The main issue in real time voice applications over Internet is to investigate a lossless playout without jitter while maintaining playout delay as small as possible. Existing playout algorithms estimate network delay by using timestamps and determine the playout schedule only at the beginning of each talkspurt. Also their scheduled playout time is determined based on a fixed upper playout delay bound over a talkspurt. The sliding adaptive playout algorithm we propose can estimate jitter without using timestamps and its playout time is allowed to slide to a certain extent. The aim of sliding playout schedule is to determine the scheduled playout time at the beginning of each talkspurt based on the playout delay expected under the normal condition where the degree of actual jitter is below the magnitude which is not quite large in relation to variations in the "baseline" delays. Then the proposed algorithm can be effectively applied to minimize the scheduled playout delay while improving the voice quality against a spike which may occur at the beginning of a talkspurt as well as a spike which occurs in the middle of a talkspurt. We develop an analytical model of the general adaptive playout algorithms and analyze the playout buffer performance for different degrees of jitter, different values of the scheduled playout delay, different maximum lengths of delay spikes, and arbitrary tolerable ranges of sliding. Based on the analytical results, we suggest the specific values of parameters used in the sliding algorithm.

Providing quality-of service (QoS) guarantees in VoIP applications has become an urgent demand in wireless and mobile networks. One of the important issues is to find a simple quality metric fitted to low power mobile devices such as smart phones. This paper considers the gap ratio (the proportion of the accumulated gap periods over the whole call session) as a simple quality metric. Our study aims to find the optimum packet count threshold between two adjacent lost packets (referred to as Gmin in RTCP-XR), which is needed for the purpose of identifying whether the current packet at the receiver belongs to the gap state or the burst state, because quality prediction errors depend on the Gmin value when the gap ratio is used as a simple quality metric. Based on this metric, we propose an accounting model that can be a candidate accounting metric useful for a quality-based accounting mechanism.

Currently, a correspondent host will have difficulties in establishing a direct session path to a mobile host because of the partial deployment of MIPv6-aware mobile hosts. Even MIPv6-aware hosts will spend up to several seconds to obtain the new location of the mobile host during Layer 3 (L3) handover. This paper proposes an application-level mobility management scheme that can solve the problems related to the increase of Internet traffic end-to-end delay under the current situation that most of the mobile devices are MIPv6-non-aware. The proposed Secure Mobility Management Application (SMMA) enables the updates of care-of address to be faster and more reliable even when L3 handovers occur frequently. SMMA uses a cross-layer approach for session mobility management with the support of Binding Updates to the home agent via IPSec tunnels. The main feature of SMMA is to handle the session-related mobility management for which operation starts just after the completion of name resolution as a pre-call mobility management, which operates in conjunction with the DNS. Our session-related mobility management introduces three new signaling messages: SS-Create for session state creation, SS-Refresh for session state extension and SS-Renewal for updating new care-of address at the mid-session. Finally, this paper analyzes the work load imposed on a mobile host to create a session state and the security strength of the SS-Renewal message, which depends on the key size used.

The playout delay for voice over IP applications is adjusted on every talkspurt. The parameter β that controls the delay/packet loss ratio is usually fixed, based on high jitter conditions. In this letter, a β-adaptive playout algorithm is presented, where the β is adjusted. The buffering delays and lateness rates are compared against the existing algorithm with the fixed β. We show that the β-adaptive system improves the lateness loss performance, especially for low jitter conditions, while maintaining almost identical buffering delay/lateness loss performance when jitter is high.

The basic playout scheme (BAS) is designed not to take into account network impairment information during silence periods. We propose a jitter-robust playout mechanism (RST), which uses silence description (SID) packets. The lateness loss percentages are compared between the BAS and the RST algorithms. We report that the accuracy of the playout schedule calculation in the BAS is getting worse as the previous silence interval increases and our proposed RST algorithm is more effective in removing high jitter than the BAS. Under high jitter Internet conditions, the accuracy of the estimates and therefore the resulting of VoIP playout quality can be significantly improved by using the SID packets in the playout schedule recalculation.

To ensure secure mobile communication, the communicating entities must know their mutual identities. The entities which need to be identified in a mobile communication system are mobile devices and the network. Third Generation Partnership Project (3GPP) has specified Evolved Packet System Authentication and Key Agreement (EPS AKA) procedure for the mutual authentication of user and the Long Term Evolution (LTE) network. EPS AKA certainly overcomes most of the vulnerabilities in the Global System for Mobile Communications (GSM) and Universal Mobile Telecommunication System (UMTS) access procedures. However, the LTE access procedure still has security weaknesses against some of the sophisticated security threats, such as, Denial-of-Service (DoS) attacks, Man-in-the-Middle (MitM) attacks, rogue base station attacks and fails to ensure privacy protection for some of the important parameters. This paper proposes an improved security framework for the LTE access procedure by ensuring the confidentiality protection of International Mobile Subscriber Identity (IMSI) and random-challenge RAND. Also, our proposed system is designed to reduce the impact of DoS attacks which try to overwhelm the network with useless computations. We use a one-time shared key with a short lifetime between the UE and MME to protect IMSI and RAND privacy. Finally, we explore the parameters design for the proposed system which leads to satisfy the requirements imposed on computational load and latency as well as security strength.