Voice Delay Considerations for DSL
Multimedia Access Network Design and Operation
S. Wright A. J. Vernon
Architecture & Emerging Technology Ridge Consulting Group Inc.,
BellSouth, Atlanta, GA Atlanta, GA
Steven.Wright@BellSouth.com ajv@RidgeConsultingGroup.com
Abstract—This paper discusses the carrier perspective on
baseline capabilities and technology tradeoffs in the capacity
planning methods in the context of the evolution of the DSL
Access network. Capacity improvements are being made possible
by new modem technologies and deeper deployments (shorter
loop lengths). DSL networks are evolving from a service-specific
(Internet Access) infrastructure to one that must support
multiple services. Emerging multimedia service requirements
(e.g., for voice) are considered from the perspective of the
additional design constraints (e.g., delay) that impact DSL
network design and its operation.
Keywords-DSL, NGN,QoS, Delay, Capacity Planning
I. INTRODUCTION
Service providers, equipment vendors and others are
expending significant effort to develop both a vision statement
and an architectural framework for Next Generation Networks
(NGN) and services, resulting in standardization efforts in
bodies such as ITU-T, 3GPP, ETSI, and ATIS (see e.g. [3]).
The NGN envisages a multi-provider, multi-access, QoS-
enabled service environment based on IP connectivity,
including existing and evolving DSL network infrastructure.
The NGN standards work introduces the notion of an IP
Connectivity Access Network (IPCAN) to provide access to
NGN services. These NGN notions introduce additional
functionality to DSL access networks (e.g. Quality of Service
(QoS), mobility and a Session Initiation Protocol (SIP) control
plane based on the IP Multimedia System (IMS)). While
IPCAN conformance criteria are still under discussion, DSL
networks are expected to evolve in compliance.
Recent work at the DSL Forum [1] has created a focus on
the development of architectures and the selection of
appropriate mechanisms to support QoS-enabled IP services.
While future improvements to the bandwidth available on the
DSL line (e.g., ADSL, ADSL2, ADSL2+, VDSL2) may be
deployed by operators and may appear to diminish the need for
QoS mechanisms, any deployed bandwidth tends to be
consumed [2]. While strict QoS guarantees are not envisioned
in this environment, the QoS capabilities required are expected
to address the stricter performance requirements for latency-
and jitter-sensitive multimedia services (e.g., Voice over
Internet Protocol (VoIP) and gaming [13]). Deeper
deployments (that is, shortening copper loop lengths) and loop
bonding provide additional mechanisms to improve DSL
capacity. Bonding can occur at multiple levels in the protocol
stack, for example, at the modems with G.Bond [11], at the
BRAS through MLPPP, at an Ethernet switching device
0-7803-9319-8/05/$20.00 (c)2005 IEEE. 111
A. Whited
Architecture & Emerging Technology
BellSouth, Atlanta, GA
Albert.Whited@BellSouth.com
through 802.3ad, or at an IP service layer using ECMP—each
with different tradeoffs in terms of efficiency and end-user
impact.
Given the current industry focus, we assume the service
provider wishes to offer a “triple play” suite of voice, data, and
entertainment video services over a DSL infrastructure. In this
paper we consider one illustrative deployment scenario where
broadcast entertainment video is channelized separately from
the DSL access network (e.g. via satellite, hybrid Fiber Coax,
Fiber to the Curb, MMDS), and video on demand or PVR
video content is delivered via a “best-effort” data service (e.g.
tcp, udp). Two QoS classes are assumed, with voice receiving
either preemptive or non-preemptive priority over best-effort
data.
The paper is organized as follows. Section II reviews some
of the critical traffic demand considerations, focusing on delay
issues. Section III discusses some of the implications for DSL
network capacity design, primarily through the consideration of
VoIP as a multimedia service component. Conclusions
regarding DSL link configuration, CODEC selection options,
and implications of alternative broadcast delivery strategies are
discussed in Section IV.
II. DSL TRAFFIC DEMAND CONSIDERATIONS
A. DSL Network Delay Performance Objectives
DSL network performance, as expressed through Quality of
Service (QOS) metrics must be translated into appropriate
Quality of Experience (QOE) metrics to determine whether the
end user requirements are met. Mean Opinion Survey (MOS)
scores are the accepted QOE metrics for voice and the E-model
provides an empirical mechanism to translate delay metrics into
MOS scores [9]. The primary focus of this QoS analysis is
delay. ITU Y.1541 [10] identifies end-to-end network
performance objectives for IP networks, including delay
parameters, and Y.1541 Appendix X provides some example
speech quality calculations for packet voice calls, albeit with a
different reference model than this document. ITU G.114
provides planning values for the delays associated with various
transmission elements in the PSTN [8]. G.114 Appendix II
provides guidance on one-way delay for packet voice systems
for the intra-and inter-regional cases; however, it does not
make recommendations on apportioning this delay. In the
regional case (defined to have a diameter less than 5000 km,
such as North America) the “mouth-to-ear” one-way delay
should be less than 150 ms. In this paper we assume the voice
call traverses an inter-exchange carrier as well as a local
exchange carrier at the far end, and allocate the delay objective
equally across the three service providers. This results in a
50ms delay objective for voice service across the DSL network
(from the voice client to the PSTN gateway, as illustrated in
Figure 4)
While bounds exist for worst-case delays in packets
networks [7], engineering a network based on those bounds
would certainly lead to cost-prohibitive over-provisioning.
Instead of only considering an upper bound on worst-case
delay, the approach adopted here is based on the stochastic
analysis of the delay bound, as presented in [5], which
specifically accounts for voice having preemptive or non-
preemptive priority over data traffic (for example, voice is
transported in the Expedited Forwarding (EF) class in the IETF
DIFFSERV model). Following [5] and their arguments on delays
and the intelligibility of received speech, we adopt a statistical
upper- bound on the maximum delay for the voice services:
Pr[Access Network Delay ≥ 50 ms] < 10 −5 (1)
B. Traffic Demand Characteristics
The data traffic demand is assumed to be derived from the
requirement to support Internet access. Although the DSL
Forum architecture permits multiple data classes of
differentiated service traffic, we assume all data traffic is
treated as “Best Effort”.
Voice traffic characteristics depend upon the CODEC
selection; we consider both G.711 and G.729A CODECs without
the use of voice activity detection. Both schemes use an 8 kHz
sampling rate and a 20 ms packet formation time, but G.729A
uses 16-bit samples and 16:1 compression ratio, resulting in a
voice data rate of 8 kbps (compared with 64 kbps for G.711).
While this implies significant efficiency, G.729A does incur a
5 ms look-ahead delay penalty.
Entertainment video is primarily a unidirectional service
with significant bandwidth demands depending upon codec
selection. Entertainment services are typically considered as
broadcast or multicast services, but emerging trends towards
Video-on-Demand (VoD) and the user oriented content search
capabilities demonstrated by Personal Video Recorders (PVRs)
seem to be changing the entertainment video service concept
towards a unicast model. Modeling video traffic sources can be
fairly complex, (see [12] for a tutorial) depending on CODEC
selection (e.g. MPEG-4), operational parameters (e.g. video
profile), and system level service design choices (e.g.
Multilevel encoding for different video resolutions).
Entertainment video is also sensitive to errors, with stringent
demands for the users QoE resulting in requirements for error
concealment techniques in the codecs and error control
mechanisms (e.g. FEC /Interleave). The predictive nature of
video frame construction in MPEG magnifies the effect of a
single packet error, potentially damaging multiple video frames
constructed at later time, expanding the error over several
frames.
Of course, the traffic demands for the different multimedia
service components supported by the DSL network may be
constrained by admission control – either a generic rate control
or a session control based on the NGN SIP architecture. This
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would limit the peak traffic load that the infrastructure would
need to support.
C. Delay Model for DSL Access Network
A model for the transport of voice in the access network and
the relevant delay components is shown in Figure 4. The
model assumes that the voice streams are
terminated/originated at a (PSTN) gateway or border proxy.
Link delays, for example Duplink, include processing overheads
at the transmitting node, if any. In the core IP transport
network, voice transport delay is obviously a function of how
the IP traffic is routed; therefore, the index h is used to
represent a router hop (a hop includes the processing and
scheduling delays at the transmitting router). From Figure 4 it
is quite easy to see that the equation governing the access
network delay (D) of voice packets is:
D = D pack + Denc + Dhome + DDSL
(2)
+ Duplink + ∑D
h∈Path
h + Degress + D play + Ddec
For transport links, the delay on a link is the sum of the delays
due to transmission, queuing, interleaving, bonding,
propagation, and node processing. It will be assumed that the
interleaving delay (Ilink), link bonding delay (Blink), propagation
delay (Plink) and the node processing delay (Slink) are all
negligible and thus the delay on a link is assumed to
be Dlink = Tlink + Qlink . Also, to avoid potential packet loss,
the play-out (Dplay) and decode (Ddec) delays at the PSTN
gateway are assumed to equal the maximum queuing delay
(Q*) [5]. Obviously, transmission delay (Tlink), and queuing
delay (Qlink) depend on the length of the packets and the arrival
process of the packets to the link, both of which are functions
of the type of CODEC used. Per the objectives discussed in
Section II.A, , the delay must be bounded to ensure that
application performance is sufficient:
Denc + D pack + Thome + TDSL + Tuplink
(3)
+ ∑T
h∈Path
h + Tegress + 2Q * ≤ 50ms
D. Analysis and Example Results
Considering the upstream toward the PSTN gateway, the
transmission delays are dominated by the transmission delay of
DSL line. In a worst-case scenario (256 kbps upstream data
rate), the sum of the transmission delays is approximately 10
ms. Accounting for CODEC-specific delays, Equation 4 implies
the following constraint on maximum queuing delay:
Q * ≤ 
10 ms G.711 (4)
7.5 ms G.729
Analysis of queuing delay is based on the methodology
presented in [5] and [6]. For worst-case delay analysis, it is
assumed that a voice packet will see the residual transmission
time of a 1500-byte TCP payload, in addition to the intra-class
queuing delays resulting from the phase differences between
the voice streams. For G.711 CODECs, Figures 1 and 2 provide
example complementary cumulative distribution functions for
the maximum queuing delays on a DSL link and a DS3
DSLAM uplink, respectively. In both figures, the parameter K
denotes the number of active voice streams. Results for
G.729A CODECs and the other links in the network (see
Figure 4) can be similarly obtained, but those results are
omitted for the sake of brevity. In the next section, these and
similar results are applied to the issue of DSL network design.
Figure 1 DSL Link Queuing Delay CCDF: Preemptive
(dashed) and Non-Preemptive (solid)
Figure 2 DS3 Link Queuing delay CCDF: Preemptive
(dashed) and Non-Preemptive (solid)
III. DSL NETWORK DESIGN METHODOLOGY
Access networks are blocking aggregation networks: the
aggregate bandwidth of subscriber interfaces is greater than the
aggregate bandwidth of core network interfaces. Capacity
planning for Internet access services in DSL networks is
typically based on an assumption of heavily asymmetric traffic
dominated by downloads. The asymmetry assumption may be
challenged in future by changing data usage patterns, for
example, a growing emphasis on peer-peer communication
applications rather than client-server download applications.
On the other hand, entertainment video services would make
the aggregate traffic more asymmetric.
A. Bandwidth-Constrained Capacity Planning for Voice
services over DSL
For the PSTN access networks, congestion was managed by
a combination of factors including (a) using admission control
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to gracefully constrain excess demand, and (b) provisioning
capacity based on (i) well-defined traffic characteristics and
demand assumptions and (ii) well-known theory to translate
these into capacity requirements (e.g., Erlang tables [16]). As
an example, consider a DSLAM with 100 lines each supporting
a VOIP service with demand estimated at 0.1 ERL. At a blocking
probability of 0.5%, the uplink capacity required for the 10 ERL
aggregate demand at the DSLAM is 21 circuits. The bandwidth
required for the voice circuits can be calculated using the
applicable overheads for the protocol stacks on the link (see
Figure 4). The bandwidth required for the voice service must
be compared with that available on the physical facility and
that required for other services. In this case we assume priority
for the voice service and elasticity of the best-effort data
service, but some minimum throughput capacity is still
required for the best effort data class. For a typical DS3 uplink,
21 G.711 voice streams (each requiring 106 kbps, including
link overheads) is well within the capacity available. Note that
efforts to shorten DSL loop lengths result in additional stages
of aggregation: if the DSL links were aggregated by an FTTC
implementation with 10 voice lines per node, the voice capacity
required at the node uplinks would be 5 circuits. Aggregation
characteristics of the data class are beyond the scope of this
paper. Greater dispersal of the aggregation reduces the node
size and consequently impacts the trunking efficiency. The
admission control mechanism should recognize each potential
choke point in the network and admit accordingly for proper
service operation.
B. Delay-Constrained CapacityPlanning for DSL
Bandwidth dimensioning is only a single facet of the
capacity planning process for packet-switched systems: delay-
sensitive applications, such as voice, require that the delay also
be engineered, and it is quite possible that delay may be the
more limiting factor. As an example consider a DSLAM
uplink carrying G.711 voice streams that is allocated 1ms for
voice queuing delay and the probability this budget is exceeded
cannot be greater than 10-5. Using the model and analysis
approach described earlier, Figure 3 plots the voice-stream
capacity as a function of the link type and illustrates that for
both preemptive and non-preemptive voice priority, delay
constraints are more limiting than bandwidth constraints; in
fact, the service may not be feasible on some lower-speed links.
For packet-voice services, the end-to-end delay
requirements must be accounted for when determining the
maximum voice-stream capacity for a link and thresholds for
all the links in the path must be determined. This is
complicated by two issues: (i) all links (except the egress link)
are expected to support traffic other than voice, and the delay
requirements for these other applications could be much less
stringent: in the absence of strict voice priority mechanisms,
both the aggregate utilization of the link and the data traffic
characteristics (such as potential self-similarity) must be
addressed; (ii) a capacity planning process based on the sum of
individual link delay budgets equaling the end-to-end delay
requirement can potentially result in over-engineering (since
the upper bound on worst-case end-to-end delay is the sum of a
series of random variables).
 The first issue is readily mitigated by implementing the
necessary QoS mechanisms to ensure that voice is sufficiently
insulated from the data traffic; for example, using Expedited
Forwarding (EF) class in the IETF DIFFSERV model. From an
operational perspective, delay-constrained capacity planning
must make a trade-off between the complexity and expense of
the planning process and the resulting capital savings from
optimum asset utilization. The second issue can thus be
addressed by managing this trade-off. As intimated by the
partial set of results provided in the previous section, it is clear
that (in the upstream direction) queuing delays on the DSL link
will dominate. If the delay allocation to the other links could
be limited to approximately 1 ms or less, then capacity
planning process could be reduced to a simple set of
engineering guidelines; for example:
Pr[ q~home ≥ 1 ms] < 10 −5 ,
 10 ms G.711
Pr  q~dsl ≥  −5
 < 10 ,
  7 . 5 ms G . 729 
Pr[ q~uplink ≥ 1 ms] < 10 −5 ,
H non-TDM interfaces (e.g. Ethernet), and are potentially
Pr[∑ q~h ≥ 1 ms] < 10 −5
h =1
Results obtained but not shown indicate that when voice is
given priority over data on links with speeds at or above a DS3,
a 1ms delay budget can be easily achieved without significantly
limiting voice-stream capacity below the bandwidth-based
constraint; thus, there is no impact on the utilization of those
assets (see Figure 3 for example capacities). Those results also
indicated that when non-DSL links are constrained to 1ms and
network topology limits the number of IP network hops to less
than five, the error that would be introduced by not accounting
for the end-to-end delay being a sum of random variables is
negligible (from a network engineering perspective). Delay-
constrained capacity planning can thus be simplified to the
form presented in Equation 5.
Voice Stream Capacity
Delay-Limited
Bandwidth-Limited
(Preemptive Priority)
Link Type
Figure 3 DSLAM Uplink: Delay-Constrained Link
Capacity
C. Operational Considerations for Streaming Media over
DSL
Packet networks, by their nature, introduce variations in
delay depending on traffic conditions. Variance in frequency
or delay variation—jitter—is typically addressed through
buffering, adding to the end-to-end delay of a service,
counteracting the effect of QoS.
Introduction of QoS-based real-time streaming services
may necessitate service level agreements (SLAs) or objectives
(SLOs). In both cases, time of day stamping is required with
precision in order to establish a suitable measurement
infrastructure to support the relevant delay metrics. Similarly,
devices beyond the domain of the network synchronization
distribution may require some network reference in order to
properly ascertain one-way delay, enhance QoS, or provide a
timing reference for circuit-packet boundary devices or the
enterprise edge.
(5)
Stable frequency distribution has typically been provided
by physical layer mechanisms in the TDM network. Such
mechanisms are not always available in packet networks with
expensive to retrofit. The Network Time Protocol and IEEE
1588 provide mechanisms to distribute Time of Day
information, if stable local frequency references are available.
Tolerances for time and frequency references for the SLA
measurement infrastructure are not yet standardized.
IV. CONCLUSIONS AND FURTHER WORK
In this paper, the implications for the management of voice
delay in a DSL-based multimedia service offering has been
presented and an approach to capacity planning that makes a
trade-off between the operational expense associated with
complexity and the capital savings associated with optimal
asset utilization has been proposed.
Whereas some assumptions were made to simplify the
number of elements contributing to delay, the methodology
highlights how other sources of delay, such as interleaving and
link bonding, can have a significant impact of the ability to
meet a 50 ms end-to-end delay target. When interleaving is
used, typical DSL equipments offer settings in the range 0-16
ms and thus the additional delay cannot be ignored. Link
bonding arrangements such as G.Bond introduce a fixed delay
of less than 1 ms, but higher layer link bonding approaches are
expected to introduce greater delays as well as potential
interactions with link scheduling mechanisms supporting QoS.
Delay-Limited While G.Bond may have negligible impact on delay,
(Non-preemptive Priority)
alternative approaches may require further analysis.
The impact of preemption on the best effort data traffic
class is also a subject for further study. Alternative approaches
including MTU size adjustments are also being considered.
This paper focused on the capacity planning methods for the
case of one delay sensitive service (voice) and one delay
tolerant service (best effort). Further analysis is ongoing to
support the multi-service scenarios with multiple-delay
sensitive services (voice, video), with different traffic
characteristics. Entertainment services, in particular, raise
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questions regarding capacity planning methods for supporting
multicast multimedia streams.
Efforts to improve peak bandwidths by reducing loop
lengths will reduce the node size at the first aggregation point.
While efficient network operation is important, traditional
trunking efficiency metrics may be inappropriate in a multi-
service environment. Rather than simply considering the
bandwidth required at each stage of aggregation, it may be
more appropriate to consider distance sensitive metrics for
network efficiency.
In terms of future work, we note that in certain
circumstances the delay characteristic of service controls may
actually be as or more constraining than the delay characteristic
of the service packets themselves. For example, calls for
changes in per-flow policy may result in abandoned sessions or
web-sites if the set-up time of an enabling policy is too long.
Similarly, too long a network response to requests for joining
or leaving multicast streams may generate a very negative
impression among subscribers. We note that anecdotally
multicast video services hold as an objective channel change
response times of less than one second. Channel change delays
in excess of 1.5 seconds often result in a lowered MOS score
for stream QoE and, subsequently, elevated subscriber churn.
Whether there is a capacity constraint implied by such a
signaling delay constraint is a subject for further study.
ACKNOWLEDGMENT
The authors wish to acknowledge the many helpful
comments and insights provided by colleagues and reviewers.
REFERENCES
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of QoS enabled IP services”, TR-59 Rev. 1, Sept. 2003
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[8] ITU-T Recommendation G.114 (2003) "One-Way Transmission Time".
[9] ITU-T Recommendation G.107 (2003) "The E-Model, a computational
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[10] ITU-T Recommendation Y.1541 (2002) "Network Performance
Objectives for IP-based services".
[11] Draft ITU-T Recommendation G.Bond, work in progress.
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[13] S. Wright, S. Tischer, "Architectural Considerations in Online Game
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[14] IEEE Std 1588-2002 “IEEE Standard for a Precision Clock
Synchronization Protocol for Networked Measurment and Control
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[15] D.Mills “Network Time Protocol (Version 3) Specification,
Implementation and Analysis”, March 1992, RFC 1305
[16] J.Bellamy, “Digital Telephony”, 2nd Ed., Appendix D Traffic Tables, p.
541 J.Wiley & Sons, Inc., New York NY.1990

Ddec Dplay Degress

∑D
h∈Path
h
Duplink Ddsl Dhome Denc+ Dpack

Gateway Client
Play-out
De- CO LAN DSLAM DSL Home
Decode IP DSLAM Modem Network Packetize Encode
packetize (Egress) Router Router Uplink Link
Buffer Network

Central Office Central Office Customer Premise

G.711 G.711
RTP 8 bytes 8 bytes RTP
20 bytes 20 bytes
UDP 12 bytes 40 bytes 12 bytes UDP
IP 20 bytes IP 20 bytes IP 20 bytes IP 20 bytes IP
MPLS (4 bytes) MPLS
L2TP 8 bytes L2TP PPP 2 bytes PPP PPP 2 bytes PPP
PPPoE 6 bytes PPPoE
802.3 MAC 29 bytes 802.3 HDLC 12 bytes HDLC HDLC 12 bytes HDLC Ethernet 16 bytes Ethernet 802.3 29 bytes 802.3 MAC
RFC 2684 8 bytes RFC 2684

AAL5 8 bytes AAL5

ATM 65 bytes ATM
PHY PHY PHY PHY PHY PHY PHY PHY PHY PHY
69 bytes 54 bytes 54 bytes 105 bytes 69 bytes

Figure 4: Voice Transport Model and Transport Protocol Overhead

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