Search results for "Voice over IP"
showing 10 items of 25 documents
Congestion Avoidance Using DYnamic COdec MAnagement: A solution for ISP
2005
The large diffusion of VoIP is an essential key for the success of emerging Internet service providers. These ISPs conflict with the interests of historical and predominant network maintainers which often control the network infrastructure and telephone services. To win this competition, emerging ISPs have to adopt strategic QoS solutions which will capture the attention of network users and phone clients. This paper presents a solution which is able to obtain the same performance of a 64 Kbps channel reducing the costs for over-provisioning network bandwidth. DyCoMa is a mechanism studied for VoIP applications in network with a limited bandwidth for multimedia services. It works preventing…
VoiP performance analysis in IEEE802.16 networks
2012
WiMAX, as known as IEEE standard 802.16, is a wide range broadband wireless access network which has a significant good support for the quality of service. According to IEEE standard 802.16e WiMAX has support also for mobility. One of the key advantages of the WiMAX network is the possibility to assign QoS parameters as connection based. A good example of traffic type having strict QoS demands is VoIP. VoIP will probably be a killer application in the future's broadband wireless networks because of its cost efficiency compared to circuit switched networks. In this paper, we analyze by extensive simulations how QoS is applied per connection, especially for the VoIP connection.
PCP: An End-to end Measurement Based Call Admission Control for Real-Time Services Over IP Networks
2000
Distributed end-to-end measurement based connection admission control mechanisms have been recently proposed. The goal of these schemes is to provide tight QoScon trol on a per connection basis by means of measurements taken by the edge nodes and priority based forwarding procedure at internal nodes. Since the additional flows handling procedures are implemented at the border routers and the forwarding mechanisms are for flows aggregates only, the approach is fully scalable and compatible with the IETF Differentiated Service proposal. The aim of this paper is to propose specific schemes and to investigate the advantages and limits of the approach by analyzing the basic mechanisms and evalua…
Extended HSUPA coverage and enhanced battery saving opportunities with multiple TTI lengths
2010
3GPP has specified that terminals can be configured to use either 2 or 10 ms transmission time interval in high speed uplink packet access systems. The purpose of this paper is to evaluate the benefit of exploiting a mixture of both of the transmissions time intervals within a cell instead of only one. The study is quantified by means of studying the achievable coverage of voice over IP and possible battery saving benefits. The analysis is conducted with a system level simulator modeling network and terminal behavior in detail. The paper indicates that utilizing a mixture of both transmission time intervals can extend coverage whilst providing enhanced battery saving opportunities.
Increasing the VoIP Capacity through MAP Overhead Reduction in the IEEE 802.16 OFDMa Systems
2010
One of the main issues with supporting VoIP service over 802.16 networks is the signalling overhead caused by the downlink MAP messages due to frequent transmissions and small packets. To decrease the MAP overhead, the 802.16 standard proposes some mechanisms, such as the compressed MAP and sub-MAPs. In this paper, we show by means of extensive dynamic simulations that sub-MAPs can reduce dramatically the signalling overhead associated with VoIP traffic and significantly improve overall VoIP capacity. At the same time, since sub-MAPs are more sensitive to packet drops, they tend to increase the number of HARQ retransmissions in downlink and transmission delays in the uplink direction.
Adaptive Contention Resolution for VoIP Services in the IEEE 802.16 Networks
2007
In the IEEE 802.16 networks, a subscriber station can use the contention slots to send bandwidth requests to the base station. The contention resolution mechanism is controlled by the backoff start/end values and a number of the request transmission opportunities. These parameters are set by the base station and are announced to subscriber stations in the management messages. In the case of the VoIP services, it is critical that the contention resolution occurs within the specified time interval to meet the VoIP QoS requirements. Thus, it is the responsibility of the base station to set correct contention resolution parameters to ensure the QoS requirements. This paper presents analytical c…
Advanced voice and data solutions for evolution of cellular network system
2014
<title>Revenue-maximization-based adaptive WFQ</title>
2002
In the future Internet, di erent applications such as Voice over IP (VoIP) and Video-on-Demand (VoD) arise with di erent Quality of Service (QoS) parameters including e.g. guaranteed bandwidth, delay jitter, and latency. Different kinds of service classes (e.g. gold, silver, bronze) arise. The customers of di erent classes pay di erent prices to the service provider, who must share resources in a plausible way. In a router, packets are queued using a multi-queue system, where each queue corresponds to one service class. In this paper, an adaptive Weighted Fair Queue based algorithm for traAEc allocation is presented and studied. The weights in gradient type WFQ algorithm are adapted using r…
Endpoint Admission Control with Delay Variation Measurements for QoS in IP Networks.
2002
In this paper we describe a novel Endpoint Admission Control scheme (EAC) for IP telephony. EAC mechanisms are driven by independent measurements taken by the edge nodes on a flow of packets injected in the network to probe the source to destination path. Our scheme is characterized by two fundamental features. First, it does not rely on any additional procedure in internal network routers other than the capability to apply different service priorities to probing and data packets. Second, the connection admission decision is based on the analysis of the probing flow delay variation statistics. Simulation results, which focus on a IP telephony scenario, show that, despite the lack of core ro…
PCP-DV: An End-to end Admission Control Mechanism for IP Telephony
2001
In this paper we describe a novel endpoint admission control mechanism for IP telephony:the PCP-DV which is characterized by two fundamental features. First, it does not rely on any additional procedure in internal network routers other than the capability to apply different service priority to probing and data packets. Second, the triggering mechanism for the connection admission decision is based on the analysis of the delay variation statistics over the probing flow. Numerical results for an IP telephony traffic scenario prove that 99th delay percentiles not greater than few ms per router are guaranteed even in overload conditions.