VoIP on 3GPP LTE Network: A Survey

As wireless access networks evolve towards an all-IP architecture, the principles of operations of communication services (specifically voice services), which have hitherto been circuit switched are being revisited. Voice over Internet Protocol (VoIP) has been identified as a solution and is potentially capable of completely replacing existing phone networks.  However, as opposed to circuit switching technology, the call quality obtained via packet switching through IP has not been encouraging due to certain issues. The increasing demands on data rates, mobility, coverage and better service quality, led to the evolution in Radio Access Technologies (RATs) to an era of last-mile fourth generation (4G) access technologies among which is Long Term Evolution (LTE). LTE is an all-IP network initially meant for carrying data only, while carriers would be able to support voice traffic either by utilizing 2G or 3G systems or by using VoIP. This paper seeks to describe all options for providing VoIP services as a method of voice transfer over the LTE network. Keywords: 4G, Circuit switching, Convergence, LTE, Packet switching, RAT, VoIP

VOIP Model for ICT Rural Communities Telecentre in Sintok

Transmission of Voice over Internet Protocol (VoIP) on packet switching networks is one of the rapidly emerging real-time applications. VoIP is a formation of audio and voice communication. It receive voice signal activities then encoded in digital form and divided into small parts of information as like voice data network packets. These data network packets are decoded and transmitted voice in signals then sender and receiver having a voice conversion. In a voice conversion, the clients send and receive packets in a bidirectional method. Each client work as a sender and as a receiver depends on the direction of traffic flow over network. The aim of this proposal is to propose a VOIP model for ICT rural community’s telecaster in Sintok

Network load and packet loss optimization during handoff using multi-scan approach

Handoff is a critical function that enables mobile nodes to stay connected to the wireless network by switching the data connection from one WLAN to another. During handoff the communication may be degraded or interrupted due to the high packets loss. To prevent packet loss during handoff, a handoff management scheme that employs a transport protocol has been proposed. It supports multiple connections for Voice Over IP communication and makes handoff decision based on the number of frame retransmission on the MAC layer. Moreover, the handoff scheme uses the multi-scan technique that enables mobile nodes to use two WLAN interfaces for channel scanning and multi-path transmission rather than single WLAN interface. This technique introduces extra network overhead during multi-path transmission. This work optimizes the network overhead and packet loss and keeps VoIP communication at an acceptable level

Survey of unified approaches to integrated-service networks

The increasing demand for communication services, coupled with recent technological advances in communication media and switching techniques, has resulted in a proliferation of new and expanded services. Currently, networks are needed which can transmit voice, data, and video services in an application-independent fashion. Unified approaches employ a single switching technique across the entire network bandwidth, thus, allowing services to be switched in an application-independent manner. This paper presents a taxonomy of integrated-service networks including a look at N-ISDN, while focusing on unified approaches to integrated-service networks.The two most promising unified approaches are burst and fast packet switching. Burst switching is a circuit switching-based approach which allocates channel bandwidth to a connection only during the transmission of "bursts" of information. Fast packet switching is a packet switching-based approach which can be characterized by very high transmission rates on network links and simple, hardwired protocols which match the rapid channel speed of the network. Both approaches are being proposed as possible implementations for integrated-service networks. We survey these two approaches, and also examine the key performance issues found in fast packet switching. We then present the results of a simulation study of a fast packet switching network

Location management and multimedia communication service based on mobile IP and cellular IP network

[[abstract]]Wireless communication that provides voice only is not sufficient to support the necessity of user. It is an important feature of future wireless communication to offer this capability through mobile Internet. Mobile IP allows mobile hosts to change their location and reduce the losing probability of data packets in wireless communication networks. However, Mobile IP still have some defects in handoff and route aspects. Therefore, Cellular IP protocol is proposed for routing of IP diagrams to mobile stations and fast handoff control in a limited geographical area. In this paper, a handoff method is proposed to improve Quality of Service and resource switching management to reduce data packet loss for mobile multimedia communication in hierarchical network. In the future, all-IP network and mobile multimedia communication are two important characteristics, so that IP macromobility and micromobility network architecture are combined for data packets transfer. A Soft-handoff method is also presented to improve Quality of Service (QoS) and resource switching management to reduce data packet loss.[[notice]]補正完畢[[notice]]補正完畢[[conferencetype]]國際[[conferencedate]]20030327~20030329[[iscallforpapers]]Y[[conferencelocation]]Xi'an, Chin

The Contributory Effect of Latency on the Quality of Voice Transmitted over the Internet

Deployment of Voice over Internet Protocol (VoIP) is rapidly growing worldwide due to the new services it provides and cost savings derived from using a converged IP network. However, voice quality is affected by bandwidth, delay, latency, jitter, packet loss e.t.c. Latency is the dominant factor that degrades quality of voice transfer. There is therefore strong need for a study on the effect of Latency with the view to improving Quality of Voice (QoV) in VoIP network. In this work, Poisson probability theorem, Markov Chain, Probability distribution theorems and Network performance metric were used to study the effect of latency on QoS in VoIP network. This is achieved by considering the effect of latency resulting from several components between two points in multiple networks. The NetQoS Latency Calculator, Net-Cracker Professional® for Modeling and Matlab/Simulink® for simulating network were tools used and the results obtained compare favourably well with theoretical facts

Comparing the Efficiency of IP and ATM Telephony

Circuit switching, suited to providing real-time services due to the low and fixed switching delay, is not cost effective for building integrated services networks bursty data traffic because it is based on static allocation of resources which is not efficient with bursty data traffic. Moreover, since current circuit switching technologies handle flows at rates which are integer multiples of 64 kb/s, low bit rate voice encoding cannot be taken advantage of without aggregating multiple phone calls on a single channel. This work explores the real-time efficiency of IP telephony, i.e. the volume of voice traffic with deterministically guaranteed quality related to the amount of network resources used. IP and ATM are taken into consideration as packet switching technology for carrying compressed voice and it is compared to circuit switching carrying PCM (64 Kb/s) encoded voice. ADPCM32 is the voice encoding scheme used throughout most of the paper. The impact of several network parameters, among which the number of hops traversed by a call, on the real-time efficiency is studie

Signalling in voice over IP Networks

Voice signalling protocols have evolved, keeping with the prevalent move from circuit to packet switched networks. Standardization bodies have provided solutions for carrying voice traffic over packet networks while the main manufacturers are already providing products in workgroup, enterprise, or operator portfolio. This trend will accrue in next years due to the evolution of UMTS mobile networks to an “all-IP” environment. In this paper we present the various architectures that are proposed for signalling in VoIP, mainly: H.323, SIP and MGCP. We also include a brief summary about signalling in classical telephone networks and, at the end, we give some ideas about the proposed “all-IP” architectures in UMTS 3G mobile networks.Publicad