Voice-over-IP Performance in UTRA Long

Term Evolution Downlink

Jani Puttonen1, Tero Henttonen2, Niko Kolehmainen3, Kennett Aschan2,
Martti Moisio2 and Petteri Kela3
1

Magister Solutions Ltd, c/o Mattilanniemi 6-8,

40101 Jyvskyl, Finland. Email: firstname.lastname@magister.fi
2

Nokia, P.O.BOX 45, FIN-00045 Nokia Group, Finland

Email: firstname.lastname@nokia.com

University of Jyvskyl, Dept. of Mathematical Information Technology

P.O. Box 35, 40014 University of Jyvskyl, Finland. Email: firstname.lastname@jyu.fi
AbstractIn this paper, we study Voice-over-IP (VoIP)
performance in UTRA Long Term Evolution (LTE) Downlink
(DL). We have utilized fully dynamic system simulations to study
the VoIP Adaptive Multi-Rate (AMR) 12.2 codec capacity in four
different 3GPP simulation cases. The effects of Link Adaptation
(LA), packet bundling, control channel capacity and number of
HARQ processes on VoIP capacity have also been considered.
The results present the absolute VoIP capacity numbers of LTE
DL. We also show that LA together with packet bundling
provides clear gain on the VoIP capacity, because more VoIP
packets can be scheduled in each TTI. Also, the control channel
limitations can be effectively compensated by packet bundling.
Index TermsVoIP, LTE, system simulations

I. INTRODUCTION
The Evolved UTRAN (E-UTRAN) or the UTRAN Long
Term Evolution (LTE) specifications are being finalized in
3GPP. LTE aims at ambitious goals of e.g. peak data rate of
100 Mbps in downlink and 50 Mbps in uplink, increased cell
edge user throughput, improved spectral efficiency, scalable
bandwidth from 1.25 MHz to 20 MHz, etc. [1].
The main principles of E-UTRA downlink, uplink and the
core network have been decided already. LTE supports both
time (TDD) and frequency division duplex (FDD) modes, but
in this article we concentrate on FDD. Orthogonal Frequency
Division Multiple Access (OFDMA) has been selected for the
downlink multiple access technology and Single Carrier
Frequency Division Multiple Access (SC-FDMA) for uplink
[1]. To achieve the objectives set for LTE, advanced Radio
Resource Management (RRM) functions have been defined.
The algorithms include e.g. Hybrid ARQ (HARQ), Link
Adaptation (LA), Channel Quality Indication (CQI) and
Packet Scheduling (PS). More on these can be found e.g. from
[2].
E-UTRAN is optimized for packet data transfer and the
core network is purely packet switched, so speech is
transmitted purely with Voice-over-IP (VoIP). VoIP traffic
consists of talk-spurts and silent periods, with relatively small

978-1-4244-1645-5/08/$25.00 2008 IEEE

packets transmitted quite rarely. The Adaptive Multi-Rate

(AMR) codec provides quite bursty traffic; one VoIP packet at
20 ms intervals during talk spurt and one Silence Indicator
(SID) packet at 160 ms intervals during silence period. EUTRAN is expected to support a very high number of VoIP
users and the Quality-of-Service (QoS) for VoIP is determined
by maximum End-to-End delay and tolerable packet loss.
These facts set challenges to the resource allocation of VoIP
users, for both PS and LA algorithms. Also, the capacity of
Physical Downlink Control Channel (PDCCH) induces some
restrictions, at least with higher system bandwidths. These
restrictions become most relevant with dynamic packet
scheduling, since each allocation consumes signaling
resources from PDCCH. Thus, several persistent resource
allocation schemes, such as fully persistent scheduling, talkspurt based persistent scheduling and semi-persistent
scheduling have also been proposed in 3GPP [3]. However,
these scheduling types limit the gain from multi-user and
frequency domain scheduling. VoIP service in E-UTRAN has
been studied e.g. in [4] and [5].
The objective of this article is to provide the baseline VoIP
performance results of E-UTRAN FDD downlink with
dynamic packet scheduling. The effect of different features,
such as system bandwidth, LA, Control Channel (CC)
capacity, packet bundling and HARQ processes, on VoIP
capacity are studied using simulations. The simulation results
are gathered from fully dynamic system simulator, which
models the UE mobility, RRM functionalities and their
interactions with the system.
The paper is organised as follows: Chapter II discusses the
general aspects of VoIP in LTE and related modeling. Chapter
III lists the simulation assumptions including a short
description of the simulator. Chapter IV presents the
simulation results and analysis. Finally, Chapter V reviews the
main conclusions.

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II. VOICE-OVER-IP IN LTE

VoIP has at least three characteristics that need
consideration in LTE (as well as in any wireless system):
Bursty low bitrate traffic, strict packet delay-based QoS and
high number of simultaneous users). These issues set
challenges to the RRM functions. Next, we discuss these
characteristics as well as the required RRM functions in more
detail.
A. High Capacity Demand
The requirements of E-UTRA and E-UTRAN are described
in TR.25.813 [6]. The service related requirements for VoIP
are:
The E-UTRA should efficiently support various types
of service. These must include currently available
services like web-browsing, FTP, video-streaming or
VoIP, and more advanced services (e.g. real-time video
or push-to-talk) in the Packet Switched domain.
VoIP should be supported with at least as good radio
backhaul efficiency and latency as voice over UMTS
Circuit Switched (CS) networks.
Voice and other real-time services supported in the CS
domain in Release 6 shall be supported by E-UTRAN
via the packet switched domain with at least equal
quality as supported by UTRAN (e.g. in terms of
guaranteed bit rate) - over the whole speed range.
B. Strict packet delay-based QoS
The system capacity for VoIP service is limited by the
outage limits defined in TR 25.814 [1] and updated in 3GPP
contribution R1-070674 [7]:
The system capacity is defined as the number of users
in a cell when more than 95% of the users are satisfied
A single VoIP user is in outage if less than 98% of its
speech frames are delivered successfully within 50 ms
air interface delay.
According to [8], the maximum acceptable mouth-to-ear
delay for voice is on the order of 250 ms. Assuming that the
delay for Core Network is approximately 100 ms, the tolerable
delay for Radio Link Control (RLC) and Medium Access
Control (MAC) buffering, scheduling and detection should be
strictly lower than 150 ms. Hence, assuming that both end
users are E-UTRAN users, tolerable delay for buffering and
scheduling is lower than 80 ms. A delay bound of 50 ms (for
delay from eNB to UE) has been chosen for the 3GPP
performance evaluations to better account for variability in
network end-to-end delays.
C. Bursty low bitrate traffic

Since this backlog can start accumulating easily, leading to

resource stalling for several users, scheduling should take care
that the buffering delay of each VoIP user is taken into
account in the scheduling decisions.
Since VoIP packets are relatively small (regardless of the
used AMR codec), there are some challenges in allocating the
resources; 2-4 symbols of each carrier in each Physical
Resource Block (PRB) are reserved for control data (reference
symbols, allocation information, HARQ ACK/NACK
channels), depending on the need for allocation signaling.
With the demand for several users to be scheduled
simultaneously, the control channel capacity might become a
limit for the VoIP capacity due to lack of signaling bits.
D. Packet scheduling and link adaptation
Since VoIP is strictly delay-restricted service, the PS needs
to take the buffering delay of UEs into account. As presented
in e.g. [9], dynamic packet scheduling provides good
frequency domain and multi-user gain for best effort type
traffic. However, because of the VoIP service characteristics
discussed before, several persistent type scheduling algorithms
(such as fully persistent, talk-spurt based persistent and semipersistent scheduling) have been proposed in 3GPP. These
scheduling mechanisms limit or even lack entirely the gain
from multi-user and frequency domain scheduling, but work
around a difficult problem of the PDCCH capacity restricting
the overall VoIP capacity. However, also dynamic PS may be
improved for improving the VoIP capacity with control
channel restrictions. With packet bundling the eNb may decide
to bundle one or more VoIP packets into one L1 PDU
improving the spectral efficiency together with LA due to
better resource utilization.
E.

Handovers and mobility

E-UTRAN utilizes a UE assisted hard handover algorithm

for mobility: UE measures downlink signal quality and sends
the measurement reports to eNB either periodically or when an
event triggers.(such as another eNB becoming stronger than
tha current eNB). The eNB then makes the final handover
decisions based on the received measurement reports.
Typically, measurement averaging, handover margins and
timers are used in order to avoid excess or ping-pong
handovers.
During a handover the old serving eNB flushes HARQ
Stop-and-Wait (SAW) buffers, which means that VoIP
packets still waiting for a retransmission will be discarded
permanently. Also, a UE cannot be scheduled while the
handover is in progress, which may lead to additional delays
for PDUs. After a connection to the new eNB is established
both HARQ and PS processed continue normally.

In the context of this article, we consider VoIP traffic as

provided by AMR codec. The AMR VoIP traffic is quite
bursty: Theres one VoIP packet at 20 ms intervals during talk
spurt and one SID packet at 160 ms intervals during silence
period. Thus, for any given TTI, only few of the active users
need to be scheduled. At the same time, each unscheduled user
contributes to a backlog of scheduling requests for later TTIs.

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C. Simulation cases

III. SIMULATION ASSUMPTIONS AND MODELING

A. System simulator description
We have used a fully dynamic system simulator for
studying the VoIP performance. Both E-UTRAN downlink
and uplink are simulated with TTI (1 ms) resolution.
Simulator contains detailed modeling of RRM, mobility and
handovers as well as traffic models. Exponential Effective
SINR Mapping (EESM) interface is used as link-to-system
interface [10].
B. Scenario setup and related modeling
The VoIP capacity evaluation is based on the UTRAN LTE
downlink parameters and assumptions described in [1]. All the
simulation cases were run in a three tier diamond-pattern
macro scenario with 19 3-sector sites, i.e. a total amount of 57
cells. Users are uniformly dropped and move within the 21
cells in the middle. The 26 cells at the edge of the scenario are
just generating interference at the same magnitude as the
average load in the center cells. The VoIP capacities are
presented in all 3GPP defined macro simulation cases shown
in TABLE I [1]. Note, that Cases 1, 2 and 3 are modified to
have only 5 MHz bandwidth.

The VoIP capacity depends on several different features,

such as:
1. Bandwidth: The used system bandwidth determines
the total amount of frequency domain resources
2. Link adaptation: With LA each TB can be optimized
in terms of spectral efficiency and BLER.
3. Delay threshold: Since VoIP is delay-critical service
determined by the delay threshold.
4. Number of control channels: The number of
maximum schedulable users in a TTI depends on the
control channel capacity.
5. Packet bundling: The amount of VoIP packets
bundled per UE L1 PDU may improve the resource
utilization, especially with control channel limitations.
6. HARQ processes: LTE requires small round trip
times, which is provided by fast L1 retransmissions by
HARQ.

TABLE I. 3GPP SIMULATION CASE DEFINITIONS.

Case
1
2
3
4

CF
(GHz)
2.0
2.0
2.0
0.9

ISD
(m)
500
500
1732
1000

BW
(MHz)
5.0
5.0
5.0
1.25

PLoss
(dB)
20
10
20
10

Speed
(kmph)
3
30
3
3

A set of common parameters for the simulations is

presented in TABLE II. We utilize a de-coupled Time Domain
(TD) and Frequency Domain (FD) packet scheduler presented
e.g. in [11]. We utilize Round Robin (RR) in the Time
Domain and Even Resources (ER) in the Frequency Domain.
RR chooses users with longest time since last scheduling time
instant for FD-PS scheduling candidates. ER first sorts the
candidate users based on the buffering delay. Then, for each
user in turn, the PRBs are sorted according to user experienced
CQI and each user is allocated enough PRBs to be able to
transmit a VoIP packet, or more if the PS decides to bundle
more than one packet. LA tries to maximize the spectral
efficiency by choosing a best Modulation and Coding Set
(MCS) for a scheduled user based on instantaneous radio
channel conditions.
VoIP AMR 12.2 traffic model is modeled with both active
and silence periods. Packets are modeled to include Real-time
Transport Protocol (RTP), Robust Header Compression
(ROHC), Packet Data Convergence Protocol (PDCP), RLC
and MAC headers in the total packet size. The VoIP traffic
model parameters have been presented in TABLE III.
TABLE IV shows the parameters varied in the simulations.

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TABLE II. COMMON PARAMETERS.

Parameter description
Scenario / network / direction

Parameter value
57 cells, Synchronous
reuse 1 network, DL

UE velocity
UE receiver type
Channel model
Simulation length

3 km/h
MRC 1x2
TU 20
1M steps = 72
seconds
14 (with 4
control
symbols)
1 ms
12
FDD
Off
Asynchronous, with
Chase combining
3
Off
5 ms
2 ms
2 PRBs
1 dB
QPSK 2/3

Symbols per subframe

Subframe length (TTI)

Carriers per PRB
Duplexing
Power control
HARQ mode
HARQ max retransmissions
ARQ
CQI measurement interval
CQI reporting delay
CQI reporting resolution
CQI error variance
Initial MCS (LA off)
Possible MCSs (LA on)

LA
TD packet scheduler
FD packet scheduler
Segmentation
Hard handover margin
Hard handover sliding window size

QPSK 1/3, , 2/3

16QAM , 2/3, 4/5
64QAM , 2/3, 4/5
Outer Loop LA
BLER target 0.2
Round Robin
Even Resources
Off
3 dB
200 ms

TABLE III. VOIP AMR 12.2 PARAMETERS.

Parameter description
VoIP packet
SID packet
Voice Activity Factor
Call length
Talk spurt

LA provides about 44% to 78% gain over static MCS of

QPSK 2/3 depending on the 3GPP simulation case. This is
because with LA a VoIP packet might fit in fewer PRBs for
UEs with good radio conditions, thus more VoIP packets can
be sent each TTI in average. On the other hand, UEs with bad
radio channel conditions can utilize more robust MCS
improving BLER of the transmissions.
When the system bandwidth is decreased by a factor of four
(i.e. case 4 with 1.25 MHz bandwidth) the VoIP capacity is
decreased by a factor of five. This is due to better packet
bundling gain with higher bandwidths. In 3GPP Case 4 LA
provides less gain, because with 1.25 MHz bandwidth the
system is not control channel limited and packet bundling does
not provide any gain.

Parameter value
38 bytes / 20 ms
14 bytes / 160 ms
50 %
Neg.exp. distr, mean 20 s
Neg.exp. distr. mean 2 s

TABLE IV. VARIED SIMULATION PARAMETERS.

Parameter description
Simulation case
Delay threshold
Number of control channels
Packet bundling
Link adaptation
HARQ processes

Parameter values
1, 2, 3, 4
40, 50, 60, 80, 100 ms
6, 8, 10
On, off
On, off
7, 8, 9

B. Effect of number of control channels and packet

bundling

IV. SIMULATION RESULTS

A. VoIP capacity in different 3GPP cases
The VoIP capacities in four different 3GPP cases are shown
in Figure 1. The capacity for Case 1 and Case 3 is about 300
UEs/cell with LA, showing that the larger ISD in Case 3 does
not lower the VoIP capacity. This indicates simply that the
capacity in Case 3 is limited by other factors than transmission
power and noise.
When the UE velocity is increased to 30 km/h in Case 2, the
VoIP capacity drops by about 42% with LA and by about 35%
without LA. There are three main factors contributing to this
capacity loss. First, the FD-PS performance is worse due to
less accurate CQI information the PRB allocation becomes
more random. Second, the bad CQI affects also LA and less
optimal MCS is selected. Third, HO performance is slightly
worse with higher speed as the UE is connected to a nonoptimal cell more often. Note that the second point is valid
only with LA, which explains why the capacity loss without
LA is less than with LA.

Figure 1. VoIP capacity in different 3GPP cases.

With 1.25 MHz bandwidth (Case 4) the PDCCH capacity is

not limiting the VoIP capacity due to low traffic channel
capacity. However, with higher system bandwidths the
PDCCH capacity might become a limiting factor, at least if
dynamic scheduling is used.
According to Figure 2, with 5 MHz bandwidth and without
LA, the number of control channels clearly limits the VoIP
capacity without packet bundling. The capacity gain over 6
control channels is about 33% and 80% with 8 and 10 control
channels, respectively. On the other hand, with LA the gains
are 20% with 8 control channels and 21% with 10 control
channels. Smaller the PDCCH capacity, the more relative gain
we get from joint packet bundling and LA. With 6 control
channels the capacity gain with LA and packet bundling is
106%, with 8 control channels 86% and 10 control channels
only about 39%. In Figure 3 the PDU size distribution with
250 UEs per cell is shown. It can be seen that with 6 control
channels much more VoIP packets are bundled (PDU size is
2* 38 bytes = 76 bytes) than with 8 or 10 control channels.

Figure 2. Effect of PDCCH capacity, packet bundling

and link adaptation on VoIP capacity.

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C. Effect of delay bound

V. CONCLUSION

VoIP service is strictly delay critical and the radio interface

delay threshold is specified to be 50 ms in 3GPP. However, as
can be seen in Figure 4, the delay threshold has only a little
impact on the VoIP capacity. Only the 40 ms delay threshold
affects the VoIP capacity and with other delay thresholds the
effect is only visible after the capacity point in the UE
satisfaction curve. This is because the 95% satisfaction rate is
reached at a point where the unsatisfied users are unsatisfied
due to packet losses, so regardless of the delay bound, the
same users are unsatisfied.

We have presented the basic VoIP downlink capacity

results in different 3GPP simulation cases. Also, we have
studied the effect of multiple features on VoIP capacity, such
as the effect of delay threshold, packet bundling, control
channel capacity and number of HARQ processes on VoIP
capacity.
The main conclusions from the results are that
VoIP downlink capacity is maximally about 60 UEs per
cell with 1.25 MHz system bandwidth and about 300
UEs per cell with 5 MHz.
Link adaptation together with packet bundling provides
about 44-78% gain over the static MCS of QPSK 2/3
depending a little from the simulated case,
In general the higher the system bandwidth the higher
the gain is from packet bundling and link adaptation,
VoIP capacity is clearly control channel limited but
packet bundling can quite effectively compensate the
limitations.
Future work includes studying the VoIP capacity with
persistent scheduling algorithms and dynamic PDCCH. Also
more realistic mixed traffic scenarios are to be studied.
REFERENCES
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Figure 3. L1 PDU size distribution with 250 UEs/cell.

Figure 4. Effect of different delay thresholds.

D. Effect of HARQ processes
The number of HARQ SAW channels for DL is being
defined at 3GPP. We simulated the VoIP capacity with 7-9
SAW channels and according to the results the number of
SAW channels does not seem to have any effect to the results.
This suggests that 7 SAW channel is already enough for VoIP,
as expected: the delay bound of 50 ms means that at most 2-3
packets should be waiting retransmission.

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