--- 1/draft-ietf-rtcweb-ip-handling-07.txt 2018-06-12 17:13:08.133500418 -0700 +++ 2/draft-ietf-rtcweb-ip-handling-08.txt 2018-06-12 17:13:08.157500997 -0700 @@ -1,19 +1,19 @@ Network Working Group J. Uberti Internet-Draft Google Intended status: Standards Track G. Shieh -Expires: October 20, 2018 Facebook - April 18, 2018 +Expires: December 5, 2018 Facebook + June 3, 2018 WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handling-07 + draft-ietf-rtcweb-ip-handling-08 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. @@ -21,21 +21,21 @@ 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 https://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 October 20, 2018. + This Internet-Draft will expire on December 5, 2018. Copyright Notice Copyright (c) 2018 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 @@ -49,30 +49,30 @@ 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2 3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2 4. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 5. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4 5.1. Principles . . . . . . . . . . . . . . . . . . . . . . . 4 5.2. Modes and Recommendations . . . . . . . . . . . . . . . . 5 6. Implementation Guidance . . . . . . . . . . . . . . . . . . . 6 6.1. Ensuring Normal Routing . . . . . . . . . . . . . . . . . 6 - 6.2. Determining Host Candidates . . . . . . . . . . . . . . . 6 + 6.2. Determining Host Candidates . . . . . . . . . . . . . . . 7 7. Application Guidance . . . . . . . . . . . . . . . . . . . . 7 8. Security Considerations . . . . . . . . . . . . . . . . . . . 7 - 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 - 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7 + 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 + 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 11.1. Normative References . . . . . . . . . . . . . . . . . . 8 11.2. Informative References . . . . . . . . . . . . . . . . . 8 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 9 - Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 10 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 11 1. Introduction One of WebRTC's key features is its support of peer-to-peer connections. However, when establishing such a connection, which involves connection attempts from 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 @@ -113,23 +113,27 @@ interfaces (a "split-tunnel" VPN), WebRTC will discover not only the public address for the VPN, but also the ISP public address over which the VPN is running. 2. If the client is behind a Network Address Translator (NAT), the client's private IP addresses, often [RFC1918] addresses, can be learned. 3. If the client is behind a proxy (a client-configured "classical application proxy", as defined in [RFC1919], Section 3), but - direct access to the Internet is also supported, WebRTC's STUN - checks will bypass the proxy and reveal the public IP address of - the client. + direct access to the Internet is permitted, WebRTC's STUN checks + will bypass the proxy and reveal the public IP address of the + client. This concern also applies to the "enterprise TURN + server" scenario described in [RFC7478], Section 2.3.5.1, if, as + above, direct Internet access is permitted. However, when the + term "proxy" is used in this document, it is always in reference + to an [RFC1919] proxy server. Of these three concerns, #1 is the most significant, because for some users, the purpose of using a VPN is for anonymity. However, different VPN users will have different needs, and some VPN users (e.g., corporate VPN users) may in fact prefer WebRTC to send media traffic directly, i.e., not through the VPN. #2 is considered to be a less significant concern, given that the local address values often contain minimal information (e.g., 192.168.0.2), or have built-in privacy protection (e.g., the @@ -166,85 +170,88 @@ information in a way that might violate user expectations. 5. Detailed Design 5.1. Principles The key principles for our framework are stated below: 1. By default, WebRTC traffic should follow typical IP routing, i.e., WebRTC should use the same interface used for HTTP traffic, - and only the system's 'typical' public addresses should be - visible to the application. However, in the interest of optimal - media quality, it should be possible to enable WebRTC to make use - of all network interfaces to determine the ideal route. + and only the system's 'typical' public addresses (or those of an + enterprise TURN server, if present) should be visible to the + application. However, in the interest of optimal media quality, + it should be possible to enable WebRTC to make use of all network + interfaces to determine the ideal route. 2. By default, WebRTC should be able to negotiate direct peer-to- peer connections between endpoints (i.e., without traversing a NAT or relay server), by providing a minimal set of local IP addresses to the application for use in the ICE process. This ensures that applications that need true peer-to-peer routing for bandwidth or latency reasons can operate successfully. However, it should be possible to suppress these addresses (with the resultant impact on direct connections) if desired. - 3. By default, WebRTC traffic should not be sent through proxy - servers, due to the media quality problems associated with + 3. By default, WebRTC traffic should not be sent through application + proxy servers, due to the media quality problems associated with sending WebRTC traffic over TCP, which is almost always used when - communicating with proxies, as well as proxy performance issues - that may result from proxying WebRTC's long-lived, high-bandwidth - connections. However, it should be possible to force WebRTC to - send its traffic through a configured proxy if desired. + communicating with such proxies, as well as proxy performance + issues that may result from proxying WebRTC's long-lived, high- + bandwidth connections. However, it should be possible to force + WebRTC to send its traffic through a configured proxy if desired. 5.2. Modes and Recommendations Based on these ideas, we define four specific modes of WebRTC behavior, reflecting different media quality/privacy tradeoffs: Mode 1: Enumerate all addresses: WebRTC MUST use all network interfaces to attempt communication with STUN servers, TURN servers, or peers. This will converge on the best media path, and is ideal when media performance is the highest priority, but it discloses the most information. Mode 2: Default route + associated local addresses: WebRTC MUST follow the kernel routing table rules, which will typically cause media packets to take the same route as the - application's HTTP traffic. In addition, the private IPv4 - and IPv6 addresses associated with the kernel-chosen - interface MUST be discovered and provided to the - application. This ensures that direct connections can still - be established in this mode. + application's HTTP traffic. If an application TURN server + is present, the preferred route MUST be through this TURN + server. Once an interface has been chosen, the private IPv4 + and IPv6 addresses associated with this interface MUST be + discovered and provided to the application. This ensures + that direct connections can still be established in this + mode. Mode 3: Default route only: This is the the same as Mode 2, except that the associated private addresses MUST NOT be provided; the only IP addresses gathered are those discovered via mechanisms like STUN and TURN (on the default route). This may cause traffic to hairpin through a NAT, fall back to an application TURN server, or fail altogether, with resulting quality implications. - Mode 4: Force proxy: This is the same as Mode 3, but all WebRTC - media traffic is forced 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 media quality, in addition to any performance - considerations associated with sending all WebRTC media - through the proxy server. + Mode 4: Force proxy: This is the same as Mode 3, but when the + application's HTTP traffic is sent through an application + proxy, WebRTC media traffic MUST also be proxied. 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 media + quality, in addition to any performance considerations + associated with sending all WebRTC media through the proxy + server. Mode 1 MUST only be used when user consent has been provided. The details of this consent are left to the implementation; one potential mechanism is to tie this consent to getUserMedia consent. - Alternatively, implementations can provide a specific mechanism to obtain user consent. In cases where user consent has not been obtained, Mode 2 SHOULD be used. These defaults provide a reasonable tradeoff that permits trusted WebRTC applications to achieve optimal network performance, but gives applications without consent (e.g., 1-way streaming or data channel applications) only the minimum information needed to achieve direct @@ -247,25 +254,27 @@ These defaults provide a reasonable tradeoff that permits trusted WebRTC applications to achieve optimal network performance, but gives applications without consent (e.g., 1-way streaming or data channel applications) only the minimum information needed to achieve direct connections, as defined in Mode 2. However, implementations MAY choose stricter modes if desired, e.g., if a user indicates they want all WebRTC traffic to follow the default route. Note that the suggested defaults can still be used even for organizations that want all external WebRTC traffic to traverse a - proxy, simply by setting an organizational firewall policy that - allows WebRTC traffic to only leave through the proxy. This provides - a way to ensure the proxy is used for any external traffic, but - avoids the performance issues associated with Mode 4 (where all media - is forced through said proxy) for intra-organization traffic. + proxy or enterprise TURN server, simply by setting an organizational + firewall policy that allows WebRTC traffic to only leave through the + proxy or TURN server. This provides a way to ensure the proxy or + TURN server is used for any external traffic, but still allows direct + connections (and, in the proxy case, avoids the performance issues + associated with forcing media through said proxy) for intra- + organization traffic. 6. Implementation Guidance This section provides guidance to WebRTC implementations on how to implement the policies described above. 6.1. Ensuring Normal Routing When trying to follow typical IP routing, the simplest approach is to bind the sockets used for peer-to-peer connections to the wildcard @@ -390,26 +398,35 @@ [RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013, . [RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer Protocol (HTTP/1.1): Message Syntax and Routing", RFC 7230, DOI 10.17487/RFC7230, June 2014, . + [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- + Time Communication Use Cases and Requirements", RFC 7478, + DOI 10.17487/RFC7478, March 2015, + . + [RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089, DOI 10.17487/RFC8089, February 2017, . Appendix A. Change log + Changes in draft -08: + + o Discuss how enterprise TURN servers should be handled. + Changes in draft -07: o Clarify consent guidance. Changes in draft -06: o Clarify recommendations. o Split implementation guidance into two sections.