4G MOBILE COMMUNICATION SYSTEMS

The success of Second-generation (2G) mobile systems in the previous decade prompted the development of third-generation (3G) mobile systems. While 2G systems such as GSM, IS-95, and cdmaOne were designed to carry speech and low-bit-rate data, 3G systems were designed to provide higher data-rate services. During the evolution from 2G to 3G, a range of wireless systems, including General Packet Radio Services (GPRS). International Mobile Telecommunications-2000 (IMT-2000). Bluetooth, WLAN, and HiperLAN, have been developed. All these systems were designed independently, targeting different service types, data rates, and users. As these systems have their own merits and demerits, there is no single system that is good enough to replace all the other technologies. Researchers are making efforts to establish 4G systems that integrate existing and newly developed wireless systems as a more feasible option. Different research programs, such as Mobile Virtual Centre of Excellence (VCE), MIRAI, and DoCoMo, have their own visions for 4G features and implementations.

At present, plethora of wireless technologies with their own merits and demerits exist globally; the upcoming 4G mobile communications system is foreseeing potentially a smooth merger of these technologies with a goal to support cost effective seamless communication at high data rate supported with global roaming and user customized personal services.

As discussed 4G will be an IP based wireless network replacing the old Signaling System 7 (SS7) telecommunications protocol, which is considered massively redundant as shown in figure 1 This is because SS7 signal transmission consumes a larger part of network bandwidth even when there is no signalling traffic for the simple reason that it uses a call setup mechanism to reserve bandwidth, rather time/frequency slots in the radio waves. IP networks, on the other hand, are connectionless and use the slots only when they have data to send. Hence there is optimum usage of the available bandwidth.

Figure 1: Example of IP based networks

 

The goal of 4G will be to replace the entire core of cellular networks with a single worldwide cellular network completely standardized based on the (Internet Protocol) IP for video, packet data utilizing Voice over IP (VoIP) and multimedia services. The newly standardized networks would provide uniform video, voice, and data services to the cellular handset or handheld Internet appliance, based entirely on IP (Internet Protocol).

4G Networks—Adoption and Dangers

4G is short for fourth-generation cellular communication system. There is no set definition for the specifics of 4G. The 4G will be a fully IP-based integrated system of systems and network of networks, achieved after the convergence of wired and wireless networks as well as computer, consumer electronics, communication technology, and several other convergences. These will be capable of providing 100 Mbps and 1Gbps, respectively, in outdoor and indoor environments with end-to-end QoS and high security, offering any kind of services anytime, anywhere, at affordable cost and one billing. The Wireless World Research Forum (WWRF) defines 4G as a network that operates on Internet technology, combines it with other applications and technologies such as Wi-Fi and WiMAX, and runs at speeds ranging from 100 Mbps (in cell-phone networks) to 1 Gbps (in local Wi-Fi networks). 4G is not just one defined technology or standard, but rather a collection of technologies and protocols to enable the highest throughput and lowest cost wireless network possible

Fourth generation networks are likely to use a combination of WiMAX and WiFi. Technologies employed by 4G may include SDR (Software-Defined Radio) receivers, OFDM (Orthogonal Frequency Division Multiplexing). OFDMA (Orthogonal Frequency Division Multiple Access). MIMO (multiple input/multiple output) technologies, UMTS (Universal Mobile Telecommunications Service) and TD-SCDMA (Time Division Synchronous Code Division Multiple Access). All of these delivery methods are typified by high rates of data transmission and packet-switched transmission protocols. 3G technologies, by contrast, are a mix of packet and circuit-switched networks. When fully implemented, 4G is expected to enable pervasive computing in which simultaneous connections to multiple high-speed networks provide seamless handoffs throughout a geographical area. Network operators may employ technologies such as cognitive radio and wireless mesh networks to ensure connectivity and efficiently distribute both network traffic and spectrum

Moreover, the objective to 4G is to offer seamless multimedia services to users accessing an all IP-based infrastructure through heterogeneous access technologies. IP is assumed to act as an adhesive for providing global connectivity and mobility among networks. 4G will more resemble a conglomerate of the existing technologies rather than an entirely new standard. An all IP-based 4G wireless network has inherent advantages over its predecessors. It is compatible with, and independent of the actual radio access technology. With IP, one basically gets rid of the lock-in between the core networking protocol and the radio protocol. IP tolerates a variety of radio protocols. It lets one design a core network that gives complete flexibility in the access network type.

The goal of this chapter is to study development, transition, and challenges in 4G implementation and mobility management issues. Mobility management has been recognized as one of the most important and challenges problems for a seamless access to wireless network and mobile service. It is the fundamental technology used to automatically support mobile terminals and join their services while simultaneously roaming freely without the disruption of communication. Mobility management operations are introduced, along with the discussions of key research issues and possible solutions.

Consequences for Handover Strategies

The auditory results help to answer the research questions raised at the beginning of this section:

1. The most important network characteristic is the packet loss rate. The second most important characteristic is the audio bandwidth. Switching the audio bandwidth is roughly equivalent to the quality degradation of 5-10% packet loss, in both narrowband and wideband conditions. The degradations due to network handover (make-before-break) alone - without codec switching - are negligible in comparison to packet loss and bandwidth switching.
2. Switching codecs is advantageous if the packet loss rate is high, and if the changeover helps to reduce the packet loss impact. Switching codecs in order to take profit of a larger audio bandwidth is advantageous only if a sufficiently long period of WB speech transmission remains. From the limited results of our tests, it seems that this minimal length is around 30 seconds. Unfortunately, it is not possible to know the remaining length of a call. As a workaround, handover strategies could consider the perceived quality and use this value to estimate multiple scenarios (using different remaining lengths), and then select the most beneficial for the user, assuming past and current conditions. An interaction could be observed between packet-loss and audio-bandwidth degradations:
   • If packet loss is high (low basic quality), the impact of audio bandwidth on perceived quality is low.
   • If packet loss is low (high basic quality), the impact of audio bandwidth on overall call quality is high.
3. Parametric quality prediction models like the E-model are not yet able to estimate the speech quality resulting from codec changeovers. Signal-based models like WB-PESQ do a better job, but do not always perform as well on in-scope data.

The results may help to design efficient network handover and codec changeover strategies. In bad network conditions (high packet loss rate), a handover should be made if the packet loss rate can be reduced by this step. In this situation, it is not important whether the audio bandwidth can be maintained or not; the reduction of packet loss should be the ultimate goal. In contrast to this, a handover can also be fruitful in good network conditions (low packet loss rate). In this case, switching to a higher audio bandwidth can help to significantly improve quality. The improvement is most effective if it occurs early in the call; the remaining call duration should be more than 30 seconds. Changing from WB to NB is always linked to a loss in quality; the pure network handover without codec changeover, however, does not significantly impact the perceived quality.

Estimation of Quality Judgments

Auditory tests like the ones described in the previous sections are expensive and time-consuming; thus, we checked the available quality prediction models as to whether they provide valid estimations also for NGMN handover situations. Two different types of models have been checked: A parametric model which estimates quality on the basis of the parameters describing the network conditions, and a signal-based model which estimates quality as a perceptually-weighted distance between the input and the output signals of the network under test.

We used an extended version of the E-model as a parametric model. It is based on the original algorithm described in ITU-T Rec. G. 107 for NB networks and has been modified to take into account WB transmission (by linearly extending the underlying transmission rating scale from 100 to 129), WB speech codecs (by defining codec-specific equipment impairment factors), and packet loss. The necessary modifications are recommended in the most recent update of ITU-T Rec. G. 107, Amendment 1. As the E-model does not yet consider NB↔WB transitions, we decided to calculate two separate scores for the samples in which such NB↔WB transitions occur, then calculate an average of the underlying transmission ratings, and transform this back to the MOS scale. As a signal-based type of model, we used the WB extension of the PESQ model which is described in ITU-T Rec. P.862.2 .

It has to be emphasized that both types of prediction models have not yet been validated for the scenarios investigated here. In addition, they estimate instantaneous speech quality of short samples, and not of entire conversations. Thus, we applied the predictions only to Tests 1b and 2b. The parametric E-model uses packet loss as an input parameter, which has not been manipulated in a controlled way in Test 2b; thus, for this test, only the signal-based model can be used. We compare the model estimations to the auditory test results in terms of MOS, and calculated the Pearson correlation r and the root mean squared error σ for each test. The results are given in Table 1.

Table 1: Pearson Correlation and Root Mean Squared Error between auditory and estimated MOS 

Test	WB E-model		WB-PESQ
	r	σ	R	σ
1b	0.58	1.44	0.96	0.44
2b	n.a.	n.a.	0.89	0.70

The parametric E-model does only use general information on the network condition (average packet-loss percentage Ppl, codec type) as an input; consequently, the predictions are not accurate. In contrast to this, WB-PESQ is able to recognize speech signal degradations caused by the network handover, codec changeover and packet loss. For Test 1b, this leads to very good prediction accuracy, even better than the value of r = 0.93 which is obtained for in-scope data (Rix et al., 2006). The prediction accuracy is slightly lower for Test 2b which contains conditions with G.722.2 coding at 12.65 Kbit/s; WB-PESQ has been shown to have bigger problems in predicting the effects of this codec compared to the 23.05 Kbit/s bit-rate used in Test 1a/b.

Description of the Experiments | 4G Networks

In NGMN scenarios, speech quality is expected to vary during a call, as a result of changing conditions in the connection, network handover and/or codec changeover. Thus, in order to quantify quality, the entire length of a call has to be considered. Standard listening-only tests which make use of speech samples of 4-8 s length (ITU-T Rec. P.800, 1996) are not suitable for this purpose. On the other hand, conversational tests - despite being comparable to normal telephone usage and thus being ecologically valid - place a content-related focus on the user's attention; in such a situation, users are generally less analytic in their judgments, and it might happen that subtle perceptual differences get blurred.

As a compromise, we opted for a twofold test protocol: (a) We simulated conversations of 60s length by concatenating 5 meaningful speech segments alternating with pauses, playing them back to the test participants, asking them to answer content-related questions during the pauses, and asking for an overall quality judgment at the end of the simulated conversation; this approach has been developed by Berger and is now recommended for call-quality measurement in ETSI Technical Report 102 5 06 v. 1.1.1 (2006); (b) The composing segments of the simulated conversations and some additional segments of approx. 6 s length were presented to the participants in a standard listening-only context, asking for an overall quality rating after each sample. We carried out two tests of the first type (Tests la and 2a) and two corresponding tests of the second type (Tests lb and 2b). The following subsections describe the test conditions, set-up and participant group in more detail.

Test Conditions

In the experimental work, Test 1a concentrated on WB/NB transitions and the effects of packet loss on perceived quality. It contained two conditions with pure NB and WB calls, 4 conditions where packet loss continuously increases until the middle (3rd segment) of the call, and then switching occurs to a loss-free network with a different codec (or not), and 4 conditions where NB→WB or WB→NB transitions occur at the beginning (2nd segment), or at the end (4th segment) of a call. We consider packet loss rates of 10-20%to be realistic constraints when a handover should be executed at latest. Table 1 summarizes the conditions. The corresponding Test 1b contains all segments of the simulated conversations, plus additional samples with similar degradations, addressing also Flash-OFDM networks. The resulting list of 26 segments for this test is presented in Table 3.

Table 1: Test la conditions. H: HSDPA; W: WLAN; Ppl: packet loss in %; switching at the beginning (beg.), middle (mid.) or end of a simulated call 

No.	Network(s)	Codec(s)	Ppl per segment
1	H	G.711	0
2	W	G.722.2	0
3	H	G.711	0,10,20,10,10
4	W	G.722.2	0,10,20,10,10
5	H→W mid.	G.711→G.722.2	0,10,20,0,0
6	W→H mid.	G.722.2→G.711	0,10,20,0,0
7	H→W beg.	G.711→G.722.2	0
8	H→W end	G.711→G.722.2	0
9	W→H beg.	G.722.2→G.711	0
10	W→H end	G.722.2→G.711	0

Test 2a was designed to put a magnifier on the most interesting findings of the first test. It focused on the switching position within a simulated conversation, as well as on additional packet loss rates (see Table 2). The corresponding Test 2b with short samples also included different network load and high packet-loss-rate scenarios for limited WLAN networks (overall 27 conditions).

Table 2: Test 2a conditions. See Table 1 for explanations 

No.	Network(s)	Codec(s)	Ppl per segment
1	W	G.722.2	0
2	H	G.711	0
3	H→W beg.	G.711→G.722.2	0
4	H→W mid.	G.711→G.722.2	0
5	W→H mid.	G.722.2→G.711	0,3,3,0,0
6	W→H mid.	G.722.2→G.711	0,5,5,0,0
7	W→H mid.	G.722.2→G.711	0,10,10,0,0
8	W→H→W	G.722.2→G.711	0
9	H→W→H→W	G.711→G.722.2	0
10	W	G.722.2	0,0/5,5,5/0,0

Table 3: Test 1b conditions. H: HSDPA; W: WLAN; F: Flash-OFDM; Ppl: packet loss in %; codec switching directly before (b.) or after (a.) network handover 

No.	Network(s)	Codec(s)	Ppl per network
1	F->H	G.722.2	0
2	F->H	G.722.2	10,0
3	F->H(a.)	G.722.2->G.711	0
4	F->H(b.)	G.722.2->G.711	0
5	H	G.711	0
6	H	G.711	10,0
7	H	G.711	20,0
8	H	G.711->G.722.2	0
9	H	G.722.2->G.711	0
10	H->F	G.711	0
11	H->F	G.711	10,0
12	H->F(a.)	G.711->G.722.2	0
13	H->F(b.)	G.711->G.722.2	0
14	H->W	G.711	0
15	H->W	G.711	10,0
16	H->W(a.)	G.711->G.722.2	0
17	H->W(b.)	G.711->G.722.2	0
18	H->W(b.)	G.711->G.722.2	20,0
19	W	G.722.2	0
20	W	G.722.2	10,0
21	W	G.722.2	20,0
22	W->H	G.722.2	0
23	W->H	G.722.2	10,0
24	W->H(a.)	G.722.2->G.711	0
25	W->H(b.)	G.722.2->G.711	0
26	W->H(b.)	G.722.2->G.711	20,0

Test Setup

Tests 1a/b and 2a/b were carried out at distinct points in time, with different participant groups. Test participants were invited to a sound-insulated laboratory, were instructed about the purpose of the test, and listened to the samples in three sessions of approx. 25 min. each (2 sessions for parts a, 1 session for parts b). Speech samples were presented over a Sennheiser HMD 410 headset at a comfortable listening level, with a background level below 35 dB(A) (ITU-T Rec. P.800, 1996). At the end of each simulated conversation of part a, as well as after each sample of part b, participants had to rate the overall quality on a 5-point absolute category scale, with 5 corresponding to "excellent" and 1 to "bad" quality. The test set-up and scale followed mainly the requirements given in ITU-T Rec. P.800  and ETSI Technical Report 102 506 v. 1.1.1. 13 participants took part in Test 1a, 24 in Test 1b, 14 in Test 2a, and 17 in Test 2b. They were recruited from the normal telephone-user population, did not report any hearing impairment, and received a voucher in return for their effort.

Testbed | 4G NETWORKS

The Mobisense testbed was explicitly deployed to assess the user perceived quality of mobility in NGNs using VoIP as the case application. In order to do this, the Mobisense testbed should support experiments involving the collection of speech samples, the execution of network handovers and codec changeovers, gathering of network traces, and the modification of network conditions. The following functional requirements can be derived from these processes:

• The mobile terminal should have connectivity to at least two different wireless access technologies like WiFi, Flash-OFDM, WiMAX, UMTS/HSDPA or GPRS/ EDGE.
• The mobile terminal should be able to seamlessly hand off a connection between these networks, as it can be done with Mobile IPv4 or SIP (IETF RFC 3261, 2002).
• Network traces should be collected during the voice calls to allow further evaluation of the network conditions and codec changeovers.
• There should be the possibility to degrade the network conditions artificially, i.e., to increase delay, jitter, packet loss, and additional traffic to the network connection in use.
• It is desired to use a broadband sharing environment (using IEEE 802.11 technology).
• A VoIP call has to be established between the mobile terminal and a fixed host with a stable Internet connection to measure only the impact of the wireless path on audio quality.
• The audio stream of the VoIP call must be recorded for further analysis and for the listening and conversational tests.
• The VoIP client should support wideband and narrowband speech codecs for different transmission bandwidths.
• The VoIP client must be able to switch between different speech codecs during an ongoing call.
• A jitter buffer monitoring to track the codec frames in the VoIP application should be implemented.

Some of these requirements were realized using off-the-shelf hardware and software components. However, other parts were specially implemented for the Mobisense testbed. For example, the PJPROJECT client (PJSIP, n.a.) was extensively modified to cope with the demanded features such as codec changeover and jitter buffer monitoring.

Hardware and Software Components

The Mobisense testbed uses Mobile IPv4 as a solution to enable seamless handovers between different radio access technologies. Mobile IPv4 requires a Home Agent, a Mobile Node, and a Correspondent Node to build a working system. The access networks emulate an integrated NGN conformed by the following technologies: LAN, WiFi, Flash-OFDM, and UMTS/HSDPA. The CN and MN were deployed on laptops with Linux 2.6.18.2, and the HA is a Cisco 2620XM router with Cisco IOS 12.2(8r), supporting MIPv4.

Mobile IPv4 was deployed as the mobility management protocol because all involved access networks support IPv4. As Mobile IP client, the SecGo implementation (SecGo, n.a.) was chosen because it provides a telnet interface for remote control (Telnet, n.a.) and supports NAT traversal (IETF RFC 3519,2003). As the VoIP framework, the PJPROJECT was selected and extensive modifications have been made to fulfill the testbed requirements. The tcpdump (tcpdump, n.a.) and Wireshark (Wireshark, n.a.) tools are used for trace collection and evaluation, netem (netem, n.a.) is used to enable changing the network characteristics in terms of adding delays, packet loss, packet duplication, packet corruption, and packet re-ordering. Finally, a TCP-based client/server application has been implemented to centralize the overall control of the test settings. In addition, an UDP sender has been implemented to enable the remote control of the VoIP client.

Testbed Deployment

Figure 1 shows the Mobisense testbed network architecture and its hardware components. The Mobisense testbed supports connectivity to six networks as attachment points to the Internet, which are based on different (wireless) technologies: two LAN networks (one home and one foreign network), two foreign WLAN networks, one foreign UMTS/HSDPA network, and one foreign Flash-OFDM network. The home LAN network is directly connected to the Internet and constitutes the fixed attachment point for the HA and CN; the MN can roam between the five remaining networks. The MN could connect to the foreign LAN and can then roam to any of the two foreign WLAN networks, supplied by a DSL line as the backhaul. Moreover, the MN can log on to a public (foreign) UMTS/HSDPA network, as well as communicate with the Flash-OFDM RadioRouter that forwards packets to the Internet via the Flash-OFDM internal Home Agent. The Flash-OFDM access network is provided by the BIB3R testbed (Steuer et al., 2006) and it is based on Flarion RadioRouter version 1.1 (a testing license in the UMTS frequency band has been granted by the German regulator enabling operational measurements).

 
Figure 1: Mobisense 4G network experimental setup

The Mobile Node may roam between LAN, WLAN, UMTS/HSDPA, and Flash-OFDM foreign networks. All of them apply IPv4 protocol stack. The mobility is served by the Mobile IPv4 protocol providing transparent mobility for overlying protocols. The deployed Mobile IPv4 infrastructure consists of a HA and a MN and does not involve additional Foreign Agents. Therefore, the MN individually obtains IP addresses and acts directly as one end of the Mobile IPv4 tunnel (reverse tunneling).

Arising from the complex setting described in this section, two further requirements were posed to the design: (1) the CN, acting as the counterpart of the VoIP call generated by the MN, has to be remotely controlled and (2) all software components should have an interface for remote management, connecting to the MN and CN. The central point of the architecture is a controlling script run on the MN that synchronizes the execution of all other software components. For example, the controlling script starts tcp dump to collect network trace s, telnet to communicate with the Mobile IP client, a UDP sender to control the PJPROJECT client, and the TCP client/server. The TCP client/server is employed twofold. At first, it runs the PJPROJECT client as a background process. Secondly, it controls the software installed on the Correspondent Node. Thus, the TCP client/ server is also integrated to the CN, where it starts tcpdump and the PJPROJECT client, and controls the UDP sender.

The Mobisense project aims to evaluate the effects of mobility on the perceived quality of real time services. Therefore, the testbed was designed to provide the following features:

• Access to different network technologies
• Ability to change the network connection (network handover)
• Possibility to artificially change network conditions
• Record network traces
• Share available bandwidth using DSL and WLAN technologies
• Perform VoIP calls over the networks
• Support for different voice codecs (e.g. narrowband and wideband)
• Possibility to change voice codecs (codec changeover)
• Ability to record voice samples
• Ability to record internal parameters of the VoIP application

With these features there is much potential to evaluate different effects and their impact on the quality of VoIP calls such as different network, different codecs, network handovers and codec changeovers. In order to evaluate the perceived quality, experiments with the above listed phenomena can be conducted. The resulting audio samples are recorded and then used to run auditory tests with human listeners. During these listening tests, people are asked to rate the quality of different samples. These results can be used as an input for quality prediction models. Beyond this, the network-centric effects of the network handovers and codec changeovers can be evaluated by means of the network trace recording ability of the testbed. Through its open nature, the testbed can be easily extended towards different directions such as the evaluation of multimedia applications.

Methodology | USER PERCEPTION IN 4G NETWORKS

The presented study follows a two fold evaluation approach in order to link NGMN characteristics to user perception. This approach is presented in Figure 1. It is based on speech samples processed under specified experimental conditions, using the Mobisense testbed. According to the twofold evaluation approach, the process is divided into two orthogonal stages - perception and networking analysis. On the perception layer, the audio samples are recorded and judged in an auditory experiment to obtain average quality scores (MOSs). On the networking layer, network traces are collected on the involved network stations and used for a subsequent network trace analysis. Finally, the results of both layers are merged to develop a quality prediction model that provides speech quality estimates, including the impact of mobility events in NGMNs. Design of such models enables a successful service adaptation and mobility management in real-time.

 

Figure 1: Twofold speech sample evaluation approach for NGMNs

It has to be emphasized that this experimental study is limited to the listening-only situation. The reason is that we would like to first understand what happens in the passive context, before stepping to a more interactive situation (i.e. conversational tests). As it was mentioned, interactive scenarios are more realistic, but they are also more difficult to evaluate because the results of conversational tests depend on a number of influencing factors, like the interactivity of the test scenario (free vs. more structured task-oriented conversation) and the instruction to the test subjects (i.e. whether they have been made aware of the fact that delay may occur). Thus, despite the online capability of our testbed, we left the evaluation of delay effects for further study.