draft-ietf-rtcweb-ip-handling-07.txt   draft-ietf-rtcweb-ip-handling-08.txt 
Network Working Group J. Uberti Network Working Group J. Uberti
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track G. Shieh Intended status: Standards Track G. Shieh
Expires: October 20, 2018 Facebook Expires: December 5, 2018 Facebook
April 18, 2018 June 3, 2018
WebRTC IP Address Handling Requirements WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-07 draft-ietf-rtcweb-ip-handling-08
Abstract Abstract
This document provides information and requirements for how IP This document provides information and requirements for how IP
addresses should be handled by WebRTC implementations. addresses should be handled by WebRTC implementations.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
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Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
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time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." 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 Notice
Copyright (c) 2018 IETF Trust and the persons identified as the Copyright (c) 2018 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(https://trustee.ietf.org/license-info) in effect on the date of (https://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2 3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 2
4. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 4. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
5. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4 5. Detailed Design . . . . . . . . . . . . . . . . . . . . . . . 4
5.1. Principles . . . . . . . . . . . . . . . . . . . . . . . 4 5.1. Principles . . . . . . . . . . . . . . . . . . . . . . . 4
5.2. Modes and Recommendations . . . . . . . . . . . . . . . . 5 5.2. Modes and Recommendations . . . . . . . . . . . . . . . . 5
6. Implementation Guidance . . . . . . . . . . . . . . . . . . . 6 6. Implementation Guidance . . . . . . . . . . . . . . . . . . . 6
6.1. Ensuring Normal Routing . . . . . . . . . . . . . . . . . 6 6.1. Ensuring Normal Routing . . . . . . . . . . . . . . . . . 6
6.2. Determining Host Candidates . . . . . . . . . . . . . . . 6 6.2. Determining Host Candidates . . . . . . . . . . . . . . . 7
7. Application Guidance . . . . . . . . . . . . . . . . . . . . 7 7. Application Guidance . . . . . . . . . . . . . . . . . . . . 7
8. Security Considerations . . . . . . . . . . . . . . . . . . . 7 8. Security Considerations . . . . . . . . . . . . . . . . . . . 7
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 8
11.1. Normative References . . . . . . . . . . . . . . . . . . 8 11.1. Normative References . . . . . . . . . . . . . . . . . . 8
11.2. Informative References . . . . . . . . . . . . . . . . . 8 11.2. Informative References . . . . . . . . . . . . . . . . . 8
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 9 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 9
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 10 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 11
1. Introduction 1. Introduction
One of WebRTC's key features is its support of peer-to-peer One of WebRTC's key features is its support of peer-to-peer
connections. However, when establishing such a connection, which connections. However, when establishing such a connection, which
involves connection attempts from various IP addresses, WebRTC may involves connection attempts from various IP addresses, WebRTC may
allow a web application to learn additional information about the allow a web application to learn additional information about the
user compared to an application that only uses the Hypertext Transfer user compared to an application that only uses the Hypertext Transfer
Protocol (HTTP) [RFC7230]. This may be problematic in certain cases. Protocol (HTTP) [RFC7230]. This may be problematic in certain cases.
This document summarizes the concerns, and makes recommendations on This document summarizes the concerns, and makes recommendations on
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interfaces (a "split-tunnel" VPN), WebRTC will discover not only interfaces (a "split-tunnel" VPN), WebRTC will discover not only
the public address for the VPN, but also the ISP public address the public address for the VPN, but also the ISP public address
over which the VPN is running. over which the VPN is running.
2. If the client is behind a Network Address Translator (NAT), the 2. If the client is behind a Network Address Translator (NAT), the
client's private IP addresses, often [RFC1918] addresses, can be client's private IP addresses, often [RFC1918] addresses, can be
learned. learned.
3. If the client is behind a proxy (a client-configured "classical 3. If the client is behind a proxy (a client-configured "classical
application proxy", as defined in [RFC1919], Section 3), but application proxy", as defined in [RFC1919], Section 3), but
direct access to the Internet is also supported, WebRTC's STUN direct access to the Internet is permitted, WebRTC's STUN checks
checks will bypass the proxy and reveal the public IP address of will bypass the proxy and reveal the public IP address of the
the client. 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 Of these three concerns, #1 is the most significant, because for some
users, the purpose of using a VPN is for anonymity. However, users, the purpose of using a VPN is for anonymity. However,
different VPN users will have different needs, and some VPN users different VPN users will have different needs, and some VPN users
(e.g., corporate VPN users) may in fact prefer WebRTC to send media (e.g., corporate VPN users) may in fact prefer WebRTC to send media
traffic directly, i.e., not through the VPN. traffic directly, i.e., not through the VPN.
#2 is considered to be a less significant concern, given that the #2 is considered to be a less significant concern, given that the
local address values often contain minimal information (e.g., local address values often contain minimal information (e.g.,
192.168.0.2), or have built-in privacy protection (e.g., the 192.168.0.2), or have built-in privacy protection (e.g., the
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information in a way that might violate user expectations. information in a way that might violate user expectations.
5. Detailed Design 5. Detailed Design
5.1. Principles 5.1. Principles
The key principles for our framework are stated below: The key principles for our framework are stated below:
1. By default, WebRTC traffic should follow typical IP routing, 1. By default, WebRTC traffic should follow typical IP routing,
i.e., WebRTC should use the same interface used for HTTP traffic, i.e., WebRTC should use the same interface used for HTTP traffic,
and only the system's 'typical' public addresses should be and only the system's 'typical' public addresses (or those of an
visible to the application. However, in the interest of optimal enterprise TURN server, if present) should be visible to the
media quality, it should be possible to enable WebRTC to make use application. However, in the interest of optimal media quality,
of all network interfaces to determine the ideal route. 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- 2. By default, WebRTC should be able to negotiate direct peer-to-
peer connections between endpoints (i.e., without traversing a peer connections between endpoints (i.e., without traversing a
NAT or relay server), by providing a minimal set of local IP NAT or relay server), by providing a minimal set of local IP
addresses to the application for use in the ICE process. This addresses to the application for use in the ICE process. This
ensures that applications that need true peer-to-peer routing for ensures that applications that need true peer-to-peer routing for
bandwidth or latency reasons can operate successfully. However, bandwidth or latency reasons can operate successfully. However,
it should be possible to suppress these addresses (with the it should be possible to suppress these addresses (with the
resultant impact on direct connections) if desired. resultant impact on direct connections) if desired.
3. By default, WebRTC traffic should not be sent through proxy 3. By default, WebRTC traffic should not be sent through application
servers, due to the media quality problems associated with proxy servers, due to the media quality problems associated with
sending WebRTC traffic over TCP, which is almost always used when sending WebRTC traffic over TCP, which is almost always used when
communicating with proxies, as well as proxy performance issues communicating with such proxies, as well as proxy performance
that may result from proxying WebRTC's long-lived, high-bandwidth issues that may result from proxying WebRTC's long-lived, high-
connections. However, it should be possible to force WebRTC to bandwidth connections. However, it should be possible to force
send its traffic through a configured proxy if desired. WebRTC to send its traffic through a configured proxy if desired.
5.2. Modes and Recommendations 5.2. Modes and Recommendations
Based on these ideas, we define four specific modes of WebRTC Based on these ideas, we define four specific modes of WebRTC
behavior, reflecting different media quality/privacy tradeoffs: behavior, reflecting different media quality/privacy tradeoffs:
Mode 1: Enumerate all addresses: WebRTC MUST use all network Mode 1: Enumerate all addresses: WebRTC MUST use all network
interfaces to attempt communication with STUN servers, TURN interfaces to attempt communication with STUN servers, TURN
servers, or peers. This will converge on the best media servers, or peers. This will converge on the best media
path, and is ideal when media performance is the highest path, and is ideal when media performance is the highest
priority, but it discloses the most information. priority, but it discloses the most information.
Mode 2: Default route + associated local addresses: WebRTC MUST Mode 2: Default route + associated local addresses: WebRTC MUST
follow the kernel routing table rules, which will typically follow the kernel routing table rules, which will typically
cause media packets to take the same route as the cause media packets to take the same route as the
application's HTTP traffic. In addition, the private IPv4 application's HTTP traffic. If an application TURN server
and IPv6 addresses associated with the kernel-chosen is present, the preferred route MUST be through this TURN
interface MUST be discovered and provided to the server. Once an interface has been chosen, the private IPv4
application. This ensures that direct connections can still and IPv6 addresses associated with this interface MUST be
be established in this mode. 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 Mode 3: Default route only: This is the the same as Mode 2, except
that the associated private addresses MUST NOT be provided; that the associated private addresses MUST NOT be provided;
the only IP addresses gathered are those discovered via the only IP addresses gathered are those discovered via
mechanisms like STUN and TURN (on the default route). This mechanisms like STUN and TURN (on the default route). This
may cause traffic to hairpin through a NAT, fall back to an may cause traffic to hairpin through a NAT, fall back to an
application TURN server, or fail altogether, with resulting application TURN server, or fail altogether, with resulting
quality implications. quality implications.
Mode 4: Force proxy: This is the same as Mode 3, but all WebRTC Mode 4: Force proxy: This is the same as Mode 3, but when the
media traffic is forced through a proxy, if one is application's HTTP traffic is sent through an application
configured. If the proxy does not support UDP (as is the proxy, WebRTC media traffic MUST also be proxied. If the
case for all HTTP and most SOCKS [RFC1928] proxies), or the proxy does not support UDP (as is the case for all HTTP and
WebRTC implementation does not support UDP proxying, the use most SOCKS [RFC1928] proxies), or the WebRTC implementation
of UDP will be disabled, and TCP will be used to send and does not support UDP proxying, the use of UDP will be
receive media through the proxy. Use of TCP will result in disabled, and TCP will be used to send and receive media
reduced media quality, in addition to any performance through the proxy. Use of TCP will result in reduced media
considerations associated with sending all WebRTC media quality, in addition to any performance considerations
through the proxy server. associated with sending all WebRTC media through the proxy
server.
Mode 1 MUST only be used when user consent has been provided. The Mode 1 MUST only be used when user consent has been provided. The
details of this consent are left to the implementation; one potential details of this consent are left to the implementation; one potential
mechanism is to tie this consent to getUserMedia consent. mechanism is to tie this consent to getUserMedia consent.
Alternatively, implementations can provide a specific mechanism to Alternatively, implementations can provide a specific mechanism to
obtain user consent. obtain user consent.
In cases where user consent has not been obtained, Mode 2 SHOULD be In cases where user consent has not been obtained, Mode 2 SHOULD be
used. used.
These defaults provide a reasonable tradeoff that permits trusted These defaults provide a reasonable tradeoff that permits trusted
WebRTC applications to achieve optimal network performance, but gives WebRTC applications to achieve optimal network performance, but gives
applications without consent (e.g., 1-way streaming or data channel applications without consent (e.g., 1-way streaming or data channel
applications) only the minimum information needed to achieve direct applications) only the minimum information needed to achieve direct
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These defaults provide a reasonable tradeoff that permits trusted These defaults provide a reasonable tradeoff that permits trusted
WebRTC applications to achieve optimal network performance, but gives WebRTC applications to achieve optimal network performance, but gives
applications without consent (e.g., 1-way streaming or data channel applications without consent (e.g., 1-way streaming or data channel
applications) only the minimum information needed to achieve direct applications) only the minimum information needed to achieve direct
connections, as defined in Mode 2. However, implementations MAY connections, as defined in Mode 2. However, implementations MAY
choose stricter modes if desired, e.g., if a user indicates they want choose stricter modes if desired, e.g., if a user indicates they want
all WebRTC traffic to follow the default route. all WebRTC traffic to follow the default route.
Note that the suggested defaults can still be used even for Note that the suggested defaults can still be used even for
organizations that want all external WebRTC traffic to traverse a organizations that want all external WebRTC traffic to traverse a
proxy, simply by setting an organizational firewall policy that proxy or enterprise TURN server, simply by setting an organizational
allows WebRTC traffic to only leave through the proxy. This provides firewall policy that allows WebRTC traffic to only leave through the
a way to ensure the proxy is used for any external traffic, but proxy or TURN server. This provides a way to ensure the proxy or
avoids the performance issues associated with Mode 4 (where all media TURN server is used for any external traffic, but still allows direct
is forced through said proxy) for intra-organization traffic. connections (and, in the proxy case, avoids the performance issues
associated with forcing media through said proxy) for intra-
organization traffic.
6. Implementation Guidance 6. Implementation Guidance
This section provides guidance to WebRTC implementations on how to This section provides guidance to WebRTC implementations on how to
implement the policies described above. implement the policies described above.
6.1. Ensuring Normal Routing 6.1. Ensuring Normal Routing
When trying to follow typical IP routing, the simplest approach is to When trying to follow typical IP routing, the simplest approach is to
bind the sockets used for peer-to-peer connections to the wildcard bind the sockets used for peer-to-peer connections to the wildcard
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[RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow [RFC7016] Thornburgh, M., "Adobe's Secure Real-Time Media Flow
Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013, Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,
<https://www.rfc-editor.org/info/rfc7016>. <https://www.rfc-editor.org/info/rfc7016>.
[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer [RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Message Syntax and Routing", Protocol (HTTP/1.1): Message Syntax and Routing",
RFC 7230, DOI 10.17487/RFC7230, June 2014, RFC 7230, DOI 10.17487/RFC7230, June 2014,
<https://www.rfc-editor.org/info/rfc7230>. <https://www.rfc-editor.org/info/rfc7230>.
[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478,
DOI 10.17487/RFC7478, March 2015,
<https://www.rfc-editor.org/info/rfc7478>.
[RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089, [RFC8089] Kerwin, M., "The "file" URI Scheme", RFC 8089,
DOI 10.17487/RFC8089, February 2017, DOI 10.17487/RFC8089, February 2017,
<https://www.rfc-editor.org/info/rfc8089>. <https://www.rfc-editor.org/info/rfc8089>.
Appendix A. Change log Appendix A. Change log
Changes in draft -08:
o Discuss how enterprise TURN servers should be handled.
Changes in draft -07: Changes in draft -07:
o Clarify consent guidance. o Clarify consent guidance.
Changes in draft -06: Changes in draft -06:
o Clarify recommendations. o Clarify recommendations.
o Split implementation guidance into two sections. o Split implementation guidance into two sections.
 End of changes. 16 change blocks. 
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