Phenomena in 4G Networks

After introducing the main concepts related to mobility in 4G networks and the relevant methods and models linked to the evaluation of QoE, the potential effect that mobility across heterogeneous networks may have on the QoE of mobile users in 4G networks. There are three main sources for potential degradations in overall call quality: (1) changes in the underlying network conditions, (2) network handover, and (3) codec adaptation to network changes. These phenomenona will be briefly described next, as they are the core conditions studied in this work.

Changes in Network Conditions

A main characteristic of 4G Networks is heterogeneity. This translates into potentially severe changes in network conditions when seamlessly roaming across wireless technologies. The variations in packet loss, delays and bandwidth to which mobile applications are confronted with in 4G networks may lead to changes in the user's quality experience. Variations in network conditions are common, but the magnitude of the changes will increase in 4G networks, and new methods to deal with this will be needed in order to maintain an acceptable QoE. A first step towards building these new methods is to understand the impact of these variations on user experience. This chapter discusses this impact for an ongoing VoIP call when a user moves between independent wireless access networks.

Network Handovers

In Next Generation Mobile Networks (NGMN), the convergence of a multitude of different network technologies will provide the user with transparent and ubiquitous access to data and media services. The independence of network and service layers also introduces a new level of mobility. Equipped with appropriate terminals, users may move through geographical areas covered by different wireless network technologies while service access is preserved. In order to guarantee the provision of seamless connectivity especially for time-critical services like media streaming or Voice-over-IP (VoIP), sophisticated mobility-enabling protocols like Mobile IP are needed which ascertain a fast and robust roaming between heterogeneous wireless networks. This roaming between disparate wireless technologies (vertical handovers) while keeping ongoing sessions forces a new level of application level adaptation, causing an effect on user perception. This effect has not been thoroughly quantified as it is not present on existing communication systems.

Codec Adaptation

Current voice over IP codecs can adapt to variations in the underlying networks, but may not cope with the kind of changes caused by vertical handovers, as these could result in the need to change codec families (e.g., narrowband to wideband). In this case, for an ongoing VoIP call the network handover can occur in conjunction with the appropriate application layer adaptation in order to account for the newly encountered conditions. Most important, a codec re-negotiation might be enforced to meet sudden changes in network conditions. This should ideally be done in such a manner that neither the call nor the audio stream is interrupted during the handover. Thus, the continuity of the speech is maintained, resulting in an optimal speech quality.

Special attention has to be paid to these aspects as they produce new auditory artifacts that can invalidate speech quality models in the context of forthcoming communications systems, causing mismatches in the design of networks and mobile devices and applications.

Assessment of QoE and User Perception Models

Quality has been defined as the result of a perception and judgment process, during which the user compares the perceived characteristics of the speech sound (the so-called "auditory event") to the expected or desired characteristic. Because of the necessity of perception and judgment processes, subjective tests are still the only valid and reliable means for the purpose of quantifying the impact of different types of degradations on perceived overall quality (QoE). However, instrumental models such as PESQ and the E-model have shown to provide valid estimations of the results of perceptual tests, within the limits of applications they have been de signed for. As a consequence, these models are widely used instead of auditory tests, e.g. for network planning and monitoring. Still, their range of validity needs to be respected, and this is why they cannotsimply be applied without further validation to NGMN scenarios.

In the case of transmitted speech, which is the focus of the present chapter, the perceived quality of the given conditions is collected in auditory tests. In a listening-only situation, for instance, a selection of test participants is asked to judge the quality of a number of processed speech samples. The text material, read by different speakers, is chosen according to the aim of the experiment, and the recorded clean speech files are processed by the system of interest and are finally presented to the listeners. Their task is to judge upon the perceived quality of the processed speech sample, providing a quantitative measure of the QoE.

The Telecommunication Standardization Sector of the International Telecommunication Union (ITU-T) provides information about how such tests have to be performed in detail. There, it is specified how to choose "balanced" speech material, "normal" speakers as well as "normal-hearing" test participants. In order to increase the reliability of the experiment, the recording and play-back situation is specified, as well as further test parameters which might have a significant impact on the measurement results.

The judgments of the listeners are usually limited to the identification or scaling of pre-defined properties of the percept. Therefore, a set of predefined scales are available. In the area of speech transmission, a 5-point category scale is usually employed (Absolute Category Rating, ACR). For the collection of overall quality ratings, this 5-point scale is labeled with the attributes "Excellent", "Good", "Fair", "Poor", and "Bad". Subsequently, the ratings are averaged overall participants, leading to an arithmetic mean (Mean Opinion Score, MOS) for each processing condition.

In contrast to the listening-only situation, a bi-directional communication system is employed in real-world conversation situations. Thus, the ecologically most valid method for quality assessment is the conduction of conversation experiments with two interacting participants. Here, the interlocutors are asked to have a conversation on an arbitrary or pre-defined topic. By means of, e.g., the abovementioned ACR scale, the opinion about the just finished connection is supposed to be judged. 

Another standardized paradigm, which can be regarded as a trade-off between the "artificial" listening-only experiments and the quite complex conversation tests and it is also called the "CallQuality" method. In this process, the perceived quality of a "simulated" telephone conversation is assessed. The participants are asked to listen to short extracts of a normal conversation, and verbally answer questions regarding the content of the stimulus they just heard. After five of these stimuli, they rate the quality of the entire simulated conversation on the ACR scale. The answering part was introduced to come close to a real conversation with its turn-takings, and to distract subjects from concentrating on the quality until rating it (the test participants are instructed to try to put themselves into the position of an interlocutor).

Once the perceptual effects have been quantified, instrumental quality prediction models can be developed that are capable of estimating the subjective ratings. One type of recommended models by ITU-T, the so-called "Perceptual Evaluation of Speech Quality", is based on the application-layer speech signals. Here, the quality of transmitted speech in a listening-only situation is estimated by comparing the clean and degraded signals on a perceptual level, i.e. by taking advantage of psycho-acoustic knowledge, such as the Bark-scale transform, loudness functions, time/frequency masking, asymmetries of "positive" and "negative" error components, as well as insensitivities of certain variations in delay, the spectrum, and the amplitude. PESQ has been extended towards wideband speech by applying a flat input filter and a different mapping function.

Currently, the requirements for a successor of PESQ are discussed in Study Group 12 of ITU-T which is supposed to improve known drawbacks of PESQ and is valid for an even wider range of distortions (e.g., audio bandwidth, time-warping). NGMN-specific degradations, however, will not explicitly be covered by this model.

For conversational speech quality, the ITU-T recommends the so-called E-Model, which is a parametric model usually employed for offline quality estimation, e.g. for network planning. The MOS values are estimated on the basis of transmission channel parameters commonly known in classical telephonometry (e.g., loudness ratings, noise levels), but also in the context of packet-based networks (e.g., packet loss rates). These parameters are subsumed by so-called impairment factors, which are assumed to be additive on a psychological scale, the so-called transmission rating scale (R-scale). The eventual quality estimates are then obtained by a non-linear transformation of the R-values, i.e. the summations of the perceived impairments.

Until now, none of these models have been validated to correctly predict the effects of NGMN handovers and/or codec changeovers on user perception. However, such instrumental models are indispensable to rapidly design network handover and codec changeover strategies which provide an optimum quality to the user. The auditory investigations presented in this chapter can therefore be considered as a basis for the development of NGMN-capable quality models.