Опубликованные RFC, связи обновления и замены, официальные статусы и русские переводы.
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A JSON Meta Application Protocol (JMAP) Subprotocol for WebSocket
This document defines a binding for the JSON Meta Application Protocol (JMAP) over a WebSocket transport layer. The WebSocket binding for JMAP provides higher performance than the current HTTP binding for JMAP.
Secure Device Install
Deploying a new network device in a location where the operator has no staff of its own often requires that an employee physically travel to the location to perform the initial install and configuration, even in shared facilities with "remote-hands" (or similar) support. In many cases, this could be avoided if there were an easy way to transfer the initial configuration to a new device while still…
Proxy Mobile IPv6 Extensions for Distributed Mobility Management
Distributed Mobility Management solutions allow networks to be set up in such a way that traffic is distributed optimally and centrally deployed anchors are not relied upon to provide IP mobility support. There are many different approaches to address Distributed Mobility Management -- for example, extending network-based mobility protocols (like Proxy Mobile IPv6) or client-based mobility protoco…
Research Directions for Using Information-Centric Networking (ICN) in Disaster Scenarios
Information-Centric Networking (ICN) is a new paradigm where the network provides users with named content instead of communication channels between hosts. This document outlines some research directions for ICN with respect to applying ICN approaches for coping with natural or human-generated, large-scale disasters. This document is a product of the Information-Centric Networking Research Group (…
ICMPv6 Errors for Discarding Packets Due to Processing Limits
Network nodes may discard packets if they are unable to process protocol headers of packets due to processing constraints or limits. When such packets are dropped, the sender receives no indication, so it cannot take action to address the cause of discarded packets. This specification defines several new ICMPv6 errors that can be sent by a node that discards packets because it is unable to process…
DNS-Based Service Discovery (DNS-SD) Privacy and Security Requirements
DNS-SD (DNS-based Service Discovery) normally discloses information about devices offering and requesting services. This information includes hostnames, network parameters, and possibly a further description of the corresponding service instance. Especially when mobile devices engage in DNS-based Service Discovery at a public hotspot, serious privacy problems arise. We analyze the requirements of …
Network File System (NFS) Version 4 Minor Version 1 Protocol
This document describes the Network File System (NFS) version 4 minor version 1, including features retained from the base protocol (NFS version 4 minor version 0, which is specified in RFC 7530) and protocol extensions made subsequently. The later minor version has no dependencies on NFS version 4 minor version 0, and is considered a separate protocol. This document obsoletes RFC 5661. It substan…
Special Use Domain Name 'ipv4only.arpa'
NAT64 (Network Address and Protocol Translation from IPv6 Clients to IPv4 Servers) allows client devices using IPv6 to communicate with servers that have only IPv4 connectivity. The specification for how a client discovers its local network's NAT64 prefix (RFC 7050) defines the special name 'ipv4only.arpa' for this purpose. However, in its Domain Name Reservation Considerations section (Section 8.…
TLS Certificate Compression
In TLS handshakes, certificate chains often take up the majority of the bytes transmitted. This document describes how certificate chains can be compressed to reduce the amount of data transmitted and avoid some round trips.
Zstandard Compression and the 'application/zstd' Media Type
Zstandard, or "zstd" (pronounced "zee standard"), is a lossless data compression mechanism. This document describes the mechanism and registers a media type, content encoding, and a structured syntax suffix to be used when transporting zstd-compressed content via MIME. Despite use of the word "standard" as part of Zstandard, readers are advised that this document is not an Internet Standards Track…
Guidelines for Defining Packet Timestamps
Various network protocols make use of binary-encoded timestamps that are incorporated in the protocol packet format, referred to as "packet timestamps" for short. This document specifies guidelines for defining packet timestamp formats in networking protocols at various layers. It also presents three recommended timestamp formats. The target audience of this document includes network protocol desi…
Non-interactive Emergency Calls
Use of the Internet for emergency calling is described in RFC 6443, 'Framework for Emergency Calling Using Internet Multimedia'. In some cases of emergency calls, the transmission of application data is all that is needed, and no interactive media channel is established: a situation referred to as 'non-interactive emergency calls', where, unlike most emergency calls, there is no two-way interactiv…
Working Group GitHub Administration
The use of GitHub in IETF working group processes is increasing. This document describes uses and conventions for working groups that are considering starting to use GitHub. It does not mandate any processes and does not require changes to the processes used by current and future working groups not using GitHub.
Working Group GitHub Usage Guidance
This document provides a set of guidelines for working groups that choose to use GitHub for their work.
Message Session Relay Protocol (MSRP) over Data Channels
This document specifies how a Web Real-Time Communication (WebRTC) data channel can be used as a transport mechanism for the Message Session Relay Protocol (MSRP) and how the Session Description Protocol (SDP) offer/answer mechanism can be used to negotiate such a data channel, referred to as an MSRP data channel. Two network configurations are supported: the connection of two MSRP data channel en…
Guidelines for Using the Multiplexing Features of RTP to Support Multiple Media Streams
The Real-time Transport Protocol (RTP) is a flexible protocol that can be used in a wide range of applications, networks, and system topologies. That flexibility makes for wide applicability but can complicate the application design process. One particular design question that has received much attention is how to support multiple media streams in RTP. This memo discusses the available options and…
A Solution Framework for Private Media in Privacy-Enhanced RTP Conferencing (PERC)
This document describes a solution framework for ensuring that media confidentiality and integrity are maintained end to end within the context of a switched conferencing environment where Media Distributors are not trusted with the end-to-end media encryption keys. The solution builds upon existing security mechanisms defined for the Real-time Transport Protocol (RTP).
Encrypted Key Transport for DTLS and Secure RTP
Encrypted Key Transport (EKT) is an extension to DTLS (Datagram Transport Layer Security) and the Secure Real-time Transport Protocol (SRTP) that provides for the secure transport of SRTP master keys, rollover counters, and other information within SRTP. This facility enables SRTP for decentralized conferences by distributing a common key to all of the conference endpoints.
Evaluation Test Cases for Interactive Real-Time Media over Wireless Networks
The Real-time Transport Protocol (RTP) is a common transport choice for interactive multimedia communication applications. The performance of these applications typically depends on a well-functioning congestion control algorithm. To ensure a seamless and robust user experience, a well-designed RTP-based congestion control algorithm should work well across all access network types. This document d…
Evaluating Congestion Control for Interactive Real-Time Media
The Real-Time Transport Protocol (RTP) is used to transmit media in telephony and video conferencing applications. This document describes the guidelines to evaluate new congestion control algorithms for interactive point-to-point real-time media.
Test Cases for Evaluating Congestion Control for Interactive Real-Time Media
The Real-time Transport Protocol (RTP) is used to transmit media in multimedia telephony applications. These applications are typically required to implement congestion control. This document describes the test cases to be used in the performance evaluation of such congestion control algorithms in a controlled environment.
SDP: Session Description Protocol
This memo defines the Session Description Protocol (SDP). SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. This document obsoletes RFC 4566.
T.140 Real-Time Text Conversation over WebRTC Data Channels
This document specifies how a Web Real-Time Communication (WebRTC) data channel can be used as a transport mechanism for real-time text using the ITU-T Protocol for multimedia application text conversation (Recommendation ITU-T T.140) and how the Session Description Protocol (SDP) offer/answer mechanism can be used to negotiate such a data channel, referred to as a T.140 data channel. This documen…
Negotiation Data Channels Using the Session Description Protocol (SDP)
Data channel setup can be done using either the in-band Data Channel Establishment Protocol (DCEP) or some out-of-band non-DCEP protocol. This document specifies how the SDP (Session Description Protocol) offer/answer exchange can be used to achieve an out-of-band non-DCEP negotiation for establishing a data channel.
Interactive Connectivity Establishment Patiently Awaiting Connectivity (ICE PAC)
During the process of establishing peer-to-peer connectivity, Interactive Connectivity Establishment (ICE) agents can encounter situations where they have no candidate pairs to check, and, as a result, conclude that ICE processing has failed. However, because additional candidate pairs can be discovered during ICE processing, declaring failure at this point may be premature. This document discusse…
Best Practices for Securing RTP Media Signaled with SIP
Although the Session Initiation Protocol (SIP) includes a suite of security services that has been expanded by numerous specifications over the years, there is no single place that explains how to use SIP to establish confidential media sessions. Additionally, existing mechanisms have some feature gaps that need to be identified and resolved in order for them to address the pervasive monitoring th…
Sending Multiple RTP Streams in a Single RTP Session: Grouping RTP Control Protocol (RTCP) Reception Statistics and Other Feedback
RTP allows multiple RTP streams to be sent in a single session but requires each Synchronization Source (SSRC) to send RTP Control Protocol (RTCP) reception quality reports for every other SSRC visible in the session. This causes the number of RTCP reception reports to grow with the number of SSRCs, rather than the number of endpoints. In many cases, most of these RTCP reception reports are unnece…
Sending Multiple Types of Media in a Single RTP Session
This document specifies how an RTP session can contain RTP streams with media from multiple media types such as audio, video, and text. This has been restricted by the RTP specifications (RFCs 3550 and 3551), and thus this document updates RFCs 3550 and 3551 to enable this behaviour for applications that satisfy the applicability for using multiple media types in a single RTP session.
A Framework for Session Description Protocol (SDP) Attributes When Multiplexing
The purpose of this specification is to provide a framework for analyzing the multiplexing characteristics of Session Description Protocol (SDP) attributes when SDP is used to negotiate the usage of a single 5-tuple for sending and receiving media associated with multiple media descriptions. This specification also categorizes the existing SDP attributes based on the framework described herein.
Indicating Exclusive Support of RTP and RTP Control Protocol (RTCP) Multiplexing Using the Session Description Protocol (SDP)
This document defines a new Session Description Protocol (SDP) media-level attribute, 'rtcp-mux-only', that can be used by an endpoint to indicate exclusive support of RTP and RTP Control Protocol (RTCP) multiplexing. The document also updates RFC 5761 by clarifying that an offerer can use a mechanism to indicate that it is not able to send and receive RTCP on separate ports.