Rtcweb Workgroup RFCs

Browse Rtcweb Workgroup RFCs by Number

RFC7478 - Web Real-Time Communication Use Cases and Requirements: This document describes web-based real-time communication use cases. Requirements on the browser functionality are derived from the use cases.; This document was developed in an initial phase of the work with rather minor updates at later stages. It has not really served as a tool in deciding features or scope for the WG's efforts so far. It is being published to record the early conclusions of the WG. It will not be used as a set of rigid guidelines that specifications and implementations will be held to in the future.
RFC7675 - Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness: To prevent WebRTC applications, such as browsers, from launching attacks by sending traffic to unwilling victims, periodic consent to send needs to be obtained from remote endpoints.; This document describes a consent mechanism using a new Session Traversal Utilities for NAT (STUN) usage.
RFC7742 - WebRTC Video Processing and Codec Requirements: This specification provides the requirements and considerations for WebRTC applications to send and receive video across a network. It specifies the video processing that is required as well as video codecs and their parameters.
RFC7874 - WebRTC Audio Codec and Processing Requirements: This document outlines the audio codec and processing requirements for WebRTC endpoints.
RFC7875 - Additional WebRTC Audio Codecs for Interoperability: To ensure a baseline of interoperability between WebRTC endpoints, a minimum set of required codecs is specified. However, to maximize the possibility of establishing the session without the need for audio transcoding, it is also recommended to include in the offer other suitable audio codecs that are available to the browser.; This document provides some guidelines on the suitable codecs to be considered for WebRTC endpoints to address the use cases most relevant to interoperability.
RFC8825 - Overview: Real-Time Protocols for Browser-Based Applications: This document gives an overview and context of a protocol suite intended for use with real-time applications that can be deployed in browsers -- "real-time communication on the Web".; It intends to serve as a starting and coordination point to make sure that (1) all the parts that are needed to achieve this goal are findable and (2) the parts that belong in the Internet protocol suite are fully specified and on the right publication track.; This document is an applicability statement -- it does not itself specify any protocol, but it specifies which other specifications implementations are supposed to follow to be compliant with Web Real-Time Communication (WebRTC).
RFC8826 - Security Considerations for WebRTC: WebRTC is a protocol suite for use with real-time applications that can be deployed in browsers -- "real-time communication on the Web". This document defines the WebRTC threat model and analyzes the security threats of WebRTC in that model.
RFC8827 - WebRTC Security Architecture: This document defines the security architecture for WebRTC, a protocol suite intended for use with real-time applications that can be deployed in browsers -- "real-time communication on the Web".
RFC8828 - 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.
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.
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.
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.
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.
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.
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.
RFC8854 - WebRTC Forward Error Correction Requirements: This document provides information and requirements for the use of Forward Error Correction (FEC) by WebRTC implementations.
RFC9429 - 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.; This specification obsoletes RFC 8829.