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":
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