Voice over IP - WLAN, 3G and LTE issues

WIRELESS NETWORKS, CHALMERS 2011

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Voice over IP - WLAN, 3G and LTE issues Baran Kiziltan, Majid Khan and Francesco M. Velotti [email protected], [email protected] and [email protected]

Abstract—The aim of this paper is to give a basic introduction on VoIP- WLAN, 3G and LTE issues, and thoroughly describe QoS, problems in different wireless network techniques and scenarios to get the best quality in real-time. The attention is focused on the Quality of Service in three different wireless networks, so, there will be proposed some solutions to improve these issues. Index Terms—WLAN, 3G, LTE, Voice over IP

I. I NTRODUCTION

V

OICE over Internet Protocol (VoIP), also known as IP telephony or Internet telephony, is a set of protocols to transport voice traffic over IP-based packet-switched networks with acceptable quality of service (QoS) and reasonable cost. Wireless Local Area Networks (WLANs) have become a part of everyday technology. This has now been deployed around the world. Voice over WLAN (VoWLAN) has been emerging as an infrastructure to provide low-cost wireless voice services. However, since the performance characteristics of wireless networks are much worse than wire line counterparts, and the IEEE 802.11-based WLAN was not originally designed to support delay-sensitive voice traffic. Third generation (3G) packet switched UMTS/WCDMA networks with High Speed Downlink Packet Access (HSPDA) is being installed worldwide. With the introduction of HSDPA in 3G networks, packed switched wireless systems will allow dynamic resource sharing therefore more efficient use of bandwidth and improved network efficiency will be possible. Since voice applications are real-time, they are intolerant of lengthy delays, packet losses and jitter (delay variation). All these problems degrade the quality of the voice transmitted. These QoS issues over 3G wireless networks will be assessed with respect to network load, packet switching, buffer length and packet segmentation under certain protocols, such as Adaptive Multi-Rate Speech Codec, HC (Header Compression), RTP (Real Time Protocol) and RLC (Radio Link Control Layer).

The challenges for achieving this include typical VoIP related QoS (Quality of Service) problems, such as delay, delay variation (i.e. jitter), packet loss and additional overhead brought by the VoIP protocol stack [1]. The QoS problems for VoIP over LTE will be analyzed by comparing physical layer techniques and try to obtain the best one in terms of VoIP quality, and compared with some simulation or data. When talking about quality on WLAN it is useful to distinguish between scenarios. If your WLAN access point keeps crashing then you could say QoS for you is poor. To get best access point (AP), one can describe the IEEE family 802.11 to find what technique should be used. The paper is organized as follow. Section II we focus our attention on issues that afflict VoIP in WLAN networks. Section III we consider QoS over 3G networks. Section IV we analyze QoS and physical layer techniques to improve VoIP quality over LTE networks. Finally, Section V we present our conclusion and possible future implementations. II. VO IP OVER WLAN The Wireless Local Area Network (WLAN) becomes popular to support high-data-rate Internet access for users in proximity of an access point (AP). The main advantages of WLAN is simplicity, flexibility and cost effectiveness. VoWLAN applications use the infrastructure based on WLANs. There is a variety of standards defined in the IEEE 802.1 [2]. The most deployed standard is 802.11b, whereas 802.11g is receiving acceptance because of the high rate and backward compatibility with 802.11b. WLANs are only specified at the physical layer and part of the data link layer. Reason why security and QoS on WLAN is hard is because of all IP routing, session control etc is outside the scope of WLAN, since both security and QoS are clearly needed end to end then higher layer solution needs to interface o the WLAN capabilities VoIP is real time applications and WLAN is not basically


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TABLE I QOS

Fig. 1.

VoIP over WLAN [3]

made for real time application because the QoS which is big issue with WLAN is important and main portion of VoIP applications. Challenges VoWLAN: there are many challenge in VoWLAN like Quality of Services (QoS), security etc QoS in VoWLAN consist of these three thing which is discussed below. A. Packet Loss The total number of packet transmit over the network is not receive to the end point or destination, so it means the some data or packet loos or not received by the destination. There are two main sources of packet losses:one is network packet losses, mainly due to network congestion (router buffer overflow), link failures and rerouting, transmission errors, etc; and the other is discarded packet losses for packets experienced excessive delay. B. Delay The time taken by a packet to reach from a source to destination, delay can be occurred from different sources like delay at source, delay at receiver, delay in network. Delay at source and receiver is due to coding like changing analog to digital and digital to analog and packetization, while network delay is due to transmission, queuing and propagation. C. Jitter The variation of time between packet transmit from source to reach destination, means one packet reach in 100 ms and one reach in 125 ms to handle this problem jet-buffer is used at receiving end and it has two type static jet-buffer which hardware base and dynamic base which is software base and can be handle by administrator .but should take care about jet-buffer because some time it is also becoming reason for delay like memory over-flows etc. The following are the some measurement and recommendation of ITU-T G.114 for a VoIP call for the three attribute which is define in tab. I [5]. “Factors such as packet delay, jitter, packet loss and network latency can noticeably affect the quality of UDP- based

Packet delay

Packet loss

Jitter

≤ 150 ms

≤ 1%

≤ 25 ms

services such as VoIP and video streaming. Contrary to TCP-based services such as HTTP, SMTP, etc, a steady stream of data packets is crucial for VoIP connections, where even slight connectivity problems can cause noise or echo”. “Quality of any service depends on the traffic flow as well as the network of terminating partners. Following are some issues should be considered to provide better-quality service. Number of calls managed simultaneously by the network The alternate way to transfer the call to it desired destination in case of any fault/failure occurred in the network CODECs for coding and encoding purposes. Overall setup of the network” [6]. D. Original IEEE 802.11 MAC layer The original IEEE802.11 has no idea QoS especially for voice data application have no sensitivity about Delay jitter. The basic MAC layer use distributed coordination function (DCF) and Point coordination function (PCF ) to share medium with station both have several limitation [2]. DCF relies on CSMA/CA and optional 802.11 RTS/CTS to share the medium between station. The problem in DCF is that if many station want to communicate at the same time there is always a collision occur and it is based on collision avoidance means it has to wait the medium to be free which produce delay and if collision occur it is waste of the bandwidth and make communication slow. some problem in DCF: • there is no QoS guaranty and priority between data traffic like voice and data; • if a station sense medium and it is free and get medium to communicate no other can’t communicate until it didn’t let free the medium if a station has slow bit rate it will capture the medium for along time. PCF is the other coordination which is define by basic IEEE802.11. It is optional. PCF is used only in infrastructure mod in which all station are connected by one center object called Access Point (AP). PCF define two frame Contention Free Period (CFP) and Contention Period (CP). In the CP DCP is used. To give the right of communicate over the medium the CFP send Contention-Free-Poll (CF-Poll) to station at time one packet each. The AP is coordinator. PCF has a little


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bit QoS management but have no idea of the different class of traffic. E. IEEE 802.11e This standard define enhancement in the original IEEE802.11 Mac layer DCF and ECF with new coordination function Hybrid Coordination Function (HCF). It proposed priority and class based traffic means the voice and multimedia application data class will have high priority during transmission compare with other data like email data class in a shared wireless medium etc. There are two method to access the channel to communicate like original IEEE802.11 MAC. HCF Controlled Channel Access (HCCA) and Enhanced Distributed Channel Access (EDCA)[6]. With the EDCA the data which have high priority will have have chance to send early then the low priority data and station having EDCA implemented will have to wait less to send data. It work mostly like PCF. In PCF scenario the interval between to beacon frame is divide into two period CFP and an CP, the HCCA is allowing the CFP to initiated almost any time during CP. This kind of CFP is called Access Phase (CAP) in 802.11e. The AP will initiate CAP any time which it want and can receive frame from other contention-free manner. The CAC is a method which will decide whether a new connection will be allow to established or not, it will be decide on the basis of capacity of WLAN means if the new connection is allowed what will be MOS or quality of over all call which would be specified. So the CAC will maintain the over all quality of Voice of VoWLAN. For infrastructure mode of VoWLAN the CAC can be implemented in AP. Codec is used to convert voice signal to digitally encoded version compress it on the sender end and then reverse the processes on the receiver end. These codec are standardized by International Telecommunication (ITUT). There are many codec technique which is used in VoIP for Encoding and Decoding. Some coding and it different result are mentioned in below table fig.2 which has been calculated with different IEEE 802.11 standard with 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 different code which different bit rate per kbps and with have different WLAN IEEE802.11 like a,b,g and got different MOS and found how much simultaneously connection or calls can be established at time on per AP. Among we observe that Codec G.711 have high MOS rate and with reasonable simultaneously calls at time. But it should

Fig. 2.

VoIP over WLAN

be care about the bandwidth of connection is ok for requirement of codec selected like G.711 require at least 128 bit for both way communication. MOS is International Telecommunications Union Telecommunication Standardization sector (ITU-T) approved which gives a numerical indication of the perceived quality of the media received after being transmitted and eventually compressed using codec. The WLAN are working on radio wave which are open which can eavesdrops and some one can manage to use it illegally like crack the secret key. F. IEEE 802.11i The IEEE802.11e enhance the security issue of original WLAN and put forward the WAP2, it using Advanced Encryption Standard (AES) block cipher. The WEP and WAP were using RC4 stream cipher. The IEEE802.11e replace the issue of Authentication and privacy issue with more detail and security adjustment [9]. Different VLAN can be used to separate Voice traffic and data traffic: it will solve the space problem and voice device can be protected from external network. Separate VLAN will have private addresses which will hide phone device from directly connected to public network; QoS trust boundary extension to voice devicesQoS trust boundaries can be extended to voice devices without extending these trust boundaries and, in turn, QoS features to PCs and other data devices; protection from malicious network attacks-Subnet access control, can provide protection for voice devices from malicious internal and external network attacks such as worms, denial of service (DoS) attacks, and attempts by data devices to gain access to priority queues; ease of management and configuration-Separate VLANs for voice and data devices at the access layer provide ease of management and simplified QoS configuration. III. VO IP OVER 3G Traditionally, real-time services (e.g. voice) are transported over dedicated channels because of their


Fig. 3.

Transport of speech in IP

delay sensitivity while data is transported over shared channels because of its transmitted in short, uneven spurts. In order to carry voice on IP networks, appropriate protocols must be used. The main protocols are Real Time Protocol (RTP), User Datagram Protocol (UDP) and Internet Protocol (IP) [11]. In Fig. 1, the voice frames are generated in the application layer, encoded and encapsulated within payload of an RTP SDU. The RTP PDU is encapsulated into an UDP SDU, which is delivered to the IP layer. Adaptive Multi-Rate Speech Codec (AMR) is a codec with 8 narrow-band speech encoding modes with bit rates between 4.75 and 12.2 kbps. If the data rate is 12.2 kbps, the AMR codec generates packets of 244 bits which represent voice frames of 20 ms [12]. Since the AMR codec encodes and decodes digital speech data with an optimum power and bandwidth consumption, the Internet Engineering Task Force (IETF) has approved the RTP payload format for AMR. Real Time Protocol (RTP) is an end to end transport protocol, used to transport multimedia traffic in IP networks, supporting unicast and multicast traffic. In the case of VoIP service, it is implemented together with UDP/IP [11]. Since RTP does not provide any reliability mechanisms and other layers should be implemented. AMR and RTP the main performance parameters for VoIP quality that are described earlier, can be measured by the RTP protocol. RTP AMR and RTP The main performance parameters for VoIP quality that are described earlier, can be measured by the RTP protocol. User Datagram Protocol (UDP) as a transport layer protocol for VoIP over Internet Protocol (IP), UDP is used to avoid any retransmission delays. On the other hand, it provides no reliability on datagram delivery. The UDP header size is standardized in 8 bytes and 20

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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 is important to use bandwidth efficiently. On the other hand, large headers of the protocols used when voice data is sent over the wireless network where a high bit error rate (BER) due to fading and mobility is present. Robust Header Compression (ROHC) protocol has been developed for this problem. The effective compression makes use of the fact that majority of the fields in the combined IP, UDP and RTP header either remain constant or introduce constant change throughout a session.

A. QoS Analysis One main parameter for assessing packet loss is FER (Frame Error Rate). Although packet loss is undesired some loss can be tolerated since error-concealment techniques can be used. Buffer length can also cause packet loss due to discarding of delayed packets. On the other hand buffer length also may also increase the delay where for acceptable conversational quality, the maximum end-to-end delay should be around 250-300 ms [13]. Therefore buffer length takes an important role short buffering time will risk buffer underflows causing jitter, and long buffering time causes long delay and buffer overflows. Too short buffering time may also cause increased packet loss due to loss of segmented packets. Simulation with parameters specified for 2 different simulations can be seen in tab. II. From the first simulation it can be seen that for different error probabilities, ranging from 1% to 10%, packet loss is directly related to the load on the wireless network. With the increase number of network users, applied packet switching technique is not feasible. Therefore it can be said that delay and delay jitter mainly depends on both Round Robin switching


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TABLE II S IMULATION PARAMETERS Parameter

Value (Simulation 1)

Simulation runs

10000

Value (Simulation 2) 6min30s of speech

Load

Variable

One user

Channel error

Variable

Variable

12.2kbps

12.2kbps

20ms

20ms

probability AMR source data rate AMR voice frame duration Call duration

120s

390s

Silence

Voice on/off

Silence Descriptor (SID)

periods

(mean duration 3s)

(160 ms intervals)

AMR voice

244bits

244bits

Protocol Stack

RT P + U DP + IP v4

RT P + U DP + IP v6

size

= 40 byte

= 60 byte

Header

Robust HC

Robust HC

Unacknowledged

Unacknowledged

Mode

Mode

3

None

4

None

Round-Robin

None

packet payload

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

size

Compression (HC) RLC mode Maximum number of MAC-hs retransmissions Number of MAC-hs H- ARQ

Fig. 7.

parallel processes Packet scheduling

Simulation Results for Simulation 2[1]

algorithm Delay budget

100ms

Predefined jitter buffer (FIFO algorithm)

packet loss ratio increases on the wireless channel, total packet loss rate increases. The reason for occurrence of erroneous packets in loss-free simulation is due to the packet segmentation at RLC (Radio Link Control Layer), where packets larger than one TTI (Time Transfer Interval) are segmented over several TTIs, introducing longer transmission delays and packet drops. IV. VO IP OVER LTE

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

technique and Hybrid-ARQ mechanism where the main features of MAC-hs (Medium Access Controlhigh speed) protocol of HSDPA are retransmission of erroneous packets which is handled by H-ARQ and sequential delivery of the packets to the upper layer [14]. This reasoning can also be seen in the PDF of delay jitter for 5 fixed users on the network in figure 7. In simulation 2, a predetermined buffer is implemented; therefore average network delay is constant for different error probabilities. On the other hand as

There are two important conditions must be met to ensure an adequate VoIP quality: 1) delay from sender to receiver must be as low as possible; 2) packet loss must be between 1% to 3%. So, in LTE, end-to-end Quality of Service is based on two parameters that formalize these two conditions. First, Layer 2 Packet Delay Budget is specified for every connection and for every User Equipment (UE). Second, Layer 2 Packet Loss Ratio is defined in order to guarantee the above specification. Hence, if a VoIP connection has a L2PDB of 100 ms and a L2PLR of 2% it mens that the QoS level for a subscriber is satisfactory. [16] In wireless networks, like LTE, the principal cause of issues is the path between the radio base-station and


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the UE. In fact, there are many new physical layer techniques made to try to avoid the bit errors and the delay, for example: Hybrid automatic repeat request (HARQ) or advanced channel coding. Given that LTE is strongly dependent on HARQ, reducing the bit errors, the delay over the connection link will be reduced as well. [16] LTE Hybrid ARQ is a physical layer technique to increase robustness against transmission errors, and to increase capacity. It is part of the MAC layer but the soft-combining operation is handled by the physical layer. In this technique, the erroneously received packet is stored in a buffer memory and later combined with the retransmission to obtain a single, combined packet which is more reliable than its constituents. Decoding of the error-correcting code operates an the combined signal. If the decoding fails, a retransmission is requested. [17] There are four kind of HARQ schemes, the first one is called Type I HARQ and it is based on the use of Cyclic Redundancy Check (CRC), the second one is Type I CC HARQ because it is the same of the first one, but it uses a Chase combining technique, the third one is Type II Full IR (Incremental Redundancy) and it gradually decreases coding rate in each transmission by sending additional redundancy bits [18] and the last one is Type III Partial IR and as the previous, it gradually decreases coding rate by sending additional redundancy bits, but each bit maintains self-decodability in each retransmission [18]. Type I CC HARQ is a scheme that, when the receiver finds an error, it discards erroneous packets and sends a retransmission request to the transmitter. The entire packet is retransmitted. The packets are combined based on either the weighted SNR?s of individual bits, in which case the technique is termed Chase combining [19]. Type II Full IR is a scheme, where retransmission requests consist only of parity bits. The receiver combines additional parity bits from retransmission with bits of the first transmission resulting in lower rates, before FEC decoding is attempted [20]. Type III Partial IR is a schemes, in which individually transmitted packets are self-decodable and each packet differs in coded bits from the previous transmission. In Type III ARQ, packets are only combined after decoding has been attempted on the individual packet [21].

Fig. 8. Packet error ratio in function of SNR for a modulation 64QAM with gray coding 34 coding rate. [18]

Given that, we analyzed simulation results of [18] showed in 8 and we can say that to achieve an acceptable quality of service, based on our previous parameters, the best choice it will be HARQ Type III Partial IR. V. C ONCLUSION In this paper we focused on QoS issues of VoIP over WLAN, 3G and LTE and tried to analyze these issues by comparing different studies and proposing new ideas for future work. As a result, conclusion for this paper can be discussed in 3 parts. VoIP over WLAN is real time application and very sensitive to delay, packet loss and jitter but on the other hand security is also an issue. This is because WLAN is mainly on physical and MAC layer where the security is handled in the upper layer. However this kind of problem can be overcome if IEEE802.11e standard is implemented where “Call Admission Control” will monitor the voice quality. In addition, by adding G.711 coding technique, high quality multiple simultaneous calls will be possible. On the other hand for more secure VoIP applications IEEE802.11i should be implemented and appropriate coding technique such as G.711 should be simulated for future work. VoIP over 3G, end-to-end QoS analysis of two similar simulations under same protocols shows that current 3G networks offer an adequate level of quality for VoIP services. However, to improve this some further analysis can be carried out. •

As the number of user increases packet switching in HSPDA becomes more important. Therefore more capable Expo-Linear packet switching technique can be simulated. This technique calculates the user priority not only based on ranking users according to their instantaneous channel quality, relative to


•

•

their own average channel conditions but also the delay bound. Therefore, it is able to meet the different QoS requirements of real time users [15]. MAC-hs protocol enables retransmissions which causes decrease on QoS. On the other hand introducing predetermined TTIs also causes segmentation problems related to RLC creating delay and packet loss. Therefore an adaptive TTI and buffering should be simulated for future work. Only then MAC-hs protocols retransmission can be fully effective since RLC does not guarantee delivery. Current Packet Loss Concealment techniques are effective only for small numbers of consecutive lost packets, for example a total of 20-30 milliseconds of speech, and for low packet loss rates. Therefore a further study on intelligent PLC where a learning technique can overcome packet loss issues.

VoIP over LTE, QoS analysis is mainly based on variants of H-ARQ to improve Eb/N o over a low packet error rate which is usually 10−3 for voice and 10−6 for data transmissions. The trade-offs in this assessment was between memory usage and SNR. Standard H-ARQ needs almost no memory but provides very little SNR improvement on the other hand Type-II Full Incremental Redundancy requires high memory but provides more than 10 dB improvement compared to the standard HARQ. Therefore Type-III Partial IR where the retransmitted packet can be chase combined with previous packets to increase the diversity gain, is the main candidate for future work. R EFERENCES [1] Renaud Cuny, Ari Lakaniemi, VoIP in 3G Networks: An Endto-End Quality of Service Analysis, Nokia Research Center. [2] INTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS 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 online 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, Wireless Telecommunications Research Group - GTEL, Department of Teleinformatics Engineering - DETI, Federal University of Ceara, 2006. [12] 3GPP, Mandatory speech codec speech processing functions; amr speech codec; error concealment of lost frames, 3rd GenerationPartnership Project, 1999. [Online]. Available: http://www.3gpp.org

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[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 Telecommunications Engineering, 2005. [15] Matthias Malkowski, Andreas Kemper, Xiaohua Wang, “Performance of Scheduling Algorithms for HSDPA”, Communication Networks, RWTH Aachen University. [16] Capacity Enhancement of VoIP over LTE by Stochastic Adaptive Modulation and Coding, K.Homayounfar and B. Rohani, 10-10 Cendex Center , Singapore. [17] E. Dahlman. 3G Evolution: HSPA and LTE for Mobile Broadband. Academic Press, 2008. [18] Kian Chung Beh, Angela Doufexi, Simon Armour, “PERFORMANCE EVALUATION OF HYBRID ARQ SCHEMES OF 3GPP LTE OFDMA SYSTEM”, The 18th Annual IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC?07). [19] D. Chase, “Code combining; A maximum likelihood decoding approach 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 code combining”, 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 Combining methods in Fading Channels”, Dept. of Electrical Eng. and Telecomm, 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)