Internet DRAFT - draft-ietf-pwe3-congcons
draft-ietf-pwe3-congcons
PWE3 YJ. Stein
Internet-Draft RAD Data Communications
Intended status: Informational D. Black
Expires: January 25, 2015 EMC Corporation
B. Briscoe
BT
July 24, 2014
Pseudowire Congestion Considerations
draft-ietf-pwe3-congcons-02
Abstract
Pseudowires (PWs) have become a common mechanism for tunneling
traffic, and may be found in unmanaged scenarios competing for
network resources both with other PWs and with non-PW traffic, such
as TCP/IP flows. It is thus worthwhile specifying under what
conditions such competition is safe, i.e., the PW traffic does not
significantly harm other traffic or contribute more than it should to
congestion. We conclude that PWs transporting responsive traffic
behave as desired without the need for additional mechanisms. For
inelastic PWs (such as TDM PWs) we derive a bound under which such
PWs consume no more network capacity than a TCP flow. We also
propose employing a transport circuit breaker
[I-D.fairhurst-tsvwg-circuit-breaker] that shuts down a TDM PW
consistently surpassing this bound, as the emulated TDM service
itself would be be of insufficient quality.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
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This Internet-Draft will expire on January 25, 2015.
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Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. PWs Comprising Elastic Flows . . . . . . . . . . . . . . . . 4
3. PWs Comprising Inelastic Flows . . . . . . . . . . . . . . . 5
4. Security Considerations . . . . . . . . . . . . . . . . . . . 16
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 16
6. Informative References . . . . . . . . . . . . . . . . . . . 17
Appendix A. Loss Probabilities for TDM PWs . . . . . . . . . . . 18
Appendix B. Effect of Packet Loss on Voice Quality for TDM PWs . 19
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 22
1. Introduction
A pseudowire (PW)(see [RFC3985]) is a construct for tunneling a
native service, such as Ethernet or TDM, over a Packet Switched
Network (PSN), such as IPv4, IPv6, or MPLS. The PW packet
encapsulates a unit of native service information by prepending the
headers required for transport in the particular PSN (which must
include a demultiplexer field to distinguish the different PWs) and
preferably the 4 byte PWE3 control word.
PWs have no bandwidth reservation or control mechanisms, meaning that
when multiple PWs are transported in parallel, and/or in parallel
with other flows, there is no defined means for allocating resources
for any particular PW, or for preventing negative impact of a
particular PW on neighboring flows. Mechanisms for provisioning PWs
in service provider networks are well understood and will not be
discussed further here.
While PWs are most often placed in MPLS tunnels, there are several
mechanisms that enable transporting PWs over an IP infrastructure.
These include:
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UDP/IP encapsulations defined for TDM PWs
([RFC4553][RFC5086][RFC5087]),
L2TPv3 based PWs,
MPLS PWs directly over IP according to RFC 4023 [RFC4023],
MPLS PWs over GRE over IP according to RFC 4023 [RFC4023].
Whenever PWs are transported over IP, they may compete for network
resources with neighboring congestion-responsive flows (e.g., TCP
flows). In this document we study the effect of PWs on such
neighboring flows, and discover that the negative impact of PW
traffic is generally no worse than that of congestion-responsive
flows, ([RFC2914],[RFC5033]}.
At first glance one may consider a PW transported over IP to be
considered as a single flow, on a par with a single TCP flow. Were
we to accept this tenet, we would require a PW to back off under
congestion to consume no more bandwidth than a single TCP flow under
such conditions (see [RFC5348]). However, since PWs may carry
traffic from many users, it makes more sense to consider each PW to
be equivalent to multiple TCP flows.
The following two sections consider PWs of two types.
Elastic Flows: Section 2 concludes that the response to congestion
of a PW carrying elastic (e.g., TCP) flows is no different to the
combined behaviour of the set of the same elastic flows were they
not encapsulated within a PW.
Inelastic Flows: Section 3 considers the case of inelastic constant
bit-rate (CBR) TDM PWs ([RFC4553][RFC5086] [RFC5087]) competing
with TCP flows. Such PWs require a preset amount of bandwidth,
that may be lower or higher than that consumed by an otherwise
unconstrained TCP flow under the same network conditions. In any
case, such a PW is inable to respond to congestion in a TCP-like
manner; on the other hand, the total bandwidth it consumes remains
constant and does not increase to consume additional bandwidth as
TCP rates back off. If the bandwidth consumed by a TDM PW is
considered detrimental, the only available remedy is to completely
shut down the PW, by using a transport circuit breaker mechanism.
However, we will show that even before such an action is
warranted, the PW will become unable to faithfully emulate the
native TDM service; for example, when a TDM service is carrying
voice grade telephony channels, the voice quality will degrade to
below useful levels.
Thus, in both cases, pseudowires will not inflict significant harm on
neighboring TCP flows, as in one case they respond adequately to
congestion, and in the other they would be shut down due to being
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unable to emulate the native service before harming neighboring
flows.
2. PWs Comprising Elastic Flows
In this section we consider Ethernet PWs that primarily carry
congestion-responsive traffic. We show that we automatically obtain
the desired congestion avoidance behavior, and that additional
mechanisms are not needed.
Let us assume that an Ethernet PW aggregating several TCP flows is
flowing alongside several TCP/IP flows. Each Ethernet PW packet
carries a single Ethernet frame that carries a single IP packet that
carries a single TCP segment. Thus, if congestion is signaled by an
intermediate router dropping a packet, a single end-user TCP/IP
packet is dropped, whether or not that packet is encapsulated in the
PW.
The result is that the individual TCP flows inside the PW experience
the same drop probability as the non-PW TCP flows. Thus the behavior
of a TCP sender (retransmitting the packet and appropriately reducing
its sending rate) is the same for flows directly over IP and for
flows inside the PW. In other words, individual TCP flows are
neither rewarded nor penalized for being carried over the PW. An
elastic PW does not behave as a single TCP flow, as it will consume
the aggregated bandwidth of its component flows; yet if its component
TCP flows backs off by some percentage, the bandwidth of the PW as a
whole will be reduced by the very same percentage, purely due to the
combined effect of its component flows.
This is, of course, precisely the desired behavior. Were individual
TCP flows rewarded for being carried over a PW, this would create an
incentive to create PWs for no operational reason. Were individual
flows penalized, there would be a deterrence that could impede
pseudowire deployment.
There have been proposals to add additional TCP-friendly mechanisms
to PWs, for example by carrying PWs over DCCP. In light of the above
arguments, it is clear that this would force the PW down to the
bandwidth of a single flow, rather than N flows, and penalize the
constituent TCP flows. In addition, the individual TCP flows would
still back off due to their end points being oblivious to the fact
that they are carried over a PW. This would further degrade the
flow's throughput as compared to a non-PW-encapsulated flow, in
contradiction to desirable behavior.
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3. PWs Comprising Inelastic Flows
Inelastic PWs, such as TDM PWs ([RFC4553][RFC5086][RFC5087]), are
potentially more problematic than the elastic PWs of the previous
section. Being constant bit-rate (CBR), TDM PWs can not be made
responsive to congestion. On the other hand, being CBR, they also do
not attempt to capture additional bandwidth when neighboring TCP
flows back off.
Since a TDM PW continuously consumes a constant amount of bandwidth,
if the bandwidth occupied by a TDM PW endangers the network as a
whole, the only recourse is to shut it down, denying service to all
customers of the TDM native service. We can accomplish this by
employing a transport circuit breaker, by which we mean an automatic
mechanism for terminating a flow to prevent negative impact on other
flows and on the stability of the network
[I-D.fairhurst-tsvwg-circuit-breaker]. Note that a transport circuit
breaker is intended as a protection mechanism of last resort, just as
an electrical circuit breaker is only triggered when absolutely
necessary. We should mention in passing that under certain
conditions it may be possible to reduce the bandwidth consumption of
a TDM PW. A prevalent case is that of a TDM native service that
carries voice channels that may not all be active. Using the AAL2
mode of [RFC5087] (perhaps along with connection admission control)
can enable bandwidth adaptation, at the expense of more sophisticated
native service processing (NSP).
In the following we will show that for many cases of interest a TDM
PW, treated as a single flow, will behave in a reasonable manner
without any additional mechanisms. We will focus on structure-
agnostic TDM PWs [RFC4553] although our analysis can be readily
applied to structure-aware PWs (see Appendix A).
In order to quantitatively compare TDM PWs to TCP flows, we will
compare the effect of TDM PW packets with that of TCP packets of the
same packet size and sent at the same rate. This is potentially an
overly pessimistic comparison, as TDM PW packets are frequently
configured to be short in order to minimize latency, while TCP
packets are free to be much larger.
There are two network parameters relevant to our discussion, namely
the one-way delay D and the packet loss rate PLR. The one-way delay
of a native TDM service consists of the physical time-of-flight plus
125 microseconds for each TDM switch traversed; and is thus very
small as compared to typical PSN network-crossing latencies. Many
protocols and applications running over TDM circuits thus expect
extremely low delay, and thus in our comparisons we will only
consider delays of a few milliseconds.
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Regarding packet loss, the TDM PW RFCs specify behaviors upon
detecting a lost packet. Structure-agnostic transport has no
alternative to outputting an "all-ones" AIS pattern towards the TDM
circuit, which, when long enough in duration, is recognized by the
receiving TDM device as a fault indication (see Appendix A).
International standards place stringent limits on the number of such
faults tolerated. Calculations presented in the appendix show that
only loss probabilities in the realm of fractions of a percent are
relevant for structure-agnostic transport (see Appendix A).
Structure-aware transport regenerates frame alignment signals thus
hiding AIS indications resulting from infrequent packet loss.
Furthermore, for TDM circuits carrying voice channels the use of
packet loss concealment algorithms is possible (such algorithms have
been previously described for TDM PWs). However, even structure-
aware transport ceases to provide a useful service at about 2 percent
loss probability. Hence, in our comparisons we will only consider
PLRs of 1 or 2 percent.
RFC 5348 on TCP Friendly Rate Control (TFRC) [RFC5348] provides a
simplified formula for TCP throughput as a function of delay and
packet loss rate.
S
X = ------------------------------------------------
R ( sqrt(2p/3) + 12 sqrt(3p/8) p (1+32p^2) )
where
X is average sending rate in Bytes per second,
S is the segment (packet payload) size in Bytes,
R is the round-trip time in seconds,
p is the packet loss probability (i.e., PLR/100).
We can now compare the bandwidth consumed by TDM pseudowires with
that of a TCP flow for given packet loss and delay. The results are
depicted in the accompanying figures (available only in the PDF
version of this document). In Figures 1 and 2 we see the
conventional rate vs. packet loss plot for low-rate TDM (both T1 and
E1) traffic, as well as TCP traffic with the same payload size (64 or
256 Bytes respectively). Since the TDM rates are constant (T1 and E1
having payload throughputs of 1.544 Mbps and 2.048 Mbps
respectively), and the TDM service can only be faithfully emulated
using SAToP up to a PLR of about half a percent, the T1 and E1
pseudowires occupy line segments on the graph. On the other hand,
the TCP rate equation produces rate curves dependent on both delay
and packet loss.
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We see that in general for large packet sizes, short delays, and low
packet loss rates, the TDM pseudowires consume much less bandwidth
than TCP would under identical conditions. Only for small packets,
long delays, and high packet loss ratios, do TDM PWs potentially
consume more bandwidth, and even then only marginally. Similarly, in
Figures 3 and 4 we repeat the exercise for higher rate E3 and T3
(rates 34.368 and 44.736 Mbps respectively) pseudowires, allowing
delays and PLRs suitable appropriate for these signals. We see that
the TDM pseudowires consume much less bandwidth than TCP, for all
reasonable parameter combinations.
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Figure 1 E1/T1 PWs vs. TCP for segment size 64B
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Figure 2 E1/T1 PWs vs. TCP for segment size 256B
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Figure 3 T3/E3 PWs vs. TCP for segment size 536B
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Figure 4 T3/E3 PWs vs. TCP for segment size 1024B
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We can use the TCP rate equation to determine precise conditions
under which a TDM PW consumes no more bandwidth than a TCP flow
between the same endpoints would consume under identical conditions.
Replacing the round-trip delay with twice the one-way delay D,
setting the bandwidth to that of the TDM service BW, and the segment
size to be the TDM fragment (taking into account the PWE3 control
word), we obtain the following condition for a TDM PW.
4 S
D < -----------
BW f(p)
where
D is the one-way delay,
S is the TDM segment size (packet excluding overhead) in Bytes,
BW is TDM service bandwidth in bits per second,
f(p) = sqrt(2p/3) + 12 sqrt(3p/8) p (1+32p^2).
One may view this condition as defining an operating envelope for a
TDM PW, as a TDM PW that occupies no more bandwidth than a TCP flow
causes no more congestion than that TCP flow would. Under this
condition it is safe to place the TDM PW along with congestion-
responsive traffic such as TCP, without causing additional
congestion. on the other hand, were the TDM PW to consume
significantly more bandwidth a TCP flow, it could contribute
disproportionately to congestion, and its mixture with congestion-
responsive traffic might be inappropriate.
We derived this condition assuming steady-state conditions, and thus
two caveats are in order. First, the condition does not specify how
to treat a TDM PW that initially satisfies the condition, but is then
faced with a deteriorating network environment. In such cases one
additionally needs to analyze the reaction times of the responsive
flows to congestion events. Second, the derivation assumed that the
TDM PW was competing with long-lived TDM flows, because under this
assumption it was straightforward to obtain a quantitative comparison
with something widely considered to offer a safe response to
congestion. Short-lived TCP flows may find themselves disadvantaged
as compared to a long-lived TDM PW satisfying the condition.
We see in Figures 5 and 6 that TDM pseudowires carrying T1 or E1
native services satisfy the condition for all parameters of interest
for large packet sizes (e.g., S=512 Bytes of TDM data). For the
SAToP default of 256 Bytes, as long as the one-way delay is less than
10 milliseconds, the loss probability can exceed 0.3 or 0.6 percent.
For packets containing 128 or 64 Bytes the constraints are more
troublesome, but there are still parameter ranges where the TDM PW
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consumes less than a TCP flow under similar conditions. Similarly,
Figures 7 and 8 demonstrate that E3 and T3 native services with the
SAToP default of 1024 Bytes of TDM per packet satisfy the condition
for a broad spectrum of delays and PLRs.
Note that violating the condition for a short amount of time is not
sufficient justification for shutting down the TDM PW. While TCP
flows react within a round trip time, PW commissioning and
decommissioning are time consuming processes that should only be
undertaken when it becomes clear that the congestion is not
transient.
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Figure 5 TCP Compatibility areas for T1 SAToP
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Figure 6 TCP Compatibility areas for E1 SAToP
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Figure 7 TCP Compatibility areas for E3 SAToP
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Figure 8 TCP Compatibility areas for T3 SAToP
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4. Security Considerations
This document does not introduce any new congestion-specific
mechanisms and thus does not introduce any new security
considerations above those present for PWs in general.
5. IANA Considerations
This document requires no IANA actions.
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6. Informative References
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41, RFC
2914, September 2000.
[RFC3985] Bryant, S. and P. Pate, "Pseudo Wire Emulation Edge-to-
Edge (PWE3) Architecture", RFC 3985, March 2005.
[RFC4023] Worster, T., Rekhter, Y., and E. Rosen, "Encapsulating
MPLS in IP or Generic Routing Encapsulation (GRE)", RFC
4023, March 2005.
[RFC4553] Vainshtein, A. and YJ. Stein, "Structure-Agnostic Time
Division Multiplexing (TDM) over Packet (SAToP)", RFC
4553, June 2006.
[RFC5033] Floyd, S. and M. Allman, "Specifying New Congestion
Control Algorithms", BCP 133, RFC 5033, August 2007.
[RFC5086] Vainshtein, A., Sasson, I., Metz, E., Frost, T., and P.
Pate, "Structure-Aware Time Division Multiplexed (TDM)
Circuit Emulation Service over Packet Switched Network
(CESoPSN)", RFC 5086, December 2007.
[RFC5087] Stein, Y(J)., Shashoua, R., Insler, R., and M. Anavi,
"Time Division Multiplexing over IP (TDMoIP)", RFC 5087,
December 2007.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC
5348, September 2008.
[G775] International Telecommunications Union, "Loss of Signal
(LOS), Alarm Indication Signal (AIS) and Remote Defect
Indication (RDI) defect detection and clearance criteria
for PDH signals", ITU Recommendation G.775, October 1998.
[G826] International Telecommunications Union, "Error Performance
Parameters and Objectives for International Constant Bit
Rate Digital Paths at or above Primary Rate", ITU
Recommendation G.826, December 2002.
[P862] International Telecommunications Union, "Perceptual
evaluation of speech quality (PESQ): An objective method
for end-to-end speech quality assessment of narrow-band
telephone networks and speech codecs", ITU Recommendation
G.826, February 2001.
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[I-D.stein-pwe3-tdm-packetloss]
Stein, Y(J). and I. Druker, "The Effect of Packet Loss on
Voice Quality for TDM over Pseudowires", October 2003.
[I-D.fairhurst-tsvwg-circuit-breaker]
Fairhurst, G., "Network Transport Circuit Breakers",
draft-fairhurst-tsvwg-circuit-breaker-01 (work in
progress), May 2014.
Appendix A. Loss Probabilities for TDM PWs
ITU-T Recommendation G.826 [G826] specifies limits on the Errored
Second Ratio (ESR) and the Severely Errored Second Ratio (SESR). For
our purposes, we will simplify the definitions and understand an
Errored Second (ES) to be a second of time during which a TDM bit
error occurred or a defect indication was detected. A Severely
Errored Second (SES) is an ES second during which the Bit Error Rate
(BER) exceeded one in one thousand (10^-3). Note that if the error
condition AIS was detected according to the criteria of ITU-T
Recommendation G.775 [G826] a SES was considered to have occurred.
The respective ratios are the fraction of ES or SES to the total
number of seconds in the measurement interval.
For both E1 and T1 TDM circuits, G.826 allows ESR of 4% (0.04), and
SESR of 1/5% (0.002). For E3 and T3 the ESR must be no more than
7.5% (0.075), while the SESR is unchanged.
Focusing on E1 circuits, the ESR of 4% translates, assuming the worst
case of isolated exactly periodic packet loss, to a packet loss event
no more than every 25 seconds. However, once a packet is lost,
another packet lost in the same second doesn't change the ESR,
although it may contribute to the ES becoming a SES. Assuming an
integer number of TDM frames per PW packet, the number of packets per
second is given by packets per second = 8000 / (frames per packet),
where prevalent cases are 1, 2, 4 and 8 frames per packet. Since for
these cases there will be 8000, 4000, 2000, and 1000 packets per
second, respectively, the maximum allowed packet loss probability is
0.0005%, 0.001%, 0.002%, and 0.004% respectively.
These extremely low allowed packet loss probabilities are only for
the worst case scenario. In reality, when packet loss is above
0.001%, it is likely that loss bursts will occur. If the lost
packets are sufficiently close together (we ignore the precise
details here) then the permitted packet loss rate increases by the
appropriate factor, without G.826 being cognizant of any change.
Hence the worst-case analysis is expected to be extremely pessimistic
for real networks. Next we will go to the opposite extreme and
assume that all packet loss events are in periodic loss bursts. In
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order to minimize the ESR we will assume that the burst lasts no more
than one second, and so we can afford to lose no more than packet per
second packets in each burst. As long as such one-second bursts do
not exceed four percent of the time, we still maintain the allowable
ESR. Hence the maximum permissible packet loss rate is 4%. Of
course, this estimate is extremely optimistic, and furthermore does
not take into consideration the SESR criteria.
As previously explained, a SES is declared whenever AIS is detected.
There is a major difference between structure-aware and structure-
agnostic transport in this regards. When a packet is lost SAToP
outputs an "all-ones" pattern to the TDM circuit, which is
interpreted as AIS according to G.775 [G775]. For E1 circuits, G.775
specifies for AIS to be detected when four consecutive TDM frames
have no more than 2 alternations. This means that if a PW packet or
consecutive packets containing at least four frames are lost, and
four or more frames of "all-ones" output to the TDM circuit, a SES
will be declared. Thus burst packet loss, or packets containing a
large number of TDM frames, lead SAToP to cause high SESR, which is
20 times more restricted than ESR. On the other hand, since
structure-aware transport regenerates the correct frame alignment
pattern, even when the corresponding packet has been lost, packet
loss will not cause declaration of SES. This is the main reason that
SAToP is much more vulnerable to packet loss than the structure-aware
methods.
For realistic networks, the maximum allowed packet loss for SAToP
will be intermediate between the extremely pessimistic estimates and
the extremely optimistic ones. In order to numerically gauge the
situation, we have modeled the network as a four-state Markov model,
(corresponding to a successfully received packet, a packet received
within a loss burst, a packet lost within a burst, and a packet lost
when not within a burst). This model is an extension of the widely
used Gilbert model. We set the transition probabilities in order to
roughly correspond to anecdotal evidence, namely low background
isolated packet loss, and infrequent bursts wherein most packets are
lost. Such simulation shows that up to 0.5% average packet loss may
occur and the recovered TDM still conform to the G.826 ESR and SESR
criteria.
Appendix B. Effect of Packet Loss on Voice Quality for TDM PWs
Packet loss in voice traffic can cause in gaps or artifacts that
result in choppy, annoying or even unintelligible speech. The
precise effect of packet loss on voice quality has been the subject
of detailed study in the VoIP community, but VoIP results are not
directly applicable to TDM PWs. This is because VoIP packets
typically contain over 10 milliseconds of the speech signal, while
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multichannel TDM packets may contain only a single sample, or perhaps
a very small number of samples.
The effect of packet loss on TDM PWs has been previously reported
[I-D.stein-pwe3-tdm-packetloss]. In that study it was assumed that
each packet carried a single sample of each TDM timeslot (although
the extension to multiple samples is relatively straightforward and
does not drastically change the results). Four sample replacement
algorithms were compared, differing in the value used to replace the
lost sample:
1. replacing every lost sample by a preselected constant (e.g., zero
or "AIS" insertion),
2. replacing a lost sample by the previous sample,
3. replacing a lost sample by linear interpolation between the
previous and following samples,
4. replacing the lost sample by STatistically Enhanced INterpolation
(STEIN).
Only the first method is applicable to SAToP transport, as structure
awareness is required in order to identify the individual voice
channels. For structure aware transport, the loss of a packet is
typically identified by the receipt of the following packet, and thus
the following sample is usually available. The last algorithm posits
the LPC speech generation model and derives lost samples based on
available samples both before and after each lost sample.
The four algorithms were compared in a controlled experiment in which
speech data was selected from English and American English subsets of
the ITU-T P.50 Appendix 1 corpus [P.50App1] and consisted of 16
speakers, eight male and eight female. Each speaker spoke either
three or four sentences, for a total of between seven and 15 seconds.
The selected files were filtered to telephony quality using modified
IRS filtering and downsampled to 8 KHz. Packet loss of 0, 0.25, 0.5,
0.75, 1, 2, 3, 4 and 5 percent were simulated using a uniform random
number generator (bursty packet loss was also simulated but is not
reported here). For each file the four methods of lost sample
replacement were applied and the Mean Opinion Score (MOS) was
estimated using PESQ [P862]. Figure 5 depicts the PESQ-derived MOS
for each of the four replacement methods for packet drop
probabilities up to 5%.
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Figure 5 PESQ derived MOS as a function of packet drop probability
For all cases the MOS resulting from the use of zero insertion is
less than that obtained by replacing with the previous sample, which
in turn is less than that of linear interpolation, which is slightly
less than that obtained by statistical interpolation.
Unlike the artifacts speech compression methods may produce when
subject to buffer loss, packet loss here effectively produces
additive white impulse noise. The subjective impression is that of
static noise on AM radio stations or crackling on old phonograph
records. For a given PESQ-derived MOS, this type of degradation is
more acceptable to listeners than choppiness or tones common in VoIP.
If MOS>4 (full toll quality) is required, then the following packet
drop probabilities are allowable:
zero insertion - 0.05 %
previous sample - 0.25 %
linear interpolation - 0.75 %
STEIN - 2 %
If MOS>3.75 (barely perceptible quality degradation) is acceptable,
then the following packet drop probabilities are allowable:
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zero insertion - 0.1 %
previous sample - 0.75 %
linear interpolation - 3 %
STEIN - 6.5 %
If MOS>3.5 (cell-phone quality) is tolerable, then the following
packet drop probabilities are allowable:
zero insertion - 0.4 %
previous sample - 2 %
linear interpolation - 8 %
STEIN - 14 %
Authors' Addresses
Yaakov (Jonathan) Stein
RAD Data Communications
24 Raoul Wallenberg St., Bldg C
Tel Aviv 69719
ISRAEL
Phone: +972 (0)3 645-5389
Email: yaakov_s@rad.com
David L. Black
EMC Corporation
176 South St.
Hopkinton, MA 69719
USA
Phone: +1 (508) 293-7953
Email: david.black@emc.com
Bob Briscoe
BT
B54/77, Adastral Park
Martlesham Heath
Ipswich IP5 3RE
UK
Phone: +44 1473 645196
Email: bob.briscoe@bt.com
URI: http://bobbriscoe.net/
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