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Multi-layer Realistic Voice Capacity Evaluation in LTE Rel. 9 and Performance Comparison with PMR and GSM Alina Alexandra Florea, Laurent Martinod, Philippe Mege Security & Communication Solutions Cassidian (an EADS Company) Elancourt, France [email protected] Hang Nguyen Wireless Network & Multimedia Services Telecom SudParis Evry, France [email protected]  Abstract— This paper proposes a multi-layer realistic voice capacity evaluation method for long term evolution (LTE) Release 9 downlink transmission. The voice spectral efficiency is also compared with existent technologies, private mobile radio (PMR) and global system for mobile communications (GSM). The realistic system model considers both transmission protocols overhead, including internet protocol (IP) and robust header compression (ROHC), and air frame overhead. The LTE standard compliant layer one parameters are not appropriate for low bit-rate voice transmission. We propose to optimize this algorithm by considering the minima normalized values for both coding rate and transport block size. The newly computed values are optima for PMR voice transmission. The LTE voice communications capacity is given for the realistic and ideal values of the protocol stack overhead. The realistic results are compared to GSM and PMR voice spectral efficiencies, with the appropriate voice coders. LTE Release 9 seems promising for future data transfers, but is not yet favourable to voice services. Even though the air frame is highly configurable, the protocol stack overhead and the standard resource allocation constraints are not designed for real-time voice packets transmission. The ROHC compressed protocol overhead remains large for voice frames and demands important air resources. On the other h and, resource allocation is not foreseen for small packets transmission. Considering the proposed allocation algorithm, results show that the voice capacity may double if the protocol overhead only were reduced to ideal values.  Keywords: GSM; LTE; PMR, voice over IP. I. INTRODUCTION The long term evolution (LTE) standard has been proposed by the T hird Generation Partnershi p Project (3GPP) as the next step towards the fourth generation (4G) technologies. It represents the radio network evolution, part of the evolved packet system (EPS) [1], which also includes the evolved packet core (EPC) network standardization. LTE is expected to provide smaller transmission and connection delays, increased bit rate even at cell edge, better spectral efficiency, transparent mobility towards any network and reasonable mobile power consumption. Optimized for high data rates, the technology does not provide any voice specific mechanism. Since one of its main characteristics is the end-to-end internet protocol (IP) transmission, the envisaged solution for voice communications seems to be voice over internet protocol (VoIP). But up to now, radio mobile networks used circuit switched mechanisms, which were end-to-end controlled, using a low bandwidth and ensuring small delays. Voice transmission over LTE is one hot topic in today's research. Consistent effort has been put in finding different results and proposals to answer two main questions:  what would the real LTE efficiency be for conversational voice transmission using the architecture defined by today's standard, and  what would the best LTE architecture be for conversational VoIP transmission? In [2], the authors provide a thorough overview of the LTE protocol stack and their use with voice transmission. The radio link control (RLC) protocol must be used in unacknowledged mode (UM). Because some voice coders may tolerate high packet error rates, the medium access layer (MAC) must allow the transmission of up to 10 -2  packet error rates to higher layers. The LTE overheads for transport control protocol (TCP) used for data transmissions and VoIP services are estimated and compared. The robust header compression (ROHC) protocol can compress up to 42 % of the overhead, as low as 3 bytes, for the adaptive multi-rate (AMR) voice coder with TCP acknowledgem ent. The impact of the IP/TCP overhead and the LTE protocol stack configuration are considered. However, their impact on the overall voice capacity is not investigated. In [3], the LTE radio interface and its options are discussed. The authors highlight the importance of the scheduler on the overall downlink (DL) system performance. The paper further analyses the LTE spectral efficiency and its improvement when key radio LTE features are introduced. In a frequency division duplex (FDD) configuration and a 10 MHz deployment bandwidth (BW), the target throughput numbers evolve from 1.07 bps/Hz/sector up to 1.73 bps/Hz/sector in DL and from 0.46 to 1.05 in uplink (UL). Nevertheless, these estimations are given considering only radio layer configuration and deployment parameters. Upper layers control overhead and voice coders configurations are not taken into account. This work has been supported in part by the National Agency for Research & Technology (ANRT) in the context of the PhD of A.A. Florea, supervised by SAMOVAR Laboratory, Telecom SudParis, and Doctoral School of Informatics, Telecommunications & Electronics (EDITE) of Pierre & Marie Curie University (UPMC), Paris, France.

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Multi-layer Realistic Voice Capacity Evaluation inLTE Rel. 9 and Performance Comparison with PMR

and GSM

Alina Alexandra Florea, Laurent Martinod, Philippe MegeSecurity & Communication Solutions

Cassidian (an EADS Company)Elancourt, France

[email protected]

Hang NguyenWireless Network & Multimedia Services

Telecom SudParisEvry, France

[email protected]

Abstract — This paper proposes a multi-layer realistic voicecapacity evaluation method for long term evolution (LTE)Release 9 downlink transmission. The voice spectral efficiency is

also compared with existent technologies, private mobile radio(PMR) and global system for mobile communications (GSM).The realistic system model considers both transmission protocolsoverhead, including internet protocol (IP) and robust headercompression (ROHC), and air frame overhead. The LTEstandard compliant layer one parameters are not appropriate forlow bit-rate voice transmission. We propose to optimize thisalgorithm by considering the minima normalized values for bothcoding rate and transport block size. The newly computed valuesare optima for PMR voice transmission. The LTE voicecommunications capacity is given for the realistic and idealvalues of the protocol stack overhead. The realistic results arecompared to GSM and PMR voice spectral efficiencies, with theappropriate voice coders. LTE Release 9 seems promising forfuture data transfers, but is not yet favourable to voice services.Even though the air frame is highly configurable, the protocolstack overhead and the standard resource allocation constraintsare not designed for real-time voice packets transmission. TheROHC compressed protocol overhead remains large for voiceframes and demands important air resources. On the other hand,resource allocation is not foreseen for small packets transmission.Considering the proposed allocation algorithm, results show thatthe voice capacity may double if the protocol overhead only werereduced to ideal values.

Keywords: GSM; LTE; PMR, voice over IP.

I. INTRODUCTION

The long term evolution (LTE) standard has been proposedby the Third Generation Partnership Project (3GPP) as the nextstep towards the fourth generation (4G) technologies. Itrepresents the radio network evolution, part of the evolvedpacket system (EPS) [1], which also includes the evolvedpacket core (EPC) network standardization. LTE is expected toprovide smaller transmission and connection delays, increasedbit rate even at cell edge, better spectral efficiency, transparentmobility towards any network and reasonable mobile powerconsumption. Optimized for high data rates, the technologydoes not provide any voice specific mechanism. Since one of

its main characteristics is the end-to-end internet protocol (IP)transmission, the envisaged solution for voice communicationsseems to be voice over internet protocol (VoIP). But up to now,

radio mobile networks used circuit switched mechanisms,which were end-to-end controlled, using a low bandwidth andensuring small delays.

Voice transmission over LTE is one hot topic in today'sresearch. Consistent effort has been put in finding differentresults and proposals to answer two main questions:

• what would the real LTE efficiency be forconversational voice transmission using thearchitecture defined by today's standard, and

• what would the best LTE architecture be forconversational VoIP transmission?

In [2], the authors provide a thorough overview of the LTE

protocol stack and their use with voice transmission. The radiolink control (RLC) protocol must be used in unacknowledgedmode (UM). Because some voice coders may tolerate highpacket error rates, the medium access layer (MAC) must allowthe transmission of up to 10 -2 packet error rates to higherlayers. The LTE overheads for transport control protocol (TCP)used for data transmissions and VoIP services are estimatedand compared. The robust header compression (ROHC)protocol can compress up to 42 % of the overhead, as low as 3bytes, for the adaptive multi-rate (AMR) voice coder with TCPacknowledgement. The impact of the IP/TCP overhead and theLTE protocol stack configuration are considered. However,their impact on the overall voice capacity is not investigated. In[3], the LTE radio interface and its options are discussed. Theauthors highlight the importance of the scheduler on the overalldownlink (DL) system performance. The paper further analysesthe LTE spectral efficiency and its improvement when keyradio LTE features are introduced. In a frequency divisionduplex (FDD) configuration and a 10 MHz deploymentbandwidth (BW), the target throughput numbers evolve from1.07 bps/Hz/sector up to 1.73 bps/Hz/sector in DL and from0.46 to 1.05 in uplink (UL). Nevertheless, these estimations aregiven considering only radio layer configuration anddeployment parameters. Upper layers control overhead andvoice coders configurations are not taken into account.

This work has been supported in part by the National Agency forResearch & Technology (ANRT) in the context of the PhD of A.A. Florea,supervised by SAMOVAR Laboratory, Telecom SudParis, and DoctoralSchool of Informatics, Telecommunications & Electronics (EDITE) of Pierre& Marie Curie University (UPMC), Paris, France.

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Voice over IP is highly encouraged and operators mustpropose new transmission architectures for VoIP within LTEbefore migrating from existent radio networks to all-IPnetworks. A series of proposals have emerged, a few of whichhave been considered by the 3GPP: circuit switched (CS)fallback [4], IP multimedia subsystem (IMS) telephony(MMTel) [5], and IMS telephony with handover to CS domain(SRVCC) [6]. The not 3GPP supported are the internet-based

voice services and the voice over LTE generic access(VoLGA). In [7], these technologies are discussed, comparingthe benefits for the operator and the end-user. The advantage ofMMTel is its transparency for any kind of services whether thedevice is mobile or fixed and would represent a true IP solutionfor LTE. SRVCC and CS are based on the usage of existentcircuit-switched radio networks. The internet-based services donot benefit from a standardized inter-operator interface and donot support CS handover and emergency numbers if needed.VoLGA proposes to connect existent technologies to LTE viaone gateway, but the end-to-end call would not benefit from anall-IP architecture. The authors concentrate on the differentsolutions for voice transmission, but do not mention voicecapacity. To the best of our knowledge, no objective andrealistic study has been performed on VoIP transmission whichconsider both protocol stack and air frame configurations.

As the public networks evolve towards the 4Garchitectures, other types of networks, as the private mobileradio (PMR), are also growing towards high data ratesapplications. The PMR have been used by different privatesectors since the 1960's. Well-known standards are terrestrialtrunked radio (TETRA) [8] and TETRAPOL [9],[10]. TETRAhas evolved to a third generation (3G) architecture, TETRAenhanced data system (TEDS) [11], introducing IP for higherdata rates services.

LTE represents an interesting broadband perspective for the

PMR world. However, PMR standards must comply withspecific requirements for public safety, emergency cases andindustrial use. A key element in PMR, voice communicationsare characterized by short, very concise messages that must beclear and intelligible. Network accessibility and the number ofcommunications are prioritized, therefore low bit rate voicecoders with a very robust channel protection are used. CurrentPMR voice coders bit rates are of 5 kbps approximately, butfuture target bit rates may go as low as 2.45 kbps, e.g.Advanced Multi-Band Excitation (AMBE).

This paper evaluates the Rel. 9 LTE voice capacity andcompares its spectral efficiency with circuit-switched radiotechnologies, PMR and global system for mobile

communications (GSM). We model a multi-layer realisticvoice transmission system, considering both the protocol stackand the air frame overheads. We propose several adjustmentsto the 3GPP proposed algorithm for finding the layer oneallocation parameters: the transport block size, the coding rateand the radio resources. We further compute the realistic LTEvoice capacity and evaluate the protocol overhead impact onthe overall capacity. For small voice packets transmission, theprotocol overhead remains consistent as it demands up to 50%of the radio allocated resources even using ROHC. Voicecapacity is further restricted by the air resources allocationstrategy. The standard does not even envisage very small

frames transmission. Finally, we evaluate voice spectralefficiency for LTE and compare these results with deployedGSM and PMR networks. LTE is not optimized for real-timePMR voice communications, the 3GPP proposed protocols andallocation options are not suitable to small packetstransmission.

This paper is organized as follows. Section II describes theproposed multi-layer realistic LTE capacity evaluationalgorithm. Section III gives the real and ideal results for thePMR use case. The spectral efficiency comparison is furtherintroduced with the corresponding hypotheses and parameters.We conclude in section IV.

II. PROPOSED MULTI -LAYER REALISTIC LTE CAPACITYEVALUATION

In this section, we will present the proposed multi-layeralgorithm for the realistic evaluation of downlink (DL) LTEvoice capacity. Subsection A describes the realistictransmission system model and explains 3GPP LTE standardsparameters. In subsection B, we point out standard weakness

regarding small frame allocation: the resource allocationalgorithm and its choice of the transport blocks (TB), and wepropose its optimization.

A. Transmission System DescriptionThe system model is represented in Fig. 1, as defined by the

3GPP standards.

The application layer voice coder emits compressed voiceframes at every period of time, T voice . These packets areconsecutively prepared for transmission using: real-timetransport protocol (RTP), user datagram protocol (UDP) andIP. At layer three, the packet data convergence protocol(PDCP) context is negotiated and the ROHC compression is

performed. The PDCP header and the radio link control (RLC)header are added. At layer two, the medium access control(MAC) header is included. The newly obtained packetrepresents the necessary data packet size for layer one, calledtransport block size (TBS), here denoted by TBS necessary .

For all accepted modulation orders, M , and for all possibleradio allocations, N PRB , the 3GPP standard defines all possibleTBSs for which layer one mechanisms are configured [15].These values must be determined using system definedparameters: the number of channel bits that can be used for thefirst transmission, N ch, and the scheduled coding rate, Rscheduled .According to the 3GPP algorithm [12], the TBS must verify

ch

CRC dardized sscheduled N

N TBS R +− tanmin (1)

where TBS standardized denotes a valid TB size as standardized in[15] and N CRC the size of the layer one cyclic redundancy check(CRC) in number of bits.

The layer one receives a valid TBS and, after CRCattachment, performs the specific mechanisms for wirelessinterface transmission: turbo encoding, rate matchingmodulation and the corresponding symbol mapping to the air

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frame resources, before orthogonal frequency divisionmultiplexing (OFDM) signal generation.

The air frame resource allocation for each user has beendefined by higher layers. N PRB physical resource blocks (PRBs)are allocated in the frequency bandwidth and one sub-frame of1 ms is used for time emission, i.e. 2 sub-slots. One PRBrepresents 12 sub-carriers (SCs) in the frequency domain and 1sub-slot in the time domain, i.e. 6 OFDM symbols if anextended cyclic prefix (CP) is used. Using one layer mapping,one quadrature phase shift keying (QPSK) symbol is mappedon one resource element (RE). One RE is defined as oneOFDM symbol on one sub-carrier frequency. Fig. 2 gives thegraphical representation of one time sub-frame spanning onefrequency PRB. Each sub-frame contains pilot REs, physicaldownlink control channel (PDCCH) REs and the physicaldownlink shared channel (PDSCH) REs. The PDSCH REsonly are used for user data transmission.

The REs dedicated to PDCCH, pilot and synchronisationsymbols are considered air frame overhead and represent animportant percentage of the frame capacity, especially in thelower bandwidths. We have considered the air frame overheadas fixed and not susceptible of modifications. We have notcounted the synchronisation channels, occupying the centre 6PRBs of the used bandwidth in the first and sixth sub-frames.

The number of channel bits necessary for each user datatransmission, N ch, is strictly derived from the modulation order,

M , the scheduled radio bandwidth allocation, N PRB , and thenumber of REs per PRB available in the PDSCH, N REs/PRB , as

PRBPRB / RE ch N N N ×= . (2)

These values are defined for the deployed LTE bandwidth, N BW PRBs.

B. Proposed Multi-layer Realistic Algorithm Description

In this subsection, we propose several modifications to the3GPP standard allocation algorithm. This 3GPP algorithmcomputes the maximum TBS value that minimizes (1) for onlyone given radio block allocation, N PRB , and for the number ofchannel bits, N ch, that can be transmitted using this radio block.

But, for a realistic evaluation of LTE voice capacity used asPMR network, one element is essential: the use of a very lowbit rate voice coder. Using the standard given algorithm, wehave concluded that the obtained TBS value, which wouldperfectly respect (1) and the specified conditions, would beconsiderably larger than the voice packet. The problem withthis algorithm and given TBSs is that they are not optimized forsmall voice frames transmission, when considering robust

coding rates.We describe hereafter the proposed algorithm, step by step,

considering all layers parameters and their computation.

Step 1) Compute the voice packet bits length, P voice , considering thevoice coder bit rate, R vocoder , and the voice frame time length, T voice ,

P voice = R vocoder * T voice .

Step 2) Calculate the TBS necessary by adding the higher layer overhead,H protocols ,

TBS necessary .= P voice + H protocols .

Step 3) Set modulation order, M , and system required code rate,R scheduled .

Step 4) Set air frame parameters:• C OFDM , number of OFDM symbols dedicated to control fields,

PDCCH,

• N pilots , number of REs dedicated to pilots,

• S OFDM , number of OFDM symbols per sub-frame

Step 5) Compute the number of REs per PRB pair per sub-frame

N REs/PRB = 12* ( S OFDM - C OFDM )- N pilots .

Step 6) Load TBS_table valid values as given in TBS table 7.1.7.1-1from 3GPP TS 36.213 V9.0.

1 subframe = 1 ms

1 PRB(12 SCs)

C OFDM DOFDM

P D C C H

1 RE =1 SC x

1 symbol

PilotRE

PDSCH

(Voice Packet)

Figure 2. Resource grid possible allocation.

Voice bit frame

RTP header

UDP header

IP header

ROHC compression

PDCP header

RLC header

MAC header

Transmission Protocols

Layer 3

Layer 2

Application Layer

TBS necessary

CalculateTBS standardized

TBS standardized

CRC

Turbo coding 1/3

Rate matching

Modulation (QPSK)

Radio ressources mapping &pre-coding

OFDM

Wireless interface

Layer 1

Figure 1. Transmission system model.

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Step 7) For all possible allocation values , with i from 1 to N BW , findTBS(i) which verifies

0)(min >−necessaryTBS iTBS

where TBS(i) spans column i in TBS_table .

Step 8) For i from 1 to N BW , normalize all coding rates corresponding tovalues from TBS_vector as

i N i N PRB RE ch ×=

/ )(;

)()(

)(i N N iTBS

Rich

CRC scheduled rate

+−=δ ;

)max()(

)(rate

ratenormalized

ii R

δ

δ = .

Step 9) For i from 1 to N BW , normalize TBS values as

)max(*)(

)()(

TBS iTBS

TBS iTBS iTBS

necessarynormalized

−= .

Step 11) Compute the minimum TBS value closest to TBS necessary ,

which is 3GPP standardized as{ }( ))min(tan normalized normalized dardized s TBS RindexTBS TBS +=

The proposed voice allocation algorithm considers allpossible radio blocks allocations, from one to the maximumnumber of PRBs per bandwidth, N BW . All possible radioallocations are envisaged for the closest values of TBS thatcorrespond to specific code rates. We choose the optimumstandardized TBS as to minimize packet padding and obtain acode rate as close as possible to Rscheduled . Therefore, thepossible TBS values are normalized together with theircorresponding rates, and their normalized minima are finallychosen as optimum for transmission. The corresponding radio

blocks allocation, N PRB, is chosen for user resource mapping.

III. MULTI -LAYER REALISTIC LTE CAPACITY EVALUATIONAND COMPARISON RESULTS

Considering the above explained algorithm, we havecomputed the realistic LTE voice capacity and compared itsspectral efficiency in one of the used bandwidths, 1.4 MHz,with PMR and GSM radio networks.

A. LTE Voice Capacity EvaluationTable I presents the results for the realistic LTE voice

capacity with their specific simulation configurationparameters. The capacity is computed for three deploymentbandwidths, N BW : 1.4 MHz, 3 MHz and 5 MHz, which areinteresting for PMR. The chosen voice coder is the AMBE 2.45kbps. For the same air frame overhead, we are comparing theobtained capacities computed using two possible protocolsoverheads: one realistic ROHC compressed overhead and onetheoretical ideal overhead. Table I also gives the effectivespeech code rate, defined as

ch

protocolsnecessaryaudio N

H TBS R

−= , (3)

and layer one code rate

ch

CRC dardized slayer N

N TBS R

+= tan

1_. (4)

We note that voice capacity is very sensitive to the largeprotocol stack overhead. This may strongly influence the finalair frame resources partition. The higher layer's overhead andthe physical layer's CRC may represent more than 50 % of thetotal frame size for an average ROHC compression, therefore

TABLE I LTE DL VOICE CAPACITY EVALUATIONRESULTS .

System parameters Computedvalues

Realisticvalues

Idealvalues

Rvocoder 2450 bpsT voice 20 msVoice encoder AMBEP voice 49 bits

Transmission protocols RTP + UDP +IP 40 bytes 40bytesPDCP ROHC compression 3 bytes 1 byte

PDCP header 1 byte 0RLC header 1 byte 0MAC header 1 byte 0

Higher layer overhead H protocols 6 bytes 1 byte

Necessary LTE TBS TBS necessary 97 bits 57bits

Overhead percentage H/TBS necessary 49.5 % 14 %

TBS for QPSK 1/3 TBS standardized 104 bits 72bits

PHY CRC N CRC 24 bits 0 bits N BW 6 N ch 336 168

audio R 0,15 0,291_layer R 0,38 0,43

N PRB 2 1

Number users/sub-frame 3 6

1,4 MHzC OFDM =4

Numberusers/20 ms

period60 120

N BW 15 N ch 384 192

audio R 0,13 0,26

1_layer R 0,33 0,38

N PRB 2 1

Number users/sub-frame 7 15

3 MHzC OFDM =3

Numberusers/20 ms

period140 300

N BW 25 N ch 384 192

audio R 0,13 0,26

1_layer R 0,33 0,38

N PRB 2 1 Number users/

sub-frame 12 25

LTE DL

S OFDM =12;N pilots =8;

(PSS 1,SSS 2,

PBCH 3 notconsidered)

5 MHzC OFDM =3

Numberusers/20 ms

period240 500

1 Primary Synchronization Channel2 Secondary Synchronization Channel

3 Physical Broadcast Channel

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by adjusting the overhead to lower values closer to "ideal"percentages, the LTE capacity can approach its double.

LTE is not yet optimised for the usage of small voicepackets either. The small TBS choice is limited whenconstrained by a certain coding rate value. Our results showthat it is difficult to choose the closest TBS of the smallnecessary voice TBS. This means that the next closest TBS isconsiderably larger compared to the useful bits. Importantresources would be wasted if it were chosen for transmissionon the air frame. Radio resources allocation strategy is notoptimized either, since a minimum of one PRB pair must bereserved for each user. One PRB pair may prove to be largewhen using a very low bit rate voice coder and a very robustmodulation and coding scheme.

Regarding the air frame overhead, the performances may berestrained because of a limited PDCCH capacity (2 to 6 percontrol channel per sub-frame). But using persistent or semi-persistent allocation allows one PDCCH field to reserve userresources for a certain number of incoming sub-frames. Thismay be interpreted as a "virtual voice circuit" establishment.Therefore, even though LTE capacity depends on the numberof OFDM symbols assigned to control channels and also topilot signals, we do not consider the air frame overhead as anobstacle for voice communications.

B. LTE Voice Spectral Efficiency ComparisonTable II gives the PMR and GSM voice spectral

efficiencies, calculated considering EADS Cassidiandocuments. Table III gives LTE spectral efficiency results,obtained using the proposed algorithm.

We first consider the absolute channels capacity per carrier,followed by the cluster size and consequently the reuse offrequencies, and their respective voice spectral efficiencies

computed for different voice coders. We suppose we arecomparing the performances during 1 second (1 call lasts 1

second) and the technologies are all deployed in a ∆ f-BW =1.4MHz bandwidth.

Since it is unusual to compare circuit switched technologiesperformances with packet switched ones, we will only considerspecial LTE cases in which hypotheses are clearly stated andspeech transmission is comparable with PMR and GSM.

Hypotheses for PMR, GSM and LTE results are

• voice coders: AMBE 2.45 kbps, Algebraic Code ExcitedLinear Prediction (ACELP) 4.6 kbps, Regular Pulse(RP) CELP 6 kbps, Adaptive Multi-Rate (AMR) 12.2kbps, as specified in Tables II and III,

• radio access: FDD for each of the presentedtechnologies, with different possible time divisionmultiple access (TDMA) timeslots per frequency carrier;

For PMR and GSM only, we suppose

• 1 channel δ f denotes the reserved frequency bandwidthper time slot during one communication;

• the air throughput remains constant in kbpsindependently of the communication because only onetype of channel coding and modulation is used.

For packet switched LTE, we consider

• H protocols =6 bytes and N CRC =24 bits;• DL 1.4 MHz bandwidth configuration,• QPSK modulation with 1/3 coding rate; it has been

shown in EADS Cassidian studies that, for LTE tomatch PMR cell coverage in voice broadcast, an MCS ofQPSK 1/3 is necessary for a soft-input soft-output(SISO) signal to noise ratio (SNR) equal to 2dB;

• 1 transmission channel bandwidth is represented by onereserved frequency block ( N PRB , 180 x N PRB kHz, with

N PRB = 2,4,6) per 1 sub-frame for the duration of thecommunication (either dynamic or persistent allocation);

TABLE II PMR AND GSM PERFORMANCES RESULTS .

Technology TETRA TETRAPOL 10kHz TETRAPOL12,5 kHz GSM 900

Channel δ f 25 kHz 10 kHz 12,5 kHz 200 kHzTDMA 4 1 1 8

Number communications/ δ f : 4 1 1 8Number communications/ ∆ f-BW 224 140 112 1400/200*8 = 56

Reuse factor 16 12 12 9Communications/ ∆ f-BW /cell, C BW-cell 14 11,66 9,33 6,22

Voice codec bit rate, Rvocoder ACELP 4,6 kbps RPCELP 6 kbps RPCELP 6 kbps AMR 12,2Voice spectral efficiency, SE voice 0,046 bits/s/Hz/cell 0,05 bits/s/Hz/cell 0,04 bits/s/Hz/cell 0,054 bits/s/Hz/cell

TABLE III LTE PERFORMANCES COMPARISON RESULTS WITH PMR AND GSM.

Technology LTE (QPSK 1/3) LTE (QPSK 1/3) LTE (QPSK 1/3) LTE (QPSK 1/3)Channel δ f 2x180 kHz 4x180 kHz 4x180 kHz 6x180 kHz

TDMA 20 30 20 20Number communications/ δ f 20 30 20 20

Number communications/ ∆ f-BW 60 30 20 20Reuse factor 1 1 1 1

Communications/ ∆ f-BW /cell, C BW-cell 60 30 20 20Voice codec bit rate, Rvocoder AMBE 2,45 kbps ACELP 4,6 kbps RPCELP 6 kbps AMR 12,2

Voice spectral efficiency, SE voice 0,105 bits/s/Hz/cell 0,099 bits/s/Hz/cell 0,086 bits/s/Hz/cell 0,174 bits/s/Hz/cell

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• even if LTE has 6 PRBs in 1.4 MHz, if one user needs N PRB and N PRB is not a divisor of 6, we suppose only 6divided by N PRB such users may be scheduled in thisbandwidth;

• each user will need to transmit a new encoded voiceframe after T milliseconds, periodicity supposed equalto the TDMA transmission value for LTE;

• reuse factor equal to 1, which is true for datatransmission case. The actual reuse factor for speechcommunications will be larger, its theoretical valuebeing estimated around 3 or 4.

We compute the voice spectral efficiency percommunication and per cell as

BW f

cell BW vocoder voice

C RSE −

∆×

= (5)

where C BW-cell denotes the number of voice communications perbandwidth and per cell.

The results show that the number of possiblecommunications in LTE may be superior to those in PMR andGSM. But this number will be reduced once the real reusefactor for speech will be used. Even so, in the optimistic caseof all frequency bandwidth being used in every cell, thespectral efficiency of LTE is hardly the double of that of PMRand GSM. We also note that with the voice coder bit rateevolving from 2.45 kbps to 12.2 kbps, LTE spectral efficiencyvalues slightly increase, as the air resource allocations aresimilar.

IV. CONCLUSION

The all IP architecture is one of the innovations introducedby LTE. One way of improving the core network performanceand reduce latency is by using a flat architecture with moredirect routing from mobile device to end system. The 3GPPpropose the ROHC protocol for RTP/UDP/IP typical headercompression for multimedia applications as video and speech.The ROHC is a negotiation protocol for compressioncontinuously changes its status from very stable (3 bytes) tounstable (40 bytes). PMR systems can not tolerate these largeoverheads since the typical used voice bit rate is around 5 kbps,with a future target bit rate of 2,45 kbps. An overhead of 6bytes represents almost 50 % of the TBS at the physical layer, alarge percentage of the frame sent on the air interface. The IPprotocol, even though a desirable solution in the core network,it will not be efficient on the radio link and ROHC is notenough as a header compression protocol.

The standard offers a large choice of TBSs. But LTEresource allocation is not optimized for small voice packetstransmission, as given TBS values are considerably large andradio blocks allocations options are limited. In its current state,LTE Rel. 9 is not an appropriate solution for PMR voicesystems.

Better performances may be obtained with an improvedprotocol stack. A larger choice in TBS values and a moreflexible radio allocation may considerably improve results aswell. Nevertheless, promised data rates will be obtained inlarger bandwidths, where the control and pilot channelsoverhead occupies a small percentage, and with MIMO and

adaptive antenna systems, benefitting from larger radiochannels and smaller guard bands between carriers.

REFERENCES [1] 3GPP Technical Specification Group Services and System Aspects,

Service requirements for the Evolved Packet System (EPS), 3GPP TS22.278 V9.6.0, Sept. 2010.

[2] A. Larmo, M. Lindström, M. Meyer, G. Pelletier, J. Torsner, H.

Wiemann, “The LTE link-layer design,” IEEE Communications Magazine , Vol. 47, pp. 52-59, Apr. 2009.

[3] D. Astély, E. Dahlman, A. Furuskär, Y. Jading, M. Lindström, S.Parkvall, “LTE: the evolution of mobile broadband,” IEEECommunications Magazine , Vol. 47, pp. 44-51, Apr. 2009.

[4] 3GPP Technical Specification Group Services and System Aspects,Circuit Switched (CS) fallback in Evolved Packet System (EPS), Stage 2 ,3GPP TS 23.272 V10.5.0, Sept. 2011.

[5] 3GPP Technical Specification Group Core Network and Terminals; IMSmultimedia telephony communication service and supplementaryservices, Stage 3 , 3GPP TS 24.173 V11.0.0, Sept. 2011.

[6] 3GPP Technical Specification Group Core Network and Terminals; Evolved Packet System (EPS); 3GPP Sv interface (MME to MSC, andSGSN to MSC) for SRVCC, 3GPP TS 29.280 V11.0.0, Sept. 2011.

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LTE,” in MIPRO , Opatija, Croatia, 2010.[8] Terrestrial Trunked Radio (TETRA) Voice plus Data, Part 2: Air

Interface , ETSI EN 300 392-2 V3.2.1, Sept. 2007.[9] TETRAPOL Specifications, Part 1: General Network Design; Reference

Model , TETRAPOL Forum, PAS 0001-1-1, V3.0.4, Nov. 1999.[10] TETRAPOL Specifications, Part 2: Radio Air Interface , TETRAPOL

Forum, PAS 0001-2, V3.0.0, Nov. 1999.[11] TETRA Release 2, Designer's Guide; TETRA High-Speed Data (HSD);

TETRA Enhanced Data System (TEDS) , ETSI TR 102.580 V1.1.1, Oct.2007.

[12] 3GPP Evolved Universal Terrestrial Radio Access (E-UTRA); User Equipment (UE) radio transmission and reception , 3GPP TS 36.101V9.2.0, Dec. 2009.

[13] 3GPP Evolved Universal Terrestrial Radio Access (E-UTRA); PhysicalChannels and Modulation , 3GPP TS 36.211 V9.0.0, Dec. 2009.

[14] 3GPP Evolved Universal Terrestrial Radio Access (E-UTRA); Multiplexing and channel coding , 3GPP TS 36.212 V9.0.0, Dec. 2009.

[15] 3GPP Evolved Universal Terrestrial Radio Access (E-UTRA); Physicallayer procedures , 3GPP TS 36.213 V9.0.1, Dec. 2009.

[16] 3GPP Evolved Universal Terrestrial Radio Access (E-UTRA); Medium Access Control (MAC) protocol specification , 3GPP TS 36.321 V9.1.0,Dec. 2009.

[17] H. Holma, A. Toskala, LTE for UMTS: OFDMA and SC-FDMA Based Radio Access , John Wiley and Sons Ltd., UK, 2009.

[18] EDGE, HSPA, LTE: Broadband Innovation , 3G Americas white paper,Sep. 2008.

[19] Methodology for the assessment of PMR systems in terms of spectrumefficiency, operation and implementation – CEPT, Bucharest 1997