Internet DRAFT - draft-zanaty-rmcat-app-interaction
draft-zanaty-rmcat-app-interaction
RMCAT WG M. Zanaty
Internet-Draft Cisco
Intended status: Informational V. Singh
Expires: January 5, 2015 Aalto University
S. Nandakumar
Cisco
Z. Sarker
Ericsson AB
July 4, 2014
RTP Application Interaction with Congestion Control
draft-zanaty-rmcat-app-interaction-01
Abstract
Interactive real-time media applications that use the Real-time
Transport Protocol (RTP) over the User Datagram Protocol (UDP) must
use congestion control techniques above the UDP layer since it
provides none. This memo describes the interactions and conceptual
interfaces necessary between the application components that relate
to congestion control, including the RTP layer, the higher-level
media codec control layer, and the lower-level transport interface,
as well as components dedicated to congestion control functions.
Status of This Memo
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This Internet-Draft will expire on January 5, 2015.
Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Key Words for Requirements . . . . . . . . . . . . . . . . . 3
3. Conceptual Model . . . . . . . . . . . . . . . . . . . . . . 4
4. Implementation Model . . . . . . . . . . . . . . . . . . . . 5
5. Interfaces and Interactions . . . . . . . . . . . . . . . . . 6
5.1. Config - Codec Interactions . . . . . . . . . . . . . . . 6
5.2. Config - RTP/RTCP Interactions . . . . . . . . . . . . . 6
5.3. Codec - RTP Interactions . . . . . . . . . . . . . . . . 6
5.4. Codec - CC Interactions . . . . . . . . . . . . . . . . . 7
5.5. RTP - CC Interactions . . . . . . . . . . . . . . . . . . 9
5.6. CC - UDP Interactions . . . . . . . . . . . . . . . . . . 9
5.7. CC - Shared State Interactions . . . . . . . . . . . . . 10
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 10
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 10
8. Security Considerations . . . . . . . . . . . . . . . . . . . 10
9. References . . . . . . . . . . . . . . . . . . . . . . . . . 11
9.1. Normative References . . . . . . . . . . . . . . . . . . 11
9.2. Informative References . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 13
1. Introduction
Interactive real-time media applications most commonly use RTP
[RFC3550] over UDP [RFC0768]. Since UDP provides no form of
congestion control, which is essential for any application deployed
on the Internet, these RTP applications have historically implemented
one of the following options at the application layer to address
their congestion control requirements.
1. For media with relatively low packet rates and bit rates, such as
many speech codecs, some applications use a simple form of
congestion control that stops transmission permanently or
temporarily after observing significant packet loss over a
significant period of time, similar to the RTP circuit breakers
[I-D.ietf-avtcore-rtp-circuit-breakers].
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2. Some applications have no explicit congestion control, despite
the clear requirements in RTP and its profiles AVP [RFC3551] and
AVPF [RFC4585], under the expectation that users will terminate
media flows that are significantly impaired by congestion (in
essence, human circuit breakers).
3. For media with substantially higher packet rates and bit rates,
such as many video codecs, various non-standard congestion
control techniques are often used to adapt transmission rate
based on receiver feedback.
4. Some experimental applications use standardized techniques such
as TCP-Friendly Rate Control (TFRC) [RFC5348]. However, for
various reasons, these have not been widely deployed.
The RTP Media Congestion Avoidance Techniques (RMCAT) working group
was chartered to standardize appropriate and effective congestion
control for RTP applications. It is expected such applications will
migrate from the above historical solutions to the RMCAT solution(s).
The RMCAT requirements [I-D.ietf-rmcat-cc-requirements] include low
delay, reasonably high throughput, fast reaction to capacity changes
including routing or interface changes, stability without over-
reaction or oscillation, fair bandwidth sharing with other instances
of itself and TCP flows, sharing information across multiple flows
when possible [I-D.welzl-rmcat-coupled-cc], and performing as well or
better in networks which support Active Queue Management (AQM),
Explicit Congestion Notification (ECN), or Differentiated Services
Code Points (DSCP).
In order to meet these requirements, interactions are necessary
between the application's congestion controller, the RTP layer, media
codecs, other components, and the OS UDP stack. This memo discusses
these interactions, presents a conceptual model of the required
interfaces based on a simplified application decomposition, and
proposes specific information exchange across these interfaces along
with corresponding component behavior.
Note that RTP can also operate over other transports with integrated
congestion control such as TCP [RFC5681] and DCCP [RFC4340], but that
is beyond the scope of RMCAT and this memo.
2. Key Words for Requirements
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 [RFC2119].
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3. Conceptual Model
It is useful to decompose an RTP application into several components
to facilitate understanding and discussion of where congestion
control functions operate, and how they interface with the other
components. The conceptual model in Figure 1 consists of the
following components.
+----------------------------+
| +-----Config-----+ |
| | | | |
| | Codec | |
| | | | | | |
| APP +---RTP | RTCP---+ |
| | | | | |
| | | | | |
| +---Congestion-------|---Shared
| Control | State
+----------------------------+
|
+----------------------------+
| OS UDP |
+----------------------------+
Figure 1
o APP: Application containing one or more RTP streams and the
corresponding media codecs and congestion controllers. For
example, a WebRTC browser.
o Config: Configuration specified by the application that provides
the media and transport parameters, RTP and RTCP parameters and
extensions, and congestion control parameters. For example, a
WebRTC Javascript application may use the 'constraints' API to
affect the media configuration, and SDP applications may negotiate
the media and transport parameters with the remote peer. This
determines the initial static configuration negotiated in session
establishment. The dynamic state may differ due to congestion or
other factors, but still must conform to limits established in the
config.
o Codec: Media encoder/decoder or other source/sink for the RTP
payload. The codec may be, for example, a simple monaural audio
format, a complex scalable video codec with several dependent
layers, or a source/sink with no live encoding/decoding such as a
mixer which selectively switches and forwards streams rather than
mixes media.
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o RTP: Standard RTP stack functions, including media packetization /
depacketization and header processing, but excluding existing
extensions and possible new extensions specific to congestion
control (CC) such as absolute timestamps or relative transmission
time offsets in RTP header extensions. RTCP: Standard RTCP
functions, including sender reports, receiver reports, extended
reports, circuit breakers [I-D.ietf-avtcore-rtp-circuit-breakers],
feedback messages such as NACK [RFC4585] and codec control
messages such as TMMBR [RFC5104], but excluding existing
extensions and possible new extensions specific to congestion
control (CC) such as REMB [I-D.alvestrand-rmcat-remb] (for
receiver-side CC), ACK (for sender-side CC), absolute and/or
relative timestamps (for sender-side or receiver-side CC), etc.
o Congestion Control: All functions directly responsible for
congestion control, including possible new RTP/RTCP extensions,
send rate computation (for sender-side CC), receive rate
computation (for receiver-side CC), other statistics, and control
of the UDP sockets including packet scheduling for traffic
shaping/pacing.
o Shared State: Storage and exchange of congestion control state for
multiple flows within the application and beyond it.
o OS: Operating System containing the UDP socket interface and other
network functions such as ECN, DSCP, physical interface events,
interface-level traffic shaping and packet scheduling, etc.
4. Implementation Model
There are advantages and drawbacks to implementing congestion control
in the application layer. It avoids OS dependencies and allows for
rapid experimentation, evolution and optimization for each
application. However, it also puts the burden on all applications,
which raises the risks of improper or divergent implementations. One
motivation of this memo is to mitigate such risks by giving proper
guidance on how the application components relating to congestion
control should interact.
Another drawback of congestion control in the application layer is
that any decomposition, including the one presented in Figure 1, is
purely conceptual and illustrative, since implementations have
differing designs and decompositions. Conversely, this can be viewed
as an advantage to distribute congestion control functions wherever
expedient without rigid interfaces. For example, they may be
distributed within the RTP/RTCP stack itself, so the separate
components in Figure 1 are combined into a single RTP+RTCP+CC
component as shown in Figure 2.
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+----------------------------+
| +-----Config |
| | | |
| | Codec |
| APP | | |
| +---RTP+RTCP+CC------|---Shared
+----------------------------+ State
|
+----------------------------+
| OS UDP |
+----------------------------+
Figure 2
The conceptual model in Figure 1 will be used throughout this memo to
establish clearer boundaries between functions. But actual
implementations may be closer to the looser model in [Singh12].
5. Interfaces and Interactions
5.1. Config - Codec Interactions
The primary interactions between the config and the codec that are
relevant to congestion control are the multiplexing of media streams
[I-D.ietf-mmusic-sdp-bundle-negotiation] and RTP/RTCP [RFC5761] on
the same UDP port.
The config also establishes limits for the codec such as maximum bit
rate and other codec-specific parameters. For example, a video codec
config often sets limits on maximum resolution and frame rate as well
as bit rate.
5.2. Config - RTP/RTCP Interactions
The config establishes the negotiated RTP and RTCP attributes and
extensions such as Extended Reports (XR), reduced size [RFC5506],
codec control [RFC5104], transmission time [RFC5450], etc.
5.3. Codec - RTP Interactions
Packetization of codec frames into RTP packets can be an important
interaction. Some network interfaces may benefit from small packet
sizes well below the MTU, while others may benefit from large packets
approaching the MTU. Equalizing packet sizes of a frame may also be
beneficial in some cases, rather than a combination of large and
small packets. For example, in some FEC schemes, the FEC bandwidth
overhead depends on the largest source packet size. Equalizing the
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source packet sizes can yield lower overhead than a combination of
large and small packets.
5.4. Codec - CC Interactions
Allowed Rate (from CC to Codec): The max transmit rate allowed over
the next time interval. The time interval may be specified or may
use a default, for example, one second. The rate may be specified in
bytes or packets or both. The rate must never exceed permanent
limits established in session signaling such as the SDP bandwidth
attribute [RFC4566] nor temporary limits in RTCP such as TMMBR
[RFC5104] or REMB [I-D.alvestrand-rmcat-remb]. This is the most
important interface among all components, and is always required in
any RMCAT solution. In the simplest possible solution, it may be the
only CC interface required.
Media Elasticity (from Codec to CC): Many live media encoders are
highly elastic, often able to achieve any target bit rate within a
wide range, by adapting the media quality. For example, a video
encoder may support any bit rate within a range of a few tens or
hundreds of kbps up to several Mbps, with rate changes registering as
fast as the next video frame, although there may be limitations in
the frequency of changes. Other encoders may be less elastic,
supporting a narrower rate range, coarser granularity of rate steps,
slower reaction to rate changes, etc. Other media, particularly some
audio codecs, may be fully inelastic with a single fixed rate. CC
can beneficially use codec elasticity, if provided, to plan Allowed
Rate changes, especially when there are multiple flows sharing CC
state and bandwidth.
Startup Ramp (from Codec to CC, and from CC to Codec): Startup is an
important moment in a conversation. Rapid rate adaptation during
startup is therefore important. The codec should minimize its
startup media rate as much as possible without adversely impacting
the user experience, and support a strategy for rapid rate ramp. The
CC should allow the highest startup media rate as possible without
adversely impacting network conditions, and also support rapid rate
ramp until stabilizing on the available bandwidth. Startup can be
viewed as a negotiation between the codec and the CC. The codec
requests a startup rate and ramp, and the CC responds with the
allowable parameters which may be lower/slower. The RMCAT
requirements also include the possibility of bandwidth history to
further accelerate or even eliminate startup ramp time. While this
is highly desirable from an application viewpoint, it may be less
acceptable to network operators, since it is in essence a gamble on
current congestion state matching historical state, with the
potential for significant congestion contribution if the gamble was
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wrong. Note that startup can often commence before user interaction
or conversation to reduce the chance of clipped media.
Delay Tolerance (from Codec to CC): An ideal CC will always minimize
delay and target zero. However, real solutions often need a real
non-zero delay tolerance. The codec should provide an absolute delay
tolerance, perhaps expressed as an impairment factor to mix with
other metrics.
Loss Tolerance (from Codec to CC): An ideal CC will always minimize
packet loss and target zero. However, real solutions often need a
real non-zero loss tolerance. The codec should provide an absolute
loss tolerance, perhaps expressed as an impairment factor to mix with
other metrics. Note this is unrecoverable post-repair loss after
retransmission or forward error correction.
Throughput Sensitivity (from Codec to CC): An ideal CC will always
maximize throughput. However, real solutions often need a trade-off
between throughput and other metrics such as delay or loss. The
codec should provide throughput sensitivity, perhaps expressed as an
impairment factor (for low throughputs) to mix with other metrics.
Rate Stability (from Codec to CC): The CC algorithm must strike a
balance between rate stability and fast reaction to changes in
available bandwidth. The codec should provide its preference for
rate stability versus fast and frequent reaction to rate changes,
perhaps expressed as an impairment factor (for high rate variance
over short timescales) to mix with other metrics.
Forward Error Correction (FEC): Simple FEC schemes like XOR Parity
codes [RFC5109] may not handle consecutive or burst loss well. More
complex FEC schemes like Reed-Solomon [RFC6865] or Raptor [RFC6330]
codes are more effective at handling bursty loss. The sensitivity to
packet loss therefore depends on the media (source) encoding as well
as the FEC (channel) encoding, and this sensitivity may differ for
different loss patterns like random, periodic, or consecutive loss.
Expressing this sensitivity to the congestion controller may help it
choose the right balance between optimizing for throughput versus low
loss.
Probing for Available Bandwidth: FEC can also be used to probe for
additional available bandwidth, if the application desires a higher
target rate than the current rate. FEC is preferable to synthetic
probes since any contribution to congestion by the FEC probe will not
impact the post-repair loss rate of the source media flow while
synthetic probes may adversely affect the loss rate [Nagy14]. Note
that any use of FEC or retransmission must ensure that the total flow
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of all packets including FEC, retransmission and original media never
exceeds the Allowed Rate.
5.5. RTP - CC Interactions
RTP Circuit Breakers: The intent behind RTP circuit breakers
[I-D.ietf-avtcore-rtp-circuit-breakers] is to provide a kill switch
of last resort, not true congestion control. The breakers should
never trip when an effective congestion control is operating. This
may impose some boundaries on RMCAT solutions to ensure the
congestion control never approaches situations which may trigger the
breakers.
RTCP Feedback: The primary method of communicating CC information is
RTCP.
RTP Header Extensions: While RTCP is likely to be the primary carrier
of CC feedback, the RMCAT requirements also include the possibility
of using RTP header extensions in bidirectional flows for CC
feedback. Transmission time [RFC5450], or possibly absolute time,
also use header extensions, as would any per packet priority markings
expected to survive across different networks and administrative
domains.
5.6. CC - UDP Interactions
Pacing / Shaping: Simple pacing / shaping strategies delay the
transmission of packets to equalize inter-packet time intervals,
assuming the bottleneck is most sensitive to packet rate. More
complex pacing strategies may go beyond simple even distribution of
transmission times. For example, Sprout [Winstein13] attempts to
predict the optimal transmission time (and rate) with the highest
probability of success for each packet based on channel statistics.
Pacing may be always on, or adaptively enabled / disabled based on
congestion state to minimize delay. Pacing may be performed within
the CC for a single flow or across multiple flows. It may also be
performed across all or selective traffic over the network interface
if the OS supports interface-level traffic shaping.
Detection of Transport Capabilities: The CC can query the OS for
useful transport capabilities such as ECN, DSCP, traffic shaping,
etc. This may also aid upper layers in making better decisions such
as whether or not to multiplex media streams. For example, if audio
can be given differentiated network treatment from video when using
separate ports.
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ECN: If the OS and transport path support ECN, the CC can react
faster than a loss-based CC and more reliably to congestion onset and
abatement.
DSCP: If the OS and transport path support DSCP, the CC can map per-
packet priority from RTP header extensions to DSCP (and layer 2 QoS
if available) for better network handling under congestion.
AQM: If AQM is present in the bottleneck, and working effectively,
there should be little or no excess delay observed when varying the
transmission rate. The loss of such delay signals may hinder the
performance of congestion control algorithms that are highly
dependent on delay variation for adapting transmission rate. If the
application has knowledge of the presence of AQM, through any means
which are beyond the scope of this memo, it should communicate this
to the CC. The CC may use this to alter its signal collection and
rate adaptation strategies. The CC must not rely solely on this as a
reliable indicator. It must continue to monitor statistics to
validate this application hint, and react appropriately if the
statistics suggest different network behavior.
5.7. CC - Shared State Interactions
Multiple Flows: Sharing state across multiple flows within the
application can yield better CC decisions. Sharing state across even
more flows beyond the application can yield even better CC decisions.
The actual benefits and mechanisms of state sharing and coupled CC
are described in [I-D.welzl-rmcat-coupled-cc].
Weighted Fairness: An important consideration in CC of multiple flows
is their relative application-specified weights. Within an
application, it is likely the different flows have different rate
requirements, so equal bandwidth sharing may not be fair nor
desirable, and weighted fairness may be required.
6. Acknowledgements
The RMCAT design team discussions contributed to this memo.
7. IANA Considerations
This memo includes no request to IANA.
8. Security Considerations
Amplification attacks often use UDP traffic to launch denial of
service attacks. Attackers may attempt to subvert congestion control
protocols in UDP applications to launch amplification attacks by
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signaling more bandwidth than is actually available. For example,
sending a victim a forged REMB or a few fast ACKs may result in the
victim sending a high rate RTP stream. Attacks on conference servers
could lead to further amplification if it distributes the streams to
many others. One mitigation is to use SRTCP for congestion control
messages where supported. Even if SRTCP is only authenticated not
encrypted, SRTCP packets should always pass authentication checks
before any message contents are interpreted. Non-secure RTCP should
be avoided where possible.
9. References
9.1. Normative References
[I-D.alvestrand-rmcat-remb]
Alvestrand, H., "RTCP message for Receiver Estimated
Maximum Bitrate", draft-alvestrand-rmcat-remb-03 (work in
progress), October 2013.
[I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", draft-ietf-
avtcore-rtp-circuit-breakers-05 (work in progress),
February 2014.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-07 (work in progress), April 2014.
[I-D.ietf-rmcat-cc-requirements]
Jesup, R., "Congestion Control Requirements For RMCAT",
draft-ietf-rmcat-cc-requirements-04 (work in progress),
April 2014.
[I-D.ietf-rmcat-eval-criteria]
Singh, V. and J. Ott, "Evaluating Congestion Control for
Interactive Real-time Media", draft-ietf-rmcat-eval-
criteria-01 (work in progress), March 2014.
[I-D.welzl-rmcat-coupled-cc]
Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion
control for RTP media", draft-welzl-rmcat-coupled-cc-03
(work in progress), May 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, March 2009.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
9.2. Informative References
[Nagy14] Nagy, M., Singh, V., Ott, J., and L. Eggert, "Congestion
Control using FEC for Conversational Multimedia
Communication", Proc. of 5th ACM Internation Conference on
Multimedia Systems , 3 2014.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340, March 2006.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
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[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC
5348, September 2008.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009.
[RFC6330] Luby, M., Shokrollahi, A., Watson, M., Stockhammer, T.,
and L. Minder, "RaptorQ Forward Error Correction Scheme
for Object Delivery", RFC 6330, August 2011.
[RFC6865] Roca, V., Cunche, M., Lacan, J., Bouabdallah, A., and K.
Matsuzono, "Simple Reed-Solomon Forward Error Correction
(FEC) Scheme for FECFRAME", RFC 6865, February 2013.
[Singh12] Singh, V., Ott, J., and C. Perkins, "Congestion Control
for Interactive Media: Control Loops & APIs", Proc. of
IAB/IRTF Workshop on Congestion Control for Interactive
RTC , 7 2012.
[Winstein13]
Winstein,, K., Sivaraman,, A., and H. Balakrishnan,
"Stochastic Forecasts Achieve High Throughput and Low
Delay over Cellular Networks", Proc. of the 10th USENIX
Symposium on Networked Systems Design and Implementation ,
4 2013.
Authors' Addresses
Mo Zanaty
Cisco
Raleigh, NC
USA
Email: mzanaty@cisco.com
Varun Singh
Aalto University
Espoo, FIN
Finland
Email: varun@comnet.tkk.fi
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Suhas Nandakumar
Cisco
San Jose, CA
USA
Email: snandaku@cisco.com
Zaheduzzaman Sarker
Ericsson AB
Luleae
Sweden
Email: zaheduzzaman.sarker@ericsson.com
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