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Abstract —In 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 studythe VoIP Adaptive Multi-Rate (AMR) 12.2 codec capacity in fourdifferent 3GPP simulation cases. The effects of Link Adaptation(LA), packet bundling, control channel capacity and number ofHARQ processes on VoIP capacity have also been considered.The results present the absolute VoIP capacity numbers of LTEDL. We also show that LA together with packet bundlingprovides clear gain on the VoIP capacity, because more VoIPpackets can be scheduled in each TTI. Also, the control channellimitations can be effectively compensated by packet bundling.

Index Terms —VoIP, LTE, system simulations

I. INTRODUCTION

The Evolved UTRAN (E-UTRAN) or the UTRAN LongTerm Evolution (LTE) specifications are being finalized in3GPP. LTE aims at ambitious goals of e.g. peak data rate of100 Mbps in downlink and 50 Mbps in uplink, increased celledge 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 bothtime (TDD) and frequency division duplex (FDD) modes, butin this article we concentrate on FDD. Orthogonal FrequencyDivision Multiple Access (OFDMA) has been selected for thedownlink multiple access technology and Single CarrierFrequency Division Multiple Access (SC-FDMA) for uplink[1]. To achieve the objectives set for LTE, advanced RadioResource Management (RRM) functions have been defined.The algorithms include e.g. Hybrid ARQ (HARQ), LinkAdaptation (LA), Channel Quality Indication (CQI) andPacket Scheduling (PS). More on these can be found e.g. from[2].

E-UTRAN is optimized for packet data transfer and thecore network is purely packet switched, so speech istransmitted purely with Voice-over-IP (VoIP). VoIP trafficconsists of talk-spurts and silent periods, with relatively small

packets transmitted quite rarely. The Adaptive Multi-Rate(AMR) codec provides quite bursty traffic; one VoIP packet at20 ms intervals during talk spurt and one Silence Indicator(SID) packet at 160 ms intervals during silence period. E-UTRAN is expected to support a very high number of VoIPusers 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 VoIPusers, for both PS and LA algorithms. Also, the capacity ofPhysical Downlink Control Channel (PDCCH) induces somerestrictions, at least with higher system bandwidths. Theserestrictions become most relevant with dynamic packetscheduling, since each allocation consumes signalingresources from PDCCH. Thus, several persistent resourceallocation schemes, such as fully persistent scheduling, talk-

spurt based persistent scheduling and semi-persistentscheduling have also been proposed in 3GPP [3]. However,these scheduling types limit the gain from multi-user andfrequency 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 withdynamic packet scheduling. The effect of different features,such as system bandwidth, LA, Control Channel (CC)capacity, packet bundling and HARQ processes, on VoIPcapacity are studied using simulations. The simulation resultsare gathered from fully dynamic system simulator, whichmodels the UE mobility, RRM functionalities and their

interactions with the system.The paper is organised as follows: Chapter II discusses thegeneral aspects of VoIP in LTE and related modeling. ChapterIII lists the simulation assumptions including a shortdescription of the simulator. Chapter IV presents thesimulation results and analysis. Finally, Chapter V reviews themain conclusions.

Voice-over-IP Performance in UTRA LongTerm Evolution Downlink

Jani Puttonen , Tero Henttonen , Niko Kolehmainen , Kennett Aschan ,Martti Moisio 2 and Petteri Kela 3

1Magister Solutions Ltd, c/o Mattilanniemi 6-8,40101 Jyväskylä, Finland. Email: firstname.lastname@magister.fi

2 Nokia, P.O.BOX 45, FIN-00045 Nokia Group, FinlandEmail: firstname.lastname@nokia.com

3University of Jyväskylä, Dept. of Mathematical Information TechnologyP.O. Box 35, 40014 University of Jyväskylä, Finland. Email: firstname.lastname@jyu.fi

978-1-4244-1645-5/08/$25.00 ©2008 IEEE 2502

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II. VOICE -OVER -IP IN LTE

VoIP has at least three characteristics that needconsideration in LTE (as well as in any wireless system):Bursty low bitrate traffic, strict packet delay-based QoS andhigh number of simultaneous users). These issues setchallenges to the RRM functions. Next, we discuss thesecharacteristics as well as the required RRM functions in moredetail.

A. High Capacity Demand

The requirements of E-UTRA and E-UTRAN are describedin TR.25.813 [6]. The service related requirements for VoIPare:

• The E-UTRA should efficiently support various typesof service. These must include currently availableservices like web-browsing, FTP, video-streaming orVoIP, and more advanced services (e.g. real-time videoor 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-UTRANvia the packet switched domain with at least equalquality as supported by UTRAN (e.g. in terms ofguaranteed bit rate) - over the whole speed range.

B. Strict packet delay-based QoS

The system capacity for VoIP service is limited by theoutage limits defined in TR 25.814 [1] and updated in 3GPPcontribution 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 itsspeech frames are delivered successfully within 50 msair interface delay.

According to [8], the maximum acceptable mouth-to-eardelay for voice is on the order of 250 ms. Assuming that thedelay for Core Network is approximately 100 ms, the tolerabledelay for Radio Link Control (RLC) and Medium AccessControl (MAC) buffering, scheduling and detection should bestrictly lower than 150 ms. Hence, assuming that both endusers are E-UTRAN users, tolerable delay for buffering andscheduling is lower than 80 ms. A delay bound of 50 ms (fordelay from eNB to UE) has been chosen for the 3GPP

performance evaluations to better account for variability innetwork end-to-end delays.

C. Bursty low bitrate traffic

In the context of this article, we consider VoIP traffic as provided by AMR codec. The AMR VoIP traffic is quite bursty: There’s one VoIP packet at 20 ms intervals during talkspurt and one SID packet at 160 ms intervals during silence

period. Thus, for any given TTI, only few of the active usersneed to be scheduled. At the same time, each unscheduled usercontributes to a backlog of scheduling requests for later TTIs.

Since this backlog can start accumulating easily, leading toresource stalling for several users, scheduling should take carethat the buffering delay of each VoIP user is taken intoaccount in the scheduling decisions.

Since VoIP packets are relatively small (regardless of theused AMR codec), there are some challenges in allocating theresources; 2-4 symbols of each carrier in each PhysicalResource Block (PRB) are reserved for control data (reference

symbols, allocation information, HARQ ACK/NACKchannels), depending on the need for allocation signaling.With the demand for several users to be scheduledsimultaneously, the control channel capacity might become alimit 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 needsto take the buffering delay of UEs into account. As presentedin e.g. [9], dynamic packet scheduling provides goodfrequency domain and multi-user gain for best effort typetraffic. However, because of the VoIP service characteristicsdiscussed before, several persistent type scheduling algorithms(such as fully persistent, talk-spurt based persistent and semi-

persistent scheduling) have been proposed in 3GPP. Thesescheduling mechanisms limit or even lack entirely the gainfrom multi-user and frequency domain scheduling, but workaround a difficult problem of the PDCCH capacity restrictingthe overall VoIP capacity. However, also dynamic PS may beimproved for improving the VoIP capacity with controlchannel restrictions. With packet bundling the eNb may decideto bundle one or more VoIP packets into one L1 PDUimproving the spectral efficiency together with LA due to

better resource utilization.

E. Handovers and mobility

E-UTRAN utilizes a UE assisted hard handover algorithmfor mobility: UE measures downlink signal quality and sendsthe measurement reports to eNB either periodically or when anevent triggers.(such as another eNB becoming stronger thantha current eNB). The eNB then makes the final handoverdecisions based on the received measurement reports.Typically, measurement averaging, handover margins andtimers are used in order to avoid excess or ping-ponghandovers.

During a handover the old serving eNB flushes HARQStop-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 thehandover is in progress, which may lead to additional delaysfor PDUs. After a connection to the new eNB is established

both HARQ and PS processed continue normally.

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III. SIMULATION ASSUMPTIONS AND MODELING

A. System simulator description

We have used a fully dynamic system simulator forstudying the VoIP performance. Both E-UTRAN downlinkand uplink are simulated with TTI (1 ms) resolution.Simulator contains detailed modeling of RRM, mobility andhandovers as well as traffic models. Exponential EffectiveSINR Mapping (EESM) interface is used as link-to-systeminterface [10].

B. Scenario setup and related modeling

The VoIP capacity evaluation is based on the UTRAN LTEdownlink parameters and assumptions described in [1]. All thesimulation cases were run in a three tier diamond-patternmacro scenario with 19 3-sector sites, i.e. a total amount of 57cells. Users are uniformly dropped and move within the 21cells in the middle. The 26 cells at the edge of the scenario are

just generating interference at the same magnitude as theaverage load in the center cells. The VoIP capacities are

presented in all 3GPP defined macro simulation cases shownin TABLE I [1]. Note, that Cases 1, 2 and 3 are modified tohave only 5 MHz bandwidth.

TABLE I. 3GPP SIMULATION CASE DEFINITIONS.

Case CF(GHz)

ISD(m)

BW(MHz)

PLoss(dB)

Speed(kmph)

1 2.0 500 5.0 20 32 2.0 500 5.0 10 303 2.0 1732 5.0 20 34 0.9 1000 1.25 10 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 presentede.g. in [11]. We utilize Round Robin (RR) in the TimeDomain and Even Resources (ER) in the Frequency Domain.RR chooses users with longest time since last scheduling timeinstant for FD-PS scheduling candidates. ER first sorts thecandidate users based on the buffering delay. Then, for eachuser in turn, the PRBs are sorted according to user experiencedCQI and each user is allocated enough PRBs to be able totransmit a VoIP packet, or more if the PS decides to bundlemore than one packet. LA tries to maximize the spectralefficiency by choosing a best Modulation and Coding Set(MCS) for a scheduled user based on instantaneous radiochannel conditions.

VoIP AMR 12.2 traffic model is modeled with both activeand silence periods. Packets are modeled to include Real-timeTransport Protocol (RTP), Robust Header Compression(ROHC), Packet Data Convergence Protocol (PDCP), RLCand MAC headers in the total packet size. The VoIP trafficmodel parameters have been presented in TABLE III.

TABLE IV shows the parameters varied in the simulations.

C. Simulation cases

The VoIP capacity depends on several different features,such as:

1. Bandwidth : The used system bandwidth determinesthe total amount of frequency domain resources

2. Link adaptation: With LA each TB can be optimizedin terms of spectral efficiency and BLER.

3. Delay threshold : Since VoIP is delay-critical servicedetermined by the delay threshold.

4. Number of control channels: The number ofmaximum schedulable users in a TTI depends on thecontrol channel capacity.

5. Packet bundling: The amount of VoIP packets bundled per UE L1 PDU may improve the resourceutilization, especially with control channel limitations.

6. HARQ processes: LTE requires small round triptimes, which is provided by fast L1 retransmissions byHARQ.

TABLE II. COMMON PARAMETERS.

Parameter description Parameter valueScenario / network / direction 57 cells, Synchronous

reuse 1 network, DL

UE velocity 3 km/hUE receiver type MRC 1x2Channel model TU 20Simulation length 1M steps = 72

secondsSymbols per subframe 14 (with 4

controlsymbols)

Subframe length (TTI) 1 msCarriers per PRB 12Duplexing FDDPower control OffHARQ mode Asynchronous, with

Chase combiningHARQ max retransmissions 3ARQ OffCQI measurement interval 5 msCQI reporting delay 2 msCQI reporting resolution 2 PRBsCQI error variance 1 dBInitial MCS (LA off) QPSK 2/3

Possible MCSs (LA on) QPSK 1/3, ½, 2/316QAM ½, 2/3, 4/564QAM ½, 2/3, 4/5

LA Outer Loop LABLER target 0.2

TD packet scheduler Round RobinFD packet scheduler Even ResourcesSegmentation OffHard handover margin 3 dBHard handover sliding window size 200 ms

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TABLE III. VOIP AMR 12.2 PARAMETERS.

Parameter description Parameter valueVoIP packet 38 bytes / 20 msSID packet 14 bytes / 160 msVoice Activity Factor 50 %Call length Neg.exp. distr, mean 20 sTalk spurt Neg.exp. distr. mean 2 s

TABLE IV. VARIED SIMULATION PARAMETERS.

Parameter description Parameter valuesSimulation case 1, 2, 3, 4Delay threshold 40, 50, 60, 80, 100 ms

Number of control channels 6, 8, 10Packet bundling On, offLink adaptation On, offHARQ processes 7, 8, 9

IV. SIMULATION R ESULTS

A. VoIP capacity in different 3GPP cases

The VoIP capacities in four different 3GPP cases are shownin Figure 1. The capacity for Case 1 and Case 3 is about 300UEs/cell with LA, showing that the larger ISD in Case 3 doesnot lower the VoIP capacity. This indicates simply that thecapacity 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 thiscapacity loss. First, the FD-PS performance is worse due toless accurate CQI information – the PRB allocation becomes

more random. Second, the bad CQI affects also LA and lessoptimal MCS is selected. Third, HO performance is slightlyworse with higher speed as the UE is connected to a non-optimal cell more often. Note that the second point is validonly with LA, which explains why the capacity loss withoutLA is less than with LA.

Figure 1. VoIP capacity in different 3GPP cases.

LA provides about 44% to 78% gain over static MCS ofQPSK 2/3 depending on the 3GPP simulation case. This is

because with LA a VoIP packet might fit in fewer PRBs forUEs with good radio conditions, thus more VoIP packets can

be sent each TTI in average. On the other hand, UEs with badradio channel conditions can utilize more robust MCSimproving 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 isdecreased 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 thesystem is not control channel limited and packet bundling doesnot provide any gain.

B. Effect of number of control channels and packetbundling

With 1.25 MHz bandwidth (Case 4) the PDCCH capacity isnot limiting the VoIP capacity due to low traffic channelcapacity. However, with higher system bandwidths thePDCCH capacity might become a limiting factor, at least ifdynamic scheduling is used.

According to Figure 2, with 5 MHz bandwidth and withoutLA, the number of control channels clearly limits the VoIPcapacity without packet bundling. The capacity gain over 6control channels is about 33% and 80% with 8 and 10 controlchannels, respectively. On the other hand, with LA the gainsare 20% with 8 control channels and 21% with 10 controlchannels. Smaller the PDCCH capacity, the more relative gainwe get from joint packet bundling and LA. With 6 controlchannels the capacity gain with LA and packet bundling is106%, with 8 control channels 86% and 10 control channelsonly about 39%. In Figure 3 the PDU size distribution with

250 UEs per cell is shown. It can be seen that with 6 controlchannels much more VoIP packets are bundled (PDU size is2* 38 bytes = 76 bytes) than with 8 or 10 control channels.

Figure 2. Effect of PDCCH capacity, packet bundlingand link adaptation on VoIP capacity.

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C. Effect of delay bound

VoIP service is strictly delay critical and the radio interfacedelay threshold is specified to be 50 ms in 3GPP. However, ascan be seen in Figure 4, the delay threshold has only a littleimpact on the VoIP capacity. Only the 40 ms delay thresholdaffects the VoIP capacity and with other delay thresholds theeffect is only visible after the capacity point in the UEsatisfaction curve. This is because the 95% satisfaction rate isreached at a point where the unsatisfied users are unsatisfieddue to packet losses, so regardless of the delay bound, thesame users are unsatisfied.

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 beingdefined at 3GPP. We simulated the VoIP capacity with 7-9SAW channels and according to the results the number ofSAW 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.

V. CONCLUSION

We have presented the basic VoIP downlink capacityresults in different 3GPP simulation cases. Also, we havestudied the effect of multiple features on VoIP capacity, suchas the effect of delay threshold, packet bundling, controlchannel capacity and number of HARQ processes on VoIPcapacity.

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 300UEs per cell with 5 MHz.

• Link adaptation together with packet bundling providesabout 44-78% gain over the static MCS of QPSK 2/3depending a little from the simulated case,

• In general the higher the system bandwidth the higherthe gain is from packet bundling and link adaptation,

• VoIP capacity is clearly control channel limited but packet bundling can quite effectively compensate thelimitations.

Future work includes studying the VoIP capacity with persistent scheduling algorithms and dynamic PDCCH. Alsomore realistic mixed traffic scenarios are to be studied.

R EFERENCES

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