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Estimating quality of service of mixed traffic in 3G networks
Abstract
This paper focuses on the performance of traffic mix which consists of both voice and data services. The paper deals with intra-frequency soft handover in a 3G networks where mobile stations are connected simultaneously to several base stations. The QoS of voice for mobile users in the mixed traffic environment is estimated. The results are compared with the performance obtained by deploying a scheduling scheme based on weight. The scheme is simulated with different number of users. It is observed that the scheduling scheme could improve the overall network performance. Although the number of users is increased, the end-to-end delay can be reduced significantly. The results show that the scheme can improve the system capacity while maintaining the acceptable end-to-end delay.
... An access control protocol was proposed for an integrated voice, video and non real-time data traffic on the forward link (cell-site to mobile). The QoS of voice for mobile users in the mixed traffic environment was estimated in [9]. The results were compared with the performance obtained by deploying a scheduling scheme based on weight. ...
Third Generation (3G) mobile devices and services will transform wireless communications into on-line, real-time connectivity. 3G wireless technology will allow an individual to have immediate access to location-specific services that offer information on demand. 3G wireless systems such as 3G-1X, 1xEV-DO and 1xEV-DV provide support for a variety of high-speed data applications. The success of these services critically relies on the capability to ensure an adequate QoS experience to users at an affordable price. The traffic being transferred within 3G mobile networks will be composed by different information flows with various constraints on the required QoS (bit rate, delays, etc.). In this paper, we give an evaluation of the performance of 3G networks when providing access to voice and data services. In particular, we analyze through simulations a typical scenario where voice and data services affect the packet traffic in 3G system approach.
The architecture of the IP Multimedia Subsystem (IMS) defined by the Third Generation Partnership Project (3GPP) provides a solid base for solutions for interconnecting fixed and wireless networks and supplying new services over them, such as voice over IP (VolP). Compared with circuit-switched voice, the main end-user benefit of VolP is the ease with which end users can enrich basic voice service with multimedia services during an ongoing session. Improved end-user flexibility will also probably lead to an increase in traffic and operator revenues. VolP also has the potential to help operators lower their operating costs by converging their networks into a single all-IP network. The challenges associated with deploying VolP over mobile systems include matching the efficiency and quality of circuit-switched voice service while maintaining the inherent flexibility of IP-based services. In the process, one must consider every aspect of the VolP realization, from compression of IP header overhead to radio access bearer, including quality enhancements and the handling of jitter. The authors provide an overview of VolP and its transmission over mobile radio links with available packet-based access (GPRS/EDGE, WCDMA) and dedicated bearers (HSDPA/E-UL). They also describe the capabilities a radio network must have to provide a flexible packet-switched voice service with adequate quality and capacity.
Radio Network Planning and Optimisation for UMTS comprehensively explains how to dimension, plan and optimise UMTS (Universal Mobile Telecommunications System) networks. It introduces the properties of the spread spectrum system and provides a general overview of the physical layer of UTRA FDD. The radio network planning process for WCDMA is clearly presented and detailed information on how to dimension, plan and rollout a 3G network, both theoretically and practically is provided. This valuable text examines current and future radio network management issues and their impact on network performance as well as the relevant capacity and coverage enhancement methods.
- Includes automation examples of radio resource management
- Focuses on UTRA FDD and introduces UTRA TDD, GPRS and EDGE and examines their interaction and synergy
- Provides an excellent source of information for those considering future cellular networks where Quality of Service (QoS) is of paramount importance
- Analyses the radio network planning challenges and opportunities for both greenfield and existing operators
- Includes an accompanying CD-ROM featuring a static radio network simulator implemented in MATLAB
Authoritative and instructive, this text will have instant appeal to wireless operators and network and terminal manufacturers. It will also be essential reading for university students, frequency regulation bodies and everyone interested in radio network planning and optimisation, especially RF network systems engineering professionals.
This paper presents an analytical study on voice over Internet protocol (VoIP) performance in a third generation (3G) network evaluated under certain IP conditions. The mouth-to-ear delay is analysed using different types of coding schemes to estimate the actual voice packet arrival as traffic sources are very bursty in nature. The simulation results closely match to the theoretical model and it also indicates that encoder types have an impact on the system performance. The stability of a system is also affected by the rate of voice traffic
As mobile communication involves transferring an ongoing call from one cell to another as the user moves through the coverage area, it is important to study the factors that affect the overall performance. This paper presents the process of voice over Internet Protocol (VoIP) in a third generation (3G) network with performance was evaluation. The paper also discusses the quality of voice when compression technique is used to reduce the amount of required bandwidth. Each packet takes a different amount of time to go from the source to its destination. The actual voice packet arrival is estimated and the delay from source to destination is calculated. The results show the impact of packet arrival rate and traffic load on the overall performance.
The ubiquity and convergence of mobile telephony and the Internet leads many to believe that future isochronous telephony data will be transported in the form of voice-over-IP (VoIP) datagrams. This paper investigates the modelling and aggregation of multiple VoIP streams. The relationship between the number of VoIP sources, output link rate and certain teletraffic metrics is investigated. Thereafter, an exemplar network consisting of a small cluster of wireless cells is considered and a number of different handover strategies enacted. The frequency and duration of handovers are varied to investigate the effects on system performance when VoIP calls are ongoing. The findings in this paper could be extremely useful when dimensioning networks as they provide insight into the QoS obtainable for the individual streams. This is a critical issue for real-time applications such as VoIP.
One important issue in 3G wireless systems is to port voice service to the new packet-data-based platform while retaining the quality of service (QoS) that users associate with present-day circuit-switched realizations. Flexibility in packet switching introduces difficulty in maintaining QoS requirements, in particular, the delay requirement. In this work, we conduct research on delay performance of Voice over Internet Protocol (VoIP) services over high speed downlink packet access that is used in 3G systems to provide high speed packet data service. We evaluate the VoIP network capacity under different packet delay budgets that associated with the radio access channel. We also investigate the impact of rate profile, spreading factor, code capacity, multiplexing capacity and header compression algorithm.
This paper presents the results of a quality of service (QoS) study for the VoIP service over 3G WCDMA networks. An end-to-end simulation platform has been used for this purpose. The simulations have been using an adaptive multi-rate (AMR) speech codec at 12.2 kbit/s with combination of RTP, UDP and IPv6 protocols. The simulated transmission path includes two radio links (uplink and downlink), connected with a packet switched core network and UTRAN radio access networks with several different radio transmission conditions. Furthermore, robust header compression is applied in both radio links. The results include buffering statistics, end-to-end delay estimates, and packet loss statistics.
There has been a significantly increased interest in building
future UMTS systems based on IP (Internet protocol) technologies. The
major motivation is to have a common service platform and transport for
3G networks based on IP. This is also reflected in the establishment of
the 3G.IP industry group and current 3GPP standardisation activities.
This paper presents an overview of this trend and summarises the major
technical issues and challenges, with emphasis on migration towards an
IP based UMTS system. This paper consists of three parts. It starts with
a description of `all-IP based UMTS networks', including the scope of
applying IP technologies and considerations of candidate architectures.
An architecture is presented for all-IP based UMTS networks in which a
packet multimedia service domain is overlaid on top of UMTS packet
switch transport. The second part highlights an overview of the major
technical challenges. This covers issues in voice/multimedia services,
quality of service, mobility management, legacy system support and
transport efficiency including radio efficiency. Finally the paper
finishes with a discussion on the realisation and migration from UMTS
Release '99 (R'99) network to such an all-IP based UMTS architecture
The introduction of IP-based real-time services in next-generation mobile systems requires coupling mobility with quality of service. The mobility of the node can disrupt or even intermittently disconnect an ongoing real-time session. The duration of such an interruption is called disruption time or handover latency, and can heavily affect user satisfaction. Therefore, this delay needs to be minimized to provide good quality of VoIP services. In this article, we focus on network-layer mobility and mobile IP since it is a natural candidate for providing such mobility. We evaluate different low-latency schemes based on mobile IP and compare their performances in terms of disruption time for VoIP services. Low-latency handoffs are performed by anticipating and/or postponing the mobile IP registration process. With these methods, disruption time is reduced to 200 ms in most considered cases.
Wireless Internet access has gained significant attention as wireless/mobile communications and networking become widespread. The voice over IP service is likely to play a key role in the convergence of IP-based Internet and mobile cellular networks. We explore different mobility management schemes from the perspective of VoIP services, with a focus on Mobile IP and session initiation protocol. After illustrating the signaling message flows in these two protocols for diverse cases of mobility management, we propose a shadow registration concept to reduce the interdomain handoff (macro-mobility) delay in the VoIP service in mobile environments. We also analytically compute and compare the delay and disruption time for exchanging signaling messages associated with the Mobile IP and SIP-based solutions.