air trafﬁc control (atc) is an aeronautical service ... · of the international civil aviation...

Adaptation of the E-Model forSatellite Internet ProtocolRadio Calls in Air TrafficControl

SPYROS APOSTOLACOSAPOSTOLOS MELIONESinAccessAthens, GreeceandNational Technical University of AthensAthens, Greece

STEFANO BADESSIEuropean Space AgencyNoordwijk, the Netherlands

GEORGE STASSINOPOULOS, Senior Member, IEEENational Technical University of AthensAthens, Greece

This paper proves that voice over Internet protocol (IP)technology can be combined with satellite transmission in air trafficcontrol (ATC) applications provided that the systems used toimplement this approach have been meticulously designed to meetthe end-to-end delay requirements and to certify the quality ofservice they offer under strenuous conditions. We propose a modifiedE-model and prove it can be used as a radio network planning tool,because once the requirements that were drawn from it werefulfilled, the users of the resulting service accepted it as sufficient interms of call quality. The approach has been applied on theSATWAYS system, cofunded by the European Space Agency, anddesigned and implemented to extend the coverage of radiofrequencies within airspaces when the commonly used landlinecommunication infrastructure is either too costly or impossible toinstall and maintain. The paper provides evidence that the way to anall-IP ATC network is wide open.

Manuscript received January 31, 2013; revised August 22, 2013; releasedfor publication June 13, 2014.

DOI. No. 10.1109/TAES.2014.130064.

Refereeing of this contribution was handled by M. Rice.

This work was supported by the European Space Agency under GrantNo. 20379/06/NL/US of the Advanced Research in TelecommunicationsSystems framework.

Authors’ addresses: S. Apostolacos, inAccess, Maroussi, Athens,GR-15125 Greece, E-mail: ([email protected]); A. Meliones,Department of Digital Systems, University of Piraeus, GR-18534Greece, E-mail: ([email protected]); S. Badessi, ESA ESRIN, Frascati,I-00044 Italy, E-mail: ([email protected]); G. Stassinopoulos,School of Electrical and Computer Engineering, National TechnicalUniversity of Athens, 15780 Greece, E-mail: ([email protected]).

0018-9251/15/$26.00 C© 2015 IEEE

I. INTRODUCTION

Air traffic control (ATC) is an aeronautical serviceprovided to pilots by qualified flight controllers, whoguide aircraft, which are either on the ground or airborne,with the primary goals of avoiding collisions, organizingand predicting the flow of traffic, and provisioning othernecessary support [1]. Collision avoidance is often calledseparation, which is the maintenance of minimumdistances among aircraft. In most countries, ATC servicesare provided by the respective national civil aviationauthorities. The regions of an airspace for which flightcontrollers are responsible for maintaining the separationare characterized as controlled airspace, while all othersare collectively called uncontrolled airspace. Dependingon the type of flight and airspace, the controllers eitherissue commands, which the pilots are obliged to follow, orjust provide information to aid the pilot, with the latterhaving the last word. There are three basic components ofan ATC: communication, navigation, and surveillance(CNS). All three must be operational for a controlledairspace to be safe to fly in.

Voice communications are one of the three primaryinstruments in ATC, without which it is practicallyimpossible to fly an aircraft without compromising safety.The infrastructure that is necessary to implement a voiceservice in ATC is a network of radio transceivers, installedon the ground and tuned at specific frequencies so as toprovide full geographical coverage at all altitudes. Thepositions and the characteristics of these VHF transceiversare carefully selected using specialized tools so as tomaximize the coverage and capacity of the network, sincebecause of the nature of radio communications, whensomeone emits on a certain frequency, everybody elsetuned to the same channel can listen to that person.Redundancy setups are also mandatorily employed.

The voice to and from these remote VHF transceivershas to be relayed to the flight controller positions, usingsophisticated switching equipment collectively denoted asthe voice communications system (VCS), whichdynamically creates connections among pilots andcontrollers according to the responsibility sector of thelatter. The sector and frequency plans are updateddepending on the season, time of the day, and flightdensity. During their flight, the pilots change sectors,frequencies, and consequently controllers using aprocedure called handoff, which resembles the oneemployed in a cellular mobile telephony network.

The interconnection between the remote radio systemsand the VCS is accomplished using ground-based analogor digital switching circuits of the public switchedtelephone network (PSTN), which are leased by the civilaviation authority from a communication service provider.The architecture is presented in Fig. 1.

Over time and because of technological advances invoice transport, such as the introduction of voice overInternet protocol (VoIP), leased lines became overlyexpensive and inflexible. Consequently, in the context of

IEEE TRANSACTIONS ON AEROSPACE AND ELECTRONIC SYSTEMS VOL. 51, NO. 1 JANUARY 2015 81

Fig. 1. ATC voice communications infrastructure.

both the Single European Sky Air Traffic ManagementResearch (SESAR) [2] and the Next Generation AirTransportation System (NextGen, driven by NASA andthe Federal Aviation Administration, or FAA) [3] projects,discussions regarding the adoption of packet-switchedtransport for aeronautical radio communications wereinitiated. The advantages are numerous. First, the voiceand data transport networking infrastructure can beunified, yielding lower costs and better management. Thereliability is enhanced, because redundancy can beachieved in a variety of ways permitted bypacket-switched data transport. For example, the VCSpresented in Fig. 1, which is a single point of failure in theleased-line scenario, can now be distributed. Moreover,when functionality and interoperability standards areestablished, the civil aviation authorities will becomeindependent from the equipment suppliers, leading toreduced prices and implementation of new services.Finally, the adoption of mobile Internet protocol (IP)techniques brings ATC one step closer to the “oneflight–one controller” vision, eradicating the need forhandovers, which is an important source ofcontroller-related deadly errors in flight control.

The use of VoIP in ATC (referred to herein as radioover IP, or RoIP) was assessed recently by the EuropeanCommission’s SESAR program [4] and the U.S. FAA [5],both launching respective standardization efforts regardingits use in operational environments under the supervisionof the International Civil Aviation Organization (ICAO),Eurocontrol (the European civil aviation organization),and the European Organisation for Civil AviationEquipment (EuroCAE) [6–8]. The related architecture,ratified by the Vienna Convention [6] is presented inFig. 2. Because it is impossible to abandon the currentlydeployed communication infrastructure and to facilitatethe migration efforts, the Vienna Convention predicts theuse of specialized equipment (gateways, called RoIPgateways herein) to achieve interoperability between older,non-IP enabled equipment and IP networks. These RoIPgateways are the main focus of the remainder of this paper.

Though the introduction of RoIP technologies in ATChas the previously mentioned advantages, it does not solvethe problems related to radio coverage. For countries witha complex terrain, such as Greece (a mountainous countrywith a huge sea area around it), it is extremely difficult to

Fig. 2. VoIP ATC architecture ratified by Vienna Convention.

provide satisfactory radio coverage, especially for lowaltitudes where visual flight rule (VFR, where the pilotdrives the aircraft using his eyes) flights operate becauseof the large number of VHF transceivers andcorresponding networking infrastructure that has to beinstalled. As a result, many accidents, especially involvinghelicopters, occur each year, resulting in several injuriesand deaths. One especially problematic phase of ahelicopter flight at present is the approach to the remoteheliport, because the controller seldom receives feedbackon whether the pilot has successfully landed. This meansthat the flight plan is closed over mobile phones, which isoutside regulations and extremely error prone.Consequently, in the case of an accident, the responsetimes are measured in hours, not minutes. Because of theincrease of VFR flight density over the last years, thesituation has become increasingly difficult to tackle.

The alternative, which is explored herein, is theadoption of satellite communications to address thisproblem. SATCOM has been in use for ATC for a numberof years in areas where ground-based radio coverage isimpossible to achieve (e.g., oceanic areas [9]), and mostaircraft are equipped with satellite communication means.Furthermore, SATCOM has a prominent place in thefuture ATC system outlined by SESAR [4], because it willbe used to implement a number of new services. TheEuropean Space Agency (ESA) has kick-started the IRISproject [10], which, in direct connection to SESAR,studies this particular domain. In any case, this approach isnot expected to resolve the previously mentionedproblems in the near future for a number of reasons. First,the installation of new avionics is required so as to allowaccess to satellite circuits [11]. The cost for such atransition is huge, and for the airlines to ratify it, certainadditional advantages that come with it, such as theconcept of free flight [12], will have to be enabled. Thiswill not happen, according to [2], before 2025. In addition,even if airlines are finally convinced, the necessarychanges to small aircraft are expected to occur later. Aparallel effort led by Eurocontrol and the FAA is the

82 IEEE TRANSACTIONS ON AEROSPACE AND ELECTRONIC SYSTEMS VOL. 51, NO. 1 JANUARY 2015

Communications Operating Concept and Requirements(COCR) for the Future Radio System, now at version 2.0(see [30]), which targets at the modernization ofair-to-ground communications to enable data transportover the existing VHF radio infrastructure (VHF data link,or VDL), also mandating changes to the aircraft avionicsand the ground segment foreseen after a 2020 milestone.

On the contrary, the introduction of VoIP technologiesfor radio communication transport over the groundsegment is nonintrusive, and this infrastructure will bemaintained as an additional redundant path even whendirect satellite links to the aircraft are in place.

The principal question that the work presented hereinattempts to answer is whether it would be possible tocombine RoIP with satellite transmission technologies toimplement parts of the ground segment of the future ATCnetworking infrastructure whenever it is not possible toinstall and maintain other networking means. An answerto this question is extremely important for the civilaviation authorities of countries like Greece, which havegreat difficulties in deploying ground-basedtelecommunication infrastructure, because they would beprovided with the capability to extend their network withautonomous remote VHF transceivers at minimum costand with full compatibility with the future vision of ATC;a power supply is the only requirement.

At a first glance, the problem looks simple. Thetransmission of VoIP data over satellite IP networks isoperational for at least 10 years, so the extension toimplement radio communications seemed obvious, at leastuntil the first survey of the literature was carried out,revealing that the issue has not yet been tackled.

The basic function of ATC, as presented earlier, is tomaintain a minimum separation among aircraft so as toavoid collisions. The denser the traffic, the smaller thedistance has to be. The same applies to the time availablefor voice communications between pilots and flightcontrollers if a minimum margin for error is maintained.Everything plays its role; the phrasing, the syllable rate,the human response time, and the communicationinfrastructure are all decisive factors that determine themaximum achievable flight density.

Given the high criticality of time in ATC, thecombination of RoIP and satellite seems risky, because theadditive induced delay could incur serious flight safetyissues and reduce achievable flight densities. In addition,the cost of satellite capacity mandates the compression ofvoice, which reduces its quality and intelligibility,increasing the minimum number of exchanges betweenthe two ends and correspondingly increasing their reactiontimes. One final hurdle obstructing this research path isthat humans have to be involved in the evaluation of everyproposed solution, because their performance affects theresults. Consequently, a prototype system has to beimplemented, deployed, and commissioned in order for itto be tested. To construct such a system, it is necessary tohave specific requirements correlating the parameters ofthe implementation, with the foreseen impact on the

perception of the user based on a certain model, the questfor which in existing published research work has provedfruitless.

The structure of the approach presented herein startswith a presentation of the factors affecting the quality of avoice service and the E-model, which relates them tophysically measured parameters of the communicationpath used to implement it. Then, specific simplifications ofthe E-model so that it can be used to quantify radio callquality are proposed, and specific requirements that willhave to be fulfilled by the designed system are recorded.In the next section, details regarding the implementationof a RoIP gateway, taking into consideration theserequirements, are presented. Finally, the results of afull-scale, real-life experimental validation of the useracceptance of that system are recorded, together with a setof conclusions and future work recommendations.

II. VOICE QUALITY OF SERVICE AND THE E-MODEL

The parameters that affect the quality of service (QoS)classification of a voice service like the one based on RoIPover satellite technology are the call quality, call setupquality, and service availability. The call setup quality ismainly characterized by the call setup time, which isperceived by the user as the responsiveness of the serviceand the probability to reject a call, while the call quality ischaracterized by the overall transmission quality duringthe call. Call quality is affected by two importantparameters, namely, the end-to-end delay, which mainlyaffects the interactivity of a conversation (the mouth of thespeaker to the ear of the listener), and the end-to-endspeech quality (end-to-end voice performance), which isthe one-way speech quality as perceived in anoninteractive situation. Service availability is thepercentage of a predetermined time that the voice serviceis available. A suitable framework for call quality analysisin the context of a RoIP over satellite system is theE-model, presented in [13] and [14].

The E-model takes into consideration the impairmentsimposed during the transmission of speech to calculate theso-called R factor. A higher R factor corresponds to abetter call quality, 0 being the worst value, 80 the minimalquality for telephone calls through the PSTN (toll quality),and 100 being the best achievable value. The transmissionrating factor R comprises of five terms (uncorrelatedimpairment factors), which combine different types ofimpairments. According to the psychoacousticalbackground theory, their effect is additive:

R = Ro − Is − Id − Ie + A

where Ro represents the basic signal-to-noise ratio,including noise sources such as circuit noise and roomnoise, Is is a combination of all impairments, which occurmore or less simultaneously with the voice signal, Id

represents the impairments caused by delay, and theequipment impairment factor Ie represents impairmentscaused by low bit-rate codecs. Finally, the advantage factor

APOSTOLACOS ET AL.: E-MODEL ADAPTATION FOR SATELLITE IP RADIO CALLS IN ATC 83

A allows compensation of impairment factors when thereare other advantages of access to the user. The Ro, Is, andId terms are further subdivided into specific impairmentvalues that are influenced by a number of parameters, aspresented within the basic model of the end-to-end voicepath for a telephone connection included in [13]. TheE-model applies to each transmission direction separately,so in asymmetric scenarios, two R factors (one pertransmission direction) have to be calculated.

III. RADIO CALL QUALITY ANALYSIS USINGTHE E-MODEL

The E-model has been developed to measure the callquality within telephone networks, and its adaptation tosystems involving radios has not been, to the authors’knowledge, addressed in the literature. An approach,together with the assumptions made, is attempted in theensuing, where the E-model, like any other voice qualitymeasurement algorithm, is based on segmenting a voicecommunication path into two transmission directions andthen quantifying the call quality for each one separately.This in turn implies that any bidirectional communicationscheme, such as radio communications, can theoreticallybe studied using the E-model as a tool. Finally, radio voiceoccupies the same bandwidth as telephone calls(300 Hz–3.4 kHz). We should, however, keep in mind thatadopting the E-model for radio call quality analysis can beconsidered to provide pessimistic evaluations becauseQoS frameworks for telephony apply for a voice servicetargeted at the public, but people who use aeronauticalvoice communication are highly trained officers who areaware of the operational difficulties, so they can and havebeen trained to cope with them.

A. The End-to-End Voice Path

At present, the Hellenic Civil Aviation Authority(HCAA), similarly to all other civil aviation authoritiesworldwide, operates a large number of remote radiostations controlling air traffic over the Greek air spacethrough 4-wire leased lines relaying the radio audio to andfrom its headquarters in Athens. The audio is relayedthrough a short 4-wire section from the remote radiostation to the nearest PSTN switch, enters the digitalswitching infrastructure, and then is delivered digitally toa central switching station at HCAA headquarters. Theend-to-end path is shown in Fig. 1.

A RoIP over satellite system can be deployed tohard-to-reach areas where leased-line connections arecostly or impossible to install and maintain as acomplement to existing HCAA infrastructure. Thisend-to-end architecture is shown in Fig. 3. Standing onthat, the system components affecting the call quality forradio calls are 1) the radio equipment on the aircraft and atthe remote site, 2) the RoIP gateway installed at theremote site, 3) any IP networking equipment at the remotesite (switches, routers, etc.), 4) IP networking equipmentat HCAA headquarters, 5) the RoIP gateway installed at

Fig. 3. Radio call using RoIP over satellite.

HCAA headquarters, 6) the two-way communicationequipment at HCAA headquarters, 7) The 4-wire sectionsconnecting communication equipment with the RoIPgateways on both ends, 8) the earth segment of the satellitelink on both sides (satellite modem equipment), and 9) thespace segment (the satellite).

B. Adaptation of the E-Model to Radio Calls

In the context of Figs. 1 and 3 and whenever the pilotwishes to contact the flight controller at HCAAheadquarters, the pilot presses the push-to-talk (PTT)button of the radio equipment and talks in the microphone.During this talking period, a sidetone path provides thepilot’s voice to the earpiece. The sidetone andtransmit/receive audio levels are configurable by the userusing controls of the radio terminal. The aircraft radiotransmitter emits the pilot voice modulated on a specificVHF frequency. The radio receiver on the ground, tuned atthe same frequency, picks up the emission and translatesthe existence of valid energy content on that frequency toa squelch signal (an incoming audio notification)presented, along with the received voice, at the groundradio equipment output. Both the squelch signal and theaudio are relayed to the flight controller position at HCAAheadquarters using either leased lines or RoIP oversatellite. When the controller wants to respond, thecontroller presses the PTT button and repeats the sameactions, using the transmitter on the ground to excite theaircraft receiver.

Taking into consideration the description of a radiocommunication presented earlier, the model of a telephoneconnection presented in [13] has to be modified torepresent the end-to-end voice paths of Figs. 1 and 3, andparameters involved in R factor calculations that do notapply for radio calls have to be removed. One importantmodification in this direction is related to telephoneconnections involving the two 2-wire sections at the sendand receive side and one 4-wire section between, whichrepresents the PSTN infrastructure, and the two directionsof transmission in a radio call being separated from oneend to the other so that 2- to 4-wire and 4- to 2-wireconversions do not exist. Moreover, while telephoneconnections are full duplex, radio communication isalways half-duplex, as presented earlier. Consequently,echo can be avoided, so impairment factors related to echocan also be eliminated.

There is a hidden echo source in the voice paths ofFigs. 1 and 3. When the radio transmitter on the groundemits the flight controller voice toward the airborne radio,


Fig. 4. ITU-T Recommendation G.107 end-to-end voice path modeladapted to radio calls.

the radio receiver on the ground, tuned at the samefrequency, considers the emission as incoming audiorelayed back to the flight controller. This feedback ispresently used to implement the sidetone for the flightcontroller; the delays the signal experiences in that case donot exceed 10 ms because of the use of leased lines.Should the delays involved in satellite transmission beinserted in the voice path, however, the result would beunacceptable. The other transmission direction does notpresent such problems, because only acoustic couplingcould occur at the flight controller position. However, thisis not possible because the flight controller has the PTTbutton depressed when not talking, so the feedback path isdisrupted.

When the flight controller talks, the pilot listens, soecho cancellation is not necessary and an elementary echosuppressor can be implemented, consisting of cutting offthe return path within the RoIP gateway installed at theremote site when a PTT from the flight controller ispresent. The sidetone between the flight controller’smicrophone and earpiece can be implemented within theRoIP gateway installed at HCAA headquarters. Based onthe analysis presented earlier, the model of a telephoneconnection presented in [13] can be modified to representa radio connection as per Fig. 4, depending on thetransmission direction.

C. Calculation of the R Factor for Radio Calls

1) The Basic Signal-to-Noise Ratio Ro: The basicsignal-to-noise ratio Ro is defined, according to [13], as

Ro = 15 − 1.5(SLR + No)

where the term No (in dBm0p) is the power addition ofdifferent noise sources:

No = 10log[10

Nc10 + 10

Nos10 + 10

Nor10 + 10

Nf o

10

]

In the pilot–to–flight controller transmission direction,the microphone of the latter is off. Therefore, the formulafor the Nor factor presented in [13] can be significantlysimplified, because the environment noise at the receive

side does not affect the listener via the sidetone because ofthe absence of sidetone during reception. So

Nor = RLR − 121 + Pre + 0.008(Pre − 35)2

where

Pre = Pr + 10log10

[1 + 10

10−LST R10

] ∼= Pr

Here, Pr is the level of the room noise at the receptionside, measured at the position of the listener’s ear. Thevalue of Pr is equal to the noise level in the flight controlroom reduced by the attenuation achieved by the headsetused by the flight controller. According to [15], theenvironment noise level varies significantly with anaverage value of 80 dB(A-weighted decibels), while theisolation provided by the flight controller headset is15–30 dB. In the ensuing, we use a value of 22 dB forheadset isolation, which is compatible with thespecifications of a large number of headsets in the market.

The Nos factor, referred to in [13], is calculated usingthe formula

Nos = Ps − SLR − Ds − 100

+ 0.004 (Ps − OLR − Ds − 14)2

where Ps is the level of the room noise at the transmissionside measured at the microphone position. A survey of theliterature regarding noise level in the pilot cabin ofairborne vehicles has been reported in [16], in which theauthors conclude to a high noise level value of 90 dB(A).The Ds factor defines the suppression of the room noiseachieved by the microphone. EuroCAE [15] indicates anoise suppression ratio of at least 20 dB featured bymicrophones used in ATC, which will have to besubtracted from the aforementioned cabin noise level tocalculate the noise level measured at the pilot microphone.The resulting value (70 dB(A)) is used in the ensuing.

Furthermore, it should be taken into account that thevoice signal level at the pilot side is at least 7 dB largerthan the typical value measured in the telephone network[15], which increases equivalently the signal-to-noiseratio, because the E-model has been designed fortelephone calls. This can be accounted for in Ro,increasing it by 7 dB in the pilot–to–flight controllertransmission direction.

The optimal values referred to in [13] for the receiveloudness rating (RLR, + 2 dB) and send loudness rating(SLR, + 8 dB) can be achieved by the flight controller andpilot radio equipment using existing programmable gains.Therefore, overall loudness rating (OLR) is set to+ 10 dB, because OLR = SLR + RLR.

Because of the absence of relevant data, Nc and Nfor

(Nfo = Nfor + RLR) is set to the values proposed in [13],i.e., −70 and −64 dBm0p, respectively. As is obvious inthe ensuing, the contribution of these parameters (circuitnoise and noise floor) to Ro is negligible; therefore, apotential variation from the exact values is not importantto the analysis.


TABLE IContribution of Various Parameters to Calculating No

Value (dBm0p; Value (dBm0p; (FlightPilot–to–Flight Controller Controller–to–Pilot

Parameter Transmission Direction) Transmission Direction)

Nos −29.536 −45.376Nor −56.768 −39.2Nc −70 −70Nfo −62 −62No −29.525 −38.24

It is obvious that Ro is independent of thetelecommunication infrastructure and only depends onequipment and operational environment parameters in theend-to-end paths shown in Fig. 4. Using the valuescalculated in this section, Table I presents the contributionof each factor to calculating the value of No for bothtransmission directions. The differences between the Nocomponents are huge because of the logarithmic scale. Asexpected, in the pilot–to–flight controller transmissiondirection, No is dominated by the environment noise at thetransmission side, i.e., the cabin noise of the aircraft. Inthe reverse direction, the noise of the flight control room issuppressed by 20 dB by the microphone, while the noise atthe reception side (pilot cabin) is determined by theisolation achieved by the closed type headset used by thepilot as protection from the high noise level (equal to22 dB, as mentioned earlier). Again, No is dominated bythe environment noise inside the pilot cabin.

2) The Simultaneous Impairment Factor Is:According to [13], the factor Is is the sum of allimpairments, which may occur more or lesssimultaneously with the voice transmission and is dividedinto three further specific impairment factors:

Is = Iolr + Ist + Iq

Referring to [13], Iolr represents the decrease in qualitycaused by too-low values of OLR and can be calculated bythe following formula:

Iolr = 20

[{1 +

(Xolr

8

)8} 18

− Xolr

8

]

where Xolr = OLR + 0.2 (64 + No − RLR).The optimal values for OLR (+10 dB) and RLR

(+2 dB) according to [13] can be achieved by the flightcontroller and pilot using existing programmable gains ontheir equipment. In that case, Iolr equals to 0.0157 and0.0343 in the pilot–to–flight controller transmissiondirection and in the reverse direction, respectively. Thesevalues were calculated using the No values of the previousparagraph and are independent of the telecommunicationinfrastructure used. They only depend on equipment andoperational environment parameters in the end-to-endpaths shown in Fig. 4. This effectively zeros out Iolr, whichcan safely be omitted from R factor calculations hereafter.

The analysis continues with Ist, which represents theimpairment caused by a nonoptimum sidetone, or the

Fig. 5. Ist value vs. STMR value.

leakage of one’s voice into that person’s earpiece,providing reassurance that the line is live during aconversation. Sidetone in telephony is created by thetelephone device, so it is set to a value by design andcannot be altered by the user. Though sidetone is alsoapplicable to radio communications, its level isconfigurable, so the sidetone masking rating (STMR),which is the ratio of a defined input acoustic signal to theresulting acoustic output signal, can also be adjusted.

The formulas involved in calculating Ist can also begreatly simplified because there is no echo in radiocommunication due to its half-duplex nature. The talkerecho loudness rating (TELR) can thus be set to a largevalue (ideally infinite) so as to zero out the second term ofthe formula for STMRo calculation [13]. This leads to theconclusion that STMRo is equal to STMR, so the equationfor Ist can be rewritten as

Ist =12

[1+

(ST MRo−13

6

)8] 18

−28

[1+

(ST MRo

19.4

)35] 135

−13

[1 +

(ST MRo − 3

33

)13] 113

+ 29

where

ST MRo = −10 log[10− ST MR

10 + e− T4 10− T ELR

10

]

Plotting, in Fig. 5, Ist versus the allowable STMRvalues as per [13], one can reach the conclusion that thereis an optimum value for STMR that does not depend onother parameters and can lead Ist to 0. Even negative Ist

values can be achieved, though the improvement is hardlyworth accounting for. The programmability of the sidetonelevel allows fine tuning so as to zero out Ist, which will nolonger be included in R factor calculations.

3) The Delay Impairment Factor Id: The delayimpairment factor Id represents all impairments because ofdelay of voice signals. It can be expressed as Id = Idte +Idle + Idd. As presented within [13], the factor Idte givesan estimate for the impairments because of talker echo,while the factor Idle represents impairments because oflistener echo. Both impairment factors are similarlymodeled and can thus be similarly simplified, taking into


consideration the effect that the absence of echo has on theformulas used for their calculation, because radiocommunication is inherently half-duplex. As alreadystated, the absence of echo on both ends of a radiocommunication leads in an infinite TELR as far as talkerecho is concerned and an infinite weighted echo path loss(WEPL) as far as listener echo is concerned. According tothe formulas of [13], infinite TELR and WEPL lead toinfinite Re and Rle, respectively, irrelevant to all otherparameters used in the calculations. Therefore,√

(Roe−Re)2

4+100 ≈ Re

2and

√(Ro−Rle)2

4+ 169 ≈ Rle

2

Idte and Idle can thus be modified for infinite TELR andWEPL, because Idte = −(1 − eT) and Idle = 0. So, bothfactors can be set to 0 for subsequent calculations;consequently, Id = Idd. More specifically, setting Idle to 0 isa pessimistic approach (the minimum value for Idle is −1for T → ∞).

The third factor (Idd) represents the impairment causedby too-long absolute delay Ta, which occurs even withperfect echo cancellation (or no echo, as in our case) andis calculated as

Idd = 25

{(1 + X6)

16 − 3

(1 +

[X

3

]6) 16

+ 2

},

X = log(

Ta

100

)log 2

, for Ta > 100 ms

and Idd = 0 for Ta ≤ 100 ms.This parameter is extremely important in the case of

high-delay links such as satellite communications and isrevisited within the following sections. In conclusion, themain outcome is that Id can only be affected by Idd.

4) The Value of the R Factor for Radio Calls OverLeased Lines: The simplifications proposed within theprevious sections can now be used to derive the value ofthe R factor for radio calls over leased lines according tothe end-to-end architecture of Fig. 1 for both transmissiondirections. All that is needed is to define values for theabsolute end-to-end delay (Ta), the impairment factor (Iq),and the advantage factor (A). As far as the first parameteris concerned, measurements carried out by HCAA indicatean upper limit of 20 ms for both transmission directions.For the calculation of the value of the second parameter,the number of quantization distortion units (QDUs) has tobe defined. The end-to-end voice path presented in Fig. 1contains at a minimum two analog-to-digital anddigital-to-analog conversions, where the digital value iscodified according to 8-bit A-law/μ-law pulse-codemodulation (PCM), because it has to be conveyed over thetelephone network. In this case, the QDU number is equalto 2 according to [13]. If more analog sections exist withinthe PSTN network, this number may be larger.

With this parameterization the value of the R factor,considering A = 0, is equal to 51.596 in the pilot–to–flightcontroller transmission direction and 57.668 in theopposite transmission direction. These values represent

the typical performance of an aeronautical radiocommunication system as far as voice quality is concernedand are used in the ensuing as a reference point. It isevident that the R factor is considerably lower comparedto the one expected from a voice service for the public(R = 80, according to [17]). This difference demonstratesthe training level of flight controllers in achieving reliablevoice communications in such harsh environments.

5) The Value of the R Factor Using RoIP OverSatellite: Once the leased-line reference point has beenestablished, the R factor for radio calls employing RoIPover satellite is calculated, using the end-to-endcommunication path presented in Fig. 3. The No parameterhas the same values for both transmission directions, andthe simplifications proposed within the previous sectionsas far as the Is and Id impairment factors are concernedequally apply. The prominent differentiations are asfollows:

• The need for the use of a low bit-rate voice codecbecause of the high cost of satellite bandwidth,introducing a nonzero value for the Ie factor

• The value of the Iq factor, because in this case onlyone conversion from analog to digital, and vice versa,exists, whose accuracy can be arbitrarily selected as adesign option

• The value of the Idd factor, because satellitetransmission has a considerably higher end-to-end delay

The following sections analyze the contribution of thepreviously mentioned differentiations to the R factor whencompared to the leased-line scenario for both transmissiondirections.

a) The Ie impairment factor: The research on theapplicability of low bit-rate voice coding in aeronauticalradio communications started in the 1990s as part of thedevelopment of VDL technology, which is a digitalmodulation technique to transmit voice and data betweenaircraft and ground stations using existing VHF channels(adopted in the context of COCR [30]). At first, the use ofproprietary codecs was proposed [16]. The quick adoptionof VoIP technology has, however, made imperative theneed to endorse known and tested solutions alreadysupported by equipment manufacturers. Friedman-Berget al. [18] present the subjective evaluation of severalvoice codecs (G.711 at 64 kbps, G.726 at 32 kbps, G.729at 8 kbps, G.723.1 at 5.3 kbps, and G.726 at 16 kbps). Theuser ratings follow the values of the Ie impairment factoraccording to [13], provided that one encoding/decodingpair exists along the whole transmission path.

Using this set of voice codecs, a selection criterioncalled capacity efficiency had to be defined as the ratio ofthe value of the Ie factor over the necessary bandwidth toconvey one voice channel using a specific voice encodingscheme. Larger values denote that a voice codec uses upmore bandwidth compared to the voice quality it offerswhen compared with another codec presenting a smallervalue for this metric. Comparing the values of this metricwith [18] and taking into consideration the cost of satellite


Fig. 6. Iq value vs. Ro vs. QDU.

bandwidth, the remainder of the analysis presented hereinencompasses only G.729 and G.723.1 as possibilities.More evidence leading to the selection of one of them ispresented in the ensuing. The Ie values stated within [13](which are the same irrespective of the transmissiondirection) apply only when the packet loss is below 0.5%[19]. The fulfillment of this requirement is proven inforthcoming sections using the results from field testing.

b) The Iq impairment factor: As has already beenstated, the analog-to-digital and digital-to-analogconversion method can be arbitrarily selected in the caseof radio voice transmission using RoIP over satellite bythe designer of the RoIP gateway. Consequently, the valueof the Iq impairment factor can be controlled in contrast tothe leased-line scenario (Fig. 1), where using 8-bit PCM ismandated by the call being bound to use of the PSTN.Should 16-bit linear PCM be selected, the number ofQDUs is equal to 0, according to [13]. Fig. 6 presents thevalue for the Iq factor when the number of QDUs is 0 forthe whole range of the values of the Ro factor. Itapproaches 0 in any case, so Iq can be omitted fromsubsequent calculations, provided that 16-bit linear PCMencoding is employed, irrespective from the transmissiondirection.

c) The Idd impairment factor: Delay is expected to bethe most important factor that affects the end-to-end voicequality and consequently the acceptance of the service byits users. ICAO relates the voice communication delaywith the minimum separation among aircraft through thenotion of required communication performance (RCP)[20], which has an operational, not a technical,interpretation. This means that a very high-delaycommunication link, such as one involving a satellite, mayaffect the achievable airspace density so as to maintain anacceptable flight safety level. The RCP takes intoconsideration the human factors associated with theresponse times of the pilot and flight controller, the flightprocedures, and the operational environment, because itmodels the duration, continuity, availability, and integrityof the relay of commands, approvals, information, ordemands. The RCP does not deal with the underlying

communication technology directly but only defines a setof performance parameters to ensure that thecommunication system can empower its users to fulfillspecific procedures related to ATC with an acceptablemargin of safety.

Cardosi [21] uses published and unpublished data fromoperational communications within the American airspaceto establish bounds on the human performance necessaryto maintain a certain RCP. These bounds are thenprocessed to extract specific requirements on the employedcommunication system. The results indicate that anunderlying communication system with a roundtrip delayequal to 0.77 s (385 ms one way, assuming symmetricalbehavior) is sufficient to support even the most demandingoperational scenario (emergency communications) at thesmallest allowable separation of 5 nautical miles (NM).

One more source of information regarding the issue ofacceptable delay in aeronautical radio communications isthe existing literature on the subjective user evaluation ofexisting high-delay links under simulated conditions. In[22], the methodology and results of such a studyconducted by the FAA are presented. Ten flight controllersand three one-way delay scenarios were tested (290, 390,and 790 ms in the flight controller–to–pilot transmissiondirection and 260, 360, and 760 ms in the reversetransmission direction). The differences according to thetransmission direction are because a VDL3 link was beingsimulated and the call setup procedures are differentaccording to which party initiates the radio call. A denseairspace was then simulated, and the results wereevaluated using the controller acceptance rating scale(CARS) method (see [29]), indicating that no differencesin user acceptance existed between the 290/260-ms andthe 390/360-ms scenarios, while for the 790/760-msscenario, the users were largely dissatisfied.

One final examined publication was [23], whoserationale is closest to the work presented herein because itevaluates the effect of satellite transmission onaeronautical radio communications, though it does notexamine IP transport. The maximum simulated one-waydelay was equal to 380 ms, close to the one used in the twopreviously mentioned studies. No significant degradationof service was reported by the users of the system.

Summing up this survey of the literature, the bound onthe one-way delay that is used in the ensuing is 375 ms,which is the strictest according to the previous sections.Using this value and the equation for Idd in [13] for valuesof Ta greater than 100 ms, the worst-case Id can becalculated to be equal to 22.033.

d) Delay budget allocation: As stated earlier, delay isthe main point of concern, and because satellitecommunication is involved, it is critical for useracceptance. Once the one-way absolute delay requirementof 375 ms has been set, the end-to-end delay is brokendown to a number of components, and requirements arestated for each one.

The first such component is the propagation delay(time required for the electrical signal to travel along a


transmission medium as a function of the geographicaldistance). Propagation delay for satellite transmission(earth to satellite to earth) is 250 ms for a transmissionfrom Greece to Greece over a transparent (bent-pipetransponder) geostationary satellite.

The second component is jitter, which is the variabilityin the packet delivery time experienced when usingpacketized transmission systems. It is caused by thedifferent routes different packets carrying speechsegments of the same conversation can take through thenetwork or queuing of data, voice, and other voice streamson the same route. This effect depends strongly on theimplementation of specific mechanisms for transport,queuing, or prioritization. To counteract the jitter effect,packets are collected in a buffer at the receive side on bothtransmission directions. This buffer rearranges the timelyorder of the packets, thus ensuring that they are deliveredat regular intervals to the voice decoder, even if theyexperience variation in the time they are under way in thenetwork. If the delivery time of a packet exceeds thelength of the receive buffer, then this packet “comes toolate” with respect to the size of this buffer and isdiscarded. Hence, the speech carried in this packet is lost,which has a negative impact on speech transmissionquality. The jitter buffer induces further delay to the voicestream (the jitter buffer delay) depending on the way inwhich it operates and on network conditions. Because ofthe unknown network characteristics, a delay introducedby the jitter buffer not exceeding 35 ms is targeted whilethe packet loss is bound to a maximum of 0.5%, arequirement established in Section III.C.5.a so that thevalue of Ie is the same as the one in [13], which was usedin the calculations of previous sections.

The third component is the fixed delay introduced bynetworking equipment, such as the RoIP gateway andsatellite modem/router presented in Fig. 3. A realisticrequirement is that this delay will not exceed 50 ms oneach transmission direction and end to end.

e) Delay budget optimization: One last componentthat imposes delays on the end-to-end voice path is themechanism, which compresses linear PCM voice using avoice codec and encapsulates it into IP packets. Once acertain acceptable delay bound has been allocated to eachof the other components presented in the previous section,the voice codec and packetization interval can be selectedso that the end-to-end delay can be kept below 375 ms.

The voice coding algorithm introduces a standarddelay. In addition, a voice frame output by a voice codecrepresents a certain time span of voice data. As far as theG.723.1 codec is concerned, its algorithmic delay is37.5 ms and the codec interval equals 30 ms [24], while forG.729, the numbers are 15 and 10 ms, respectively [25].

Moreover, each RoIP packet carries a payload thatdepends on the voice codec used and the time span thepacket covers (number of voice frames per packet), as wellas overhead, such as the real-time transport protocol (RTP)header (fixed at 12 bytes/packet), the user datagramprotocol header (fixed at 8 bytes/packet), the IP header

(fixed at 20 bytes/packet), and the satellite link layeroverhead (which depends on the network technologyemployed as transport). It is thus usual practice toaccumulate voice data before putting them into a frame fortransmission to reduce the amount of overhead, and this,in turn, increases the end-to-end delay.

The final step is thus to optimize the number of voiceframes per IP packet and the employed voice codec so asto constrain the end-to-end delay below 375 ms.Subtracting the required delay limits on all intermediatelinks of the end-to-end chain from this number, themaximum limit on the sum of the packetization and thecodec algorithmic delays can be set to 40 ms. The G.723.1voice codec will have to be excluded because, even at 1voice frame per packet, the sum of the algorithmic andpacketization delays is 67.5 ms. The only applicable codecis thus G.729 at packetization intervals of 1 and 2 voiceframes per packet, yielding delays of 25 and 35 ms,respectively.

f) R factor value for radio calls using RoIP oversatellite: Up to this point we have presented alladaptations, simplifications, and considerations that allowthe calculation of the R factor for radio calls using RoIPover satellite, enabling the comparison with the leased-linescenario. Ro was calculated in Section III.C.1; Ie wascalculated in Section III.C.5.a for both transmissiondirections, assuming the use of a G.729 low bit-rate codec;and Idd was calculated in Section III.C.5.c for bothtransmission directions. The corresponding R factorvalues can thus be calculated to be equal to 20.254 in thepilot–to–flight controller transmission direction and26.327 in the reverse direction. Comparing these valueswith the respective ones for the leased-line case (SectionIII.C.4), we can conclude that the respective advantagefactors are equal to 31.342 for both transmissiondirections. What remains to be proven is 1) whether therequirements set forth within the earlier sections arefulfilled and 2) whether the users of the proposed systemjustify the expected deterioration of voice quality againstthe advantages of introducing this RoIP over satelliteservice, i.e., to validate that the calculated advantagefactor can be accounted for. The only way to prove thefirst allegation is through field testing, while the secondcan only be validated using subjective testing of theachievable voice quality. The exact methodology andresults of field and subjective tests of a RoIP over satellitesystem, as well as pertinent conclusions, are presented inSections VI–VIII.

IV. BANDWIDTH REQUIREMENT DERIVATION

The earlier analysis has indicated the preferable codecand the allowable packetization intervals to be used forradio calls. Based on this data, the required bit rate, whichis the same for both transmission directions, can becalculated. The use of header compression is mandatory,because satellite bandwidth is too scarce and consequentlytoo costly to be wasted on carrying headers that rarely


change. Using modern techniques for robust headercompression (ROHC) can reduce the overhead in RTPfrom 40 to 4 bytes at most, as shown in [26] (even 2 bytesin some cases, which has been considered highlyoptimistic and was not used during system design).

Taking into consideration the gain from ROHC headercompression and the real-time transport control protocol(RTCP) overhead (fixed at 5% of the bit rate) and omittingthe negligible bandwidth necessary for signaling data, therequired worst-case bit rate for a G.729-encoded voicestream has been calculated equal to 11.8 or 10.1 kbps pertransmission direction depending on whether thepacketization interval is 1 or 2 voice frames per packet.

V. SYSTEM DESIGN AND IMPLEMENTATION DETAILS

The authors, together with an experienceddevelopment team, designed and implemented within theSATWAYS project, cofunded by the European SpaceAgency, a RoIP over satellite communication solution (theSATWAYS system) among remote radio sites and the ATCcenter to address the problems presented in Section I,taking into consideration the outcome of the analysispresented in Sections II–IV. The RoIP gateway wasdeveloped around the inAccess Linux-based RSC-10intelligent remote site controller. The RSC-10 wascomplemented with a custom voice extension interfacemodule and appropriate kernel drivers. This interfacemodule incorporates four radio ports, which can interfaceboth VHF radio and voice recorder equipment employing16-bit PCM digitization, two ports for telephones, a digitalsignal processor (DSP), and an field-programmable gatearray for bus adaptation purposes. Commercially availableDSP software could not fulfill the requirements pertinentto RoIP transport, so custom firmware was developed tooffload the central processing unit from time-critical andprocessing power–intensive tasks, such as voice coding,echo cancellation, telephony signaling generation anddetection, gain adjustment, and mixing. Specializedsoftware to implement the transport of voice between theEthernet port of the RSC-10 and the voice interfaces wasdeveloped, implementing RTP, RTCP, and sessioninitiation protocol for call control with certified QoSwhen executed on a real-time Linux platform. Forinformation regarding the system and softwarearchitecture, consult [27].

The very small aperture terminal IP network was builtaround the SkyWAN system from ND SatCom. SkyWANis a demand assigned multiple access system, offeringextended Diffserv-based QoS features at the IP level(multiple QoS classes with precise trimming of the bit rateavailable for each packet stream instantiation); star, mesh,and mixed network configurations; and redundancy persite and hub location.

The satellite network was designed so as to fulfill therequirements presented in the previous sections. Thesatellite of choice was HellasSat II, whose beams coverthe whole of Greece. The SATWAYS system was required

Fig. 7. Antenna of SATWAYS master and client stations at AreaControl Center of Greek FIR, Hellenicon, Athens, and Patmos.

Fig. 8. Test setup for field testing of SATWAYS solution.

to operate with a satellite bandwidth of 536 kHz on theextended Ku band (uplink frequency: 13 981.146 GHz,downlink frequency: 12 737.146 GHz) offered by ESA.The link budget analysis was performed by HellasSatusing as inputs the carrier data presented earlier, theavailability (target set to 99.9%, to allow some headroomrelated to the requirements of [28]), and the bit error rate(less than 1 in 107, as per [28]). It indicated a necessaryantenna diameter of 1.8 m and a transmit power of 1.7 W.Then, ND SatCom carried out the design of the satellitenetwork around the satellite carrier, which indicated anend-to-end delay of 250 ms and a maximum jitter of 30 mspeak to peak. The satellite modems were configured toapply header compression to RTP packets and to prioritizevoice traffic. The total available bit rate was calculated tobe equal to 481.825 kbps, so taking into consideration thenecessary bit rate per direction calculated in Section IV,about 20 concurrent radio calls can be accommodated.

VI. FIELD TESTING OF THE SATWAYS SOLUTION

As stated in Section III.C.5.f, field testing was judgedas necessary to validate the technical approach.Coexistence of the two ends of the communication pathwas necessary to estimate the performance of the solution,so they have both been deployed to the same location(Area Control Center of the Greek flight informationregion, or FIR, at Hellenicon, Athens, Fig. 7, left). Afterthe commissioning and related testing have beencompleted, a set of field tests was carried out using the testsetup presented in Fig. 8, employing RF signal duplexersto allow both satellite router/modems to modulate theirsignals onto the same intermediate frequency. In this way,the same antenna, transmit amplifier, and low-noise blockcould be used by both ends while they were colocated.

The two RoIP gateways were configured to use theG.729 codec with a packetization interval of 1 voice frameper RTP packet and an adaptive jitter buffer, as indicatedwithin Sections III.C.5.d and III.C.5.e. The satellite


TABLE IIResults of Field Testing of SATWAYS Solution

Trial Min E2E Max E2E Mean E2E Call Tx Tx Packet Rx Packet P2PID Delay (ms) Delay (ms) Delay (ms) Measured Direction Loss (%) Loss (%) jitter (ms)

1 352.3 370.3 360.0 One call R-H 0 0.009 5.11H-R 0 0.003 0.23

2 354.4 370.6 359.1 1st call R-H 0.327 0.347 7.31H-R 0.328 0.351 0.28

2nd call R-H 0.315 0.331 7.21H-R 0.315 0.330 0.31

3 356.8 366.3 362.4 One call R-H 0 0.006 1.49H-R 0 0.003 5.54

4 357.6 367.7 363.7 1st call R-H 0 0.003 1.37H-R 0 0.003 5.41

2nd call R-H 0 0.006 1.38H-R 0 0.006 5.68

5 381.9 412.4 391.3 One call R-H 0 0.006 1.44H-R 0 0.133 17.93

6 385.7 406.7 392.8 1st call R-H 0 0.002 1.42H-R 0 0.272 14.50

2nd call R-H 0 0.020 1.39H-R 0 0.220 14.37

E2E = end to end, Tx = transmitted, Rx = received, P2P = peak to peak.

modem and hub were configured according to Section V.An oscilloscope was then used to measure the end-to-endone-way delay, monitoring a dedicated signal generated atone end and captured at the other. This signal had toremain synchronized with the voice traffic so that it is indirect analogy with the delay experienced by voicesignals. The obvious choice is the PTT signal, which wasgenerated by the radio terminal emulator on one end,conveyed within RTP packets to the other end, andregenerated there. The PTT signal is ideal because it canbe captured with an oscilloscope, yielding extremelyaccurate results.

Two workstations were used to generate Ethernettraffic so as to congest the satellite link to 100%, as well asto create heavy interrupt and system load (greater than70%) on the RoIP gateways throughout the experiments.Such a situation is not likely to happen in normaloperation, but it provides good evidence regarding systemstability and performance under adverse conditions.

Table II depicts the results of the field tests. Allexperiments were repeated 100 times. Trials 1 and 2involved one and two radio calls, respectively, between theemulated remote site and the HCAA headquarters in theabsence of system and network load. Trials 3 and 4 repeattrials 1 and 2, respectively, in the presence of system load.Trials 5 and 6 repeat trials 3 and 4, respectively, in thepresence of both system and network load up to satellitecongestion.

The end-to-end delay was measured as the average of100 on/off cycles of the PTT signal. Its maximum valueremained under 375 ms, including the jitter buffer delay.Only the maximum values measured in trials 5 and 6reached close to 400 ms. This behavior can be attributed tothe increased peak-to-peak jitter because of the networkload driving the jitter buffer to larger sizes. More

specifically, the jitter buffer size is expected to be twicethe maximum encountered jitter in similar testingconditions. Comparing the difference of 30 ms in theend-to-end delay among trials 5 and 6 and the remainingtests with the respective jitter measurements (14.5 ms),this rule of thumb applies. However, these worst case trialsare not representative of typical operating conditions.Furthermore, no dependency on the number of concurrentcalls is demonstrated.

Jitter measurements unveiled a maximumpeak-to-peak jitter of 18 ms. The output jitter of theSATWAYS system has been measured to be less than1 ms, which is considered minimal except when it ishighly loaded with Ethernet traffic. This is normal,because the transmission of voice packets is oftenpreempted by the transmission of data packets via theEthernet. However, this raised jitter figure (in theneighborhood of 8 ms) is masked quite efficiently by thejitter induced by the satellite link (trials 3–6). In contrast,end-to-end jitter is in general low (below 7.5 ms);however, it is unexpectedly increased when the satellitelink gets congested (trials 5 and 6). Finally, the end-to-endpacket loss remained at all times below the 0.5% limitused in Sections III.C.5.a and III.C.5.d. Consequently, theequipment impairment factors that were used for thecalculations are valid. The increased end-to-end packetloss measured in trial 2 can be attributed to the increasedbit error rate of the satellite link. Wind bursts are the mostlikely cause, because this test was carried out duringwinds of intensity up to 8 beauforts. The packet lossmeasured in trials 5 and 6 is increased because the jitterwas also increased due to link congestion, which forcedsome voice packets to drop out of the jitter buffer (thiseffect can more accurately be described as delay spikes).All measurements were carried out with a fully congested


satellite link, demonstrating the proper operation of theQoS features of the satellite modem equipment.

VII. SUBJECTIVE TESTING OF THE SATWAYS SYSTEM

Though the results of the field testing activitypresented in the previous section clearly indicatecompliance to the requirements set forth before the designof the solution, this does not imply that the modificationsto the E-model proposed within this paper apply whenhumans are involved. The only way to collect proof on thematter was to put the SATWAYS system to the test withactual pilots and flight controllers. The Patmos island inthe southeastern Aegean Sea was selected as the targetremote site, and the SATWAYS system was installed there(Fig. 7, right). The operational evaluation lasted 3 monthsduring the summer of 2010, an exceptionally busy periodfor the Greek airspace.

After system commissioning, flight controllers wereinstructed to evaluate the radio call quality through aprocedure called radio check in ICAO terminology. Thefirst step was to contact the pilot of a plane or helicopterflying in the area at a frequency currently used for ATC.Then, the pilot was requested to contact the flightcontroller on the frequency used by the SATWAYSsystem. Afterward, the pilot was requested to grade thecall quality, and the grades from both parties on a scale of1–10, based on their prior experience with radio systems,were recorded by the flight controller on properly designedquestionnaires. The pilot was then instructed to return tothe previously used frequency. The questionnaires and thepostprocessing procedure were designed according to theCARS method devised by the U.S. FAA to test newsystems for radio communications in ATC [29].

Both VFR and instrument flight rules (IFR, where thepilot relies solely on instrumentation to fly the aircraft)flights had to be contacted for testing purposes, becausethis would enlarge the test sample and create morerepresentative results. The geographical and altitudedispersion of the targets, as recorded within the reports, isquite large. IFR flights are by far more demanding interms of conversational interactivity because of smallerseparations for the maintenance of which the flightcontroller is responsible, higher aircraft speeds, andconsequently stricter requirements on the end-to-enddelay. Officially, the involvement of just two controllerswas requested (one for VFR and one for IFR flights), buteventually more than 15 controllers operated the system.This shows their enthusiasm, as well as the usefulness ofthe system.

Table III presents 21 user ratings of the system with anaverage of 8.67 on a scale from 1 to 10 (10 is best)regarding voice quality and an average of 1.44–1.61 on ascale from 1 to 10 (1 is best) regarding the effect of delayon their perception, effectiveness, timing, correctphraseology, and speech rate. The average values indicatean exceptional performance, and the opinion of most usershas been more or less the same, because the deviation of

their grades is small. The actual number of times that theSATWAYS system has been used for operational activitiesis larger compared to the number of the written reports;however, the controllers were rating it in writing onlywhen their workload was reduced and flight safety was notcompromised by their actions.

The user ratings were correlated with logs recorded bythe system. Therefore, all equipment used wassynchronized to a network time protocol server withinHCAA’s internal network, which also provided time to theclocks at controller positions. The results are presented inTable III. The recorded maximum jitter values are slightlyhigher than the larger ones of those recorded withinTable II by 3 ms, which is expected to translate to 6 ms ofadditional end-to-end delay. Even in that case, theend-to-end delay, though it cannot be measured becausethe two stations are set apart, is expected to be below the375-ms limit, taking into consideration the measurementspresented within Table II. The maximum jitter values arealmost constant and equal per transmission direction, abehavior that is different from the one depicted in Table II,yet the experiment is different because the two stations areset apart. The packet loss values remained close to themeasurements of Section VI. Reports 16–18 indicatepacket loss values considerably larger than the 0.5%requirement (a tenfold increase). The reason was traced,with the help of HellasSat, to be interference from anarmy-operated satellite transmitter. However, thesedeviations do not seem to affect the radio call quality asfar as the flight controllers are concerned, becauseexceptional grades were awarded in any case.

Before the evaluation of the SATWAYS system and tofactor out the contribution of insufficient radio coverage touser ratings, HCAA was asked to provide radio coveragemaps to the flight controllers. Fig. 9 illustrates as anexample the radio coverage at an altitude of 2000 ft. Thesame maps were used by the authors to cross-check thereport comments, because some of them were possiblyrelated to radio coverage issues. For instance, in Reports14 and 15, several tests have been carried out at variousaltitudes and distances from Patmos. Almost allcommunications presented no problems, and the mapsindicate good radio coverage from the flight level of 2000up to 20 000 ft. However, the flight controller commentsindicate significant noise when the aircraft was at 12 000ft, 65 NM west of Patmos, and the radio coverage studyindicates that this location may be outside of radiocoverage. Furthermore, in Report 17, controller commentsindicate that when the target was at a distance of 15 NMnorth of Kalymnos and a flight level of 6500 ft,communication was problematic. The radio coveragestudy indicates a shadow in that direction, induced by theisland of Lipsi.

Once the radio coverage issues were identified by thecontrollers and factored out from their evaluation, thisparticularly selective target group started using theSATWAYS system to control flights like any other system.Reports 15, 17, and 18, for example, indicate that voice


TABLE IIIResults of Subjective Testing of SATWAYS Solution

Max Max Average AverageJitter Jitter Packet Packet

Trial Flight Type Patmos HCAA Loss LossID Level of Flight Time Date Q1 Q2 Q3 Q4 Q5 Q6 (ms) (ms) Patmos (%) HCAA (%)

1 32000 IFR 9:10 20/7/2010 9 2 2 2 2 2 20.625 20.375 0.2561 0.00592 32000 IFR 9:20 20/7/2010 9 2 2 1 1 1 20.625 20.375 0.2561 0.00593 36000 IFR 9:30 20/7/2010 8 3 4 3 3 2 20.625 20.375 0.2561 0.00594 20000 IFR 9:40 20/7/2010 8 2 2 2 2 2 20.625 20.375 0.2561 0.00595 20000 IFR 9:40 20/7/2010 8 2 2 2 2 2 20.625 20.375 0.2561 0.00596 24000-9000 IFR 21:10 21/7/2010 8 2 2 2 2 2 20.625 20.625 0.6340 0.53587 25000-8500 IFR 21:20 21/7/2010 8 2 2 2 2 2 20.625 20.625 0.6340 0.53588 16500 VFR 11:00 25/7/2010 9 2 2 2 2 2 20.625 20.625 0.6519 0.01019 6900 VFR 11:10 25/7/2010 9 2 2 2 2 2 20.625 20.625 0.6519 0.0101

10 4800 VFR 6:25 26/7/2010 9 1 1 1 1 1 20.625 21.25 0.4334 1.143011 4500 VFR 6:30 26/7/2010 9 1 1 1 1 1 20.625 21.25 0.4334 1.143012 12500 VFR 5:00 1/8/2010 9 1 1 1 1 1 20.75 22.375 0.4395 0.738113 9000 VFR 5:20 1/8/2010 9 1 1 1 1 1 20.75 22.375 0.4395 0.738114 5500-12000 VFR 5:30 1/8/2010 8 1 1 1 1 1 20.75 22.375 0.4395 0.738115 9000 VFR 5:20 1/8/2010 9 1 1 1 1 1 20.75 22.375 0.4395 0.738116 10400 VFR 9:00 1/8/2010 9 1 1 1 1 1 20.625 18.375 0 3.504317 6500 VFR 8:10 13/8/2010 9 1 1 1 1 1 16.5 79.625 5.4262 5.756418 gnd-2000-gnd VFR 8:20 13/8/2010 9 1 1 1 1 1 16.5 79.625 5.4262 5.756419 13500 VFR 10:50 30/8/2010 9 1 1 1 1 1 N/A 24.25 N/A 0.002620 11500 VFR 10:50 30/8/2010 9 1 1 1 1 1 N/A 24.25 N/A 0.002621 9500 VFR 10:55 30/8/2010 9 1 1 1 1 1 N/A 24.25 N/A 0.0026

Average: 8.71 1.48 1.52 1.43 1.43 1.38Standard deviation: 0.46 0.6 0.75 0.6 0.6 0.5

Q1: How do you rate the quality of the voice during this radio call?Q2: To what extent did the delays interfere with the effectiveness of your communications during this radio call?Q3: To what extent did the delays interfere with the timing of your control instructions during this radio call?Q4: To what extent did the delays interfere with your using correct phraseology during this radio call?Q5: To what extent did the delays interfere with your speech quality (or clarity) during this radio call?Q6: To what extent did the delays increase your speech rate during this radio call?N/A = not applicable.

Fig. 9. Patmos VHF transceiver radio coverage at altitude of 2000 ft.

communication between the flight controller and the pilotwas ongoing for a prolonged period, during which theaircraft has traveled tenths of miles within the Greekairspace.

Postprocessed data indicate a system that operates asexpected and validate the E-model modificationspresented herein. It is remarkable to mention that radiotransceivers operating over landlines often present scoresby far lower than the ones achieved by the SATWAYS

system, mostly because of inadequate performance of thecommunication infrastructure because of a variety ofimpairments injected along the route of the leased line.

VIII. CONCLUSIONS AND FUTURE WORK

The work presented herein reaches a series ofimportant conclusions. First, the findings of previousstudies that delays in the region of 375 to 390 ms do notaffect the operational capabilities of ATC are confirmed.However, while previous studies have been carried out insimulation environments, the results presented herein weregathered using pilots and flight controllers in operationalscenarios, indicating that the use of systems introducingsuch delays is possible in the most demanding situations.In addition, the use of low bit-rate voice codecs does notseem to adversely affect the call quality.

Moreover, it is proved that VoIP technology can becombined with satellite transmission in ATC applications,provided that the systems used to implement this approachhave been meticulously designed to meet the end-to-enddelay requirements and to certify the QoS they offer understrenuous conditions.

Furthermore, it is proved that the proposedmodifications to the E-model can be related to the


experience of the user. Consequently, this modifiedE-model can be used as a radio network planning tool,because once the requirements that were drawn from itwere fulfilled, the users of the resulting service accepted itas sufficient in terms of call quality.

Finally, and most importantly, concrete evidence hasbeen presented that both pilots and flight controllers haveincreased margins of resilience to the degradation of thevoice quality introduced by increased end-to-end delay andpacket loss, because their primary concern is to reliablyconvey their messages. More specifically, an advantagefactor (A) equal to 31.342 for both transmission directionsas presented in Section III.C.5.f can be accountedfor so that the radio calls through a RoIP over satellitesystem present equivalent call quality when comparedto traditional systems based on leased lines. This value,however large it may look for applications targeting thepublic, such as telephony, is proved to apply; moreover,it seems to have a small contribution to user ratings. Themost probable cause is that, because of the half-duplexnature of radio communications, the effect of delay onthe user perception of call quality is largely reduced whencompared to telephone conversations, so the reductionof the R factor is largely overestimated by the E-model.

Future research areas were also identified through thework presented herein. First, it is judged as necessary tooperationally evaluate the effect of delay on call qualityacross its whole expected range and not just using specificvalues. This aspect is considered of exceptionalimportance in view of the adoption of both VoIP andsatellite transmission technologies in ATC, because suchdata, together with the E-model modifications presentedherein, could be used to engineer an invaluabletransmission planning tool. At the moment, the adoptionof VoIP is being considered by all standardization bodiesrelated to ATC. ICAO is addressing the issue within Doc9896 (Draft Manual for the ATN Using Internet ProtocolSuite Standards and Protocols), and Eurocontrol isfocused on it within EuroCAE ED-136–ED-138. Finally,SESAR, which is designing the next-generation ATMnetwork for Europe, has readily identified the movetoward the use of VoIP and is recommending it over oldertechnologies.

Another particularly interesting domain is the study ofthe effect of multiple satellite access procedures throughthe deployment of the SATWAYS system to more pointsso as to optimize the joint use of the satellite resource andexplore issues related to call blocking and serviceavailability, together with novel call control strategies.

Moreover, and because of the safety-critical nature ofthe targeted application, the effect of interference on thesatellite link will have to be evaluated. Such an event has,as already mentioned, been experienced during fieldtesting. The study of intentional interference or satellitejamming is by nature challenging, and the first step in thatdirection would be an analysis of the sensitivity against aspecific type of interference, which at the moment canonly be guessed (e.g., clean carrier transmission,

cross-polar from orthogonal polarization network, andintermodulation).

One last research area is the use of voice codecs withvoice activity detection so as to further reduce the necessarybandwidth. The adoption of such an approach maylead to syllable clipping, which in the case of aeronauticalcommunications, may have detrimental effectson flight safety. Therefore, extensive testing has to becarried out to carefully quantify its impact on call quality.

As a final word, the adoption of the SATWAYS systemby its users has been enthusiastic, indicating that the roadtoward an all-IP ATC future is wide open. This issummarized in the comments of Report 18, as recorded bythe responsible flight controller:

“It was a flight towards the heliport of the Patmosisland. The flight was controlled from its departure fromthe heliport of the Mykonos island until its landing to theheliport of the Patmos island. At a distance of 15 nauticalmiles east of RIPLI [the code name of a region of the Greekairspace], the pilot and the controller were reporting a 5/5communication. At a distance of 20 nautical miles west ofPatmos and at an altitude of 1500 feet, the pilot and thecontroller were reporting a 5/5 communication. Beginningthe descent towards the heliport, both the reception andtransmission were exceptional. On the ground, thecommunication was exceptional and the flight plant wasclosed this way. The pilot reported his surprise to mebecause he has never had any radio coverage in thisparticular region in the past. I judge that the attempt toconvey radio communications over satellite is successfuland covers the heliport with voice comms to ground level.”

ACKNOWLEDGMENT

We thank Marinos Kardaris, director generalof Air Navigation, and Kostas Simeakis, chief of thetelecommunications division at HCAA, for their concen-trated know-how, professionalism, and support during thedesign, implementation, and deployment of the solution.

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Spyros Apostolacos received a diploma in electrical, electronic, and computerengineering from the National Technical University of Athens (NTUA), where he isfinalizing his Ph.D. thesis. From 1998 to 2000, he worked as a technical consultant toLucent Technologies, the Netherlands, focusing on the design and implementation ofwireless and high-speed wireline transmission systems. Since November 2000, he hasbeen working with inAccess as technical manager of the Communication Systemsgroup.


Apostolos Meliones received a diploma with honors in computer engineering andinformatics from the University of Patras, in 1994, and a Ph.D. degree in distributedsystems and telecommunications from NTUA in 1999. From 1994 to 1999, he held aresearch position at NTUA. From 2001 until 2005, he was technical manager ofnetwork embedded systems at Ellemedia Technologies, a Lucent product developmentpartner. In 2006, he joined inAccess as research and development manager. He was theinitiator, project manager, or member of the management board of several EuropeanUnion and national research activities. Since 2009, he has been a faculty member of theDepartment of Digital Systems at the University of Piraeus. He is the author of 70research papers in the areas of distributed computing, telecommunication andembedded systems, and pervasive computing. He is the developer of the Linux ATMdevice driver for the PowerQUICC-II family of communication processors, which hasbeen widely used worldwide.

Stefano Badessi was born in Rome, Italy, in October 1957. He is presently employed aspayload data ground segment operations manager in the Directorate of EarthObservation Programmes of the European Space Agency (ESRIN, Frascati, Italy). Hepreviously worked at ESA as antenna engineer and then as applications senior engineerin the Telecommunications Directorate.

George Stassinopoulos born in Athens, Greece, in 1951. He received a degree inelectrical engineering from ETH Zurich in 1974 and a Ph.D. degree in automaticcontrol from the Department of Computing and Control, Imperial College, London, in1977. He is a professor at the Division of Computer Science, National TechnicalUniversity of Athens. He has industrial experience in the design and manufacture ofmicroprocessor based industrial controllers in the cement industry, as well as incomputer networking and industrial process control. His current research interests are inthe fields of communication networks, local area networks, wireless broadbandnetworks, ad hoc networks, web services, and security. He leads the embedded systemslabs, with particular focus on communication and media applications. He hasparticipated in many national research programs dealing with communication networksand services at national, European, and international levels (EU RACE, frameworkprograms). He has more than 90 publications in the preceding areas.


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