Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track G. Shieh
Expires: August 15, 2018 Facebook
February 11, 2018

WebRTC IP Address Handling Requirements


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

Status of This Memo

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This Internet-Draft will expire on August 15, 2018.

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Table of Contents

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.,, 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:

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 ( 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:

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

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997.

11.2. Informative References

[I-D.ietf-rtcweb-transports] Alvestrand, H., "Transports for WebRTC", Internet-Draft draft-ietf-rtcweb-transports-17, October 2016.
[RFC1918] Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G. and E. Lear, "Address Allocation for Private Internets", BCP 5, RFC 1918, DOI 10.17487/RFC1918, February 1996.
[RFC1919] Chatel, M., "Classical versus Transparent IP Proxies", RFC 1919, DOI 10.17487/RFC1919, March 1996.
[RFC1928] Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D. and L. Jones, "SOCKS Protocol Version 5", RFC 1928, DOI 10.17487/RFC1928, March 1996.
[RFC3986] Berners-Lee, T., Fielding, R. and L. Masinter, "Uniform Resource Identifier (URI): Generic Syntax", STD 66, RFC 3986, DOI 10.17487/RFC3986, January 2005.
[RFC4941] Narten, T., Draves, R. and S. Krishnan, "Privacy Extensions for Stateless Address Autoconfiguration in IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", RFC 5245, DOI 10.17487/RFC5245, April 2010.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P. and D. Wing, "Session Traversal Utilities for NAT (STUN)", RFC 5389, DOI 10.17487/RFC5389, October 2008.
[RFC5766] Mahy, R., Matthews, P. and J. Rosenberg, "Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)", RFC 5766, DOI 10.17487/RFC5766, April 2010.
[RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013.
[RFC7230] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol (HTTP/1.1): Message Syntax and Routing", RFC 7230, DOI 10.17487/RFC7230, June 2014.
[RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089, DOI 10.17487/RFC8089, February 2017.

Appendix A. Change log

Changes in draft -05:

Changes in draft -04:

Changes in draft -03:

Changes in draft -02:

Changes in draft -01:

Changes in draft -00:

Authors' Addresses

Justin Uberti Google 747 6th St S Kirkland, WA 98033 USA EMail:
Guo-wei Shieh Facebook 1101 Dexter Ave Seattle, WA 98109 USA EMail: