WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-04

Document Type Active Internet-Draft (rtcweb WG)
Last updated 2017-07-03
Replaces draft-shieh-rtcweb-ip-handling
Stream IETF
Intended RFC status Proposed Standard
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Document shepherd Sean Turner
IESG IESG state I-D Exists
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Send notices to Sean Turner <sean@sn3rd.com>
Network Working Group                                          J. Uberti
Internet-Draft                                                  G. Shieh
Intended status: Standards Track                                  Google
Expires: January 4, 2018                                    July 3, 2017

                WebRTC IP Address Handling Requirements
                    draft-ietf-rtcweb-ip-handling-04

Abstract

   This document provides information and requirements for how IP
   addresses should be handled by WebRTC implementations.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on January 4, 2018.

Copyright Notice

   Copyright (c) 2017 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
   (http://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.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   2
   3.  Problem Statement . . . . . . . . . . . . . . . . . . . . . .   2
   4.  Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  Detailed Design . . . . . . . . . . . . . . . . . . . . . . .   4
   6.  Application Guidance  . . . . . . . . . . . . . . . . . . . .   6
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .   6
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   6
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   7
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .   7
     10.1.  Normative References . . . . . . . . . . . . . . . . . .   7
     10.2.  Informative References . . . . . . . . . . . . . . . . .   7
   Appendix A.  Change log . . . . . . . . . . . . . . . . . . . . .   8
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   9

1.  Introduction

   One of WebRTC's key features is its support of peer-to-peer
   connections.  However, when establishing such a connection, which
   involves connectivity tests using various IP addresses, WebRTC may
   allow a web application to learn additional information about the
   user compared to an application that only uses the Hypertext Transfer
   Protocol (HTTP) [RFC7230].  This may be problematic in certain cases.
   This document summarizes the concerns, and makes recommendations on
   how WebRTC implementations should best handle the tradeoff between
   privacy and media performance.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

3.  Problem Statement

   In order to establish a peer-to-peer connection, WebRTC
   implementations use Interactive Connectivity Establishment (ICE)
   [RFC5245], which gathers and exchanges all the IP addresses it can
   discover, using techniques like Session Traversal Utilities for NAT
   (STUN) [RFC5389] and Traversal Using Relays around NAT (TURN)
   [RFC5766], in order to check the connectivity of each local-address-
   remote-address pair and select the best one.  The addresses that are
   gathered usually consist of an endpoint's private physical/virtual
   addresses and its public Internet addresses.

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