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Versions: (draft-johnston-tram-stun-origin) 00 01 02 03 04 05 06

TRAM Working Group                                           A. Johnston
Internet-Draft                                                     Avaya
Intended status: Standards Track                               J. Uberti
Expires: June 5, 2015                                             Google
                                                               J. Yoakum
                                                                K. Singh
                                                                   Avaya
                                                        December 2, 2014


               An Origin Attribute for the STUN Protocol
                     draft-ietf-tram-stun-origin-03

Abstract

   STUN, or Session Traversal Utilities for NAT, is a protocol used to
   assist other protocols traverse Network Address Translators or NATs.
   STUN, and STUN extensions such as TURN, or Traversal Using Relays
   around NAT, and ICE, Interactive Communications Establishment, have
   been around for many years but with WebRTC, Web Real-Time
   Communications, STUN and related extensions are about to see major
   deployments and implementation due to these protocols being
   implemented in browsers.  This specification defines an ORIGIN
   attribute for STUN that can be used in similar ways to the HTTP
   header field of the same name.  WebRTC browsers utilizing STUN and
   TURN would include this attribute which would provide servers with
   additional information about the STUN and TURN requests they receive.
   This specification defines the usage of the STUN ORIGIN attribute for
   web and SIP contexts.

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 June 5, 2015.

Copyright Notice



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   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
   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 . . . . . . . . . . . . . . . . . . . . . . . . .  3
     1.1.  Requirements Language  . . . . . . . . . . . . . . . . . .  5
   2.  STUN ORIGIN attribute  . . . . . . . . . . . . . . . . . . . .  5
     2.1.  STUN Usage . . . . . . . . . . . . . . . . . . . . . . . .  6
     2.2.  TURN Usage . . . . . . . . . . . . . . . . . . . . . . . .  6
     2.3.  NAT Behavior Discovery Usage . . . . . . . . . . . . . . .  6
     2.4.  ICE Usage  . . . . . . . . . . . . . . . . . . . . . . . .  6
     2.5.  Media Keep-Alive Usage . . . . . . . . . . . . . . . . . .  6
     2.6.  SIP Keep-Alive Usage . . . . . . . . . . . . . . . . . . .  7
     2.7.  Multiple Origins . . . . . . . . . . . . . . . . . . . . .  7
   3.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  7
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . .  7
   5.  Implementation Status  . . . . . . . . . . . . . . . . . . . .  8
   6.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 10
   7.  Normative References . . . . . . . . . . . . . . . . . . . . . 10
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12



















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1.  Introduction

   STUN, or Session Traversal Utilities for NAT, is a protocol used to
   assist other protocols traverse Network Address Translators or NATs.
   TURN, or Traversal Using Relays around NAT [RFC5766], is a STUN
   extension [RFC5389] that allows endpoints to acquire a relayed
   address for media flows.  It is most commonly used in conjunction
   with ICE, Interactive Connectivity Establishment [RFC5245], which is
   used to establish peer-to-peer flows between endpoints through NATs
   and firewalls.

   STUN defines three authentication modes, depending on the STUN usage.
   For STUN binding requests sent between peers, such as for ICE
   connectivity checks, a short term authentication method is
   recommended.  Each peer contributes random strings which are
   exchanged over signaling and used to authenticate the connectivity
   checks.  For TURN, a usage of STUN used to acquire and refresh relay
   addresses, a long term authentication method is recommended.  This
   authentication is similar to SIP Digest [RFC3261], which involves an
   authentication challenge for each request.  A server, upon receipt of
   a TURN request, generates an authentication challenge that includes a
   realm and nonce.  The client resends the TURN request supplying a
   user name and password based on the realm indicated by the server.
   For a STUN binding request sent to a STUN server, no authentication
   is recommended, as generating the response is less work for a server
   than the server utilizing the short term or long term authentication
   approach.  In addition, the resource requirements of operating a STUN
   server are minimal.

   WebRTC, Web Real-Time Communications, adds peer-to-peer real-time,
   interactive voice and video media capabilities and data channels to
   browsers [I-D.ietf-rtcweb-overview] without a plugin or download, and
   allows web developers to access this functionality using JavaScript
   API calls [WebRTC-API].  WebRTC includes STUN, TURN, and ICE client
   functionality built into browsers.  For a session established between
   two browsers, if either browser is behind a NAT, a STUN server is
   necessary.  Public STUN servers are currently available and a web
   application can suggest a particular STUN server be used.  In other
   cases, a TURN server is needed to establish a peer connection.  In
   this case, TURN credentials need to be available to the browser for
   the long term authentication approach.  A TURN server for WebRTC
   might serve a number of different domains and realms.

   From the perspective of the web application provider, providing
   service for a number of different domains and realms, it is useful to
   know something about the source of the STUN request when processing
   the request.  For a web application provider STUN or TURN server, the
   server will have no idea which web pages or sites are sending binding



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   requests to the service.  In conventional applications, the SOFTWARE
   attribute would provide some identifying information to the service,
   but that no longer works when the browser is the application.  For a
   web application provider TURN server, the TURN server does not know
   which realm to include in an authentication challenge.

   In the web world, HTTP requests have the concept of origin.  The
   origin of a web page, as defined in [RFC6454], is defined by the
   URI's scheme, host or IP address, and port portions.  The HTTP Origin
   header field inserted by the web browser carries this information and
   is useful information for servers that receive HTTP requests
   generated via JavaScript.  For example, Cross Origin Resource
   Sharing, CORS, allows an HTTP server to serve HTTP requests from
   multiple origins.

   This specification proposes extending the origin concept to STUN
   requests.  STUN requests generated by a web browser would include the
   origin of the HTTP page that is initiating the Peer Connection.
   Using this extra information, a STUN server could use the origin to
   determine which STUN binding requests to respond to, reducing the
   load on a STUN server.  Using this information, a TURN server could
   use the origin to determine which realm to include in the
   authentication challenge.  A TURN server can also use the origin
   information for logging and analytics, and also as additional
   information after authentication for providing service.  This
   specification also defines an origin for SIP and XMPP users of STUN
   and TURN.

   An important use case that the STUN Origin helps solve is the
   operation of a multi-tenanted TURN server (i.e. a TURN server that
   serves multiple, perhaps tens of thousands of different domains).
   The problem associated with this use case is described in Section 4.5
   of [I-D.ietf-tram-auth-problems].  While it is possible for a TURN
   server to use the same authentication credentials across many
   domains, a more likely (and more manageable) scenario is to have
   separate credentials for each domain, and hence a different realm for
   each domain.  With the TURN server configured with a mapping between
   a domain (conveyed in the Origin) and the realm string (to be used in
   the authentication challenge), a single TURN URI could be used across
   all domains, and the resulting JavaScript code would be portable.

   Note that the origin information is most useful as a hint in initial
   STUN and TURN requests as received by a server.  However, origin
   information still has value throughout the session even after
   authentication for logging and other purposes.

   The following sections of this document define the STUN ORIGIN
   attribute and define its usage.



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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.  STUN ORIGIN attribute

   This specification defines how to apply the web origin concept and
   syntax of [RFC6454] to the STUN protocol.

   This specification defines a new Attribute to the STUN protocol
   [RFC5389].  The attribute is called ORIGIN and uses the syntax
   defined in Section 15 of [RFC5389].  The number used for this in the
   type field is 0x802F, chosen in the comprehension optional range.
   The value of ORIGIN is a variable-length value.  It MUST contain a
   UTF-8 [RFC3629] encoded sequence of characters.  Senders MAY include
   multiple ORIGIN attributes in a request, and receivers MUST support
   parsing and receiving multiple ORIGIN attributes.  The size of
   ORIGINs included in a STUN message can have a major impact on the
   size of a STUN packet, and could potentially cause UDP fragmentation.
   HTTP origins are less than 267 characters (the maximum 253 character
   domain name plus 8 characters for the URI scheme plus 5 characters
   for the port number).

   For a web browser (HTTP User Agent), the contents of the ORIGIN
   attribute is the unicode-serialization of an origin defined in
   Section 6.1 of [RFC6454].  The origin value included is the same as
   the Origin header field for an HTTP request generated from the web
   page that is creating the Peer Connection.  It does not include any
   string terminating (\x00) character in the serialization.

   For a SIP User Agent [RFC3261] using STUN and TURN, the ORIGIN
   attribute is set to be the URI of the registrar server used by the
   User Agent (i.e. the Request-URI of a REGISTER method).  This is the
   full Request-URI component of the SIP ABNF defined in [RFC3261].  For
   example, a SIP user "sip:bob@biloxi.example.com" might register with
   the Request-URI of "sip:registrar.biloxi.example.com".

   For an XMPP client [RFC6120] using STUN and TURN, the ORIGIN
   attribute is an XMPP URI [RFC5122] representing the full domainpart
   of the client's Jabber ID (JID) as defined in the ABNF of [RFC6122];
   for example, if the client's JID is "juliet@im.example.com/balcony"
   then the ORIGIN attribute would be "xmpp:im.example.com".

   Other contexts can define a usage of the ORIGIN attribute to use an
   appropriate URI or URL.



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   If an ORIGIN attribute is not present in a request, it is up to the
   server how to handle the request.  For example, it could assume a
   default Origin.

2.1.  STUN Usage

   For STUN requests sent without authentication to a STUN server (i.e.
   STUN binding requests sent to a STUN server), the STUN client SHOULD
   include the ORIGIN attribute.  A STUN server can derive additional
   information for logging and analytics about the request through the
   ORIGIN attribute, such as the source of the request.  For example, an
   enterprise STUN server might only reply to STUN binding requests from
   certain domains.

2.2.  TURN Usage

   For STUN requests sent using the long-term authentication method,
   such as TURN [RFC5766] allocate requests, the STUN client SHOULD
   include the ORIGIN attribute.  Including the ORIGIN attribute in the
   Send method is NOT RECOMMENDED.  A TURN server can use the ORIGIN
   attribute to determine which REALM to include in the authentication
   challenge.  A TURN server can also use the ORIGIN attribute after
   authentication to provide appropriate service.  See the section below
   on "Multiple Origins."

2.3.  NAT Behavior Discovery Usage

   For the NAT Behavior Discovery Usage in [RFC5780], the ORIGIN
   attribute SHOULD be included in requests sent to a STUN server.  This
   usage is most similar to the STUN Usage described earlier.

2.4.  ICE Usage

   For STUN requests sent using the short-term authentication method,
   such as ICE connectivity checks [RFC5245], the use of the ORIGIN
   attribute is NOT RECOMMENDED.  No valid use cases for the ORIGIN
   attribute have been identified to date.

2.5.  Media Keep-Alive Usage

   For media keep-alive STUN requests described in Section 20 of
   [RFC5245], the use of the ORIGIN attribute is NOT RECOMMENDED.  No
   valid use cases for the ORIGIN attribute have been identified to
   date.







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2.6.  SIP Keep-Alive Usage

   For SIP keep-alive STUN requests described in [RFC5626], the use of
   the ORIGIN attribute is NOT RECOMMENDED.  No valid use cases for the
   ORIGIN attribute have been identified to date.

2.7.  Multiple Origins

   Multiple Origins for HTTP Requests are described in Section 7.2 of
   [RFC6454].  Multiple origins can occur when the same resource is
   fetched by multiple origins at the same time (e.g. multiple tabs,
   windows, frames, etc.).  In the context of WebRTC, it doesn't make
   sense for a STUN binding or TURN allocation to be shared across
   origins (e.g.  Peer Connections).  Based on their definitions,
   multiple SIP and XMPP origins also do not apply here.


3.  IANA Considerations

   This specification, if approved, adds a new value to the IANA "STUN
   Attributes Registry" created by [RFC5389].  The ORIGIN attribute
   value is 0x802F.


4.  Security Considerations

   The security considerations of [RFC6454] apply to this extension.
   Servers using the information present in the STUN ORIGIN attribute
   need to realize that this attribute could be set arbitrarily by a
   non-browser client or modified by an intermediary.  The method
   proposed in this document is not meant to replace existing STUN
   authentication mechanisms but to provide additional information to
   the server for logging and analytics and how to handle the request
   after authentication.

   Just as browsers do not allow a web application to set the Origin
   header field via JavaScript, browsers should not allow a web
   application through JavaScript to set the STUN ORIGIN attribute.

   While the STUN MESSAGE-INTEGRITY attribute can provide integrity
   protection for all attributes present in a STUN request, MESSAGE-
   INTEGRITY is not present in the initial STUN message sent.  As a
   result, an ORIGIN attribute could be modified or removed from a STUN
   request without the server knowing.  DTLS or TLS transport SHOULD be
   used when integrity protection for the ORIGIN attribute is important.

   The STUN ORIGIN attribute does have privacy implications.  The
   recipient of the STUN request learns the web origin of the user.  In



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   addition, an on-path attacker could determine this information by
   inspecting STUN messages between the STUN client and STUN server,
   depending on the transport used.  This information is often available
   in other messages sent by the browser, such as DNS or HTTP requests.
   However, in cases where secure HTTP is used, including the ORIGIN
   attribute over an unencrypted transport could leak this information.
   STUN has a defined TLS transport; however, TLS transport is generally
   unsuitable for the real-time media flows that follow STUN requests
   and must use the same transport.  The DTLS transport for STUN
   [I-D.ietf-tram-stun-dtls] provides a very good privacy solution to
   this problem.  In cases where privacy is paramount, the ORIGIN
   attribute SHOULD NOT be included or only included if DTLS or TLS
   transport is used.

   The STUN ORIGIN attribute also has privacy implications in that the
   origin information is shared with a STUN or TURN server which
   otherwise might not know this information.  This information could be
   used to track usage of real-time communication services.  A STUN or
   TURN server will always know the public IP address of each user, but
   the ORIGIN attribute provides more information about which service or
   provider is being used.  The particular STUN and TURN servers used
   are usually selected by the real-time communications service provider
   (i.e. the web provider for WebRTC or the SIP or XMPP service
   provider).  In addition, they are usually also run by the same
   provider, or by a trusted partner, especially for TURN.  However, a
   service or provider using an untrusted third party STUN or TURN
   server needs to recognize that the operator of the third party STUN
   or TURN server will learn the identity of the service or provider
   through this extension.


5.  Implementation Status

   Note to RFC Editor: Please remove this entire section prior to
   publication, including the reference to RFC 6982.

   This section records the status of known implementations of the
   protocol defined by this specification at the time of posting of this
   Internet-Draft, and is based on a proposal described in [RFC6982].
   The description of implementations in this section is intended to
   assist the IETF in its decision processes in progressing drafts to
   RFCs.  Please note that the listing of any individual implementation
   here does not imply endorsement by the IETF.  Furthermore, no effort
   has been spent to verify the information presented here that was
   supplied by IETF contributors.  This is not intended as, and must not
   be construed to be, a catalog of available implementations or their
   features.  Readers are advised to note that other implementations may
   exist.



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   According to [RFC6982], "this will allow reviewers and working groups
   to assign due consideration to documents that have the benefit of
   running code, which may serve as evidence of valuable experimentation
   and feedback that have made the implemented protocols more mature.
   It is up to the individual working groups to use this information as
   they see fit".

   Two proof-of-concept implementations have been created in support of
   this proposed standard.  One provides a WebRTC enabled browser that
   includes the appropriate STUN ORIGIN Attribute with the Origin
   insight known to the browser in STUN/TURN messages sent to servers.
   The other provides an example of a multiple realms capable TURN
   server that takes advantage of Origin insight provided in the STUN
   ORIGIN Attribute.

   A Chrome browser implementation has been created by Graham Yoakum and
   Ryan Yoakum (Skobalt LLC) and is freely licensed under the standard
   terms of the open source Chromium and WebRTC projects.  This proof-
   of-concept version of the Google Chrome browser (nicknamed 'Chromeo')
   sends Origin insight in STUN and TURN messages using the proposed new
   STUN ORIGIN attribute with a value of 0x802F (as initially proposed,
   however that value is easily changed in a single line of code).
   'Chromeo' includes a Chrome flag to enable and disable this unique
   feature (and is by default disabled to prevent any non-intentional
   use of this feature until the standard is finalized).  This
   implementation is based on is draft-johnston-tram-stun-origin-02.

   Coordinated changes to both the WebRTC and Chromium open source
   projects have been formally submitted for consideration.  The two
   submitted change lists together implement the complete browser proof-
   of-concept.  'Chromeo' has been built for Linux and STUN protocol
   behavior has been verified using WireShark traces illustrating that
   proper STUN Origin attributes are being included in STUN/TURN
   messages sent by the browser to servers (screen captures of STUN
   messages illustrating the Origin attribute and content are
   available).

   The WebRTC and Chromium open source projects can be found at:
   http://www.webrtc.org/ and http://www.chromium.org/

   Google can choose to accept or modify the changes proposed for Chrome
   and other browser vendors can access and take advantage of the
   publicly available WebRTC and Chromium open source submissions as
   desired.  Hopefully this will enable browsers to quickly implement
   STUN Origin enhancements.

   A multiple realms capable advanced open source Origin enabled TURN
   server (named 'Coturn') has been created by Oleg Moskalenko and is



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   freely licensed under the New BSD license.  This reference
   implementation and proof-of-concept provides a clone (a spin-off) of
   the rfc5766-turn-server project adding Origin-based multiple realms
   support.

   'Coturn' is backward-compatible with rfc5766-turn-server project but
   the code is more complex and it uses a different (also more complex)
   database structure.  It is the intent to add all IETF TRAM TURN
   server related capabilities to this project as they mature.  'Coturn'
   is publicly available and can be found at:
   https://code.google.com/p/coturn/


6.  Acknowledgements

   Thanks to John Selbie, Tirumaleswar Reddy, Simon Perreault, Marc
   Petit-Huguenin, Andy Hutton, and Oleg Moskalenko for their feedback
   and reviews.  Special thanks to Graham Yoakum and Ryan Yoakum of
   Skobalt LLC and Oleg Moskalenko of rfc5766-turn-server project for
   contributing open source proof-of-concept implementations for a
   Chrome web browser and a multiple realms capable TURN server, quickly
   demonstrating feasibility.


7.  Normative References

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-13
              (work in progress), November 2014.

   [I-D.ietf-tram-auth-problems]
              Reddy, T., R, R., Perumal, M., and A. Yegin, "Problems
              with STUN long-term Authentication for TURN",
              draft-ietf-tram-auth-problems-05 (work in progress),
              August 2014.

   [I-D.ietf-tram-stun-dtls]
              Petit-Huguenin, M. and G. Salgueiro, "Datagram Transport
              Layer Security (DTLS) as Transport for Session Traversal
              Utilities for NAT (STUN)", draft-ietf-tram-stun-dtls-05
              (work in progress), June 2014.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.



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              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3489]  Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
              "STUN - Simple Traversal of User Datagram Protocol (UDP)
              Through Network Address Translators (NATs)", RFC 3489,
              March 2003.

   [RFC3629]  Yergeau, F., "UTF-8, a transformation format of ISO
              10646", STD 63, RFC 3629, November 2003.

   [RFC4020]  Kompella, K. and A. Zinin, "Early IANA Allocation of
              Standards Track Code Points", RFC 4020, February 2005.

   [RFC5122]  Saint-Andre, P., "Internationalized Resource Identifiers
              (IRIs) and Uniform Resource Identifiers (URIs) for the
              Extensible Messaging and Presence Protocol (XMPP)",
              RFC 5122, February 2008.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [RFC5626]  Jennings, C., Mahy, R., and F. Audet, "Managing Client-
              Initiated Connections in the Session Initiation Protocol
              (SIP)", RFC 5626, October 2009.

   [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.

   [RFC5780]  MacDonald, D. and B. Lowekamp, "NAT Behavior Discovery
              Using Session Traversal Utilities for NAT (STUN)",
              RFC 5780, May 2010.

   [RFC6066]  Eastlake, D., "Transport Layer Security (TLS) Extensions:
              Extension Definitions", RFC 6066, January 2011.

   [RFC6120]  Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Core", RFC 6120, March 2011.

   [RFC6122]  Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Address Format", RFC 6122, March 2011.



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   [RFC6454]  Barth, A., "The Web Origin Concept", RFC 6454,
              December 2011.

   [RFC6982]  Sheffer, Y. and A. Farrel, "Improving Awareness of Running
              Code: The Implementation Status Section", RFC 6982,
              July 2013.

   [WebRTC-API]
              Bergkvist, A., Burnett, D., Jennings, C., and A.
              Narayanan, "WebRTC 1.0: Real-time Communication Between
              Browsers", W3C Working Draft http://www.w3.org/TR/webrtc/,
              2013, <http://www.w3.org/TR/2013/WD-webrtc-20130910/>.


Authors' Addresses

   Alan Johnston
   Avaya
   St. Louis, MO
   USA

   Phone:
   Email: alan.b.johnston@gmail.com


   Justin Uberti
   Google
   Kirkland, WA
   USA

   Phone:
   Email: justin@uberti.name


   John Yoakum
   Avaya
   Cary, NC
   USA

   Phone:
   Email: yoakum@avaya.com










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   Kundan Singh
   Avaya
   San Francisco, CA
   USA

   Phone:
   Email: kundan10@gmail.com












































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