) headers. [STANDARDS-TRACK]

This document contains requirements for robust IP/UDP/RTP (Internet Protocol/User Datagram Protocol/Real-Time Transport Protocol) header compression to be developed by the ROHC (Robust Header Compression) WG. It is based on the ROHC charter, discussions in the WG, the 3GPP document "3GPP TR 23.922", version 1.0.0 of October 1999, as well as contributions from 3G.IP. This memo provides information for the Internet community.

RFC 1240, "OSI connectionless transport services on top of UDP", was published as a Proposed Standard in June 1991 but at this time there do not seem to be any implementations which follow RFC 1240. In addition there is a growing concern over using UDP-based transport protocols in environments where congestion is a possibility This memo provides information for the Internet community.

This document describes a method for compressing the headers of IP/UDP/RTP datagrams to reduce overhead on low-speed serial links. [STANDARDS-TRACK]

This document summarizes a range of possible techniques for the repair of continuous media streams subject to packet loss. This memo provides information for the Internet community. This memo does not specify an Internet standard of any kind.

This memo presents a techniques for efficiently implementing the Internet Checksum in hardware. It includes PLD code for programming a single, low cost part to perform checksumming at 1.26 Gbps. This memo provides information for the Internet community. This memo does not specify an Internet standard of any kind.

TCP/IPX allows TCP/IP applications to run over IPX networks by letting TCP and UDP run over IPX. And this memo specifies the packet format and operational procedures for running TCP and UDP over IPX. This document defines an Experimental Protocol for the Internet community. This does not specify an Internet standard of any kind.

This paper describes a simple proposal which provides a long-term solution to Internet addressing, routing, and scaling. This memo provides information for the Internet community. It does not specify an Internet standard.

This document describes a protocol for running OSI Connectionless service on UDP. [STANDARDS-TRACK]

This RFC defines a proposed standard protocol to support NetBIOS services in a TCP/IP environment. Both local network and internet operation are supported. Various node types are defined to accommodate local and internet topologies and to allow operation with or without the use of IP broadcast. This RFC describes the NetBIOS-over-TCP protocols in a general manner, emphasizing the underlying ideas and techniques. Detailed specifications are found in a companion RFC, "Protocol Standard For a NetBIOS Service on a TCP/UDP Transport: Detailed Specifications".

This RFC defines a proposed standard protocol to support NetBIOS services in a TCP/IP environment. Both local network and internet operation are supported. Various node types are defined to accommodate local and internet topologies and to allow operation with or without the use of IP broadcast. This RFC gives the detailed specifications of the netBIOS-over-TCP packets, protocols, and defined constants and variables. A more general overview is found in a companion RFC, "Protocol Standard For NetBIOS Service on TCP/UDP Transport: Concepts and Methods".