One document matched: draft-schwartz-rtcweb-return-03.xml
<?xml version="1.0" encoding="US-ASCII"?>
<!DOCTYPE rfc SYSTEM "rfc2629.dtd">
<?xml-stylesheet type="text/xsl" href="rfc2629.xslt" ?>
<?rfc toc="yes" ?>
<?rfc symrefs="yes" ?>
<?rfc iprnotified="no" ?>
<?rfc strict="yes" ?>
<?rfc compact="yes" ?>
<?rfc sortrefs="yes" ?>
<?rfc colonspace="yes" ?>
<?rfc rfcedstyle="no" ?>
<?rfc tocdepth="4"?>
<rfc category="std" docName="draft-schwartz-rtcweb-return-03" ipr="trust200902">
<front>
<title abbrev="RETURN">
Recursively Encapsulated TURN (RETURN) for Connectivity and Privacy in
WebRTC
</title>
<author fullname="Benjamin M. Schwartz" initials="B. M." surname="Schwartz">
<organization>Google</organization>
<address>
<postal>
<street>747 6th Ave S</street>
<city>Kirkland</city>
<region>WA</region>
<code>98033</code>
<country>USA</country>
</postal>
<email>bemasc@webrtc.org</email>
</address>
</author>
<date day="2" month="September" year="2014" />
<area>RAI</area>
<abstract>
<t>In the context of WebRTC, the concept of a local TURN proxy has been
suggested, but not reviewed in detail. WebRTC applications are already
using TURN to enhance connectivity and privacy. This document explains how
local TURN proxies and WebRTC applications can work together.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>TURN <xref target="RFC5766"></xref> is a protocol for communication
between a client and a TURN server, in order to route UDP traffic to and
from one or more peers. As noted in <xref target="RFC5766"></xref>, the
TURN relay server “typically sits in the public Internet”. In
a WebRTC context, if a TURN server is to be used, it is typically provided
by the application, either to provide connectivity between users whose NATs
would otherwise prevent it, or to obscure the identity of the participants
by concealing their IP addresses from one another.</t>
<t>In many enterprises, direct UDP transmissions are not permitted between
clients on the internal networks and external IP addresses, so media must
flow over TCP. To enable WebRTC services in such a situation, clients
must use TURN-TCP, or TURN-TLS. These configurations are not ideal: they
send all traffic over TCP, which leads to higher latency than would
otherwise be necessary, and they force the application provider to operate
a TURN server because WebRTC endpoints behind NAT cannot typically act as
TCP servers. These configurations may result in especially bad behaviors
when operating through TCP or HTTP proxies that were not designed to carry
real-time media streams.</t>
<t>To avoid forcing WebRTC media streams through a TCP stage, enterprise
network operators may operate a TURN server for their network, which can
be discovered by clients using TURN Auto-Discovery
<xref target="I-D.ietf-tram-turn-server-discovery"></xref>, or through a
proprietary mechanism. This TURN server may be placed inside the network,
with a firewall configuration allowing it to communicate with the public
internet, or it may be operated by the a third party outside the network,
with a firewall configuration that allows hosts inside the network. to
communicate with it. Use of the specified TURN server may be the only
way for clients on the network to achieve a high quality WebRTC
experience. This scenario is required to be supported by the WebRTC
requirements document
<xref target="I-D.ietf-rtcweb-use-cases-and-requirements"></xref> Section
3.3.5.1.</t>
<t>When the application intends to use a TURN server for identity
cloaking, and the enterprise network administrator intends to use a TURN
server for connectivity, there is a conflict. In current WebRTC
implementations, TURN can only be used on a single-hop basis in each
candidate,
but using only the enterprise's TURN server reveals information about the
user (e.g. organizational affiliation), and using only the application's
TURN server may be blocked by the network administrator, or may require
using TURN-TCP or TURN-TLS, resulting in a significant sacrifice in
latency.</t>
<t>To resolve this conflict, we introduce Recursively Encapsulated TURN,
a procedure that allows a WebRTC endpoint to route traffic through
multiple TURN servers, and get improved connectivity and privacy in
return.</t>
</section>
<section title="Goals">
<t>These goals are requirements on this document (not on implementations
of the specification).</t>
<section title="Connectivity">
<t>As noted in
<xref target="I-D.ietf-rtcweb-use-cases-and-requirements"></xref>
Section 3.3.5.1 and requirement F20, a WebRTC browser endpoint MUST be
able to direct UDP
connections through a designated TURN server configured by enterprise
policy (a “proxy”).</t>
<t>It MUST be possible to configure a WebRTC endpoint that supports
proxies to achieve connectivity no worse than if the endpoint were
operating at the proxy's address.</t>
<t>For efficiency, network administrators SHOULD be able to prevent
browsers from attempting to send traffic through routes that are already
known to be blocked.</t>
</section>
<section title="Privacy">
<t>To prevent WebRTC peers from determining each others' IP addresses,
applications MUST have the ability to direct all traffic through an
application-specified TURN server.</t>
<t>A compatible WebRTC browser MAY attempt to prevent a hostile web page
from determining the endpoint's public IP address. (This requirement is
documented in <xref target="I-D.ietf-rtcweb-security"></xref> Section
4.2.4. Note that the measures
proposed here are not sufficient by themselves to achieve this goal.
Implementing this specification in current browsers would still leave
many other ways for a malicious website to determine the endpoint's IP
address. Operating-system-wide VPN configurations are therefore
currently preferred for this purpose.)</t>
<t>A compatible WebRTC browser MAY allow the user to prevent
non-malicious web pages from accidentally revealing the IP address of
remote peers to a local passive network adversary. This ability SHOULD
NOT reduce performance when it is not in use. (Due to the difficulty of
distinguishing between stupidity and malice, this goal is principally
aspirational.)</t>
</section>
</section>
<section title="Concepts">
<t>To achieve our goals, we introduce the following new concepts:</t>
<section title="Proxy">
<t>In this document a “proxy” is any TURN server that was
provided by any mechanism other than through the standard
WebRTC-application ICE candidate provisioning API
<xref target="I-D.ietf-rtcweb-jsep"></xref>. If a proxy is to be used,
it will be the destination of traffic generated by the client. There is
no analogue to the transparent/intercepting HTTP proxy configuration,
which modifies traffic at the network layer. Mechanisms to configure a
proxy include Auto-Discovery
<xref target="I-D.ietf-tram-turn-server-discovery"></xref> and local
policy (<xref target="I-D.ietf-rtcweb-jsep"></xref>, "ICE candidate
policy").</t>
<t>In an application context, a proxy may be “active”
(producing candidates) or “inactive” (not in use, having no
effect on the context).</t>
</section>
<section title="Virtual interface">
<t>A typical WebRTC browser endpoint may have multiple network
interfaces available, such as wired ethernet, wireless ethernet, and
WAN. In this document, a “virtual interface” is a procedure
for generating ICE candidates that are not simply generated by a
particular physical interface. A virtual interface can produce
“host”, “server-reflexive”,
and “relay” candidates, but may be restricted to only some
type of candidate (e.g. UDP-only).</t>
</section>
<section title="Proxy configuration leakiness">
<t>“Leakiness” is an attribute of a proxy configuration.
This document defines two values for the “leakiness” of a
proxy configuration: “leaky” and “sealed”. Proxy
configuration, including leakiness, may be set by local policy
(<xref target="I-D.ietf-rtcweb-jsep"></xref>,
“ICE candidate policy”) or other mechanisms.</t>
<t>A leaky configuration adds a proxy and also allows the browser to use
routes that transit directly via the endpoint's physical interfaces (not
through the proxy). In a leaky configuration, setting a proxy augments
the available set of ICE candidates. Multiple leaky-configuration
proxies may therefore be active simultaneously.</t>
<t>A sealed proxy configuration requires the browser to route all WebRTC
traffic through the proxy, eliminating all ICE candidates that do not go
through the proxy. Only one sealed proxy may be active at a time.</t>
</section>
<section title="Sealed proxy rank">
<t>In some configurations, an endpoint may be subject to multiple sealed
proxy settings at the same time. In that case, one of those settings
will have highest rank, and it will be the active proxy. In a given
application context (e.g. a webpage), there is at most one active sealed
proxy. This document does not specify a representation for rank.</t>
</section>
</section>
<section title="Diagrams">
<figure anchor="basic_figure"
title="Basic WebRTC ICE candidates (no proxy)">
<preamble>This figure shows the connections that provide the ICE
candidates for WebRTC in the basic configuration (no proxy). This
figure is provided in order to serve as a baseline against which to
compare the candidate routes that make use of a proxy.</preamble>
<artwork><![CDATA[
+-------------+ * *
|UDP generator| * * +----+
| host+----+--O-----O.....+STUN|
|relay srflx| | * * +----+
+--+-------+--+ | * *
| | | * LAN *
| \-------/ * *
| * * *
| +------+ * * +------+ *
\------+ TURN +==============+ TURN +-----O
|client| * * |server| *
+------+ * * +------+ *
.. STUN packets *** Network interface
-- Bare UDP content link *O* Candidate port
== TURN encapsulated link
]]></artwork>
</figure>
<figure anchor="proxy_figure"
title="WebRTC ICE candidates using a proxy">
<preamble>This figure shows the connections that provide the ICE
candidates for WebRTC on the virtual interface that represents a
proxy.</preamble>
<artwork><![CDATA[
+-------------+ * +------+ * +----+
|UDP generator| * |Proxy | * .+STUN|
| host+-------+ TURN | * * . +----+
|relay srflx| * |Client| * * .
+--+-------+--+ * | | * +------+ * . +------+ *
| | * | | * |Proxy | *. | App | *
| \----------+ +######+ TURN +????O=====+TURN +-----O
| * | | * |Server| * |Server| *
| +------+ * | | * +------+ * +------+ *
| | App | * | | * *
\------+ TURN +====+ | * *
|client| * | | *
+------+ * +------+ *
.. STUN packets *** Network interface
-- Bare UDP content link *O* Candidate port
== TURN encapsulated UDP content link
## RETURN double-encapsulated link
?? Mixed content link
]]></artwork>
</figure>
</section>
<section title="Requirements">
<section title="ICE candidates produced in the presence of a proxy">
<t>When a proxy is configured, by Auto-Discovery or a proprietary means,
the browser MUST NOT report a “relay” candidate representing
the proxy. Instead, for each active proxy, the browser MUST connect to
the proxy and then, if the connection is successful, treat the TURN
tunnel as a UDP-only virtual interface.</t>
<t>For a virtual interface representing a TURN proxy, this means that
the browser MUST report the public-facing IP address and port acquired
through TURN as a “host” candidate, the browser MUST perform
STUN through the TURN proxy (if STUN is configured), and it MUST perform
TURN by recursive encapsulation through the TURN proxy, resulting in
TURN candidates whose “raddr” and “rport”
attributes match the acquired public-facing IP address and port on the
proxy.</t>
<t>Because the virtual interface has some additional overhead due to
indirection, it SHOULD have lower priority than the physical interfaces
if physical interfaces are also active. Specifically, even host
candidates generated by a virtual interface SHOULD have priority 0 when
physical interfaces are active (similar to
<xref target="RFC5245"></xref> Section 4.1.2.2, “the local
preference for host candidates from a VPN interface SHOULD have a
priority of 0”).</t>
</section>
<section title="Leaky proxy configuration">
<t>If the active proxy for an application is leaky, the browser should
undertake the standard ICE candidate discovery mechanism
<xref target="RFC5245"></xref> on the available physical and virtual
interfaces.</t>
</section>
<section title="Sealed proxy configuration">
<t>If the active proxy for an application is sealed, the browser MUST
NOT gather or produce any candidates on physical interfaces. The WebRTC
implementation MUST direct its traffic from those interfaces only to the
proxy, and perform ICE candidate discovery only on the single virtual
interface representing the active proxy.</t>
</section>
<section title="Proxy rank">
<t>Any browser mechanism for specifying a proxy SHOULD allow the caller
to indicate a higher rank than the proxy provided by Auto-Discovery
<xref target="I-D.ietf-tram-turn-server-discovery"></xref>.</t>
</section>
<section title="Multiple physical interfaces">
<t>Some operating systems allow the browser to use multiple interfaces
to contact a single remote IP address. To avoid producing an excessive
number of candidates, WebRTC endpoints MUST NOT use multiple physical
interfaces to connect to a single proxy simultaneously. (If this were
violated, it could produce a number of virtual interfaces equal to the
product of the number of physical interfaces and the number of active
proxies.)</t>
<t>For strategies to choose the best interface for communication with a
proxy, see
<xref target="I-D.reddy-mmusic-ice-best-interface-pcp"></xref>. Similar
considerations apply when connecting to an application-specified TURN
server in the presence of physical and virtual interfaces.</t>
</section>
<section title="Unspecified leakiness">
<t>If a proxy configuration mechanism does not specify leakiness,
browsers SHOULD treat the proxy as leaky. This is similar to current
WebRTC implementations' behavior in the presence of SOCKS and HTTP
proxies: the candidate allocation code continues to generate UDP
candidates that do not transit through the proxy.</t>
</section>
<section title="Interaction with SOCKS5-UDP">
<t>The <xref target="RFC1928">SOCKS5 proxy standard</xref> permits
compliant SOCKS proxies to support UDP traffic. However, most
implementations of SOCKS5 today do not support UDP. Accordingly, WebRTC
browsers MUST by default (i.e. unless deliberately configured otherwise)
treat SOCKS5 proxies as leaky and having lower rank than any configured
TURN proxies.</t>
</section>
<section title="Encapsulation overhead, fragmentation, and Path MTU">
<t>Encapsulating a link in TURN adds overhead on the path between the
client and the TURN server, because each packet must be wrapped in a
TURN message. This overhead is sometimes doubled in RETURN proxying.
To avoid excessive overhead, client implementations SHOULD use
ChannelBind and ChannelData messages to connect and send data through
proxies and application TURN servers when possible. Clients MAY buffer
messages to be sent until the ChannelBind command completes (requiring
one round trip to the proxy), or they MAY use CreatePermission and Send
messages for the first few packets to reduce startup latency at the cost
of higher overhead.</t>
<t>Adding overhead to packets on a link decreases the effective
Maximum Transmissible Unit on that link. Accordingly, clients that
support proxying MUST NOT rely on the effective MTU complying with the
Internet Protocol's minimum MTU requirement.</t>
<t>ChannelData messages have constant overheard, enabling consistent
effective PMTU, but Send messages do not necessarily have constant
overhead. TURN messages may be fragmented and reassembled if they are not
marked with the Don't Fragment (DF) IP bit or the DONT-FRAGMENT TURN
attribute. Client implementors should keep this in mind, especially if
they choose to implement PMTU discovery through the proxy.</t>
</section>
<section title="Interaction with alternate TURN server fallback">
<t>As per <xref target="RFC5766"></xref>, a TURN server MAY respond to
an Allocate request with an error code of 300 and an ALTERNATE-SERVER
indication. When connecting to proxies or application TURN servers,
clients SHOULD attempt to connect to the specified alternate server in
accordance with <xref target="RFC5766"></xref>. The client MUST route
a connection to the alternate server through the proxy if and only if
the original connection attempt was routed through the proxy.</t>
</section>
</section>
<section title="Examples">
<section title="Firewalled enterprise network with a basic application">
<t>In this example, an enterprise network is configured with a firewall
that blocks all UDP traffic, and a TURN server is advertised for
Auto-Discovery in accordance with
<xref target="I-D.ietf-tram-turn-server-discovery"></xref>.
The proxy leakiness of the TURN server is unspecified, so the browser
treats it as leaky.</t>
<t>The application specifies a STUN and TURN server on the public net.
In accordance with the ICE candidate gathering algorithm
<xref target="RFC5245">RFC 5245</xref>, it receives a set of candidates
like:
<list style="numbers">
<t>A host candidate acquired from one interface.
<list style="symbols">
<t>e.g. candidate:1610808681 1 udp 2122194687 [internal ip addr
for interface 0] 63555 typ host generation 0</t>
</list></t>
<t>A host candidate acquired from a different interface.
<list style="symbols">
<t>e.g. candidate:1610808681 1 udp 2122194687 [internal ip addr
for interface 1] 54253 typ host generation 0</t>
</list></t>
<t>The proxy, as a host candidate.
<list style="symbols">
<t>e.g. candidate:3458234523 1 udp 24584191 [public ip addr for
the proxy] 54606 typ host generation 0</t>
</list></t>
<t>The virtual interface also generates a STUN candidate, but it is
eliminated because it is redundant with the host candidate, as noted
in <xref target="RFC5245"></xref> Sec 4.1.2..</t>
<t>The application-provided TURN server as seen through the virtual
interface. (Traffic through this candidate is recursively
encapsulated.)
<list style="symbols">
<t>e.g. candidate:702786350 1 udp 24583935 [public ip addr of the
application TURN server] 52631 typ relay raddr [public ip addr for
the proxy] rport 54606 generation 0</t>
</list></t>
</list>
There are no STUN or TURN candidates on the physical interfaces, because
the application-specified STUN and TURN servers are not reachable
through the firewall.</t>
<t>If the remote peer is within the same network, it may be possible to
establish a direct connection using both peers' host candidates. If the
network prevents this kind of direct connection, the path will instead
take a “hairpin” route through the enterprise's proxy, using
one peer's physical “host” candidate and the other's virtual
“host” candidate, or (if that is also disallowed by the
network configuration) a “double hairpin” using both
endpoints' virtual “host” candidates.</t>
</section>
<section title="Conflicting proxies configured by Auto-Discovery and local
policy">
<t>Consider an enterprise network with TURN and HTTP proxies advertised
for Auto-Discovery with unspecified leakiness (thus defaulting to
leaky). The browser endpoint configures an additional TURN proxy by a
proprietary local mechanism.</t>
<t>If the locally configured proxy is leaky, then the browser MUST
produce candidates representing any physical interfaces (including
SSLTCP routes through the HTTP proxy), plus candidates for both UDP-only
virtual interfaces created by the two TURN servers.</t>
<t>There MUST NOT be any candidate that uses both proxies. Multiple
configured proxies are not chained recursively.</t>
<t>If the locally configured proxy is “sealed”, then the
browser MUST produce only candidates from the virtual interface
associated with that proxy.</t>
<t>If both proxies are configured for “sealed” use, then the
browser MUST produce only candidates from the virtual interface
associated with the proxy with higher rank.</t>
</section>
</section>
<section title="Security Considerations">
<t>This document describes web browser behaviors that, if implemented
correctly, allow users to achieve greater identity-confidentiality during
WebRTC calls under certain configurations.</t>
<t>If a site administrator offers the site's users a TURN proxy, websites
running in the users' browsers will be able to initiate a UDP-based WebRTC
connection to any UDP transport address via the proxy. Websites'
connections will quickly terminate if the remote endpoint does not reply
with a positive indication of ICE consent, but no such restriction applies
to other applications that access the TURN server. Administrators should
take care to provide TURN access credentials only to the users who are
authorized to have global UDP network access.</t>
<t>TURN proxies and application TURN servers can provide some privacy
protection by obscuring the identity of one peer from the other.
However, unencrypted TURN provides no additional privacy from an observer
who can monitor the link between the TURN client and server, and even
encrypted TURN (<xref target="I-D.ietf-tram-stun-dtls"></xref> Section
4.6) does not provide significant privacy from an observer who sniff
traffic on both legs of the TURN connection, due to packet timing
correlations.</t>
</section>
<section title="IANA Considerations">
<t>This document requires no actions from IANA.</t>
</section>
<section title="Acknowledgements">
<t>Significant review, including the virtual-interface formulation, was
provided by Justin Uberti. Thanks to Harald Alvestrand, Phillip
Hancke, and Tirumaleswar Reddy for suggestions to improve the content and
presentation.</t>
</section>
</middle>
<back>
<references title="Normative References">
<reference anchor="RFC5245">
<front>
<title>Interactive Connectivity Establishment (ICE): A Protocol for
Network Address Translator (NAT) Traversal for Offer/Answer
Protocols</title>
<author fullname="J. Rosenberg" initials="J." surname="Rosenberg">
<organization>jdrosen.net</organization>
</author>
<date month="April" year="2010" />
</front>
<seriesInfo name="RFC" value="5245" />
<format target="http://www.rfc-editor.org/rfc/rfc5245.txt" type="TXT" />
</reference>
<reference anchor="RFC5766">
<front>
<title>Traversal Using Relays around NAT (TURN): Relay Extensions to
Session Traversal Utilities for NAT (STUN)</title>
<author fullname="Rohan Mahy" initials="R." surname="Mahy">
<organization></organization>
</author>
<author fullname="Philip Matthews" initials="P." surname="Matthews">
<organization>Alcatel-Lucent</organization>
</author>
<author fullname="J. Rosenberg" initials="J." surname="Rosenberg">
<organization>jdrosen.net</organization>
</author>
<date month="April" year="2010" />
</front>
<seriesInfo name="RFC" value="5766" />
<format target="http://www.rfc-editor.org/rfc/rfc5245.txt" type="TXT" />
</reference>
<reference anchor="RFC1928">
<front>
<title>SOCKS Protocol Version 5</title>
<author fullname="Marcus Leech" initials="M." surname="Leech">
<organization>Bell-Northern Research Ltd</organization>
</author>
<author initials="M." surname="Ganis">
<organization>International Business Machines</organization>
</author>
<author initials="Y." surname="Lee">
<organization>NEC Systems Laboratory</organization>
</author>
<author initials="R." surname="Kuris">
<organization>Unify Corporation</organization>
</author>
<author initials="D." surname="Koblas">
<organization></organization>
</author>
<author initials="L." surname="Jones">
<organization>Hewlett-Packard Company</organization>
</author>
<date month="March" year="1996" />
</front>
<seriesInfo name="RFC" value="5766" />
<format octets="285120"
target="http://www.rfc-editor.org/rfc/rfc1928.txt" type="TXT" />
</reference>
<reference anchor="I-D.ietf-rtcweb-jsep">
<front>
<title>Javascript Session Establishment Protocol</title>
<author fullname="Justin Uberti" initials="J" surname="Uberti">
<organization></organization>
</author>
<author fullname="Cullen Jennings" initials="C" surname="Jennings">
<organization></organization>
</author>
<date day="13" month="February" year="2014" />
<abstract>
<t>This document describes the mechanisms for allowing a Javascript
application to control the signaling plane of a multimedia session
via the interface specified in the W3C RTCPeerConnection API, and
discusses how this relates to existing signaling protocols.</t>
</abstract>
</front>
<seriesInfo name="Internet-Draft" value="draft-ietf-rtcweb-jsep-06" />
<format target="http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-06.txt"
type="TXT" />
</reference>
</references>
<references title="Informative References">
<reference anchor="I-D.ietf-tram-turn-server-discovery">
<front>
<title>TURN Server Auto Discovery</title>
<author fullname="Prashanth Patil" initials="P" surname="Patil">
<organization>Cisco Systems, Inc.</organization>
</author>
<author fullname="Tirumaleswar Reddy" initials="T" surname="Reddy">
<organization>Cisco Systems, Inc.</organization>
</author>
<author fullname="Dan Wing" initials="D" surname="Wing">
<organization>Cisco Systems, Inc.</organization>
</author>
<date day="24" month="July" year="2014" />
<abstract>
<t>Current Traversal Using Relays around NAT (TURN) server discovery
mechanisms are relatively static and limited to explicit
configuration. These are usually under the administrative control
of the application or TURN service provider, and not the enterprise
or the ISP, the network in which the client is located. Enterprises
and ISPs wishing to provide their own TURN servers need auto
discovery mechanisms that a TURN client could use with no or minimal
configuration. This document describes two such mechanisms for TURN
server discovery.</t>
</abstract>
</front>
<seriesInfo name="Internet-Draft"
value="draft-ietf-tram-turn-server-discovery-00" />
<format target="http://www.ietf.org/internet-drafts/draft-ietf-tram-turn-server-discovery-00.txt"
type="TXT" />
</reference>
<reference anchor="I-D.reddy-mmusic-ice-best-interface-pcp">
<front>
<title>Improving ICE Interface Selection Using Port Control Protocol
(PCP) Flow Extension</title>
<author fullname="Tirumaleswar Reddy" initials="T" surname="Reddy">
<organization>Cisco Systems, Inc.</organization>
</author>
<author fullname="Dan Wing" initials="D" surname="Wing">
<organization>Cisco Systems, Inc.</organization>
</author>
<author fullname="Bill VerSteeg" initials="B" surname="VerSteeg">
<organization>Cisco Systems, Inc.</organization>
</author>
<author fullname="Reinaldo Penno" initials="R" surname="Penno">
<organization>Cisco Systems, Inc.</organization>
</author>
<author fullname="Varun Singh" initials="V" surname="Singh">
<organization>Aalto University</organization>
</author>
<date day="10" month="October" year="2013" />
<abstract>
<t>A host with multiple interfaces needs to choose the best
interface for communication. Oftentimes, this decision is based on
a static configuration and does not consider the link
characteristics of that interface, which may affect the user
experience.</t>
<t>This document describes a mechanism for an endpoint to query the
link characteristics from the access router (the router at the other
end of the endpoint's access link) using a Port Control Protocol
(PCP) Flow Extension. This information influences endpoint's
Interactive Connectivity Establishment (ICE) candidate selection
algorithm.</t>
</abstract>
</front>
<seriesInfo name="Internet-Draft"
value="draft-ietf-tram-turn-server-discovery-00" />
<format target="http://www.ietf.org/internet-drafts/draft-ietf-tram-turn-server-discovery-00.txt"
type="TXT" />
</reference>
<reference anchor="I-D.ietf-rtcweb-use-cases-and-requirements">
<front>
<title>Web Real-Time Communication Use-cases and Requirements</title>
<author fullname="Christer Holmberg" initials="C" surname="Holmberg">
<organization>Ericsson</organization>
</author>
<author fullname="Stefan Hakansson" initials="S" surname="Hakansson">
<organization>Ericsson</organization>
</author>
<author fullname="Goran AP Eriksson" initials="G" surname="Eriksson">
<organization>Ericsson</organization>
</author>
<date day="12" month="February" year="2014" />
<abstract>
<t>This document describes web based real-time communication
use-cases. Requirements on the browser functionality are derived
from the use-cases.</t>
</abstract>
</front>
<seriesInfo name="Internet-Draft"
value="ietf-rtcweb-use-cases-and-requirements-14" />
<format target="http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirements-14.txt"
type="TXT" />
</reference>
<reference anchor="I-D.ietf-tram-stun-dtls">
<front>
<title>Datagram Transport Layer Security (DTLS) as Transport for
Session Traversal Utilities for NAT (STUN)</title>
<author fullname="Marc Petit-Huguenin" initials="M"
surname="Petit-Huguenin">
<organization>Jive Communications</organization>
</author>
<author fullname="Gonzalo Salgueiro" initials="G" surname="Salgueiro">
<organization>Cisco Systems</organization>
</author>
<date day="27" month="June" year="2014" />
<abstract>
<t>This document specifies the usage of Datagram Transport Layer
Security (DTLS) as a transport protocol for Session Traversal
Utilities for NAT (STUN). It provides guidances on when and how to
use DTLS with the currently standardized STUN Usages. It also
specifies modifications to the STUN URIs and TURN URIs and to the
TURN resolution mechanism to facilitate the resolution of STUN URIs
and TURN URIs into the IP address and port of STUN and TURN servers
supporting DTLS as a transport protocol. This document updates RFC
5389 and RFC 5928.</t>
</abstract>
</front>
<seriesInfo name="Internet-Draft"
value="ietf-rtcweb-use-cases-and-requirements-14" />
<format target="http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirements-14.txt"
type="TXT" />
</reference>
<reference anchor="I-D.ietf-rtcweb-security">
<front>
<title>Security Considerations for WebRTC</title>
<author fullname="Eric Rescorla" initials="E" surname="Rescorla">
<organization>RTFM, Inc.</organization>
</author>
<date day="4" month="July" year="2014" />
<abstract>
<t>The Real-Time Communications on the Web (RTCWEB) working group is
tasked with standardizing protocols for real-time communications
between Web browsers, generally called "WebRTC". The major use cases
for WebRTC technology are real-time audio and/or video calls, Web
conferencing, and direct data transfer. Unlike most conventional
real-time systems (e.g., SIP-based soft phones) WebRTC communications
are directly controlled by a Web server, which poses new security
challenges. For instance, a Web browser might expose a JavaScript
API which allows a server to place a video call. Unrestricted access
to such an API would allow any site which a user visited to "bug" a
user's computer, capturing any activity which passed in front of
their camera. This document defines the WebRTC threat model and
analyzes the security threats of WebRTC in that model.</t>
</abstract>
</front>
<seriesInfo name="Internet-Draft"
value="ietf-rtcweb-security-07" />
<format target="http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-security-07.txt"
type="TXT" />
</reference>
</references>
</back>
</rfc>
| PAFTECH AB 2003-2026 | 2026-04-24 10:19:42 |