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C. Simulation cases
III. SIMULATION ASSUMPTIONS AND MODELING
The VoIP capacity depends on several different features,
such as:
A. System simulator description 1. Bandwidth: The used system bandwidth determines
the total amount of frequency domain resources
We have used a fully dynamic system simulator for 2. Link adaptation: With LA each TB can be optimized
studying the VoIP performance. Both E-UTRAN downlink in terms of spectral efficiency and BLER.
and uplink are simulated with TTI (1 ms) resolution. 3. Delay threshold: Since VoIP is delay-critical service
Simulator contains detailed modeling of RRM, mobility and determined by the delay threshold.
handovers as well as traffic models. Exponential Effective 4. Number of control channels: The number of
SINR Mapping (EESM) interface is used as link-to-system maximum schedulable users in a TTI depends on the
interface [10]. control channel capacity.
5. Packet bundling: The amount of VoIP packets
B. Scenario setup and related modeling
bundled per UE L1 PDU may improve the resource
The VoIP capacity evaluation is based on the UTRAN LTE utilization, especially with control channel limitations.
downlink parameters and assumptions described in [1]. All the 6. HARQ processes: LTE requires small round trip
simulation cases were run in a three tier diamond-pattern times, which is provided by fast L1 retransmissions by
macro scenario with 19 3-sector sites, i.e. a total amount of 57 HARQ.
cells. Users are uniformly dropped and move within the 21
cells in the middle. The 26 cells at the edge of the scenario are TABLE II. COMMON PARAMETERS.
just generating interference at the same magnitude as the
average load in the center cells. The VoIP capacities are Parameter description Parameter value
presented in all 3GPP defined macro simulation cases shown Scenario / network / direction 57 cells, Synchronous
in TABLE I [1]. Note, that Cases 1, 2 and 3 are modified to reuse 1 network, DL
have only 5 MHz bandwidth. UE velocity 3 km/h
TABLE I. 3GPP SIMULATION CASE DEFINITIONS. UE receiver type MRC 1x2
Channel model TU 20
Case CF ISD BW PLoss Speed Simulation length 1M steps = 72
(GHz) (m) (MHz) (dB) (kmph) seconds
1 2.0 500 5.0 20 3
Symbols per subframe 14 (with 4
2 2.0 500 5.0 10 30
control
3 2.0 1732 5.0 20 3 symbols)
4 0.9 1000 1.25 10 3
Subframe length (TTI) 1 ms
A set of common parameters for the simulations is Carriers per PRB 12
presented in TABLE II. We utilize a de-coupled Time Domain Duplexing FDD
(TD) and Frequency Domain (FD) packet scheduler presented Power control Off
e.g. in [11]. We utilize Round Robin (RR) in the Time HARQ mode Asynchronous, with
Domain and Even Resources (ER) in the Frequency Domain. Chase combining
RR chooses users with longest time since last scheduling time HARQ max retransmissions 3
instant for FD-PS scheduling candidates. ER first sorts the ARQ Off
candidate users based on the buffering delay. Then, for each CQI measurement interval 5 ms
user in turn, the PRBs are sorted according to user experienced CQI reporting delay 2 ms
CQI and each user is allocated enough PRBs to be able to CQI reporting resolution 2 PRBs
transmit a VoIP packet, or more if the PS decides to bundle CQI error variance 1 dB
more than one packet. LA tries to maximize the spectral Initial MCS (LA off) QPSK 2/3
efficiency by choosing a best Modulation and Coding Set
Possible MCSs (LA on) QPSK 1/3, ½, 2/3
(MCS) for a scheduled user based on instantaneous radio
channel conditions. 16QAM ½, 2/3, 4/5
VoIP AMR 12.2 traffic model is modeled with both active
and silence periods. Packets are modeled to include Real-time 64QAM ½, 2/3, 4/5
Transport Protocol (RTP), Robust Header Compression LA Outer Loop LA
(ROHC), Packet Data Convergence Protocol (PDCP), RLC BLER target 0.2
and MAC headers in the total packet size. The VoIP traffic TD packet scheduler Round Robin
model parameters have been presented in TABLE III. FD packet scheduler Even Resources
TABLE IV shows the parameters varied in the simulations. Segmentation Off
Hard handover margin 3 dB
Hard handover sliding window size 200 ms
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TABLE III. VOIP AMR 12.2 PARAMETERS.
LA provides about 44% to 78% gain over static MCS of
Parameter description Parameter value QPSK 2/3 depending on the 3GPP simulation case. This is
VoIP packet 38 bytes / 20 ms because with LA a VoIP packet might fit in fewer PRBs for
SID packet 14 bytes / 160 ms UEs with good radio conditions, thus more VoIP packets can
Voice Activity Factor 50 % be sent each TTI in average. On the other hand, UEs with bad
Call length Neg.exp. distr, mean 20 s radio channel conditions can utilize more robust MCS
Talk spurt Neg.exp. distr. mean 2 s improving BLER of the transmissions.
When the system bandwidth is decreased by a factor of four
TABLE IV. VARIED SIMULATION PARAMETERS. (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
Parameter description Parameter values bundling gain with higher bandwidths. In 3GPP Case 4 LA
Simulation case 1, 2, 3, 4 provides less gain, because with 1.25 MHz bandwidth the
Delay threshold 40, 50, 60, 80, 100 ms system is not control channel limited and packet bundling does
Number of control channels 6, 8, 10 not provide any gain.
Packet bundling On, off
Link adaptation On, off B. Effect of number of control channels and packet
HARQ processes 7, 8, 9 bundling
IV. SIMULATION RESULTS 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
A. VoIP capacity in different 3GPP cases PDCCH capacity might become a limiting factor, at least if
dynamic scheduling is used.
The VoIP capacities in four different 3GPP cases are shown According to Figure 2, with 5 MHz bandwidth and without
in Figure 1. The capacity for Case 1 and Case 3 is about 300 LA, the number of control channels clearly limits the VoIP
UEs/cell with LA, showing that the larger ISD in Case 3 does capacity without packet bundling. The capacity gain over 6
not lower the VoIP capacity. This indicates simply that the control channels is about 33% and 80% with 8 and 10 control
capacity in Case 3 is limited by other factors than transmission channels, respectively. On the other hand, with LA the gains
power and noise. are 20% with 8 control channels and 21% with 10 control
When the UE velocity is increased to 30 km/h in Case 2, the channels. Smaller the PDCCH capacity, the more relative gain
VoIP capacity drops by about 42% with LA and by about 35% we get from joint packet bundling and LA. With 6 control
without LA. There are three main factors contributing to this channels the capacity gain with LA and packet bundling is
capacity loss. First, the FD-PS performance is worse due to 106%, with 8 control channels 86% and 10 control channels
less accurate CQI information – the PRB allocation becomes only about 39%. In Figure 3 the PDU size distribution with
more random. Second, the bad CQI affects also LA and less 250 UEs per cell is shown. It can be seen that with 6 control
optimal MCS is selected. Third, HO performance is slightly channels much more VoIP packets are bundled (PDU size is
worse with higher speed as the UE is connected to a non- 2* 38 bytes = 76 bytes) than with 8 or 10 control channels.
optimal 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.
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C. Effect of delay bound V. CONCLUSION
VoIP service is strictly delay critical and the radio interface We have presented the basic VoIP downlink capacity
delay threshold is specified to be 50 ms in 3GPP. However, as results in different 3GPP simulation cases. Also, we have
can be seen in Figure 4, the delay threshold has only a little studied the effect of multiple features on VoIP capacity, such
impact on the VoIP capacity. Only the 40 ms delay threshold as the effect of delay threshold, packet bundling, control
affects the VoIP capacity and with other delay thresholds the channel capacity and number of HARQ processes on VoIP
effect is only visible after the capacity point in the UE capacity.
satisfaction curve. This is because the 95% satisfaction rate is The main conclusions from the results are that
reached at a point where the unsatisfied users are unsatisfied • VoIP downlink capacity is maximally about 60 UEs per
due to packet losses, so regardless of the delay bound, the cell with 1.25 MHz system bandwidth and about 300
same users are unsatisfied. 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
[1] “Physical Layer Aspects for Evolved UTRA”, 3GPP Technical Report
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and M. Wahlqvist, “Technical Solutions for the 3G Long-Term
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[3] D. Jiang, H. Wang, E. Malkamäki and E. Tuomaala, ”Principle and
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[6] "Requirements for Evolved UTRA (E-UTRA) and Evolved UTRAN (E-
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The number of HARQ SAW channels for DL is being 4, September 2005, pp. 2306–2311.
[11] P. Kela, J. Puttonen, N. Kolehmainen, T. Ristaniemi, T. Henttonen, and
defined at 3GPP. We simulated the VoIP capacity with 7-9 M. Moisio, “Dynamic Packet Scheduling Performance in UTRA Long
SAW channels and according to the results the number of Term Evolution Downlink,” in Proceedings of the International
SAW channels does not seem to have any effect to the results. Symposium on Wireless Pervasive Computing (ISWPC’08), May 2008,
This suggests that 7 SAW channel is already enough for VoIP, to be published.
as expected: the delay bound of 50 ms means that at most 2-3
packets should be waiting retransmission.
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