WIRELESS NETWORKS, CHALMERS 2011 1

Voice over IP - WLAN, 3G and LTE issuesBaran Kiziltan, Majid Khan and Francesco M. Velotti

Abstract—The aim of this paper is to give a basicintroduction on VoIP- WLAN, 3G and LTE issues, andthoroughly describe QoS, problems in different wirelessnetwork techniques and scenarios to get the best qualityin real-time. The attention is focused on the Quality ofService in three different wireless networks, so, there willbe proposed some solutions to improve these issues.

Index Terms—WLAN, 3G, LTE, Voice over IP

I. INTRODUCTION

VOICE over Internet Protocol (VoIP), also knownas IP telephony or Internet telephony, is a set

of protocols to transport voice traffic over IP-basedpacket-switched networks with acceptable quality ofservice (QoS) and reasonable cost.

Wireless Local Area Networks (WLANs) havebecome a part of everyday technology. This has nowbeen deployed around the world. Voice over WLAN(VoWLAN) has been emerging as an infrastructure toprovide low-cost wireless voice services. However, sincethe performance characteristics of wireless networks aremuch worse than wire line counterparts, and the IEEE802.11-based WLAN was not originally designed tosupport delay-sensitive voice traffic.

Third generation (3G) packet switchedUMTS/WCDMA networks with High Speed DownlinkPacket Access (HSPDA) is being installed worldwide.With the introduction of HSDPA in 3G networks, packedswitched wireless systems will allow dynamic resourcesharing therefore more efficient use of bandwidth andimproved network efficiency will be possible. Sincevoice applications are real-time, they are intolerantof lengthy delays, packet losses and jitter (delayvariation). All these problems degrade the quality of thevoice transmitted. These QoS issues over 3G wirelessnetworks will be assessed with respect to network load,packet switching, buffer length and packet segmentationunder certain protocols, such as Adaptive Multi-RateSpeech Codec, HC (Header Compression), RTP (RealTime Protocol) and RLC (Radio Link Control Layer).

The challenges for achieving this include typicalVoIP related QoS (Quality of Service) problems, suchas delay, delay variation (i.e. jitter), packet loss andadditional overhead brought by the VoIP protocol stack[1].

The QoS problems for VoIP over LTE will beanalyzed by comparing physical layer techniques andtry to obtain the best one in terms of VoIP quality, andcompared with some simulation or data.When talking about quality on WLAN it is useful todistinguish between scenarios. If your WLAN accesspoint keeps crashing then you could say QoS for you ispoor. To get best access point (AP), one can describethe IEEE family 802.11 to find what technique shouldbe used.

The paper is organized as follow. Section II we focusour attention on issues that afflict VoIP in WLANnetworks. Section III we consider QoS over 3G net-works. Section IV we analyze QoS and physical layertechniques to improve VoIP quality over LTE networks.Finally, Section V we present our conclusion and possi-ble future implementations.

II. VOIP OVER WLAN

The Wireless Local Area Network (WLAN) becomespopular to support high-data-rate Internet access forusers in proximity of an access point (AP). The mainadvantages of WLAN is simplicity, flexibility andcost effectiveness. VoWLAN applications use theinfrastructure based on WLANs. There is a varietyof standards defined in the IEEE 802.1 [2]. The mostdeployed standard is 802.11b, whereas 802.11g isreceiving acceptance because of the high rate andbackward compatibility with 802.11b.

WLANs are only specified at the physical layer andpart of the data link layer. Reason why security and QoSon WLAN is hard is because of all IP routing, sessioncontrol etc is outside the scope of WLAN, since both se-curity and QoS are clearly needed end to end then higherlayer solution needs to interface o the WLAN capabilitiesVoIP is real time applications and WLAN is not basically

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Fig. 1. VoIP over WLAN [3]

made for real time application because the QoS which isbig issue with WLAN is important and main portion ofVoIP applications. Challenges VoWLAN: there are manychallenge in VoWLAN like Quality of Services (QoS),security etc QoS in VoWLAN consist of these three thingwhich is discussed below.

A. Packet Loss

The total number of packet transmit over the networkis not receive to the end point or destination, so it meansthe some data or packet loos or not received by the desti-nation. There are two main sources of packet losses:oneis network packet losses, mainly due to network conges-tion (router buffer overflow), link failures and rerouting,transmission errors, etc; and the other is discarded packetlosses for packets experienced excessive delay.

B. Delay

The time taken by a packet to reach from a sourceto destination, delay can be occurred from differentsources like delay at source, delay at receiver, delayin network. Delay at source and receiver is due tocoding like changing analog to digital and digital toanalog and packetization, while network delay is dueto transmission, queuing and propagation.

C. Jitter

The variation of time between packet transmit fromsource to reach destination, means one packet reach in100 ms and one reach in 125 ms to handle this problemjet-buffer is used at receiving end and it has two typestatic jet-buffer which hardware base and dynamic basewhich is software base and can be handle by administra-tor .but should take care about jet-buffer because sometime it is also becoming reason for delay like memoryover-flows etc. The following are the some measurementand recommendation of ITU-T G.114 for a VoIP call forthe three attribute which is define in tab. I [5]. “Factorssuch as packet delay, jitter, packet loss and networklatency can noticeably affect the quality of UDP- based

TABLE IQOS

Packet delay Packet loss Jitter≤ 150 ms ≤ 1% ≤ 25 ms

services such as VoIP and video streaming. Contrary toTCP-based services such as HTTP, SMTP, etc, a steadystream of data packets is crucial for VoIP connections,where even slight connectivity problems can cause noiseor echo”. “Quality of any service depends on the trafficflow as well as the network of terminating partners.Following are some issues should be considered toprovide better-quality service. Number of calls managedsimultaneously by the network The alternate way totransfer the call to it desired destination in case of anyfault/failure occurred in the network CODECs for codingand encoding purposes. Overall setup of the network”[6].

D. Original IEEE 802.11 MAC layer

The original IEEE802.11 has no idea QoS especiallyfor voice data application have no sensitivity about Delayjitter. The basic MAC layer use distributed coordina-tion function (DCF) and Point coordination function(PCF ) to share medium with station both have severallimitation [2]. DCF relies on CSMA/CA and optional802.11 RTS/CTS to share the medium between station.The problem in DCF is that if many station want tocommunicate at the same time there is always a collisionoccur and it is based on collision avoidance means it hasto wait the medium to be free which produce delay andif collision occur it is waste of the bandwidth and makecommunication slow. some problem in DCF:

• there is no QoS guaranty and priority between datatraffic like voice and data;

• if a station sense medium and it is free andget medium to communicate no other can’tcommunicate until it didn’t let free the mediumif a station has slow bit rate it will capture themedium for along time.

PCF is the other coordination which is define by ba-sic IEEE802.11. It is optional. PCF is used only ininfrastructure mod in which all station are connectedby one center object called Access Point (AP). PCFdefine two frame Contention Free Period (CFP) andContention Period (CP). In the CP DCP is used. Togive the right of communicate over the medium the CFPsend Contention-Free-Poll (CF-Poll) to station at timeone packet each. The AP is coordinator. PCF has a little

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bit QoS management but have no idea of the differentclass of traffic.

E. IEEE 802.11e

This standard define enhancement in the originalIEEE802.11 Mac layer DCF and ECF with new coordi-nation function Hybrid Coordination Function (HCF). Itproposed priority and class based traffic means the voiceand multimedia application data class will have highpriority during transmission compare with other data likeemail data class in a shared wireless medium etc. Thereare two method to access the channel to communicatelike original IEEE802.11 MAC. HCF Controlled Chan-nel Access (HCCA) and Enhanced Distributed ChannelAccess (EDCA)[6]. With the EDCA the data whichhave high priority will have have chance to send earlythen the low priority data and station having EDCAimplemented will have to wait less to send data. It workmostly like PCF. In PCF scenario the interval betweento beacon frame is divide into two period CFP and anCP, the HCCA is allowing the CFP to initiated almostany time during CP. This kind of CFP is called AccessPhase (CAP) in 802.11e. The AP will initiate CAP anytime which it want and can receive frame from othercontention-free manner. The CAC is a method whichwill decide whether a new connection will be allowto established or not, it will be decide on the basisof capacity of WLAN means if the new connection isallowed what will be MOS or quality of over all callwhich would be specified. So the CAC will maintain theover all quality of Voice of VoWLAN. For infrastructuremode of VoWLAN the CAC can be implemented in AP.Codec is used to convert voice signal to digitally encodedversion compress it on the sender end and then reversethe processes on the receiver end. These codec arestandardized by International Telecommunication (ITU-T). There are many codec technique which is used inVoIP for Encoding and Decoding. Some coding and itdifferent result are mentioned in below table fig.2 whichhas been calculated with different IEEE 802.11 standardwith sample period 20, and voice activity detection active[8].

The above data int the table is calculated by [9]Connect 802 VoIP Bandwidth Provisioning Calculator: from the table we got different result from differentcode which different bit rate per kbps and with havedifferent WLAN IEEE802.11 like a,b,g and got differentMOS and found how much simultaneously connection orcalls can be established at time on per AP. Among weobserve that Codec G.711 have high MOS rate and withreasonable simultaneously calls at time. But it should

Fig. 2. VoIP over WLAN

be care about the bandwidth of connection is ok forrequirement of codec selected like G.711 require at least128 bit for both way communication. MOS is Inter-national Telecommunications Union TelecommunicationStandardization sector (ITU-T) approved which givesa numerical indication of the perceived quality of themedia received after being transmitted and eventuallycompressed using codec. The WLAN are working onradio wave which are open which can eavesdrops andsome one can manage to use it illegally like crack thesecret key.

F. IEEE 802.11iThe IEEE802.11e enhance the security issue of orig-

inal WLAN and put forward the WAP2, it using Ad-vanced Encryption Standard (AES) block cipher. TheWEP and WAP were using RC4 stream cipher. TheIEEE802.11e replace the issue of Authentication andprivacy issue with more detail and security adjustment[9]. Different VLAN can be used to separate Voice trafficand data traffic: it will solve the space problem andvoice device can be protected from external network.Separate VLAN will have private addresses which willhide phone device from directly connected to publicnetwork; QoS trust boundary extension to voice devices-QoS trust boundaries can be extended to voice deviceswithout extending these trust boundaries and, in turn,QoS features to PCs and other data devices; protectionfrom malicious network attacks-Subnet access control,can provide protection for voice devices from maliciousinternal and external network attacks such as worms,denial of service (DoS) attacks, and attempts by datadevices to gain access to priority queues; ease of man-agement and configuration-Separate VLANs for voiceand data devices at the access layer provide ease ofmanagement and simplified QoS configuration.

III. VOIP OVER 3GTraditionally, real-time services (e.g. voice) are

transported over dedicated channels because of their

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Fig. 3. Transport of speech in IP

delay sensitivity while data is transported over sharedchannels because of its transmitted in short, unevenspurts. In order to carry voice on IP networks,appropriate protocols must be used. The main protocolsare Real Time Protocol (RTP), User Datagram Protocol(UDP) and Internet Protocol (IP) [11]. In Fig. 1, thevoice frames are generated in the application layer,encoded and encapsulated within payload of an RTPSDU. The RTP PDU is encapsulated into an UDP SDU,which is delivered to the IP layer.

Adaptive Multi-Rate Speech Codec (AMR) is a codecwith 8 narrow-band speech encoding modes with bitrates between 4.75 and 12.2 kbps. If the data rateis 12.2 kbps, the AMR codec generates packets of244 bits which represent voice frames of 20 ms [12].Since the AMR codec encodes and decodes digitalspeech data with an optimum power and bandwidthconsumption, the Internet Engineering Task Force(IETF) has approved the RTP payload format for AMR.

Real Time Protocol (RTP) is an end to end transportprotocol, used to transport multimedia traffic in IPnetworks, supporting unicast and multicast traffic. In thecase of VoIP service, it is implemented together withUDP/IP [11]. Since RTP does not provide any reliabilitymechanisms and other layers should be implemented.AMR and RTP the main performance parameters forVoIP quality that are described earlier, can be measuredby the RTP protocol. RTPAMR and RTP The main performance parameters forVoIP quality that are described earlier, can be measuredby the RTP protocol.

User Datagram Protocol (UDP) as a transport layerprotocol for VoIP over Internet Protocol (IP), UDP isused to avoid any retransmission delays. On the otherhand, it provides no reliability on datagram delivery.The UDP header size is standardized in 8 bytes and 20

Fig. 4. FER for several loads and channel error for Simulation 1[11]

bytes for IPv4 or 40 bytes for IPv6.

Header Compression (HC) in 3G networks it isimportant to use bandwidth efficiently. On the otherhand, large headers of the protocols used when voicedata is sent over the wireless network where a high biterror rate (BER) due to fading and mobility is present.Robust Header Compression (ROHC) protocol has beendeveloped for this problem. The effective compressionmakes use of the fact that majority of the fields inthe combined IP, UDP and RTP header either remainconstant or introduce constant change throughout asession.

A. QoS Analysis

One main parameter for assessing packet loss is FER(Frame Error Rate). Although packet loss is undesiredsome loss can be tolerated since error-concealmenttechniques can be used. Buffer length can also causepacket loss due to discarding of delayed packets. Onthe other hand buffer length also may also increase thedelay where for acceptable conversational quality, themaximum end-to-end delay should be around 250-300ms [13]. Therefore buffer length takes an important roleshort buffering time will risk buffer underflows causingjitter, and long buffering time causes long delay andbuffer overflows. Too short buffering time may alsocause increased packet loss due to loss of segmentedpackets. Simulation with parameters specified for 2 dif-ferent simulations can be seen in tab. II.

From the first simulation it can be seen that fordifferent error probabilities, ranging from 1% to 10%,packet loss is directly related to the load on the wirelessnetwork. With the increase number of network users,applied packet switching technique is not feasible.Therefore it can be said that delay and delay jittermainly depends on both Round Robin switching

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TABLE IISIMULATION PARAMETERS

Parameter Value (Simulation 1) Value (Simulation 2)Simulation runs 10000 6min30s

of speechLoad Variable One user

Channel error Variable Variableprobability

AMR source 12.2kbps 12.2kbps

data rateAMR voice 20ms 20ms

frame durationCall duration 120s 390s

Silence Voice on/off Silence Descriptor (SID)periods (mean duration 3s) (160 ms intervals)

AMR voice 244bits 244bits

packet payloadsize

Protocol Stack RTP + UDP + IPv4 RTP + UDP + IPv6

size = 40 byte = 60 byte

Header Robust HC Robust HCCompression (HC)

RLC mode Unacknowledged UnacknowledgedMode Mode

Maximum number 3 Noneof MAC-hs

retransmissionsNumber of 4 None

MAC-hs H- ARQparallel processesPacket scheduling Round-Robin None

algorithmDelay budget 100ms Predefined jitter

buffer (FIFO algorithm)

Fig. 5. Mean packet delay for several loads and channel forSimulation 1[11]

technique and Hybrid-ARQ mechanism where themain features of MAC-hs (Medium Access Control-high speed) protocol of HSDPA are retransmissionof erroneous packets which is handled by H-ARQand sequential delivery of the packets to the upperlayer [14]. This reasoning can also be seen in the PDFof delay jitter for 5 fixed users on the network in figure 7.

In simulation 2, a predetermined buffer is imple-mented; therefore average network delay is constantfor different error probabilities. On the other hand as

Fig. 6. PDF of the mean packet delay jitter for several channel errorfor Simulation 1[11]

Fig. 7. Simulation Results for Simulation 2[1]

packet loss ratio increases on the wireless channel, totalpacket loss rate increases. The reason for occurrence oferroneous packets in loss-free simulation is due to thepacket segmentation at RLC (Radio Link Control Layer),where packets larger than one TTI (Time Transfer Inter-val) are segmented over several TTIs, introducing longertransmission delays and packet drops.

IV. VOIP OVER LTE

There are two important conditions must be met toensure an adequate VoIP quality:

1) delay from sender to receiver must be as low aspossible;

2) packet loss must be between 1% to 3%.

So, in LTE, end-to-end Quality of Service is basedon two parameters that formalize these two conditions.First, Layer 2 Packet Delay Budget is specified forevery connection and for every User Equipment (UE).Second, Layer 2 Packet Loss Ratio is defined in orderto guarantee the above specification. Hence, if a VoIPconnection has a L2PDB of 100 ms and a L2PLRof 2% it mens that the QoS level for a subscriber issatisfactory. [16]

In wireless networks, like LTE, the principal causeof issues is the path between the radio base-station and

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the UE. In fact, there are many new physical layertechniques made to try to avoid the bit errors and thedelay, for example: Hybrid automatic repeat request(HARQ) or advanced channel coding. Given that LTE isstrongly dependent on HARQ, reducing the bit errors,the delay over the connection link will be reduced aswell. [16]

LTE Hybrid ARQ is a physical layer technique toincrease robustness against transmission errors, and toincrease capacity. It is part of the MAC layer but thesoft-combining operation is handled by the physicallayer. In this technique, the erroneously received packetis stored in a buffer memory and later combinedwith the retransmission to obtain a single, combinedpacket which is more reliable than its constituents.Decoding of the error-correcting code operates an thecombined signal. If the decoding fails, a retransmissionis requested. [17]

There are four kind of HARQ schemes, the first one iscalled Type I HARQ and it is based on the use of CyclicRedundancy Check (CRC), the second one is Type I CCHARQ because it is the same of the first one, but it uses aChase combining technique, the third one is Type II FullIR (Incremental Redundancy) and it gradually decreasescoding rate in each transmission by sending additionalredundancy bits [18] and the last one is Type III PartialIR and as the previous, it gradually decreases codingrate by sending additional redundancy bits, but each bitmaintains self-decodability in each retransmission [18].

Type I CC HARQ is a scheme that, when the receiverfinds an error, it discards erroneous packets and sendsa retransmission request to the transmitter. The entirepacket is retransmitted. The packets are combined basedon either the weighted SNR?s of individual bits, inwhich case the technique is termed Chase combining[19].

Type II Full IR is a scheme, where retransmissionrequests consist only of parity bits. The receivercombines additional parity bits from retransmission withbits of the first transmission resulting in lower rates,before FEC decoding is attempted [20].

Type III Partial IR is a schemes, in which individuallytransmitted packets are self-decodable and each packetdiffers in coded bits from the previous transmission.In Type III ARQ, packets are only combined afterdecoding has been attempted on the individual packet[21].

Fig. 8. Packet error ratio in function of SNR for a modulation64QAM with gray coding 3

4 coding rate. [18]

Given that, we analyzed simulation results of [18]showed in 8 and we can say that to achieve an acceptablequality of service, based on our previous parameters, thebest choice it will be HARQ Type III Partial IR.

V. CONCLUSION

In this paper we focused on QoS issues of VoIP overWLAN, 3G and LTE and tried to analyze these issuesby comparing different studies and proposing new ideasfor future work. As a result, conclusion for this papercan be discussed in 3 parts.VoIP over WLAN is real time application and verysensitive to delay, packet loss and jitter but on the otherhand security is also an issue. This is because WLAN ismainly on physical and MAC layer where the securityis handled in the upper layer. However this kind ofproblem can be overcome if IEEE802.11e standardis implemented where “Call Admission Control” willmonitor the voice quality. In addition, by adding G.711coding technique, high quality multiple simultaneouscalls will be possible. On the other hand for more secureVoIP applications IEEE802.11i should be implementedand appropriate coding technique such as G.711 shouldbe simulated for future work.

VoIP over 3G, end-to-end QoS analysis of two similarsimulations under same protocols shows that current 3Gnetworks offer an adequate level of quality for VoIPservices. However, to improve this some further analysiscan be carried out.

• As the number of user increases packet switching inHSPDA becomes more important. Therefore morecapable Expo-Linear packet switching techniquecan be simulated. This technique calculates the userpriority not only based on ranking users accordingto their instantaneous channel quality, relative to

WIRELESS NETWORKS, CHALMERS 2011 7

their own average channel conditions but also thedelay bound. Therefore, it is able to meet thedifferent QoS requirements of real time users [15].

• MAC-hs protocol enables retransmissions whichcauses decrease on QoS. On the other hand intro-ducing predetermined TTIs also causes segmenta-tion problems related to RLC creating delay andpacket loss. Therefore an adaptive TTI and buffer-ing should be simulated for future work. Onlythen MAC-hs protocols retransmission can be fullyeffective since RLC does not guarantee delivery.

• Current Packet Loss Concealment techniques areeffective only for small numbers of consecutive lostpackets, for example a total of 20-30 millisecondsof speech, and for low packet loss rates. Thereforea further study on intelligent PLC where a learningtechnique can overcome packet loss issues.

VoIP over LTE, QoS analysis is mainly based on variantsof H-ARQ to improve Eb/No over a low packet errorrate which is usually 10−3 for voice and 10−6 fordata transmissions. The trade-offs in this assessmentwas between memory usage and SNR. Standard H-ARQneeds almost no memory but provides very little SNRimprovement on the other hand Type-II Full IncrementalRedundancy requires high memory but provides morethan 10 dB improvement compared to the standard H-ARQ. Therefore Type-III Partial IR where the retransmit-ted packet can be chase combined with previous packetsto increase the diversity gain, is the main candidate forfuture work.

REFERENCES

[1] Renaud Cuny, Ari Lakaniemi, VoIP in 3G Networks: An End-to-End Quality of Service Analysis, Nokia Research Center.

[2] INTERNATIONAL JOURNAL OF COMMUNICATION SYS-TEMS Int. J. Commun. Syst. 2006; 19:491?508.

[3] http://uwanted.blogspot.com/2006/09/wireless-voip.html[4] Recommendation of ITU-T G.114[5] 1998 - 2011 Paessler AG.[6] www.advancedvoip.com[7] IInt. J. Commun. Syst. 2006; 19:491?508 Published on-

line in Wiley InterScience (www.interscience.wiley.com). DOI:10.1002/dac.801

[8] http://www.ozvoip.com/voip-codecs/[9] http://www.connect802.com/voipbandwidth.php

[10] Voice over WLAN Campus Test Architecture Cisco[11] Leonardo Ramon N. Sousa, Marcone L. Carvalho, Emanuel B.

Rodrigues, Leonardo Sampaio and Francisco R. P. Cavalcanti,Quality of Service Evaluation of VoIP over HSDPA, WirelessTelecommunications Research Group - GTEL, Department ofTeleinformatics Engineering - DETI, Federal University ofCeara, 2006.

[12] 3GPP, Mandatory speech codec speech processing func-tions; amr speech codec; error concealment of lost frames,3rd GenerationPartnership Project, 1999. [Online]. Available:http://www.3gpp.org

[13] IETF Differentiated Services (DiffServ) Working Group,http://www.ieft.org/html.charters/diffserv-charter.html.

[14] Robert Bestak, “Performance Analysis of MAC-hs Protocol”,Czech Technical University in Prague, Department of Telecom-munications Engineering, 2005.

[15] Matthias Malkowski, Andreas Kemper, Xiaohua Wang, “Perfor-mance of Scheduling Algorithms for HSDPA”, CommunicationNetworks, RWTH Aachen University.

[16] Capacity Enhancement of VoIP over LTE by Stochastic Adap-tive Modulation and Coding, K.Homayounfar and B. Rohani,10-10 Cendex Center , Singapore.

[17] E. Dahlman. 3G Evolution: HSPA and LTE for Mobile Broad-band. Academic Press, 2008.

[18] Kian Chung Beh, Angela Doufexi, Simon Armour, “PER-FORMANCE EVALUATION OF HYBRID ARQ SCHEMESOF 3GPP LTE OFDMA SYSTEM”, The 18th Annual IEEEInternational Symposium on Personal, Indoor and Mobile RadioCommunications (PIMRC?07).

[19] D. Chase, “Code combining; A maximum likelihood decodingapproach for combining an arbitrary number of noisy packets”,IEEE Trans. Commun., vol. 33, pp. 385 to 393, May 1985.

[20] S. Kallel, “Analysis of Type II Hybrid ARQ Schemes with codecombining”, IEEE Trans. on Commun., vol. 38, No. 8, Aug.1990.

[21] Kingsley Oteng-Amoako, Jinhong Yuan, Saeid Nooshabadi,“Selective Hybrid-ARQ turbo schemes with various Combiningmethods in Fading Channels”, Dept. of Electrical Eng. andTelecomm, University of NSW, Sydney 2052, Australia.

Group-13 Voice over IP - WLAN, 3G and LTE issues

Question:

What are the Quality of Service issues for voice communications over different wireless technologies?

Answer:

- Delay

- Packet Loss

- Jitter (Delay Variation)