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Versions: 00 draft-iab-cc-workshop-report

Network Working Group                                      H. Tschofenig
Internet-Draft                                                 L. Eggert
Intended status: Informational                                 Z. Sarker
Expires: April 18, 2013                                 October 15, 2012


Report from the IAB/IRTF Workshop on Congestion Control for Interactive
                        Real-Time Communication
               draft-tschofenig-cc-workshop-report-00.txt

Abstract

   This document provides a summary of the IAB/IRTF Workshop on
   'Congestion Control for Interactive Real-Time Communication', which
   took place in Vancouver, Canada, on July 28, 2012.  The main goal of
   the workshop was to foster a discussion on congestion control
   mechanisms for interactive real-time communication.  This report
   summarizes the discussions and lists the conclusions and
   recommendations to the Internet Engineering Task Force (IETF)
   community.  The views and positions in this report are those of the
   workshop participants and do not necessarily reflect the views and
   positions of the authors, the Internet Architecture Board (IAB) or
   the Internet Research Task Force (IRTF).

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 April 18, 2013.

Copyright Notice

   Copyright (c) 2012 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



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   (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
   2.  History and Current Status . . . . . . . . . . . . . . . . . .  5
   3.  Recommendations  . . . . . . . . . . . . . . . . . . . . . . .  6
     3.1.  Changes to Network Infrastructure  . . . . . . . . . . . .  6
     3.2.  Avoiding Self-Inflicted Queuing  . . . . . . . . . . . . .  7
   4.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 10
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 11
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 12
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 13
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 13
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 13
   Appendix A.  Program Committee . . . . . . . . . . . . . . . . . . 14
   Appendix B.  Workshop Material . . . . . . . . . . . . . . . . . . 15
   Appendix C.  Accepted Position Papers  . . . . . . . . . . . . . . 16
   Appendix D.  Workshop Participants . . . . . . . . . . . . . . . . 19
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22
























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1.  Introduction

   Any application that sends significant amounts of data over the
   Internet is expected to implement reasonable congestion control
   behavior.  Although the specific mechanisms depend on the application
   or protocol, the motivations for congestion control are well
   understood and documented in RFC 2914 [RFC2914] and RFC 5405
   [RFC5405].  The goals are:

   1.  Preventing congestion collapse.

   2.  Allowing multiple flows to share the network fairly.

   The Internet has been used for interactive real-time communication
   for decades, most of which is being transmitted using RTP over UDP,
   often over provisioned capacity and/or using only rudimentary
   congestion control mechanisms.

   The development and upcoming widespread deployment of web-based real-
   time media communication - where RTP is used to and from web browsers
   to transmit audio, video and data - will likely result in substantial
   new Internet traffic.  Due to the projected volume of this traffic,
   as well as the fact that it is more likely to use unprovisioned
   capacity, it is essential that it is transmitted with robust and
   effective congestion control mechanisms.

   Designing congestion control mechanisms that perform well under a
   wide variety of traffic mixes and over network paths with widely
   varying characteristics is not easy.  Prevention of congestion
   collapse can be achieved through "circuit breaker" mechanisms, but
   for media flows that are supposed to coexist with a user's other
   ongoing communication sessions, a congestion control mechanism that
   shares capacity fairly in the presence of a mix of TCP, UDP and other
   protocol flows is needed.

   Many additional complications arise.  Here are some examples:

   1.  Real-time interactive media sessions require low latencies,
       whereas streaming media can use large play-out buffers.

   2.  RTCP feedback about an RTP session typically arrives much less
       frequently than, for example, TCP ACKs for a given TCP
       connection.  The RTP/RTCP control loop can lead to a longer
       reaction time.

   3.  Media codecs can usually only adjust their output rates in a much
       more coarse-grained fashion than, for example, TCP, and user
       experience suffers if encoding rates are switched too frequently.



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       Codecs typically have a minimum sending rate as well.

   4.  Some bits of an encoded media stream are more important than
       others.  For example, losing or dropping an I-frame of a video
       stream is more problematic than dropping a P-frame.

   5.  Ramping up the transmission rate can be problematic.  Simply
       increasing the output rate of the codec without knowledge whether
       the network path can sustain transmission at the increased rate
       runs the danger of incurring a significant amount of packet loss
       that can cause playback artifacts.

   6.  A congestion control scheme for interactive media needs to handle
       bundles of interrelated flows (audio, video, data), in a way that
       accommodates the preferences of the application in the event of
       congestion.

   7.  The desire to provide a congestion control mechanism that can be
       efficiently implemented inside an application (e.g., a web server
       and a browser) imposes additional restrictions.

   8.  There are explicit congestion signals (such as ECN), and there
       are indirect indications of congestion (such as packet delay and
       loss).  However, congestion is not the only source of these
       indirect signals.  Care must be taken to account for each of
       these signals, particularly if various applications react on the
       same signals.

   9.  Network elements and end device operating systems often create
       buffers to better support TCP-based applications.  These buffers
       introduce additional communication delay, which harms the small
       delay budget available for real-time applications.

   Note that this document is a report on the proceedings of the
   workshop.  The views and positions documented in this report are
   those of the workshop participants and do not necessarily reflect the
   views of the authors.














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2.  History and Current Status

   The currently deployed RTP-based RTC systems use congestion control
   schemes that are ad hoc at best.  As the Internet moves towards an
   RTCWeb standard for interoperable audio/video conferencing, it needs
   a standardized and most importantly effective mechanism for
   congestion control.

   The most immediate need is for a control mechanism that balances a
   real-time flow fairly among other competing traffic.  For instance,
   under congested conditions, it would be inappropriate to use far more
   bandwidth than flows from other applications are using, particularly
   lasting gross over-utilization of the network.  Simple "circuit
   breaker" -style control mechanisms can prevent congestion collapse by
   disabling a circuit that far exceeds a fair share and making the
   circuit remain disabled for some time.  But does not in any way
   enable concurrent flows to share available network capacity sensibly.

   Therefore, in addition to circuit breakers, more complete methods for
   media congestion control are needed.  The focus of the workshop was
   on discussing ideas for these more complete congestion control
   methods.Note that the primary focus of the workshop is RTCWeb-style
   audio/video flows.  The assumed underlying flows (in most attendees
   minds) were digital audio/video data encapsulated in RTP over UDP
   over IP.  Most of the concepts presented at the workshop could be
   used with different media types over different transports, but this
   was the generally assumed baseline for discussions.
























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3.  Recommendations

   The participants suggested to explore two complementary solution
   tracks, as described in the two sub-sections below.

3.1.  Changes to Network Infrastructure

   Like for all other traffic on the network, better data plane
   infrastructure improves the perceived quality of the best-effort
   service the Internet provides for RTCWeb flows.  The IETF has already
   developed several technologies that would be of immediate usefulness,
   if they were deployed.  The workshop participants expressed the hope
   that due to the volume and importance of RTCWeb traffic, some of
   these technologies might finally see widespread use.

   The first and by far most important improvement is traffic
   segregation, i.e., the ability to use different queues for different
   traffic types.  Specifically jitter/delay sensitive and throughput
   maximizing protocols need to be in different queues.  It is not
   possible for a single queue/AQM to be optimal for both.

   Explicit Congestion Notification (ECN) allows routers along the end-
   to-end path to signal the onset of congestion and allows applications
   to respond early, avoiding losses and keep queue sizes short and
   therefore end-to-end delay low.  ECN is implemented on some end
   system stacks and routers, but frequently not enabled.

   Different mechanisms have been developed to facilitate traffic
   segregation.  Differentiated services are one possibility in this
   space if applications start to mark outgoing traffic appropriately
   and routers would segregate traffic accordingly, browsers could more
   directly control the relative importance of their various flows, and
   avoid self-competition.  Compared to ECN, however, DiffServ is far
   more difficult to deploy meaningfully end-to-end, especially given
   that DSCPs have no defined end-to-end meaning and packets can be re-
   marked.  Quality-of-service (QoS) signaling together with resource
   reservation facilities would enable a fine-grained and flexible way
   to indicate resource needs to network elements but it is also by far
   the most heavyweight proposal, and unlikely to be viable in the
   global Internet.

   However, any change to network infrastructure will take time
   particularly if the interest of the involved stakeholders is not
   aligned (as it is the case for network operators and over-the-top
   application providers).  It is therefore imperative that RTCWeb
   congestion control provides adequate improvement in the absence of
   any of the aforementioned schemes.




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3.2.  Avoiding Self-Inflicted Queuing

   This approach tries to ensure that the network does not get congested
   by focusing on the traffic sent by a single host independent of any
   changes to networks, as mentioned in the earlier chapter.  A single
   congestion manager within the end host or the browser could already
   help to coordinate various congestion control activities and to
   ensure a more coordinated approach between different applications,
   and different flows.

   The following activities were suggested during the workshop.

   Reacting to all Congestion Signals:

      To initiate the congestion control process it is important to
      detect congestion in the communication path.  Congestion in the
      comminication can be detected using either an explicit mechanism
      or an implicit mechanism.  The explicit mechanism involves direct
      congestion signaling usually from the congested network node, such
      as ECN.  The ECN bits are set in the IP header by the network node
      before it starts to drop the packets and gives end hosts to react
      to the congestion.  In case of implicite mechanism, a particular
      characteristics or event in the network traffic can be inferred as
      a congestion signal.  For example, packet loss events or observed
      delay increase or jitter in the communication path can be taken as
      an implicit congestion signals.  These measurements can also be
      made available in a variety of different protocols, such as RTCP
      reports or transport protocols.  Communication endpoints can
      trigger congestion control on the event of these phenomena.  It is
      also possible to correlate these events to detect congestion in
      the communication path.  The implicit and explicit signals can
      come in any sequence and at any time during the ongoing media
      session.  It is recommended to react to all the possible implicit
      congestion signals, ideally from all applications at the end host,
      to avoid self- inflicted queues.

   Delay- and Loss-based Algorithms:

      The main goal of designing a congestion control algorithm for
      real-time conversational media is to achieve low latency.  An
      explicit signaling based congestion control algorithm perhaps is
      the best fit for this purpose as the network nodes are directly
      involved in the process.  However, in case of absence of ECN
      support in the end-to-end communication path delay based
      algorithms are needed where the increased delay in the
      communication path is taken into account as implicit congestion
      signal.  While it is difficult to design algorithms based on delay
      since many wireless networks show a wide delay variation even in a



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      non-congested network.  The workshop participants recommended that
      more research should be done to better understand non-congestion
      related delay variation in the network.  General consensus among
      the workshop participants was that when latency-based techniques
      were competing against loss-based techniques, the loss-based
      techniques got all the bandwidth.

   Algorithm Evaluation:

      The Internet consists of heterogeneous networks, which also
      includes misconfigured, and unmanaged network nodes.  Bandwidth
      and latency varies a lot.  Not only this, different services
      deployed using RTP/UDP have different requirements in terms of
      media quality.  A congestion control algorithm needs to perform
      well not only in simulators but also in today's Internet.  To
      achieve this it is recommended to test the algorithms with real-
      world loss and delay figures to be sure that the desired audio/
      video rates are attainable using the proposed algorithms for the
      desired services.

   Media Characteristics in Focus:

      Interactive real-time voice and video data is inherently variable.
      Usually the content of the media and service requirements dictate
      the media coding.  The codec may be bursty and not all frames are
      equally important (e.g., I-frames are more important than
      P-frames).  Thus, codecs have limited room for adaptation.
      Congestion control for audio and video codecs is therefore
      different to congestion control applied for bulk transfers.  The
      latter are allows buffering of media irrespective to their content
      and more fine-grained control for a congestion control algorithm.
      In the workshop these limitations were brought up and the workshop
      participants recommended that a congestion controller needs to be
      aware of these constraints.  However, further investigation is
      needed to decide what information needs to be exchanged between a
      codec and the congestion manager.

   Startup Behaviour:

      The startup media quality is very important for real-time
      interactive application and for user-perceived application
      performance.  The startup behavior of these is also different to
      other traffic.  By nature the real-time interactive communication
      applications want to provide a smooth user experience and maintain
      best media quality to ease the interaction.  While it may be
      desirable from a user experience point of view to immediately
      start streaming video with high-definition quality and audio of a
      wideband codec this will have impacts on the bandwidth of the



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      already ongoing flows.  As such, it would be ideal to start slow
      enough to avoid excessive congestion to other flows but fast
      enough to offer a good user experience.  The sweetspot, however,
      yet has to be found.

   In addition, there were some discussions on how to make RTCWeb-style
   flows fair to other RTCWeb-style flows, other TCP based flows, LEDBAT
   flows, and other flows.  This is a much more difficult problem than
   simply making RTCWeb flows avoid congestion.  Changing the way TCP is
   used in browsers or other HTTP-based applications would already help,
   for example, by avoid opening too many concurrent TCP connections,
   and improved interworking with video streaming and file sharing
   software.  The work on HTTP 2.0 with SPDY is already the step in the
   right direction since SPDY makes makes use of a more aggressive form
   of multiplexing instead of opening a larger number of TCP
   connections.



































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4.  Acknowledgments

   We would like to thank the participants and the paper authors of the
   position papers for their input.

   Additionally, we would like to thank the following persons for their
   review comments: Michael Welzl, John Leslie, Mirja Kuehlewind, Matt
   Mathis, Mary Barnes











































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5.  IANA Considerations

   This memo includes no request to IANA.
















































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6.  Security Considerations

   While two position papers focused on security congestion control
   security itself was not on the agenda of the workshop.  As such,
   nothing beyond the material contained in those position papers can be
   reported.













































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7.  References

7.1.  Normative References

   [RFC2914]  Floyd, S., "Congestion Control Principles", BCP 41,
              RFC 2914, September 2000.

   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405,
              November 2008.

7.2.  Informative References

   [I-D.ietf-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "RTP Congestion Control: Circuit
              Breakers for Unicast Sessions",
              draft-ietf-avtcore-rtp-circuit-breakers-00 (work in
              progress), October 2012.

































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Appendix A.  Program Committee

   This workshop was organized by Harald Alvestrand, Bernard Aboba, Mary
   Barnes, Gonzalo Camarillo, Spencer Dawkins, Lars Eggert, Matthew
   Ford, Randell Jesup, Cullen Jennings, Jon Peterson, Robert Sparks,
   and Hannes Tschofenig.













































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Appendix B.  Workshop Material

   o  Main Workshop Page:
      http://www.iab.org/activities/workshops/cc-workshop/

   o  Position Papers:
      http://www.iab.org/activities/workshops/cc-workshop/papers/

   o  Slides:
      http://www.iab.org/activities/workshops/cc-workshop/slides/









































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Appendix C.  Accepted Position Papers

   1.   "One control to rule them all" by Michael Welzl

   2.   "Congestion Avoidance Through Deterministic" by Pier Luca
        Montessoro, Riccardo Bernardini, Franco Blanchini, Daniele
        Casagrande, Mirko Loghi, and Stefan Wieser

   3.   "Congestion Control in Real Time Media - Context" by Harald
        Alvestrand

   4.   "Improving the Interactive Real-Time Video Communication with
        Network Provided Congestion Notification" by ANM Zaheduzzaman
        Sarker, Ingemar Johansson

   5.   "Multiparty Requirements in Congestion Control for Real-Time
        Interactive Media" by Magnus Westerlund

   6.   "On Fairness, Delay and Signaling of Different Approaches to
        Real-time Congestion Control" by Stefan Holmer

   7.   "RTP Congestion Control and RTCWeb Application Feedback" by Ted
        Hardie

   8.   "Issues with Using Packet Delays and Inter-arrival Times for
        Inference of Internet Congestion" by Wesley M. Eddy

   9.   "Impact of TCP on Interactive Real-Time Communication" by Ilpo
        Jarvinen, Binoy Chemmagate, Laila Daniel, Aaron Yi Ding, Markku
        Kojo, and Markus Isomaki

   10.  "Security Concerns For RTCWeb Congestion Control" by Dan York

   11.  "Vendors Considered Harmfull" by Cullen Jennings, Suhas
        Nandakumar, and Hein Phan

   12.  "Network-Assisted Dynamic Adaptation" by Xiaoqing Zhu and Rong
        Pan

   13.  "Congestion Control for Interactive Real-Time Applications" by
        Sanjeev Mehrotra, and Jin Li

   14.  "There is No Magic Transport Wand" by John Leslie

   15.  "Towards Adaptive Congestion Management for Interactive Real-
        Time Communications" by Dirk Kutscher, and Miriam Kuehlewind





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   16.  "Enlarge the pre-congestion spectrum usage?" by Xavier Marjou,
        and Emile Stephan

   17.  "Congestion control for users who don't have first-class
        internet access" by Maire Reavy

   18.  "Realtime Congestion Challenges" by Randell Jesup

   19.  "Congestion Control for Interactive Media: Control Loops & APIs"
        by Varun Singh, Joerg Ott, and Colin Perkins

   20.  "Some Notes on Threat Modelling Congestion Management" by Eric
        Rescorla

   21.  "Timely Detection of Lost Packets" by Ali C. Begen

   22.  "Congestion Control Considerations for Data Channels" by Michael
        Tuexen

   23.  "Position paper on CC for Interactive RT" by Matt Mathis

   24.  "Overall Considerations for Congestion Control" by M. Zanaty, B.
        VerSteeg, B. Christensen, D. Benham, A. Romanow

   25.  "Fairness Considerations for Congestion Control" by Mo Zanaty

   26.  "Media is not Data: The Meaning of Fairness for Competing
        Multimedia Flows" by Timothy B. Terriberry

   27.  "Thoughts on Real-Time Congestion Control" by Murari Sridharan

   28.  "Congestion Control for Interactive Real-Time Flows on Today's
        Internet" by Keith Winstein, Anirudh Sivaraman, and Hari
        Balakrishnan

   29.  "Congestion Control Principles for CoAP" by Carsten Bormann,
        Klaus Hartke

   30.  "Erasure Coding and Congestion Control for Interactive Real-Time
        Communication" by Pierre-Ugo Tournoux, Tuan Tran Thai, Emmanuel
        Lochin, Jerome Lacan, Vincent Roca

   31.  "Video Conferencing Specific Considerations for RTP Congestion
        Control" by Stephen Botzko and Mary Barnes

   32.  "The Internet is Broken, and How to Fix It" by Jim Gettys





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   33.  "Deployment Considerations for Congestion Control in Real-Time
        Interactive Media Systems" by Jari Arkko

















































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Appendix D.  Workshop Participants

   We would like to thank the following workshop participants for
   attending the workshop:

   o  Mat Ford

   o  Bernard Aboba

   o  Alissa Cooper

   o  Mary Barnes

   o  Lars Eggert

   o  Harald Alvestrand

   o  Gonzalo Camarillo

   o  Robert Sparks

   o  Cullen Jennings

   o  Dirk Kutscher

   o  Carsten Bormann

   o  Michael Welzl

   o  Magnus Westerlund

   o  Colin Perkins

   o  Murari Sridharan

   o  Klaus Hartke

   o  Pier Luca Montessoro

   o  Xavier Marjou

   o  Vincent Roca

   o  Wes Eddy

   o  Ali C. Begen





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   o  Mo Zanaty

   o  Jin Li

   o  Dave Thaler

   o  Bob Briscoe

   o  Barry Leiba

   o  Jari Arkko

   o  Stewart Bryant

   o  Martin Stiemerling

   o  Russ Housley

   o  Marc Blanchet

   o  Zaheduzzaman Sarker

   o  Xiaoqing Zhu

   o  Randell Jesup

   o  Eric Rescorla

   o  Suhas Nandakumar

   o  Hannes Tschofenig

   o  Bill VerSteeg

   o  Sean Turner

   o  Keith Winstein

   o  Jon Peterson

   o  Maire Reavy

   o  Michael Tuexen

   o  Stefan Holmer

   o  Joerg Ott




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   o  Timothy Terriberry

   o  Benoit Claise

   o  Ted Hardie

   o  Stephen Botzko

   o  Matt Mathis

   o  David Benham

   o  Jim Gettys

   o  Spencer Dawkins

   o  Sanjeev Mehrotra

   o  Adrian Farrel

   o  Greg White

   o  Markku Kojo

   We also had remote participants, namely

   o  Emmanuel Lochin

   o  Mark Handley

   o  Anirudh Sivaraman

   o  John Leslie

   o  Varun Singh
















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Authors' Addresses

   Hannes Tschofenig
   Linnoitustie 6
   Espoo,   02600
   Finland

   Phone: +358 (50) 4871445
   Email: Hannes.Tschofenig@gmx.net
   URI:   http://www.tschofenig.priv.at


   Lars Eggert
   Sonnenallee 1
   Kirchheim,   85551
   Germany

   Phone: +49 151 12055791
   Email: lars@netapp.com
   URI:   http://eggert.org/


   Zaheduzzaman Sarker
   Lulea,   SE-971 28
   Sweden

   Phone: +46 10 717 37 43
   Email: zaheduzzaman.sarker@ericsson.com























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