Internet Engineering Task Force T. Stach, Ed.
Internet-Draft A. Hutton
Intended status: Informational Siemens Enterprise Communications
Expires: March 24, 2014 J. Uberti
Google
September 20, 2013
RTCWEB Considerations for NATs, Firewalls and HTTP proxies
draft-hutton-rtcweb-nat-firewall-considerations-02
Abstract
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.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on March 24, 2014.
Copyright Notice
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Provisions Relating to IETF Documents
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
1.1. Requirements Language . . . . . . . . . . . . . . . . . . 3
2. Considerations for NATs/Firewalls independent of HTTP proxies 4
2.1. NAT/Firewall open for outgoing UDP and TCP traffic . . . 4
2.2. NAT/Firewall open only for TCP traffic . . . . . . . . . 4
2.3. NAT/Firewall open only for TCP on restricted ports . . . 5
3. Considerations for NATs/Firewalls in presence of HTTP proxies 6
3.1. HTTP proxy with NAT/Firewall open for
outgoing UDP and TCP traffic . . . . . . . . . . . . . . 6
3.2. HTTP proxy with NAT/Firewall open only for TCP traffic . 6
3.3. HTTP proxy assisted TURN server connection . . . . . . . 6
3.3.1. TURN server connection via TCP . . . . . . . . . . . 6
3.3.2. TURN server connection via UDP . . . . . . . . . . . 8
4. Other Approaches . . . . . . . . . . . . . . . . . . . . . . 8
4.1. TURN server connection via WebSocket . . . . . . . . . . 8
4.2. Port Control Protocol . . . . . . . . . . . . . . . . . . 9
4.3. HTTP Fallback for RTP Media Streams . . . . . . . . . . . 9
5. Requirements for RTCWEB-enabled browsers . . . . . . . . . . 9
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 10
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 10
8. Security Considerations . . . . . . . . . . . . . . . . . . . 10
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 10
9.1. Normative References . . . . . . . . . . . . . . . . . . 10
9.2. Informative References . . . . . . . . . . . . . . . . . 11
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 12
1. Introduction
WebRTC is a web-based technique for direct interactive rich
communication using audio, video, and data between two peer browsers.
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
[RFC5245] in order to establish a media path between two WebRTC peers
in the presence of such NATs/FWs.
When WebRTC is deployed by the corporate IT department one can assume
that the corporate IT configures the corporate NATs, Firewalls, DPI
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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 [RFC4787], the peers will introduce a TURN server
[RFC5766] 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.
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.
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
[draft-ietf-rtcweb-use-cases-and-requirements].
The TURN server in the public internet is not sufficient to establish
connectivity for RTP-based media [RFC3550] and the WebRTC data
channel [draft-ietf-rtcweb-data-channel] 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) [RFC2616].
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.
This document examines impact of NAT/FW policies in Section 2.
Additional impacts due to the presence of a HTTP proxy are examined
in Section 3.
1.1. Requirements Language
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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 RFC 2119 [RFC2119].
2. Considerations for NATs/Firewalls independent of HTTP proxies
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 Section 3.
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.
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.
2.1. NAT/Firewall open for outgoing UDP and TCP traffic
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.
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.
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.
2.2. NAT/Firewall open only for TCP traffic
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.
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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
[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.
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.
Note that the second case is not to be confused with TURN/TCP
[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.
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
2.3. NAT/Firewall open only for TCP on restricted ports
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.
Open issue: Although
[draft-ietf-rtcweb-use-cases-and-requirements] 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.
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.
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3. Considerations for NATs/Firewalls in presence of HTTP proxies
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.
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.
3.1. HTTP proxy with NAT/Firewall open for outgoing UDP and TCP traffic
As in Section 2.1 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
Section 2.1 apply with respect to the traversal of the NAT/FW.
3.2. HTTP proxy with NAT/Firewall open only for TCP traffic
As in Section 2.2 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 Section 2.2 apply with respect to the traversal of the NAT/FW.
3.3. HTTP proxy assisted TURN server connection
3.3.1. TURN server connection via TCP
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 [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.
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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.
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 [draft-ietf-httpbis-p2-semantics] describes tunneling to ports
other than 80/443 and the corresponding security impacts.
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
[RFC2817] as this would require the TURN server to support HTTP. We
rather envisage the following sequence:
o the browser opens a TCP connection to the HTTP proxy,
o the browser issues a HTTP CONNECT request to the HTTP proxy with
the TURN server address in the Request URI, for example
CONNECT turn_server.example.com:5349 HTTP/1.1 Host:
turn_server.example.com:5349
o 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,
o the browser can do a TLS handshake over the virtual end-to-end TCP
connection with the TURN server.
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.
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.
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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.
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 [RFC5766] 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.
3.3.2. TURN server connection via UDP
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.
4. Other Approaches
4.1. TURN server connection via WebSocket
The WebRTC client could connect to a TURN server via WebSocket
[RFC6455] as described in [draft-chenxin-behave-turn-WebSocket].
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.
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.
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4.2. Port Control Protocol
As a further alternative, the Port Control Protocol (PCP) [RFC6887]
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.
4.3. HTTP Fallback for RTP Media Streams
As an alternative to using a TURN server it was proposed to send RTP
directly over HTTP [draft-miniero-rtcweb-http-fallback]. 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.
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.
5. Requirements for RTCWEB-enabled browsers
For the purpose of relaying WebRTC media streams or data channels a
browser needs to be able to
o connect to a TURN server via UDP, TCP and TLS,
o support the HTTP CONNECT method and request that a proxy establish
a tunneled TCP connection to a TURN server on its behalf,
o connect to a TURN server through this HTTP proxy-tunnelled TCP
connection,
o connect to the TURN server via application specified ports other
than the default STUN ports including the HTTP(s) ports,
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?
o upgrade the HTTP proxy-tunneled connection to the TURN server to
use TLS,
o 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),
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o switch the usage of the HTTP proxy-tunneled connection with the
TURN server from HTTP to STUN/TURN,
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.
o use a preconfigured or standardized port range for UDP-based media
streams or data channels,
o learn from the proxy configuration script about the presence of a
local TURN server and use it for sending UDP traffic to the
internet,
o as an option and if needed, support ICE-TCP for TCP-based direct
media connection to the WebRTC peer.
6. Acknowledgements
The authors want to thank Heinrich Haager for all his input during
many valuable discussions.
Furthermore, the authors want to thank for comments and suggestions
received from Bernard Aboba, Xavier Marjou, Dan Wing, ...
7. IANA Considerations
This memo includes no request to IANA.
8. Security Considerations
TBD
9. References
9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
[RFC2817] Khare, R. and S. Lawrence, "Upgrading to TLS Within HTTP/
1.1", RFC 2817, May 2000.
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[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4787] Audet, F. and C. Jennings, "Network Address Translation
(NAT) Behavioral Requirements for Unicast UDP", BCP 127,
RFC 4787, January 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
9.2. Informative References
[RFC6062] Perreault, S. and J. Rosenberg, "Traversal Using Relays
around NAT (TURN) Extensions for TCP Allocations", RFC
6062, November 2010.
[RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC
6455, December 2011.
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
"TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, March 2012.
[RFC6887] Wing, D., Cheshire, S., Boucadair, M., Penno, R., and P.
Selkirk, "Port Control Protocol (PCP)", RFC 6887, April
2013.
[draft-chenxin-behave-turn-WebSocket]
Xin. Chen , "Traversal Using Relays around NAT (TURN)
Extensions for WebSocket Allocations ", 2013, <http://
tools.ietf.org/html/draft-chenxin-behave-turn-WebSocket>.
[draft-ietf-httpbis-p2-semantics]
R. Fielding, J. Reschke , "Hypertext Transfer Protocol
(HTTP/1.1): Semantics and Content ", 2013, <http://
tools.ietf.org/html/draft-ietf-
httpbis-p2-semantics-23#section-4.3.6>.
[draft-ietf-rtcweb-data-channel]
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R. Jesup, S. Loreto, M. Tuexen , "RTCWeb Data Channels ",
2013, <http://tools.ietf.org/html/draft-ietf-rtcweb-data-
channel>.
[draft-ietf-rtcweb-use-cases-and-requirements]
C. Holmberg, S. Hakansson, G. Eriksson , "Web Real-Time
Communication Use-cases and Requirements ", 2013, <http://
tools.ietf.org/html/draft-ietf-WebRTC-use-cases-and-
requirements>.
[draft-miniero-rtcweb-http-fallback]
L. Miniero , "HTTP Fallback for RTP Media Streams ", 2012,
<http://tools.ietf.org/html/draft-miniero-rtcweb-http-
fallback>.
Authors' Addresses
Thomas Stach (editor)
Siemens Enterprise Communications
Dietrichgasse 27-29
Vienna 1030
AT
Email: thomas.stach@siemens-enterprise.com
Andrew Hutton
Siemens Enterprise Communications
Technology Drive
Nottingham NG9 1LA
UK
Email: andrew.hutton@siemens-enterprise.com
Justin Uberti
Google
747 6th Ave S
Kirkland, WA 98033
US
Email: justin@uberti.name
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