One document matched: draft-stein-pwe3-tdm-packetloss-00.txt
Network Working Group Y(J) Stein
Internet-Draft I. Druker
Expires: March 2, 2003 RAD Data Communications
September 1, 2002
The Effect of Packet Loss on Voice Quality for TDM over Pseudowires
draft-stein-pwe3-tdm-packetloss-00.txt
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Copyright (C) The Internet Society (2002). All Rights Reserved.
Abstract
The effect of packet loss on voice quality has been the subject of
detailed study in the VoIP community, but these results are not
directly applicable to TDM transport as being studied in the PWE WG.
The present document presents an analysis of packet loss for the TDM
over PW case, and demonstrates that packet loss of a few percent can
be tolerated. We propose that robustness to packet loss of a few
percent be a requirement for any proposed method for transport of TDM
over pseudowires.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Measures of Voice Quality . . . . . . . . . . . . . . . . . . 4
3. Packet Loss Replacement Algorithms . . . . . . . . . . . . . . 5
4. Experimental Results . . . . . . . . . . . . . . . . . . . . . 6
5. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . . 8
6. Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 10
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 10
Full Copyright Statement . . . . . . . . . . . . . . . . . . . 12
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1. Introduction
There are several sources of packet loss in PSNs. Routers purposely
drop packets when they detect errors in them, or when they need to
manage congestion, leading to typical packet loss rates of between 1
and 20 percent. Real-time streams have an additional source of
packet loss, namely reordered packet rejection at the PE. Non-real-
time data communications is not overly effected by packet loss, due
to retransmission mechanisms, but real-time constraints usually
prohibit retransmission.
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, and the development
of packet loss concealment algorithms have been the subject of
detailed study in the VoIP community. Their results can be
summarized as follows: 1) One percent packet loss causes perceived
voice quality to drop from toll-quality to cell-phone quality. 2)
Above two percent, packet loss is the dominant cause of voice quality
deterioration, compressed and uncompressed speech becoming comparable
in quality. 3) Packet length is not a significant factor (at least
for lengths typically employed in VoIP). 4) By using appropriate
packet loss concealment algorithms (PLC) five percent packet loss of
uncompressed speech can be comparable or better than cell-phone
quality.
Unfortunately, these results are not directly applicable to TDM
transport as being studied in the PWE WG [TDMoIP,CESoPSN,SONET-VT].
This is because VoIP packets typically contain between 80 samples (10
milliseconds) and 240 samples (30 milliseconds) of the speech signal,
while multichannel TDM packets may contain only a single sample, or
perhaps a very small number of samples.
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2. Measures of Voice Quality
Perceived voice quality is a psychophysical quantity that depends on
the physiology and psychology of the listener. The most universally
accepted subjective measure of voice quality is the mean opinion
score (MOS) defined by the ITU-T for telephone quality speech in
[P.800], and by the ITU-R for higher fidelity audio in [BS.1116-1].
It is found by averaging the reported grades of multiple listeners,
each of which rates the audio on a five point quality scale, with
MOS=1 being unintelligible, and MOS=5 meaning excellent quality. Due
to the 4 KHz bandwidth limitation and the logarithmic amplitude
characteristics of the 64 Kbps DS0 digital channel, telephony voice
is rated lower than 5, with 4 to 4.5 being considered "toll-quality".
MOS ratings of 3.5 to 4 are considered acceptable to many listeners,
and cellular telephones audio is readily accepted at about MOS=3.5
due to the added convenience of the cellular medium. Speech quality
lower than MOS=3 is considered acceptable only for special
applications, such as encrypted military communications.
The problem with MOS is that being a subjective measure it is time
consuming and costly to measure. Objective measures, ones that can
be computed by algorithms based on the signal samples, are preferable
if they correlate well with the subjective measures. The ITU-T has
standardized two such measures for telephony quality speech, namely
PSQM [P.861] and PESQ [P.862], while the ITU-R has decided on PEAQ
[BS.1387] for higher fidelity radio quality audio. These objective
measures utilize models of the biological auditory system and have
been shown to correlate well with subjective measurements of MOS.
PSQM was developed for lab comparison of different speech codecs and
does not take such factors as delay or packet loss into account.
PESQ specifically performs end-to-end speech quality assessment and
was therefore chosen for our experiment.
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3. Packet Loss Replacement Algorithms
In this section we discuss algorithms for concealing packet loss when
it occurs. For concreteness we will assume in the following
discussion that packets carry single samples of each TDM timeslot.
The extension to multiple samples is relatively straightforward, and
turns out not to drastically change the results of the next section.
The simplest ploy to implement is to blindly insert a constant value
in place of any lost speech samples. Since we can assume that the
input signal is zero-mean (i.e. contains no DC component) minimal
distortion is attained when this constant is chosen to be zero. This
is in fact precisely what happens when a G.711 mu-law codec receives
a word containing all-ones, as would be the case if AIS were to be
received (but unfortunately is not true for A-law).
A slightly more sophisticated technique is to replace the missing
sample with the previous one. This method is somewhat more
justifiable in the VoIP case where the quasistationarity of the
speech signal means that the missing buffer is expected to be similar
to the previous one. Even in the single sample case it is decidedly
better than replacement by zero due to the typical low-pass quality
of speech signals, and to the fact that during intervals with
significant high frequency content (e.g. fricatives) the error is
less noticeable.
A packet is usually declared lost following the reception of the next
packet, hence the both the sample prior to the missing one, and that
following it are available. This enables us to estimate the missing
sample value by interpolation, the simplest type of which is linear
interpolation, whereby the missing sample is replaced by the average
of the two surrounding values. This serves to conceal the packet
loss event. More complex interpolation, such as quadratic
interpolation or splines can be used as well, but for the purposes of
this analysis we will restrict ourselves to the linear case.
More sophisticated methods of packet concealment are based on model-
based prediction. Standardized speech compression algorithms have
had integral packet loss concealment methods for some time, and more
recently the ITU-T has standardized a packet loss concealment method
for uncompressed speech [G.711App1]. For such algorithms to function
previous sample values must be saved in a circular buffer or re-
extracted from the system jitter buffer. For the purposes of the
experiment described in Section 4, we need only to estimate the value
of a single missing sample, and so relatively simple modeling is
sufficient. We used an interpolation model based on second order
statistics of the previous 30 samples. Details of this algorithm
will be reported elsewhere.
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4. Experimental Results
In order to quantify the anecdotal results we have observed in real-
world deployments, we have carried out a controlled experiment to
measure the effect of packet loss on voice quality. We first
describe the methodology we employed.
The 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.
A uniform random number generator was used to simulate packet loss.
Packet loss of 0, 0.25, 0.5, 0.75, 1, 2, 3, 4 and 5 percent were
tested. In the simulations reported here we disallowed loss of
successive packets; bursty packet loss (where the probability of
groups of missing samples is much higher than would be expected from
the average packet loss rate) was also simulated but is not reported
here.
For each file four methods of lost sample replacement were applied
and PESQ software was then used to estimate the MOS rating. A graph
depicting the PESQ derived MOS as a function of packet loss for the
four lost packet replacement algorithms cases is available in ps and
pdf formats at http://www.dspcsp.com/tdmoip/pl.ps and
http://www.dspcsp.com/tdmoip/pl.pdf respectively.
We obtained the following qualitative and quantitative results.
1) 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.
2) 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, this type of degradation is more
acceptable to listeners than choppiness or tones common in VoIP.
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3) If MOS>4 (full toll quality) is required, then the following
packet losses are allowable:
zero insertion - 0.05 %
previous sample - 0.25 %
linear interpolation - 0.75 %
statistical interpolation - 2 %
4) If MOS>3.75 (barely perceptible quality degradation) is tolerable,
then the following packet losses are allowable:
zero insertion - 0.1 %
previous sample - 0.75 %
linear interpolation - 3 %
statistical interpolation - 6.5 %
5) If MOS>3.5 (cell-phone quality) is sufficient, then the following
packet losses are allowable:
zero insertion - 0.4 %
previous sample - 2 %
linear interpolation - 8 %
statistical interpolation - 14 %
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5. Discussion
The most undemanding approach to handling packet loss in TDM over PW
is to generate Alarm Indication Signal (AIS) whenever a packet is
lost. This results in insertion of constant values, and extremely
low tolerance. Transport methods that respect frame structure, such
as AAL1, employ "frame replay", which increases the perceived voice
quality and has the added benefit that CAS signaling integrity is
guaranteed.
The linear and statistical interpolation methods can only be employed
when the TDM is transported in the PW in a framed and structured
fashion, i.e. that the timeslot signal values are readily available
for manipulation. This rules out unframed transport and non-byte-
oriented transport (including some methods of transporting T1 links).
In addition, complex encapsulations that impede the extraction of
required samples, may hinder the use of these methods.
Assuming a processor with hardware companding and which can perform
an addition and a shift in a single cycle (e.g. a DSP), linear
interpolation requires a single cycle per timeslot per sample loss,
or 8000 L instruction cycles per second, where L is the packet loss
percentage. An entire 30 channel E1 link will thus require 0.24 L
MIPS, and an entire 24 channel T1 link 0.192 L MIPS. For example at
2% packet loss, an average processing power of 1 MIPS will suffice
for 208 E1 trunks or 260 T1 trunks. Even using a processor that
requires 10 instructions to process an interpolation, dedicating 1
MIPS will enable fixing 20 E1s or 26 T1s.
The statistical interpolation method requires the computation of
energy, single and dual lag autocorrelations, which for a history
buffer of N samples involves approximately 3N multiplications and
additions. For processors with MAC operations (e.g. a DSP) this
translates to 0.024 N L MIPS per timeslot (0.72 N L MIPS per E1 or
0.576 N L MIPS per T1). N must be chosen large enough to capture the
signal statistics, but not so large that the statistics would be
expected to change significantly in normal speech. Numbers in the
range 10 to 100 are reasonable. For example, using N=30 and once
again assuming 2% packet loss, the processing drain would be 0.432
MIPS per E1 and 0.3456 MIPS per T1.
Although statistical interpolation is consistently better than simple
linear interpolation, the additional MIPS would probably only be
justifiable when the packet loss rate is particularly high.
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6. Summary
Packet loss is to be expected in any packet switched network, but
does not degrade most data traffic since retransmission mechanisms
compensate for it with no ill effects other than a reduction in
effective data transfer rate. Unfortunately, real-time traffic such
as TDM can frequently not tolerate the added latency retransmission
incurs.
Conventional TDM networks dedicate highly synchronous circuits to
voice calls. Hence there is never packet loss, and even individual
bit slips are tightly controlled. Telephony customers have grown
accustomed to telephone service quality, and will not consent to
lower quality unless there are other major advantages (e.g.
mobility, significantly lower price).
Market acceptance of TDM transport over PW will depend on service
providers being able to offer SLAs with meaningful voice quality
guarantees, while deploying networks with some reasonable amount of
packet loss.
We have shown that by using simple packet loss concealment
techniques, methods of transporting TDM over PW can function under a
few percent packet loss without dramatic degradation of voice
quality.
Since the voice quality is not a major obstacle, it is mandatory that
the protocols employed not introduce additional impediments to
operation at realistic packet loss rates.
We therefore propose that robustness to packet loss of a few percent
be a requirement for any proposed method for pseudowire transport of
TDM.
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7. References
[BS.1116-1] ITU-R Recommendation BS.1116-1 (1994-1997) Methods for
the Subjective Assessment of Small Impairments in Audio Systems
Including Multichannel Sound
[BS.1387] ITU-R Recommendation BS.1387 (1998) Method for Objective
Measurements of Perceived Audio Quality
[CESoPSN] draft-vainshtein-cesopsn-03.txt (2002) TDM Circuit
Emulation Service over Packet Switched Network (CESoPSN), Alexander
("Sasha") Vainshtein et al, work in progress
[G.711App1] ITU-T Recommendation G.711 - Appendix I (1999) A high
quality low-complexity algorithm for packet loss concealment with
G.711
[P.50App1] ITU-T Recommendation P.50 - Appendix I (1998) Artificial
Voices - Test Signals
[P.800] ITU-T Recommendation P.800 (1996) Methods for Subjective
Determination of Transmission Quality
[P.861] ITU-T Recommendation P.861 (1998) Objective Quality
Measurement of Telephone-band (300-3400 Hz) Speech Codecs
[P.862] ITU-T Recommendation P.862 (2001) Perceptual evaluation of
speech quality (PESQ), an objective method for end-to-end speech
quality assessment of narrow-band Telephone Networks and Speech
Codecs
[SONET-VT] draft-ietf-pwe3-sonet-vt-00.txt (2002) TDM Service
Specification for Pseudo-Wire Emulation Edge to Edge (PWE3), Prayson
Pate et al, work in progress
[TDMoIP] draft-anavi-tdmoip-04.txt (2002) TDM over IP, Yaakov
(Jonathan) Stein et al, work in progress
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Authors' Addresses
Yaakov (Jonathan) Stein
RAD Data Communications
24 Raoul Wallenburg St., Bldg C
Tel Aviv 69719
ISRAEL
Phone: +972 3 6455389
EMail: yaakov_s@rad.co.il
Ilya Druker
RAD Data Communications
24 Raoul Wallenburg St., Bldg C
Tel Aviv 69719
ISRAEL
Phone: +972 3 7657061
EMail: ilya_d@rad.co.il
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