36 IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS, VOL. 24, NO.
1, JANUARY 2006
An Analysis of VoIP Service Using 1 EV-DO
Revision A System
Qi Bi, Senior Member, IEEE, Pi-Chun Chen, Yang Yang, and Qinqing Zhang, Senior Member, IEEE
Abstract—While voice-over-Internet protocol (VoIP) on wireline
network is maturing, VoIP on wireless mobile network is still in its
infancy. This disparity is due to the fact that the wireline band-
width is abundant and can be traded off for delay performance
and overhead, whereas bandwidth in wireless mobile network is
still a scarce resource. With the deployment of 1 EV-DO revision
0 (DOr0) worldwide, the spectrum efficiency has been significantly
improved. However, DOr0 still lacks of features essential for VoIP.
For this reason, 1 EV-DO revision A (DOrA) has been standard-
ized in the 3GPP2 with many improvements favorable for VoIP
implementation. In this paper, we identify challenges and explore
the feasibility of implementing VoIP using DOrA. We develop both
analytical and simulation models to evaluate the VoIP capacity and
delay performance over the air interface.
Index Terms—Air interface capacity, CDMA2000, voice-over-
Internet protocol (VoIP), wireless communication, 1 EV-DO.
I. INTRODUCTION
V OICE-over-Internet protocol (VoIP) has made consider-
able progress in wireline networks in the last decade [1],
[2]. VoIP in wireless has drawn much interest recently because
of the convergence of all IP architecture in wireless and wireline
networks. Initial VoIP efforts [3], [4] were focused on the wire-
less local area networks (LANs) since the achievable data rate is
close to that of the wireline network. Call admission and band-
width allocation are designed to support large voice capacity
and to meet the delay requirements.
Despite the success of VoIP in wireline and wireless LAN net-
works, little progress was made on wireless cellular networks.
This is because VoIP implementation adds considerable IP and
other overhead, which decreases spectral efficiency. In addi-
tion, VoIP service requires stringent end-to-end quality-of-ser-
vice (QoS) support that is not yet available to ensure the tight
delay constraint. With the deployment of 1 EV-DO revision
0 (DOr0) [5], [6] worldwide, the spectrum efficiency for wire-
less mobile data applications has been significantly improved.
However, DOr0 still lacks the capabilities essential to VoIP im-
plementation. Recognizing this, 1 EV-DO revision A (DOrA)
[7] has been standardized with many improvements favorable to
VoIP implementation.
In this paper, we explore the possibility of implementing VoIP
using DOrA. We identify the challenges that hinder the imple-
mentation of VoIP on wireless network in general, and inves-
tigate possible solutions to meet these challenges. We develop
Manuscript received October 7, 2004; revised June 3, 2005.
The authors are with Bell Laboratories, Lucent Technologies, Whippany,
NJ 07981 USA (e-mail: qbi@lucent.com; pc2000@lucent.com; yyang7@
lucent.com; qinqing@lucent.com).
Digital Object Identifier 10.1109/JSAC.2005.858882
0733-8716/$20.00 © 2006 IEEE
simulation and analytical models to evaluate the expected per-
formance including voice capacity and resultant delays when
VoIP is implemented using DOrA.
II. VOIP IMPLEMENTATION USING 1 EV-DO DORA
When VoIP is implemented using DOrA, the coverage, ca-
pacity, and voice quality should be similar to that provided by
the CDMA2000 system. These requirements present challenges
to system designs. In this section, we examine some of the so-
lutions and establish assumptions, design criteria, and the resul-
tant system model to facilitate the analysis that follows.
A. Speech Coder and Silence Suppression
In this paper, we assume that an enhanced variable rate codec
(EVRC) speech coder [8] is used for VoIP. The EVRC speech
encoder generates frames with four variable data rates, i.e., full,
1/2, 1/4, and 1/8 rate, with the probabilities of 38%, 5%, 0%,
and 57%, respectively. Further, silence suppression is assumed
by not or rarely transmitting the 1/8 rate packets, resulting in a
43% voice activity factor.
B. IP and Other Packet Overhead
For VoIP applications, voice frames have to be placed into
IP packets. The added protocol overhead represents an in-
tolerable amount of spectral inefficiency for wireless mobile
networks. For this reason, IP header compression is necessary.
Header compression algorithms have been standardized for
CDMA2000 network and DOrA. We assume that header com-
pression is utilized for both forward link (FL) and reverse link
(RL). To demonstrate the importance of header compression,
consider a full rate EVRC frame that has 171 bits. Together
with 24 cyclic redundancy bits and six coding tail bits, the total
payload is 201. Without header compression, the overhead will
be about 360 bits, which is more than the payload, reducing the
spectral efficiency significantly. If the header compression is
designed in such a way that all overhead including IP overhead
can be fit into 55 bits, the encoder packet size will be 256 bits,
resulting in an overhead of only 33% when compared with
circuit-switched voice frame.
C. End-to-End QoS Support
To ensure end-to-end QoS in a packet-switched network, a
series of features have been defined in DOrA to meet the need
for different applications. Main QoS support can be categorized
into two categories: signaling mechanisms [9] to support the
call features similar to the current cellular voice calls, and trans-
port mechanisms [10] to meet comparable quality of the cellular
BI et al.: AN ANALYSIS OF VOIP SERVICE USING 1 EV-DO REVISION A SYSTEM
voices. The end-to-end QoS framework for DOrA is currently
under standardization by 3GPP2. In this paper, we assume that
the QoS architecture and mechanism [11], [12] are in place to
differentiate the VoIP flow from other applications.
D. FL Multiple Packet Size Support
In DOr0 [5], each requested data rate is associated with only
one packet size. When the sector schedules a packet, it must
transmit the specific packet size corresponding to the mobile
requested data rate regardless of the data backlog situation. As-
sume that a mobile is at a good RF location and indicates that
a high data rate corresponding to 4096-bit packet size can be
used. Since VoIP is a low data rate service, only a few hun-
dred bits will be in the buffer at each transmission. In this case,
the scheduler will have to pad the rest frame till 4096 bits. This
mismatch causes inefficient use of FL RF resource. DOrA has
the improvement by defining multiple packet sizes associated
with each requested data rate. The shorter packets have stronger
channel coding structure and, hence, better performance. The
sector has the flexibility to choose a packet size to match the
data backlog and reduce the packet transmission delay.
E. FL Packets Multiplexing From Multiple Users
The DOr0 system is optimized to support a small number
of users downloading large amounts of data from the network.
For VoIP, this assumption is violated since there might be many
users transmitting at a very low data rate. For example, an EVRC
speech coder generates 50 packets/s, and there are a total of 600
time slots per second on the FL. Therefore, each sector can only
support less than 12 VoIP users if each voice packet needs to be
transmitted within a 20 ms delay.
To overcome the shortage of time slots, DOrA introduces a
multiuser packet (MUP) feature that allows packets from up to
eight users to be packed into a single physical-layer packet. The
decision to transmit a MUP is dynamically made by the sector
on a packet-by-packet basis. Considering a VoIP packet, a MUP
packet of 1024 bits can practically accommodate up to four-user
packets, while MUP of larger sizes can support a maximum of
eight packets. Since certain data rates only support MUP packets
up to 1024 bits, it presents a practical limit of four-user MUP
packets capacity. In addition, a user’s data rate needs to be at
least 153.6 kb/s to qualify for any MUP operation. At lower rate,
only a single-user packet (SUP) can be transmitted.
F. RL HARQ OPERATION
One of the major improvements of DOrA over DOr0 is the
adoption of the hybrid ARQ feature on the RL. Each subframe
(SF) spans four time slots, and is associated with an interlace
channel index of 1, 2, and 3. The maximum number of allowed
SF transmissions is four SFs per packet. To minimize delay for
the VoIP application however, each VoIP packet needs to com-
plete its transmission within three SF transmissions. This is be-
cause there may be three voice frames arriving every 60 ms from
the EVRC source and there are nine SF in the same time period.
If each packet requires no more than three SF to complete, these
three voice frames can be completed in nine SF. Otherwise, the
37
voice frame following these three voice frames will incur addi-
tional queuing delay.
G. RL Resource Management Control
The RL link of the DOrA system is designed to tolerate a
given received interference power level. The interference power
is measured by the ratio of total received power to thermal noise
power, and is referred to as rise-over-thermal (RoT). Gener-
ally, a system operated at a higher RoT will have higher RL
capacity; however, the tradeoffs in operating at a high RoT are
higher user transmit power and a greater chance for power con-
trol instability.
III. VOIP PERFORMANCE CRITERION AND
SIMULATION ASSUMPTIONS
To study the delay and capacity performance of VoIP using
DOrA, theoretical analysis and computer simulations were car-
ried out. In this section, we summarize the assumptions and the
system model used for the VoIP performance investigation.
A. VoIP Performance Criteria
For voice service, the acceptable delay guideline has been
studied extensively and been summarized in the G.114 Stan-
dard. At this point, it is not yet clear what guideline will be
appropriate for the VoIP latency bound. The complication
stems from the fact that VoIP packet delay varies on a packet
basis, whereas the guideline from G.114 is obtained with fixed
end-to-end delay. To proceed with this study, we assume that
the frame erasure rate (FER) due to packet loss and packet
delay exceeding the target latency bound be kept within 2%.
Further, at least 98% of VoIP users in the network should meet
the above criterion, and the interference level in terms of RoT
should be kept below a given threshold.
B. Simulation Setup
A computer simulation was carried out to verify the theoret-
ical analysis and provide more comprehensive and detailed in-
formation regarding VoIP performance and system capacity. In
each run of the simulation, a certain number of mobiles are ran-
domly placed into the network with uniform area distribution.
The number of users per sector is, thus, a Poisson random vari-
able. Erlang capacity for VoIP is defined as the average number
of users per sector. The values of the system parameters used in
the simulations follow the standard recommendation [13].
IV. ANALYSIS OF VOIP ON FORWARD LINK (FL)
A. Analytical Model
In this section, we utilized the field measurements of the ex-
isting DOr0 system as reported in [19] to estimate VoIP capacity.
A simplified first-in–first-out (FIFO) scheduling model with ex-
tended bulk-service was proposed to analyze the delay and ca-
pacity performance on the FL. The FIFO scheduler was selected
to provide consistent delay and jitter performance for all VoIP
users across the RF conditions. As shown in Fig. 1, a Poisson
arrival with rate represents the aggregate traffic destined for
all VoIP users in the sector.
38 IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS, VOL. 24, NO. 1, JANUARY 2006
Fig. 1. FIFO queueing model for FL scheduler.
The arrival consists of three types of users based on different
RF conditions of the users.
•Type 1 represents traffic from users that do not qualify
for MUP operation, but multiple VoIP packets from the
same user can be transmitted in the same packet. To take
this into account, the analysis makes a minor modification
on the FIFO scheduler assumption. If there are multiple
VoIP packets destined to the same mobile in the queue,
the scheduler will transmit up to four packets together.
Defining as the average number of VoIP packets con-
tained in a Type 1 transmission, the effective traffic arrival
rate of Type 1 traffic to the scheduler is reduced by
times. The estimation of will be given later in (14).
• Type 2 represents traffic from users that qualify for the
MUP operation with up to four users per packet.
• Type 3 represents traffic from users that qualify for the
MUP operation with up to eight users per packet.
Denoting as the probability of Type traffic with
, the arrival rate for Type traffic can be written as
. As explained above, the effective traffic arrival rate
to the scheduler is . The normalized
probability for the effective traffic arrival to the scheduler
is
(1)
The bulk server serves the users in FIFO fashion, and the
service rate depends on the type and number of users in the
queue. The rule is defined as follows:
• when serving a SUP;
• , , when serving a MUP containing
type 2 or type 3 users;
• , , when serving a MUP containing
type 3 users.
In the following, we discuss the estimation of parameters used
in the model.
• The aggregate VoIP packet arrival rate
(packets/slot), where 43% is the voice packet activity
factor with silence suppression, 50 is the number of voice
frames per second from the EVRC vocoder, 600 is the
number of slots per second in DOrA, as defined in the
standards [7], and represents the average offered voice
traffic load in the sector in Erlangs. Defining as the
probability for the data rate control DRC index ,
, it is easy to see
(2)
according to [13, Table 11.7.6.1–2]. For the analysis we
used the field measurements of provided in [18]. A
simplified assumption is made that user requested DRC
rate does not change over time, so the user in poor RF
condition never recovers. This assumption is likely to lead
to conservative capacity projections, as the actual user’s
RF condition is dynamic and DRC rates adapt to the RF
conditions.
• The bulk-service rates , , can be written as
, where represents the control channel
overhead on the DOrA FL. 10% is assumed in the
analysis. is the average packet transmission length in
slots when users are contained in the packet and can be
evaluated based on the service rule described above.
• The scheduler transmits a SUP if any of the following
conditions are satisfied.
— There is only one user in the queue. In this case
for state (3)
where and are the average packet transmis-
sion length when the packet contains Type 1 traffic
and Type 2 or 3 traffic, respectively.
— There are multiple users in the queue, but the
head-of-line (HoL) user is a Type 1 user, or the user
next to the HoL is a Type 1 user. In this case
for state (4)
and can be evaluated from field measurement
on packet transmission performance
(5)
where is the probability that a SUP of DRC
index is transmitted successfully in slots. is
the number of nominal slots defined by the standard
[7] for DRC index . Table I shows some DOr0
field measurements on that is used for this
analysis. Note that some data rates are defined only
in one slot and are not shown in the table.
• The average packet transmission performance for a MUP
depends on both the number of users included in the trans-
mission and the RF conditions of these users, since a MUP
transmission can only be terminated when positive ac-
knowledgment (ACK) for all users targeted in the trans-
BI et al.: AN ANALYSIS OF VOIP SERVICE USING 1 EV-DO REVISION A SYSTEM 39

Fig. 2. State transitions from and to state m of continuous time Markov chain.

TABLE I TABLE II
DOR0 FIELD MEASUREMENT ON fb (k )g VALUES OF fh g

of is governed by the continuous-time Markov state transition

shown in Fig. 2.
Several assumptions are made to construct the Markov model.
• The discrete slot-based packet scheduling operation is ap-
proximated in the continuous time domain.
• The distribution of the type of users in the queue is depen-
dent on scheduling decisions.
Defining the PMF of as , it satisfies the following
equations:
mission are received, or the full HARQ process has been
exhausted. It can be shown that

(9)
(6)

where is the conditional probability that the scheduler

serves users when there are users in the queue and takes its
(7) value from Table II. For example, if the scheduler serves five
users when there are more than five users in the queue, it means
the five HoL users are all Type 3 users, while the sixth user is
not. Hence, for .
is the cumulative distribution function (CDF) of the Denoting the user waiting time as , its distribution
packet transmission length when the MUP contains can be evaluated from
user packets. is the nominal slot of the MUP.
is the modified probability mass function (PMF) (10)
of packet transmission length of DRC index , taking into
account of the effect of the missed ACK for the MUP
where . and
transmission
are the Laplace transform of and ,
respectively. represents the Laplace
transform of bulk-service time distribution when users are
(8)
served together. The system utilization is then the server busy
probability

where is the ACK miss rate during the MUP transmis- (11)
sion. is assumed in the study.
Meanwhile, the average number of VoIP packets that are packed
B. VoIP Performance Analysis into a Type 1 packet need to be estimated to facilitate the
analysis. As previously described, can be approximated as
We now evaluate the FL VoIP user performance. Denoting
the number of users in the scheduler queue as , the behavior (12)
40 IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS, VOL. 24, NO. 1, JANUARY 2006

where is the number of VoIP packets in the queue belonging

to the same Type 1 mobile when that mobile is scheduled. As-
suming the individual user packet arrival can be approximated
in the Poisson process, the distribution of given a total of
user packets in the queue is then

(13)

Combining (11)–(13), can be evaluated as

Fig. 3. FL delay bound for 98% of users achieving less than 2% FER.
(14)

C. System Delay Bound Evaluation

Based on the capacity criterion defined in Section III-A, we
derive the system delay bound for a target FER covering
a certain percentage of the VoIP users in the network. In the
DOrA system, the target packet error rate (PER) is 10 on FL
packet transmissions. Fast physical-layer retransmission mech-
anisms can be employed to further reduce the net PER to below
10 . As a result, the contribution of packet loss to voice frame
erasures is negligible. Instead, the primary contributor to voice
frame erasures is the excessive packet delay beyond the system
delay bound.
The FL packet latency consists of two parts.
Fig. 4. Average number of users packed in a physical packet.
• The queueing delay at the scheduler under a certain VoIP
traffic load. With the FIFO scheduler, the packet waiting
time experienced by different users is similar regardless of where and is the Laplace transform of and
their RF conditions, and the distribution is given by (10). , respectively. The system delay bound is identified as
• The packet transmission latency over the air. It depends on
the packet transmission rate, hence, the RF condition of
the user. The users in the worst RF conditions experience (18)
the worst packet latency performance.
We will mark the user who has the worst performance out of
-percent of the total users in the network as . With the sim- D. Analysis and Simulation Results
plified assumption that a user’s RF condition does not change,
the packet transmission rate index for user can be identified Fig. 3 shows the VoIP capacity estimate from the analysis and
as simulation under the quality criterion of 2% and capacity
criterion of 98%. It can be seen that VoIP capacity gain
in voice loading diminishes with higher delay-bound due to the
and (15) nonlinear increase of the queueing delay. From the curve, it is
estimated that about 35 Erlangs can be supported with a 70 ms
delay bound on the FL if the handsets are equipped with two
The distribution of the packet transmission duration (in slot receiving antennae.
units) in the interlaced structure can be obtained as Fig. 4 shows the MUP efficiency in terms of the average
number of VoIP packets contained in one physical-layer packet.
(16) As the key technique of VoIP transmission in the DOrA air in-
terface, it is clear from the figure that MUP operation plays a
critical role in absorbing the loading with more VoIP traffic.
where . Combining (10) and (15), the FL packet In Fig. 5, the lower curve shows the sector throughput when
latency distribution for user can be obtained as only voice bits are included. The middle curve shows the sector
throughput when voice bits, the overhead bits and the padding
(17) bits are included. The upper curve shows the sector throughput
BI et al.: AN ANALYSIS OF VOIP SERVICE USING 1 EV-DO REVISION A SYSTEM 41

In this expression, is the number of active users in the

current sector, is the other cell interference factor, is the
total bandwidth (in Hertz), is the required bit energy
to interference-plus-noise ratio for user , is the bit rate,
is the percentage of chip energy from the overhead channels
relative to the chip energy from the traffic channel, is the
percentage of the chip energy from the control channel relative
to the chip energy from the traffic channel, and is a binary
random variable indicating whether user is transmitting data
traffic or not. The system outage probability is, thus

Fig. 5. Aggregate sector throughput performance.

(20)
when all bits are included and all time slots are fully utilized.
From Fig. 4, the VoIP throughput efficiency is rather poor, since where is typically between 0.1–0.25, which corresponds to a
a significant gap exits between the lower and the middle curves. RoT of 6–10 dB. In this expression, the number of users is
This implies that the packets are often filled with significant a Poisson random variable with a mean of based on the
number of padding bits. This situation can be improved when lost call held (LCH) model [14], where is the average call
the system supports both VoIP traffic and best effort data. In arrival rate and is the average call holding time. is a bi-
this case, padding bits can be replaced by the best effort data nary random variable depending on the traffic activity factor.
bits, thus increasing the RF packing efficiency and application The required bit energy to interference-plus-ratio is
throughput. a random variable, depending on the coding and modulation of
Further observation can be made that there is significant the encoder packet, the target PER, and the number of trans-
reduction in sector throughput with VoIP throughput when missions introduced by the HARQ operation. The bit rate is
compared with the best effort packet data throughput. In [19], also a random variable depending on the rate control algorithm.
about 1.1 Mb/s sector throughput can be achieved when sup- The objective is to estimate the Erlang capacity that the
porting best-effort data services. When supporting VoIP, the system can provide under the outage probability.
sector throughput decreases to less than 500 kb/s. The loss is
due to scheduling the delay-sensitive VoIP packets. For best B. Calculation of the System Outage Probability
effort data packets, the scheduler is able to optimize the sector From (20), the system outage probability can be written as
throughput based on users RF condition. However, in the case
of VoIP, the scheduler is forced to serve users in poor RF
conditions under time constraint to meet the stringent delay
requirement, hence, the reduction in the sector throughput.
(21)
V. ANALYSIS OF VOIP ON REVERSE LINK
where .
A. Theoretical Analysis for VoIP Capacity Let denote the maximum number of transmissions for an
For VoIP services on the RL, each user is assigned its own encoder packet. The required to meet a target packet
high-speed data traffic channel for the entire call. Therefore, the error rate (PER) depends on the coding and modulation scheme,
traditional queueing analysis still applies to the evaluation of the and the number of transmissions. For a particular encoder packet
Erlang capacity of the system. We expand the Erlang capacity with corresponding rate , the will take a finite number
analysis for the circuit-switched voice in a code-division mul- of discrete values with probability
tiple-access (CDMA) system [14]–[16] and further enhance the
analysis by considering the effect of HARQ on system capacity. (22)
The Erlang capacity is constraint of the system outage
probability. The system outage probability is defined as the
probability that the rise over thermal RoT exceeds a certain and , where , , is the prob-
threshold. The RoT is often expressed as , where ability of an encoder packet transmitted in subpacket, and
is the so-called loading of the sector. By further derivation is the required for the encoder packet after
and decomposition, the sector loading can be expressed as transmissions to meet the target PER.
The transmission rate is also a random variable. Let de-
note the size of the transmission rate set. The probability of is
(19)
(23)
42 IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS, VOL. 24, NO. 1, JANUARY 2006

and , where , is the proba- TABLE III

bility of a user choosing rate . PARAMETERS FOR VOIP ERLANG CAPACITY ANALYSIS
Therefore, the probability distribution function (PDF) of
will take an number of discrete values as

(24)
TABLE IV
Since the required for an encoder packet is usually PROBABILITIES OF NUMBER OF TRANSMISSIONS
represented by the aggregated value over the total number of
transmissions, the average per subpacket can be com-
puted as the aggregated divided by the total number of
transmissions. Thus, (24) becomes

where is the mean of the number of users per sector. For

(25) simplicity we consider the average other cell interference level,
which is modeled as a ratio of the average number of users in
where the current sector. In other words, the other cell interference is
The Activity Factor: The activity of the channel depends on assumed to be in the variance calculation.
two factors: the on–off activity of the data traffic source, and The outage probability is given as
the scheduling of the packet transmission over the packet data
channel. Let denote the probability that the source is gen- (29)
erating a traffic encoder packet, and denote the probability
that a subpacket of the encoder packet is being transmitted over where is the error function; and are given by (27) and
the channel. depends on the number of subpacket transmis- (28), respectively.
sions for the encoder packet. It equals the ratio of the number
C. Erlang Capacity for VoIP in DOrA
of transmissions to the maximum number of transmissions per
encoder packet . The conditional probability of We apply the analytical model as described in the previous
given the number of transmissions is section and evaluate the Erlang capacity for VoIP in DOrA. The
parameters used for VoIP Erlang capacity analysis are summa-
(26) rized in Table III.
From the link level simulations with 1% target PER after
The Outage Probability: When the number of users is
a maximum of three transmissions, the probabilities of the
large, the left side of the inequality (21) or the loading can be
number of transmissions and corresponding are shown
approximated as a Gaussian random variable with mean and
in Table IV.
variance . Assuming , , and are independent random
Pilot, DRC, RRI, and DSC are overhead channels, which are
variables, the mean is
always active when the user is admitted to the system. With the
traffic to pilot ratio values in Table III, the overhead percentage
of the traffic channel is, thus, 0%, 36.45% for the
256-bit EP size.

D. Comparison of Analytical Results and Simulations

Using the above parameters, we compute the RoT distribution
under different Erlang capacity situations as shown in Fig. 6.
(27) The simulation results are also shown in the figure. We can see
that the analytical results match the simulation results very well
The variance is derived as
in terms of RoT distribution with different loadings. To meet the
outage probability of 1% at a RoT threshold of 7 dB, 35 Erlang
voice capacity can be supported.

E. Delay Analysis
Depending on the RF conditions, each packet may require
one to four SF to complete using HARQ. The number of SF
or equivalently, the transmission time for each packet, can be
described by a PMF as

(28)
(30)
BI et al.: AN ANALYSIS OF VOIP SERVICE USING 1 EV-DO REVISION A SYSTEM 43

Also, since , we have

(33)

In the above, we have provided the delay probability due to

unavailability of the SF. The probability density function
of the alignment delay is clearly the combination of this delay
and the delay due to arrival in the middle of a SF, which can be
written as

(34)

where is defined as the gate function that is equal to 1 in

the interval of 0 and 20 ms, and 0 elsewhere.
The VoIP packet transmission delay distribution can be ap-
Fig. 6. Comparison of analysis and simulation results of RL RoT.
proximated by the summation of the time alignment delay
and the delay due to the multiple SF transmissions of HARQ as
where is the time needed for SF transmissions. They are 6.7, given in (30). Assuming that these two delays are statistically
26.7, 46.7, and 66.7 ms, respectively. Further, is the proba- independent, the PDF for the packet transmission delay
bility for each SF due to HARQ. for the RL can be obtained from
In addition to the delays due to multiple SF transmissions
using HARQ, delays may occur when the voice frame arrives
at a time not necessarily at the boundary between two SFs, and (35)
when the next SF following the packet’s arrival is not avail-
able. Let us define the time interval between the voice frame
arrival and the boundary of the available SF as the time align- F. Computer Simulations
ment delay . In this paper, we assume that the time alignment
As discussed in previous sections, the EP size of 256 bits is
delay within a SF follows a uniform distribution.
mostly used to accommodate the small voice frame. To mini-
Without losing generality, let us assume that a voice frame
mize the transmission delay, power control and traffic channel
arrives in a middle of a SF. We denote the next three SFs after
power to the pilot (T2P) power ratio is set in such a way that al-
the voice frame arrival as a SF with interlace index 1, 2, and
most all packets complete successful transmission in three SFs.
3. To reduce transmission delay, we adopt the greedy algorithm
By aggregating packets from all users in the system, it is ob-
by which each voice frame is placed into the next available SF.
served that the probabilities that a packet completes transmis-
Now, let us define four mutually disjointed events and probabil-
sion in 1, 2, and 3 SF are 0.41, 0.48, and 0.11, respectively.
ities by considering the three SFs following the arrival of a voice
The probability that a packet needs four SFs is negligible. Sub-
frame. Let be the probability that the voice frame is placed
stituting these values in (30) and computing (35), the PDF for
into SF 1. The subscript indicates the fact that the delay due to
packet transmission delay of a typical user is plotted in Fig. 7.
the unavailability of the SF is zero. Let , , 2, and 3 be the
Shown in Fig. 7 are also the simulation results for delays of all
probability that the voice frame is placed into the SF . In
packets in the system when the Erlang capacity in each sector is
this case, we denote as the probability that all the SF’s with
26 and 35, respectively. As can be seen from the figure, the the-
index 1, 2, and 3 are not available. Precise analysis of the above
oretical analysis for the delay matches that from the simulations
probabilities is very involved and is beyond the scope of this
very well. As long as the T2P values can be controlled so that
paper. For this reason, we simplify the analysis by assuming that
the probability of the fourth SF transmission is negligible, al-
that the probability of transmitting the fourth SF using HARQ
most all packets in the system can be limited to within 66.7 ms,
is negligible . Under this condition, . To obtain
which includes the 60 ms HARQ delay and the 6.7 ms voice
we further assume that the system is in an equilibrium state
frame alignment delay.
in which the probability that the SF 1 is available is approxi-
For commercial services, however, it is usually important
mately identical for every three SF periods, although there may
to ensure the voice quality of each user, not the PER of all
be some small variations in reality. We obtain
packets in the system. Consider users whose PER is below a
given threshold, say 1%. One useful criterion is to require
(31) percent of users in the system to meet the above criterion. A
typical value for , for instance is 98%. Through simulations,
To obtain , we observe that the voice frame will be placed the PDF of users that can meet 1% FER is shown in Fig. 8 for
into SF3 only when SF1 and SF2 are both not available. This two system loadings. From the figure with a delay limit of 70
probability is given by ms, 98% of users can meet the performance criteria under the
two Erlang capacity values.
By combining Figs. 7 and 8 using the 35 Erlang curves, we
(32) obtain Fig. 9 in which the axis represents the percentage of
44
Fig. 7. Packet delay distribution comparisons of simulation and analysis.
Fig. 9. Percentage of users not meeting 2% FER versus average FER.
Fig. 8. User delay distribution with different Erlang loading.
total frame erasures in the system and axis represents the per-
centage of users not meeting the 2% FER. From Fig. 9, it can
be seen that the percentage of users not meeting the 2% FER
can be approximated by a linear function of the system packet
delay given in (35). The slope of the linear function is approxi-
mately 12 using the best-fit technique and the intercept is about
0. For example, about 60% users will not be able to meeting the
IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS, VOL. 24, NO. 1, JANUARY 2006
2% FER in the system if the average FER of the system is about
5%.
VI. CONCLUSION
In this paper, we evaluated the feasibility of supporting VoIP
service using the 1 EV-DO Revision A system. We proposed
an analytical model and carried out computer simulations to
study the possible capacity and delay performance. Based on the
analysis and simulations, the support of VoIP using 1 EV-DO
Rev. A appears technically attractive. The expected Erlang ca-
pacity is estimated to be comparable to that of a circuit switched
CDMA2000 system.
ACKNOWLEDGMENT
The authors would like to thank S. Vitebsky, A. Stoylar,
X. Wang, and the reviewers for their helpful discussions and
comments.
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2
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BI et al.: AN ANALYSIS OF VOIP SERVICE USING 1 EV-DO REVISION A SYSTEM
Qi Bi (SM’92) received the B.S. and M.S. degrees
from Shanghai Jiao Tong University, Shanghai,
China, and the Ph.D. from Pennsylvania State
University, University Park.
He is a Bell Laboratories Fellow in the Mobility
Solutions Unit, Lucent Technologies, Whippany,
NJ. He currently heads a team with responsibilities
of analyzing and designing the third-generation
wireless digital communication systems. He served
as the Guest Editor of Wireless Communications and
Mobile Computing (Wiley). He is also a recognized
leader outside of Lucent Technologies and has served as technical chair in
many international conferences. He holds more than 40 U.S. patents. His
present focus is in the areas of high-speed wireless data network delivering
VoIP, broadcast and multicast services, push to talk, and broadband wireless
communications.
Dr. Bi was the recipient of numerous honors including the Advanced Tech-
nology Laboratory Award in 1995 and 1996, the Bell Laboratories President’s
Gold Award in 2000 and 2002, The Bell Laboratories Innovation Team Award
in 2003, the Speaker of the Year Award from the IEEE New Jersey Coast Sec-
tion in 2004, and the Asian American Engineer of the Year Award in 2005.
He has served as the Technical Vice-Chair of the IEEE Wireless Communica-
tions and Network Conference 2003, Technical Chair for Wireless Symposium
of the IEEE GLOBECOM 2000–2002, and organizer of the First and Second
Lucent IS-95 and UMTS Technical Conference in 1999 and 2000. He served
as Feature Editor of the IEEE Communications Magazine (2001), Editor of
the IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS and the IEEE
TRANSACTION ON WIRELESS COMMUNICATIONS.
Pi-Chun Chen received the M.S. and Ph.D. degrees
in electrical engineering from the Wireless Informa-
tion Laboratory (WINLAB), Rutgers University, Pis-
cataway, NJ, in 1996 and 1999, respectively.
Upon her graduation, she joined Applied Re-
search, Telcordia Technologies, as a Research
Scientist conducting research in the area of advanced
wireless technology analysis. Her work included
the packet data protocol design to provide the
evolution path of the Personal Access Communi-
cations Systems (PACS) to 3G system and network
deployment tools’ algorithm. Since November 2000, she has been Member
of Technical Staff at Bell Laboratories, Lucent Technologies, Whippany,
NJ. She is responsible for system level performance analysis and algorithm
design for 3G communication systems. Her research interests include radio
resource management for QoS services, power control, handoff, random access
performance, admission, and congestion control.
45
Yang Yang received the B.S. degree from the Uni-
versity of Science and Technology of China, Hefei,
China, in 1994, and the M.E. and Ph.D. degrees
from Stevens Institute of Technology, Hoboken,
NJ, in 1997 and 1999, respectively, all in electrical
engineering.
Since 1999, she has been with Lucent Tech-
nologies, Whippany, NJ, as a System Engineer.
Her current research interests and activities are
focused on the performance analysis of CDMA2000
2
and 1 EV-DO systems, algorithm designs for air
interface control and optimization, traffic modeling, and engineering of the
wireless networks.
Qinqing Zhang (S’95–M’98–SM’03) received the
B.S. and M.S.E. degrees in electronics engineering
from Tsinghua University, Beijing, China, and the
M.S. and Ph.D. degrees in electrical engineering
from the University of Pennsylvania, Philadelphia.
She is Member of Technical Staff at Bell Labo-
ratories, Lucent Technologies, Whippany, NJ. Since
joining Bell Laboratories in 1998, she has been
working on the design and performance analysis of
wireline and wireless communication systems and
networks, radio resource management, algorithms
and protocol designs, and traffic engineering. She is also an Adjunct Assistant
Professor in the Department of Electrical and System Engineering, University
of Pennsylvania. She is the coauthor of Design and Performance of 3G Wireless
Networks and Wireless LANs (Springer, 2005). She has published numerous
papers in IEEE journals and conferences. She has been awarded 6 patents and
has 14 patent applications pending.
Dr. Zhang serves on the Editorial Board of the IEEE TRANSACTIONS ON
WIRELESS COMMUNICATIONS and on the technical program committees of
various IEEE conferences.