Опубликованные RFC, связи обновления и замены, официальные статусы и русские переводы.
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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.
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. This document obsoletes RFC 4583.
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. This document specifies the Binary Floor Control Protocol (BFCP). BFCP is used between…
WebRTC Forward Error Correction Requirements
This document provides information and requirements for the use of Forward Error Correction (FEC) by WebRTC implementations.
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 …
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.
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 type…
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.
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).
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.
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 Descript…
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 scenar…
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 Descriptio…
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 c…
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 refer…
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. This d…
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. This specification defines the following new Session Description Protocol (SDP) protocol identifiers (proto values): "UDP/DTLS/SC…
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 connec…
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. This document obsoletes RFCs 5245 and 6336.
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.
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.
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-base…
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.
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 feat…
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…
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.
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 a…
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. This mechanism is used to signal the association between the SDP concept of "media description" and the Web Real-Time Communication (WebRTC) concept of MediaStream/MediaStreamTrack using SDP signaling.
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.
WebRTC IP Address Handling Requirements
This document provides information and requirements for how IP addresses should be handled by Web Real-Time Communication (WebRTC) implementations.