Voice Over Internet Protocol Quality Of Service Research Articles

Improving QoS of IPTV and VoIP over IEEE 802.11n

Tremendous growth rates of Internet Protocol Television (IPTV) and Voice over Internet Protocol (VoIP) have demanded the shift of paradigm from wired to wireless applications. Increased packet loss with continuously varying wireless conditions make the transmission a challenging task in wireless environment. Our study investigates and proposes improvement in the transmission of combined IPTV and VoIP over the IEEE 802.11n WLAN. Our major contributions include the analytical and experimental investigations of (1) transport layer protocol UDP/TFRC for IPTV and VoIP, (2) optimal physical layer parameters for IPTV and VoIP, (3) proposition of wireless enhancement of TFMCC (W-TFMCC) to enhance the capacity and Quality of Service (QoS) of wireless IPTV and VoIP. Analytical and experimental evaluations show a 25% increase in capacity using TFRC with 167% more bandwidth share to TCP. Our study shows that use of W-TFMCC with optimal parameters can enhance IPTV and VoIP capacity by 44%.

The Affects of Different Queuing Algorithms within the Router on QoS VoIP application Using OPNET

Voice over Internet Protocol (VoIP) is a service from the internet services that allows users to communicate with each other. Quality of Service (QoS) is very sensitive to delay so that VOIP needs it. The objective of this research is study the effect of different queuing algorithms within the router on VoIP QoS. In this work, simulation tool “OPNET Modeler version 14.0” is used to implement the proposed network (VoIP Network). The proposed network is a private network for a company that has two locations located at two different countries around the world in order to simulate the communications within the same location as a local and the communications between two locations as a long distance and analyze VoIP QoS through measuring the major factors that affect the QoS for VoIP according to international telecommunication union (ITU) standards such as: delay, jitter and packet loss. In this research , a comparison was carried out between different queue algorithms like First in First out (FIFO), Priority queue (PQ) and Weight Fair Queuing (WFQ) and it was found that PQ and WFQ algorithms are the most appropriate to improve VoIP QoS .

QoS-aware path switching for VoIP traffic using SCTP

Voice over Internet protocol (VoIP) has been a prevalent multimedia service nowadays. It allows us to transmit voice data over IP networks. However, quality of service (QoS) is a major challenge to VoIP services. It must provide similar quality to traditional public switched telephone network or cellular phone services. Therefore, QoS related protocols have become important for real-time applications. Multi-protocol label switch (MPLS) is one of the important techniques to improve the network performance from QoS point of view. It employs label swapping to speed up packet forwarding. However, when a large number of users utilize VoIP services, the network congestion issue still exists. It causes delay, jitter and packet loss that affect VoIP QoS. In this paper, we propose a QoS-aware path switching strategy by using stream control transmission protocol (SCTP) in MPLS network to improve the VoIP traffic. This was done by employing SCTP selective acknowledgment mechanism to report the transmission parameters of primary path and to determine the criteria to switch to backup path. Simulation results show significant improvement in VoIP QoS.

Performance Analysis of Various Codecs using RSVP on VoIP Quality of Service over Variable Bandwidth

With the rapid growth of the Internet, customers are demanding multimedia applications such as telephony and video on demand, to be available on the internet. Voice over Internet Protocol (VoIP) is a rapidly emerging technology for voice communication that uses the ubiquity of IP-based networks. The greatest challenge is the provisioning of Quality of Service (QoS) over limited bandwidth. Multimedia applications need a lot more bandwidth and have different QoS requirements than the applications that were used in early years of the internet. The purpose of this paper is to check the performance of VoIP application under different Codec’s such as G.711, G.729 and G.723.1 over variable bandwidth. This paper proposes RSVP protocol for providing end to end QoS for real time applications over diverse networks. The performance is checked under RSVP and Non RSVP. Packet end to end delay and packet delay variation are evaluated through simulation. The result shows that under any bandwidth G.723.1 gives better result as compared to G.711 and G.729. Simulation is done using OPNET IT Guru Academic edition.

Enhanced Distributed Coordination Function of MAC for VoIP Services Using IEEE802.11 Networks

Voice over Internet Protocol (VoIP) is an important service with strict Quality-of-Service (QoS) requirements with in wireless local area networks. The popular Distributed Coordination Function (DCF) of IEEE802.11 Medium Access Control (MAC) protocol adopts Multiplicative Increase and linear Decrease procedure to reduce the packet collision probability in WLANs. In DCF, the size of contention window is doubled upon a collision regardless of the network loads. This paper presents an enhanced DCF scheme to improve the QoS of VoIP in WLANs. This scheme applies a threshold of the collision rate to switch between two different functions for increasing the size of contention window based on the status of network loads. The performance of this scheme investigated and compared to the original DCF using the network simulator NS-2. Under the high traffic loads the packet loss probability decreases with the enhanced DCF compared to the original DCF. Some other parameters like throughput and access delay is decreased with the enhanced DCF.

NGN 기반환경 에서의 VoIP QoS 관리체계 모델 설계

QoS(Quality of Service) is defined as collective effect of performance which determines the degree of satisfaction of a user of the service by ITU-T Rec. E.800. While the use of VoIP(Voice Over Internet Protocol) has been widely implemented, persistent problems with QoS are a very important sue which needs to be solved. This research is finding the assignment of VoIP QoS to deduct how to manage the control system and presenting the QoS control process and framework under NGN(Next Generation Network) environment. The trial framework is the modeling of the QoS measurement metrics, instrument, equipment, method of measurement, the series of cycle & the methodology about analysis of the result of measurement. This research underlines that the vulnerability of the VoIP protocol in relation to its QoS can be guaranteed when the product quality and management are controlled and measured systematically. Especially it`s very important time to maintain the research about VoIP QoS measurement and control because the big conversion of new network technology paradigm is now spreading. In addition, when the proposed method is applied, it can reduce an overall delay and can contribute to improved quality, in relation to signal, voice processing, filtering more effectively.

QoS-guaranteed burst transmission for VoIP service over optical burst switching networks

We propose a burst transmission method that guarantees the voice over Internet protocol (VoIP) service. The proposed method consists of three techniques: round-robin burst assembly with slotted scheduling, priority control with void filling, and hop-based preemption. Each technique is utilized so that the burst loss probability and the burst transmission delay satisfy VoIP quality of service (QoS). We evaluate by simulation the performance of the proposed method in NSFNET with 14 nodes. Numerical examples show that our proposed method is effective for guaranteeing the VoIP QoS while accommodating a large number of VoIP users.