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Versions: 00 01 02 03

Internet Engineering Task Force                                 T. Stach
Internet-Draft                                                 A. Hutton
Intended status: Informational                                     Unify
Expires: July 24, 2014                                         J. Uberti
                                                                  Google
                                                        January 20, 2014


       RTCWEB Considerations for NATs, Firewalls and HTTP proxies
           draft-hutton-rtcweb-nat-firewall-considerations-03

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 specifies requirements on WebRTC enabled 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.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on July 24, 2014.

Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of



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   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
     1.1.  Requirements Language . . . . . . . . . . . . . . . . . .   4
   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 with NAT/Firewall open only to proxy routed
           traffic . . . . . . . . . . . . . . . . . . . . . . . . .   6
   4.  Solutions for Further Study . . . . . . . . . . . . . . . . .   7
     4.1.  HTTP CONNECT based mechanism  . . . . . . . . . . . . . .   7
     4.2.  ALPN - Use of Application Layer Protocol Negotiation  . .   8
     4.3.  TURN server connection via WebSocket  . . . . . . . . . .   9
     4.4.  HTTP Fallback for RTP Media Streams . . . . . . . . . . .   9
     4.5.  Port Control Protocol . . . . . . . . . . . . . . . . . .   9
     4.6.  Network Specific TURN Server  . . . . . . . . . . . . . .   9
   5.  Requirements for RTCWEB-enabled browsers  . . . . . . . . . .  10
   6.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  11
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  11
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  11
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  11
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  11
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  12
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  13

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.




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   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 [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 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 WebRTC
   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.




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1.1.  Requirements Language

   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 the 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




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   directly to the peer or contacting a TURN server via TCP that then
   relays RTP.

   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 using TURN to
   request a "TCP Allocation" as described in [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 request TCP allocations at the TURN server
   since it is preferable that WebRTC media is transported over UDP as
   far as possible.  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.

   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 provide
   this information to browser.








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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 with NAT/Firewall open only to proxy routed traffic

   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.  Currently only the case of an explicit proxy is
   considered here.

   This scenario is the most complex and controversial as it requires
   the WebRTC media to be tunneled through the proxy.  However such
   techniques are already specified in RFC's and deployed an example of
   this is websockets [RFC6455] which uses the HTTP CONNECT mechanism in
   the presense of HTTP Proxies.

   This document discusses some alternative approaches to achieving
   connectivity for WebRTC media in this environment but does not
   currently make any firm recommendations as the alternatives are




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   mostly work in progress in other areas of the IETF.  Therefore it is
   not possible to make such a recommendation at this time.

4.  Solutions for Further Study

   The following sections outline and provide some analysis of various
   solutions to the issues raised regarding WebRTC media traversing
   firewalls and proxies.  All of these potential solutions require
   further analysis by the IETF RTCWEB working group and in some cases
   may require work in other IETF working groups.

   It is possible that due to different network environments that WebRTC
   browsers may need to implement more than one solution.

   NOTE - THIS ANALYSIS IS NOT COMPLETE.

4.1.  HTTP CONNECT based mechanism

   A WebRTC browser could make use of 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.

   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.

   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.  The
   following is a possible sequence of events:

   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





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   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.

   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.  The WebRTC 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.

   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.

   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.

4.2.  ALPN - Use of Application Layer Protocol Negotiation

   The application layer protocol negotiation (ALPN)
   [draft-ietf-tls-applayerprotoneg] specifies a TLS extension which
   permits the application layer to negotiate protocol selection within
   the TLS handshake.  This provides an explicit and visable indication
   of the application layer protocol associated with the TLS connection
   allowing the application protocol to be visable without relying on
   the port number to identify the protocol.

   [draft-ietf-tls-applayerprotoneg] could therefore be used to identify
   that it is WebRTC media that is contained within the TLS connection.

   ALPN is effectively an extension to the HTTP CONNECT mechanism
   decribed in Section 4.1 since the establishment of the TLS connection
   would require the use of this mechanism in the presence of a proxy as
   described in [draft-ietf-httpbis-http2].




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4.3.  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.

   This is again effectively an extension to the HTTP CONNECT mechanism
   decribed in Section 4.1 since the establishment of the webcoskets
   connection would require the use of this mechanism in the presence of
   a proxy as described in [draft-ietf-httpbis-http2].  Like the ALPN
   approach the websockets approach also includes that the purpose of
   the websockets connection is to transport WebRTC media.

4.4.  HTTP Fallback for RTP Media Streams

   As an alternative to using a TURN server
   [draft-miniero-rtcweb-http-fallback] proposed to send RTP directly
   over HTTP.  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.

4.5.  Port Control Protocol

   As a further alternative, the Port Control Protocol (PCP) [RFC6887]
   allows the client to communicate with the NAT/FW and negotiate how
   incoming IPv6 or IPv4 packets are translated and forwarded.  However,
   to be successful such a solution would require the widespread
   deployment and use of PCP enabled firewalls so this does not appear
   to be a workable solution at least for early deployments of WebRTC.

4.6.  Network Specific TURN Server

   If a network specific 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



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   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.

   Since the TURN server is under the same administrative control as the
   NAT/FW then it can be assumed that the NAT/FW allows WebRTC media
   that traverses the TURN server to traverse the NAT/FW.

   The implementation of this solution in WebRTC is actually a
   requirement specified in
   [draft-ietf-rtcweb-use-cases-and-requirements].

   The implementation of this solution in WebRTC does not remove the
   need for other solutions for the case when there is no such network
   specific TURN server.

5.  Requirements for RTCWEB-enabled browsers

   THIS SECTION IS EVEN MORE WORK IN PROGRESS THAN PREVIOUS SECTIONS.

   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 a mechanism for connecting to a TURN server in the
      presence of a firewall that only permits connections that orginate
      from a HTTP Proxy.  The mechanism is for further study.

   o  connect to the TURN server via application specified ports other
      than the default STUN ports including the HTTP(s) ports,

   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),

   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.






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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

   In case of using HTTP CONNECT to a TURN server the security
   consideration of [[draft-ietf-httpbis-p2-semantics], Section-4.3.6]
   apply.  It states that there "are significant risks in establishing a
   tunnel to arbitrary servers, particularly when the destination is a
   well-known or reserved TCP port that is not intended for Web traffic.
   ... Proxies that support CONNECT SHOULD restrict its use to a limited
   set of known ports or a configurable whitelist of safe request
   targets."

   Consequently when HTTP CONNECT is used to reach a TURN server, the
   proxy administrator SHOULD configure a whitelist of trusted TURN
   servers and/or a blacklist of TURN server known to be subject to
   fraud or other undesired behavior.

   With respect to the other discussed alternatives the security
   considerations of the corresponding RFCs and Internet Drafts apply.

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.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.



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   [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-http2]
              M. Belshe, R. Peon, M. Thomson, A. Melnikov, "Hypertext
              Transfer Protocol version 2.0", 2013,
              <http://tools.ietf.org/html/
              draft-ietf-httpbis-http2-09#section-8.3>.

   [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-25#section-4.3.6>.




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   [draft-ietf-rtcweb-data-channel]
              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-ietf-tls-applayerprotoneg]
              S. Friedl, A. Popov, A. Langley, E. Stephan, "Transport
              Layer Security (TLS) Application Layer Protocol
              Negotiation Extension", 2013, <http://tools.ietf.org/html/
              draft-ietf-tls-applayerprotoneg>.

   [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
   Unify
   Dietrichgasse 27-29
   Vienna  1030
   AT

   Email: thomas.stach@unify.com


   Andrew Hutton
   Unify
   Technology Drive
   Nottingham  NG9 1LA
   UK

   Email: andrew.hutton@unify.com











Stach, et al.             Expires July 24, 2014                [Page 13]


Internet-Draft                RTCWEB NAT-FW                 January 2014


   Justin Uberti
   Google
   747 6th Ave S
   Kirkland, WA  98033
   US

   Email: justin@uberti.name












































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