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RAI This document provides information and requirements for how IP addresses should be handled by Web Real-Time Communication (WebRTC) implementations.

Status of This Memo This is an Internet Standards Track document. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at .

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Table of Contents

• .  Introduction
• .  Terminology
• .  Problem Statement
• .  Goals
• .  Detailed Design
  • .  Principles
  • .  Modes and Recommendations
• .  Implementation Guidance
  • .  Ensuring Normal Routing
  • .  Determining Associated Local Addresses
• .  Application Guidance
• .  Security Considerations
• .  IANA Considerations
• . References
  • .  Normative References
  • .  Informative References
• Acknowledgements
• Authors' Addresses

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) . This may be problematic in certain cases. This document summarizes the concerns and makes recommendations on how WebRTC implementations should best handle the trade-off between privacy and media performance.

Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 when, and only when, they appear in all capitals, as shown here.

Problem Statement In order to establish a peer-to-peer connection, WebRTC implementations use Interactive Connectivity Establishment (ICE) . ICE attempts to discover multiple IP addresses using techniques such as Session Traversal Utilities for NAT (STUN) and Traversal Using Relays around NAT (TURN) 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 or virtual addresses and its public Internet addresses. These addresses are provided 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 additional 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 can 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 addresses, can be learned.
3. If the client is behind a proxy (a client-configured "classical application proxy", as defined in ), but 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 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 proxy server.

Of these three concerns, the first 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. The second concern is less significant but valid nonetheless. The core issue is that web applications can learn about addresses that are not exposed to the Internet; typically, these address are IPv4, but they can also be IPv6, as in the case of NAT64 . While disclosure of the IPv6 addresses recommended by is fairly benign due to their intentionally short lifetimes, IPv4 addresses present some challenges. Although private IPv4 addresses often contain minimal entropy (e.g., 192.168.0.2, a fairly common address), in the worst case, they can contain 24 bits of entropy with an indefinite lifetime. As such, they can be a fairly significant fingerprinting surface. In addition, intranet web sites can be attacked more easily when their IPv4 address range is externally known. Private IP addresses can also act as an identifier that allows web applications running in isolated browsing contexts (e.g., normal and private browsing) to learn that they are running on the same device. This could allow the application sessions to be correlated, defeating some of the privacy protections provided by isolation. It should be noted that private addresses are just one potential mechanism for this correlation and this is an area for further study. The third concern is the least common, 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 media quality. Note also that these concerns predate WebRTC; Adobe Flash Player has provided similar functionality since the introduction of Real-Time Media Flow Protocol (RTMFP) support in 2008.

Goals WebRTC's support of secure peer-to-peer connections facilitates deployment of decentralized systems, which can have privacy benefits. As a result, blunt solutions that disable WebRTC or make it significantly harder to use are undesirable. 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 trade-offs 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.

Detailed Design

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 (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) when such connections are possible. This ensures that applications that need true peer-to-peer routing for bandwidth or latency reasons can operate successfully.
3. It should be possible to configure WebRTC to not disclose private local IP addresses, to avoid the issues associated with web applications learning such addresses. This document does not require this to be the default state, as there is no currently defined mechanism that can satisfy this requirement as well as the aforementioned requirement to allow direct peer-to-peer connections.
4. 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 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.

Modes and Recommendations Based on these ideas, we define four specific modes of WebRTC behavior, reflecting different media quality/privacy trade-offs:

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. If an enterprise 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 as host candidates. This ensures that direct connections can still be established in this mode.

Mode 3 - Default route only:

This is 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 when the application's HTTP traffic is sent through a 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 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 NOT be used unless 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 (device permissions) consent, described in . 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 trade-off 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. Future documents may define additional modes and/or update the recommended default modes. Note that the suggested defaults can still be used even for organizations that want all external WebRTC traffic to traverse a 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.

Implementation Guidance This section provides guidance to WebRTC implementations on how to implement the policies described above.

Ensuring Normal Routing When trying to follow typical IP routing, as required by Modes 2 and 3, the simplest approach is to bind() the sockets used for peer-to-peer 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 still be determined as mentioned below.

Determining Associated Local Addresses When binding to a wildcard address, some extra work is needed to determine the associated local address required by Mode 2, which we define as the source address that would be used for any packets sent to the web application host (assuming that UDP and TCP get the same routing treatment). Use of the web-application host as a destination ensures the right source address is selected, regardless of where the application resides (e.g., on an intranet). First, the appropriate remote IPv4/IPv6 address is obtained by resolving the host component of the web application URI . If the client is behind a proxy and cannot resolve these IPs via DNS, the address of the proxy can be used instead. Or, if the web application was loaded from a file:// URI rather than over the network, the implementation can fall back to a well-known DNS name or IP address. Once a suitable remote IP has been determined, the implementation can create a UDP socket, bind() it to the appropriate wildcard address, and then connect() to the remote IP. Generally, this results in the socket being assigned a local address based on the kernel routing table, without sending any packets over the network. Finally, the socket can be queried using getsockname() or the equivalent to determine the appropriate local address.

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 is 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 is in use; this knowledge can be useful for diagnostic purposes.

Security Considerations This document describes several potential privacy and security concerns associated with WebRTC peer-to-peer connections and provides mechanisms and recommendations for WebRTC implementations to address these concerns.

IANA Considerations This document has no IANA actions.

References Normative References In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. A Uniform Resource Identifier (URI) is a compact sequence of characters that identifies an abstract or physical resource. This specification defines the generic URI syntax and a process for resolving URI references that might be in relative form, along with guidelines and security considerations for the use of URIs on the Internet. The URI syntax defines a grammar that is a superset of all valid URIs, allowing an implementation to parse the common components of a URI reference without knowing the scheme-specific requirements of every possible identifier. This specification does not define a generative grammar for URIs; that task is performed by the individual specifications of each URI scheme. [STANDARDS-TRACK] Session Traversal Utilities for NAT (STUN) is a protocol that serves as a tool for other protocols in dealing with Network Address Translator (NAT) traversal. It can be used by an endpoint to determine the IP address and port allocated to it by a NAT. It can also be used to check connectivity between two endpoints, and as a keep-alive protocol to maintain NAT bindings. STUN works with many existing NATs, and does not require any special behavior from them. STUN is not a NAT traversal solution by itself. Rather, it is a tool to be used in the context of a NAT traversal solution. This is an important change from the previous version of this specification (RFC 3489), which presented STUN as a complete solution. This document obsoletes RFC 3489. [STANDARDS-TRACK] If a host is located behind a NAT, then in certain situations it can be impossible for that host to communicate directly with other hosts (peers). In these situations, it is necessary for the host to use the services of an intermediate node that acts as a communication relay. This specification defines a protocol, called TURN (Traversal Using Relays around NAT), that allows the host to control the operation of the relay and to exchange packets with its peers using the relay. TURN differs from some other relay control protocols in that it allows a client to communicate with multiple peers using a single relay address. [STANDARDS-TRACK] This document provides a more complete specification of the "file" Uniform Resource Identifier (URI) scheme and replaces the very brief definition in Section 3.10 of RFC 1738. It defines a common syntax that is intended to interoperate across the broad spectrum of existing usages. At the same time, it notes some other current practices around the use of file URIs. RFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings. This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based communication. This protocol is called Interactive Connectivity Establishment (ICE). ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN). This document obsoletes RFC 5245. Informative References This document describes address allocation for private internets. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document explains "classical" and "transparent" proxy techniques and attempts to provide rules to help determine when each proxy system may be used without causing problems. This memo provides information for the Internet community. This memo does not specify an Internet standard of any kind. This memo describes a protocol that is an evolution of the previous version of the protocol, version 4 [1]. This new protocol stems from active discussions and prototype implementations. [STANDARDS-TRACK] Nodes use IPv6 stateless address autoconfiguration to generate addresses using a combination of locally available information and information advertised by routers. Addresses are formed by combining network prefixes with an interface identifier. On an interface that contains an embedded IEEE Identifier, the interface identifier is typically derived from it. On other interface types, the interface identifier is generated through other means, for example, via random number generation. This document describes an extension to IPv6 stateless address autoconfiguration for interfaces whose interface identifier is derived from an IEEE identifier. Use of the extension causes nodes to generate global scope addresses from interface identifiers that change over time, even in cases where the interface contains an embedded IEEE identifier. Changing the interface identifier (and the global scope addresses generated from it) over time makes it more difficult for eavesdroppers and other information collectors to identify when different addresses used in different transactions actually correspond to the same node. [STANDARDS-TRACK] This memo describes Adobe's Secure Real-Time Media Flow Protocol (RTMFP), an endpoint-to-endpoint communication protocol designed to securely transport parallel flows of real-time video, audio, and data messages, as well as bulk data, over IP networks. RTMFP has features that make it effective for peer-to-peer (P2P) as well as client-server communications, even when Network Address Translators (NATs) are used. The Hypertext Transfer Protocol (HTTP) is a stateless application-level protocol for distributed, collaborative, hypertext information systems. This document provides an overview of HTTP architecture and its associated terminology, defines the "http" and "https" Uniform Resource Identifier (URI) schemes, defines the HTTP/1.1 message syntax and parsing requirements, and describes related security concerns for implementations. This document describes web-based real-time communication use cases. Requirements on the browser functionality are derived from the use cases. This document was developed in an initial phase of the work with rather minor updates at later stages. It has not really served as a tool in deciding features or scope for the WG's efforts so far. It is being published to record the early conclusions of the WG. It will not be used as a set of rigid guidelines that specifications and implementations will be held to in the future.

Acknowledgements Several people provided input into this document, including , , , , , , , and .

Authors' Addresses Google

747 6th St S Kirkland WA 98033 United States of America justin@uberti.name

333 Elliott Ave W #500 Seattle WA 98119 United States of America guoweis@gmail.com