VoIP Services Research Articles

Implementing an Open Internet: Republic of Korea and U.S. Perspectives and Motivations

The Open Internet issue while not as hotly topical as a few years back continues to be a fundamental part of the policy debate here in the United States. Interestingly, in other countries, such as the Republic of Korea, this debate is very active, because of some fundamentally different implementation choices. In this paper we will explore these choices and the motivation behind them. In their Open Internet Order, the FCC adopted three basic rules in which broadband providers: must disclose network management practices; may not block lawful content/applications/services/devices; and may not unreasonably discriminate against lawful traffic. They further stated that mobile broadband providers may not block websites or applications that compete with their voice or video telephony services. Since the adoption of this Order, there has been littleIn 2011, the Korea Communications Commission (KCC) developed their open Internet guidelines; however these guidelines proved insufficient to cope with the complex issues that arose when Samsung Electronics launched Smart TV and Kakao Talk launched Voice Talk (a free mobile VoIP service). One of the major mobile carriers, Korea Telecom (KT), decided to block Smart TV, and announced that it would block Voice Talk service unless subscribers or service providers pay extra. While the KCC initially took action on KT not to block Smart TV, in June 2012 it came out with a decision allowing operators to be able to restrict access to these mVoIP services for a rational traffic management, provoking an immediate backlash from subscribers and advocates.After hearing divergent opinions through several policy debates, in July 2012, KCC published a tentative document, ‘The Guidelines for Reasonable Traffic Management & Use in the Internet,’ which gave specification to the prior basic guideline of 2011. Of note, these guidelines focus on implementation of allowable traffic management. Here, KCC allowed the mobile carriers (KT, SKT, LGU ) to charge a premium for voice service by discriminating subscribers based on the level of subscription rate, which the FCC disallows. Interestingly, the smallest mobile carrier, LGU , stated that it would allow its subscribers to access the Kakao Talk service as part of their existing data subscription plans – with no usage limitations and no additional charges – a stance that riled the larger carriers.In this paper, we will describe the contrasting approaches and then consider what motivated these differences and the consequence they might have on the overall openness and operation of the Internet. In our initial analysis, we see that the differences between these country’s approaches are driven by differing: 1) attitudes concerning Internet liberalism 2) user expectations, 3) network histories, 4) relationships between the regulator and the carriers, 4) maturity of the application industry, 6) and competitive environments in the market. We plan to expand on this analysis through a rigorous review of the legal records and scholarly work in this area. It is our hope that this work will help international efforts to harmonize of a common understanding of Internet Openness.

Traffic Analysis over a VoIP Server

This paper describes a traffic study over a VoIP server. The aim is to perform a study to know the traffic of communications of the current system installed and so what is the bandwidth needed to replace the existing traditional phone system for a new IP telephony system. After this study we used a simulator to know which is the maximum traffic supported by the VoIP server, to know if the system can support all users connected. Finally there will be a study to know what the bandwidth needed to establish a link between two VoIP servers located at two sites separated by a distance of 3 Km.

블랙홀 공격이 있는 MANET에서 패킷취합전송의 종단간 성능

본 논문에서는 블랙홀 공격이 발생되는 MANET에서 패킷취합전송의 종단간 성능을 여러 종류의 VoIP 트래픽을 대상으로 측정하고 분석하여 보았다. NS-2를 기반으로 구성한 VoIP 시뮬레이터를 이용하여 MOS, 호 연결율, 네트워크지연 및 패킷손실율 종단간 전송성능으로 측정하였다. 측정 결과의 분석을 통하여 블랙홀 공격이 있는 MANET에서 패킷취합전송의 특성을 여러 VoIP 트래픽 별로 제시하였다. 블랙홀 공격이 있는 MANET 환경에서 VoIP 서비스 구현에 요구되는 패킷취합전송의 특성을 본 논문의 결과로 제시하였다. In this paper, end-to-end performances for some VoIP traffics are measured and analyzed over MANET(Mobile Ad-hoc Networks) with packet aggregation transmission under blackhole attacks. A VoIP simulator based on NS-2 is used for performance measure. In the simulation, MOS, call connection ratio, network delay and packet loss rate are measured for end-to-end transmission performance. With analysis of measured data, characteristics of each traffics for performance parameters of packet aggregation transmission is showed on MANET under blackhole attack, As a results of this paper, some considerations be required for packet aggregation transmission are suggested to implement VoIP services on MANET under blackhole attacks.

Performance Analysis and Improvement on SIP-Based VoIP Server

The performance improvement on SIP-based VoIP server is given in this paper. The signaling server “SIP Proxy Server” and media streaming server “Media Relay Server” are applied. Firstly, the performance of VoIP server is analyzed and evaluated in detail. Then the effective method is applied to improve the performance of VoIP server. In the analysis results of SIP Proxy Server, the CPU time for the registration and call-setup sessions need 3.241 ms and 7.985 ms respectively. Moreover, in the Media Relay Server system, when the packet size from 1 to 32 bytes, the maximum throughput of packet is about 82,000 and 16,000 per second for UDP and TCP respectively. The large the packet, the less throughput is tenable. There are three methods applied to improve the performance. The capacity by 31 times is achieved. The capacity with 3,524 calls for each Media Relay Server is achieved. The improved method proposed in this paper is reasonable.

Secure Communication and VoIP Threats in Next Generation Networks

VoIP services are among key issues in the Next Generation Network (NGN) for the telecommunication domain. This technology is comprised of positive and negative aspects like similar emerging technologies. Nowadays, telecom operators provide call waiting, conference calling, call transfer, caller ID and other VoIP services based on NGN and all IP solution. Thus, VoIP security is now one of the issues over which many debates take place. In this paper we have decided to concentrate on vulnerability categories for VoIP services in NGN. We try to present the common security threats and vulnerabilities of VoIP services. Determining the vulnerabilities and their classification, together with the risks that this system is threatened by clarifies the ways to penetrate this system and enables administrators to solve the problems. The debate on this matter would be misleading without having full recognition of various specifications. In addition, to make the importance of identifying these types of vulnerabilities clear, we will classify the probability threats for VoIP.

Obtaining More Realistic Cross-Layer QoS Measurements: A VoIP over LTE Use Case

We introduce a real-time experimentation testbed in this paper which enables more realistic analysis of quality of service (QoS) in LTE networks. This testbed is envisioned for the improvement of QoS and quality of experience (QoE) through the experimentation with real devices, services, and radio configurations. Radio configurations suggested in the literature typically arise from simulations; the testbed provides a real and controlled testing environment where such configurations can be validated. The added value of this testbed goes a long way not only in the provision of more realistic results but also in the provision of QoS and QoE cross-layer measurements through the correlation of information collected at different layers: from service and IP levels to radio and protocol parameters. Analyzing the interlayer dependencies will allow us to identify optimal settings for the radio access network and service parameters. This information can be used to suggest new cross-layer optimizations to further improve quality of experience of mobile subscribers. As a use case, we examine VoIP service over LTE, which is currently an open issue.

A novel LIPA scheme for LTE VoIP services with home eNBs

The need for wireless multimedia communication motivates the development of the 4G cellular wireless networks. To this end, the 4G standard, Long Term Evolution (LTE), is specified by the 3GPP. Especially, the local IP access (LIPA) function has been defined for a home base station (HeNB) to offload traffic from the core network as otherwise packets are always routed through the packet data network gateway which is located at the back-end side. Without the LIPA function, the packet routing in an LTE network is in a detour way, even if in the communication between two user equipments (UEs) attaching to the same HeNB. To implement the LIPA function, two solutions have been proposed in the literature. However, they still suffer from a few shortcomings. In this paper, we propose a novel solution, in which an additional bearer is responsible for local data packet detection and, once detected, is transformed into a data bearer for local data packet delivery. Qualitative analysis distinguishes our solution from the two previous solutions in several aspects. Simulation results show that our method outperforms the two solutions in terms of packet transmission delay and mean opinion score (MOS). An extension is proposed to provide local IP access for UEs attaching to different serving HeNBs which are yet in the same domain. Analytical results shows that applying the extension, the domain edge router can provide hundreds of VoIP connections with the local IP access service, which is ten times more than the two solutions in the literature can do.

SIP Protocol in the IMS Network and Simulation

IP Multimedia Subsystem IMS is the core technology of next generation networks can provide real-time IP multimedia services and VoIP services through IMS, but IMS is not perfect, of which there are still some problems. IMS in real-time business a higher demand for mobility support, voice and video data usually use real-time transport protocol RTP to transfer user data protocol over UDP, requiring low transmission delay, seamless switching, as well as higher bandwidth utilization. In this paper, the IMS-related issues, the IMS network protocol.The SIP protocol simulation. Simulation results show that more of the nodes, the network send and receive packets faster, the results of this paper is the study of the IMS network, the actual reference value

Turning Need into Demand for Wi-Fi Broadband Internet Access on Trains

it is the demand of every internet user to stay connected most of the time even if the user is on the move. The core focus of this paper is on internet connectivity with moving vehicles. Wireless internet access on trains is not a standardized technology yet, therefore, it has diverse and vendor specific architecture. The authors have presented the physical architecture of internet access on moving vehicles, specifically trains and the logical architecture to give a clear idea that how the service will work at each layer of TCP/IP protocol stack. In order to observe the data rates offered on board, and experience Wi-Fi internet access onboard, the authors chose Great North Eastern Railway (GNER) route from London Kings Cross station till Peterborough (UK). For the network data rate analysis the authors chose Bandwidth Monitor a freeware used to monitor network and Internet bandwidth. The authors observed the data rates on GNER Wi-Fi internet on basis of three different scenarios i.e., Basic Browsing, Live Radio(Voice),VOIP services and video streaming Consequently, suggestions for improving quality of experience and market penetrations plans to make this technology successful in terms of generating revenues for the operators are given.

무선 멀티 홉 네트워크에서 VoIP 서비스를 위한 효율적인 패킷 결합 기법

)간의 트레이드오프 관계를 비용 함수로 표현하였다 .Key Words : VoIP, Wireless Multi-hop Network, Energy, MOS, QoSABSTRACTThis paper proposes packet aggregation scheme for considering energy consumption of mobile devices on guaranteeing quality of voice. To our purpose, we analyze VoIP service on wireless multi-hop channel in terms of delay , packet loss and energy consumption when packet aggregation scheme is applied to VoIP service. Moreover, we induce a cost function with considering of tradeoff relation between quality of voice and energy consumption.

향상된 Multi Gray-Leveling을 통한 VoIP 스팸 탐지 기법

In this paper, we propose an improved Multi Gray-Leveling scheme which reduces the problems of the existing Multi Gray-Leveling scheme suggested as a way of prevention against call spam in VoIP environment. The existing scheme having two different time period distinguishes the possibility of call spam by checking the call interval, so that it prevents the spammer`s avoidance controlling the call interval. This is the strength of the existing one but it can misunderstand the normal user as a spammer due to taking long term time period. To solve this problem, this paper proposes the upgrade scheme which utilizes the receiver`s action pattern as well as the caller`s action pattern. It has such a good strength that can do gray leveling via the collected information in the database of VoIP service provider without user`s direct involvement. Hence it can be a very effective way of VoIP spam detection.

ICT Based Communication Channels Preferences towards Knowledge Sharing among Multicultural Students

Many researches conducted in knowledge sharing have been focusing on organisational context. Yet, not many studies exist that explore knowledge sharing in education institutions. This paper provides an insight of how attitude, environment, motivation indicators, and individual factors (culture, age, and gender) would help in determining the preferences of information and communication technology (ICT) based communication channels among the students in multicultural institutions. The study uses quantitative design approach where data is collected from students studying in International Islamic University Malaysia (IIUM) Gombak Campus. The participants consisted of students originating from the regions of Middle East, West Africa, Southeast Asia and Eastern Europe. Information was solicited by the use of a survey. The survey was administered to a total of 184 participants. The research model designed for the study is based on the Theory of Planned Behaviour and Social Cognitive Theory. Factors involved in this study are attitude, environment, motivation, and demography, which includes cultural background. Six different communication channels have been looked through this study, i.e., instant messaging chat, social network, email, e-bulletin board, VOIP service, Telephone, and SMS. The results of the study can be used as a guide for identifying appropriate ICT based communication channels to be provided to the students for their knowledge sharing activities.

The Effects of Legacy Pricing Regulation on Adoption of Broadband Service in the United States

We examine and document the effect of policy-relevant factors on broadband adoption. We focus on the role of the price of basic local service on broadband adoption, to determine whether and to what extent legacy telephone policies and associated pricing distortions interfere with the contemporary goal of increased broadband adoption. Legacy universal service policy suppressed basic local telephone rates for residential customers in some areas to below-cost levels, subsidized by a variety of other revenue sources, including switched access revenues, prices for non-basic features, and business rates. The economic theory supporting this policy was that lower prices for basic local service would encourage adoption of basic local service, encouraging universal connection to the public switched telephone network. Today, however, there is effectively universal subscription to some form of local telephone service, which may include wireless or wireline services that are provided at unregulated prices; and policy goals are transitioning away from promoting basic local service to promotion of universal broadband adoption.We hypothesize that subsidized prices for basic residential services depress broadband adoption decisions. There are two reasons that we anticipate such an effect. First, broadband service and stand-alone traditional voice service are economic substitutes insofar as consumers can use broadband for voice service by purchasing an over-the-top VoIP service. Second, some customers may purchase a stand-alone voice line only or primarily for dial-up Internet access, for which broadband service is a substitute. Hence, for both reasons, we would expect to find a positive relationship between the price of basic local service and broadband adoption, meaning that where basic local service prices are lower, broadband adoption is lower. Intuitively, the logic is that, for a given price of broadband, the higher is the price of stand-alone voice service, the lower is the price hurdle to adopting broadband instead. We test this theory using quarterly household-level data from the first quarter 2006 through the second-quarter 2010 for over 70,000 household observations. The data are from the nationwide TNS surveys and bill samples. Using a multinomial logit specification, we find that higher prices for basic local service are in fact associated with higher rates of broadband adoption, controlling for availability of broadband service, broadband expenditures, household income, age of the head of household, size of household, population density of the household’s location, race, and other factors. We find that the demographic variables have largely the expected effects consistent with the literature. We also find that once other demographic and price variables are controlled for, there is not a “rural” divide - that is, a household’s propensity to adopt broadband service is higher, all else equal, in less densely populated areas, controlling for income, broadband availability, and other factors. Our results regarding the pricing variables are consistent with the hypothesis that promoting legacy policy goals of universal voice service by holding retail prices of basic local service artificially low impedes the achievement of current policy goals of universal broadband adoption. We believe our results are important because, unlike the numerous demographic variables that are known to be correlated with broadband adoption, the price of basic local service is, in many areas, subject to regulatory control and is therefore a policy variable. Our results suggest that relieving restrictions on prices of basic local service would promote the federal policy of universal broadband adoption.

이종 사설망간에 VoIP 미디어의 도.감청 보안 강화를 위한 암호화 기법 설계

기존의 IP 망을 이용하여 음성 데이터 서비스를 제공하는 VoIP는 최근 많은 관심을 받고 있다. VoIP 서비스는 다양한 보안 취약성을 포함한다. VoIP 서비스의 주요 공격 유형으로는 도 감청, DoS 공격, 스팸, 서비스 오용 공격 등을 들 수 있다. 이 중에서 도 감청으로 인한 기밀 정보의 유출은 매우 중요한 문제로 다루어지고 있다. VoIP 미디어 정보에 대한 도 감청 방지를 위해 SRTP나 ZRTP와 같은 암호화 기법이 주로 사용되고 있다. 일반적으로 VoIP 서비스는 단일 사설망 내에서의 운용과 서로 다른 사설망간의 운용으로 구분할 수 있는데, SRTP나 ZRTP는 단일 사설망 내에서의 통신 시에는 VoIP 미디어 정보를 암호화 한다. 그러나 두 가지 방법 모두 서로 다른 사설망간의 통신 시에는 암호화를 수행하지 못하는 문제를 가지고 있다. 이러한 문제를 해결하기 위해 본 논문에서는 기존의 SRTP 프로토콜의 일부를 수정하여 서로 다른 사설망간의 통신 시에도 VoIP 미디어 정보에 대한 암호화를 수행할 수 있는 암호화 기법을 제안한다. VoIP provides voice data service using existing IP networks and has received much attention recently. VoIP service has a variety of security vulnerabilities. Types of main attacks on VoIP service are tapping/interception, DoS attacks, spam, misuse of service attacks and the like. Of these, confidential information leak because of tapping/interception has been considered as a critical problem. Encryption techniques, such as SRTP and ZRTP, are mostly used to prevent tap and intercept on VoIP media information. In general, VoIP service has two service scenarios. First, VoIP service operates within a single private network. Second, VoIP service operates between different private networks. Both SRTP and ZRTP for VoIP media information within a single private network can perform encryption. But they can not perform encryption between different private networks. In order to solve this problem, in this paper, we modify SRTP protocol. And then, we propose an encryption method that can perform encryption of VoIP media information between the different private networks.

The Mixed Blessing of a Deregulatory Endpoint for the Public Switched Telephone Network

Receiving authority to dismantle the wireline public switched telephone network (“PSTN”) will deliver a mixture of financial benefits and costs to incumbent carriers. Even if these carriers continue to provide basic telephone services via wireless facilities or the Internet, they will benefit from the likely substantial relaxation of common carriage duties, no longer having to serve as the carrier of last resort and having the opportunity to decide where and what services they will provide going forward. On the other hand, incumbent carriers may have underestimated the substantial financial and marketplace advantages they also will lose in the deregulatory process. Incumbent carriers often obscure or dismiss as insignificant the substantial privileges and benefits accruing from their status as telecommunications service providers. Common carrier responsibilities include duties to interconnect with other carriers, provide service on transparent and nondiscriminatory terms and offer some low margin services. But this legal status also guarantees wireline local exchange carriers access to billions of dollars in annual universal service funding, zero or low cost access to rights of way and radio spectrum, accelerated depreciation and other tax benefits, the ability to vertically integrate throughout the “food chain” of telecommunications services and dominant status in the administration of telephone numbers, standard setting and other policy issues. Incumbents will strive to capture deregulatory benefits while retaining the many benefits previously reserved for common carriers. This paper will identify the potential problems resulting from an FCC decision to grant authority for telecommunications service providers to discontinue PSTN services. The paper also will consider whether in the absence of common carrier duties, carriers providing telephone services, including Voice over the Internet Protocol (“VoIP”), voluntarily will agree to interconnect their networks. The paper will examine Internet peering and other types of network interconnection with an eye toward assessing whether a largely unregulated marketplace can ensure ubiquitous access to PSTN replacement services. The paper concludes that private carrier interconnection models and information service regulatory oversight may not solve all disputes, or foreclose price discrimination for functionally the same type of service. Recent Internet interconnection and television program carriage disputes involving major players such as Comcast, Level 3, Fox and Cablevision, point to the possibility of increasingly contentious negotiations that could result in balkanized telecommunications networks with reversed or reduced progress in achieving universal service goals. The paper also concludes that rural access to VoIP and other voice communications services could end up costing significantly more than what urban residents pay, an efficient, but politically risky outcome.

Performance Evaluation of audio codecs using VoIP traffic in Wireless LAN using RSVP

has become an interesting topic of research in both the internet and the telecommunication industry. The tremendous increase in popularity of VoIP services is a result of huge growth in broadband access. In wired as well as in wireless communication, VoIP is expected to completely replace the traditional telephony approaches. To provide a good quality speech through VoIP applications, certain QoS parameters must be analyzed. These QoS parameters help us to evaluate the performance of various networking protocols, voice encoding schemes, etc. Numbers of QoS techniques like IntServ, DiffServ and RSVP are adopted to ensure good quality in IP based networks. In this paper, our main contribution is to analyze and evaluate the performance of various voice encoding schemes using RSVP, in a VoIP based wireless LAN. The network model designed in this paper is based on OPNET IT GURU Academic Edition. Various scenarios for different voice encoding schemes using RSVP are setup in the OPNET simulation environment. Different parameters that indicate the QoS like throughput, end to end delay, delay variations, traffic send, traffic received, etc. are calculated and analyzed in WLAN. KeywordsWLANs, QoS, IntServ, DiffServ, RSVP.

NCTU‐VT: a freeware for wireless VoIP performance measurement

AbstractVoice over IP (VoIP) is a promising low‐cost voice communication over the wireless IP network. To provide satisfactory VoIP services, the Quality of Service (QoS) of the wireless network should be guaranteed. This paper proposes a VoIP performance measurement freeware called NCTU VoIP Testing Tool (NCTU‐VT). We compare NCTU‐VT with two commercial tools SmartVoIPQoS and IxChariot in terms of packet loss, latency, and Mean Opinion Score (MOS) of the VoIP sessions in Wi‐Fi network. Our study indicates that these three tools can accurately measure VoIP performance in Wi‐Fi environment. Copyright © 2010 John Wiley & Sons, Ltd.

Formal analysis for robust anti-SPIT protection using model checking

Anti-SPIT policies counter the SPam over Internet Telephony (SPIT) by distinguishing bots launching unsolicited bulks of VoIP calls from human beings. We propose an Anti-SPIT Policy Management mechanism (aSPM) that detects spam calls and prevents VoIP session establishment by the Session Initiation Protocol (SIP). The SPIN model checker is used to formally model and analyze the robustness of the aSPM mechanism in execution scenarios with parallel SIP sessions. In case of a possible design flaw, the model checker provides a trace of the caught unexpected behavior (counterexample), that can be used for the revision of the mechanism’s design. Our SPIN model is parameterized, based on measurements from experiments with VoIP users. Non-determinism plays a key role in representing all possible anti-SPIT policy decisions, in terms of the SIP messages that may be exchanged. The model checking results provide evidence for the timeliness of the parallel SIP sessions, the absence of deadlocks or livelocks, and the fairness for the VoIP service users. These findings ensure robust anti-SPIT protection, meaning that the aSPM mechanism operates as expected, despite the occurrence of random SPIT calls and communication error messages. To the best of our knowledge, this is the first analysis for exhaustively searching security policy flaws, due to complex interactions between anti-SPIT measures and the SIP protocol services.

Comparative Analysis of the Performance of Different Codecs in a Live VoIP Network using SIP Protocol

In this paper we analyzed the impact of different codecs on the quality of VoIP services. We analyzed the performance of VoIP networks to have made an assessment to what extent are fulfilled the expectations regarding the desired quality of service and the authors applied the following methodology: we selected according to certain criteria and the same codecs implemented in a VoIP network, then selected the appropriate traffic routes and measurements carried out on them, and finally at the end is based on measurement results obtained comparative analysis and the comparisons made packet loss parameter. In the paper are discussed the main problems regarding the quality of VoIP services: and the choice of coding techniques. Measurements were performed with common used codecs for VoIP: G.711, G.723 and G.729, and the variation values of packet size and number of calls. Also, measurements were performed with optical and wireless transmission system. Finally, the main objective of this study was to select an appropriate codec to get a high quality voice and tries to compromise between price/quality of service. In conclusion, G.711 is the preferred technique that provides high voice quality (MOS greater than 4) but with a flaw need high bandwidth, and G.723 provide relatively good voice quality (MOS between 3 and 4) with a much smaller bandwidth, which means it can handle multiple users. The G.729 has a high level of compression and proper technique for many users, but only when the voice quality is not taken into account. Ill. 11, bibl. 9, tabl. 1 (in English; abstracts in English and Lithuanian).DOI: http://dx.doi.org/10.5755/j01.eee.117.1.1050

Performance of LTE network for VoIP users

With the arrival of LTE standard, it is expected that the mobile voice services paradigm will shift from the circuit switched to fully packet switched mode supporting the VoIP services. VoIP services took quite a bit of time before they were accepted as the main stream telephony service in the fixed networks. To provide VoIP services over the LTE networks with appropriate QoS, it is necessary to analyse the performance of such services and optimise the network parameters. This paper analyses the performance of VoIP services on the LTE network using the FD and the SMP packet scheduling techniques. This work identifies and analyses the features of above LTE packet scheduling techniques to enhance the QoS of VoIP services. An OPNET-based simulation model is used to analyse the performance of VoIP services on the LTE network by incorporating G.711 and G.723 speech coders. The work also studied the performance of VoIP services in variable transmission channel conditions.