Transports for WebRTC
RFC 8835
Document | Type | RFC - Proposed Standard (January 2021; No errata) | |
---|---|---|---|
Author | Harald Alvestrand | ||
Last updated | 2021-01-18 | ||
Stream | IETF | ||
Formats | plain text html xml pdf htmlized bibtex | ||
Reviews | |||
Stream | WG state | Submitted to IESG for Publication | |
Document shepherd | Cullen Jennings | ||
Shepherd write-up | Show (last changed 2016-07-07) | ||
IESG | IESG state | RFC 8835 (Proposed Standard) | |
Action Holders |
(None)
|
||
Consensus Boilerplate | Yes | ||
Telechat date | |||
Responsible AD | Alissa Cooper | ||
Send notices to | (None) | ||
IANA | IANA review state | Version Changed - Review Needed | |
IANA action state | No IANA Actions |
Internet Engineering Task Force (IETF) H. Alvestrand Request for Comments: 8835 Google Category: Standards Track January 2021 ISSN: 2070-1721 Transports for WebRTC Abstract 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. Status of This Memo This is an Internet Standards Track document. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at https://www.rfc-editor.org/info/rfc8835. Copyright Notice Copyright (c) 2021 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents 1. Introduction 2. Requirements Language 3. Transport and Middlebox Specification 3.1. System-Provided Interfaces 3.2. Ability to Use IPv4 and IPv6 3.3. Usage of Temporary IPv6 Addresses 3.4. Middlebox-Related Functions 3.5. Transport Protocols Implemented 4. Media Prioritization 4.1. Local Prioritization 4.2. Usage of Quality of Service -- DSCP and Multiplexing 5. IANA Considerations 6. Security Considerations 7. References 7.1. Normative References 7.2. Informative References Acknowledgements Author's Address 1. Introduction WebRTC is a protocol suite aimed at real-time multimedia exchange between browsers, and between browsers and other entities. WebRTC is described in the WebRTC overview document [RFC8825], which also defines terminology used in this document, including the terms "WebRTC endpoint" and "WebRTC browser". Terminology for RTP sources is taken from [RFC7656]. This document focuses on the data transport protocols that are used by conforming implementations, including the protocols used for interaction with intermediate boxes such as firewalls, relays, and NAT boxes. This protocol suite is intended to satisfy the security considerations described in the WebRTC security documents, [RFC8826] and [RFC8827]. This document describes requirements that apply to all WebRTC endpoints. When there are requirements that apply only to WebRTC browsers, this is called out explicitly. 2. Requirements Language The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here. 3. Transport and Middlebox Specification 3.1. System-Provided Interfaces The protocol specifications used here assume that the following protocols are available to the implementations of the WebRTC protocols: UDP [RFC0768]: This is the protocol assumed by most protocol elements described. TCP [RFC0793]: This is used for HTTP/WebSockets, as well as TURN/TLS and ICE-TCP. For both protocols, IPv4 and IPv6 support is assumed. For UDP, this specification assumes the ability to set the Differentiated Services Code Point (DSCP) of the sockets opened on a per-packet basis, in order to achieve the prioritizations described in [RFC8837] (see Section 4.2 of this document) when multiple media types are multiplexed. It does not assume that the DSCPs will be honored and does assume that they may be zeroed or changed, since this is a local configuration issue. Platforms that do not give access to these interfaces will not be able to support a conforming WebRTC endpoint. This specification does not assume that the implementation will have access to ICMP or raw IP. The following protocols may be used, but they can be implemented by a WebRTC endpoint and are therefore not defined as "system-provided interfaces":Show full document text