RFC Abstracts

RFC8878 - 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.
RFC8877 - 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 designers. It is expected that a new network protocol that requires a packet timestamp will, in most cases, use one of the recommended timestamp formats. If none of the recommended formats fits the protocol requirements, the new protocol specification should specify the format of the packet timestamp according to the guidelines in this document.
RFC8876 - 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 interactive media such as voice or video or text. This document describes use of a SIP MESSAGE transaction that includes a container for the data based on the Common Alerting Protocol (CAP). That type of emergency request does not establish a session, distinguishing it from SIP INVITE, which does. Any device that needs to initiate a request for emergency services without an interactive media channel would use the mechanisms in this document.
RFC8875 - 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.
RFC8874 - Working Group GitHub Usage Guidance
This document provides a set of guidelines for working groups that choose to use GitHub for their work.
RFC8873 - 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 endpoints; and a gateway configuration, which connects an MSRP data channel endpoint with an MSRP endpoint that uses either TCP or TLS. This document updates RFC 4975.
RFC8872 - 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 design trade-offs, and provides guidelines on how to use the multiplexing features of RTP to support multiple media streams.
RFC8871 - 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).
RFC8870 - 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.
RFC8869 - 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 describes test cases for evaluating performances of candidate congestion control algorithms over cellular and Wi-Fi networks.
RFC8868 - 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.
RFC8867 - 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.
RFC8866 - 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.
RFC8865 - 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 document updates RFC 8373 to specify its use with WebRTC data channels.
RFC8864 - 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.
RFC8863 - 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 discusses when these situations can occur.
RFC8862 - 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 threat model. This specification describes best practices for negotiating confidential media with SIP, including a comprehensive protection solution that binds the media layer to SIP layer identities.
RFC8861 - 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 unnecessary, since all SSRCs of an endpoint are normally co-located and see the same reception quality. This memo defines a Reporting Group extension to RTCP to reduce the reporting overhead in such scenarios.
RFC8860 - 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.
RFC8859 - 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.
RFC8858 - 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.
RFC8857 - The WebSocket Protocol as a Transport for the Binary Floor Control Protocol (BFCP)
The WebSocket protocol enables two-way real-time communication between clients and servers. This document specifies the use of Binary Floor Control Protocol (BFCP) as a new WebSocket subprotocol enabling a reliable transport mechanism between BFCP entities in new scenarios.
RFC8856 - Session Description Protocol (SDP) Format for Binary Floor Control Protocol (BFCP) Streams
This document defines the Session Description Protocol (SDP) offer/answer procedures for negotiating and establishing Binary Floor Control Protocol (BFCP) streams.
RFC8855 - The Binary Floor Control Protocol (BFCP)
Floor control is a means to manage joint or exclusive access to shared resources in a (multiparty) conferencing environment. Thereby, floor control complements other functions -- such as conference and media session setup, conference policy manipulation, and media control -- that are realized by other protocols.
RFC8854 - WebRTC Forward Error Correction Requirements
This document provides information and requirements for the use of Forward Error Correction (FEC) by WebRTC implementations.
RFC8853 - Using Simulcast in Session Description Protocol (SDP) and RTP Sessions
In some application scenarios, it may be desirable to send multiple differently encoded versions of the same media source in different RTP streams. This is called simulcast. This document describes how to accomplish simulcast in RTP and how to signal it in the Session Description Protocol (SDP). The described solution uses an RTP/RTCP identification method to identify RTP streams belonging to the same media source and makes an extension to SDP to indicate that those RTP streams are different simulcast formats of that media source. The SDP extension consists of a new media-level SDP attribute that expresses capability to send and/or receive simulcast RTP streams.
RFC8852 - RTP Stream Identifier Source Description (SDES)
This document defines and registers two new Real-time Transport Control Protocol (RTCP) Stream Identifier Source Description (SDES) items. One, named RtpStreamId, is used for unique identification of RTP streams. The other, RepairedRtpStreamId, can be used to identify which stream is to be repaired using a redundancy RTP stream.
RFC8851 - RTP Payload Format Restrictions
In this specification, we define a framework for specifying restrictions on RTP streams in the Session Description Protocol (SDP). This framework defines a new "rid" ("restriction identifier") SDP attribute to unambiguously identify the RTP streams within an RTP session and restrict the streams' payload format parameters in a codec-agnostic way beyond what is provided with the regular payload types.
RFC8850 - Controlling Multiple Streams for Telepresence (CLUE) Protocol Data Channel
This document defines how to use the WebRTC data channel mechanism to realize a data channel, referred to as a Controlling Multiple Streams for Telepresence (CLUE) data channel, for transporting CLUE protocol messages between two CLUE entities.
RFC8849 - Mapping RTP Streams to Controlling Multiple Streams for Telepresence (CLUE) Media Captures
This document describes how the Real-time Transport Protocol (RTP) is used in the context of the Controlling Multiple Streams for Telepresence (CLUE) protocol. It also describes the mechanisms and recommended practice for mapping RTP media streams, as defined in the Session Description Protocol (SDP), to CLUE Media Captures and defines a new RTP header extension (CaptureID).
RFC8848 - Session Signaling for Controlling Multiple Streams for Telepresence (CLUE)
This document is about Controlling Multiple Streams for Telepresence (CLUE) signaling. It specifies how the CLUE protocol and the CLUE data channel are used in conjunction with each other and with existing signaling mechanisms, such as SIP and the Session Description Protocol (SDP), to produce a telepresence call.
RFC8847 - Protocol for Controlling Multiple Streams for Telepresence (CLUE)
The Controlling Multiple Streams for Telepresence (CLUE) protocol is an application protocol conceived for the description and negotiation of a telepresence session. The design of the CLUE protocol takes into account the requirements and the framework defined within the IETF CLUE Working Group. A companion document, RFC 8848, delves into CLUE signaling details as well as the SIP / Session Description Protocol (SDP) session establishment phase. CLUE messages flow over the CLUE data channel, based on reliable and ordered SCTP-over-DTLS transport. ("SCTP" stands for "Stream Control Transmission Protocol".) Message details, together with the behavior of CLUE Participants acting as Media Providers and/or Media Consumers, are herein discussed.
RFC8846 - An XML Schema for the Controlling Multiple Streams for Telepresence (CLUE) Data Model
This document provides an XML schema file for the definition of CLUE data model types. The term "CLUE" stands for "Controlling Multiple Streams for Telepresence" and is the name of the IETF working group in which this document, as well as other companion documents, has been developed. The document defines a coherent structure for information associated with the description of a telepresence scenario.
RFC8845 - Framework for Telepresence Multi-Streams
This document defines a framework for a protocol to enable devices in a telepresence conference to interoperate. The protocol enables communication of information about multiple media streams so a sending system and receiving system can make reasonable decisions about transmitting, selecting, and rendering the media streams. This protocol is used in addition to SIP signaling and Session Description Protocol (SDP) negotiation for setting up a telepresence session.
RFC8844 - Unknown Key-Share Attacks on Uses of TLS with the Session Description Protocol (SDP)
This document describes unknown key-share attacks on the use of Datagram Transport Layer Security for the Secure Real-Time Transport Protocol (DTLS-SRTP). Similar attacks are described on the use of DTLS-SRTP with the identity bindings used in Web Real-Time Communications (WebRTC) and SIP identity. These attacks are difficult to mount, but they cause a victim to be misled about the identity of a communicating peer. This document defines mitigation techniques that implementations of RFC 8122 are encouraged to deploy.
RFC8843 - Negotiating Media Multiplexing Using the Session Description Protocol (SDP)
This specification defines a new Session Description Protocol (SDP) Grouping Framework extension called 'BUNDLE'. The extension can be used with the SDP offer/answer mechanism to negotiate the usage of a single transport (5-tuple) for sending and receiving media described by multiple SDP media descriptions ("m=" sections). Such transport is referred to as a BUNDLE transport, and the media is referred to as bundled media. The "m=" sections that use the BUNDLE transport form a BUNDLE group.
RFC8842 - Session Description Protocol (SDP) Offer/Answer Considerations for Datagram Transport Layer Security (DTLS) and Transport Layer Security (TLS)
This document defines the Session Description Protocol (SDP) offer/answer procedures for negotiating and establishing a Datagram Transport Layer Security (DTLS) association. The document also defines the criteria for when a new DTLS association must be established. The document updates RFCs 5763 and 7345 by replacing common SDP offer/answer procedures with a reference to this specification.
RFC8841 - Session Description Protocol (SDP) Offer/Answer Procedures for Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer Security (DTLS) Transport
The Stream Control Transmission Protocol (SCTP) is a transport protocol used to establish associations between two endpoints. RFC 8261 specifies how SCTP can be used on top of the Datagram Transport Layer Security (DTLS) protocol, which is referred to as SCTP-over-DTLS.
RFC8840 - A Session Initiation Protocol (SIP) Usage for Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (Trickle ICE)
The Interactive Connectivity Establishment (ICE) protocol describes a Network Address Translator (NAT) traversal mechanism for UDP-based multimedia sessions established with the Offer/Answer model. The ICE extension for Incremental Provisioning of Candidates (Trickle ICE) defines a mechanism that allows ICE Agents to shorten session establishment delays by making the candidate gathering and connectivity checking phases of ICE non-blocking and by executing them in parallel.
RFC8839 - Session Description Protocol (SDP) Offer/Answer Procedures for Interactive Connectivity Establishment (ICE)
This document describes Session Description Protocol (SDP) Offer/Answer procedures for carrying out Interactive Connectivity Establishment (ICE) between the agents.
RFC8838 - Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol
This document describes "Trickle ICE", an extension to the Interactive Connectivity Establishment (ICE) protocol that enables ICE agents to begin connectivity checks while they are still gathering candidates, by incrementally exchanging candidates over time instead of all at once. This method can considerably accelerate the process of establishing a communication session.
RFC8837 - Differentiated Services Code Point (DSCP) Packet Markings for WebRTC QoS
Networks can provide different forwarding treatments for individual packets based on Differentiated Services Code Point (DSCP) values on a per-hop basis. This document provides the recommended DSCP values for web browsers to use for various classes of Web Real-Time Communication (WebRTC) traffic.
RFC8836 - Congestion Control Requirements for Interactive Real-Time Media
Congestion control is needed for all data transported across the Internet, in order to promote fair usage and prevent congestion collapse. The requirements for interactive, point-to-point real-time multimedia, which needs low-delay, semi-reliable data delivery, are different from the requirements for bulk transfer like FTP or bursty transfers like web pages. Due to an increasing amount of RTP-based real-time media traffic on the Internet (e.g., with the introduction of the Web Real-Time Communication (WebRTC)), it is especially important to ensure that this kind of traffic is congestion controlled.
RFC8835 - Transports for WebRTC
This document describes the data transport protocols used by Web Real-Time Communication (WebRTC), including the protocols used for interaction with intermediate boxes such as firewalls, relays, and NAT boxes.
RFC8834 - Media Transport and Use of RTP in WebRTC
The framework for Web Real-Time Communication (WebRTC) provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context and gives requirements for which RTP features, profiles, and extensions need to be supported.
RFC8833 - Application-Layer Protocol Negotiation (ALPN) for WebRTC
This document specifies two Application-Layer Protocol Negotiation (ALPN) labels for use with Web Real-Time Communication (WebRTC). The "webrtc" label identifies regular WebRTC: a DTLS session that is used to establish keys for the Secure Real-time Transport Protocol (SRTP) or to establish data channels using the Stream Control Transmission Protocol (SCTP) over DTLS. The "c-webrtc" label describes the same protocol, but the peers also agree to maintain the confidentiality of the media by not sharing it with other applications.
RFC8832 - WebRTC Data Channel Establishment Protocol
The WebRTC framework specifies protocol support for direct interactive rich communication using audio, video, and data between two peers' web browsers. This document specifies a simple protocol for establishing symmetric data channels between the peers. It uses a two-way handshake and allows sending of user data without waiting for the handshake to complete.
RFC8831 - WebRTC Data Channels
The WebRTC framework specifies protocol support for direct, interactive, rich communication using audio, video, and data between two peers' web browsers. This document specifies the non-media data transport aspects of the WebRTC framework. It provides an architectural overview of how the Stream Control Transmission Protocol (SCTP) is used in the WebRTC context as a generic transport service that allows web browsers to exchange generic data from peer to peer.
RFC8830 - WebRTC MediaStream Identification in the Session Description Protocol
This document specifies a Session Description Protocol (SDP) grouping mechanism for RTP media streams that can be used to specify relations between media streams.
RFC8829 - JavaScript Session Establishment Protocol (JSEP)
This document describes the mechanisms for allowing a JavaScript application to control the signaling plane of a multimedia session via the interface specified in the W3C RTCPeerConnection API and discusses how this relates to existing signaling protocols.