Transports for WebRTC
RFC 8835

Document Type RFC - Proposed Standard (January 2021; No errata)
Author Harald Alvestrand 
Last updated 2021-01-18
Stream IETF
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Document shepherd Cullen Jennings
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IESG IESG state RFC 8835 (Proposed Standard)
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Responsible AD Alissa Cooper
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IANA IANA review state Version Changed - Review Needed
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Internet Engineering Task Force (IETF)                     H. Alvestrand
Request for Comments: 8835                                        Google
Category: Standards Track                                   January 2021
ISSN: 2070-1721

                         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.

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

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
   ( 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
   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",
   "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

   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
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