One document matched: draft-hutton-rtcweb-nat-firewall-considerations-02.xml
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<rfc category="info" docName="draft-hutton-rtcweb-nat-firewall-considerations-02" ipr="trust200902">
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<!-- ***** FRONT MATTER ***** -->
<front>
<title abbrev="RTCWEB NAT-FW">RTCWEB Considerations for NATs, Firewalls and HTTP proxies </title>
<author fullname="Thomas Stach" initials="T." role="editor"
surname="Stach">
<organization>Siemens Enterprise Communications</organization>
<address>
<postal>
<street>Dietrichgasse 27-29</street>
<city>Vienna</city>
<region></region>
<code>1030</code>
<country>AT</country>
</postal>
<email>thomas.stach@siemens-enterprise.com</email>
</address>
</author>
<author fullname="Andrew Hutton" initials="A."
surname="Hutton">
<organization>Siemens Enterprise Communications</organization>
<address>
<postal>
<street>Technology Drive</street>
<city>Nottingham</city>
<region></region>
<code>NG9 1LA</code>
<country>UK</country>
</postal>
<email>andrew.hutton@siemens-enterprise.com</email>
</address>
</author>
<author fullname="Justin Uberti" initials="J."
surname="Uberti">
<organization>Google</organization>
<address>
<postal>
<street>747 6th Ave S</street>
<city>Kirkland</city>
<region>WA</region>
<code>98033</code>
<country>US</country>
</postal>
<email>justin@uberti.name</email>
</address>
</author>
<date year="2013" />
<!-- Meta-data Declarations -->
<area>General</area>
<workgroup>Internet Engineering Task Force</workgroup>
<keyword>RTCWEB, NAT, Firewall, HTTP proxy</keyword>
<abstract>
<t>
This document describes mechanism to enable media stream establishment
for Real-Time Communication in WEB-browsers (WebRTC) in the presence
of network address translators,
firewalls and HTTP proxies.
HTTP proxy and firewall deployed in many private network domains introduce obstacles to
the successful establishment of media stream via WebRTC.
This document examines some of these deployment scenarios and
develops requirements on the web browsers designed to provide the best possible chance of
media connectivity between WebRTC peers.
</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>
WebRTC is a web-based technique for direct interactive rich communication using
audio, video, and data between two peer browsers.
</t>
<t>
Many organizations, e.g. an enterprise, a public service agency or a university, deploy
Network Address Translators (NAT) and firewalls (FW) at the border to the public internet.
WebRTC relies on ICE <xref target="RFC5245"></xref> in order to establish
a media path between two WebRTC peers in the presence of such NATs/FWs.
</t>
<t>
When WebRTC is deployed by the corporate IT department one can assume that
the corporate IT configures the corporate NATs, Firewalls, DPI units, TURN servers
accordingly.
If so desired by the organization WebRTC media streams can then be established to WebRTC peers
outside of the organization subject to the applied policies.
In order to cater for NAT/FWs with address and port dependent mapping characteristics
<xref target="RFC4787"></xref>,
the peers will introduce a TURN server
<xref target="RFC5766"></xref>
in the public internet as a media relay.
Such a TURN server could be deployed by the organization wanting to assert policy on WebRTC traffic.
</t>
<t>
However, there are also environments that are not prepared for WebRTC and have NAT/FW deployed
that prevent the media stream establishment although such blocking is not intentional.
These environments include e.g. internet cafes or hotels offering their customers access
to the web and have opened the well-known HTTP(S) ports but nothing else.
In such an environment ICE will fail to establish connectivity.
Re-configuration of the NAT/FW is also often impracticable or not possible.
</t>
<t>
In such an environment a WebRTC user may easily reach its WebRTC server possibly
via an HTTP proxy and start establishing a WEBTRTC session,
but will become frustrated when a media connection cannot be established.
A corresponding use case and its requirements relating to WebRTC NAT/FW traversal can be found in
<xref target="draft-ietf-rtcweb-use-cases-and-requirements"></xref>.
</t>
<t>
The TURN server in the
public internet is not sufficient to establish connectivity for RTP-based media
<xref target="RFC3550"></xref>
and the WebRTC data channel
<xref target="draft-ietf-rtcweb-data-channel"></xref>
towards external WebRTC peers since
the FW policies include blocking of all UDP based traffic and allowing only traffic
to the TCP ports 80/443 with the intent to support HTTP(S) <xref target="RFC2616"/>.
</t>
<t>
We explicitly don't address even more restricted environments, that deploy HTTP traffic validation.
This could e.g. be done by means of
DPI validation or traffic pattern analysis to determine the contents of the packets that
the traffic is, in fact, HTTP or HTTPS-looking or by
an HTTP proxy that breaks into the TLS exchange and looks for HTTP in the traffic.
However we want to address the case when access to the World Wide Web from inside an organization
is only possible via a transparent HTTP Proxy
that just tunnels traffic after e.g. enforcing an acceptable use policy.
</t>
<t>
This document examines impact of NAT/FW policies in
<xref target="pure_FW"></xref>.
Additional impacts due to the presence of a HTTP proxy are examined in
<xref target="proxy_FW"></xref>.
</t>
<section title="Requirements Language">
<t>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 <xref
target="RFC2119">RFC 2119</xref>.</t>
</section>
</section>
<section anchor="pure_FW" title="Considerations for NATs/Firewalls independent of HTTP proxies">
<t>
This section covers aspects of how NAT/FW characteristic influence the establishment of a media stream.
Additional aspects introduced by the presence of a HTTP proxy are covered in <xref target="proxy_FW"/>.
</t>
<t>
If the NATs serving caller and callee both show port and address dependent mapping
behavior the need for a TURN server arises in order to establish connectivity for media streams.
The TURN server will relay the RTP packet to the WebRTC peer using UDP.
How the RTP packets can be transported from the WebRTC client within the private network
to the TURN server depends on what the firewall will let pass through.
</t>
<t>
Other types of NATs do not require using the TURN relay.
Nevertheless, the FW rules and policies still affect how media streams can be established.
</t>
<section anchor="pure_FW_tcp_udp" title="NAT/Firewall open for outgoing UDP and TCP traffic">
<t>
This scenario assumes that the NAT/FW is transparent for all outgoing
traffic independent of using UDP or TCP as transport protocol.
This case is used as starting point for introduction of more restrictive
firewall policies.
It presents the least critical example with respect to the
establishment of the media streams.
</t>
<t>
The TURN server can be reached directly from within the private network
via the NAT/FW and
the ICE procedures will reveal that media can be sent via the TURN server.
The TURN client will send its media to the allocated resources at the TURN server via UDP.
</t>
<t>
Dependent on the port range that is used for WebRTC media streams,
the same statement would be true if the NAT/Firewall would allow UDP
traffic for a restricted UDP port range only.
</t>
</section>
<section anchor="pure_FW_tcp" title="NAT/Firewall open only for TCP traffic">
<t>
This scenario assumes that the NAT/FW is transparent for outgoing
traffic only using TCP as transport protocol. Theoretically, this gives two options
for media stream establishment dependent on the NAT's mapping
characteristics. Either transporting RTP over TCP directly to the peer or contacting a
TURN server via TCP that then relays RTP.
</t>
<t>
In the first case the browser does not use any TURN server to get through its NAT/FW.
However, the browser needs to use ICE-TCP <xref target="RFC6544"/> and
provide active, passive and/or simultaneous-open TCP candidates.
Assuming the peer also provides TCP candidates, a connectivity check
for a TCP connection between the two peers should be successful.
</t>
<t>
In the second case the browser contacts the TURN server via TCP for allocation of
an UDP-based relay address at the TURN server.
The ICE procedures will reveal that RTP media can be sent via the TURN relay using
the TCP connection between TURN client and TURN server.
The TURN server would then relay the RTP packets using UDP, as well as other UDP-based protocols.
ICE-TCP is not needed in this context.
</t>
<t>
Note that the second case is not to be confused with TURN/TCP <xref target="RFC6062"/>,
which deals with how to establish a TCP connection from a TURN server to the peer.
For this document we assume that the TURN server can reach the peer always via UDP,
possibly via a second TURN server,
in case the WebRTC peer is located in a similar environment as described in this section.
</t>
<t>
We don't see a need to support TURN/TCP since all WebRTC media is transported over UDP.
For the same reason we also prefer using TCP just as transport to the TURN server
over using the ICE-TCP with an end-to-end TCP connection </t>
</section>
<section anchor="pure_FW_http" title="NAT/Firewall open only for TCP on restricted ports">
<t>
In this case the firewall blocks all outgoing traffic except for TCP traffic to
specific ports, for example port 80 (HTTP) for HTTP or 443 for HTTPS(HTTPS).
A TURN server listening to its default ports (3478 for TCP/UDP, 5349 for TLS)
would not be reachable in this case.
However, the TURN server can still be reached when it is configured to
listen to e.g. the HTTP(S) ports.
</t>
<t>
<list><t>
Open issue: Although <xref target="draft-ietf-rtcweb-use-cases-and-requirements"></xref>
considers only a restriction to HTTP(S) similar consideration apply
to other ports or port ranges. A change to req F37 to
"The browser must be able to send streams and data to a peer in the presence of FWs
that only allows traffic via a HTTP Proxy."
has been agreed and will be in the next update does this solve the issue.
</t></list>
</t>
<t>
In addition the browser needs to be configured to contact
the TURN server over the HTTP(S) ports and/or the WebRTC client has to tell the browser
accordingly.
</t>
</section>
</section>
<section anchor="proxy_FW" title="Considerations for NATs/Firewalls in presence of HTTP proxies">
<t>
This section considers a scenario where all HTTP(S) traffic is routed via an HTTP proxy.
We assume that the HTTP proxy is tranparent and just tunnels traffic after e.g. enforcing an acceptable
use policy with respect to domains that are allowed to be reached.
We don't consider cases where the HTTP proxy is used to deploy HTTP traffic validation.
This includes DPI validation that the traffic is, in fact, HTTP or HTTPS-looking or
a HTTP proxy that breaks into the TLS exchange and looks for HTTP in the traffic.
</t>
<t>
Note: If both WebRTC clients are located behind the same HTTP proxy,
we, of course, assume that ICE would give us a direct media connection within the private network.
We don't consider this case in detail within this document.
</t>
<section anchor="proxy_FW_tcp_udp" title="HTTP proxy with NAT/Firewall open for
outgoing UDP and TCP traffic">
<t>
As in
<xref target="pure_FW_tcp_udp"/>
we assume that the NAT/FW is transparent for all outgoing traffic
independent of using UDP or TCP as transport protocol.
The HTTP proxy has no impact on the transport of media streams in this case.
Consequently, the same considerations as in
<xref target="pure_FW_tcp_udp"/>
apply with respect to the traversal of the NAT/FW.
</t>
</section>
<section anchor="proxy_FW_tcp" title="HTTP proxy with NAT/Firewall open only for TCP traffic">
<t>
As in
<xref target="pure_FW_tcp"/>
we assume that the NAT/FW is transparent only for outgoing TCP traffic.
The HTTP proxy has no impact on the transport of media streams in this case.
Consequently, the same considerations as in
<xref target="pure_FW_tcp"/>
apply with respect to the traversal of the NAT/FW.
</t>
</section>
<section anchor="proxy_relay" title="HTTP proxy assisted TURN server connection">
<section anchor="proxy_relay_turn" title="TURN server connection via TCP">
<t>
Different from the previous scenarios, we assume that the NAT/FW accepts outgoing traffic
only via a TCP connection that is initiated from the HTTP proxy.
Consequently, a browser would have to use the HTTP CONNECT method
<xref target="RFC2817"/>
and request that the HTTP proxy establishes a tunnel connection
on its behalf in order to get access to the TURN server.
The HTTP CONNECT request needs to convey the TURN Server URI or transport address.
As a result the HTTP Proxy will establish a TCP connection to the TURN server and
when successful the HTTP Proxy will answer the HTTP CONNECT request with a 200OK response.
In case of a transparent proxy, the HTTP proxy will now switch into tunneling mode
and will transparently relay the traffic to the TURN server.
</t>
<t>
We want to emphasize that by using the HTTP CONNECT method,
the TURN server only has to handle a standard TCP connection.
An update to the TURN protocol or the TURN software is not needed.
<list><t>
Open Issue: This assumes that the HTTP Proxy can handle establishing the TCP connection
to the STUN ports 3478/5349.
The syntax of the Request-URI certainly allows this, but is this also supported
in the major HTTP proxy implementations?
Section 4.3.6 of <xref target="draft-ietf-httpbis-p2-semantics"></xref> describes
tunneling to ports other than 80/443 and the corresponding security impacts.
</t>
</list>
</t>
<t>
Afterwards, the browser could upgrade the connection to use TLS,
forward STUN/TURN traffic via the HTTP proxy and use the TURN server as media relay.
Note that upgrading in this case is not to be misunderstood as usage of the HTTP UPGRADE method
as specified in <xref target="RFC2817"/>
as this would require the TURN server to support HTTP.
We rather envisage the following sequence:
</t>
<t><list style="symbols">
<t>the browser opens a TCP connection to the HTTP proxy,
</t>
<t>the browser issues a HTTP CONNECT request to the HTTP proxy with
the TURN server address in the Request URI, for example
<list><t>
CONNECT turn_server.example.com:5349 HTTP/1.1
Host: turn_server.example.com:5349
</t></list>
</t>
<t>the HTTP proxy opens a TCP connection to the TURN server and "bridges"
the incoming and outgoing TCP connections together, forming a virtual
end-to-end TCP connection,
</t>
<t>the browser can do a TLS handshake over the virtual end-to-end TCP connection
with the TURN server.
</t>
</list> </t>
<t>
If it is not possible to use HTTP CONNECT in this way it will not be possible
to establish connectivity between the WebRTC peers and
the ICE connectivity checks will fail.
</t>
<t>
Strictly speaking the TLS upgrade is not necessary, but using TLS would also prevent
the HTTP proxy from sniffing into the data stream and provides the same flow as HTTPS
and might improve interoperability with proxy servers.
Some tests (done a while ago) indicated that there are proxies performing
Deep Packet inspection (DPI) that expect to see
at least a TLS handshake and, possibly, valid TLS records.
The application has the ability to control whether TLS is used by the parameters it
supplies to the TURN URI (e.g. turns: vs. turn:), so the decision
to access the TURN server via TCP versus TLS could be left up to the application
or possibly the browser configuration script.
</t>
<t>
In contrast to using UDP or TCP for transporting the STUN messages,
the browser would now need
to first establish a HTTP over TCP connection to the HTTP proxy,
upgrade to using TLS and then switch to
using this TLS connection for transport of STUN messages.
It is also desirable that the browser detects the need to connect to the TURN server
through a HTTP proxy automatically in order to achieve seamless deployment and interoperability.
The browser should use the same proxy selection procedure for TURN as currently done for HTTP.
The user or network administrator should not be required to change browser or
proxy script configuration.
</t>
<t>
Further considerations apply to the default connection timeout of the HTTP proxy connection
to the TURN server and the timeout of the TURN server allocation.
Whereas
<xref target="RFC5766"></xref> specifies a 10 minutes default lifetime
of the TURN allocation, typical proxy connection lifetimes are in the range
of 60 seconds if no activity is detected.
Thus, if the WebRTC client wants to pre-allocate TURN ressources
it needs to refresh TURN allocations more frequently in order to keep the TCP connection to its TURN server alive.
</t>
</section>
<section anchor="proxy_relay_tcp" title="TURN server connection via UDP">
<t>
If a local TURN server under administrative control of the organization is deployed
it is desirable to reach this TURN server via UDP.
The TURN server could be specified in the proxy configuration script,
giving the browser the possibility to learn how to access it.
Then, when gathering candidates, this TURN server would always be used such that the
WebRTC client application could get UDP traffic out to the internet.
</t>
</section>
</section>
</section>
<section anchor="Alternatives" title="Other Approaches">
<section anchor="proxy_relay_ws" title="TURN server connection via WebSocket">
<t>
The WebRTC client could connect to a TURN server via WebSocket
<xref target="RFC6455"></xref>
as described in
<xref target="draft-chenxin-behave-turn-WebSocket"></xref>.
This might have benefits in very restrictive environments where HTTPS is not permitted through
the proxy.
However, such environments are also likely to deploy DPI boxes which would eventually
complain against usage of WebSocket or block WebRTC traffic based on other heuristic means.
It is also to be expected that an environment that does not allow HTTPS will also forbid
usage of WebSocket over TLS.
</t>
<t>
In addition, usage of TURN over WebSocket puts an additional burden on existing
TURN server implementation to support HTTP and WebSocket.
The resulting benefit seems rather small, thus TURN over WebSocket is left for further study.
</t>
</section>
<section anchor="PCP" title="Port Control Protocol">
<t>
As a further alternative,
the Port Control Protocol (PCP)
<xref target="RFC6887"></xref>
allows to configure how incoming IPv6 or IPv4 packets are translated and forwarded by a NAT/FW.
However, this document does not examine benefits of PCP for
the management of the local NAT/FW,
but leaves this for further study until PCP is deployed more widely.
</t>
</section>
<section anchor="RTPoverHTTP" title="HTTP Fallback for RTP Media Streams">
<t>
As an alternative to using a TURN server it was proposed to send RTP directly over HTTP
<xref target="draft-miniero-rtcweb-http-fallback"></xref>.
This approach bears some similarities with TURN as it also uses a RTP relay.
However, it uses HTTP GET and POST requests to receive and send RTP packets.
</t>
<t>
Despite a number of open issues, the proposal addreses some corner cases.
However, the expected benefit in form of an increased success rate
for establishment of a media stream seems rather small,
thus HTTP fallback is left for further study.
</t>
</section>
</section>
<section anchor="Requirements" title="Requirements for RTCWEB-enabled browsers">
<t>
For the purpose of relaying WebRTC media streams or data channels a browser needs to be able to
</t>
<t><list style="symbols">
<t>
connect to a TURN server via UDP, TCP and TLS,
</t>
<t>
support the HTTP CONNECT method and request that a proxy establish a tunneled TCP connection
to a TURN server on its behalf,
</t>
<t>
connect to a TURN server through this HTTP proxy-tunnelled TCP connection,
</t>
<t>
connect to the TURN server via application specified ports other than the default STUN ports
including the HTTP(s) ports,
<list><t>
Open issue: Do we want to allow the TURN server to relay data received via the HTTP(S) ports?
Is this a change to the TURN protocol, that needs work to be done in the BEHAVE WG?
</t></list>
</t>
<t>
upgrade the HTTP proxy-tunneled connection to the TURN server to use TLS,
</t>
<t>
use the same proxy selection procedure for TURN as currently done for HTTP
(e.g. Web Proxy Autodiscovery Protocol (WPAD) and .pac-files for Proxy-Auto-Config),
</t>
<t>
switch the usage of the HTTP proxy-tunneled connection with the TURN server from
HTTP to STUN/TURN,
<list><t>
Note: Usually, the browser would send further HTTP requests over HTTP proxy-tunneled connection.
Here, the connection in just established using HTTP CONNECT, but afterwards used for STUN/TURN messages.
</t></list>
</t>
<t>
use a preconfigured or standardized port range for UDP-based media streams or data channels,
</t>
<t>
learn from the proxy configuration script about the presence of a local TURN server and
use it for sending UDP traffic to the internet,
</t>
<t>
as an option and if needed, support ICE-TCP for TCP-based direct media connection to the WebRTC peer.
</t>
</list></t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>
The authors want to thank Heinrich Haager for all his input during many valuable discussions.
</t>
<t>
Furthermore, the authors want to thank for comments and suggestions received from
Bernard Aboba, Xavier Marjou, Dan Wing, ...
<!-- Harald Alvestrand,
Cameron Byrne,
Gustavo Garcia,
Markus Isomaki,
Hadriel Kaplan,
Reinaldo Penno,
Roy Radhika,
Uwe Rauschenbach and Tirumaleswar Reddy.-->
</t>
</section>
<!-- Possibly a 'Contributors' section ... -->
<section anchor="IANA" title="IANA Considerations">
<t>This memo includes no request to IANA.</t>
</section>
<section anchor="Security" title="Security Considerations">
<t>
TBD
</t>
</section>
</middle>
<!-- *****BACK MATTER ***** -->
<back>
<!-- References split into informative and normative -->
<references title="Normative References">
<!--?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml"?-->
&RFC2119;
&RFC2616;
&RFC2817;
&RFC3550;
&RFC4787;
&RFC5245;
&RFC5766;
</references>
<references title="Informative References">
<!-- Here we use entities that we defined at the beginning. -->
&RFC6062;
&RFC6455;
&RFC6544;
&RFC6887;
<!-- A reference written by an organization not a person. -->
<reference anchor="draft-ietf-httpbis-p2-semantics"
target="http://tools.ietf.org/html/draft-ietf-httpbis-p2-semantics-23#section-4.3.6">
<front>
<title>
Hypertext Transfer Protocol (HTTP/1.1): Semantics and Content
</title>
<author>
<organization>
R. Fielding, J. Reschke
</organization>
</author>
<date year="2013" />
</front>
</reference>
<reference anchor="draft-ietf-rtcweb-use-cases-and-requirements"
target="http://tools.ietf.org/html/draft-ietf-WebRTC-use-cases-and-requirements">
<front>
<title>
Web Real-Time Communication Use-cases and Requirements
</title>
<author>
<organization>
C. Holmberg, S. Hakansson, G. Eriksson
</organization>
</author>
<date year="2013" />
</front>
</reference>
<reference anchor="draft-ietf-rtcweb-data-channel"
target="http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel">
<front>
<title>
RTCWeb Data Channels
</title>
<author>
<organization>
R. Jesup, S. Loreto, M. Tuexen
</organization>
</author>
<date year="2013" />
</front>
</reference>
<reference anchor="draft-chenxin-behave-turn-WebSocket"
target="http://tools.ietf.org/html/draft-chenxin-behave-turn-WebSocket">
<front>
<title>
Traversal Using Relays around NAT (TURN) Extensions for WebSocket Allocations
</title>
<author>
<organization>
Xin. Chen
</organization>
</author>
<date year="2013" />
</front>
</reference>
<reference anchor="draft-miniero-rtcweb-http-fallback"
target="http://tools.ietf.org/html/draft-miniero-rtcweb-http-fallback">
<front>
<title>
HTTP Fallback for RTP Media Streams
</title>
<author>
<organization>
L. Miniero
</organization>
</author>
<date year="2012" />
</front>
</reference>
</references>
<!--
<section anchor="app-additional" title="Additional Stuff">
<t>This becomes an Appendix.</t>
</section>
Change Log
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</back>
</rfc>| PAFTECH AB 2003-2026 | 2026-04-24 01:18:22 |