692 research outputs found
A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications
This article surveys the various techniques adopted for optimising bandwidth
for VoIP services over the period 1999-2014. The improvement of bandwidth can
be realized through; silence suppression measure of repressing the silent
portions (packets) in a voice conversation using Voice Activity Detection
algorithm; by so doing, the transmission rate during the inactive periods of
speech is reduced, and thus, the mean transmission rate can be reduced. A
second measure is packet header reduction which defines a process of
multiplexing and de-multiplexing packet headers to curb excesses. Voice/ Packet
Header compression is considered the most productive of all the techniques,
offering a scheme where VoIP packets are compressed from the 40 bytes of size
to a smaller byte size of 2 bytes. When combined with aggregation, compression
potentially yields a compressed size of up to 1 byte. In either case, bandwidth
save is reached using compression and decompression codecs of varying data and
bit rates. It is envisaged that an improvement in the performance of codecs
would yield a better result in terms of enhancing results favourably in Voice
over broadband networksComment: 8 pages, 7 figures. ISSN (Print): 1694-0814 | ISSN (Online):
1694-078
Enhancement of perceived quality of service for voice over internet protocol systems
Voice over Internet Protocol (WIP) applications are becoming more and more popular in
the telecommunication market. Packet switched V61P systems have many technical advantages
over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible
use of the bandwidth, lower cost and enhanced security.
However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed
in the VoIP services. In fact, most current Vol]P services can not provide as good a voice
quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived
speech quality as do application layer impairment factors, such as codec rate and audio features.
Current perceived Quality of Service (QoS) methods are mainly designed to be used
in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a
challenge to measure perceived speech quality correctly in V61P system and to enhance user
perceived speech quality for VoIP system.
The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality
measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless
systems in the context of V61P, and to develop novel and efficient methods to enhance the user
perceived speech quality for emerging V61P services especially in mobile V61P environment.
The main contributions of the thesis are threefold:
(1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation
of PESQ performance in mobile VoIP environment was undertaken and included setting up a
PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto-
PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was
investigated and main problems causing inaccurate PESQ score (improper time-alignment in
the PESQ algorithm) were discovered
.
Calibration issues for a safe and proper PESQ testing
in mobile environment were also discussed in the thesis.
(2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented
in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters
the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end
delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms
to provide improved performance. Results show that the proposed algorithm can increase user
perceived quality without consuming too much processing power when tested in live wireless
VbIP networks.
(3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive
codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority
for the beginning of a voiced segment). The results gathered on a simulation and emulation test
platform shows that the combined method provides a better user perceived speech quality than
separate adaptive sender bit rate or packet priority marking methods
Packet aggregation for voice over internet protocol on wireless mesh networks
>Magister Scientiae - MScThis thesis validates that packet aggregation is a viable technique to increase call ca-pacity for Voice over Internet Protocol over wireless mesh networks. Wireless mesh networks are attractive ways to provide voice services to rural communities. Due to the ad-hoc routing nature of mesh networks, packet loss and delay can reduce voice quality.Even on non-mesh networks, voice quality is reduced by high overhead, associated with the transmission of multiple small packets. Packet aggregation techniques are proven to increase VoIP performance and thus can be deployed in wireless mesh networks. Kernel level packet aggregation was initially implemented and tested on a small mesh network of PCs running Linux, and standard baseline vs. aggregation tests were conducted with a realistic voice tra c pro le in hop-to-hop mode. Modi cations of the kernel were then transferred to either end of a nine node 'mesh potato' network and those tests were conducted with only the end nodes modi ed to perform aggregation duties. Packet ag-
gregation increased call capacity expectedly, while quality of service was maintained in both instances, and hop-to-hop aggregation outperformed the end-to-end con guration. However, implementing hop-to-hop in a scalable fashion is prohibitive, due to the extensive kernel level debugging that must be done to achieve the call capacity increase.Therefore, end-to-end call capacity increase is an acceptable compromise for eventual scalable deployment of voice over wireless mesh networks
Voice and Video Transmission with Mobile IPv6
Mobile IPv6 (MIPv6) is a protocol that is proposed for the future of the mobile Internet access. The aim of MIPv6 is to provide seamless communication services to mobile nodes. The aim of this study is to investigate the effect of real time applications: voice and video transmission on MIPv6 network. In this paper the implementation of MIPv6 and fast handover MIPv6 (FMIPv6) is modeled and simulated using Network Simulator 2 (NS-2) software. The performance is analyzed for three different voice coding schemes and video based on H.263 format for both MIPv6 and FMIPv6
Performance and Analysis of Transfer Control Protocol Over Voice Over Wireless Local Area Network
A thesis presented to the faculty of the College of Science and Technology at Morehead State University in partial fulfillment of the requirements for the Degree Master of Science by Rajendra Patil in August of 2008
Quality aspects of Internet telephony
Internet telephony has had a tremendous impact on how people communicate.
Many now maintain contact using some form of Internet telephony.
Therefore the motivation for this work has been to address the quality aspects
of real-world Internet telephony for both fixed and wireless telecommunication.
The focus has been on the quality aspects of voice communication,
since poor quality leads often to user dissatisfaction. The scope of the work
has been broad in order to address the main factors within IP-based voice
communication.
The first four chapters of this dissertation constitute the background
material. The first chapter outlines where Internet telephony is deployed
today. It also motivates the topics and techniques used in this research.
The second chapter provides the background on Internet telephony including
signalling, speech coding and voice Internetworking. The third chapter
focuses solely on quality measures for packetised voice systems and finally
the fourth chapter is devoted to the history of voice research.
The appendix of this dissertation constitutes the research contributions.
It includes an examination of the access network, focusing on how calls are
multiplexed in wired and wireless systems. Subsequently in the wireless
case, we consider how to handover calls from 802.11 networks to the cellular
infrastructure. We then consider the Internet backbone where most of our
work is devoted to measurements specifically for Internet telephony. The
applications of these measurements have been estimating telephony arrival
processes, measuring call quality, and quantifying the trend in Internet telephony
quality over several years. We also consider the end systems, since
they are responsible for reconstructing a voice stream given loss and delay
constraints. Finally we estimate voice quality using the ITU proposal PESQ
and the packet loss process.
The main contribution of this work is a systematic examination of Internet
telephony. We describe several methods to enable adaptable solutions
for maintaining consistent voice quality. We have also found that relatively
small technical changes can lead to substantial user quality improvements.
A second contribution of this work is a suite of software tools designed to
ascertain voice quality in IP networks. Some of these tools are in use within
commercial systems today
IMPROVING QoS OF VoWLAN VIA CROSS-LAYER BASED ADAPTIVE APPROACH
Voice over Internet Protocol (VoIP) is a technology that allows the transmission of
voice packets over Internet Protocol (IP). Recently, the integration of VoIP and
Wireless Local Area Network (WLAN), and known as Voice over WLAN
(VoWLAN), has become popular driven by the mobility requirements ofusers, as
well as by factor of its tangible cost effectiveness. However, WLAN network
architecture was primarily designed to support the transmission of data, and not for
voice traffic, which makes it lack ofproviding the stringent Quality ofService (QoS)
for VoIP applications. On the other hand, WLAN operates based on IEEE 802.11
standards that support Link Adaptive (LA) technique. However, LA leads to having a
network with multi-rate transmissions that causes network bandwidth variation, which
hence degrades the voice quality. Therefore, it is important to develop an algorithm
that would be able to overcome the negative effect of the multi-rate issue on VoIP
quality. Hence, the main goal ofthis research work is to develop an agent that utilizes
IP protocols by applying a Cross-Layering approach to eliminate the above-mentioned
negative effect. This could be expected from the interaction between Medium Access
Control (MAC) layer and Application layer, where the proposed agent adapts the
voice packet size at the Application layer according to the change of MAC
transmission data rate to avoid network congestion from happening. The agent also
monitors the quality of conversations from the periodically generated Real Time
Control Protocol (RTCP) reports. If voice quality degradation is detected, then the
agent performs further rate adaptation to improve the quality. The agent performance
has been evaluated by carrying out an extensive series ofsimulation using OPNET
Modeler. The obtained results of different performance parameters are presented,
comparing the performance ofVoWLAN that used the proposed agent to that ofthe
standard network without agent. The results ofall measured quality parameters hav
Analysing the characteristics of VoIP traffic
In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration
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