--- 1/draft-ietf-rtcweb-ip-handling-01.txt 2016-10-31 17:16:18.995731709 -0700 +++ 2/draft-ietf-rtcweb-ip-handling-02.txt 2016-10-31 17:16:19.019732305 -0700 @@ -1,18 +1,18 @@ Network Working Group J. Uberti Internet-Draft G. Shieh Intended status: Standards Track Google -Expires: September 21, 2016 March 20, 2016 +Expires: May 4, 2017 October 31, 2016 - WebRTC IP Address Handling Recommendations - draft-ietf-rtcweb-ip-handling-01 + WebRTC IP Address Handling Requirements + draft-ietf-rtcweb-ip-handling-02 Abstract This document provides best practices for how IP addresses should be handled by WebRTC applications. Status of This Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. @@ -20,21 +20,21 @@ 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 September 21, 2016. + This Internet-Draft will expire on May 4, 2017. Copyright Notice Copyright (c) 2016 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 @@ -47,22 +47,22 @@ Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2 3. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3 4. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4 5. Application Guidance . . . . . . . . . . . . . . . . . . . . 6 6. Security Considerations . . . . . . . . . . . . . . . . . . . 6 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 6 - 9. Informative References . . . . . . . . . . . . . . . . . . . 7 - Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 8 + 9. Informative References . . . . . . . . . . . . . . . . . . . 6 + Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 7 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 8 1. Introduction As a technology that supports peer-to-peer connections, WebRTC may send data over different network paths than the path used for HTTP traffic. This may allow a web application to learn additional information about the user, which may be problematic in certain cases. This document summarizes the concerns, and makes recommendations on how best to handle the tradeoff between privacy @@ -159,33 +159,31 @@ as candidates, even when binding to the wildcard addresses as mentioned above. The appropriate addresses here can be discovered by the common trick of binding sockets to the wildcard addresses, connect()ing those sockets to some well-known public IP address (one particular example being "8.8.8.8"), and then reading the bound local addresses via getsockname(). This approach requires no data exchange; it simply provides a mechanism for applications to retrieve the desired information from the kernel routing table. - 3. When used with audio and video devices, WebRTC requires explicit - user permission to access those devices. We propose that this - permission grant be expanded to include consent to allow WebRTC - to access all IP addresses associated with the user agent, for - the purpose of finding the absolute best route for media traffic. - Combining these permission grants, rather than having the user - grant permission individually, is a considered balance; this - balance takes into account that the user has placed enough trust - into the application to allow it to access their devices, that - when doing so the user typically wants to engage in a - conversational session, which benefits most from an optimal - network path, and lastly, the fact that the underlying issue is - complex, and difficult to explain meaningfully to the user. + 3. Gathering all possible candidates SHOULD only be performed when + some form of user consent has been provided; this thwarts the + typical drive-by enumeration attacks. The details of this + consent are left to the implementation; one potential mechanism + is to key this off getUserMedia consent. The getUserMedia + suggestion takes into account that the user has provided some + consent to the application already; that when doing so the user + typically wants to engage in a conversational session, which + benefits most from an optimal network path, and lastly, the fact + that the underlying issue is complex and difficult to explain, + making explicit consent for enumeration troublesome. 4. Determining whether a web proxy is in use is a complex process, as the answer can depend on the exact site or address being contacted. Furthermore, web proxies that support UDP are not widely deployed today. As a result, when WebRTC is made to go through a proxy, it typically must use TCP, either ICE-TCP [RFC6544] or TURN-over-TCP [RFC5766]. Naturally, this has attendant costs on media quality and also proxy performance. 5. RETURN [I-D.ietf-rtcweb-return] is a new proposal for explicit @@ -225,22 +223,22 @@ Mode 4: Force proxy: This forces all WebRTC media traffic through a proxy, if one is configured. If the proxy does not support UDP (as is the case for all HTTP and most SOCKS [RFC1928] proxies), or the WebRTC implementation does not support UDP proxying, the use of UDP will be disabled, and TCP will be used to send and receive media through the proxy. Use of TCP will result in reduced quality, in addition to any performance considerations associated with sending all WebRTC media through the proxy server. - We recommend Mode 1 as the default behavior only if cam/mic - permission has been granted, or Mode 2 if this is not the case. + We recommend Mode 1 as the default behavior only if the user has + provided some form of consent, as discussed above, or Mode 2 if not. Users who prefer Mode 3 or 4 should be able to select a preference or install an extension to force their browser to operate in the specified mode. Note that when a RETURN proxy is configured for the interface associated with the default route, Mode 2 and 3 will cause any external media traffic to go through the RETURN proxy. This provides a way to ensure the proxy is used for external traffic, but without the performance issues of forcing all media through said proxy. @@ -323,20 +321,26 @@ "TCP Candidates with Interactive Connectivity Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544, March 2012, . [RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013, . Appendix A. Change log + Changes in draft -02: + + o Recommendations -> Requirements + + o Updated text regarding consent. + Changes in draft -01: o Incorporated feedback from Adam Roach; changes to discussion of cam/mic permission, as well as use of proxies, and various editorial changes. o Added several more references. Changes in draft -00: