draft-ietf-rtcweb-ip-handling-05.txt   draft-ietf-rtcweb-ip-handling-06.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: August 15, 2018 Facebook Expires: September 2, 2018 Facebook
February 11, 2018 March 1, 2018
WebRTC IP Address Handling Requirements WebRTC IP Address Handling Requirements
draft-ietf-rtcweb-ip-handling-05 draft-ietf-rtcweb-ip-handling-06
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
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/. Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
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 August 15, 2018. This Internet-Draft will expire on September 2, 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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Table of Contents Table of Contents
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
6.2. Determining Host Candidates . . . . . . . . . . . . . . . 6
7. Application Guidance . . . . . . . . . . . . . . . . . . . . 7 7. Application Guidance . . . . . . . . . . . . . . . . . . . . 7
8. Security Considerations . . . . . . . . . . . . . . . . . . . 7 8. Security Considerations . . . . . . . . . . . . . . . . . . . 7
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 7 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 7
11.1. Normative References . . . . . . . . . . . . . . . . . . 7 11.1. Normative References . . . . . . . . . . . . . . . . . . 8
11.2. Informative References . . . . . . . . . . . . . . . . . 7 11.2. Informative References . . . . . . . . . . . . . . . . . 8
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 9 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 9
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 10 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 10
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
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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. In addition, the private IPv4
and IPv6 addresses associated with the kernel-chosen and IPv6 addresses associated with the kernel-chosen
interface MUST be discovered and provided to the interface MUST be discovered and provided to the
application. This ensures that direct connections can still application. This ensures that direct connections can still
be established in this mode. 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 addressses 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 all WebRTC
media traffic is forced through a proxy, if one is media traffic is forced through a proxy, if one is
configured. If the proxy does not support UDP (as is the configured. If the proxy does not support UDP (as is the
case for all HTTP and most SOCKS [RFC1928] proxies), or the case for all HTTP and most SOCKS [RFC1928] proxies), or the
WebRTC implementation does not support UDP proxying, the use WebRTC implementation does not support UDP proxying, the use
of UDP will be disabled, and TCP will be used to send and of UDP will be disabled, and TCP will be used to send and
receive media through the proxy. Use of TCP will result in receive media through the proxy. Use of TCP will result in
reduced media quality, in addition to any performance reduced media quality, in addition to any performance
considerations associated with sending all WebRTC media considerations associated with sending all WebRTC media
through the proxy server. through the proxy server.
The recommended defaults are as follows: Mode 1 MUST only be used when user consent has been provided. The
details of this consent are left to the implementation; one potential
Mode 1 MUST only be used when user consent has been provided; this mechanism is to tie this consent to getUserMedia consent.
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 In cases where user consent has not been obtained, Mode 2 SHOULD be
used. This allows applications to still achieve direct connections used.
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 These defaults provide a reasonable tradeoff that permits trusted
that want all external WebRTC traffic to traverse a proxy, simply by WebRTC applications to achieve optimal network performance, but gives
setting an organizational firewall policy that allows WebRTC traffic applications without consent (e.g., 1-way streaming or data channel
to only leave through the proxy. This provides a way to ensure the applications) only the minimum information needed to achieve direct
proxy is used for any external traffic, but avoids the performance connections, as defined in Mode 2. However, implementations MAY
issues of Mode 4 (where all media is forced through said proxy) for choose stricter modes if desired, e.g., if a user indicates they want
intra-organization traffic. 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.
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
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 p2p connections to the wildcard addresses bind the sockets used for peer-to-peer connections to the wildcard
(0.0.0.0 for IPv4, :: for IPv6), which allows the OS to route WebRTC addresses (0.0.0.0 for IPv4, :: for IPv6), which allows the OS to
traffic the same way as it would HTTP traffic. STUN and TURN will route WebRTC traffic the same way as it would HTTP traffic. STUN and
work as usual, and host candidates can be determined as mentioned TURN will work as usual, and host candidates can still be determined
below. as mentioned below.
In order to discover the correct local IP addresses, implementations 6.2. Determining Host Candidates
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 When binding to a wildcard address, some extra work is needed to
selected, regardless of where the application is hosted (e.g., on an determine a suitable host candidate, which we define as the source
intranet). If the client is behind a proxy and cannot resolve the address that would be used for any packets sent to the web
IPs via DNS, the IPv4/v6 addresses of the proxy can be used instead. application host (assuming that UDP and TCP get the same routing).
If the web application was loaded from a file:// URI [RFC8089], the Use of the web application host as a destination ensures the right
implementation can fall back to a well-known DNS name or IP address. 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 [RFC3986].
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 [RFC8089], 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
tell it to 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 host candidate.
7. Application Guidance 7. Application Guidance
The recommendations mentioned in this document may cause certain The recommendations mentioned in this document may cause certain
WebRTC applications to malfunction. In order to be robust in all WebRTC applications to malfunction. In order to be robust in all
scenarios, the following guidelines are provided for applications: scenarios, the following guidelines are provided for applications:
o Applications SHOULD deploy a TURN server with support for both UDP o Applications SHOULD deploy a TURN server with support for both UDP
and TCP connections to the server. This ensures that connectivity and TCP connections to the server. This ensures that connectivity
can still be established, even when Mode 3 or 4 are in use, can still be established, even when Mode 3 or 4 are in use,
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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>.
[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 -06:
o Clarify recommendations.
o Split implementation guidance into two sections.
Changes in draft -05: Changes in draft -05:
o Separated framework definition from implementation techniques. o Separated framework definition from implementation techniques.
o Removed RETURN references. o Removed RETURN references.
o Use origin when determining local IPs, rather than a well-known o Use origin when determining local IPs, rather than a well-known
IP. IP.
Changes in draft -04: Changes in draft -04:
 End of changes. 14 change blocks. 
45 lines changed or deleted 66 lines changed or added

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