Internet DRAFT - draft-hutton-rtcweb-nat-firewall-considerations
draft-hutton-rtcweb-nat-firewall-considerations
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.
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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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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
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Justin Uberti
Google
747 6th Ave S
Kirkland, WA 98033
US
Email: justin@uberti.name
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