WebRTC

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.

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.

This specification defines a new RTP Control Protocol (RTCP) Source Description (SDES) item and a new RTP header extension.

This specification updates RFCs 3264, 5888, and 7941.

This specification obsoletes RFC 8843.

This document provides information and requirements for the use of Forward Error Correction (FEC) by WebRTC implementations.

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.

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).

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.

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".

This document provides information and requirements for how IP addresses should be handled by Web Real-Time Communication (WebRTC) implementations.

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.

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.

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.

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.

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.

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.

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.

This document describes a set of requirements that can be used to evaluate other congestion control mechanisms in order to figure out their fitness for this purpose, and in particular to provide a set of possible requirements for a real-time media congestion avoidance technique.

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.

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.

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.

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.

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.

This document describes monitoring features related to media streams in Web real-time communication (WebRTC). It provides a list of RTP Control Protocol (RTCP) Sender Report (SR), Receiver Report (RR), and Extended Report (XR) metrics, which may need to be supported by RTP implementations in some diverse environments. It lists a set of identifiers for the WebRTC's statistics API. These identifiers are a set of RTCP SR, RR, and XR metrics related to the transport of multimedia flows.

The baseline security mechanisms in the Session Initiation Protocol (SIP) are inadequate for cryptographically assuring the identity of the end users that originate SIP requests, especially in an interdomain context. This document defines a mechanism for securely identifying originators of SIP requests. It does so by defining a SIP header field for conveying a signature used for validating the identity and for conveying a reference to the credentials of the signer.

This document obsoletes RFC 4474.

This document outlines the audio codec and processing requirements for WebRTC endpoints.

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.

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.

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.

This document proposes the use of OAuth 2.0 to obtain and validate ephemeral tokens that can be used for Session Traversal Utilities for NAT (STUN) authentication. The usage of ephemeral tokens ensures that access to a STUN server can be controlled even if the tokens are compromised.

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.

As the Internet and the telephone network have become increasingly interconnected and interdependent, attackers can impersonate or obscure calling party numbers when orchestrating bulk commercial calling schemes, hacking voicemail boxes, or even circumventing multi-factor authentication systems trusted by banks. This document analyzes threats in the resulting system, enumerating actors, reviewing the capabilities available to and used by attackers, and describing scenarios in which attacks are launched.

Over the past decade, Voice over IP (VoIP) systems based on SIP have replaced many traditional telephony deployments. Interworking VoIP systems with the traditional telephone network has reduced the overall level of calling party number and Caller ID assurances by granting attackers new and inexpensive tools to impersonate or obscure calling party numbers when orchestrating bulk commercial calling schemes, hacking voicemail boxes, or even circumventing multi-factor authentication systems trusted by banks. Despite previous attempts to provide a secure assurance of the origin of SIP communications, we still lack effective standards for identifying the calling party in a VoIP session. This document examines the reasons why providing identity for telephone numbers on the Internet has proven so difficult and shows how changes in the last decade may provide us with new strategies for attaching a secure identity to SIP sessions. It also gives high-level requirements for a solution in this space.

This document provides a summary of the IAB/IRTF Workshop on 'Congestion Control for Interactive Real-Time Communication', which took place in Vancouver, Canada, on July 28, 2012. The main goal of the workshop was to foster a discussion on congestion control mechanisms for interactive real-time communication. This report summarizes the discussions and lists recommendations to the Internet Engineering Task Force (IETF) community.

The views and positions in this report are those of the workshop participants and do not necessarily reflect the views and positions of the authors, the Internet Architecture Board (IAB), or the Internet Research Task Force (IRTF).