draft-ietf-rmcat-cc-requirements-00.txt   draft-ietf-rmcat-cc-requirements-01.txt 
Network Working Group R. Jesup Network Working Group R. Jesup
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Informational July 15, 2013 Intended status: Informational December 21, 2013
Expires: January 16, 2014 Expires: June 24, 2014
Congestion Control Requirements For RMCAT Congestion Control Requirements For RMCAT
draft-ietf-rmcat-cc-requirements-00 draft-ietf-rmcat-cc-requirements-01
Abstract Abstract
Congestion control is needed for all data transported across the Congestion control is needed for all data transported across the
Internet, in order to promote fair usage and prevent congestion Internet, in order to promote fair usage and prevent congestion
collapse. The requirements for interactive, point-to-point real time collapse. The requirements for interactive, point-to-point real time
multimedia, which needs by low-delay, semi-reliable data delivery, multimedia, which needs low-delay, semi-reliable data delivery, are
are different from the requirements for bulk transfer like FTP or different from the requirements for bulk transfer like FTP or bursty
bursty transfers like Web pages, and the TCP algorithms are not transfers like Web pages.
suitable for this traffic.
This document attempts to describe a set of requirements that can be This document attempts to describe a set of requirements that can be
used to evaluate other congestion control mechanisms in order to used to evaluate other congestion control mechanisms in order to
figure out their fitness for this purpose, and in particular to figure out their fitness for this purpose, and in particular to
provide a set of possible requirements for proposals coming out of provide a set of possible requirements for proposals coming out of
the RMCAT Working Group. the RMCAT Working Group.
This document is derived from draft-jesup-rtp-congestion-reqs This document is derived from draft-jesup-rtp-congestion-reqs
[I-D.jesup-rtp-congestion-reqs]. [I-D.jesup-rtp-congestion-reqs].
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 16, 2014. This Internet-Draft will expire on June 24, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 3
3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6 3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
4. Security Considerations . . . . . . . . . . . . . . . . . . . 7 4. Security Considerations . . . . . . . . . . . . . . . . . . . 7
5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7 5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
6. References . . . . . . . . . . . . . . . . . . . . . . . . . 7 6. References . . . . . . . . . . . . . . . . . . . . . . . . . 8
6.1. Normative References . . . . . . . . . . . . . . . . . . 7 6.1. Normative References . . . . . . . . . . . . . . . . . . 8
6.2. Informative References . . . . . . . . . . . . . . . . . 7 6.2. Informative References . . . . . . . . . . . . . . . . . 8
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 8 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 8
1. Introduction 1. Introduction
The traditional TCP congestion control requirements were developed in The traditional TCP congestion control requirements were developed in
order to promote efficient use of the Internet for reliable bulk order to promote efficient use of the Internet for reliable bulk
transfer of non-time-critical data, such as transfer of large files. transfer of non-time-critical data, such as transfer of large files.
They have also been used successfully to govern the reliable transfer They have also been used successfully to govern the reliable transfer
of smaller chunks of data in "as fast as possible" mode, such as when of smaller chunks of data in as short a time as possible, such as
fetching Web pages. when fetching Web pages.
These algorithms have also been used for transfer of media streams These algorithms have also been used for transfer of media streams
that are viewed in a non-interactive manner, such as "streaming" that are viewed in a non-interactive manner, such as "streaming"
video, where having the data ready when the viewer wants it is video, where having the data ready when the viewer wants it is
important, but the exact timing of the delivery is not. important, but the exact timing of the delivery is not.
When doing real time interactive media, the requirements are When doing real time interactive media, the requirements are
different; one needs to provide the data continuously, within a very different; one needs to provide the data continuously, within a very
limited time window (no more than 100s of milliseconds end-to-end limited time window (no more than 100s of milliseconds end-to-end
delay), the sources of data may be able to adapt the amount of data delay), the sources of data may be able to adapt the amount of data
that needs sending within fairly wide margins, and may tolerate some that needs sending within fairly wide margins, and may tolerate some
amount of packet loss, but since the data is generated in real time, amount of packet loss, but since the data is generated in real time,
sending "future" data is impossible, and since it's consumed in real sending "future" data is impossible, and since it's consumed in real
time, data delivered late is useless. time, data delivered late is useless.
One particular protocol portfolio being developed for this use case While the requirements for RMCAT differ from the requirements for the
is WebRTC [I-D.ietf-rtcweb-overview], which envisions sending other flow types, these other flow types will be present in the
network. The RMCAT congestion control algorithm must work properly
when these other flow types are present as cross traffic on the
network.
One particular protocol portofolio being developed for this use case
is WebRTC [I-D.ietf-rtcweb-overview], where one envisions sending
multiple RTP-based flows between two peers, in conjunction with data multiple RTP-based flows between two peers, in conjunction with data
flows, all at the same time, without having special arrangements with flows, all at the same time, without having special arrangements with
the intervening service providers. the intervening service providers.
Given that this use case is the focus of this document, use cases Given that this use case is the focus of this document, use cases
involving noninteractive media such as YouTube-like video streaming, involving noninteractive media such as YouTube-like video streaming,
and use cases using multicast/broadcast-type technologies, are out of and use cases using multicast/broadcast-type technologies, are out of
scope. scope.
The terminology defined in [I-D.ietf-rtcweb-overview] is used in this The terminology defined in [I-D.ietf-rtcweb-overview] is used in this
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1. The congestion control algorithm must attempt to provide as-low- 1. The congestion control algorithm must attempt to provide as-low-
as-possible-delay transit for real-time traffic while still as-possible-delay transit for real-time traffic while still
providing a useful amount of bandwidth, even when faced with providing a useful amount of bandwidth, even when faced with
intermediate bottlenecks and competing flows. There may be intermediate bottlenecks and competing flows. There may be
lower limits on the amount of bandwidth that is useful, but this lower limits on the amount of bandwidth that is useful, but this
is largely application-specific and the application may be able is largely application-specific and the application may be able
to modify or remove flows in order allow some useful flows to to modify or remove flows in order allow some useful flows to
get enough bandwidth. (Example: not enough bandwidth for low- get enough bandwidth. (Example: not enough bandwidth for low-
latency video+audio, but enough for audio-only.) latency video+audio, but enough for audio-only.)
a. It should also deal well with routing changes and interface A. It should also handle routing changes and interface changes
changes (WiFi to 3G data, etc) which may radically change (WiFi to 3G data, etc) which may radically change the
the bandwidth available. bandwidth available, and react quickly, especially if there
is a reduction in available bandwidth.
2. The algorithm must be fair to other flows, both realtime flows B. The offered load may be less than the available bandwidth at
(such as other instances of itself), and TCP flows, both long- any given moment, and may vary dramatically over time,
lived and bursts such as the traffic generated by a typical web including dropping to no load and then resuming a high load,
browsing session. Note that 'fair' is a rather hard-to-define such as in a mute operation. The reaction time between a
term. change in the bandwidth available from the algorithm and a
change in the offered load is variable, and may be different
when increasing versus decreasing.
a. The algorithm must not overreact to short-term bursts (such C. The algorithm must not overreact to short-term bursts (such
as web-browsing) which can quickly saturate a local- as web-browsing) which can quickly saturate a local-
bottleneck router or link, but also clear quickly, and bottleneck router or link, but also clear quickly, and
should recover quickly when the burst ends. This is should recover quickly when the burst ends. This is
inherently at odds with the need to react quickly-enough to inherently at odds with the need to react quickly-enough to
avoid queue buildup. avoid queue buildup.
b. We will need make some evaluation of fairness, but deciding D. Similarly periodic bursty flows such as DASH or proprietary
what is "fair" is a tough question and likely to be media streaming algorithms may compete in bursts with the
partially subjective, but we should specify some of the algorithm, and may not be adaptive within a burst. They are
inputs needed in order to select among algorithms and often are layered on top of TCP. The algorithm must avoid
tunings presented as options. too much delay buildup during those bursts, and quickly
recover. Note that this traffic may on an access link, or
may cause a shift in the location of the bottleneck fir the
duration of the burst.
c. The critical issue here is to have enough information for 2. The algorithm must be fair to other flows, both realtime flows
the WG members to decide if an algorithm is "fair", and how (such as other instances of itself), and TCP flows, both long-
"unfair" it is (to other flows or to itself) in various edge lived and bursts such as the traffic generated by a typical web
and corner cases. browsing session. Note that 'fair' is a rather hard-to-define
term.
3. The algorithm should where possible merge information across 3. The algorithm should where possible merge information across
multiple RTP streams between the same endpoints, whether or not multiple RTP streams between the same endpoints, whether or not
they're multiplexed on the same ports, in order to allow they're multiplexed on the same ports, in order to allow
congestion control of the set of streams together instead of as congestion control of the set of streams together instead of as
multiple independent streams. This allows better overall multiple independent streams. This allows better overall
bandwidth management, faster response to changing conditions, bandwidth management, faster response to changing conditions,
and fairer sharing of bandwidth with other network users. and fairer sharing of bandwidth with other network users.
Alternatively, it should work with an external bandwidth control Alternatively, it should work with an external bandwidth control
framework to coordinate bandwidth usage. framework to coordinate bandwidth usage across a bottleneck,
such as draft-welzl-rmcat-coupled-cc
[I-D.welzl-rmcat-coupled-cc].
a. If possible, it should also share information and adaptation A. If possible, it should also share information and adaptation
with other non-RTP flows between the same endpoints, such as with other non-RTP flows between the same endpoints, such as
a WebRTC data channel a WebRTC data channel
b. The most correlated bandwidth usage would be with other B. The most correlated bandwidth usage would be with other
flows on the same 5-tuple, but there may be use in flows on the same 5-tuple, but there may be use in
coordinating measurement and control of the local link(s). coordinating measurement and control of the local link(s).
C. Use of information about previous flows, especially on the
same 5-tuple, may be useful input to the algorithm,
especially to startup performance of a new flow.
4. The algorithm should not require any special support from 4. The algorithm should not require any special support from
network elements (ECN, etc). As much as possible, it should network elements (ECN, etc). As much as possible, it should
leverage existing information about the incoming flows to leverage available information about the incoming flow to
provide feedback to the sender. Examples of this information provide feedback to the sender. Examples of this information
are the packet arrival times, acknowledgments and feedback, are the ECN, packet arrival times, acknowledgments and feedback,
packet timestamps, packet sizes, packet losses. Extra packet timestamps, and packet losses; all of these can provide
information could be added to the packets to provide more information about the state of the path and any bottlenecks.
detailed information on actual send times (as opposed to
sampling times), but should not be required.
a. When additional input signals such as ECN are available, A. Extra information could be added to the packets to provide
more detailed information on actual send times (as opposed
to sampling times), but should not be required.
B. When additional input signals such as ECN are available,
they should be utilized if possible. they should be utilized if possible.
5. Since the assumption here is a set of RTP streams, the 5. Since the assumption here is a set of RTP streams, the
backchannel typically should be done via RTCP; the alternative backchannel typically should be done via RTCP; one alternative
would be to include it in a reverse RTP channel using header would be to include it instead in a reverse RTP channel using
extensions. header extensions.
a. In order to react sufficiently quickly, the AVPF/SAVPF RTP A. In order to react sufficiently quickly when using RTCP for a
profile[RFC4585] MUST be used backchannel, an RTP profile such as AVPF/SAVPF that allows
sufficiently frequent feedback [RFC4585] MUST be used.
b. Note that in some cases, backchannel messages may be delayed B. Note that in some cases, backchannel messages may be delayed
until the RTCP channel can be allocated enough bandwidth, until the RTCP channel can be allocated enough bandwidth,
even under AVPF rules. This may also imply negotiating a even under AVPF rules. This may also imply negotiating a
higher maximum percentage for RTCP data or allowing RMCAT higher maximum percentage for RTCP data or allowing RMCAT
solutions to violate or modify the rules specified for AVPF. solutions to violate or modify the rules specified for AVPF.
c. Note that RTCP is of course unreliable C. Bandwidth for the feedback messages should be minimized
d. Bandwidth for the feedback messages should be minimized
(such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR) (such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR)
e. Header extensions would avoid the RTCP timing rules issues, D. Header extensions would avoid the RTCP timing rules issues,
and allow the application to allocate bandwidth as needed and allow the application to allocate bandwidth as needed
for the congestion algorithm. for the congestion algorithm.
f. Backchannel data should be minimized to avoid taking too E. Backchannel data should be minimized to avoid taking too
much reverse-channel bandwidth (since this will often be much reverse-channel bandwidth (since this will often be
used in a bidirectional set of flows). In areas of used in a bidirectional set of flows). In areas of
stability, backchannel data may be sent more infrequently so stability, backchannel data may be sent more infrequently so
long as algorithm stability and fairness are maintained. long as algorithm stability and fairness are maintained.
When the channel is unstable or has not yet reached When the channel is unstable or has not yet reached
equilibrium after a change, backchannel feedback may be more equilibrium after a change, backchannel feedback may be more
frequent and use more reverse-channel bandwidth. This is an frequent and use more reverse-channel bandwidth. This is an
area with considerable flexibility of design, and different area with considerable flexibility of design, and different
approaches to backchannel messages and frequency are approaches to backchannel messages and frequency are
expected to be evaluated. expected to be evaluated.
6. Where possible and helpful, the algorithm should leverage and 6. Flows managed by this algorithm and flows competed against at a
piggyback on other RTP/RTCP communications, such as SR/RR, rctp- bottleneck may have different DSCP markings depending on the
fb PLI, RPSI, SLI or application-specific NACK messages (such as type of traffic. A particular bottleneck or section of the
for loss information), and also reverse-direction RTP. network path may or may not honor these markings.
A. In WebRTC, a division of packets into 4 classes is
envisioned in order of priority: faster-than-audio, audio,
video, best-effort, and bulk-transfer. Typically the flows
managed by this algorithm would be audio or video in that
heirarchy, and feedback flows would be faster-than-audio.
7. The algorithm should sense the unexpected lack of backchannel 7. The algorithm should sense the unexpected lack of backchannel
information as a possible indication of a channel overuse information as a possible indication of a channel overuse
problem and react accordingly to avoid burst events causing a problem and react accordingly to avoid burst events causing a
congestion collapse. congestion collapse.
8. It should attempt to avoid bandwidth 'collapse' when facing a 8. It should attempt to avoid bandwidth 'collapse' when facing a
long-lived saturating TCP flow or flows. (I.e. a classic delay- long-lived saturating TCP flow or flows. (I.e. a classic delay-
sensitive algorithm will reduce bandwidth to keep delay down sensitive algorithm will reduce bandwidth to keep delay down
until the TCP flow has all the bandwidth). See the Cx-TCP until the TCP flow has all the bandwidth). See the Cx-TCP
algorithm discussed in a recent Transactions On Networking algorithm discussed in a recent Transactions On Networking
[cx-tcp] for an example of a delay-sensitive congestion-control [cx-tcp] for an example of a delay-sensitive congestion-control
algorithm that transitions to a loss-based mode when competing algorithm that transitions to a loss-based mode when competing
with TCP flows - at the cost of increased delay. with TCP flows - at the cost of increased delay.
9. The algorithm should be stable and low-delay when faced with 9. The algorithm should be stable and low-delay when faced with
active queue management (AQM) such as RED [RFC2309] or CoDel active queue management (AQM) algorithms. Also note that these
[I-D.nichols-tsvwg-codel] or fq-codel in the channel. algorithms may apply across multiple queues in the bottleneck,
or to a single queue
10. The algorithm should quickly adapt to initial network conditions 10. The algorithm should quickly adapt to initial network conditions
at the start of a flow. This should occur both if the initial at the start of a flow. This should occur both if the initial
bandwidth is above or below the bottleneck bandwidth. bandwidth is above or below the bottleneck bandwidth.
a. The startup adaptation may be faster than adaptation later A. The startup adaptation may be faster than adaptation later
in a flow. It should allow for both slow-start operation in a flow. It should allow for both slow-start operation
(adapt up) and history-based startup (start at a point (adapt up) and history-based startup (start at a point
expected to be at or below channel bandwidth from historical expected to be at or below channel bandwidth from historical
information, which may need to adapt down quickly if the information, which may need to adapt down quickly if the
initial guess is wrong). Starting too low and/or adapting initial guess is wrong). Starting too low and/or adapting
up too slowly can cause a critical point in a personal up too slowly can cause a critical point in a personal
communication to be poor ("Hello!"). communication to be poor ("Hello!"). Starting over-
bandwidth causes other problems for user experience, so
b. Starting over-bandwidth causes other problems for user there's a tension here.
experience, so there's a tension here.
c. Alternative methods to help startup like probing during B. Alternative methods to help startup like probing during
setup with dummy data may be useful in some applications. setup with dummy data may be useful in some applications; in
some cases there will be a considerable gap in time between
flow creation and the initial flow of data.
d. A flow may need to change adaptation rates due to network C. A flow may need to change adaptation rates due to network
conditions or changes in the provided flows (such as un- conditions or changes in the provided flows (such as un-
muting or sending data after a gap). muting or sending data after a gap).
11. It should be evaluated in how it works both with backbone-router 11. It should be evaluated in how it works both with backbone-router
bottlenecks, (asymmetric) local-loop bottlenecks, and local-lan bottlenecks, (asymmetric) local-loop bottlenecks, and local-lan
(WiFi/etc) bottlenecks, and in competition with varying numbers (WiFi/etc) bottlenecks, and in competition with varying numbers
and types of streams (TCP, TCP variants in use, LEDBAT and types of streams (TCP, TCP variants in use, LEDBAT
[I-D.ietf-ledbat-congestion], inflexible VoIP UDP flows). [I-D.ietf-ledbat-congestion], inflexible VoIP UDP flows).
12. It should be stable if the RTP streams are halted or 12. It should be stable if the RTP streams are halted or
discontinuous (VAD/DTX). discontinuous (VAD/DTX).
a. After a resumption of RTP data it may adapt more quickly A. After a resumption of RTP data it may adapt more quickly
(similar to the start of a flow), and previous bandwidth (similar to the start of a flow), and previous bandwidth
estimates may need to be aged or thrown away. estimates may need to be aged or thrown away.
3. IANA Considerations 3. IANA Considerations
This document makes no request of IANA. This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an Note to RFC Editor: this section may be removed on publication as an
RFC. RFC.
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This document is the result of discussions in various fora of the This document is the result of discussions in various fora of the
WebRTC effort, in particular on the rtp-congestion@alvestrand.no WebRTC effort, in particular on the rtp-congestion@alvestrand.no
mailing list. Many people contributed their thoughts to this. mailing list. Many people contributed their thoughts to this.
6. References 6. References
6.1. Normative References 6.1. Normative References
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower- Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-06 (work based Applications", draft-ietf-rtcweb-overview-08 (work
in progress), February 2013. in progress), September 2013.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006. 2006.
6.2. Informative References 6.2. Informative References
skipping to change at page 8, line 8 skipping to change at page 8, line 35
Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
"Low Extra Delay Background Transport (LEDBAT)", draft- "Low Extra Delay Background Transport (LEDBAT)", draft-
ietf-ledbat-congestion-10 (work in progress), September ietf-ledbat-congestion-10 (work in progress), September
2012. 2012.
[I-D.jesup-rtp-congestion-reqs] [I-D.jesup-rtp-congestion-reqs]
Jesup, R. and H. Alvestrand, "Congestion Control Jesup, R. and H. Alvestrand, "Congestion Control
Requirements For Real Time Media", draft-jesup-rtp- Requirements For Real Time Media", draft-jesup-rtp-
congestion-reqs-00 (work in progress), March 2012. congestion-reqs-00 (work in progress), March 2012.
[I-D.nichols-tsvwg-codel] [I-D.welzl-rmcat-coupled-cc]
Nichols, K. and V. Jacobson, "Controlled Delay Active Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion
Queue Management", draft-nichols-tsvwg-codel-01 (work in control for RTP media", draft-welzl-rmcat-coupled-cc-02
progress), February 2013. (work in progress), October 2013.
[RFC2309] Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,
S., Wroclawski, J., and L. Zhang, "Recommendations on
Queue Management and Congestion Avoidance in the
Internet", RFC 2309, April 1998.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. and Consequences", RFC 5506, April 2009.
[cx-tcp] Budzisz, L., Stanojevic, R., Schlote, A., Baker, F., and [cx-tcp] Budzisz, L., Stanojevic, R., Schlote, A., Baker, F., and
R. Shorten, "On the Fair Coexistence of Loss- and Delay- R. Shorten, "On the Fair Coexistence of Loss- and Delay-
Based TCP", December 2011. Based TCP", December 2011.
Author's Address Author's Address
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