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A Model-Based Approach to Evaluation of the Efficacy of FEC Coding in Combating Network Packet Losses

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Abstract

We propose a model-based analytic approach for evaluating the overall efficacy of FEC coding combined with interleaving in combating packet losses in IP networks. In particular, by modeling the network path in terms of a single bottleneck node, described as a G/M/1/K queue, we develop a recursive procedure for the exact evaluation of the packet-loss statistics for general arrival processes, based on the framework originally introduced by Cidon et al., 1993. To include the effects of interleaving, we incorporate a discrete-time Markov chain (DTMC) into our analytic framework. We study both single-session and multiple-session scenarios, and provide a simple algorithm for the more complicated multiple-session scenario. We show that the unified approach provides an integrated framework for exploring the tradeoffs between the key coding parameters; specifically, interleaving depths, channel coding rates and block lengths. The approach facilitates the selection of optimal coding strategies for different multimedia applications with various user quality-of-service (QoS) requirements and system constraints. We also provide an information-theoretic bound on the performance achievable with FEC coding in IP networks.
A Model-Based Approach to Evaluation of the
Efficacy of FEC Coding in Combating Network
Packet Losses
VENKATA GIRI.MADDIPATI(*)
GIET,RAJAHMUNDRY
DEPT.OF.COMPUTERSCINCE &ENGG,
maddipati.venkatagiri@gmail.com
RAJASEKHAR.SWARNA
GIET,RAJAHMUNDRY
DEPT.OF.COMPUTERSCINCE &ENGG,
swarnarajasekhar@gmail.com
Abstract—We propose a model-based analytic approach for evaluating the overall efficacy of FEC coding
combined with interleaving in combating packet losses in IP networks. In particular, by modeling the
network path in terms of a single bottleneck node, described as a queue, we develop a recursive procedure
for the exact evaluation of the packet-loss statistics for general arrival processes, based on the framework
originally introduced by Cidon et al., 1993. To include the effects of interleaving, we incorporate a discrete-
time Markov chain (DTMC) into our analytic framework. We study both single-session and multiple-session
scenarios, and provide a simple algorithm for the more complicated multiple-session scenario. We show that
the unified approach provides an integrated framework for exploring the tradeoffs between the key coding
parameters; specifically, interleaving depths, channel coding rates and block lengths. The approach facilitates
the selection of optimal coding strategies for different multimedia applications with various user quality-of-
service (QoS) requirements and system constraints. We also provide an information-theoretic bound on the
performance achievable with FEC coding in IP networks.
Index Terms—Autocorrelation function, FEC coding, interleaving, packet-loss processes, residual packet-loss
rates,
single-multiplexer model.
I. INTRODUCTION
THE packet transport service provided by
representative packet-switched networks, including
IP networks, is not reliable and the quality-of-service
(QoS) cannot be guaranteed. Packets may be lost due
to buffer overflow in switching nodes, be discarded
due to excessive bit errors and failure to pass the
cyclic redundancy check (CRC) at the link layer, or
be discarded by network control mechanisms as a
response to congestion somewhere in the network.
Forward error correction (FEC) coding has often
been proposed for end-to-end recovery from such
packet losses. However, the use of FEC in this
application provides a double-edged sword. From an
end user’s perspective, FEC can help recover the lost
packets in a timely fashion through the use of
redundant packets, and generally adding more
redundancy can be expected to improve performance
provided this added redundancy does not adversely
affect the network packet loss characteristics. On the
other hand, from the network’s perspective, the
widespread use of FEC schemes by end nodes will
increase the raw packet-loss rate in a network
because of the additional loads resulting from
transmission of redundant packets. Therefore, in
order to optimize the end-to-end performance, the
appropriate tradeoff, in terms of the amount of
redundancy added, and its effect on network packet-
loss processes, needs to be investigated under
specific and realistic modeling assumptions.
In this paper,we provide a study of the
overall effectiveness of packet-level FEC coding,
employing interlaced Reed-Solomon codes, in
combating network packet losses and provide an
information- theoretic methodology for determining
the optimum compromise between end-to-end
performance and the associated increase in raw
packet loss rates using a realistic model-based
analytic approach. Intuitively, for a given choice of
block length we expect that there is an optimum
choice of redundancy, or channel coding rate, since a
rate too high (low redundancy) is simply not
powerful enough to effectively recover packet losses
while a rate too low (high redundancy) results in
excessive raw packet losses due to the increased
overhead which overwhelms the packet recovery
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capabilities of the FEC code. The optimum channel
coding rate results in an optimum compromise
between these two effects.
In order to analytically investigate this
tradeoff, we consider a simplified network scenario
described in terms of a single bottleneck node,
modeled as a multiplexer. For networks with reliable
transmission media, where transmission errors are
negligible, buffer overflows due to congestion in
routers are the major cause of packet losses. In a
packet-switched network, a flow of packets crosses a
chain of routers before it reaches the destination
node. Most of the packet losses from a flow occur in
the router which has the smallest bandwidth.
Therefore, we can model the whole chain of routers
in terms of this single bottleneck node. Both
theoretical analysis and experimental results justify
this assumption [1]–[3].Asingle-multiplexer model
for this bottleneck node is widely used to analyze the
associated queueing-related packet losses, e.g., losses
due to buffer overflows and excessive delays. Since
the correlation level of the packet-loss process has
great impact on the FEC efficacy, we investigate this
dependence using the autocorrelation function of the
packet-loss process.
The performance of FEC in recovering
network packet losses has been studied in many
papers [4]–[13]. In [4], the use of redundant parity
packets was proposed to reconstruct lost data packets
and the corresponding performance evaluation
indicated that residual packet-loss rates can be
reduced up to three orders of magnitude. However, in
[4] for analysis purposes the packet-loss process
resulting from the single-multiplexer model was
assumed to be independent and, consequently, the
simulation results provided show that this simplified
analysis
considerablly overestimates the performance of FEC.
By modeling the single-multiplexer as an or queue,
Cidon et al. [5] proposed a recursive algorithm to
compute the packet-loss statistics (block errror
density), through which the exact residual packet-loss
rate after decoding was computed. Surprisingly, all
numerical results given in [5] indicate that the
resulting residual packet-loss rates with coding are
always greater than without coding, i.e., FEC is
ineffective in this application. However, in [5] only a
single parity packet is used1 and the block length was
constrained to the range . As we show later, these
conclusions are somewhat misleading and result from
inappropriate parameter choices. In [6]–[8], more
general arrival processes were considered and coding
tradeoffs assessed but performance results were
obtained using large deviation bounds to characterize
the packet-loss processes and were not exact. In [9]
[12],
Altman et al. derived analytical
formulas for the frame-loss probabilities for the
single-multiplexer model using multi-dimensional
probability generating functions and show that the
frame loss probabilities can be reduced if a
sufficiently large amount of redundancy is added.
However, contrary to [4], [5], in these works the
authors used the frame-loss probability as the
evaluation metric for FEC performance, because it
was assumed that the failure to recover any lost data
packet will lead to the loss of all the data packets in
that block. For ATM networks, this assumption is
valid since the loss of a single cell does result in the
discarding of the whole message. However, for other
networks, like IP networks, this is not the case. In IP
networks, packet-level FEC coding can be performed
across several IP packets. Even if any lost data
packets cannot be recovered, the correctly received
data packets in the same coding block may still be
useful. For example, in the application of video over
IP, when some lost video packets cannot be
recovered, the correctly received video packets in the
same coding block need not be discarded and can
even be used to estimate the information in lost video
packets using an appropriate error concealment
scheme. As shown in Section II, in many cases, the
FEC performance predicted by using the frame-loss
probability is not only quantitatively different, but
also qualitatively different, from that reflected by the
packet-loss probability. Although the asymptotic
analysis (when block size goes to infinity) for the
frame-loss rates in [9]–[12] can provide some insight
into FEC performance on IP networks, the
methodology and specific conclusions developed in
these works are not useful for a comprehensive
evaluation of FEC effectiveness on IP networks since
they are based on an inappropriate evaluation metric.
II. PRELIMINARIES
A. Single-Multiplexer Network Model
If a network’s performance is limited by a single
bottleneck node, the network can often be modeled in
terms of a single multiplexer. As illustrated in Fig. 1,
the single-multiplexer model is a queueing system
which consists of three components: 1) an arrival
process for packets from different sources with
corresponding packet arrival rates , ; 2) a buffer
which can hold up to packets, which are assumed
served in first-come-first-served (FCFS) order; and 3)
an output link with average packet service rate . We
assume if a packet finds a full buffer upon arrival, it
will be discarded. For analytical convenience, we
assume the packet service times are independent and
identically distributed (i.i.d.) with an exponential
distribution and average packet service time .With
denoting the overall packet arrival rate, the
normalized load to the system is (1) Although the
assumption of exponential service time may not be
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accurate in some situations, it enables us to obtain
some analytical results and gain some insight on the
effects of packet-loss processes due to buffer
overflows.
B. Source Model
We assume the packet arrival process for each source
is a renewal process, i.e., for any source the packet
interarrival times are i.i.d. with arbitrary probability
density function . In particular, we consider the case
of the Erlang interarrival time distribution When
(assuming is fixed), the variance of converges to 0
and the interarrival time becomes deterministic with
period . With appropriate parameter choice, this case
can represent constant bit rate (CBR) sources. For the
hyperexponential distribution, the average arrival rate
C. System Model for FEC Performance
Evaluation
Consider the communication system model illustrated
in Fig. 2. We suppose there are homogeneous and
independent sources sharing the single-multiplexer
and each source generates packets with average rate .
The FEC coder for each source applies an interlaced
Reed-Solomon code [6], [8], [15] to the packets from
the source, which means for every block of source
information packets it creates an additional parity
packets to the network. The channel coding rate is
given by . As a result of the channel coding, the
packet arrival rate into the network will increase to .
Let the random variable denote the number of lost
packets within a block. If , we assume all the lost
packets within that block can be recovered by the
channel decoder. Assume denotes the block-error
distribution, i.e., the probability that packets out of
are lost. Therefore, the expected number of lost
packets within a block is (6)
and the expected number of lost information packets
within a block is (7) Finally, the effective information
packet loss rate after channel decoding is (8)
D. Evaluation Metric: Packet-Loss Probability or
Frame-Loss Probability?
In some networks, such as ATM networks, the failure
to recover a single packet results in the loss of the
entire frame (block). In this case, the frame-loss
probability is a more suitable metric than packet-loss
probability for evaluation of FEC performance, as
used in [9]–[12]. The frame-loss probability is given
by (9) Figs. 3 and 4 illustrate comparisons of the FEC
performance predicted by packet-loss probability
with that predicted by
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Fig. 3. Performance difference between coding and
without coding as a function of the number of
information packets in each frame, single session ��
Other system parameters are as follows: system load
buffer size , 1 parity packet used
Fig. 4. Performance difference between coding and
without coding as a function of the number of
information packets in each frame, multiple session
�� _ __. Other system parameters are as follows:
system load _ _ ___, buffer size _ ___, 1 parity
packet used.
frame-loss probability. The curves show the
differences of the residual packet-loss probabilities
and frame-loss probabilities with coding and without
coding, denoted by and , respectively. Fig. 3 shows
the case of a single session and Fig. 4 shows the case
of multiple sessions . Other system parameters are
indicated in the corresponding figure captions. When
one of the differences is positive (i.e., or ), this
implies that using redundant packets does not
improve the corresponding performance metric ( or );
that is, use of FEC is ineffective. From Figs. 3 and 4
it should be clear, at least for the cases considered,
these two performance metrics can lead to totally
different conclusions, both quantitatively and
qualitatively, concerning the effectiveness of FEC. In
particular, in either case the results indicate that, for
the corresponding choice of system parameters, the
use of FEC is ineffective for all block lengths based
on but can provide improved performance based on
provided the block length is chosen large enough. As
described above, in this paper we focus on IP
networks and use packet-loss probability to evaluate
the FEC performance.
E. Autocorrelation Function of Packet-Loss
Processes
For a packet-loss process we use the autocorrelation
function to characterize the dependence between the
packet-loss events over time. Let the random
sequence represent the packet- loss process, with 1
denoting loss and 0 denoting reception. If is
stationary, then the autocorrelation function of is
given as (10) where is the lag and is the expectation
of the sequence
III. FEC PERFORMANCE WITH A SINGLE
SOURCE
A. FEC Without Interleaving
We begin our analysis with the simplest case: there
is only one user for the multiplexer . As (8)
illustrates, the key quantity in evaluating the residual
packet-loss rate after FEC decoding is , the block-
error distribution for an arbitrary number of
consecutive packets. In [5], Cidon et al. propose a
recursive algorithm to compute for the finite buffer
queue with Poisson arrivals and exponential service
times, denoted as the queue. In order to analyze the
packet losses for more general arrival patterns, in
what follows we first describe the extension of the
algorithm to the queue, i.e., the finite buffer queue
with general i.i.d. interarrival times and exponential
service times. 1) Analysis of Block-Error
Distribution: Suppose there is only one source
sharing the multiplexer , and the . packet interarrival
times are i.i.d. with arbitrary probability density
function . Since the packet service times are assumed
i.i.d. and exponentially distributed, the single-
multiplexer can be modeled as a standard queueing
system. Let be the number of packets in the buffer
just before the th packet arrives at the system.
Because of the memoryless property of the
exponential service time, is a discrete-time Markov
chain (DTMC) [16, pg. 249], illustrated in Fig. 5,
with the state space and state-transition matrix
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where , , is the probability that packets are served out
of the system during an arbitrary interarrival time,
provided that there are at least packets in the system
at the beginning of that interarrival time. It can be
shown [16, pg. 248]n that (12) subject to a
normalization constraint. Let be the stationary
distribution of the DTMC, which can be obtained as
the solution to (13) Define , , , , to be the probability
of losses in a block of packets, given that there are
packets in the system just before the first packet of
the block arrives to the system. So we have (14)
where is determined from (13).
B. FEC With Block Interleaving
FEC performance is often limited by the bursty
nature of typical packet-loss processes, and block
interleaving techniques are frequently used to reduce
the burstiness of the packet-loss processes in
networks [4], thereby improving FEC performance.
In this section, we analyze the efficacy of
interleaving in reducing the burstiness of network
packet-loss processes and in improving the FEC
performance.
1) Interleaving Operation: The operation of block
interleaving is illustrated in Figs. 12 and 13. Before
being transmitted into the network, packets are filled
into an matrix row-wise and then read-out from the
matrix column wise. Therefore, the originally
consecutive packets will be packets apart from each
other after interleaving. is called the interleaving
depth (ILDP). At the receiver the packets will be
reordered in the deinterleaver before decoding. In
what follows we analyze the corresponding block-
error distribution incorporating the interleaving
operation.
2) Analysis of Block-Error Distribution: As shown
in Fig. 12, we again assume the packet arrival process
at the single multiplexer (after interleaving) is a
renewal process, i.e., the packet interarrival time at
the multiplexer is i.i.d. with some common
probability density function . Let denote the number
of packets in the buffer just before the th packet
arrives at the multiplexer (Point in Fig. 12). Again,
we assume the packet service times are i.i.d. with an
exponential distribution and the average service rate
is . Therefore, the random sequence is a discrete-time
Markov chain with the state space , transition-
probability matrix and stationary probability as
expressed in (11) and (13), respectively. Define
another random sequence , where is the number of
packets in the buffer just before the th packet, seen by
the interleaver (Point in Fig. 12), arrives at the single-
multiplexer. Since after interleaving the originally
consecutive packets will be packets apart from each
other, is formed by selecting every th element of .
Therefore,is also a discrete-time Markov chain with
the state space , and the corresponding transition-
probability matrix , is the –step transition matrix of ,
(19) Let be the stationary distribution of , which can
be obtained as the solution to (20) subject to a
normalization constraint. Then, proceeding with the
same procedure as in Section III-A1, we can obtain
recursive expressions for computing . If an ideal
interleaver is applied, then the packet-loss process is
independent. Suppose the average packet-loss rate is ,
then the packet-loss statistics with ideal interleaving
are given by the binomial distribution (21) Once we
obtain the block-error distribution incorporating the
interleaving operation, the effective packet-loss rate
after FEC decoding can be obtained from (8). In
Appendix A, we determine the autocorrelation
function for packet-loss processes associated with the
single-multiplexer model (modeled as a queue) with
interleaving. Next we show some numerical
examples.
3) Numerical Examples: Fig. 14 illustrates the
efficacy of interleaving in reducing packet-loss
correlation associated with the single-multiplexer
model. In particular, we illustrate the packet-loss
autocorrelation function as a functionof lag for the
case in which the arrivals are Poisson with the
average loadand the buffer size is . It shows, as
expected,that interleaving can effectively reduce the
correlation of the packet-lo ss process, which means
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that interleaving can make the packet losses more
independent and isolated.
POTENTIAL OF FEC AND AN INFORMATION-
THEORETIC BOUND
In Fig. 17, we demonstrated that, with the same
packet-loss rate requirement, FEC coding with a
larger block size can support increased source traffic.
However, the source traffic that can be supported is
not unlimited because of the channel capacity
limitation imposed by the single-multiplexer
transport channel model. In what follows we develop
an information-theoretic upper bound on the FEC
performance based on the single-multiplexer network
model. In this section, we only consider the case of a
single source , although the approach can be extended
to arbitrary .
A. Channel Model for Packet Transmission Over
Networks
Consider a channel model for packet transmission
over a general packet-switched network. Assume a
packet has bits. It is either transmitted and received
by the receiver, or is lost due to network congestion
or buffer overflow. For a received packet, bit errors
may be introduced. Then packet transmission over
networks can be modeled for coding purpose in terms
of serial bit-by-bit transmission of -bit symbols either
over a binary symmetrical channel (BSC) with
crossover probability Fig. 23. Component channels of BIC
corresponding to packet delivery and loss. Fig. 24. Simplified
communication system model. (state 0) or over a binary
erasure channel (BEC) (state 1), both of which are
illustrated in Fig. 23, where is used to indicate the
erasure symbol.A lost packet corresponds to the
entire codeword symbol of bits being erased, while a
received packet means each of the bits is sequentially
transmitted over the BSC. This channel model
belongs to the class of block interference channels
(BIC), introduced by McEliece and Stark [14]. Let
represent the state space of the BIC. If the state
transitions are independent, then the Shannon
capacity of the BIC is given as [14], (26) where is the
capacity of the component channel , and the
expectation is over the state space . It follows
that(27) where is the probability of being in the loss
state and isthe binary entropy function, (28)
B. Information-Theoretic Bound on FEC
Performance
Referring to Fig. 12, suppose the interleaving is ideal,
and consequently the packet-loss process seen by the
channel decoder is independent. If we consider the
interleaver and the deinterleaver as components of
the coding channel, then the channel, consisting of
the interleaver, the single-multiplexer and the
deinterleaver, can be modeled as a BICwith
independent state transitions, as ill ustrated in Fig. 24.
Here we consider only the packet losses caused by
the buffer overflows, and assume no bit errors, i.e.,
the BSC crossover probability . Let be the packetloss
rate of thesingle-multiplexer, so . Then, from (27),
the capacity of the BIC is given by (29) Assume the
source creates packets at rate and the packetservice
rate is . Then the normalized system load before
coding is . The channel encoder applies channel
coding (notnecessarily RS codes) with coding rate to
thesource traffic.
CONCLUSIONS AND FUTURE WORK
We have analyzed the efficacy of FEC in combating
network packet losses based on a single-multiplexer
network model and demonstrated that FEC has great
potential in recovering the packet losses caused by
congestion at a bottleneck node of a packet-switched
network, provided that the coding rate and other
coding parameters are appropriately chosen. We
developed a discrete-time Markov chain model to
analyze the efficacy of interleaving in improving the
FEC performance and determined how much
interleaving depth is required for FEC to approach
the optimum performance. We derived an upper
bound on the end-to-end performance using FEC
based
on an information-theoretic methodology, which is
useful in predicting source rates that can be supported
with arbitrarily high reliability. Despite the great
potential of FEC coding in recovering network packet
losses, the implementation complexity of FEC coding
and the corresponding coding/decoding delay also
need
to be considered, which is an issue particularly
important for real-time applications. One objective
for future work is the analysis of the additional delay
caused by the FEC coding, perhaps combined with
interleaving/deinterleaving. Likewise, the application
of FEC for network transport is limited by the time-
varying and often uncertain error characteristics of
the channel, which makes the appropriate choice of
FEC coding rate difficult to determine. In real world
applications, FEC cod ers are required which can
adapt the channel code rate to the time-varying
channel conditions. This issue is also a topic for
future work.
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... However, the redundancy introduces significant overhead that increases latency and reduces the available bandwidth. Moreover, network congestion often leads to burst of losses [32], for which FEC is often ineffective or even detrimental [36]. Although FEC eliminates the retransmissions, most recent works [30][31][32] on FEC for cloud gaming apply FEC on the entire GoP, which either significantly increase the motion-to-photon latency (all GoP frames must be decoded before display), demand higher bandwidth, or decreases the visual quality. ...
... The bandwidth often shrinks when congestion occurs, leading to burst traffic loss [32]. Injecting more FEC-based redundant packets is ineffective to cope with congestion losses in the presence of burstiness [36]. As such, we introduce the source rate controller to control both the frequency of capturing the rendered frames and video encoding rate . ...
Conference Paper
Full-text available
Mobile cloud gaming enables high-end games on constrained devices by streaming the game content from powerful servers through mobile networks. Mobile networks suffer from highly variable bandwidth, latency, and losses that affect the gaming experience. This paper introduces Nebula, an end-to-end cloud gaming framework to minimize the impact of network conditions on the user experience. Nebula relies on an end-to-end distortion model adapting the video source rate and the amount of frame-level redundancy based on the measured network conditions. As a result, it minimizes the motion-to-photon (MTP) latency while protecting the frames from losses. We fully implement Nebula and evaluate its performance against the state-of-the-art techniques and latest research in real-time mobile cloud gaming transmission on a physical testbed over emulated and real wireless networks. Nebula consistently balances MTP latency (<140 ms) and visual quality (>31dB) even in highly variable environments. A user experiment confirms that Nebula maximizes the user experience with high perceived video quality, playability, and low user load.
... However, the redundancy introduces significant overhead that increases latency and reduces the available bandwidth. Moreover, network congestion often leads to burst of losses [30], for which FEC is often ineffective or even detrimental [34]. Although FEC minimizes retransmission latency, most current works [28][29][30] on FEC for cloud gaming apply FEC on the entire GoP, which either significantly increase the motion to photon delay (all GoP frames must be decoded before display), demand higher bandwidth, or decreases the visual quality. ...
... The bandwidth often shrinks when congestion occurs, leading to burst traffic loss [30]. Injecting more FEC-based redundant packets is ineffective to cope with congestion losses in the presence of burstiness [34]. As such, we introduce the source rate controller to control both the frequency of capturing the rendered frames and video encoding rate . ...
Preprint
Full-text available
Mobile cloud gaming enables high-end games on constrained devices by streaming the game content from powerful servers through mobile networks. Mobile networks suffer from highly variable bandwidth, latency, and losses that affect the gaming experience. This paper introduces Nebula, an end-to-end cloud gaming framework to minimize the impact of network conditions on the user experience. Nebula relies on an end-to-end distortion model adapting the video source rate and the amount of frame-level redundancy based on the measured network conditions. As a result, it minimizes the motion-to-photon (MTP) latency while protecting the frames from losses. We fully implement Nebula and evaluate its performance against the state of the art techniques and latest research in real-time mobile cloud gaming transmission on a physical testbed over emulated and real wireless networks. Nebula consistently balances MTP latency (<140 ms) and visual quality (>31 dB) even in highly variable environments. A user experiment confirms that Nebula maximizes the user experience with high perceived video quality, playability, and low user load.
... There are many transmission techniques for robust reliable delivery of the video data. Reed-Solomon (RS) codes have been utilized for packet protection in video streaming solutions [2][3][4][5][6][7][8][9][10][11]. To apply the unequal FEC framework for motion-compensated singlelayer coding such as MPEG-4 simple profile, the distortion model for packets with different picture types in group-of-pictures (GOP) is investigated. ...
... In [3], a simple packet distortion model is designed such that the distortion is proportional to the length of error propagation (LEP), and it is adopted for weighted fair queuing in Internet routers. The rate-distortion information in different layers of MPEG-4 fine grain scalable video is considered for unequal protection strategy [4][5][6][7][8][9][10]. The distortion drift by the loss of quality and temporal layers is estimated as performance metrics based on distortion model for FEC assignment [5,6,8,11]. ...
Article
Full-text available
In this paper, we propose an unequal Luby transform (LT) based on block weight shift (ULT-BWS) method as an unequal forward error correction method to minimize video distortion over packet-lossy networks. First, we consider unequal amount of error propagation effects from packet loss in hierarchical prediction structure to give unequal property in an LT codes. For robust video transmission over various channel status, the ULT-BWS method assigns an efficient amount of protection for frame blocks with different error propagation weights by controlling the range of more important blocks in a group of pictures. Simulation results demonstrate that the proposed ULT-BWS method gives robust performance and significantly improved video quality, compared with the conventional ULT schemes.
... The packet loss sequence within the channel was generated using the Gilbert-Elliot (GE) model. 26 The average burst packet loss length was set to 3.5, based on the realistic measurement results reported in Yu et al. 27 At the receiver node, the FEC-UDP sink module performs the FEC decoding process and records the reconstructed packets in the receiver file. Based on the receiver file, the Evalvid tool converts the received packets into video frames for video decoding purposes. ...
Article
Forward error correction (FEC) techniques are widely used to recover packet losses over unreliable networks in real-time video streaming applications. Traditional frame-level FEC encodes 1 video frame in each FEC coding window. By contrast, in the expanding-window FEC scheme, high-priority frames are included in the FEC processing of the following frames, so as to construct a larger coding window. In general, expanding-window FEC improves the recovery performance of FEC, because the high-priority frame can be protected by multiple windows and the use of a larger coding window increases the efficiency. However, the larger window size also increases the complexity of the coding and the memory space requirements. Consequently, expanding-window FEC is limited in terms of practical applications. Sliding-window FEC adopts a fixed window size in order to approximate the performance of the expanding-window FEC method, but with a reduced complexity. Previous studies on sliding-window FEC have generally adopted an equal error protection (EEP) mechanism to simplify the analysis. This paper considers the more practical case of an unequal error protection (UEP) strategy. An analytical model is derived for estimating the playable frame rate (PFR) of the proposed sliding-window FEC scheme with a Reed-Solomon erasure code for real-time non-scalable streaming applications. The analytical model is used to determine the optimal FEC configuration which maximizes the PFR value under given transmission rate constraints. The simulation results show that the proposed sliding-window scheme achieves almost the same performance as the expanding-window scheme, but with a significantly lower computational complexity.
... We use the symbol F i,j (θ) to express the probability of transition from state i to j in time interval θ F i,j (θ) = P[X (θ) = j|X (0) = i]. (18) According to the property of continuous time Markov chain, we have the following state transition matrix: ...
Article
Cloud gaming has emerged as a promising application to enable high-end game playing with thin clients. Transmission control protocol (TCP) has been pervasively adopted as the transport-layer protocol in the mainstream cloud gaming systems for video communication. However, streaming mobile cloud gaming video using TCP is challenged with several key technical barriers: 1) the performance limitations of wireless networks in bandwidth and reliability; 2) the high throughput demand and stringent delay constraint imposed by high-quality gaming video transmission; 3) the deadline violations and throughput fluctuations caused by the packet retransmission and congestion control mechanisms in TCP. To address these critical problems, this research proposes an application-layer source-FEC (Forward Error Correction) coding framework dubbed ESCOT (adaptivE Source-FEC COding over TCP). First, we analytically formulate the optimization problem of joint source-FEC coding to minimize the end-to-end distortion of real-time video communication over TCP. Second, we develop a heuristic solution for effective loss rate approximation, source rate control, and FEC coding adaptation. ESCOT is distinct from existing source-FEC coding schemes in proactively analyzing and leveraging the TCP characteristics. The proposed solution is able to effectively mitigate both consecutive and sporadic video frame drops caused by congestion and random packet losses. We conduct the performance evaluation through extensive emulations in the Exata platform using real-time gaming video encoded by H.264 codec. Experimental results show that ESCOT advances the state-of-the-art with noticeable improvements in video PSNR (Peak Signal-to-Noise Ratio), end-to-end delay, goodput, and frame success rate.
Article
Recently, federated learning (FL) has received tremendous attention in both academia and industry, in which decentralized clients collaboratively complete model training by exchanging model updates with a parameter server through the Internet. Its distributed nature well utilizes the localized data and preserves clients' privacy, but also incurs heavy communication overhead. Existing studies on model update have mostly focused on the bandwidth constraint of the communication channels. Today's Internet however is highly unreliable. Simply using Transmission Control Protocol (TCP) would lead to low network utilization under frequent losses. In this paper, we closely examine the optimal transmission strategies in FL over the realistic lossy Internet. We systematically integrate model compression, forward error correction (FEC) and retransmission towards Federated Learning with Lossy Communications (FedLC). We derive the convergence rate of FedLC under non-convex loss with the optimal transmission. We then decompose this non-convex problem and present effective practical solutions. Public datasets are exploited for performance evaluation by varying the packet loss rate from 10% to 50%. In a fixed training time budget, FedLC can improve model accuracy by 3.91% on average or reduce the communication traffic by 34.27%-47.57% in comparison with state-of-the-art baselines.
Article
Video surveillance has become an important Internet multimedia application to provide live streaming services. Energy efficiency is critical to guarantee the service time and quality of power-intensive video streaming on wireless surveillance systems. In particular, video coding and data communication constitute the majority of power dissipation in embedded multimedia devices driven by capacity-limited batteries. However, the complex power characteristics and time-varying channel status pose crucial challenges on enabling low-power high-quality video streaming. To address these critical problems, this paper presents a Power-Efficient and Network-Adaptive (PENA) framework for source-FEC (Forward Error Correction) coding in embedded systems. First, an analytical framework is developed to characterize the rate-distortion-power tradeoff for video encoding and data transfer. Second, a joint rate control and FEC coding solution is proposed to maximize video quality under power constraint. Distinct from the existing source-FEC coding algorithms, PENA is able to effectively leverage the video rate-distortion model and system power features. We conduct the system implementation and performance evaluation with real embedded surveillance devices over wireless networks. Experimental results demonstrate PENA achieves appreciable improvements over the reference source-FEC coding schemes in terms of power conservation, perceived video quality and end-to-end delay.
Article
Burst packet loss is a common problem over wired and wireless networks and leads to a significant reduction in the performance of packet-level forward error correction (FEC) schemes used to recover packet losses during transmission. Traditional FEC interleaving methods adopt the sequential coding-interleaved transmission (SCIT) process to encode the FEC packets sequentially and reorder the packet transmission sequence. Consequently, the burst loss effect can be mitigated at the expense of an increased end-to-end delay. Alternatively, the reversed interleaving scheme, namely, interleaved coding-sequential transmission (ICST), performs FEC coding in an interleaved manner and transmits the packets sequentially based on their generation order in the application. In this study, the analytical FEC model is constructed to evaluate the performance of the SCIT and ICST schemes. From the analysis results, it can be observed that the interleaving delay of ICST FEC is reduced by transmitting the source packets immediately as they arrive from the application. Accordingly, an Enhanced ICST scheme is further proposed to trade the saved interleaving time for a greater interleaving capacity, and the corresponding packet loss rate can be minimized under a given delay constraint. The simulation results show that the Enhanced ICST scheme achieves a lower packet loss rate and a higher peak signal-to-noise-ratio than the traditional SCIT and ICST schemes for video streaming applications.
Conference Paper
Unequal error protection (UEP) on streaming media broadcasting is a widely used technique over wireless environment. Forward error protection (FEC) with UEP is a prevailing scheme to improve the quality of video streaming over lossy channels. Frame-level FEC have been proposed for video streaming due to the priorities of video frames within transmission constraint on Bernoulli channels. However, in current Internet architecture, various communication and storage systems are prone to corrupt as burst of noises occurs. The corrupted data pieces may exceed the error correction capacity of FEC, and therefore degrade the efficacy of FEC. To address this problem, we proposed a model to evaluate the perceived quality of H.264/AVC video streaming over burst channels. The experiment results indicate that the proposed scheme is able to evaluate the video streaming quality in perspective of decodable frame rate (DFR) with lower complexity than alternative methods.
Article
We study forward error correction (FEC) that reduces loss probabilities of messages, based on adding redundant packets and interleaving, as proposed in [9]. Without interleaving, losses occur due to a locality phenomenon: If a packet is lost then the probability of another loss closely after that may be significantly larger than the probability of a loss much later; Thus losses tend to cluster. We study FEC with interleaving using different approaches: Ballot theorems, recursions and a purely algebraic approach, and complement it with numerical investigation.
Article
A new class of channel models with memory is presented in order to study various kinds of interference phenomena. It is shown, among other things, that when all other parameters are held fixed, channel capacityCis an {em increasing} function of the memory length, while the cutoff rateR_{0}generally is a {em decreasing} function. Calculations with various explicit coding schemes indicate thatCis better thanR_{0}as a performance measure for these channel models. As a partial resolution of thisCversusR_{0}paradox, the conjecture is offered thatR_{0}is more properly a measure of coding delay rather than of coding complexity.
Conference Paper
We study the loss probabilities of messages in an M/M/1/K queueing system where in addition to losses due to buffer overflow there are random losses on the incoming and outgoing links. We focus on the influence of adding redundant packets to the messages. We obtain analytical results that allow us to investigate when does adding redundancy decrease the loss probabilities.
Conference Paper
The use of FEC coding, possibly in conjuction with ARQ techniques, has emerged as a promising approach for video transport over ATM networks with/without wireless links for cell recovery and/or bit error recovery. Although FEC provides cell loss recovery capabilities it also introduces overhead which can possibly cause more cell losses. A methodology based on information-theoretic limits is described to maximize the number of sources multiplexed by using FEC codes
Article
The purpose of this paper is to study the loss probabilities of messages in an M/M/1/K queueing system where in addition to losses due to buffer overflow there are also random losses in the incoming and outgoing links. We focus on the influence of adding redundant packets to the messages (as in error correction coding, e.g. Reed–Solomon code, etc.). In the first part we use multi-dimensional probability generating functions for solving the recursions which generalize those introduced by Cidon et al. [IEEE Trans. Inform. Theory 39 (1) (1993) 98] for computing the loss probabilities and derive analytical formulae for a special case. In the second part of the paper we use combinatorial arguments and Ballot theorem results to alternatively obtain the loss probabilities. The analytical results allow us to investigate when does adding redundancy decrease the loss probabilities.
Article
We study the effect of adding redundancy to an input stream on the losses that occur due to buffer overflow. We consider several sessions that generate traffic into a finite capacity queue. Using multi-dimensional probability generating functions, we derive analytical formulas for the loss probabilities and provide asymptotic analysis (for large n and small or large ρ). Our analysis allows us to investigate when does adding redundancy decrease the loss probabilities. In many cases, redundancy is shown to degrade the performance, as the gain in adding redundancy is not sufficient to compensate the additional losses due to the increased overhead. We show, however, that it is possible to decrease loss probabilities if a sufficiently large amount of redundancy is added. Indeed, we show that for an arbitrary stationary ergodic input process, if ρ<1 then redundancy can reduce loss probabilities to an arbitrarily small value.