WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-05

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 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 August 15, 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 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. Terminology
• 3. Problem Statement
• 4. Goals
• 5. Detailed Design
• 5.1. Principles
• 5.2. Modes and Recommendations
• 6. Implementation Guidance
• 7. Application Guidance
• 8. Security Considerations
• 9. IANA Considerations
• 10. Acknowledgements
• 11. References
• 11.1. Normative References
• 11.2. Informative References
• Appendix A. Change log
• Authors' Addresses

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 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 attempts to discover multiple IP addresses using techniques such as Session Traversal Utilities for NAT (STUN) [RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766], and then checks the connectivity of each local-address-remote-address pair in order to select the best one. The addresses that are collected usually consist of an endpoint's private physical/virtual addresses and its public Internet addresses.

These addresses are exposed upwards to the web application, so that they can be communicated to the remote endpoint for its checks. This allows the application to learn more about the local network configuration than it would from a typical HTTP scenario, in which the web server would only see a single public Internet address, i.e., the address from which the HTTP request was sent.

The information revealed falls into three categories:

1. If the client is multihomed, additional public IP addresses for the client can be learned. In particular, if the client tries to hide its physical location through a Virtual Private Network (VPN), and the VPN and local OS support routing over multiple 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.

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 [RFC4941] IPv6 addresses recommended by [I-D.ietf-rtcweb-transports]).

#3 is the least common concern, as proxy administrators can already control this behavior through organizational firewall policy, and generally, forcing WebRTC traffic through a proxy server will have negative effects on both the proxy and on media quality.

Note also that these concerns predate WebRTC; Adobe Flash Player has provided similar functionality since the introduction of RTMFP [RFC7016] in 2008.

4. Goals

WebRTC's support of secure peer-to-peer connections facilitates deployment of decentralized systems, which can have privacy benefits. As a result, we want to avoid blunt solutions that disable WebRTC or make it significantly harder to use. This document takes a more nuanced approach, with the following goals:

• Provide a framework for understanding the problem so that controls might be provided to make different tradeoffs regarding performance and privacy concerns with WebRTC.
• Using that framework, define settings that enable peer-to-peer communications, each with a different balance between performance and privacy.
• Finally, provide recommendations for default settings that provide reasonable performance without also exposing addressing 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.
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 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.

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.

Mode 3:

Default route only: This is the the same as Mode 2, except that the associated private addressses 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.

The recommended defaults are as follows:

Mode 1 MUST only be used when user consent has been provided; this allows trusted WebRTC applications to achieve optimal network performance, but significanly limites the network information exposed to arbitrary web pages. The details of this consent are left to the implementation; one potential mechanism is to tie this consent to getUserMedia consent.

In cases where user consent has not been obtained, Mode 2 SHOULD be used. This allows applications to still achieve direct connections in many cases, even without consent (e.g., streaming or data channel applications). However, implementations MAY choose a stricter default policy in certain circumstances.

Note that these 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 of Mode 4 (where all media is forced 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.

When trying to follow typical IP routing, the simplest approach is to bind the sockets used for p2p connections to the wildcard addresses (0.0.0.0 for IPv4, :: for IPv6), which allows the OS to route WebRTC traffic the same way as it would HTTP traffic. STUN and TURN will work as usual, and host candidates can be determined as mentioned below.

In order to discover the correct local IP addresses, implementations can use the common trick of binding sockets to the wildcard addresses, connect()ing those sockets to the IPv4/IPv6 addresses of the web application (obtained by resolving the host component of its URI [RFC3986]) and then reading the bound local addresses via getsockname(). This requires no data exchange; it simply provides a mechanism for applications to retrieve the desired information from the kernel routing table.

Use of the web application IPs ensures the right local IPs are selected, regardless of where the application is hosted (e.g., on an intranet). If the client is behind a proxy and cannot resolve the IPs via DNS, the IPv4/v6 addresses of the proxy can be used instead. If the web application was loaded from a file:// URI [RFC8089], the implementation can fall back to a well-known DNS name or IP address.

7. Application Guidance

The recommendations mentioned in this document may cause certain WebRTC applications to malfunction. In order to be robust in all scenarios, the following guidelines are provided for applications:

• Applications SHOULD deploy a TURN server with support for both UDP and TCP connections to the server. This ensures that connectivity can still be established, even when Mode 3 or 4 are in use, assuming the TURN server can be reached.
• Applications SHOULD detect when they don't have access to the full set of ICE candidates by checking for the presence of host candidates. If no host candidates are present, Mode 3 or 4 above is in use; this knowledge can be useful for diagnostic purposes.

8. Security Considerations

This document is entirely devoted to security considerations.

9. IANA Considerations

This document requires no actions from IANA.

10. Acknowledgements

Several people provided input into this document, including Bernard Aboba, Harald Alvestrand, Ted Hardie, Matthew Kaufmann, Eric Rescorla, Adam Roach, and Martin Thomson.

11. References

11.1. Normative References

11.2. Informative References

Appendix A. Change log

Changes in draft -05:

• Separated framework definition from implementation techniques.
• Removed RETURN references.
• Use origin when determining local IPs, rather than a well-known IP.

Changes in draft -04:

• Rewording and cleanup in abstract, intro, and problem statement.
• Added 2119 boilerplate.
• Fixed weird reference spacing.
• Expanded acronyms on first use.
• Removed 8.8.8.8 mention.
• Removed mention of future browser considerations.

Changes in draft -03:

• Clarified when to use which modes.
• Added 2119 qualifiers to make normative statements.
• Defined 'proxy'.
• Mentioned split tunnels in problem statement.

Changes in draft -02:

• Recommendations -> Requirements
• Updated text regarding consent.

Changes in draft -01:

• Incorporated feedback from Adam Roach; changes to discussion of cam/mic permission, as well as use of proxies, and various editorial changes.
• Added several more references.

Changes in draft -00:

• Published as WG draft.

Authors' Addresses